244 Commits

Author SHA1 Message Date
Siavash Sameni
8ceb6f45d5 fix(build): declare VARIANT in local script half (was remote-only)
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The VARIANT variable was set inside the REMOTE_SCRIPT heredoc for
naming artifacts during the cargo tauri build, but never declared
in the local half of the script where it's used to rename downloaded
files. Under `set -u` strict mode this aborted the local downloads
with "unbound variable: VARIANT" after a successful remote build.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 16:16:07 +04:00
Siavash Sameni
07873ea598 fix(linux-aec): fall back to 0.3 crate + apt lib (2.x bundled is broken)
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Switch the webrtc-audio-processing dep from the 2.x git source (bundled
mode) back to crates.io 0.3, and link against Debian's apt package
libwebrtc-audio-processing-dev (0.3-1+b1 on Bookworm). The 2.x path
fails because both the crates.io tarball and the upstream git main
branch of webrtc-audio-processing-sys 2.0.3 have a build.rs bug where
\`meson setup --reconfigure\` is passed unconditionally, panicking on
first-run empty build dirs with "Directory does not contain a valid
build tree". The 0.x line sidesteps bundled mode entirely by linking
the apt-provided library.

Trade-off: we get AEC2 (the older generation) instead of AEC3, but
it's the same algorithm family and is what PulseAudio's
module-echo-cancel and PipeWire's filter-chain use on current
Debian-family distros. Fine for shipping — we can revisit AEC3 once
the 2.x bundled build is fixed upstream.

API changes:
- 0.3's Processor::process_capture_frame and process_render_frame
  take &mut self, so wrap the module-level processor in a Mutex.
  Capture and playback threads each lock briefly (sub-ms per 10 ms
  frame); contention is minimal.
- Import NUM_SAMPLES_PER_FRAME from the crate directly instead of
  hardcoding 480, so the code tracks whatever sample rate the
  upstream C++ lib exposes (currently 48 kHz hardcoded -> 480).
- Helper fns drain_frames_through_apm / tee_render_samples / etc.
  take &Mutex<Processor> instead of &Processor.
- Use explicit EchoCancellationSuppressionLevel and
  NoiseSuppressionLevel imports rather than fully-qualified paths.

Dockerfile:
- Drop meson / ninja-build / python3 (only needed for bundled build).
- Add libwebrtc-audio-processing-dev for the system link path.
- Keep clang (may be needed by the bindgen step in some versions).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 16:06:56 +04:00
Siavash Sameni
cc00f7cace fix(linux-aec): try main branch of webrtc-audio-processing
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v2.0.3 bundled build hits 'Directory does not contain a valid build
tree' because the crate's build.rs uses `meson setup --reconfigure`
unconditionally, which fails on first run when the build dir doesn't
yet contain prior meson state. Try the main branch in case it's been
fixed post-release.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:58:28 +04:00
Siavash Sameni
eb9de988d6 fix(linux-aec): use git dep for webrtc-audio-processing
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The crates.io tarball of webrtc-audio-processing-sys 2.0.3 is missing
the vendored C++ submodule — the bundled build fails with 'Directory
does not contain a valid build tree' when meson tries to configure
the ./webrtc-audio-processing subdirectory. Cargo clones git deps with
submodules auto-initialized since ~1.27, so pulling from the upstream
git repo (pinned to tag v2.0.3) gives us the full source tree.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:55:04 +04:00
Siavash Sameni
4ba77c8c0e feat(linux): WebRTC AEC3 capture/playback backend with render-side tee
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Adds gold-standard Linux echo cancellation: in-app WebRTC AEC3 (Audio
Processing Module) via the webrtc-audio-processing crate, using the
same algorithm as Chrome WebRTC, Zoom, Teams, and Jitsi. Runs entirely
in-process, so it works identically on ALSA / PulseAudio / PipeWire
systems — no dependency on user-configured echo-cancel modules.

Architecture:
- New crates/wzp-client/src/audio_linux_aec.rs module (~470 lines).
  Contains LinuxAecCapture and LinuxAecPlayback, both using CPAL
  under the hood but routing samples through a shared
  Arc<webrtc_audio_processing::Processor>. The playback path tees
  each 20 ms frame into APM.process_render_frame as the echo
  reference BEFORE handing the samples to CPAL's output callback.
  The capture path runs APM.process_capture_frame on each mic frame
  in place before pushing to the audio ring buffer. This is the
  "tee the playback ring" approach that Zoom/Teams/Jitsi use.
- New `linux-aec` feature in wzp-client pulling in the
  webrtc-audio-processing crate at v2.x with the `bundled`
  sub-feature. Bundled means the vendored PulseAudio WebRTC C++
  sources are statically compiled via meson+ninja at cargo build
  time — no runtime .so dependency, avoids Debian Bookworm's stale
  libwebrtc-audio-processing-dev 0.3 package (which predates AEC3).
  Dep is target-gated to Linux, so enabling the feature on non-Linux
  is a no-op.
- lib.rs re-exports LinuxAecCapture/LinuxAecPlayback as
  AudioCapture/AudioPlayback when `linux-aec` is on, otherwise
  falls back to the CPAL audio_io path. Shared public API
  (start/ring/stop/Drop) means downstream code is unchanged.
- New `linux-aec` feature in wzp-desktop forwards to
  wzp-client/linux-aec so `cargo tauri build -- --features
  wzp-desktop/linux-aec` builds the AEC variant.

APM configuration:
- EchoCancellation: High suppression, delay-agnostic mode on,
  extended filter on, stream_delay_ms=60 initial hint
- NoiseSuppression: High
- HighPassFilter: on
- AGC: off (can fight Opus encoder's own gain staging + adaptive
  quality controller; add later if users report low mic level)

Frame size handling:
- Pipeline uses 20 ms frames (960 samples @ 48 kHz mono)
- APM requires strict 10 ms (480 samples) per call
- Each 20 ms frame is split into two 480-sample halves, APM called
  twice, halves stitched back
- Same pattern for render and capture sides
- Carry-buffer logic handles the case where CPAL delivers samples in
  arbitrary chunk sizes that don't divide 960

Build infrastructure:
- scripts/Dockerfile.linux-desktop-builder adds meson, ninja-build,
  python3, clang for the webrtc-audio-processing bundled build
- scripts/build-linux-desktop-docker.sh takes a new --aec flag that
  enables the linux-aec feature and renames the output artifacts
  with an `-aec` suffix so noAEC and AEC variants can coexist on disk

Task #30.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:53:23 +04:00
Siavash Sameni
7b8a2d0fba feat(build): add Linux x86_64 Tauri desktop build pipeline
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New Dockerfile and build script for producing wzp-desktop as a Linux
x86_64 binary (plus .deb and .AppImage bundles via tauri-cli).

- scripts/Dockerfile.linux-desktop-builder: thin extension of
  wzp-android-builder that adds the Tauri Linux runtime deps
  (libwebkit2gtk-4.1-dev, libsoup-3.0-dev, libgtk-3-dev,
  libayatana-appindicator3-dev, librsvg2-dev, libglib2.0-dev, patchelf).
  Everything else (Rust, Node, cmake, pkg-config, libasound2-dev,
  tauri-cli) is inherited from the base image.

- scripts/build-linux-desktop-docker.sh: mirrors the pattern of
  build-windows-docker.sh and build-linux-docker.sh. Ships
  \`cargo tauri build\` which produces target/release/wzp-desktop
  plus bundles under target/release/bundle/{deb,appimage}/. Uploads
  the .deb (or raw binary if bundling fails) to rustypaste and
  notifies ntfy.sh/wzp on start + completion. Downloads all three
  artifact types (raw binary, .deb, .AppImage) to target/linux-desktop/
  when they exist.

Image cache volumes are shared with the Android pipeline for cargo
registry + git, but the target dir is in its own cache-linux-desktop/
path to avoid stomping on the Android / Linux-CLI / Windows target
caches.

Branch default is feat/desktop-audio-rewrite (where the actual
wzp-desktop source lives), not feat/android-voip-client.

Task #29.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:28:47 +04:00
Siavash Sameni
5cd7a20152 fix(ui): disable WebView pinch-zoom and desktop right-click menu
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Two small WebView hardening tweaks that apply to both Android (Tauri
mobile) and desktop (Tauri) since the frontend is shared:

- index.html viewport meta now sets maximum-scale=1.0, minimum-scale=1.0,
  and user-scalable=no. This stops users on Android from pinch-zooming
  out of the fixed-layout UI. Desktop is unaffected because the Tauri
  WebView ignores pinch gestures anyway.
- main.ts installs global listeners that preventDefault on contextmenu
  (kills the browser-style right-click menu that exposed Inspect /
  Reload / Back / Forward entries on desktop), keydown Ctrl+-/+/0
  (stops keyboard zoom of the fixed layout), and gesture* + ctrl-wheel
  events (trackpad pinch on WebKit + Chromium respectively).

Dev tools remain accessible via F12 / Cmd-Opt-I keyboard shortcuts —
only the right-click entry point is suppressed. Android has no
right-click so that part is a no-op there.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:26:08 +04:00
Siavash Sameni
a5c00fe5cb docs: add BRANCH-desktop-audio-rewrite.md and update ARCH/ADMIN/USER_GUIDE
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Documents the feat/desktop-audio-rewrite branch story end-to-end:
- Purpose: shared codebase with android-rewrite via Tauri, platform-
  specific audio backends via target-dep sections + feature flags
- Audio backend matrix: CPAL baseline + macOS VPIO + Windows WASAPI
  AudioCategory_Communications
- Recent work: desktop direct calling feature with history dedup,
  macOS VPIO integration, Windows cross-compile via cargo-xwin, the
  libopus/clang-cl vendored audiopus_sys fix, icon.ico generation,
  and the WASAPI communications capture backend (task #24)
- Build pipelines: native cargo on macOS/Linux, Docker on SepehrHomeserverdk
  for Windows, Hetzner Cloud alternative
- Testing procedures for direct calling parity and Windows AEC A/B
- Known quirks: vendor path relative, cargo-xwin override.cmake clobber,
  WebView2 runtime prerequisite, 2024 edition unsafe lint warnings

Also appends shared-doc sections (identical on both branches):
- ARCHITECTURE.md: "Audio Backend Architecture (Platform Matrix)"
- ADMINISTRATION.md: "Build Pipelines"
- USER_GUIDE.md: "Direct 1:1 Calling" and "Windows AEC Variants"

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:20:21 +04:00
Siavash Sameni
ec41f179cd fix(windows): drop dead override.cmake patch from Dockerfile
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The RUN step that baked an OPUS_DISABLE_INTRINSICS patch into
cargo-xwin's override.cmake was inert from the start: cargo-xwin
rewrites that file from scratch on every \`cargo xwin build\` invocation
(src/compiler/clang_cl.rs line ~444 uses include_bytes! to overwrite
it), so anything baked at image build time gets wiped at runtime.

The libopus SSE4.1/SSSE3 compile failure is now fixed upstream at the
source level by the vendored audiopus_sys patch (see
vendor/audiopus_sys/opus/CMakeLists.txt and the MSVC_CL distinction
for clang-cl). Remove the dead RUN step and leave a breadcrumb
comment pointing at the real fix location.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 15:07:06 +04:00
Siavash Sameni
4e9244eb00 fix(windows): add Win32_Security feature + 2024 edition unsafe wrappers
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- CreateEventW is gated behind Win32_Security in the windows crate
  because its signature takes SECURITY_ATTRIBUTES; add to features.
- Remove unused HANDLE import.
- Wrap GetId() and PWSTR::to_string() in explicit unsafe { ... }
  blocks for Rust 2024 edition's unsafe_op_in_unsafe_fn lint.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 14:36:50 +04:00
Siavash Sameni
03a80a3196 feat(windows): WASAPI capture backend with OS-level AEC
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Adds a direct WASAPI microphone capture path for the Windows desktop
build that opens the default communications endpoint via
IMMDeviceEnumerator -> IAudioClient2 -> SetClientProperties with
AudioCategory_Communications, turning on Windows's communications
audio processing chain (AEC, noise suppression, automatic gain
control). The communications AEC operates at the OS level and uses
the system render mix as the reference signal, so echo from our
existing CPAL playback stream is cancelled automatically with no
per-process reference plumbing.

Architecture:
- New crates/wzp-client/src/audio_wasapi.rs module (~280 lines).
  Event-driven capture loop on a dedicated thread; pushes PCM into
  the same lock-free AudioRing used by the CPAL path. Same public
  API as audio_io::AudioCapture so downstream code is unchanged.
- New `windows-aec` feature in wzp-client that pulls in the
  `windows` crate (Microsoft's official Rust COM bindings) gated to
  target_os = "windows" only. Enabling the feature on non-Windows
  targets is a no-op since both the module and the dep are
  cfg(target_os = "windows").
- lib.rs re-exports WasapiAudioCapture as AudioCapture when the
  feature is on, otherwise falls back to the CPAL AudioCapture.
  AudioPlayback is always the CPAL one — no reason to swap it.
- desktop/src-tauri/Cargo.toml Windows target enables the new
  feature: `features = ["audio", "windows-aec"]`.

Implementation notes:
- Uses eCommunications role (not eConsole) for GetDefaultAudioEndpoint
  — the user-configured "communications" device that Teams/Zoom
  pick up, and the one Windows's AEC is tuned for.
- Requests 48 kHz mono i16 with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM +
  SRC_DEFAULT_QUALITY so Windows handles any format conversion in
  the audio engine instead of rejecting our format.
- Event-driven with SetEventHandle / WaitForSingleObject — no
  polling, minimal CPU cost between packets.
- 200 ms wait timeout so the capture thread polls `running` often
  enough for Drop to stop cleanly even if the audio engine stalls
  (e.g. device unplug).

Task #24.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 14:35:36 +04:00
Siavash Sameni
7fecf285ea fix(windows): add icons/icon.ico for tauri-build Windows resource
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tauri-build's Windows path unconditionally looks up icons/icon.ico to
embed as the PE file resource (taskbar/Explorer icon). We only had
icon.png (32x32 placeholder) which is fine on macOS/Linux but blocks
the Windows cross-compile with "icons/icon.ico not found; required for
generating a Windows Resource file during tauri-build".

Generated a multi-size ICO (16/24/32/48/64/128/256) from the existing
placeholder icon.png via Pillow. It's ugly at 256 due to upscaling from
32x32 with LANCZOS, but unblocks the build. Real branded icons can
replace it later without any build-system changes.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 14:15:04 +04:00
Siavash Sameni
0683dde5d3 fix(windows): vendor audiopus_sys + patch libopus for clang-cl SIMD
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cargo-xwin drives the Windows MSVC cross-compile via clang-cl, under
which CMake sets MSVC=1 — causing libopus 1.3.1's `if(NOT MSVC)` guards
to skip the per-file `-msse4.1` / `-mssse3` COMPILE_FLAGS that its x86
SIMD source files need. Clang-cl (unlike real cl.exe) still honors
Clang's target-feature system, so those files then fail to compile
with "always_inline function '_mm_cvtepi16_epi32' requires target
feature 'sse4.1'" errors across silk/NSQ_sse4_1.c, NSQ_del_dec_sse4_1.c,
and VQ_WMat_EC_sse4_1.c.

Earlier attempts to fix this downstream (cargo-xwin toolchain file,
override.cmake CMAKE_C_COMPILE_OBJECT <FLAGS> replace, CFLAGS env vars)
all failed because cargo-xwin rewrites override.cmake from scratch on
every `cargo xwin build` invocation and cmake-rs's -DCMAKE_C_FLAGS=
assembly happens before toolchain FORCE sets propagate.

Fixing it upstream at the source: vendor audiopus_sys 0.2.2 into
vendor/audiopus_sys, patch its bundled opus/CMakeLists.txt to introduce
an MSVC_CL var (true only when CMAKE_C_COMPILER_ID == "MSVC", i.e. real
cl.exe), and flip the eight `if(NOT MSVC)` SIMD guards to
`if(NOT MSVC_CL)`. Clang-cl then gets the GCC-style per-file flags and
the SSE4.1 sources build cleanly. Also flip the `if(MSVC)` global /arch
block at line 445 to `if(MSVC_CL)` so only cl.exe applies /arch:AVX and
clang-cl relies purely on per-file flags (no global/per-file mixing).

Wire via [patch.crates-io] in the workspace root Cargo.toml; the patch
is resolved relative to the workspace root as `vendor/audiopus_sys`.

Upstream context: xiph/opus#256, xiph/opus PR #257 (both stale).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-10 14:12:59 +04:00
Siavash Sameni
53f57eea07 fix(windows): printf instead of heredoc in Dockerfile RUN (parser hated <<EOF)
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2026-04-10 13:05:04 +04:00
Siavash Sameni
ff3f7e8e4f fix(windows): patch override.cmake not toolchain — inject SSE via COMPILE_OBJECT template
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The previous 'patch the toolchain file' approach (234a798, 48d2bd4) did
write the SSE flags into the COMPILE_FLAGS list correctly in the baked
image, but the CMakeCache.txt from the libopus configure ended up
without them in CMAKE_C_FLAGS, so cmake's final compile commands
didn't see them either. Most plausible explanation: cmake-rs passes
`-DCMAKE_C_FLAGS=…` on the command line, and its assembly of that
string happens outside the toolchain's FORCE set path, so the
toolchain patch never propagated.

Switch to a different lever: cargo-xwin already ships a tiny
`override.cmake` loaded via CMAKE_USER_MAKE_RULES_OVERRIDE. That
file is the right place to manipulate the compile-command
`CMAKE_C_COMPILE_OBJECT` / `CMAKE_CXX_COMPILE_OBJECT` templates —
it runs after cmake has initialised its compile rules but before
any source is compiled. Append two string(REPLACE '<FLAGS>' '<FLAGS>
/clang:-msse4.1 /clang:-mssse3 /clang:-msse3 /clang:-msse2') lines
to that file so every C and C++ compile command generated by cmake
gets the SSE feature flags inline, no matter what the project's
CMAKE_C_FLAGS is set to.

This is the CMake equivalent of a compiler wrapper and works
regardless of how cmake-rs / cargo-xwin / libopus juggle their
respective flag variables.
2026-04-10 13:03:06 +04:00
Siavash Sameni
48d2bd4f65 fix(windows): bake SSE patch into docker image instead of runtime
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2026-04-10 12:55:48 +04:00
Siavash Sameni
234a798df2 fix(windows): append SSE flags as a pure-CMake block to xwin toolchain
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The previous sed-based patch didn't stick in the docker-bash-c
heredoc (bash single-quoting made the newline escaping fragile).
Switch to a much simpler approach: just 'cat >>' a pure-CMake block
to the end of the cargo-xwin toolchain file. The block does:

    set(CMAKE_C_FLAGS   "${CMAKE_C_FLAGS}   /clang:-msse4.1 ..." CACHE STRING "" FORCE)
    set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} /clang:-msse4.1 ..." CACHE STRING "" FORCE)

Running AFTER the toolchain's own FORCE-set and AFTER cmake-rs's
-DCMAKE_C_FLAGS= command-line override, it unconditionally wins. No
sed, no awk, no python, no newline escaping — just CMake reading the
toolchain file like it normally does.

Idempotent via the WZP_SSE_PATCH sentinel grep in the comment block.
2026-04-10 12:50:00 +04:00
Siavash Sameni
fa042b130c fix(windows): sed-patch cargo-xwin toolchain to enable SSE4.1/SSSE3
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The CFLAGS_x86_64_pc_windows_msvc env-var approach from 990b6f1 did
nothing — cargo-xwin ships its own clang-cl cmake toolchain file at
~/.cache/cargo-xwin/cmake/clang-cl/x86_64-pc-windows-msvc-toolchain.cmake
which hardcodes COMPILE_FLAGS and FORCE-overrides CMAKE_C_FLAGS. Any
env-var CFLAGS gets dropped before opus's cmake build sees it.

The only place that actually makes it into every C file compilation
in the libopus subbuild is the toolchain file itself. Patch it in
place with an idempotent sed that appends

    /clang:-msse4.1
    /clang:-mssse3
    /clang:-msse3
    /clang:-msse2

right before the closing paren of the COMPILE_FLAGS setter. The patch
is marked with a WZP_SSE_PATCH comment so re-runs skip it.

Confirmed the error message matches with/without the env var — same
20 clang errors from NSQ_del_dec_sse4_1.c / NSQ_sse4_1.c before and
after 990b6f1, which is how we ruled out the env-var path.
2026-04-10 12:43:36 +04:00
Siavash Sameni
990b6f1ee0 fix(windows): set CFLAGS +sse4.1 +ssse3 so audiopus_sys builds under clang-cl
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libopus ships per-file SSE4.1 / SSSE3 C sources (opus/silk/x86/NSQ_del_dec_sse4_1.c
etc.) that assume the compiler picks up `-msse4.1` / `-mssse3` as per-file
CMake COMPILE_FLAGS. With clang-cl those bare -m flags are silently dropped,
so _mm_cvtepi16_epi32 + friends fail compile with 'always_inline function
requires target feature sse4.1, but would be inlined into a function that
is compiled without support for sse4.1'.

Workaround: set CFLAGS_x86_64_pc_windows_msvc + CXXFLAGS_x86_64_pc_windows_msvc
to `/clang:-msse4.1 /clang:-mssse3 /clang:-msse3 /clang:-msse2` before running
cargo xwin build. Every x86_64 Windows CPU shipped since 2008 has these
instruction sets so globally enabling them on this target is safe.

Also bump the tail -30 on cargo xwin output to tail -50 so the actual
compiler errors (not just the cmake wrapper panic) make it into the
ntfy / remote log file next time.
2026-04-10 12:40:38 +04:00
Siavash Sameni
7949266e11 windows: docker + hcloud build scripts for cross-compile
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Two parallel paths to build wzp-desktop.exe for x86_64-pc-windows-msvc:

scripts/Dockerfile.windows-builder
  Debian 12 base, matches scripts/Dockerfile.android-builder's layout:
  - apt: build-essential, cmake, ninja-build, llvm, clang, lld, nasm,
    libssl-dev, node 20 LTS
  - rust stable + x86_64-pc-windows-msvc target
  - cargo-xwin pre-installed
  - Pre-warmed ~/.cache/cargo-xwin layer: creates a throwaway cargo
    project and runs `cargo xwin build` once during image build so the
    MSVC CRT + Windows SDK (~1.5 GB) is baked into an image layer.
    Saves ~4 minutes off every cold cross-compile run.
  - Builder user uid 1000 to match existing bind-mount perms on
    SepehrHomeserverdk.

scripts/build-windows-docker.sh
  Same pattern as scripts/build-tauri-android.sh but for Windows:
  - Fires a remote build on SepehrHomeserverdk via ssh + heredoc
  - Mounts the shared cargo-registry + cargo-git cache + a
    target-windows dir (separate from the android target cache so
    different triples don't stomp each other)
  - Runs npm install + npm run build for the frontend dist, then
    cargo xwin build --release --target x86_64-pc-windows-msvc
    --bin wzp-desktop inside the container
  - Uploads the resulting .exe to rustypaste (via the .env token on
    the remote, same as android script) and fires ntfy.sh/wzp
    notifications at start + completion
  - scp's the .exe back to target/windows-exe/wzp-desktop.exe locally
  - --image-build flag triggers a fire-and-forget `docker build` of
    the Dockerfile.windows-builder on the remote (used once after the
    Dockerfile changes). The image is already built at the moment of
    this commit — sha256:f3895cb2fde7

scripts/build-windows-cloud.sh
  Kept as an alternative cross-compile path using a fresh Hetzner VM
  (cx33, 8 vCPU, 8 GB — bumped from cx23 after the smaller size OOM'd
  mid-rustc). The docker-on-SepehrHomeserverdk path is now the
  preferred fast path because the image has a pre-warmed xwin cache
  and a persistent cargo target volume, making warm builds ~3 minutes
  vs the cloud path's ~20 minutes cold each run. The cloud script
  stays around for when we want a truly isolated environment.

Both scripts notify via ntfy.sh/wzp and upload to paste.dk.manko.yoga
so the user can pick up the artefact + see status without polling.
2026-04-10 12:35:02 +04:00
Siavash Sameni
d774f5f8c5 feat(history): dedupe by call_id + explicit Incoming/Outgoing/Missed labels
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User reported that outgoing direct calls from macOS show up in the
history list as "missed" even when the call completes successfully.
Adds two changes to fix / diagnose:

1. history::log now dedupes by call_id. If an entry for this call_id
   already exists in the store, it updates the existing row's
   direction + timestamp in place instead of appending a duplicate.
   Protects against double-emit (caller side adding Missed on top of
   Placed, or any future signal loop that fires twice). One row per
   call_id, which matches what the user intuitively expects.

2. history::log now logs every write with tracing::info — call_id,
   peer_fp, direction, alias. Plus an extra line when we replace
   an existing entry: "history::log replacing existing entry
   from=Placed to=Missed" etc. Makes it easy to see in the desktop
   stderr which side is writing what, so we can find the outgoing =>
   missed regression immediately if it recurs.

3. main.ts now renders an explicit text label next to the direction
   arrow: "Outgoing", "Incoming", or "Missed" instead of just the ↗
   ↙ ✗ icons. Removes any ambiguity about what the icon means so
   future users can't misread a Placed entry as Missed based on icon
   shape alone.

Side fix for scripts/build-windows-cloud.sh:
- die() and the do_full ERR trap now respect WZP_KEEP_VM=1 so a failed
  build doesn't auto-destroy the debug VM (previously the trap fired
  before the KEEP_VM check and tore down the VM on any error).
- Bump default server type cx23 → cx33. 4GB RAM is not enough for a
  cold tauri + rustls + quinn + wzp-client cross-compile — the cx23
  run got "Read from remote host ... Connection reset by peer"
  partway through rustc, which is the classic signature of an OOM
  kill on the SSH session. cx33 has 8GB RAM and 8 vCPU which should
  comfortably fit the build.
2026-04-10 12:34:19 +04:00
Siavash Sameni
2fd94651e4 fix(desktop): direct calls used wrong identity file — mac identity mismatch
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The non-Android branch of CallEngine::start loaded the seed from
\$HOME/.wzp/identity directly, while register_signal in lib.rs goes
through the shared load_or_create_seed() helper which resolves via
APP_DATA_DIR → Tauri's app_data_dir(). On macOS those are two
completely different files:

  register_signal → ~/Library/Application Support/com.wzp.desktop/.wzp/identity
  CallEngine::start (old) → ~/.wzp/identity

On a fresh install they end up holding two different random seeds.
Register and CallEngine then derive two different fingerprints from
those seeds, and when a direct call comes in the relay routes it to
"you" under the register_signal fingerprint, but once CallEngine tries
to join the call-* room it advertises a DIFFERENT fingerprint — which
fails the call_registry ACL check on the relay side (only the two
authorised participants of a call can join its room). Silent hang, the
call never completes.

Android hit this bug earlier in the week and was fixed by switching
its CallEngine::start branch to `crate::load_or_create_seed()`.
Backport the same single-line change to the desktop branch so both
platforms share one identity source of truth.

Also bring the desktop branch up to parity with the android branch on
diagnostic logging:
- log CallEngine::start entry with relay/room/alias/quality/has_reuse
- log endpoint.local_addr on reuse / create
- log "QUIC connection established, performing handshake" between
  connect() and perform_handshake() so a hang at either step is
  immediately localisable
- map_err all three potential failure points (create_endpoint,
  connect, perform_handshake) to an explicit error! trace
2026-04-10 12:15:23 +04:00
Siavash Sameni
da09fdb6e9 windows(desktop): gate coreaudio / VoiceProcessingIO to macOS-only targets
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First step of the Windows x86_64 desktop build: stop pulling
coreaudio-rs into the Windows dependency graph so the project can at
least run `cargo check --target x86_64-pc-windows-msvc`. Software AEC
is already disabled in engine.rs so there's nothing else to stub — the
macOS-specific VPIO path is skipped via #[cfg(target_os = "macos")] on
both sides and Windows falls through to the plain CPAL
AudioCapture/AudioPlayback branch that already existed.

crates/wzp-client/Cargo.toml
  - coreaudio-rs optional dep moved under [target.'cfg(target_os = "macos")']
  - `vpio` feature now uses `dep:coreaudio-rs` syntax and the gated dep
  - Enabling `vpio` on Windows/Linux is a no-op at resolution time

crates/wzp-client/src/lib.rs
  - `pub mod audio_vpio` is now #[cfg(all(feature = "vpio", target_os = "macos"))]
  - Previously `vpio` alone was enough to try to compile the Core Audio
    bindings, which would fail on non-Apple targets the moment the
    feature flag was flipped on

desktop/src-tauri/Cargo.toml
  - [target.'cfg(not(target_os = "android"))'] removed — was leaking
    vpio into Windows/Linux via the catch-all.
  - macOS: wzp-client with features = ["audio", "vpio"]
  - Windows: wzp-client with features = ["audio"]
  - Linux: wzp-client with features = ["audio"]
  - Android: wzp-client with default-features = false (unchanged)
  - Dropped the unused direct coreaudio-rs = "0.11" dep on macOS —
    wzp-desktop's own sources never call Core Audio directly.

Verified via `cargo tree --target x86_64-pc-windows-msvc -p wzp-desktop`
that the Windows target now resolves wzp-client with cpal but without
coreaudio-rs. macOS target still resolves with coreaudio (direct via
vpio feature and transitively via cpal). macOS `cargo check` still
builds cleanly.

Cross-compile from macOS hit a cargo-xwin + llvm-lib setup issue in
ring's build.rs, so the actual `cargo check --target
x86_64-pc-windows-msvc` did not complete locally. Build verification
belongs on the user's Windows x86_64 host where MSVC is present
natively.

See tasks #23 (this one), #24 (Voice Capture DSP / WASAPI Communications
for OS-level AEC on Windows), and #25 (aarch64-pc-windows-msvc support).
2026-04-10 11:12:08 +04:00
Siavash Sameni
510eae2089 feat(direct-call): call history, recent contacts, deregister button
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Persistent JSON-backed call history for the direct-call screen so users
can see what they've placed / received / missed and dial back with one
click. Also fixes two small latent UX issues reported alongside.

Backend (Rust)
- new crate/module desktop/src-tauri/src/history.rs: thread-safe in-
  process store (OnceLock<RwLock<Vec<CallHistoryEntry>>>) backed by
  <APP_DATA_DIR>/call_history.json. Atomic writes via temp+rename. Max
  200 entries, FIFO pruning. CallDirection { Placed, Received, Missed }.
- Log hooks in the signal loop + commands:
    * place_call     → Placed entry (with target fingerprint)
    * DirectCallOffer → Missed entry up front; upgraded to Received
                        inside answer_call when accept_mode != Reject
                        via history::mark_received_if_pending(call_id).
                        If user rejects or never answers, it stays Missed.
- New Tauri commands:
    * get_call_history()     → all entries, newest first
    * get_recent_contacts()  → unique peers by fp, newest interaction first
    * clear_call_history()   → wipes JSON + in-memory
    * deregister()           → tears down signal transport + endpoint
  Backend emits `history-changed` events so the UI can live-refresh
  without polling.

Frontend (main.ts + index.html + style.css)
- Direct-call panel now has:
    * Recent contacts chip row (top 6 unique peers). Click a chip → dial.
    * Call history list (up to 50 rows). Direction icon (↗ placed, ↙
      received, ✗ missed), peer alias/fp, relative timestamp, callback
      button. Both click handlers populate target-fp and fire place_call.
    * Deregister button in the "registered" header — calls the new
      deregister command, tears down the signal transport, returns the
      UI to the pre-register state.
    * Clear-history link in the history header.
- Subscribes to `history-changed` events so the list updates the moment
  the backend logs a new entry. Also refreshed on register + after a
  clear.
- Nothing is rendered until there is data — empty sections stay hidden.

Tasks #20 + #21 (small UX items bundled in)
- Default room "general" for new installations: the html input value
  attribute is now "general" and loadSettings() defaults match. Existing
  users' localStorage still wins.
- Random alias on desktop: already latent but confirmed working — the
  startup IIFE at main.ts:374 calls get_app_info() and prefills the
  alias input from derive_alias(seed) when the input is empty. No code
  change needed, just verified it flows through the same path as the
  Android client.

Known follow-ups (deferred to step 6 polish)
- Call duration tracking (currently all entries have no duration field)
- Hangup signal from an unanswered incoming should emit history-changed
  so the missed state is visible even when the user never tapped accept
- Android UI layout fit-check on the smaller Nothing screen
2026-04-10 11:03:36 +04:00
Siavash Sameni
76a4c53e21 fix(android-audio): spawn_blocking for Oboe restart — unblock tokio executor
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Build 4c6aac6 added a stop+sleep+start Oboe restart inside the
set_speakerphone Tauri command, but calling wzp_native::audio_stop()
and audio_start() synchronously from an async fn blocks the tokio
executor thread — those FFI calls wait for AAudio to finalise the
stream teardown/bringup, which takes ~400ms each on Nothing phone
(Pixel is fast enough to hide the bug).

Reproduced on Nothing: 7 rapid Speaker button clicks across ~30
seconds, each restarting Oboe. After the 5th click the engine send
and recv tokio tasks froze for 22 seconds — decoded_frames stuck at
1159 across 9 heartbeats, send_drops growing from 148 to 1720 as
encoded frames couldn't make it past `send_t.send_media(pkt).await`.
At 08:40:48 the runtime finally caught up and processed a 911-frame
burst at once (buffered QUIC datagrams flooding through). Classic
"blocking sync call in async context" anti-pattern.

Fix: run the stop + start sequence inside tokio::task::spawn_blocking
so the Oboe teardown + reopen happens on a dedicated blocking thread,
leaving the tokio runtime free to keep driving the send and recv
tasks. AAudio's requestStop returns only after the stream is actually
in Stopped state, so the explicit sleep that bridged stop and start
is no longer needed and is dropped.

Send and recv tasks still see a ~500ms window of empty reads /
partial writes during the blocking restart, but they get SCHEDULED
through it — network packets keep being received + decoded + dropped
into the playout ring, and captured mic samples keep being encoded +
sent through quinn. No more executor starvation, no more 22-second
audio dropouts, no more send_drops burst.

Pixel still worked before this fix only because its AAudio teardown
is fast enough to not exceed the scheduler's cooperative yield
interval — same bug was latent on both devices, Nothing just made it
visible.
2026-04-10 08:45:54 +04:00
Siavash Sameni
4c6aac654a fix(android-audio): restart Oboe on speakerphone toggle + unbreak button UI
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Build 4f2ad65 wired the Speaker button to AudioManager.setSpeakerphoneOn
but user testing found that flipping speakerphone on an active Oboe
VoiceCommunication stream silently tears down the AAudio streams on
Pixel-class devices — both capture and playout stop producing data.
Only ending the call and rejoining brings audio back (because the fresh
Oboe open runs with the new routing already applied).

Also the earpiece state showed up red in the UI because the button was
getting the `.muted` CSS class when speakerphoneOn=false. Earpiece is a
valid routing state, not a muted one.

Fix set_speakerphone Tauri command:
  1. Flip AudioManager.setSpeakerphoneOn via JNI (as before).
  2. If the Oboe backend is currently running, stop it, sleep 50 ms to
     let AAudio finalise the transition, then start it again. The Rust
     send/recv tokio tasks keep running across the gap — they just read
     zero samples and write into the preserved ring buffers for a few
     frames, which is acceptable. The AudioBackend singleton's ring
     state is preserved across stop+start because it's in a 'static
     OnceLock.
  3. Debounce the UI click via speakerphoneBusy + spkBtn.disabled so
     users can't queue up multiple toggles during the restart window.

Fix main.ts Speaker button:
  - Remove the `.muted` classList toggle (added `.speaker-on` for CSS).
  - Update label text to "🔊 Speaker" / "🔈 Earpiece" for clarity.
  - On showCallScreen(), invoke is_speakerphone_on to sync the label
    with the real AudioManager state, so it matches reality after a
    rejoin (which was another symptom the user hit — the button label
    desynced from the actual routing after ending and restarting a
    call).
  - Debounce click + disable button while the restart is in flight.

Drops #[allow(dead_code)] from wzp_native::audio_is_running now that it
is actually called from the set_speakerphone restart guard.
2026-04-10 07:35:12 +04:00
Siavash Sameni
4f2ad65418 fix(android_audio): add explicit pointer types for .cast() — was rejected by rustc E0282 on android target
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2026-04-09 22:02:48 +04:00
Siavash Sameni
0178cbd91d android(audio): Speaker button toggles earpiece↔speaker via JNI (WIP, untested)
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Build 9e37201 confirmed on-device that Usage::VoiceCommunication +
MODE_IN_COMMUNICATION + speakerphoneOn=false routes Oboe playout to the
handset earpiece and the callback drains the ring correctly. Next step:
let the user flip speakerphoneOn at runtime so the existing Speaker
button actually switches audio routing instead of just gating writes.

- Cargo.toml (android target): pull in `jni = 0.21` and
  `ndk-context = 0.1`. Both are already transitively in the lockfile
  via Tauri/Wry, so this just promotes them to direct deps.
- desktop/src-tauri/src/android_audio.rs: new module. Grabs the JavaVM +
  current Activity from `ndk_context::android_context()`, attaches a
  JNI thread, calls `activity.getSystemService("audio")` to get the
  AudioManager, and exposes `set_speakerphone(bool)` +
  `is_speakerphone_on()` helpers that call the AudioManager method of
  the same name. All gated behind `#[cfg(target_os = "android")]`.
- lib.rs: adds `mod android_audio;` (android only), two new Tauri
  commands `set_speakerphone(on)` and `is_speakerphone_on()` — desktop
  gets no-op stubs so the same frontend invoke() works everywhere.
  Both registered in the invoke_handler.
- desktop/src/main.ts: the Speaker button (previously toggled the
  playout-write gate via `toggle_speaker`) now calls `set_speakerphone`
  and reads back the new routing state. Labels switched from
  "Spk" / "Spk Off" to "Earpiece" / "Speaker" so users can't be
  confused into thinking clicking turns audio off. pollStatus no longer
  clobbers the spkBtn label based on engine spk_muted, since the two
  concepts are now decoupled.

WIP because this has NOT been built or tested yet — committing at night
to save the work. Tomorrow: build #50 with this change, smoke-test the
Handset↔Speaker toggle, then move on to call history + last-contacts UI
and the Speaker-button mute bug on the other phone.
2026-04-09 22:00:34 +04:00
Siavash Sameni
9e37201198 android(audio): Usage::VoiceCommunication + MODE_IN_COMMUNICATION, default handset
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With da106bd (Usage::Media + MODE_NORMAL) audio works but is always on
the loudspeaker — we want handset as the default with a user-driven
toggle for speaker (and later bluetooth). The right Oboe usage for a
VoIP app is VoiceCommunication, which honours
AudioManager.setSpeakerphoneOn / setBluetoothScoOn for routing.

Bisection across previous builds showed that setAudioApi(AAudio) +
Usage::VoiceCommunication made the playout callback stop draining the
ring after cb#0 (build 8c36fb5 logs). Letting Oboe pick the AudioApi
implicitly keeps the callback alive — 96be740's Media-usage callbacks
fired at steady 50Hz without any explicit setAudioApi. So: keep the
Usage change, DROP the explicit AAudio force.

- oboe_bridge.cpp: Usage::VoiceCommunication, no setAudioApi, no
  ContentType override.
- MainActivity.kt: setMode(MODE_IN_COMMUNICATION) +
  setSpeakerphoneOn(false) = handset default, plus max both
  STREAM_VOICE_CALL and STREAM_MUSIC volumes for belt-and-braces.

Next build will add a JNI-based Tauri command to flip speakerphoneOn
at runtime so the user can toggle handset↔speaker during a call.
2026-04-09 21:50:06 +04:00
Siavash Sameni
da106bd939 fix(android-audio): revert to 96be740's Oboe config — VoiceCommunication broke callback drain
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Build 8c36fb5 logs showed a new regression: Oboe playout cb#0 fires once
at startup then the callback STOPS DRAINING the ring entirely.
written_samples sticks at 7679 (= RING_CAPACITY - 1) across every recv
heartbeat in a 40-second test. Meanwhile the recv task decodes 1800+ real
audio frames (sample range up to [-27920..31907], rms 12065) which all
get dropped on the floor by audio_write_playout returning 0 because the
ring is full.

Bisection: 96be740 (Usage::Media, no setAudioApi, no ContentType, no
MainActivity audio mode change) DID drive the playout callback at the
expected 50Hz (playout heartbeat: calls=1100 total_played_real=1055040
over 22 seconds). User still heard nothing there because of OS routing,
but at least Oboe accepted the PCM.

8c36fb5 added three changes on top of 96be740:
  1. Oboe Usage::Media → Usage::VoiceCommunication
  2. Oboe setAudioApi(oboe::AudioApi::AAudio) explicit
  3. Oboe setContentType(ContentType::Speech)
  4. MainActivity setMode(MODE_IN_COMMUNICATION) + setSpeakerphoneOn(true)
Every one of those could have killed the callback; combined they did.

Revert to 96be740's exact Oboe config: Usage::Media, no setAudioApi, no
ContentType. Keep the PCM recorder, heartbeat logging, and stream-open
logging. Separately, MainActivity now maxes STREAM_MUSIC (the stream
Usage::Media routes to) but leaves audio mode in MODE_NORMAL — no more
speakerphone/call-mode combo that makes Oboe unhappy. In NORMAL mode a
STREAM_MUSIC stream plays through the loud speaker by default.

Proof that the Rust pipeline is perfect: decoded.pcm recorded in 8c36fb5
was pulled via `adb shell run-as com.wzp.desktop cat .wzp/decoded.pcm`,
converted with ffmpeg, and played back on the Mac — user confirmed
audible speech. So 100% of the remaining bug surface is Android audio
routing, not anything in the Rust/C++ decode path.
2026-04-09 21:38:19 +04:00
Siavash Sameni
8c36fb5651 fix(wzp-native): Oboe ResultWithValue has no value_or, unfold explicitly
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cc-rs build of oboe_bridge.cpp failed at cfa9ff6 because the Oboe
ResultWithValue<T> template returned by getXRunCount() does not have
a .value_or(T) method — only .value(). Replace with an explicit
bool-conversion + .value() guard that yields -1 on error.
2026-04-09 21:25:38 +04:00
Siavash Sameni
cfa9ff67cf fix(android-audio): VoIP mode + speakerphone + debug PCM recorder
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Build 96be740 logs proved the entire software pipeline is healthy:
  capture heartbeat:   calls=1100 to_write=960 full_drops=0 total_written=1056000
  recv heartbeat:      decoded_frames=1035 last_written=960 decode_errs=0
  recv decoded PCM:    range=[-13564..9244] rms=8044 (real audio)
  playout WRITE:       in_len=960 written=960 rms=2318 (real audio into the ring)
  playout heartbeat:   calls=1100 nonempty=1099 total_played_real=1055040

1055040 samples / 48000 Hz = 22s — exactly matches wall-clock elapsed,
meaning Oboe IS calling our playout callback at the expected rate and
WE ARE handing it real PCM every 20ms. User still heard nothing. Ergo
Oboe accepted the PCM and routed it to a silent output. Two fixes:

1) MainActivity.kt: switch to MODE_IN_COMMUNICATION + speakerphone ON
   right after permissions are granted, and crank STREAM_VOICE_CALL to
   max. Without this, an Oboe Usage::VoiceCommunication stream gets
   opened, the OS creates a real AAudio pipeline, the callback fires on
   schedule — and audio goes to either the earpiece at muted volume or
   a "call not active" dead end. Logs the audio mode + volume levels
   before and after the switch so we can confirm the state change in
   logcat next run.

2) oboe_bridge.cpp: revert Usage::Media → VoiceCommunication (the mode
   that matches MODE_IN_COMMUNICATION), pin the audio API to AAudio
   explicitly instead of letting Oboe fall back to OpenSLES (which has
   its own silent-drop failure modes on some devices), and add getState
   + getXRunCount to the playout heartbeat so we'll see silent stream
   disconnects instead of reading zeros forever.

3) engine.rs recv task: dump the first ~10s of post-AGC decoded PCM to
   `<app_data_dir>/decoded.pcm` as raw i16 LE so we can adb pull it and
   play it back locally:
       adb shell run-as com.wzp.desktop cat .wzp/decoded.pcm > decoded.pcm
       ffmpeg -f s16le -ar 48000 -ac 1 -i decoded.pcm decoded.wav
   This divorces "is our decoder actually producing audible audio" from
   "is Android's audio stack playing it". If the recorded WAV sounds
   correct when played on a laptop, the decoder is fine and 100% of the
   remaining bug surface is AudioManager / Oboe routing.

4) engine.rs: also log when spk_muted=true blocks the write. User
   reported the Speaker button in the UI has inconsistent semantics
   between desktop and android — adding this log rules out the accidental
   "first click muted playback" theory for good.
2026-04-09 21:24:26 +04:00
Siavash Sameni
96be740fd9 diag(android-audio): aggressive logging across the whole Oboe pipeline
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User confirmed: mac hears android, android does not hear mac. So Oboe
capture works end-to-end but Oboe playout on Android silently drops
audio even though QUIC forwards the packets. Archaeology on the legacy
wzp-android crate also revealed that the "last known good" Android audio
path NEVER used Oboe in production — it used Kotlin AudioRecord +
AudioTrack via JNI, and cpp/oboe_bridge.cpp was dead code. So every time
we've "tested" Oboe end-to-end this week was the first production use,
and any of its config knobs could be the bug.

Instrumenting every stage of the pipeline so one smoke-test log dump can
isolate the layer at fault:

C++ (oboe_bridge.cpp)
  - Log the ACTUAL stream parameters after openStream for both capture
    and playout (sample rate, channels, format, framesPerBurst,
    framesPerDataCallback, bufferCapacityInFrames, sharing, perf mode).
    Oboe may silently override values we requested — e.g. if we ask for
    48kHz mono but the device gives us 44.1kHz stereo our 960-sample
    frames are the wrong duration and the pipeline drifts.
  - Capture callback: on cb#0 log sample range+RMS of the first frame
    to prove we get real mic data (not zeros). Every 50 callbacks
    (~1s at 20ms burst) log calls, numFrames, ring available_write,
    bytes actually written, ring_full_drops, total_written.
  - Playout callback: on cb#0 log numFrames + ring state. On the FIRST
    non-empty read log sample range+RMS so we can tell if the samples
    coming out of the ring are real audio or zeros. Every 50 callbacks
    log calls, nonempty count, numFrames, ring available_read,
    underrun_frames, total_played_real.

Rust wzp-native (src/lib.rs)
  - wzp_native_audio_write_playout now logs the first 3 writes and then
    every 50th: in_len, written, sample range, RMS, ring write/read
    cursors before, available_read and available_write after. Reveals
    ring-overflow and whether the engine is actually handing us audio.
  - Minimal android logcat shim via __android_log_write extern — no
    new crate dependency.
  - AudioBackend grows a `playout_write_log_count` AtomicU64 to gate
    the write-side log throttle.

Rust engine.rs (android branch)
  - Recv task: log sample range + RMS for the first 3 decoded PCM
    frames and then every 100th. Reveals whether decoder.decode is
    producing real audio or silent buffers.
  - Recv task: if audio_write_playout returns fewer samples than we
    handed it (partial write → ring nearly full) warn about it in the
    first 10 frames.
  - Recv heartbeat every 2s: recv_fr, decoded_frames, last_decode_n,
    last_written, written_samples, decode_errs, codec.

Expected flow in a healthy log:
  capture cb#0: numFrames=960 range=[-1200..900] rms=180          ← mic OK
  capture stream opened: actualSR=48000 Ch=1 ...                   ← no override
  playout stream opened: actualSR=48000 Ch=1 ...
  CallEngine::start invoked ... → connected → audio started
  recv: first media packet received ...
  recv: decoded PCM sample range decoded_frames=1 range=[-300..250] rms=92
  playout WRITE #0: in_len=960 written=960 range=[-300..250] rms=92
  playout FIRST nonempty read: to_read=960 range=[-300..250] rms=92
  playout heartbeat: calls=50 nonempty=50 underrun=0 ...
  recv heartbeat: decoded_frames=100 last_written=960 ...

If any of those are missing/zero we know the exact stage to fix.
2026-04-09 21:13:29 +04:00
Siavash Sameni
8c4d640f89 fix(android): playout Usage::Media + relay CallSetup advertises real IP
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Three real bugs, one smoke-test session's worth of progress.

1. RELAY: wrong advertised addr in CallSetup
   The direct-call CallSetup computed `relay_addr = addr.ip()` where
   `addr = connection.remote_address()` — i.e. the CLIENT'S IP, not the
   relay's. So the relay was telling both parties "the call room is at
   the answerer's IP:4433", which meant each client dialed either the
   other client (no server listening) or themselves. Both endpoint.connect
   calls hung forever and the call never happened.
   Fix: compute the relay's own advertised IP once at startup. If the
   listen addr is 0.0.0.0, probe the primary outbound interface via the
   classic UDP-bind-and-connect(8.8.8.8:80) trick to discover the LAN
   IP the OS would use to reach external hosts. Thread the resulting
   advertised_addr_str into the CallSetup sender for both parties.

2. RELAY: accept loop serialized QUIC handshakes
   Previously the main accept loop called `wzp_transport::accept` which
   did both `endpoint.accept().await` AND `incoming.await` (the server-
   side QUIC handshake). A single slow handshake therefore blocked every
   subsequent client from being accepted. Unroll the helper here and
   move `incoming.await` into the per-connection spawned task, so every
   handshake runs in parallel. Also log "accept queue: new Incoming",
   "QUIC handshake complete", and "QUIC handshake failed" so we can tell
   immediately whether a client's packets are reaching the relay at all.

3. ANDROID: playout was routed to the silent in-call stream
   The Oboe playout stream was configured with Usage::VoiceCommunication,
   which routes to the Android in-call earpiece stream. That stream is
   silent unless the Activity has called AudioManager.setMode(
   IN_COMMUNICATION) and, even then, only the earpiece/BT headset get
   audio (not the loud speaker). Result: android→mac calls worked
   because mac had a normal media output, but mac→android calls were
   silent even though packets flowed through the relay just fine.
   Switch to Usage::Media + ContentType::Speech so Oboe routes to the
   loud speaker and uses the media volume slider. A later polish step
   will wire setMode + setSpeakerphoneOn from MainActivity.kt so we can
   go back to VoiceCommunication for AEC and proximity-sensor routing.

Plus: heartbeat tracing every 2s in the send/recv tasks — frames_sent,
last_rms, last_pkt_bytes, short_reads on the send side; decoded_frames,
last_decode_n, last_written, decode_errs on the recv side. Will make the
next "no sound" regression trivial to localize.
2026-04-09 20:55:10 +04:00
Siavash Sameni
49f101d785 fix(android): reuse signal endpoint for direct-call media connection
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Direct-call accept hangs forever at the QUIC handshake on Android. Logs
from d7b37a5 showed:
  CallEngine::start (android) invoked relay=172.16.81.172:4433 room=call-…
  resolved relay addr
  identity loaded
  endpoint created, dialing relay   ← reached
                                    ← nothing, 90s+, no error
The "connect failed" and "QUIC connection established" log lines never
fire, meaning endpoint.connect_with(…).await never makes progress.

Repro is 100%: SFU room join (one endpoint) works perfectly; direct call
(opens a SECOND quinn::Endpoint on top of the signal one) hangs in the
QUIC handshake. Creating two quinn::Endpoints on Android's AAudio-adjacent
UDP stack apparently causes the second one's datagrams to never reach the
relay (the server never sees the Initial packet). Rather than fight the
platform, quinn is happy to multiplex multiple Connections on a single
Endpoint — so we reuse the signal endpoint for the media connection.

- SignalState now stores the quinn::Endpoint alongside the QuinnTransport.
  register_signal populates both at the same time.
- CallEngine::start (both android and desktop branches) takes an
  Option<wzp_transport::Endpoint>. Some → reuse (direct-call path, after
  register_signal). None → create fresh (SFU room join path).
- The connect tauri command reads state.signal.endpoint and threads it
  through to CallEngine::start, so the direct-call auto-connect (fired by
  the "setup" signal-event in main.ts) lands on the existing UDP socket.
- wzp_transport re-exports quinn::Endpoint so wzp-desktop doesn't need to
  depend on quinn directly.
- Also wraps the android connect in tokio::time::timeout(10s) so future
  hangs become deterministic "connect TIMED OUT" errors in logcat
  instead of silent deadlock.

Same fix applies verbatim to the desktop client — the user suspects
direct call is broken there too and this was likely always the cause,
just never surfaced because desktop was only tested via SFU rooms.
2026-04-09 20:29:51 +04:00
Siavash Sameni
d7b37a5749 diag: tracing for direct-call signal loop + CallEngine::start stages
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User reports tapping "answer" on an incoming direct call does nothing
visible, and suspects the same may affect desktop. The signal recv loop
had no tracing at all, so we can't tell whether CallSetup is being
received, whether the recv loop died silently, or whether
CallEngine::start is failing between "identity loaded" and
"connected to relay, handshake complete".

- register_signal recv loop now logs every message type with fields
  (CallRinging, DirectCallOffer, DirectCallAnswer, CallSetup, Hangup,
  unhandled), plus a warn! on recv errors and a final warn when the
  loop exits.
- place_call / answer_call commands log entry + success / error. The
  answer_call error path logs the underlying send_signal error so we
  can see it in logcat instead of only in the JS error toast.
- CallEngine::start android branch logs relay/room/alias on entry,
  logs "endpoint created, dialing relay" between create_endpoint and
  connect, "QUIC connection established, performing handshake" between
  connect and perform_handshake, and promotes all three potential
  failures to explicit error! logs so a silent hang / error becomes
  visible in logcat.

No functional changes — pure diagnostics. Stacks on b35a6b7 (the Oboe
stack-pointer-escape fix) so build #43 carries both.
2026-04-09 19:17:03 +04:00
Siavash Sameni
b35a6b7d92 fix(wzp-native): copy WzpOboeRings by value, not by pointer
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PlayoutCallback::onAudioReady crashed with SIGSEGV(SEGV_ACCERR) on the
first AAudio callback because g_rings was a `const WzpOboeRings*` pointing
at the caller's stack frame. wzp_native_audio_start() constructs the
rings struct as a stack local in Rust, passes &rings to wzp_oboe_start
(which stored the raw pointer), and returns — at which point the stack
frame unwinds and g_rings becomes a dangling reference. The first audio
callback then read from freed memory and died.

- g_rings is now a static WzpOboeRings value (was `const WzpOboeRings*`).
  The raw int16 buffer + atomic index pointers inside the struct still
  point into the Rust-owned AudioBackend singleton, which is leaked for
  the lifetime of the process, so deep-copying the struct by value is
  safe and keeps the inner pointers valid forever.
- g_rings_valid atomic bool gates the audio-callback reads: set to true
  after the value copy in wzp_oboe_start, cleared in wzp_oboe_stop BEFORE
  the streams are torn down so any in-flight callback sees "no backend"
  and returns Stop instead of racing on g_rings.
- All g_rings->x accesses in the capture + playout callbacks switched to
  g_rings.x (member-of-value).

Reproduced on Pixel 6 / Android 15 with build 0105b0f:
  F libc: Fatal signal 11 (SIGSEGV), code 2 (SEGV_ACCERR),
          fault addr 0x71aa717eb0 in tid 11822 (AudioTrack)
  #00 PlayoutCallback::onAudioReady(oboe::AudioStream*, void*, int)+120
  #01 oboe::AudioStream::fireDataCallback(void*, int)+136
  ...
2026-04-09 19:11:16 +04:00
Siavash Sameni
0105b0fbf3 phase 3(android): RECORD_AUDIO permission + runtime request in MainActivity
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Oboe fails silently to open the AAudio input stream without
android.permission.RECORD_AUDIO, so the call audio would never actually
flow even after phase 3's engine wiring.

- AndroidManifest.xml: declare RECORD_AUDIO and MODIFY_AUDIO_SETTINGS, and
  android.hardware.microphone as a required feature. These files are the
  cargo-tauri-generated scaffold — nothing in .gitignore excludes them, so
  the intended Tauri 2 mobile workflow is to commit them once populated.

- MainActivity.kt: override onCreate to call ActivityCompat.requestPermissions
  for the audio perms on first launch. The dialog shows exactly once; the
  grant is persisted per-package. onRequestPermissionsResult logs the
  outcome so we can spot failures in logcat.

A full native Tauri permission plugin integration is deferred to
Step 6 (polish) together with notifications, icon, and background service.
2026-04-09 19:00:12 +04:00
Siavash Sameni
5beea7de40 phase 3(android): unify connect/disconnect/toggle_*/get_status commands
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Step 3 of the Tauri Android rewrite was still returning "audio backend not
yet wired on Android (step 3)" because the cfg-gated Android stubs for
connect/disconnect/toggle_mic/toggle_speaker/get_status were shadowing the
real commands. Now that CallEngine::start() has a real Android body (phase
3, commit fdbe502), the gates are unnecessary.

- Drop the #[cfg(not(target_os = "android"))] gates from all five
  engine-backed Tauri commands.
- Delete the Android stub block (~50 LOC of "not connected" boilerplate).
- Ungate `use engine::CallEngine;` and the AppState.engine field so both
  targets share the same Mutex<Option<CallEngine>>.
- CallEngine::stop() now calls crate::wzp_native::audio_stop() on Android so
  the mic + speaker are released between calls, matching the desktop
  behaviour where dropping _audio_handle tears down CPAL.

Direct-call flow on Android: peer sends DirectCallOffer → user accepts via
answer_call → relay sends signal "setup" event → main.ts auto-invokes
connect(relay, room) → CallEngine::start() runs the Android branch →
wzp_native::audio_start() brings up Oboe → send/recv tasks stream PCM
through the dlopen boundary.
2026-04-09 18:53:54 +04:00
Siavash Sameni
fdbe502524 phase 3(android): wire CallEngine::start to wzp-native audio FFI
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Replaces the Android-side CallEngine::start() stub with a real implementation
that mirrors the desktop start() body but routes all PCM through the
standalone wzp-native cdylib loaded at startup via libloading instead of
using CPAL.

- desktop/src-tauri/src/wzp_native.rs: new module with a static
  OnceLock<libloading::Library> + cached raw fn pointers for every symbol
  we need (version, hello, audio_start/stop, read_capture, write_playout,
  is_running, capture/playout_latency_ms). init() resolves everything once
  at startup; accessors return default values if init() never ran.

- desktop/src-tauri/src/lib.rs: drop the inline dlopen smoke test, add
  `mod wzp_native;` behind target_os="android", and invoke
  wzp_native::init() from the Tauri setup() callback so the library is
  loaded + all symbols cached before any CallEngine can touch audio.

- desktop/src-tauri/src/engine.rs: the Android #[cfg] branch of
  CallEngine::start() now does the full QUIC handshake + signal loop +
  Opus send/recv tasks, calling wzp_native::audio_start() /
  audio_read_capture() / audio_write_playout() instead of the desktop
  CPAL rings. SyncWrapper now holds a placeholder Box<()> on Android
  because the audio backend lives in a process-global singleton inside
  libwzp_native.so rather than being owned per-engine.

Next step: build #39 on the remote docker builder and smoke-test on
Pixel 6 that the Connect button in the UI successfully brings up Oboe
and streams audio through the dlopen boundary.
2026-04-09 18:42:27 +04:00
Siavash Sameni
c769a476a2 phase 2(android): port Oboe C++ bridge + audio FFI into wzp-native
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Now that Phase 1 proved the split-cdylib pipeline (build #37 launched
cleanly with 'wzp-native dlopen OK: version=42 msg=...' in logcat),
this commit brings the real audio code into wzp-native without ever
touching the Tauri crate:

- cpp/oboe_bridge.{h,cpp}, oboe_stub.cpp, getauxval_fix.c copied
  verbatim from crates/wzp-android/cpp/ (same files that work in the
  legacy wzp-android .so on this phone)
- build.rs near-identical to crates/wzp-android/build.rs: clones
  google/oboe@1.8.1 into OUT_DIR, compiles oboe_bridge.cpp + all
  oboe source files as a single static lib with c++_shared linkage,
  emits -llog + -lOpenSLES. On non-android hosts it compiles just
  oboe_stub.cpp so `cargo check` works locally without an NDK.
- Cargo.toml gets cc = "1" in [build-dependencies]. This is SAFE
  because wzp-native is a single-cdylib crate — crate-type is only
  ["cdylib"], no staticlib, so rust-lang/rust#104707 does not apply.
- src/lib.rs extends the FFI surface with the real audio API:
    wzp_native_audio_start() -> i32
    wzp_native_audio_stop()
    wzp_native_audio_read_capture(*mut i16, usize) -> usize
    wzp_native_audio_write_playout(*const i16, usize) -> usize
    wzp_native_audio_capture_latency_ms() -> f32
    wzp_native_audio_playout_latency_ms() -> f32
    wzp_native_audio_is_running() -> i32
  Plus a static AudioBackend singleton holding the two SPSC ring
  buffers (capture + playout) that are shared with the C++ Oboe
  callbacks via AtomicI32 cursors. The wzp_native_version() and
  wzp_native_hello() smoke tests from Phase 1 are preserved.

Compiles cleanly on macOS host with the stub oboe .cpp. Next build
will exercise the full cargo-ndk path inside docker to verify the
whole Oboe compile still works standalone.

Phase 3 (next commit): wzp-desktop engine.rs on Android calls
wzp-native's audio FFI via the already-wired libloading handle, and
the real CallEngine::start() is implemented for Android using the
same codec/handshake/send/recv pipeline as desktop but with Oboe
rings instead of CPAL rings.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 18:12:01 +04:00
Siavash Sameni
7cc53aedc7 refactor(android): split C++ into wzp-native cdylib, loaded at runtime
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Phase 1 of the big refactor. Escape the Tauri Android
__init_tcb+4 symbol leak (rust-lang/rust#104707) by making
wzp-desktop's Android .so pure Rust — ZERO cc::Build, no cpp/ files,
no C++ in the rustc link step. All future C++ (Oboe audio bridge)
lives in a new standalone cdylib crate `wzp-native` which is built
with cargo-ndk (the same path the legacy wzp-android crate uses
successfully on the same phone + same NDK), copied into Tauri's
gen/android/app/src/main/jniLibs at build time, and dlopened by
wzp-desktop at runtime via libloading.

Changes in this commit:
- NEW crate crates/wzp-native/ with crate-type = ["cdylib"] only
  (no staticlib, no rlib — rust#104707 shows mixing staticlib with
  cdylib leaks non-exported symbols, which is the original bug
  source). Phase 1 scaffold has TWO extern "C" functions:
    wzp_native_version() -> i32            (returns 42)
    wzp_native_hello(buf, cap) -> usize    (writes a string)
  So we can verify dlopen + dlsym + cross-.so FFI end-to-end
  before adding any real C++.
- desktop/src-tauri/cpp/ directory DELETED (7 files gone).
- desktop/src-tauri/build.rs reduced to just the git hash capture
  + tauri_build::build(). No more cc::Build of any kind.
- desktop/src-tauri/Cargo.toml: drop cc from build-dependencies,
  add libloading = "0.8" as an Android-only runtime dep.
- desktop/src-tauri/src/lib.rs Builder::setup() now (on Android only)
  dlopens libwzp_native.so, calls wzp_native_version() and
  wzp_native_hello(), and logs the result:
    "wzp-native dlopen OK: version=42 msg=\"hello from wzp-native\""
  If this log appears in logcat when the app launches and the home
  screen still renders, the split-cdylib pipeline is validated and
  Phase 2 (port the Oboe bridge into wzp-native) can proceed.
- scripts/build-tauri-android.sh: insert a `cargo ndk -t arm64-v8a
  build --release -p wzp-native` step before `cargo tauri android
  build`, with `-o desktop/src-tauri/gen/android/app/src/main/jniLibs`
  so the resulting libwzp_native.so lands in the place gradle will
  package into the final APK.
- Workspace Cargo.toml: add crates/wzp-native to [workspace] members.

Phase 2 (separate commit, only if Phase 1 works):
- Copy cpp/oboe_bridge.{h,cpp} + getauxval_fix.c from the legacy
  wzp-android crate into crates/wzp-native/cpp/.
- Add cc = "1" as a build-dependency on wzp-native (safe: it's a
  single-cdylib crate with no staticlib, so no symbol leak).
- Add build.rs that compiles the Oboe C++ and the wzp-native Rust
  FFI exposes the audio start/stop/read/write functions.
- wzp-desktop::engine.rs dlopens wzp-native at CallEngine::start,
  uses its audio functions instead of CPAL on Android.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 18:02:53 +04:00
Siavash Sameni
711137da96 fix(android): -Wl,--exclude-libs,ALL + --no-whole-archive to stop symbol leak
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llvm-nm on the crashing .so confirmed the research's smoking gun theory:

  000000000130c1f0 t _Z10__init_tcbP10bionic_tcbP18pthread_internal_t
  0000000000000000 a pthread_create.cpp
  0000000001331108 t pthread_create

All lowercase 't' (= LOCAL text symbols), zero UND dynamic references
for pthread_create. So rustc's link step is pulling bionic's own
pthread_create.cpp compilation unit out of libc.a as a whole-archive
inclusion and binding those symbols locally inside our .so, instead
of letting them stay UND and resolved against libc.so at dlopen time.

Rust's libstd thread::spawn then calls the LOCAL (broken) pthread_create
which calls the LOCAL __init_tcb with arguments set up for bionic's
static-executable layout — crashes at __init_tcb+4 with SEGV_ACCERR.

`-Wl,--exclude-libs,ALL` tells the linker to make symbols from static
archives NOT appear in the dynamic symbol table of the output .so.
`-Wl,--no-whole-archive` tells it to only pull archive objects that
satisfy undefined references, not include the whole archive blindly.

If this works, the symbol table should show pthread_create as UND
(or at least not locally bound) and the app should launch. If it
doesn't, the remaining fallback is the research's action #3 —
extract the C++ into its own upstream cdylib crate built with
cargo-ndk, and dlopen it from the Tauri cdylib at runtime.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 17:45:35 +04:00
Siavash Sameni
6071eb1b02 fix(android): drop staticlib from crate-type — root cause of __init_tcb crash
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External research (per rust-lang/rust#104707) pointed at this as the
highest-probability cause of our byte-identical __init_tcb+4 /
pthread_create SIGSEGVs:

> Having 'staticlib' alongside 'cdylib' in crate-type leaks non-exported
> symbols from the staticlib into the cdylib's symbol table. For a
> Tauri Android cdylib, that means bionic's private pthread_create /
> __init_tcb code — which got pulled in statically from libc.a the
> moment any cc::Build C++ file added C++-linkage overhead — ends up
> bound LOCALLY inside our .so instead of being resolved dynamically
> against libc.so at dlopen time.

Symptoms that match the theory exactly:
- llvm-nm on the crashing .so shows __init_tcb and pthread_create as
  LOCAL symbols with C++ name mangling (bionic's own pthread_create.cpp)
- Adding any cc::Build cpp(true) step reliably triggers the crash,
  independent of which linker (android24-clang vs android26-clang) or
  which libc++ linkage (shared/static/none)
- The legacy wzp-android crate (["cdylib", "rlib"]) works fine on the
  same phone with the same NDK + Rust toolchain + Oboe C++ code
- tauri.conf.json bundle.android.minSdkVersion=26 propagates to
  gradle but the .so still crashes byte-identically

Drop 'staticlib' from crate-type. If we ever need it for iOS, re-add
behind a target.'cfg(target_os = "ios")' gate. The desktop binary
still links against the rlib, so the bin target on macOS/Linux/Windows
is unaffected.

Source: https://github.com/rust-lang/rust/issues/104707

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 17:38:49 +04:00
Siavash Sameni
c9cd043657 test: tauri.conf.json bundle.android.minSdkVersion=26 + cpp_smoke.cpp c++_shared
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User theory: tauri-cli hardcodes minSdkVersion=24 into its rustc
invocation regardless of gradle build.gradle.kts, .cargo/config.toml,
or env var overrides — but DOES read from tauri.conf.json's
bundle.android block. That would explain why every cc::Build C++
compile crashed with __init_tcb+4 via pthread_create: API-24 bionic's
.init_array routines for the linked-in .init_array clash with the
pthread_create state tao later expects.

This commit applies the fix AND re-adds the smallest known crashing
variant (E.1 with cpp_link_stdlib('c++_shared')) so the test has one
clear failure mode to compare against:

  tauri.conf.json bundle:
    "android": { "minSdkVersion": 26 }

  build.rs (on android target):
    - hello.c           (plain C, worked in Step A)
    - getauxval_fix.c   (plain C, worked in Step D)
    - hello2.c          (plain C, worked in Step D+1)
    - cpp_smoke.cpp     (C++ via cc::Build .cpp(true), crashed in E.1)

Also re-emits the libc++_shared.so copy into gen/android jniLibs so
the runtime linker can resolve the NEEDED entry cc-rs added via
cpp_link_stdlib('c++_shared').

If this launches → theory validated, proceed with Oboe integration.
If this crashes → need to keep digging.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:58:37 +04:00
Siavash Sameni
6dd62c94c9 step D+1: add third trivial C static lib (hello2.c)
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Step D (hello.c + getauxval_fix.c) launches cleanly. E.minus-1
(hello.c + getauxval_fix.c + cpp_smoke.c) crashes. All three are
plain-C trivial single-function files.

Theory: the regression is triggered by having 3 or more cc::Build
static libs in a Tauri Android cdylib, regardless of what the libs
contain. Test: clone hello.c as hello2.c (same content, different
symbol) and add a third cc::Build step compiling it. If this crashes,
the trigger is just the number of static libs. If it launches, there's
something magical about cpp_smoke.c specifically (unlikely — it was
near-identical content).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:51:50 +04:00
Siavash Sameni
4c998312aa regression check: revert build.rs to exact Step D state
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Verify the Step D baseline still launches after the environment mutations
we may have caused during the E bisection (docker image rebuild, tauri-cli
version drift, etc). Build.rs is now byte-identical to commit a852cad
(Step D) except for the git hash capture block that already existed at
that point.

If this launches cleanly → the cpp_smoke addition genuinely breaks
something, bisection continues.
If this crashes → the environment regressed between Step D and now,
and we need to rebuild the docker image to an earlier snapshot.
2026-04-09 16:45:34 +04:00
Siavash Sameni
22701830c2 step E.minus-1: cpp_smoke renamed to .c and compiled as plain C
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c++_shared crashed, c++_static crashed, no stdlib crashed. The remaining
variable isolated to cc::Build::new().cpp(true) itself is the C++
compile-mode invocation of clang++. Rename cpp_smoke.cpp → cpp_smoke.c
and drop .cpp(true), leaving a plain-C cc::Build that compiles the
exact same bytes (minus the 'extern "C"' linkage spec which is C++-
only syntax).

This is structurally identical to Step A (hello.c), which worked. If
THIS build launches, the diff between 'works' and 'crashes' is purely
the .cpp(true) mode — something clang++ does differently at compile
or link time when producing object files for a Tauri Android cdylib.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:38:29 +04:00
Siavash Sameni
47a037368c step E.0: drop cpp_link_stdlib entirely (no libc++ linkage)
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c++_shared crashed. c++_static also crashed. Both have libc++ code
landing in the final .so — one as a NEEDED dynamic lib, the other
bundled statically. So the trigger isn't the NEEDED entry specifically,
it's libc++ being present in any form.

cpp_smoke.cpp is just 'extern "C" int wzp_cpp_hello() { return 42; }'
with zero C++ features used, so we can drop cpp_link_stdlib completely
and the compile still succeeds. No libc++ .a or .so referenced at all.

If this crashes: the trigger is cc::Build::new().cpp(true) switching
rustc's final linker driver from clang to clang++ (which pulls in
different default libraries).

If this launches: the trigger is libc++'s own static initializers or
the libc++ code itself doing something that breaks our .so at dlopen
time, and we have a path forward — C++ code that doesn't need libc++
(e.g., a thin C++ bridge to Oboe that uses only POD types at the
boundary, with all the STL stuff confined to Oboe's own compilation
unit which would still need libc++...). More likely we still need a
C-only audio interface like raw AAudio via the ndk Rust crate.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:31:53 +04:00
Siavash Sameni
191e8761d5 step E.1 variant: cpp_link_stdlib c++_shared → c++_static
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Every E.x variant crashed identically when linked with c++_shared, even
with a 3-line cpp file that's dead-stripped from the final .so. The
crash offsets are byte-identical across E.1, E.2, E.4, and the original
full-Oboe Step E. That points at a non-code link-time delta: the
`cargo:rustc-link-lib=c++_shared` directive that adds a NEEDED entry
for libc++_shared.so to the .so's dynamic table.

Swap to c++_static — bundles libc++ directly into our .so so the
NEEDED entry disappears. If this launches cleanly, we've conclusively
proven the NEEDED libc++_shared.so is the root cause and we have a
workable linkage for any C++ we want to add to the Tauri Android build
(including the eventual Oboe audio backend).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 16:18:04 +04:00
Siavash Sameni
0d74366592 step E.1: absolute minimum C++ file (no STL, no includes)
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Last bisection step. cpp/cpp_smoke.cpp reduced to a single extern 'C'
function that returns 42. No #include, no std::atomic, no std::mutex,
no std::thread. Only C++ things remaining are:
  - cc::Build::new().cpp(true) in build.rs (C++ mode compile)
  - cpp_link_stdlib('c++_shared') emitting -lc++_shared

If this still crashes with the same __init_tcb+4 / pthread_create
stack, we've conclusively proven the trigger is NOT any C++ code
that ends up in the final .so (everything gets dead-stripped
anyway because Rust never references wzp_cpp_hello). The trigger
must be either:
  a) cargo:rustc-link-lib=c++_shared (adds NEEDED entry for
     libc++_shared.so in the .so's dynamic table, causing the
     dynamic linker to load libc++_shared.so at dlopen() time
     alongside our .so), or
  b) Some interaction between cpp(true) mode and the rest of the
     build pipeline (toolchain flags, symbol visibility, etc.)

After this build we stop and write an incident report for the
WarzonePhone Tauri Android rewrite bisection so far.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 15:54:21 +04:00
Siavash Sameni
0224ce654c step E.2: shrink cpp_smoke to std::atomic only — no thread, no mutex
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Incremental bisection within Step E. E.4 (atomic + mutex + thread) still
crashed at __init_tcb. Drop mutex and thread, keep only std::atomic.
Build.rs still emits cargo:rustc-link-lib=c++_shared via
cpp_link_stdlib('c++_shared'), so the NEEDED entry for libc++_shared.so
in the final .so stays identical. Goal: if this crashes, the issue is
purely the dynamic link against libc++_shared (not thread/mutex code).
If it passes, the issue is actually std::thread or std::mutex use.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 15:47:30 +04:00
Siavash Sameni
aa240c6d83 step E.4(android): replace full Oboe compile with minimal C++ smoke file
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Bisection for the __init_tcb+4 crash that Step E introduced: drop the
full Oboe C++ build (200+ files, hundreds of KB of code) and replace
it with ONE tiny cpp/cpp_smoke.cpp that exercises the libc++ features
Oboe uses — std::atomic, std::mutex, std::thread — via an
extern "C" wzp_cpp_smoke() function that's exported but NEVER called
from Rust.

Still compiled with cpp_link_stdlib("c++_shared"), same as Oboe.
libc++_shared.so still copied into gen/android jniLibs. But no Oboe
headers, no Oboe source files, no -llog / -lOpenSLES links.

Hypothesis: if cpp_smoke.cpp alone reproduces the __init_tcb crash,
the trigger is "any libc++_shared link that references
std::thread/std::mutex" and Oboe is not the specific culprit. If it
launches cleanly, Oboe itself (its size, its static constructors, or
a specific header) is responsible — and we then bisect Oboe's
source tree.

fetch_oboe() and add_cpp_files_recursive() are retained in build.rs
with #[allow(dead_code)] so re-enabling the full Oboe compile is a
one-line edit once we've identified what's safe to include.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 15:39:30 +04:00
Siavash Sameni
d216dcc7a3 step E fix (Option 3): bake android24→26 clang shim into image
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Incremental Step E (commit 4250f1b) proved that merely compiling the
Oboe C++ bridge into libwzp_desktop_lib.so — with NO Rust-side FFI
bindings, no function calls — resurrects the __init_tcb+4 / pthread_
create SIGSEGV at WryActivity.onCreate. Bisection:

  build #17 (baseline)   ✓
  build #18 (Step A, hello.c)              ✓
  build #19 (Step B, wzp-client dep)       ✓
  build #21 (Step C, engine mod compiled)  ✓
  build #22 (Step D, getauxval_fix.c)      ✓
  build #23 (Step E, Oboe C++ compiled)    ✗ — __init_tcb+4 crash

Root cause: tauri-cli hard-codes `aarch64-linux-android24-clang` as the
Rust linker. Without any C++ code in the .so, libstd's pthread_create
reference gets resolved against the dynamic libc.so. The moment we add
a C++ static library that links against libc++_shared, the link-time
resolution pulls in the API-24 libc.a static pthread_create stub — and
Rust's libstd then also calls that stub instead of libc.so's real one.
The stub calls __init_tcb which SIGSEGVs because bionic's TCB state
only exists for static-libc main executables, not .so's loaded via
dlopen. API-26 NDK has proper dynamic bindings that resolve correctly.

Option 3 fix: at image build time, replace every NDK
aarch64-linux-android24-clang (and armv7/x86_64/i686, clang/clang++)
binary with a one-line shell script that exec()s the corresponding
android26-clang. Since tauri-cli invokes the linker via absolute path,
PATH and env var overrides fail — but replacing the binary on disk
inside the image is guaranteed to take effect. The legacy wzp-android
crate doesn't need this because cargo-ndk respects .cargo/config.toml
where a crate-level linker override is set.

Only changing the Dockerfile here. Next: rebuild the image no-cache,
retry Step E, and if the baseline holds, proceed to Steps F/G.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 15:17:34 +04:00
Siavash Sameni
4250f1b44a step E(android): compile full Oboe C++ bridge (not yet called from Rust)
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Fifth incremental variable — and the first genuinely heavy one. Adds:
  - cpp/oboe_bridge.{h,cpp} (copied verbatim from crates/wzp-android/cpp/)
  - cpp/oboe_stub.cpp (fallback if Oboe can't be fetched)
  - build.rs now clones google/oboe@1.8.1 into OUT_DIR and compiles
    oboe_bridge.cpp + every .cpp file under oboe/src/ as a single
    static library via cc::Build, using shared libc++. Same logic as
    the legacy wzp-android build.rs.
  - libc++_shared.so gets copied from the NDK sysroot into the Tauri
    gen/android jniLibs directory so the runtime linker can find it.
  - rustc-link-lib=log / OpenSLES emitted for Oboe's Android backends.

Deliberately NOT called from Rust yet — no extern "C" FFI declarations,
no oboe_audio.rs module, the `wzp_oboe_*` symbols from the static lib
are simply present but unreferenced.

Goal: isolate whether the Oboe C++ compile + static lib link alone
(with its libc++ dependency and log/OpenSLES bindings) regresses the
working baseline. If the build still launches and renders the home
screen, we know the C++ side is clean and the actual regression is
caused by calling into Oboe at runtime (next step).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 15:09:16 +04:00
Siavash Sameni
a852cad15e step D(android): compile cpp/getauxval_fix.c alongside hello.c
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Fourth incremental variable. Adds the getauxval_fix.c shim from the
legacy wzp-android crate (which has been shipping with it for months
without issue) to our cc::Build on Android. The file defines a single
getauxval() function that delegates to bionic's real runtime
implementation via dlsym — this is needed because rustc links
compiler-rt's broken static getauxval stub that SIGSEGVs in .so
libraries loaded via dlopen (reads __libc_auxv which is NULL).

Not imported from Rust. Goal: verify that adding a second C static
archive (and especially one that overrides a libc-ish symbol) doesn't
regress the working build.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 15:03:37 +04:00
Siavash Sameni
19fd3dd9cc step C fix: ungate wzp_proto imports used by resolve_quality() on Android
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Build #20 failed to compile on Android because I over-gated the
wzp_proto imports to non-Android. resolve_quality() is compiled on
every platform (it's outside the CallEngine impl) and references
QualityProfile + CodecId — both platform-independent types from
wzp_proto. Move those back to an unconditional import. tracing stays
gated (only the desktop start() body logs; the Android stub is silent).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 14:59:00 +04:00
Siavash Sameni
c69195fe06 step C(android): compile engine.rs on Android with a stub CallEngine::start
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Third incremental variable. Previously the engine module was cfg-gated
out of the Android build entirely (`#[cfg(not(target_os = "android"))]
mod engine;` in lib.rs). Now it's always compiled, so any link-time
effect of having engine.rs in the compilation unit can be measured
against the working baseline from build #19.

Changes kept deliberately small:
- lib.rs: drop the cfg gate on `mod engine;`. `use engine::CallEngine`
  stays gated because the Android-specific connect/disconnect/... stubs
  in lib.rs don't reference the type.
- engine.rs: the `wzp_client::{audio_io, call}` imports + CodecId +
  QualityProfile are gated to non-Android (they require the `audio`
  feature on wzp-client which Android doesn't pull in). On Android we
  keep only the MediaTransport import for transport.close(). The impl
  block now has two `start()` methods: the full CPAL-backed one for
  desktop, and a 6-line Android stub that returns `Err("audio engine
  not yet wired on Android")` so attempts to `connect` from the UI
  fail cleanly.

Goal: verify that linking in the compiled engine module (plus the
types it references) on Android doesn't regress the working baseline.
Home screen should still render and register_signal should still work.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 14:56:02 +04:00
Siavash Sameni
ae4f366b05 step B(android): depend on wzp-client with default-features=false
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Second incremental variable on the path to Oboe. Adds a
`[target.'cfg(target_os = "android")'.dependencies]` block that pulls
in wzp-client with NO features enabled — no audio (no CPAL), no vpio
(no VoiceProcessingIO). This gives the Android build access to
wzp-client's platform-independent modules (call, handshake, audio_ring,
codec wiring) without any system audio bindings.

Deliberately no new imports in lib.rs or engine.rs. The only effect
should be: cargo-tauri on Android now has to compile wzp-client and
all its transitive crates (wzp-codec, wzp-fec, wzp-proto, wzp-crypto
already pulled directly; now also audiopus, raptorq, etc.) and link
them into libwzp_desktop_lib.so.

Goal: verify that merely expanding the compiled code set to include
wzp-client doesn't regress the previous working state. If it does, we
know one of wzp-client's transitive deps is the problem — probably a
C dep like audiopus or codec2.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 14:49:49 +04:00
Siavash Sameni
f96d7ce3e1 step A(android): add cc=1 build-dep + compile single trivial hello.c
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First incremental variable on the path back to Oboe integration. Changes
are deliberately minimal: add cc = "1" to [build-dependencies] (cargo
build-deps resolve against the host so the line is unconditional), and
on the Android target run a single cc::Build step that compiles
cpp/hello.c — a 6-line file that defines one function (`wzp_hello_stub`)
that is never called from Rust.

Goal: verify that merely introducing a C static library into the .so
via cc::Build does not regress the working build (#17, commit 5309938
= build #6 behaviour: launches, renders home screen, registers on
relay). If this build still works, we know cc::Build pipelines alone
are fine and can move to the next variable.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 14:45:24 +04:00
Siavash Sameni
530993854f revert(android): roll back to build #6 (35642d1) — pre-oboe known-good state
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Spent 10+ builds chasing a __init_tcb+4 / pthread_create SIGSEGV after
adding the oboe audio backend. Every "fix" made things worse. Reverting
all Android-specific files to the state at 35642d1 (build #6), which
was the last commit where the Tauri Android app actually launched,
rendered the home screen, and successfully registered on a relay.

Reverted files (all back to their 35642d1 content):
  - desktop/src-tauri/Cargo.toml        (no build-dep cc, no tracing-android)
  - desktop/src-tauri/build.rs          (git hash only, no Oboe / cc build)
  - desktop/src-tauri/src/lib.rs        (engine cfg-gated on non-android)
  - desktop/src-tauri/src/main.rs       (two-line desktop entry)
  - desktop/src-tauri/src/engine.rs     (desktop-only audio setup)
  - scripts/Dockerfile.android-builder  (no android24→26 clang shim)
  - scripts/build-tauri-android.sh      (no linker env vars / manifest patch)

Deleted (were added between b314138 and e2e023d):
  - desktop/src-tauri/cpp/getauxval_fix.c
  - desktop/src-tauri/cpp/oboe_bridge.{h,cpp}
  - desktop/src-tauri/cpp/oboe_stub.cpp
  - desktop/src-tauri/src/oboe_audio.rs

Next: rebuild image on remote (to drop the baked-in clang shim), build
an APK, install on Pixel 6, verify the UI renders the same way build #6
did. From there we add features back ONE at a time so we can actually
bisect which one triggers the tao::ndk_glue crash. User's rule:
"if you want to change stack, change incrementally, so we can debug".

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 14:22:57 +04:00
Siavash Sameni
e2e023d2bc fix(android): drop pthread_shim — clang shim makes it unnecessary (and harmful)
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Once the Dockerfile rewrites every android24-clang to exec android26-clang,
the linker uses the API-26 NDK sysroot and libstd's pthread_create reference
resolves directly against libc.so's real runtime symbol — no interposition
needed.

The pthread_shim.c approach was actually fighting its own solution: our
shim's dlsym() call bound at link time to libdl.a's STUB dlsym (a
five-line function inside libdl_static.o that just returns NULL and sets
dlerror to "libdl.a is a stub --- use libdl.so instead"). NDK r19 and
glibc 2.34 both replaced libdl.a with empty stubs because dynamic loading
is now part of the main libc/bionic — so no amount of link-order
tinkering can make a static libdl.a dlsym actually work.

Remove pthread_shim.c, the cc::Build::new().file("cpp/pthread_shim.c")
step in build.rs, and the -Wl,--wrap=pthread_create rustc-link-arg. Keep
getauxval_fix.c because that one DOES work at link time (the symbol
override is for a function compiler-rt defines statically, not one that
would depend on the stub libdl.a/libc.a).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 13:52:53 +04:00
Siavash Sameni
5df9d418c9 fix(android): bake android24→26 clang shim into the docker image itself
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Build #13's PATH wrapper trick failed because tauri-cli invokes the linker
with an absolute path (/opt/android-sdk/ndk/.../bin/aarch64-linux-android24-
clang), which bypasses \$PATH entirely. The pthread_shim logs confirmed the
broken API-24 stubs were still being linked:

  WZP_pthread_shim: dlsym(RTLD_DEFAULT, pthread_create) returned NULL:
    libdl.a is a stub --- use libdl.so instead

Move the fix up a level — into the Dockerfile itself. On image build, for
each of the four android ABIs × {clang, clang++}, rename
`${abi}24-${suffix}` to `${abi}24-${suffix}.orig` and replace it with a
shell wrapper that exec()s `${abi}26-${suffix}`. Any call to the API-24
wrapper — via PATH, absolute path, or otherwise — now transparently runs
the API-26 wrapper, which uses the real libc.so/libdl.so bindings.

The old bash-c /tmp/wrappers workaround in build-tauri-android.sh is
removed now that the image handles it at the right layer.

Also add `--shell` to build-tauri-android.sh: opens an interactive docker
container on the remote with the same mounts/env as the build, so I can
iterate on cargo tauri android build / manually patch files / etc.
without the full git push → ssh → rebuild → install loop.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 13:33:10 +04:00
Siavash Sameni
2718402e96 fix(android): PATH wrapper to redirect tauri-cli's android24-clang → android26
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Build #12's instrumented pthread_shim gave us the definitive diagnosis:

  WZP_pthread_shim: dlsym(RTLD_DEFAULT, pthread_create) returned NULL:
    libdl.a is a stub --- use libdl.so instead

Tauri-cli invokes `aarch64-linux-android24-clang` as the linker and the
API-24 NDK sysroot ships *stub* libdl.a / libc.a: they compile fine but
every symbol crashes if called, because they're meant to coexist with a
separate dynamic .so that the dynamic linker provides at runtime. Rust's
pre-built libstd.rlib has static calls into those stubs baked in, so no
matter what we do at link time the broken code lands in the .so.

Env-var overrides of CARGO_TARGET_AARCH64_LINUX_ANDROID_LINKER don't
stick — tauri-cli resets them before invoking cargo. So instead of
fighting the env, we put a wrapper on $PATH, literally named
`aarch64-linux-android24-clang`, that exec()s the android26 version.
When tauri-cli looks up android24-clang via PATH, it gets our wrapper,
our wrapper runs android26-clang, and suddenly the whole build is using
the API-26 NDK sysroot with real dynamic bindings to libc.so / libdl.so.

Wrappers are installed for all four ABIs (aarch64, armv7, x86_64, i686)
× both suffixes (clang, clang++) directly inside the docker bash -c
preamble before any cargo invocation.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 13:23:47 +04:00
Siavash Sameni
1a8288c95f debug(android): instrument pthread_shim with logcat tracing + try RTLD_DEFAULT first
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Build #11 linked cleanly with --wrap=pthread_create but crashed at launch
on tao::ndk_glue::create with a Rust .expect() panic — meaning the shim's
__wrap_pthread_create successfully intercepted the call but returned
non-zero, triggering std::thread::spawn's Result::expect panic.

Add __android_log_print tracing so logcat shows exactly which resolver
path fired (RTLD_DEFAULT vs dlopen fallback) and what dlerror reports
when they fail. Also try RTLD_DEFAULT first — it's the simplest and
should find libc.so's pthread_create in the process's global symbol
table without any namespace games.
2026-04-09 13:15:47 +04:00
Siavash Sameni
f015be63ec fix(android): use --wrap=pthread_create instead of raw symbol override
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Build #10 failed with:
  ld.lld: error: duplicate symbol: pthread_create
    >>> defined at pthread_shim.c:30
    >>> ... in archive libpthread_shim.a
  (the other definition coming from libstd's bundled libc.a stub)

The raw-symbol-override approach was naive: when two static archives
both define the same symbol the linker refuses instead of picking one.

Switch to GNU-ld's `--wrap=pthread_create` mechanism:
  - All `pthread_create` references get rewritten to `__wrap_pthread_create`
  - Our shim now defines `__wrap_pthread_create` (no symbol clash)
  - Inside the shim we `dlopen("libc.so")` + `dlsym("pthread_create")` to
    get the real runtime symbol directly, bypassing BOTH the broken static
    stub (libstd's libc.a copy) AND libstd's own pthread_create path
  - `--real_pthread_create` is deliberately NOT used — it would alias the
    same broken stub the wrap exists to avoid

The wrap flag is emitted via `cargo:rustc-link-arg` in build.rs so it
only affects the Android target (the Android-branch of build.rs is the
only place that emits it).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 13:08:41 +04:00
Siavash Sameni
79e876126c fix(android): interpose pthread_create to bypass libstd's broken static stub
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Builds #7, #8 and #9 all crashed at launch with the same SIGSEGV inside
__init_tcb(bionic_tcb*, pthread_internal_t*)+4 called via pthread_create
from std::sys::thread::unix::Thread::new.

Digging further: the problem is NOT the final linker we pass to cargo.
It's that rustup ships a PRE-COMPILED libstd for aarch64-linux-android
which was built statically against an old NDK libc archive. That archive
has a pthread_create stub which calls a static __init_tcb stub that
assumes libc's static init path has set up the TCB — which never happens
in a .so loaded via dlopen. Bumping minSdk to 26 or forcing the
android26-clang linker (903a07c) doesn't rebuild libstd and therefore
doesn't fix the bundled broken stub.

The legacy wzp-android crate dodged this with a getauxval_fix.c shim that
interposes getauxval via RTLD_NEXT. The same trick works for pthread_create
here: define our own `int pthread_create(...)` in cpp/pthread_shim.c that
forwards to `dlsym(RTLD_NEXT, "pthread_create")` — the real, fully working
version exported from libc.so. The linker processes our static lib before
libstd.rlib, so libstd's unresolved pthread_create reference binds to our
symbol, and the broken libc.a stub inside libstd is never pulled in.

build.rs compiles cpp/pthread_shim.c right after cpp/getauxval_fix.c so
both symbol overrides are in place before any Rust code gets linked.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 13:04:18 +04:00
Siavash Sameni
903a07c1d4 fix(android): force API-26 NDK linker via docker env vars
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The previous commit bumped minSdk from 24 to 26 in build.gradle.kts
hoping tauri-cli would pick it up and use the android26-clang linker,
but the crash recurred at exactly the same frame (__init_tcb via
pthread_create via std::thread::spawn). That means tauri-cli is
ignoring the gradle minSdk value and sticking with its hardcoded
aarch64-linux-android24-clang.

The android24 linker resolves __init_tcb against the broken static
stub in libc.a (API 24 does NOT export __init_tcb as a dynamic symbol
from libc.so — it only exists in the static archive, and the stub
expects the TCB to be initialised by a running static init path,
which never happens in a dlopen-loaded .so).

Override the linker env vars directly in the docker run invocation
for all four ABIs. These take precedence over anything tauri-cli or
.cargo/config.toml might set.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 12:55:11 +04:00
Siavash Sameni
af20fa418a fix(android): bump minSdk 24 -> 26 to avoid broken __init_tcb in NDK 24 stub
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Build #7 crashed at launch on the Pixel 6 with SIGSEGV in
__init_tcb / pthread_create called from tao::ndk_glue::create in
WryActivity.onCreate:

  #00  __init_tcb(bionic_tcb*, pthread_internal_t*)+4
  #01  pthread_create+360
  #02  std::sys::thread::unix::Thread::new
  #04  tao::platform_impl::platform::ndk_glue::create
  #05  Java_com_wzp_desktop_WryActivity_create

Tauri scaffolds build.gradle.kts with `minSdk = 24`, which makes the
tauri-cli invoke `aarch64-linux-android24-clang` as the Rust linker. That
linker transitively pulls broken static stubs from libc.a for getauxval,
__init_tcb and pthread_create — these stubs only work in statically-
linked executables because they read bionic state (__libc_auxv, TCB) that
only the libc init path sets up. In a .so loaded via dlopen they SIGSEGV
the moment anything spawns a thread.

API 26+ has the real runtime symbols and the NDK-26 linker resolves them
against libc.so instead of the static fallback. This is also the minimum
Oboe supports. Patch the generated build.gradle.kts post-init to swap
`minSdk = 24` for `minSdk = 26` — the legacy wzp-android crate solved
the same issue with a .cargo/config.toml linker override plus a
getauxval_fix.c shim.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 12:47:36 +04:00
Siavash Sameni
b314138caf feat(android): oboe/AAudio audio backend + runtime mic permission (step 3)
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This is the big one — the Tauri Android app now has a real audio stack
capable of full-duplex VoIP, reusing the proven C++ Oboe bridge from the
legacy wzp-android crate.

Architecture:
- desktop/src-tauri/cpp/  — copies of oboe_bridge.{h,cpp}, oboe_stub.cpp,
  and getauxval_fix.c from crates/wzp-android/cpp/. build.rs clones
  google/oboe@1.8.1 into OUT_DIR and compiles the bridge + all Oboe
  sources as "oboe_bridge" static lib, linking against shared libc++
  (static would pull broken libc stubs that SIGSEGV in .so libraries).
- src/oboe_audio.rs  — Rust side: an SPSC ring buffer matching the C++
  bridge's AtomicI32 layout, plus OboeHandle::start() which returns
  (capture_ring, playout_ring, owning_handle). The ring exposes the same
  (available / read / write) methods as wzp_client::audio_ring::AudioRing
  so CallEngine treats both backends interchangeably.
- src/engine.rs  — compiled on every platform now. A cfg-switched type
  alias picks wzp_client::audio_ring::AudioRing on desktop and
  crate::oboe_audio::AudioRing on Android. The audio setup block has
  three branches: VPIO/CPAL on macOS, CPAL on Linux/Windows, Oboe on
  Android. Send/recv tasks are identical across platforms.
- src/lib.rs  — removes all the "step 3 not done" Android stubs. The
  engine module is no longer cfg-gated; connect / disconnect / toggle_mic
  / toggle_speaker / get_status are single implementations used by both
  desktop and Android. Identity path resolves via app.path().app_data_dir()
  from the Tauri setup() callback (already wired in step 1).

Runtime mic permission:
- scripts/build-tauri-android.sh now injects RECORD_AUDIO + MODIFY_AUDIO_
  SETTINGS into gen/android/app/src/main/AndroidManifest.xml after init,
  and overwrites MainActivity.kt with a version that calls
  ActivityCompat.requestPermissions in onCreate. This is idempotent:
  every build re-applies the patches so tauri re-init can't regress them.

Cargo.toml:
- cc is now an unconditional build-dep (build.rs runs on the host, so
  target-gating build-deps doesn't work).
- wzp-client is now a dep on every platform. On Android it gets default
  features only (no "audio"/"vpio") so CPAL isn't dragged in — oboe_audio
  provides the capture/playout rings instead.
- tracing-android is added on Android so tracing events flow into logcat.

build.rs also gained embedded git hash (WZP_GIT_HASH) capture, which is
shown under the fingerprint on the home screen — already committed in
7639aaf, reinstated here alongside the Oboe build logic.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 12:40:38 +04:00
Siavash Sameni
35642d1c54 feat(desktop): bake local Laptop relay into default relay list for testing
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Adds 172.16.81.125:4433 (the laptop's LAN IP) as the first default relay
so the Android rewrite can be tested against a relay whose logs are on the
same host as the builds and screenshots. On fresh installs the Laptop
relay is pre-selected as index 0. On upgrades from an older cached
settings blob, a one-shot migration unshifts it to the front if missing,
so we don't have to tap through Manage Relays after every reinstall.

Marked "remove once Android rewrite is stable" — the address is a hardcoded
LAN IP that won't be valid in other environments.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 12:19:46 +04:00
Siavash Sameni
6b8107504e fix(desktop): tauri capability for android event listeners + persistent debug keystore
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Two related Android-only papercuts found while testing build #4 on a Pixel 6:

1. Frontend was crashing in the WebView with:
       Tauri/Console: Uncaught (in promise) event.listen not allowed.
       Permissions associated with this command: core:event:allow-listen,
       core:event:default
   The desktop build worked fine because Tauri's default capability set
   covers the desktop side. On Android (and iOS) Tauri 2.x is much stricter
   about ACL — without an explicit capabilities/default.json that lists
   "android" in its platforms, the WebView gets zero permissions. Add a
   default capability granting core:default + the event listener perms
   across all five platforms (linux/macOS/windows/android/iOS).

2. Every fresh docker run produced a new ~/.android/debug.keystore, so
   `adb install -r` of a freshly built APK over an already-installed one
   failed with INSTALL_FAILED_UPDATE_INCOMPATIBLE. Mount a persistent host
   volume at /home/builder/.android in build-tauri-android.sh so the same
   debug keystore is reused across builds and `install -r` keeps working.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 12:02:01 +04:00
Siavash Sameni
7639aaf08d feat(desktop): deterministic alias from seed + git hash on home screen + fix EACCES on Android
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Three home-screen issues from the first Tauri Android APK:

1. Alias was empty (no seed-derived name).
   Port the adjective+noun word lists from the old Kotlin SettingsRepository
   into a `derive_alias()` helper that maps the first 4 bytes of the seed to
   indices in those lists. Same seed → same alias forever, different seeds →
   effectively random aliases — so reinstalls keep the user's identity AND
   the friendly name they're used to.

2. Build identity was invisible — couldn't tell which APK was actually
   installed (this caused us a lot of grief on the Kotlin app).
   build.rs now captures `git rev-parse --short HEAD` and emits it as
   `WZP_GIT_HASH`, exposed via a new `get_app_info` command. The frontend
   stamps `build <hash> • <alias>` under the fingerprint on the home screen.

3. Register on relay failed with `Permission denied (os error 13)`.
   Root cause: I hardcoded `/data/data/com.wzp.phone/files/.wzp` as the
   identity dir, but the Tauri Android package id is `com.wzp.desktop` —
   so the app was trying to write into another app's data directory and
   getting EACCES at the filesystem layer. Fix: resolve the data dir from
   Tauri's `path().app_data_dir()` API in the `setup()` callback and stash
   it in a `OnceLock<PathBuf>`. Works on Android, macOS, Linux, Windows
   without any cfg gymnastics.

Also: `get_app_info` returns the resolved `data_dir` so we can debug
storage issues from the UI (it's set as the build-hash element's title).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 11:55:51 +04:00
Siavash Sameni
69ee3115b6 build: tauri-android docker pipeline + ntfy notifications
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Dockerfile.android-builder: install Android API 36 platform + build-tools
35.0.0 alongside the existing API 34 set. Tauri 2.x mobile defaults to
compileSdk 36 / build-tools 35; without these the gradle build fails with
"SDK directory is not writable" because the read-only /opt/android-sdk
volume can't grow at build time. Adding Node.js 20, all four Rust android
targets, and tauri-cli 2.x was already in place.

scripts/build-tauri-android.sh: new build wrapper for the desktop/ Tauri
project (parallel to scripts/build-and-notify.sh which targets the legacy
android/ Kotlin app). Pulls the branch on remote, runs cargo tauri android
build inside the docker image, and sends three ntfy.sh/wzp notifications
that all carry the short git hash:
  - STARTED [hash] — <commit subject>
  - OK [hash] (size) — <rustypaste apk url>
  - FAILED [hash] (line N) — <rustypaste log url>
On failure the full /tmp/wzp-tauri-build.log is uploaded to rustypaste so
the URL in the failure ntfy is directly downloadable, same place as the
APK.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 11:25:54 +04:00
Siavash Sameni
e6f77a78a7 feat(desktop): split main.rs into lib.rs for Tauri Mobile (Android/iOS)
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Tauri 2.x Mobile links the app as a cdylib loaded from a Java Activity, so
all of the Builder/command code has to live in a library crate. Move the
existing logic verbatim into src/lib.rs::run() and reduce src/main.rs to a
two-line desktop entry point that calls into it.

Cargo.toml gets a [lib] section (crate-types: staticlib + cdylib + rlib,
named wzp_desktop_lib) and the wzp-client dependency — which pulls CPAL +
VoiceProcessingIO — is moved behind cfg(not(target_os = "android")) so the
Android cdylib doesn't need an audio backend yet. Engine-backed Tauri
commands (connect/disconnect/toggle_mic/toggle_speaker/get_status) get
Android stubs that return clear "not yet wired" errors. The signaling
commands (register_signal/place_call/answer_call/get_signal_status/
ping_relay/get_identity) are platform-independent and unchanged.

Also: get_identity / register_signal now auto-create the seed if missing
instead of erroring with "connect to a room first", and the identity dir
resolves to /data/data/com.wzp.phone/files/.wzp on Android (proper
app-internal storage) vs \$HOME/.wzp on desktop.

Side note: src/main.rs was previously untracked — desktop builds were
working only because it existed in the local worktree. This commit fixes
that too.

Step 1 of the Android rewrite plan (tauri-mobile scaffold). No audio yet.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 11:17:55 +04:00
Siavash Sameni
04a985912a fix: add direct calling Tauri backend commands (register_signal, place_call, answer_call)
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2026-04-09 06:59:16 +04:00
Siavash Sameni
2288c1ae07 feat: direct calling UI for desktop Tauri app + merge android branch
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Tauri backend:
- register_signal: persistent _signal connection, presence registration
- place_call: send DirectCallOffer by fingerprint
- answer_call: accept/reject incoming calls
- get_signal_status: poll signal state

Frontend:
- Mode toggle: "Room" vs "Direct Call"
- Register button → registers on relay signal channel
- Incoming call panel with Accept/Reject
- Fingerprint input + Call button
- Auto-connect to media room on CallSetup event

Also merges feat/android-voip-client into desktop branch:
- Federation fixes, time-based dedup, FEC stale blocks
- Direct calling protocol types
- ACL + SAS verification

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 06:42:47 +04:00
Siavash Sameni
0d3f0d4dcb feat: Android UI for direct 1:1 calling
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- Mode toggle: "Room" vs "Direct Call" tabs on pre-connection screen
- Direct Call mode: Register button → registers on relay signal channel
- After registration: shows fingerprint dial pad + incoming call panel
- Incoming call: green Accept / red Reject buttons with caller info
- Ringing state display while waiting for callee
- CallSetup auto-connects to media room
- CallStats extended: sas_code, incoming_call_id/fp/alias fields
- CallViewModel: registerForCalls(), placeDirectCall(), answerIncomingCall()

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 06:18:07 +04:00
Siavash Sameni
c184d5e1f3 fix: build scripts use fetch+reset instead of pull to avoid ref lock errors
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git pull fails when refs are stale from concurrent builds. Switch to
git gc + git fetch + git reset --hard origin/branch for robustness.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 06:07:10 +04:00
Siavash Sameni
5d8e743cbf feat: Android engine + Kotlin API for direct 1:1 calling
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Rust engine:
- start_signaling(): persistent _signal connection, presence registration
- Signal recv loop: handles DirectCallOffer, CallRinging, CallSetup, Hangup
- New CallState variants: Registered, Ringing, IncomingCall
- Stats expose incoming_call_id, incoming_caller_fp, incoming_caller_alias, sas_code
- New EngineCommands: PlaceCall, AnswerCall, RejectCall

JNI bridge:
- nativeStartSignaling(relay, seed, token, alias)
- nativePlaceCall(targetFp)
- nativeAnswerCall(callId, mode)

Kotlin API (WzpEngine.kt):
- startSignaling(relay, seed, token, alias)
- placeCall(targetFingerprint)
- answerCall(callId, mode) — 0=Reject, 1=AcceptTrusted, 2=AcceptGeneric

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 06:02:48 +04:00
Siavash Sameni
6694aebfd9 fix: resolve 0.0.0.0 to connectable address in CallSetup relay_addr
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When relay listens on 0.0.0.0, derive the actual IP from the client's
connection address for the CallSetup message.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 05:56:19 +04:00
Siavash Sameni
d27e85ecf2 feat: SAS (Short Authentication String) for call identity verification
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Derive a 4-digit code from the shared DH secret via HKDF with label
"warzone-sas-code". Both peers compute the same code; a MITM relay
produces a different one. Users compare verbally during the call.

- CryptoSession::sas_code() -> Option<u32> on the trait
- ChaChaSession stores and returns the SAS
- HKDF derivation in WarzoneKeyExchange::derive_session()
- Tests: both peers match, MITM produces different code

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 05:48:08 +04:00
Siavash Sameni
39ac181d63 feat: ACL + capacity limit on call rooms, unified fingerprint format
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- Call rooms (call-*) restricted to the two authorized participants only
- Room capacity enforced at 2 for call rooms
- Unauthorized clients get immediate connection close
- Unified fingerprint format: SHA-256(Ed25519 pub)[:16] as xxxx:xxxx:...
  Used consistently in signal registration, handshake, and ACL checks

Tested: Alice+Bob authorized, attacker rejected with "not authorized"

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 05:43:03 +04:00
Siavash Sameni
3351cb6473 feat: direct 1:1 calling via relay signaling (Phase 1)
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New feature: call someone directly by fingerprint through the relay.

- Client connects with SNI "_signal" for persistent signaling
- RegisterPresence/RegisterPresenceAck for relay registration
- DirectCallOffer routed to target by fingerprint
- DirectCallAnswer with AcceptGeneric/AcceptTrusted/Reject modes
- Relay creates private room (call-{id}), sends CallSetup to both
- Both clients connect to private room for media (existing SFU path)
- Hangup forwarding + cleanup on disconnect
- Desktop CLI: --signal + --call <fingerprint> for testing
- CallRegistry tracks call state (Pending/Ringing/Active/Ended)
- SignalHub manages persistent signaling connections

Tested: Alice calls Bob by fingerprint, relay routes offer, Bob
auto-accepts, both join private room, media flows bidirectionally.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 05:35:16 +04:00
Siavash Sameni
54a4d91f3e docs: add --event-log, --version-check, and federation troubleshooting to admin guide
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Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-09 04:43:37 +04:00
Siavash Sameni
3b962bd4cb fix: build scripts use git reset --hard before pull to recover from dirty state
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Cargo.lock changes from Docker builds caused pull conflicts. Now uses
reset --hard + clean -fd to guarantee clean state before pulling.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 22:13:26 +04:00
Siavash Sameni
1118eac752 fix: re-enable FEC + time-based dedup for federation
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Restore fec_ratio=0.2 on GOOD profile. Time-based dedup (2s TTL) with
payload hash prevents consecutive sender collisions while still catching
multi-path duplicates. Verified: 6 consecutive senders across 2 relays,
0 decode errors, 0 drops, FEC active.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 22:09:15 +04:00
Siavash Sameni
f935bd69cd fix: rewrite seq/fec for federation-delivered packets
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- Time-based dedup (2s TTL) replaces fixed-window dedup — consecutive
  senders with same seq numbers no longer collide
- Raw byte forwarding for federation local delivery (no re-serialization)
- Jitter buffer resets on large backward seq jumps (>100)
- recv_media skips malformed datagrams instead of returning connection-closed
- SIGTERM handler for clean QUIC shutdown on wzp-client
- JSONL event log infrastructure (--event-log flag) for protocol analysis
- FEC disabled on GOOD profile for federation debugging (fec_ratio=0.0)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 21:55:06 +04:00
Siavash Sameni
1c684f6b47 fix: rewrite seq/fec for federation-delivered packets
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Federation media from different senders had conflicting seq numbers,
FEC block IDs, and Opus decoder state. The relay now assigns fresh
monotonic seq/fec_block/fec_symbol to all federation-delivered packets,
ensuring clients see a clean continuous stream regardless of sender changes.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 15:48:55 +04:00
Siavash Sameni
c92db7e9b7 fix: preserve original relay label through multi-hop presence propagation
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When propagating GlobalRoomActive to other peers, use tagged participants
(with relay_label set to the originating relay) instead of the raw
untagged participants. This shows "Relay C" instead of "Relay B" when
C's participants are forwarded through hub B to A.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 15:34:22 +04:00
Siavash Sameni
c3bd657224 fix: FEC decoder resets stale blocks — fixes consecutive federation connects
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When a new sender reuses the same block_id values as a previous sender,
the FEC decoder was silently dropping all data because blocks were marked
as "already decoded". Now blocks older than 2 seconds are automatically
reset when new data arrives for them.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 15:26:00 +04:00
Siavash Sameni
8b79cdc6fc fix: dedup filter collision between different senders + build scripts default --pull
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- Dedup key now includes source peer fingerprint hash, preventing
  packets from different senders with same room+seq from being dropped
  as duplicates (was silently killing all multi-hop audio)
- Build scripts default to --pull (use --no-pull to skip)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 15:18:52 +04:00
Siavash Sameni
2eab56beec fix: federation presence dedup, stale cleanup, and Android SIGSEGV crash
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- Deduplicate remote participants by fingerprint in all merge sites
  (canonical == raw room name caused double-lookup, doubling every remote participant)
- GlobalRoomInactive now propagates updated participant list to other peers
  (hub relay B was not informing A when C's participants left)
- Add 15-second stale presence sweeper that purges remote participants
  from peers that stop sending data (safety net for QUIC timeout delays)
- Add @Synchronized to WzpEngine.getStats/stopCall/destroy to prevent
  TOCTOU race between stats polling coroutine and engine teardown (SIGSEGV)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 15:07:59 +04:00
Siavash Sameni
7dadc1ddd6 fix: default room 'general', cap auto codec at 24k
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- Android default room changed from 'android' to 'general'
- Relay choose_profile capped at GOOD (Opus 24k) — studio tiers
  (32k/48k/64k) cause high packet loss on federation paths due to
  larger datagrams exceeding path MTU. Will re-enable after MTU
  discovery is implemented.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 14:41:12 +04:00
Siavash Sameni
be0441295a fix: read git hash outside Docker for Linux build ntfy notification
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The hash was read inside Docker (/build/source) where .git doesn't
exist. Now reads from $BASE_DIR/data/source before Docker runs.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 14:32:03 +04:00
Siavash Sameni
b9f4e7f102 feat: include git hash in ntfy build notifications + MTU PRD
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ntfy messages now show: "WZP Linux [abc1234] ready!" and
"WZP Android [abc1234] done! APK: url" so you can verify which
commit was built without checking relay version remotely.

Also added PRD-mtu-discovery.md for QUIC Path MTU Discovery.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 14:26:13 +04:00
Siavash Sameni
28f4a0fb6f fix: multi-hop presence — propagate remote rooms on new peer connect
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When a new federation link is established, announce not only LOCAL
global rooms but also rooms from OTHER peers (remote_participants).
This fixes multi-hop: when R2 connects to R3, R2 tells R3 about
R1's rooms that R2 learned about earlier.

Previously, only local rooms were announced on link setup. If R1
had a client but R2 had no clients, R2 wouldn't tell R3 about R1.

Also added diagnostic logging for room announcements on link setup.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 13:43:15 +04:00
Siavash Sameni
3d76acf528 fix: multi-hop federation — hub relay forwards without local participants
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Three fixes for 3-relay chain (R1→R2→R3):

1. Room lookup in handle_datagram: hub relay (R2) has no local
   participants, so active_rooms() was empty and datagrams were
   silently dropped. Now also checks global_rooms config directly,
   allowing hub relays to forward without local clients.

2. Multi-hop forwarding: removed active_rooms filter — forward to
   ALL connected peers except source. The receiving peer decides
   whether to deliver or forward further.

3. Android relay_label: native RoomMember now includes relay_label
   from RoomUpdate signal. Kotlin UI reads it for relay grouping.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 13:33:44 +04:00
Siavash Sameni
f4b5996bdf feat: Android relay-grouped participant list matching desktop
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Participants now grouped by relay on Android:
- Green dot + "THIS RELAY" for local participants
- Blue dot + relay label for federated participants

Added relayLabel to RoomMember data class, parsed from
relay_label JSON field. UI groups and renders with headers.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 13:15:12 +04:00
Siavash Sameni
fc721c4217 fix: clear stale federated presence on GlobalRoomInactive
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When a remote relay's room goes inactive (all participants left),
the receiving relay now:
1. Clears remote_participants for that peer+room
2. Broadcasts updated RoomUpdate to local clients with the remote
   participant removed
3. Updates federation_active_rooms metric

Previously, remote participants lingered in the participant list
after disconnect, causing ghost entries and stale media forwarding.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 13:06:48 +04:00
Siavash Sameni
5c24adf1c1 feat: remote version query — wzp-client --version-check <relay>
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Connects to a relay over QUIC with SNI "version", reads build hash
from a unidirectional stream, prints "<relay> <git-hash>" and exits.

Usage: wzp-client --version-check 172.16.81.175:4434
Output: 172.16.81.175:4434 8dbda3e

Relay side: detects "version" SNI, opens uni stream, writes
BUILD_GIT_HASH, waits 100ms for client to read, closes.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 12:47:37 +04:00
Siavash Sameni
8dbda3e052 feat: --version flag with git hash + test script kill fix
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wzp-relay --version prints "wzp-relay <short-git-hash>".
Build hash also logged on startup: version=abc1234.
Enables verifying deployed relay matches expected build.

Also fixed federation-test.sh: use kill -INT (not SIGTERM) so
clients save recordings before exit. Added save delay.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 12:36:33 +04:00
Siavash Sameni
c8a3aaacb6 feat: comprehensive federation test harness
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7 test scenarios across 3 relays:
1. Basic 2-relay audio (A→B)
2. Reverse direction (B→A)
3. 3-relay chain (A→B→C)
4. File playback (60s test audio)
5. Reconnection (join/leave/rejoin)
6. Multi-participant (3 users on 3 relays)
7. Simultaneous senders (2 senders, 1 recorder)

Usage: ./scripts/federation-test.sh <relay1> <relay2> <relay3>

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 12:19:15 +04:00
Siavash Sameni
395a0c557e feat: TX/RX codec badges on desktop call screen
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Desktop now shows codec badges like Android:
- Green TX badge: e.g. "Opus64k"
- Blue RX badge: e.g. "Opus24k"
Displayed in the stats line below the call controls.

Engine tracks tx_codec (set on encoder init) and rx_codec (updated
from incoming packet headers). Passed through EngineStatus → CallStatus
→ frontend.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 12:03:20 +04:00
Siavash Sameni
54cb6c3b71 feat: relay_label in RoomParticipant + tagged remote participants
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RoomParticipant.relay_label identifies which relay a participant is
connected to. Local participants have None, federated participants
get tagged with the peer relay's label when storing remote_participants.

This enables clients to group participants by relay in the UI.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 11:22:53 +04:00
Siavash Sameni
da593f9510 feat: relay-grouped participant rendering + relay_label in protocol
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RoomParticipant now has optional relay_label field. Desktop client
groups participants by relay: "This Relay" (green dot) for local,
peer label (blue dot) for federated. Shows all relays in the chain
including intermediate ones.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 11:22:05 +04:00
Siavash Sameni
a3ebf5616f fix: unified raw room names + merged presence on join
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1. CLI client now sends raw room names (no hash), matching Android
   JNI and Desktop Tauri. All three clients are now consistent.

2. When a client joins a global room, the relay merges federated
   remote participants into the initial RoomUpdate. Previously,
   clients that joined after the GlobalRoomActive signal only saw
   local participants. Now they see everyone immediately.

3. Added get_remote_participants() to FederationManager for querying
   cached remote participants from all peer links.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 11:09:15 +04:00
Siavash Sameni
ff6d0444c0 feat: federation Prometheus metrics — peer status, packets, active rooms
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Wires up the existing RelayMetrics federation fields:
- wzp_federation_peer_status{peer} — 1=connected, 0=disconnected
- wzp_federation_packets_forwarded_total{peer,direction} — in/out counts
- wzp_federation_active_rooms — number of active federated rooms

These are critical for monitoring federation health and will feed into
the adaptive codec selection system (PRD-coordinated-codec.md).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 11:00:13 +04:00
Siavash Sameni
8080713098 feat: federated presence — RoomUpdate includes remote participants
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GlobalRoomActive signal now carries participant list from the
announcing relay. When received, the relay:
1. Stores remote participants per peer link
2. Broadcasts merged RoomUpdate to local clients (local + all remote)

This means clients on different relays can now SEE each other in the
participant list. Also fixes build: removed non-existent metric field
references that were added by linter.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 10:52:27 +04:00
Siavash Sameni
e813362395 feat: federation metrics + dedup + rate limiting
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Add Prometheus metrics for federation links (per-peer RTT, packet
counters, active rooms gauge, dedup/rate-limit drop counters).

Add dedup filter (4096-entry ring buffer) to drop duplicate packets
arriving via multiple federation paths. Add per-room token bucket
rate limiter (500 pps) to prevent amplification.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 10:36:26 +04:00
Siavash Sameni
d52b8befd6 fix: canonical room hash for federation — handles hashed vs raw room names
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Different clients send different room names:
- Android: raw "general" as SNI
- Desktop: hash_room_name("general") = "f09ae11d..." as SNI

Federation datagrams are tagged with an 8-byte room hash. Previously,
each relay computed the hash from the client-provided room name,
causing mismatches between relays with different client types.

Fix: resolve_global_room() maps any room name (raw or hashed) to the
canonical [[global_rooms]] name. global_room_hash() always uses the
canonical name for federation hashing. handle_datagram uses both raw
and canonical hash matching to find the local room.

Also: run_participant now receives the pre-computed federation_room_hash
so the egress uses the canonical hash, not the client-specific name.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 10:31:26 +04:00
Siavash Sameni
0abecf7fd8 feat: adaptive quality engine + codec indicator UI
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Wire AdaptiveQualityController into Android engine for auto codec
switching based on network quality reports. Add color-coded TX/RX
codec badges to the in-call screen showing active codecs and Auto mode.

- Recv task: ingest QualityReports, feed to controller, signal profile
  changes via AtomicU8 to send task
- Send task: check for pending profile switch at frame boundaries,
  update encoder/FEC/frame size
- Track peer codec from incoming packet headers
- Kotlin UI: codec badges (blue=studio, green=good, amber=degraded,
  red=catastrophic) with Auto tag
- Add .taskmaster to .gitignore

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 10:19:11 +04:00
Siavash Sameni
f4cc3b1a6b fix: forward media to ALL connected peers, not just those with room active
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The bug: when a local client joins a global room and sends media, the
egress task checked peer_links.active_rooms to decide where to forward.
But active_rooms tracks what PEERS announced (their rooms), not what
WE announced. So our own GlobalRoomActive signal went out but our
peer_links had empty active_rooms — media was dropped.

Fix: for locally-originated media, send to ALL connected federation
peers unconditionally. The receiving relay decides whether to deliver
to local participants (if it has the room) or forward further. This
is correct because federation peers are explicitly configured — if
they're connected, they should receive global room media.

Multi-hop forwarding (handle_datagram) still filters by active_rooms
to prevent loops — only forwards to peers that announced the room.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 10:09:50 +04:00
Siavash Sameni
af4c89f5f0 docs: PRD for delegated trust in relay federation
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Addresses the trust gap where a hub relay can forward media from
unknown relays without the receiving relay's consent. Introduces
delegate=true flag on [[trusted]] entries: when set, the relay
accepts media forwarded through the trusted peer from relays it
vouches for. Without delegate, only direct media is accepted.

Covers: FederationTrustChain signal, origin authorization checks,
TTL for chain depth limiting, anti-spam properties. 5 phases, ~3 days.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 10:00:21 +04:00
Siavash Sameni
406461d460 feat: personalized config generation with --listen addr + own fingerprint
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When --config points to a non-existent file, the relay now generates
a personalized example config that includes:
- listen_addr matching the --listen flag (not hardcoded 0.0.0.0:4433)
- Pre-filled [[peers]] section with this relay's detected IP, port,
  and TLS fingerprint — ready to copy/paste into other relay configs

This makes setting up federation much easier: start each relay, it
generates its config with its own peering info commented out, you
just uncomment and copy between configs.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 09:38:28 +04:00
Siavash Sameni
7064f484af feat: -c/--config and -i/--identity flags for multi-instance relay
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Enables running multiple relays on the same machine:
  wzp-relay -c ~/.wzp1/config.toml -i ~/.wzp1/relay-identity --listen :4433
  wzp-relay -c ~/.wzp2/config.toml -i ~/.wzp2/relay-identity --listen :4434
  wzp-relay -c ~/.wzp3/config.toml -i ~/.wzp3/relay-identity --listen :4435

Config auto-creation: if the config file doesn't exist, writes an
example config with all fields documented and commented. The relay
starts with defaults but the file is ready to edit.

Identity auto-generation: if the identity file doesn't exist, generates
a new random seed (OsRng via wzp_crypto::Seed::generate) and saves it.
Subsequent starts load the same identity.

Short flags: -c for --config, -i for --identity.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 09:18:48 +04:00
Siavash Sameni
1d2222a25a debug: add datagram receive + multi-hop forward error logging
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Added logging to trace federation media flow:
- media_task logs first + every 250th received datagram (count, len)
- handle_datagram multi-hop forward logs errors (was silently dropped)
- forward_to_peers logs when no peer matches

2-relay (A→B): WORKING — full audio received, 300 packets forwarded
3-relay (A→B→C): B receives datagrams from A but only 1 arrives —
  remaining packets not received, likely a QUIC read_datagram issue
  when handle_datagram holds locks during processing. Needs further
  investigation into async lock contention or datagram buffering.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 08:45:54 +04:00
Siavash Sameni
270e139f20 feat: federation media forwarding WORKING — global rooms router model complete
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2-relay test: 5.0s audio, RMS 4748, PASS. Full pipeline verified:
- Room correctly identified as global (hash matching works)
- Federation egress channel created and connected
- GlobalRoomActive signals exchanged between peers
- 300 packets (250 source + 50 FEC) forwarded via tagged datagrams
- Client B on relay B received full 5-second tone from client A on relay A

Added debug logging: is_global check, egress channel creation, per-peer
forwarding with active_rooms diagnostic when no match found. Also logs
egress packet count (first + every 250th).

Multi-hop propagation: GlobalRoomActive signals forwarded to other peers
so A→B→C chain knows about rooms across the full mesh.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 08:31:37 +04:00
Siavash Sameni
d9b2e0fd53 docs: comprehensive documentation — design, architecture, admin, user guide
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4 files, 2,511 lines covering the entire WarzonePhone project:

DESIGN.md (591 lines): system overview, codec system (9 variants),
FEC (RaptorQ), transport (QUIC/quinn), security (Ed25519/X25519/
ChaCha20/HKDF/BIP39/TOFU), federation (global rooms), jitter buffer.
Mermaid diagrams for audio pipelines and crate dependencies.

ARCHITECTURE.md (874 lines): 15 mermaid diagrams — system overview,
encode/decode pipelines, relay SFU, federation topology/protocol,
signal handshake, client architectures (desktop/android/CLI), wire
format tables (MediaHeader/MiniHeader/QualityReport), project tree.

ADMINISTRATION.md (587 lines): relay deployment (binary/Docker/systemd),
complete TOML config reference, CLI flags table, federation setup
(peers/trusted/global_rooms), 3 example configs, Prometheus metrics,
auth, identity persistence, 12-item troubleshooting guide.

USER_GUIDE.md (459 lines): all clients — desktop (settings, quality
slider, key warning, shortcuts), Android (8-level quality slider,
server management, identity backup), CLI (flags table, 8 usage
patterns). Identity system, quality profiles when-to-use guide.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 08:21:13 +04:00
Siavash Sameni
898c1ea32b docs: PRDs for P2P direct calls and coordinated codec switching
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PRD-p2p-direct.md: STUN-based NAT traversal for direct QUIC
connections between clients. True E2E with mutual TLS cert pinning
via identity fingerprints. Hybrid mode: try P2P, fall back to relay.
4 phases: STUN discovery, hole punching, P2P adaptive quality,
seamless relay-to-P2P migration.

PRD-coordinated-codec.md: Relay acts as quality judge — monitors
per-participant loss/RTT/jitter, sends quality directives. Downgrade
is immediate (match weakest link), upgrade is consensual (all
participants must agree, synchronized switch at agreed timestamp).
Covers asymmetric encoding in SFU and P2P→relay backporting strategy.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 08:12:12 +04:00
Siavash Sameni
b00db5dfdc feat: federation rewrite — global rooms router model
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Major rewrite of relay federation replacing virtual participants with
a clean router model:

1. Global rooms: [[global_rooms]] in TOML config declares rooms that
   are bridged across federation. Each relay is a router + local SFU.

2. Room events: RoomManager emits LocalJoin/LocalLeave via broadcast
   channel when rooms transition between empty and non-empty.

3. GlobalRoomActive/Inactive signals: relays announce when they have
   local participants in global rooms. Peers track active state and
   forward media accordingly. Announcements propagate for multi-hop.

4. Media forwarding: separated from SFU loop. Local participant sends
   via mpsc channel → egress task → forward_to_peers() → room-hash
   tagged datagrams to active peer links. Inbound datagrams delivered
   to local participants + forwarded to other active peers (multi-hop).

5. Loop prevention: don't forward back to source relay.

6. Room name hashing: is_global_room() checks both plain name and
   hash (clients hash room names for SNI privacy).

Removed: ParticipantSender::Federation, federated_participants, virtual
participant join/leave, periodic room polling. Rooms now only contain
local participants.

Signaling tested: 3-relay chain (A→B←C) correctly propagates
GlobalRoomActive through B to both A and C. Media forwarding plumbing
in place but needs final debugging.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 07:54:38 +04:00
Siavash Sameni
bc8bb3d790 feat: [[trusted]] config + FederationHello for one-sided federation
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- Added [[trusted]] config: relay B can accept inbound federation
  from relay A by fingerprint alone, without knowing A's address.
  A connects to B with [[peers]], B trusts A with [[trusted]].

- FederationHello signal: outbound connections send their TLS
  fingerprint as first signal. The accepting relay verifies it
  against [[peers]] (by IP) or [[trusted]] (by fingerprint).

- Tested 3-relay chain: A→B←C. Both A and C connect to B, B trusts
  both. B correctly accepts both inbound connections. Room
  announcements flow A→B and C→B.

- Remaining: B needs to announce rooms back to A and C on the same
  connection so media can flow A→B→C. Currently A has no virtual
  participant for B, so media doesn't reach B's SFU for forwarding.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 06:49:20 +04:00
Siavash Sameni
ea51d068e6 feat: --debug-tap for relay packet header logging
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Adds --debug-tap <room> flag (or debug_tap in TOML config) that logs
every media packet's header metadata passing through a room. Use '*'
for all rooms.

Output (via tracing target "debug_tap"):
  TAP room=... dir=in addr=... seq=31 codec=Opus24k ts=520
      fec_block=5 fec_sym=1 repair=false len=65 fan_out=1

Shows: direction, source address, sequence number, codec ID, timestamp,
FEC block/symbol, repair flag, payload size, and fan-out count.
No decryption needed — headers are not encrypted.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 06:34:22 +04:00
Siavash Sameni
7271942c6a feat: federation media forwarding working — audio crosses between relays
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Added debug logging to federation signal path. Fixed the announce/recv
flow: outbound link's announce_task sends FederationRoomJoin, peer's
inbound signal_task receives it and creates virtual participant.

Tested: two relays on localhost with mutual TOML config, client A
sends tone via relay A, client B records via relay B — audio received
through federation (0.1s/RMS 7291/PASS).

Room announcement delay is ~1s (poll interval). The full pipeline:
client join → room created → announce_task detects → sends signal →
peer receives → creates virtual participant → SFU loop forwards
media via room-hash-tagged datagrams → peer demuxes → local delivery.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 06:26:49 +04:00
Siavash Sameni
da84ed332c docs: PRD for protocol analyzer — relay debug tap + full analyzer tool
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Two tools:
1. --debug-tap on relay: logs packet header metadata (seq, codec, ts,
   FEC, repair, size) per room without decryption. 0.5 day effort.
2. wzp-analyzer standalone: joins room as observer, decodes audio,
   shows TUI with per-participant waveforms + quality stats + FEC
   recovery rates. Capture/replay and HTML reports. 5-8 days total.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 05:57:27 +04:00
Siavash Sameni
e50925e05a fix: IP-based peer matching for inbound federation + room announcements
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- Inbound federation connections now matched by source IP against
  configured peer URLs (QUIC clients don't present TLS certs, so
  fingerprint matching fails for inbound direction).
- Added periodic room announcement task (1s poll) that sends
  FederationRoomJoin to peers when new rooms appear with local
  participants. Handles rooms created after federation link is up.
- Added find_peer_by_addr() to FederationManager.

Federation link topology: each relay pair has 2 connections (outbound
from each side). Outbound sends signals, peer's inbound receives them.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-08 05:49:37 +04:00
Siavash Sameni
6be36e43c2 feat: relay federation infrastructure — room bridging, loop prevention, peer connections
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Phase 1 of relay federation:

1. Signal messages: FederationRoomJoin/Leave/ParticipantUpdate added
   to SignalMessage enum for relay-to-relay room coordination.

2. Room changes: ParticipantOrigin (Local/Federated) tracking, loop
   prevention (federated media only forwards to local participants),
   ParticipantSender::Federation with 8-byte room-hash prefixed
   datagrams, merged participant lists (local + remote), new methods:
   join_federated(), update_federated_participants(), local_senders(),
   active_rooms(), local_participants().

3. FederationManager: connects to configured peers via QUIC with SNI
   "_federation", reconnects with exponential backoff (5s-300s),
   exchanges FederationRoomJoin signals, runs recv loops for both
   signals and media datagrams, creates virtual participants in rooms.

4. Accept-side: _federation SNI handling in main.rs, unknown peer
   gets helpful "add to relay.toml" log message, recognized peers
   handed off to FederationManager.

TODO: TLS fingerprint verification — currently outbound connections
use client_config() which doesn't present a cert, so inbound
verification fails. Need mutual TLS or URL-based peer matching.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 22:30:18 +04:00
Siavash Sameni
2f2720802d feat: TOML config file with federation peers + --config flag
The relay now supports loading configuration from a TOML file via
--config <path>. CLI flags override TOML values. All fields have
serde defaults so a minimal config only needs what you want to change.

Example relay.toml:
  listen_addr = "0.0.0.0:4433"
  [[peers]]
  url = "193.180.213.68:4433"
  fingerprint = "1a:39:38:..."
  label = "Pangolin EU"

Federation hint on startup now shows TOML format with TLS fingerprint
(not Ed25519 identity fingerprint), since TLS fingerprint is what
peers actually verify. Configured peers are logged on startup.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 22:13:56 +04:00
Siavash Sameni
087bfd2335 feat: deterministic TLS certificate from relay identity seed
The relay's TLS certificate is now derived from the persisted
Ed25519 seed via HKDF, so the same seed produces the same cert
and the same TLS fingerprint across restarts. This fixes the
"Server Key Changed" warnings on every relay restart.

Implementation: HKDF-SHA256(seed, "wzp-tls-ed25519") → Ed25519
signing key → PKCS8 DER → rcgen KeyPair → self-signed cert.

Also adds tls_fingerprint() helper (SHA-256 of DER cert, hex with
colons) and prints it on startup. This is the prerequisite for
relay federation (peers verify each other by TLS fingerprint).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 22:10:08 +04:00
Siavash Sameni
0a05e62c7f feat: relay prints federation peering config on startup
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On startup, the relay detects its outbound IP (via UDP socket trick)
and prints a ready-to-copy YAML snippet for other relays to federate:

  federation: to peer with this relay, add to peers config:
    - url: "193.180.213.68:4433"
      fingerprint: "a5d6:e3c6:..."

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 21:37:10 +04:00
Siavash Sameni
b97f32ce46 docs: PRD for relay federation (multi-relay mesh) + identity fix
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Documents the relay TLS identity bug (cert regenerates on restart
because server_config() creates a new keypair every time, ignoring
the persisted Ed25519 seed) and the full federation design:

- YAML config with mutual peer trust (url + fingerprint)
- QUIC connections between peers, fingerprint verification
- Room bridging: media forwarding for shared room names
- Merged participant presence across relays
- Helpful log message for unconfigured peer connection attempts
- No transcoding, no re-encryption, no central coordinator

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 21:33:05 +04:00
Siavash Sameni
d66d583583 docs: PRD for adaptive quality control (auto codec)
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Covers the full design for runtime codec switching based on network
conditions: 3-tier basic (GOOD/DEGRADED/CATASTROPHIC), extended
5-tier with studio levels, and bandwidth probing. Details the
existing QualityAdapter infrastructure, what's missing (report
ingestion, profile switch loop, cross-task signaling via AtomicU8),
and implementation plan for both Android and desktop engines.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 21:25:33 +04:00
Siavash Sameni
d06cf66538 fix: auto codec, force-ping button, relay delete button
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1. Auto codec: new "Auto" position on quality slider (JNI index 7).
   When selected, the engine uses the relay's chosen_profile from
   CallAnswer instead of the local preference. Slider now has 8
   positions: Studio 64k → Auto → Codec2 1.2k.

2. Force ping: added refresh button (↻) in Manage Relays dialog
   header. Calls pingAllServers() to re-check all relays on demand.

3. Delete relay fix: the X button was inside a Surface(onClick=...)
   which swallowed the touch event. Replaced with a separate Surface
   that properly intercepts the click.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 21:22:24 +04:00
Siavash Sameni
7bddc6b5a6 fix: advertise studio profiles in desktop handshake supported_profiles
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Same fix as Android — the CallOffer now includes STUDIO_64K/48K/32K
so the relay can negotiate studio quality levels.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 21:06:48 +04:00
Siavash Sameni
c8bcc5c974 fix: advertise studio profiles in handshake supported_profiles
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The CallOffer only advertised GOOD/DEGRADED/CATASTROPHIC. When a
client uses a studio profile, the relay's choose_profile couldn't
pick it. Now advertises all 6 profiles (studio 64k/48k/32k + good +
degraded + catastrophic) in both Android engine and shared handshake.

Also: the relay MUST be rebuilt with the new CodecId variants,
otherwise it will fail to deserialize CallOffer messages containing
studio QualityProfiles in supported_profiles.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 19:39:31 +04:00
Siavash Sameni
760126b6ab fix: remove duplicate Kotlin imports causing build failure
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Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 19:17:33 +04:00
Siavash Sameni
53f8bf8fff feat: full quality tiers + slider UI + key-change warning on Android
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1. Wire protocol: add Opus 32k/48k/64k (CodecId 6/7/8) + STUDIO
   profiles with is_opus() helper. Opus enc/dec accept all Opus variants.

2. JNI bridge: expand profile_from_int to 7 levels (0-6) mapping to
   GOOD, DEGRADED, CATASTROPHIC, Codec2_3200, STUDIO_32K/48K/64K.

3. Settings UI: replace radio buttons with Material3 Slider — 7 stops
   from Studio 64k (green) to Codec2 1.2k (dark red), matching desktop.

4. Key-change warning: AlertDialog on connect when server fingerprint
   has changed. Shows old vs new fingerprint, Accept New Key or Cancel.
   Accepting saves the new fingerprint and proceeds with the call.

5. Engine recv: handle studio codec IDs in auto-switch path.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 19:11:29 +04:00
Siavash Sameni
3b85604b41 docs: PRDs for local recording + mixer and studio quality tiers
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PRD-local-recording.md: Dual-path architecture for podcast-quality
interviews — local lossless WAV recording alongside live call, with
sync markers for post-session alignment, resumable upload to a
self-hosted mixer service that produces normalized multi-track output.

PRD-studio-quality.md: Documents the Opus 32k/48k/64k studio tiers,
when to use them, cross-codec interop, and backward compatibility.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 18:32:24 +04:00
Siavash Sameni
a8c2011445 feat: add Opus 32k/48k/64k studio quality tiers
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Adds three new codec IDs (Opus32k=6, Opus48k=7, Opus64k=8) and
corresponding STUDIO_32K, STUDIO_48K, STUDIO_64K quality profiles.
All use 20ms frames with minimal FEC (10%) for maximum quality on
good networks.

Updated across: wire protocol (codec_id.rs), encoder/decoder
(opus_enc/dec.rs), adaptive codec switch (call.rs), CLI
(--profile studio-64k), desktop engine + UI slider (8 quality
levels from Studio 64k green to Codec2 1.2k red).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 18:31:05 +04:00
Siavash Sameni
ded49bdb7b feat: replace browser confirm with proper key-change warning dialog
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When the relay's server key changes (e.g. after restart), show a
styled in-app warning dialog instead of the ugly browser confirm().
The dialog shows old vs new fingerprints and lets the user accept
the new key or cancel. Accepting updates the saved fingerprint and
refreshes the relay button state.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 18:19:53 +04:00
Siavash Sameni
b3cdad0c75 fix: copy libc++_shared.so from NDK when cargo-ndk skips it
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cargo-ndk doesn't always copy libc++_shared.so into jniLibs. The
build script now finds it in the NDK and copies it manually if
missing, preventing the build check from failing.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 18:06:28 +04:00
Siavash Sameni
fa3c7f1cef fix: dynamic frame sizing for non-default quality profiles on Android
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The send loop was hardcoded to 960 samples (20ms/Opus24k), causing
DEGRADED (Opus 6k, 40ms) and CATASTROPHIC (Codec2 1200, 40ms) to
fail — the encoder needed 1920 samples but only got 960.

Changes:
- capture_buf, ring read threshold, and timestamp increment are now
  computed from profile.frame_duration_ms (960 for 20ms, 1920 for 40ms)
- decode_buf sized to MAX_FRAME_SAMPLES (1920) to handle any incoming codec
- recv codec switch now uses correct QualityProfile per codec (was
  inheriting original profile's frame_duration_ms, breaking cross-codec)
- added ComfortNoise guard on recv path

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 18:00:27 +04:00
Siavash Sameni
369347ce54 fix: remove unused FRAME_SAMPLES_20MS constant in desktop engine
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Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 17:54:13 +04:00
Siavash Sameni
44f04b55e8 feat: quality slider in settings with color gradient
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Replace the quality dropdown with a range slider in the settings
panel. The slider goes from Auto (green) through Opus 24k, Opus 6k
(yellow), Codec2 3.2k (orange) to Codec2 1.2k (dark red). The
track uses a green-to-red gradient and the label color updates
to match the selected level. Removed the quality dropdown from
the connect screen — quality is now settings-only.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 17:50:46 +04:00
Siavash Sameni
85c2146760 feat: quality profile selection in desktop settings
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Adds a Quality dropdown (Auto / Opus 24k / Opus 6k / Codec2 3.2k /
Codec2 1.2k) to both the connect screen and settings panel. The
selected profile is passed through to the engine which configures
the encoder and decoder accordingly.

The desktop engine recv path now auto-switches the decoder codec
when incoming packets use a different codec than expected, enabling
cross-codec interop between clients on different quality settings.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 17:44:17 +04:00
Siavash Sameni
96ccb4f333 fix: auto-switch decoder codec to match incoming packets
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The CallDecoder now inspects each incoming packet's codec_id and
automatically switches the audio decoder if it differs from the
current profile. This enables cross-codec interop where one client
sends Opus and the other sends Codec2 — previously the receiver
would try to decode with the wrong codec, producing garbled audio.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 15:35:31 +04:00
Siavash Sameni
95a905e1b5 feat: add --profile/--codec flag to CLI for forcing codec selection
Enables debugging Codec2 by allowing forced codec selection from CLI.
Supports: good, degraded, catastrophic, codec2-3200, codec2-1200.
Frame size, timing, and jitter buffer are all adjusted dynamically
based on the selected profile.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 15:35:31 +04:00
Siavash Sameni
f7ccb67b02 fix: desktop ping closes endpoint properly, prevents resource leaks
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Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 15:00:32 +04:00
Siavash Sameni
4df08eadbd fix: don't block connect on offline ping — always allow connection attempt
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Server may be reachable even if ping failed (transient timeout).
User should always be able to try connecting. Fingerprint change
still shows confirm dialog (accept/reject).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 14:20:38 +04:00
Siavash Sameni
6d776097c8 feat: relay ping handling, identity persistence, linux build script (backport)
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Backported from feat/android-voip-client:
- Relay: SNI "ping" connections handled gracefully (no timeout errors)
- Relay: identity persisted in ~/.wzp/relay-identity (stable fingerprint)
- Linux fire-and-forget build script (Hetzner VM)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 11:45:27 +04:00
Siavash Sameni
68b56d9172 fix: ping every 5min (was 5s), clean endpoint on failure, never block connect
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- Ping interval: 5 minutes (was 5 seconds — too aggressive)
- Rust ping_relay: explicitly close endpoint + shutdown runtime on failure
- Connect button works regardless of ping status (never blocked)
- Ping failure doesn't corrupt engine state

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 11:40:14 +04:00
Siavash Sameni
7973c8c6a3 fix: ntfy failure notification on build error (trap ERR)
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Both Android and Linux build scripts now send ntfy notification
when build fails, not just on success.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 11:23:32 +04:00
Siavash Sameni
3e9539e5da fix: add libasound2-dev to Docker image for Linux audio builds
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Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 11:16:39 +04:00
Siavash Sameni
a1ccb3f390 feat: Linux x86_64 fire-and-forget Docker build on SepehrHomeserverdk
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Same Docker image as Android build. Separate cache dirs (cache-linux/)
to avoid conflicts when running both builds simultaneously.

Builds: wzp-relay, wzp-client, wzp-client-audio, wzp-web, wzp-bench
Uploads tar.gz to rustypaste, notifies ntfy.sh/wzp.

Usage:
  ./scripts/build-linux-docker.sh --pull         # fire and forget
  ./scripts/build-linux-docker.sh --pull --install # wait + download

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 11:09:01 +04:00
Siavash Sameni
7751439e2b feat: relay identity persistence + Linux build script
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Relay identity:
- Stored in ~/.wzp/relay-identity (hex-encoded 32-byte seed)
- Generated on first run, reused on restart
- Fingerprint stays consistent across relay restarts

Linux build script (scripts/build-linux-notify.sh):
- Fire and forget: Hetzner VM → build all binaries → upload to rustypaste → ntfy notify → destroy VM
- Builds: wzp-relay, wzp-client, wzp-client-audio, wzp-web, wzp-bench
- Packages as tar.gz, uploads to rustypaste
- --keep flag to preserve VM

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 11:05:49 +04:00
Siavash Sameni
20bc290c18 fix: relay handles ping connections gracefully (no timeout errors)
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Relay recognizes SNI "ping" and returns immediately — no handshake,
no stream accept, no timeout error logs. Client closes after QUIC
connect for RTT measurement.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 11:01:03 +04:00
Siavash Sameni
a8dc350a65 feat: codec selection in settings (Opus / Opus Low / Codec2)
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- Settings UI: radio buttons for encode codec selection
- Persisted via SettingsRepository
- Passed through WzpEngine.startCall(profile=) → JNI → Rust CallStartConfig
- Decode always accepts all codecs (per-packet codec_id switch)
- 0 = Opus 24k (GOOD), 1 = Opus 6k (DEGRADED), 2 = Codec2 1.2k (CATASTROPHIC)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 10:50:01 +04:00
Siavash Sameni
00fa109f07 feat: codec2 support — adaptive encoder/decoder, per-packet codec switch
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Android engine:
- Use wzp_codec::create_encoder/create_decoder (factory) instead of
  hardcoded OpusEncoder/OpusDecoder
- Recv path: auto-switch decoder based on incoming packet's codec_id
- Supports mixed-codec rooms (one client Opus, another Codec2)

Desktop client already uses factory functions — no changes needed.

Codec selection via QualityProfile:
- GOOD: Opus 24kbps
- DEGRADED: Opus 6kbps
- CATASTROPHIC: Codec2 1200bps

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 10:34:14 +04:00
Siavash Sameni
1e40dec468 feat: periodic server ping every 5s while app is open
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Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 10:13:51 +04:00
Siavash Sameni
aecef0905d feat: fire-and-forget build script with ntfy + rustypaste
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- Uploads build script to remote, runs in tmux (survives SSH drop)
- Builds Rust + APK in Docker
- Validates both .so files present before APK build
- Uploads APK to rustypaste
- Sends ntfy.sh/wzp notification with download URL
- --install flag: waits + downloads + adb installs locally
- --rust flag: force clean Rust rebuild
- --pull flag: git pull before building

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 10:00:49 +04:00
Siavash Sameni
18f7faa279 fix: ping as engine instance method — same lifecycle as call
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Ping was a static JNI method that loaded the .so before nativeInit,
crashing jemalloc. Now ping is an instance method on WzpEngine:

- Engine is created once (nativeInit), reused for both ping and call
- pingRelay() uses same tokio runtime pattern as startCall()
- Auto-pings all servers on app launch (after engine init)
- No process restart needed
- TOFU fingerprints saved on first successful ping

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 09:49:33 +04:00
Siavash Sameni
eeb85aeac2 feat: ping-and-exit for server RTT, remove broken UDP ping
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- Ping button: pings all servers via native QUIC, saves RTT + fingerprint
  to SharedPreferences, then exits process (System.exit)
- On restart: loads saved ping results (no native .so loading needed)
- Avoids jemalloc crash: native lib only loaded once per process lifetime
- Removed broken UDP probe (QUIC servers don't respond to it)
- SettingsRepository: savePingRtt/loadPingRtt for cached results
- PingResult: added reachable field

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 09:31:02 +04:00
Siavash Sameni
00b405aa87 feat: debug recording off by default, toggle in settings
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- AudioPipeline.debugRecording defaults to false (was true)
- SettingsRepository: persist debug_recording preference
- CallViewModel: debugRecording StateFlow + setter, wired to AudioPipeline
- Only records PCM + RMS when explicitly enabled in settings

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 09:01:43 +04:00
Siavash Sameni
d09e21965e feat: pure Kotlin UDP ping — periodic every 5s, no JNI crash
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Replace WzpEngine.pingRelay() (JNI, loads native .so, crashes jemalloc
on Android 16 MTE) with pure Kotlin DatagramSocket UDP probe.

- RelayPinger: sends QUIC Version Negotiation trigger packet, measures
  RTT from response. No native lib, no JNI, zero crash risk.
- Periodic: pings all servers every 5 seconds via coroutine
- Server fingerprint: filled lazily on first real QUIC connection
  (TOFU still works, just delayed)
- Lock status: OFFLINE when ping fails, NEW until first connection

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 08:57:27 +04:00
Siavash Sameni
97bcc79f9b feat: desktop-style UI + docker build scripts, fix ping crash
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- InCallScreen rewrite matching desktop dark theme layout
- Removed auto-ping LaunchedEffect (loading native .so early via
  pingRelay crashes jemalloc on Android 16 MTE)
- Added Docker build scripts (Dockerfile.android-builder + build-android-docker.sh)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 08:19:45 +04:00
Siavash Sameni
9f7962a6cd fix: vec allocation for desktop AudioRing (match Android fix)
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Same fix as Android: Box::new([0i16; 16384]) allocates 32KB on the
stack before moving to heap. Use vec![].into_boxed_slice() for
direct heap allocation.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-07 05:26:59 +04:00
Siavash Sameni
264ef9c4d4 feat: relay ping with RTT, server TOFU, lock icons (Phase 2 backport)
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Rust JNI:
- nativePingRelay: QUIC connect with 3s timeout, returns RTT + server
  certificate fingerprint as JSON. Static method, no engine needed.

Kotlin:
- WzpEngine.pingRelay() static wrapper
- SettingsRepository: TOFU fingerprint persistence (tofu_{address} keys)
- CallViewModel: pingAllServers() coroutine, lockStatus() helper,
  PingResult/LockStatus data types
- InCallScreen: server chips show lock icon + RTT color (green/yellow),
  "Ping All" button

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 22:43:53 +04:00
Siavash Sameni
a9adb5cfd7 feat: identicons, tap-to-copy fingerprint, recent rooms (Phase 1 backport)
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Backport from desktop client to Android:

Identicons:
- New Identicon.kt composable: deterministic 5x5 symmetric Canvas pattern
  from fingerprint hash (same algorithm as desktop identicon.ts)
- Participant list shows identicon + name + tappable fingerprint
- Settings page shows identicon next to fingerprint

CopyableFingerprint:
- Tap any fingerprint text to copy to clipboard with Toast feedback
- Used in participant list and settings page

Recent rooms:
- SettingsRepository: persists last 5 (relay, room) pairs
- CallViewModel: saves on startCall, exposes as StateFlow
- InCallScreen: clickable chips that fill room + select matching server

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 22:37:46 +04:00
Siavash Sameni
a39b074d6e fix: DirectByteBuffer as class field — survives ART JIT OSR
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Previous attempt allocated DirectByteBuffer as local variables inside
runCapture/runPlayout. ART's JIT On-Stack Replacement nulled them
when recompiling the hot loop mid-execution.

Fix: allocate as class fields on AudioPipeline (captureDirectBuf,
playoutDirectBuf). Object fields live on the heap, immune to OSR
stack frame replacement.

Eliminates JNI array copies (GetShortArrayRegion/SetShortArrayRegion)
from the audio hot path, preventing ART GC SIGBUS crashes on
Android 16 with concurrent mark-compact GC.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 22:22:54 +04:00
Siavash Sameni
9cab6e2347 ci: skip build on CI-only file changes
Add paths-ignore for .gitea/** so build.yml doesn't waste runner time
when only workflow files are modified.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 22:13:29 +04:00
Siavash Sameni
8c9befb15d ci: skip build on CI-only file changes
Add paths-ignore for .gitea/** so build.yml doesn't waste runner time
when only workflow files are modified.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 22:12:32 +04:00
Siavash Sameni
5e93cb74f2 fix: filter tracing to INFO for wzp crates, WARN for jni crate
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The jni crate emits VERBOSE logs for every JNI method lookup (~10 lines
per call, 100+ calls/sec on audio threads). This floods logcat, consumes
CPU, and triggers system kills. Filter to only show INFO+ for our crates
and WARN+ for everything else.

Also fix build script: clean full Rust target to ensure libc++_shared.so
is always copied by cargo-ndk.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 21:37:29 +04:00
Siavash Sameni
b56b4a759c revert: use ShortArray audio path (DirectByteBuffer causes null ptr crash)
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DirectByteBuffer.clear() crashes with null pointer in ART's JIT OSR
compiled code on Android 16. Revert AudioPipeline to use the original
ShortArray writeAudio/readAudio path.

The DirectByteBuffer JNI functions remain in WzpEngine.kt and
jni_bridge.rs for future use once the OSR issue is resolved.

The original SIGBUS from ART GC is rare (~1 crash per 8 min call)
and doesn't warrant the DirectByteBuffer approach until we can
allocate the buffer as a class field outside the hot loop.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 21:17:15 +04:00
Siavash Sameni
6f99841cc7 fix: cloud build script — filter by server name, rsync upload, cx33
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- Filter hcloud by SERVER_NAME to avoid touching other servers
- Use rsync instead of tar (handles submodules, no macOS xattr spam)
- Default server type cx33
- Release APK failure is non-fatal (debug APK still produced)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 20:00:10 +04:00
Siavash Sameni
3f869a4cd7 ci: add GitHub mirror workflow
Automatically pushes branches and tags to github.com:manawenuz/wzp.git
on every push to Forgejo. Uses GH_SSH_KEY secret for authentication.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 19:50:39 +04:00
Siavash Sameni
3b0811ce2e ci: add GitHub mirror workflow
Automatically pushes branches and tags to github.com:manawenuz/wzp.git
on every push to Forgejo. Uses GH_SSH_KEY secret for authentication.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 19:49:59 +04:00
Siavash Sameni
9eed94850d fix: DirectByteBuffer audio path — eliminate JNI array copies
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Adds nativeWriteAudioDirect / nativeReadAudioDirect JNI functions
that accept a DirectByteBuffer instead of ShortArray. The buffer's
native memory is accessed directly by Rust via pointer — no
GetShortArrayRegion / SetShortArrayRegion, no GC-managed array
copies on the audio hot path.

This fixes SIGBUS crashes on Android 16 where ART's concurrent
mark-compact GC crashes when flipping thread roots during JNI
array operations on MAX_PRIORITY audio threads.

Old ShortArray methods kept for backward compatibility.
AudioPipeline switched to use Direct variants.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 19:29:08 +04:00
Siavash Sameni
5e9718aeb2 docs: incident report — SIGBUS in ART GC during audio JNI calls
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Android 16's concurrent mark-compact GC crashes when flipping
thread roots on our MAX_PRIORITY audio threads during JNI calls
(AudioRecord.read / AudioTrack.write). Not our code — all crash
frames are in libart.so.

Proposed fixes:
- Short term: DirectByteBuffer to reduce JNI transitions
- Long term: Oboe native audio from Rust (no JNI, no GC)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 19:21:32 +04:00
Siavash Sameni
3093933602 fix: build script works on Ubuntu 24.04 (cmake 3.28) too
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cmake 3.28 works when ANDROID_NDK is set (not just ANDROID_NDK_HOME).
Relaxed version check from <=3.26 to <=3.30.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 19:00:06 +04:00
Siavash Sameni
4c6c909732 feat: comprehensive Android build script for Debian 12
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Documents WHY each version is pinned:
- cmake 3.25: 3.27+ rewrote Android-Determine.cmake with bugs
- NDK 26.1: NDK 27 scudo crashes on MTE devices (Nothing A059)
- JDK 17: Gradle 8.5 + AGP 8.2.0 official support
- ANDROID_NDK: cmake checks this, not ANDROID_NDK_HOME

Idempotent, works from clone or existing tree.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 18:37:12 +04:00
Siavash Sameni
33fab9a049 fix: vec allocation for AudioRing, catch_unwind on tracing init, profiling
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- AudioRing: use vec![].into_boxed_slice() instead of Box::new([]) to
  avoid 32KB stack allocation that crashes scudo on Android
- JNI bridge: wrap tracing_subscriber init in catch_unwind to survive
  sharded_slab allocation failures on some devices
- Engine: per-step encode profiling (avg_agc_us, avg_opus_us, avg_fec_us,
  avg_send_us) logged every 5 seconds in send stats

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 15:41:46 +04:00
Siavash Sameni
31d2306915 feat: per-step encode profiling in send task stats
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Adds average microsecond timings for each encode step:
- avg_agc_us: AGC processing
- avg_opus_us: Opus encoding
- avg_fec_us: FEC encode + repair generation
- avg_send_us: QUIC send_media
- avg_total_us: sum of above

Logged every 5 seconds in send stats. Resets each interval.
Use to identify which step is bottlenecking the encode loop
on devices where fps drops below 50.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 14:18:33 +04:00
Siavash Sameni
2263e898e5 fix: port AudioRing reader-detects-lap fix to desktop client
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Same fix as Android (4af7c5f): writer never touches read_pos,
reader self-corrects when lapped. Power-of-2 capacity (16384),
bitmask indexing, overflow/underrun counters.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 13:42:33 +04:00
Siavash Sameni
4af7c5f94c fix: AudioRing cursor desync + capture thread use-after-free
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AudioRing (reader-detects-lap architecture):
- Writer NEVER touches read_pos — fixes SPSC invariant violation
- Reader self-corrects when lapped (snaps read_pos forward)
- Power-of-2 capacity (16384 = 341ms) with bitmask indexing
- Added overflow_count and underrun_count diagnostics
- Wired ring health into engine stats and periodic logging

Capture thread use-after-free (drain latch):
- Added CountDownLatch(2) to AudioPipeline
- Audio threads count down after exiting their loops
- teardown() awaits latch (200ms timeout) before destroy()
- Guarantees no in-flight JNI calls when native handle is freed
- stopAudio() no longer nulls pipeline (teardown handles it)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 13:28:34 +04:00
Claude
6597b5bd86 docs: incident report + fix spec for capture thread use-after-free crash
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SIGSEGV on hangup: capture thread calls writeAudio() via JNI after
teardown() has freed the native engine handle. TOCTOU race between
the nativeHandle==0L check and destroy() on the ViewModel thread.

Fix: CountDownLatch(2) — audio threads count down after exiting loops,
teardown() awaits before destroy(). 2 Kotlin files, no Rust changes.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 09:21:35 +00:00
Claude
ae9d8526dd docs: implementation spec for AudioRing SPSC desync fix
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Complete spec for fixing the playout ring buffer cursor race that
causes 12-16s bidirectional silence mid-call. Includes exact code,
memory ordering rationale, unit tests, and verification steps.

Any agent can implement from this document alone.

See also: debug/INCIDENT-2026-04-06-playout-ring-desync.md

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 09:16:47 +00:00
Siavash Sameni
9ab57ba037 merge: fj/feat/android-voip-client — congestion fix, AEC toggle, debug logging
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Merged 10 commits from Android branch:
- Send task crash fix on QUIC congestion (continue instead of break)
- AEC toggle + NoiseSuppressor on Android
- Debug reporter for crash diagnostics
- Mic mute crackling fix
- Participant dedup in UI
- Proper QUIC connection close on hangup
- Null alias display fix
- Tracing → Android logcat
- Incident reports for send-task crash and playout ring desync

Conflict resolved in room.rs: kept Android's improved debug logging
(recv gap tracking, lock contention, forward latency, send errors)
inside our media_task async block for parallel signal handling.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 13:13:43 +04:00
Siavash Sameni
7806d4ec04 feat: identicons, server fingerprints, lock status (TOFU)
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Identicon generator:
- Deterministic 5x5 symmetric pattern from fingerprint hash
- HSL-derived colors, rendered as inline SVG
- Click any identicon to copy its fingerprint to clipboard
- Used for participants, user identity, and relay servers

Server identity (TOFU — Trust On First Use):
- Ping returns server fingerprint (QUIC peer certificate hash)
- First contact: auto-saved as known fingerprint
- Subsequent pings: compared against known fingerprint
- Lock icons: locked (verified), unlocked (new), warning (changed), red (offline)
- Fingerprint mismatch shows confirmation dialog before connecting

UI updates:
- Participants show identicons instead of letter avatars
- User identity shows identicon + fingerprint on connect screen
- Manage Relays shows identicon per server with lock status
- Relay button shows lock icon instead of colored dot

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 13:02:42 +04:00
Siavash Sameni
d31b81a21d fix: replace relay dropdown with direct dialog on click
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- Click relay button opens Manage Relays dialog directly (no dropdown)
- Click a relay in the dialog to select it (highlighted with accent border)
- × button to delete, Add Relay button to add new
- Removed all dropdown menu code and CSS

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 12:53:13 +04:00
Claude
4d54b6f9e4 docs: incident reports for send-task crash and playout ring desync
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Two root-caused bugs documented with full evidence:

1. Send task fatal exit on QUIC congestion (FIXED in 2092245)
   - send_media() Err(Blocked) caused break → killed entire call
   - Now drops packet and continues

2. Playout ring buffer cursor desync (ROOT-CAUSED, fix pending)
   - AudioRing::write() mutates read_pos from producer thread on overflow
   - Violates SPSC contract → reader/writer fight over read_pos
   - Causes 12-16s bidirectional silence ~25-30s into call
   - Both clients affected simultaneously

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 08:52:14 +00:00
Siavash Sameni
c268ce419a fix: relay dialog overflow — stack inputs, full-width Add button
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- Dialog fits within 360px window (was overflowing at 420px)
- Add inputs stacked: name + host:port in a row, "Add Relay" button below
- Text overflow with ellipsis on relay names and addresses
- Proper min-width: 0 on flex children to prevent overflow

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 12:49:26 +04:00
Siavash Sameni
61b6e67610 feat: relay server dropdown with status indicators and manage dialog
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- Relay selector as dropdown with green/yellow/red status dots
  (green < 200ms, yellow > 200ms, red = offline, gray = unknown)
- All relays pinged on startup, RTT shown next to each
- "Manage Relays..." dialog: add/remove servers, see live status
- Clicking a relay in dropdown selects it, fills connect form
- Recent room chips auto-select matching relay
- Migrates old single-relay settings format automatically
- Prevents connecting to offline relays

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 12:44:19 +04:00
Siavash Sameni
dddf5d2e2d feat: relay ping with RTT display, fix dead_code warning
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- New ping_relay Tauri command: QUIC connect with 3s timeout, returns RTT ms
- Relay status shown next to input field: "42ms" (green) or "offline" (red)
- Auto-pings on app startup and debounced on relay input change
- Fix SyncWrapper dead_code warning with #[allow(dead_code)]

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 12:41:28 +04:00
Siavash Sameni
ed272d29f8 feat: fingerprint at startup, relay+room pairs, auto-reconnect, cleanup
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#7 Fingerprint shown before connecting — new get_identity command reads
   ~/.wzp/identity at startup (generates if missing). Click to copy.

#8 Recent rooms store (relay, room) pairs — clicking a chip fills both
   fields. Settings panel shows relay alongside room name. Migrates
   old string[] format automatically.

#9 Auto-reconnect on unexpected disconnect — exponential backoff
   (1s, 2s, 4s... max 10s), up to 5 attempts. Yellow blinking dot
   shows reconnecting state. Stops if user clicks hangup.

#10 Audio handle cleanup — CPAL handles stored in SyncWrapper (no more
    mem::forget), dropped properly on CallEngine::stop().

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 12:15:05 +04:00
Claude
2b3bdae440 fix: enable Rust tracing → Android logcat via tracing-android
Rust tracing subscriber was never initialized — all info!/warn!/error!
calls in the engine went to /dev/null. This meant our send/recv health
logging was invisible and we couldn't confirm the congestion fix was
active.

Now initializes tracing-android layer on first nativeInit(), routing
all Rust logs to logcat under tag "wzp_android". Also expanded logcat
filter in DebugReporter to capture engine-level log lines.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 08:03:28 +00:00
Siavash Sameni
21f5b24cbf fix: keep audio handles alive for call duration, fix Send+Sync
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The VPIO/CPAL audio handles were dropped at the end of start(),
killing the audio unit immediately. Audio I/O stopped working
after the first frame.

- Store audio handle in CallEngine via SyncWrapper
- Drop MutexGuard before returning from status() (Send future)
- Audio streams now live for the entire call duration

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 12:00:16 +04:00
Siavash Sameni
9b733010ab fix: blocking_lock panic in status(), fingerprint copy-to-clipboard
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- Change status() from blocking_lock to async lock().await —
  fixes "Cannot block the current thread from within a runtime" panic
  that froze the call timer and broke audio
- Click fingerprint to copy to clipboard (both connect and settings screens)
- Show "Copied!" feedback on click

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 11:53:31 +04:00
Siavash Sameni
80d5bd7628 fix: survive QUIC congestion — drop packets instead of killing send task
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send_datagram() returns Err(Blocked) when the QUIC congestion window
is full. This is transient — the window reopens once ACKs arrive.
Previously, all send paths treated this as fatal (break/return),
which killed the send task and cascaded via tokio::select! to kill
the entire call.

Now: log warning, drop the packet, continue. Brief audio glitch
(20-100ms) instead of complete call death. FEC on the receiver
side recovers most dropped packets.

Fixed in:
- CLI run_live send task (continue + error counter)
- CLI run_file_mode send paths (2 locations)
- Desktop engine send task

Also hardened recv tasks: transient errors (non-closed/reset)
are survived instead of causing exit.

Matches the fix applied to Android client (engine.rs).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 11:48:20 +04:00
Siavash Sameni
4a195a923a feat: settings panel with Cmd+, shortcut (macOS standard)
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- Full settings page as modal overlay (blur backdrop)
- Opens via gear icon on connect/call screens or Cmd+, (Ctrl+, on Win/Linux)
- Escape or click outside to close
- Settings: relay, room, alias, OS AEC toggle, AGC toggle
- Identity section showing fingerprint and identity file path
- Recent rooms management (remove individual, clear all)
- Save syncs back to connect form
- Gear icon on both connect and in-call screens

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 11:44:22 +04:00
Siavash Sameni
f726f8cfa4 feat: desktop GUI enhancements — audio level, call timer, VPIO, settings
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- Audio level meter with log-scale RMS visualization
- Call duration timer
- VPIO (OS AEC) wired through to engine with fallback to CPAL
- "You" badge on own participant entry
- Recent rooms list (click to reuse)
- Enter key to connect from form fields
- Improved dark theme with pulse animation on status dot
- Settings persistence via localStorage (relay, room, alias, AEC, recent rooms)
- Fingerprint display on connect screen
- Keyboard shortcuts skip input fields

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 11:40:07 +04:00
Claude
20922455bd fix: send task crash on QUIC congestion + AEC toggle + debug reporter
Root cause: send_media() returns Err(Blocked) when QUIC congestion
window is full. The send task treated ANY send error as fatal (break),
killing the entire call. Now send errors drop the packet and continue.

Also hardened recv task to survive transient errors and added health
logging (recv gap tracking, periodic stats) to both send and recv.

Relay: added comprehensive debug logging — recv gaps, lock contention,
forward latency, send errors — all per-participant with 5s stats.

Other changes:
- AEC toggle in Settings (persisted, applied on next call)
- Debug report: records call audio (WAV), RMS histogram (CSV), logcat,
  stats. Emailed as zip via Android share intent after call ends.
- Replaced LinearProgressIndicator with Box (compose version compat)
- FileProvider for sharing debug zip attachments

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 07:38:56 +00:00
Siavash Sameni
e468454464 feat: Tauri desktop GUI app with call engine
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- New desktop/ directory with Tauri v2 + Vite + TypeScript
- Rust backend: CallEngine wrapping wzp-client audio + transport
- Web frontend: connect screen, in-call screen with participants,
  mic/speaker mute, keyboard shortcuts (m/s/q)
- Dark theme UI, settings persistence via localStorage
- Platform-aware --os-aec: warns on Windows/Linux (not yet implemented)
- Workspace updated to include desktop/src-tauri

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 11:25:54 +04:00
Siavash Sameni
d1c96cd71f feat: macOS VoiceProcessingIO for hardware AEC + delay-compensated NLMS
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- Add --os-aec flag: uses Apple VoiceProcessingIO audio unit for
  hardware echo cancellation (same engine as FaceTime)
- New vpio feature + audio_vpio.rs: combined capture+playback via VPIO
- Improved software AEC: delay-compensated leaky NLMS with Geigel DTD
  (60ms tail, 40ms delay, configurable via --aec-delay)
- Add --aec-delay flag for tuning software AEC delay compensation
- Add dev-fast Cargo profile (opt-level 2 with incremental compilation)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 11:10:10 +04:00
Siavash Sameni
1b00b5e2a4 feat: improved AEC, keyboard shortcuts, dedup participants, dev-fast profile
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AEC improvements:
- Reduce echo tail from 100ms to 30ms (3.3x faster, suited for laptops)
- Add double-talk detection: freeze adaptation when near-end speaks
- Add residual echo suppression
- Disable AEC by default in --android mode (macOS has built-in AEC)

CLI features:
- Keyboard shortcuts: m=mic mute, s=speaker mute, q=quit (raw terminal mode)
- Dedup participants in RoomUpdate display (same fingerprint+alias shown once)
- Add dev-fast profile (opt-level 2 with incremental compilation)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 10:15:23 +04:00
Claude
e6564bab57 fix: mic mute crackling + add AEC/NoiseSuppressor + dedup room participants
Mic mute: the send loop now zeros the capture buffer when muted instead
of relying on write_audio() to skip writes. Previously stale ring data
and AGC amplification of near-silence caused crackling artifacts.

AEC: attach Android's hardware AcousticEchoCanceler to the AudioRecord
session. Also attach NoiseSuppressor when available. Both are released
on capture stop.

Room UI: deduplicate participants by fingerprint so ghost entries from
stale relay state don't show duplicate names.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 06:06:35 +00:00
Siavash Sameni
cfb48df1ef feat: direct playout mode, AEC far-end, audio processing switches
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- Add --android/--direct-playout: bypass jitter buffer, decode on recv
  (matches Android engine architecture)
- Wire AEC far-end reference from decoded playout to encoder
- Add --no-aec, --no-agc, --no-fec, --no-silence, --no-denoise switches
- Fix BufferSize::Fixed(960) → Default for macOS CoreAudio compat
- Optimize wzp-codec, wzp-fec, audiopus, nnnoiseless in debug profile
- Add capture callback size diagnostic logging

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 09:48:34 +04:00
Claude
aebf9156c0 fix: dedup participants in UI, wait for QUIC close ack before exiting
UI: deduplicate room participants by fingerprint so ghost entries from
stale relay state don't show duplicates.

Engine: after select! ends, call close_now() + connection.closed() with
500ms timeout to wait for the relay to acknowledge the CONNECTION_CLOSE.
Previously the close frame was queued but the runtime died before quinn
could retransmit if the first packet was lost.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 05:40:06 +00:00
Claude
9bbaec6b35 fix: use shutdown_timeout so QUIC CONNECTION_CLOSE actually gets sent
shutdown_background() killed the tokio runtime before quinn could send the
CONNECTION_CLOSE frame on the wire, so the relay never knew the client left.
Now use shutdown_timeout(500ms) to give quinn time to flush the close frame,
matching the desktop client pattern (which uses 2s timeout).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 05:20:20 +00:00
Siavash Sameni
ba29d8354f fix: send alias via CallOffer handshake (match Android approach)
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Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 09:10:07 +04:00
Siavash Sameni
0908507a7a Merge remote-tracking branch 'origin/feat/android-voip-client' into feat/desktop-audio-rewrite 2026-04-06 09:04:55 +04:00
Siavash Sameni
860c90394d feat: rewrite desktop audio I/O with lock-free ring buffers
- Replace Mutex-based CPAL callbacks with atomic SPSC ring buffers
- Proper async send/recv loops (no block_on), 20ms playout tick
- Add signal task for RoomUpdate presence display
- Add --alias, --raw-room flags and key persistence (~/.wzp/identity)
- Add SetAlias signal variant and relay-side handling
- Graceful Ctrl+C shutdown with force-quit on second press

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 09:04:51 +04:00
Claude
dc66b60d18 fix: null alias display — Android JSONObject.optString returns literal "null"
o.optString("alias", null) returns the string "null" when the JSON value
is JSON null. Use o.isNull() check first. Also handle empty fingerprint
edge case with "unknown" fallback.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 05:04:47 +00:00
Claude
a9c4260b4e fix: close QUIC connection on hangup so relay removes participant immediately
stop_call() now calls close_now() on the stored transport handle before
killing the tokio runtime. This sends a QUIC CONNECTION_CLOSE frame so
the relay's recv loop breaks immediately, triggering leave() + RoomUpdate
broadcast. Previously the runtime was killed first, so transport.close()
never ran and the relay kept stale participants until idle timeout.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 04:58:24 +00:00
Claude
7eb136fcb3 fix: settings save button (back=discard), fix missing alias in featherchat tests
- Settings now uses draft state — changes only persist on explicit Save
- Back button discards unsaved changes
- Added applyServers() for batch server updates
- Added missing alias field to CallOffer in featherchat tests

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 04:30:23 +00:00
Claude
550a124972 fix: add missing alias arg to perform_handshake call in wzp-web
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 04:15:24 +00:00
Claude
0835c36d0f feat: settings page with persistence, client alias in handshake, fix null fingerprints
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- Add SettingsScreen with identity (alias, key backup/restore), audio defaults,
  server management, network prefs, and default room
- SettingsRepository persists all settings via SharedPreferences
- Auto-generate random display names on first launch (e.g. "Swift Wolf")
- Thread alias through CallOffer → relay handshake → RoomUpdate broadcast
- Derive caller fingerprint from identity key in relay handshake (fixes null
  fingerprints when --auth-url is not set)
- Persist identity seed for stable fingerprints across reconnects
- Add alias field to SignalMessage::CallOffer (serde default for backward compat)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 03:56:33 +00:00
Claude
6228ab32c1 ci: upload build artifacts to rustypaste
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Requires PASTE_AUTH and PASTE_URL secrets configured in Forgejo.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 02:08:13 +00:00
Claude
bd258f432a fix: remove actions/upload-artifact (unsupported on Forgejo)
Some checks failed
Build Release Binaries / build-amd64 (push) Has been cancelled
Forgejo doesn't support @actions/artifact v4. Package the tarball
and print sizes instead. Binaries can be grabbed from the runner
workspace or deployed directly.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 02:07:06 +00:00
Claude
8bf073aa80 fix: handle RoomUpdate variant in wzp-client signal type mapping
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Build Release Binaries / release (push) Has been skipped
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-06 01:54:36 +00:00
Claude
72e834b45e fix: init git submodules in CI with HTTPS fallback
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Build Release Binaries / release (push) Has been skipped
The featherchat submodule uses SSH URL which doesn't work in CI.
Convert to HTTPS via git insteadOf before submodule init.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 18:24:59 +00:00
Claude
673ffd498c fix: use catthehacker/ubuntu:act-latest for Forgejo CI runner
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Build Release Binaries / release (push) Has been skipped
The Forgejo runner needs Node.js for actions/checkout@v4.
catthehacker/ubuntu:act-latest has Node.js pre-installed.
Also install Rust in the workflow since the base image doesn't have it.
Build triggers on main + feat/* branches now.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 18:19:14 +00:00
Claude
2d4b8eebd5 feat: RoomUpdate protocol — broadcast participant list on join/leave
- Add RoomUpdate signal message to wzp-proto with participant count + list
- Add RoomParticipant struct (fingerprint + optional alias)
- Store fingerprint/alias in relay Participant struct
- Broadcast RoomUpdate to all room members on join and leave
- Add signal recv task in Android engine to handle RoomUpdate
- Surface room_participant_count + room_participants in CallStats JSON
- Show "X in room" with participant names in Android in-call UI

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 18:12:24 +00:00
Claude
a23d9f5e41 feat: foreground service, dB gain sliders, speaker routing, live network stats
- Wire CallService foreground service for background calls (microphone type)
- Add Voice Volume + Mic Gain sliders (-20 to +20 dB) applied in Kotlin
- Connect AudioRouteManager for real speaker toggle via AudioManager
- Feed quinn QUIC RTT into PathMonitor, display Loss/RTT/Jitter from live data
- Nuclear teardown between calls — recreate engine + audio pipeline each call
- Fix re-entrant teardown loop from CallService notification callback
- Park audio threads as daemons to avoid libcrypto TLS destructor crash on exit
- Remove duplicate wakelocks from Activity (service owns them now)
- Strip AEC + denoise from capture path, keep AGC only (incremental approach)
- Fix .so copy target: libwzp_android.so not libwzp.so

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 17:45:00 +00:00
Claude
b3e56ecbd8 feat: add AGC to capture + playout paths, add server UI, DNS resolve
- Wire AutoGainControl on both capture (mic → encode) and playout
  (decode → speaker) paths to normalize volume levels
- Add server list with add/remove custom server dialog
- Add IPv4/IPv6 preference toggle for DNS resolution
- Resolve DNS hostnames to IP in Kotlin before passing to Rust engine
- Revert to IP addresses for default servers (DNS still broken on QUIC)

AGC confirmed working — voice levels noticeably improved in testing.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 14:02:33 +00:00
Claude
2fa07286c3 feat: wakelock for background calls, server selector UI
- Partial wake lock + WiFi high-perf lock during calls — audio
  continues when screen is off / phone is locked
- Server selector: toggle between LAN (172.16.81.175) and Pangolin
  (pangolin.manko.yoga) before connecting
- Room name editable in idle screen

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 12:54:02 +00:00
Claude
bf91cf25bd feat: add real audio pipeline with Opus + RaptorQ FEC
- AudioPipeline: Kotlin AudioRecord/AudioTrack on JVM threads, PCM
  shuttled to Rust via lock-free ring buffers + JNI
- FEC: RaptorQ fountain codes on encode (5 frames/block, 20% repair
  ratio for GOOD profile), decoder feeds repair symbols for recovery
- Real audio level meter from mic RMS (replaces fake animation)
- Room name editable in UI (default: "android")
- Relay changed to pangolin.manko.yoga:4433
- Stats overlay shows FEC recovered count
- CallState now synced from polled stats (fixes "Connecting" stuck bug)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 12:33:59 +00:00
Claude
81c756c076 chore: switch relay to 172.16.81.175:4433 for testing
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 12:01:51 +00:00
Claude
af85a49e86 fix: eliminate all native thread creation — run everything single-threaded
pthread_create crashes on Android due to static bionic __init_tcb stubs
in the Rust std prebuilt rlibs. This is unfixable without rebuilding std.

Solution: run the entire call (QUIC connect, handshake, media send/recv)
on a single tokio current_thread runtime. The JNI startCall() now blocks,
so Kotlin dispatches it to Dispatchers.IO (JVM thread, not pthread).

Audio pipeline temporarily simplified to silence frames — will restore
once threading is solved (either via Java Thread or rebuilding std).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 09:52:28 +00:00
Claude
bae03365da fix: restore getauxval_fix.c + current_thread tokio — both needed
The getauxval override (dlsym wrapper) fixes SIGSEGV in
init_have_lse_atomics at library load time. The current_thread
tokio runtime avoids SEGV_ACCERR in pthread_create/__init_tcb.
Both fixes are required together.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 09:37:57 +00:00
Claude
9d9ce4706d fix: use current_thread tokio runtime — avoid pthread_create SEGV on Android
Multi-thread tokio runtime crashes with SEGV_ACCERR in __init_tcb
during pthread_create on Android (static bionic stubs from CRT).
Switch to current_thread runtime which runs network I/O on the
calling thread without spawning additional OS threads.

Also: clean up build.rs — use only libc++_shared.so (dynamic),
remove getauxval_fix.c hack, remove static c++/c++abi linking.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 09:27:46 +00:00
Claude
9098e28a1f fix: SIGSEGV in getauxval — override broken CRT stub with dlsym wrapper
compiler-rt's init_have_lse_atomics calls getauxval(AT_HWCAP) at
library load time. The static getauxval from the CRT reads from
__libc_auxv which is NULL in shared libraries → SIGSEGV at 0x0.

Fix: compile getauxval_fix.c that provides a getauxval() which uses
dlsym(RTLD_DEFAULT) to find the real bionic getauxval at runtime.
Also switch to libc++_shared.so (bundled in APK) to avoid pulling
in static libc stubs.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 08:39:57 +00:00
Claude
f6d51fce61 fix: target API 26 in ELF — pthread_atfork blocked by bionic at API 21
The .note.android.ident ELF section had API level 0x15 (21), causing
Android's bionic linker to block pthread_atfork (used by rand crate).
Fix: pass -P 26 to cargo-ndk and set linker to android26-clang.
Verified: ELF now shows 0x1a (26).

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 06:05:44 +00:00
Claude
a8dd0c2f57 fix: also link libc++abi for RTTI — resolve missing __class_type_info vtable
- Compile all 62 Oboe source files (was headers-only, missing symbols)
- Link libc++_static + libc++abi with NDK sysroot search path
- Bump linker target from android21 to android26 (fixes pthread_atfork)
- Link liblog + libOpenSLES for Oboe runtime deps

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 05:48:49 +00:00
Claude
64566e9acb fix: logcat-server.py SyntaxError — global declaration after use
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 05:12:28 +00:00
Claude
10eb19cd24 feat: add logcat HTTP server for remote crash debugging
Simple Python script that captures adb logcat and serves it over HTTP.
Run on laptop, read from anywhere via curl/browser.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 05:11:19 +00:00
Claude
778f4dd428 fix: link libc++ statically — crash on launch due to missing libc++_shared.so
- Set cpp_link_stdlib(None) to suppress cc crate's automatic linking
- Explicitly link both c++_static and c++abi with NDK sysroot search path
- Fixes RTTI vtable symbol (_ZTVN10__cxxabiv117__class_type_infoE) error
- Verified: only liblog.so remains as dynamic dependency

Closes #001

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 05:07:25 +00:00
Siavash Sameni
622fdee51f fix: also link libc++abi for RTTI — resolve missing __class_type_info vtable
Previous fix linked c++_static but not c++abi. Android NDK splits the
static C++ runtime into two archives: libc++_static.a (STL) and
libc++abi.a (RTTI/exceptions). Without c++abi, dlopen fails on
_ZTVN10__cxxabiv117__class_type_infoE.

Now using cpp_link_stdlib(None) to suppress cc crate auto-linking, then
explicitly linking both c++_static and c++abi via cargo:rustc-link-lib.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 09:00:14 +04:00
Claude
b204213a01 build: rebuild APK with static libc++ linking (fixes #001)
libc++_shared.so is no longer a runtime dependency — verified
via llvm-readelf. Only system libs (libdl, liblog, libm, libc) remain.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 04:56:35 +00:00
Siavash Sameni
e751af7e38 fix: link libc++ statically — crash on launch due to missing libc++_shared.so
The app crashed immediately when loading libwzp_android.so because the
cc crate's default dynamic linking produced a runtime dependency on
libc++_shared.so, which was never packaged into the APK.

Adding .cpp_link_stdlib(Some("c++_static")) to build.rs bakes the C++
runtime into libwzp_android.so directly, eliminating the missing .so.

See issues/001-libc++-shared-crash.md for full diagnosis and logcat trace.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 08:52:55 +04:00
Claude
8d5f6fe044 feat: wire QUIC transport, JNI bridge, connect UI + add docs
- Replace raw FFI with proper `jni` crate for string marshalling
- Wire QUIC transport in engine: connect to relay, crypto handshake
  (CallOffer/CallAnswer, X25519+Ed25519), send/recv MediaPackets
- Feed received packets into jitter buffer (was previously ignored)
- Add connect screen UI with CALL button (idle state) and in-call
  controls (mute, speaker, hang up, live stats)
- Hardcode relay 172.16.81.125:4433, room "android"
- Add comprehensive docs in docs/android/:
  architecture.md (8 mermaid diagrams), build-guide.md,
  debugging.md, maintenance.md, roadmap.md

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 04:43:49 +00:00
Claude
780309fede fix: crash on launch — don't auto-start call, handle null JNI strings, remove stdout tracing
- CallActivity no longer auto-starts a call on launch
- CallViewModel lazily inits engine only when startCall() is called
- nativeGetStats nullable return handled safely in Kotlin
- Removed tracing_subscriber::fmt() which panics on Android (no stdout)
- All JNI calls wrapped in try/catch on Kotlin side

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-05 02:04:23 +00:00
Claude
73ebcdd869 build: Android APK builds working — debug (8.9MB) and release (2.0MB)
- Fix C++ std::std:: double namespace in oboe_bridge.cpp
- Auto-fetch Oboe headers from GitHub in build.rs
- Configure cargo cross-compilation (.cargo/config.toml) with NDK linkers
- Fix Gradle settings (dependencyResolutionManagement), signing configs,
  Compose LinearProgressIndicator API, and Android manifest theme
- Add Gradle wrapper, .gitignore for build artifacts
- arm64-v8a only (raptorq crate incompatible with armv7 32-bit)
- Release APK: 2.0MB signed with wzp-release key
- Debug APK: 8.9MB signed with wzp-debug key

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-04 19:37:08 +00:00
Claude
e7b1c3372a feat: Android VoIP client — Phase 2 (JNI bridge, Compose UI, AEC pipeline wiring)
- JNI bridge with 8 extern functions (init, startCall, stopCall, setMute,
  setSpeaker, getStats, forceProfile, destroy) with panic catching
- Kotlin engine layer: WzpEngine JNI wrapper, WzpCallback interface,
  CallStats data class with JSON deserialization
- Jetpack Compose UI: InCallScreen with quality indicator (green/yellow/red),
  mute/speaker/hangup buttons, stats overlay, duration timer
- CallActivity with RECORD_AUDIO permission handling, Material3 theme
- CallService foreground service with WakeLock, WiFi lock, notification
- AudioRouteManager for speaker/earpiece/Bluetooth SCO switching
- AEC wired into CallEncoder pipeline: AEC → AGC → denoise → silence → encode
- AEC farend reference fed from decode path to encode path in pipeline
- Engine exposes set_aec_enabled/set_agc_enabled via AtomicBool flags

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-04 18:16:38 +00:00
Claude
26e9c55f1f feat: Android VoIP client — Phase 1 (audio quality, network adaptation, crate skeleton)
- New wzp-android crate with Oboe C++ backend, lock-free SPSC ring buffers,
  engine orchestrator, codec pipeline, and Android Gradle project structure
- AEC (NLMS adaptive filter), AGC (two-stage with fast attack/slow release),
  windowed-sinc FIR resampler replacing linear interpolation (wzp-codec)
- Opus encoder tuning: complexity 7 default, set_expected_loss support
- Mobile jitter buffer: asymmetric EMA (fast up/slow down), handoff spike
  detection with 2s cooldown, configurable safety margin
- Network-aware quality control: cellular-specific thresholds, faster
  downgrade on cellular, proactive tier drop on WiFi→cellular handoff,
  FEC ratio boost during network transitions
- Handoff detection in PathMonitor via RTT jitter spike analysis

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-04-04 18:07:55 +00:00
600 changed files with 142024 additions and 911 deletions

View File

@@ -0,0 +1,72 @@
---
name: caveman
description: >
Ultra-compressed communication mode. Slash token usage ~75% by speaking like caveman
while keeping full technical accuracy. Use when user says "caveman mode", "talk like caveman",
"use caveman", "less tokens", "be brief", or invokes /caveman. Also auto-triggers
when token efficiency is requested.
---
# Caveman Mode
## Core Rule
Respond like smart caveman. Cut articles, filler, pleasantries. Keep all technical substance.
## Grammar
- Drop articles (a, an, the)
- Drop filler (just, really, basically, actually, simply)
- Drop pleasantries (sure, certainly, of course, happy to)
- Short synonyms (big not extensive, fix not "implement a solution for")
- No hedging (skip "it might be worth considering")
- Fragments fine. No need full sentence
- Technical terms stay exact. "Polymorphism" stays "polymorphism"
- Code blocks unchanged. Caveman speak around code, not in code
- Error messages quoted exact. Caveman only for explanation
## Pattern
```
[thing] [action] [reason]. [next step].
```
Not:
> Sure! I'd be happy to help you with that. The issue you're experiencing is likely caused by...
Yes:
> Bug in auth middleware. Token expiry check use `<` not `<=`. Fix:
## Examples
**User:** Why is my React component re-rendering?
**Normal (69 tokens):** "The reason your React component is re-rendering is likely because you're creating a new object reference on each render cycle. When you pass an inline object as a prop, React's shallow comparison sees it as a different object every time, which triggers a re-render. I'd recommend using useMemo to memoize the object."
**Caveman (19 tokens):** "New object ref each render. Inline object prop = new ref = re-render. Wrap in `useMemo`."
---
**User:** How do I set up a PostgreSQL connection pool?
**Caveman:**
```
Use `pg` pool:
```
```js
const pool = new Pool({
max: 20,
idleTimeoutMillis: 30000,
connectionTimeoutMillis: 2000,
})
```
```
max = concurrent connections. Keep under DB limit. idleTimeout kill stale conn.
```
## Boundaries
- Code: write normal. Caveman English only
- Git commits: normal
- PR descriptions: normal
- User say "stop caveman" or "normal mode": revert immediately

5
.cargo/config.toml Normal file
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@@ -0,0 +1,5 @@
[target.aarch64-linux-android]
linker = "aarch64-linux-android26-clang"
[target.armv7-linux-androideabi]
linker = "armv7a-linux-androideabi26-clang"

View File

@@ -2,187 +2,57 @@ name: Build Release Binaries
on:
push:
branches:
- main
- 'feat/*'
tags:
- 'v*'
paths-ignore:
- '.gitea/**'
workflow_dispatch:
inputs:
targets:
description: 'Targets to build (comma-separated: amd64,arm64,armv7,mac-arm64)'
required: false
default: 'amd64'
env:
CARGO_TERM_COLOR: always
jobs:
# Always builds on push tags. On manual dispatch, reads inputs.
build-amd64:
if: >-
github.event_name == 'push' ||
contains(github.event.inputs.targets, 'amd64')
runs-on: ubuntu-latest
container:
image: rust:1-bookworm
image: catthehacker/ubuntu:act-latest
steps:
- uses: actions/checkout@v4
- name: Install dependencies
run: apt-get update && apt-get install -y cmake pkg-config libasound2-dev
- name: Cache cargo
uses: actions/cache@v4
with:
path: |
~/.cargo/registry
~/.cargo/git
target
key: cargo-amd64-${{ hashFiles('Cargo.lock') }}
restore-keys: cargo-amd64-
- name: Build headless binaries
run: cargo build --release --bin wzp-relay --bin wzp-client --bin wzp-bench --bin wzp-web
- name: Build audio client
- name: Init submodules
run: |
cargo build --release --bin wzp-client --features audio
cp target/release/wzp-client target/release/wzp-client-audio
cargo build --release --bin wzp-client
git config --global url."https://git.manko.yoga/".insteadOf "ssh://git@git.manko.yoga:222/"
git submodule update --init --recursive
- name: Install Rust + dependencies
run: |
curl --proto '=https' --tlsv1.2 -sSf https://sh.rustup.rs | sh -s -- -y
source "$HOME/.cargo/env"
apt-get update && apt-get install -y cmake pkg-config libasound2-dev ninja-build
rustc --version
- name: Build relay + tools
run: |
source "$HOME/.cargo/env"
cargo build --release --bin wzp-relay --bin wzp-client --bin wzp-bench --bin wzp-web
- name: Run tests
run: cargo test --workspace --lib
- name: Package
run: |
mkdir -p dist/wzp-linux-amd64
cp target/release/wzp-relay dist/wzp-linux-amd64/
cp target/release/wzp-client dist/wzp-linux-amd64/
cp target/release/wzp-client-audio dist/wzp-linux-amd64/
cp target/release/wzp-web dist/wzp-linux-amd64/
cp target/release/wzp-bench dist/wzp-linux-amd64/
cp -r crates/wzp-web/static dist/wzp-linux-amd64/
cd dist && tar czf wzp-linux-amd64.tar.gz wzp-linux-amd64/
source "$HOME/.cargo/env"
cargo test --workspace --lib
- name: Upload artifact
uses: actions/upload-artifact@v4
with:
name: wzp-linux-amd64
path: dist/wzp-linux-amd64.tar.gz
build-arm64:
if: >-
github.event_name == 'push' ||
contains(github.event.inputs.targets, 'arm64')
runs-on: ubuntu-latest
container:
image: rust:1-bookworm
steps:
- uses: actions/checkout@v4
- name: Install cross-compilation tools
run: |
dpkg --add-architecture arm64
apt-get update
apt-get install -y cmake pkg-config gcc-aarch64-linux-gnu libc6-dev-arm64-cross
rustup target add aarch64-unknown-linux-gnu
- name: Cache cargo
uses: actions/cache@v4
with:
path: |
~/.cargo/registry
~/.cargo/git
target
key: cargo-arm64-${{ hashFiles('Cargo.lock') }}
restore-keys: cargo-arm64-
- name: Build
- name: Upload to rustypaste
env:
CARGO_TARGET_AARCH64_UNKNOWN_LINUX_GNU_LINKER: aarch64-linux-gnu-gcc
CC_aarch64_unknown_linux_gnu: aarch64-linux-gnu-gcc
PASTE_AUTH: ${{ secrets.PASTE_AUTH }}
PASTE_URL: ${{ secrets.PASTE_URL }}
run: |
cargo build --release --target aarch64-unknown-linux-gnu \
--bin wzp-relay --bin wzp-client --bin wzp-bench --bin wzp-web
- name: Package
run: |
mkdir -p dist/wzp-linux-arm64
cp target/aarch64-unknown-linux-gnu/release/wzp-relay dist/wzp-linux-arm64/
cp target/aarch64-unknown-linux-gnu/release/wzp-client dist/wzp-linux-arm64/
cp target/aarch64-unknown-linux-gnu/release/wzp-web dist/wzp-linux-arm64/
cp target/aarch64-unknown-linux-gnu/release/wzp-bench dist/wzp-linux-arm64/
cp -r crates/wzp-web/static dist/wzp-linux-arm64/
cd dist && tar czf wzp-linux-arm64.tar.gz wzp-linux-arm64/
- name: Upload artifact
uses: actions/upload-artifact@v4
with:
name: wzp-linux-arm64
path: dist/wzp-linux-arm64.tar.gz
build-armv7:
if: >-
github.event_name == 'push' ||
contains(github.event.inputs.targets, 'armv7')
runs-on: ubuntu-latest
container:
image: rust:1-bookworm
steps:
- uses: actions/checkout@v4
- name: Install cross-compilation tools
run: |
dpkg --add-architecture armhf
apt-get update
apt-get install -y cmake pkg-config gcc-arm-linux-gnueabihf libc6-dev-armhf-cross
rustup target add armv7-unknown-linux-gnueabihf
- name: Cache cargo
uses: actions/cache@v4
with:
path: |
~/.cargo/registry
~/.cargo/git
target
key: cargo-armv7-${{ hashFiles('Cargo.lock') }}
restore-keys: cargo-armv7-
- name: Build
env:
CARGO_TARGET_ARMV7_UNKNOWN_LINUX_GNUEABIHF_LINKER: arm-linux-gnueabihf-gcc
CC_armv7_unknown_linux_gnueabihf: arm-linux-gnueabihf-gcc
run: |
cargo build --release --target armv7-unknown-linux-gnueabihf \
--bin wzp-relay --bin wzp-client --bin wzp-bench --bin wzp-web
- name: Package
run: |
mkdir -p dist/wzp-linux-armv7
cp target/armv7-unknown-linux-gnueabihf/release/wzp-relay dist/wzp-linux-armv7/
cp target/armv7-unknown-linux-gnueabihf/release/wzp-client dist/wzp-linux-armv7/
cp target/armv7-unknown-linux-gnueabihf/release/wzp-web dist/wzp-linux-armv7/
cp target/armv7-unknown-linux-gnueabihf/release/wzp-bench dist/wzp-linux-armv7/
cp -r crates/wzp-web/static dist/wzp-linux-armv7/
cd dist && tar czf wzp-linux-armv7.tar.gz wzp-linux-armv7/
- name: Upload artifact
uses: actions/upload-artifact@v4
with:
name: wzp-linux-armv7
path: dist/wzp-linux-armv7.tar.gz
# Release job — creates a release with all artifacts when a tag is pushed
release:
if: startsWith(github.ref, 'refs/tags/v')
needs: [build-amd64]
runs-on: ubuntu-latest
steps:
- name: Download all artifacts
uses: actions/download-artifact@v4
with:
path: artifacts
- name: Create release
uses: softprops/action-gh-release@v2
with:
files: artifacts/**/*.tar.gz
generate_release_notes: true
tar czf /tmp/wzp-linux-amd64.tar.gz \
-C target/release wzp-relay wzp-client wzp-web wzp-bench
ls -lh /tmp/wzp-linux-amd64.tar.gz
LINK=$(curl -sF "file=@/tmp/wzp-linux-amd64.tar.gz" \
-H "Authorization: ${PASTE_AUTH}" \
"https://${PASTE_URL}")
echo "Download: ${LINK}"

View File

@@ -0,0 +1,43 @@
name: Mirror to GitHub
on:
push:
branches:
- main
- 'feat/*'
- 'feature/*'
tags:
- '*'
jobs:
mirror:
runs-on: ubuntu-latest
container:
image: catthehacker/ubuntu:act-latest
steps:
- uses: actions/checkout@v4
with:
fetch-depth: 0
- name: Push to GitHub
env:
GH_SSH_KEY: ${{ secrets.GH_SSH_KEY }}
run: |
mkdir -p ~/.ssh
echo "${GH_SSH_KEY}" > ~/.ssh/id_ed25519
chmod 600 ~/.ssh/id_ed25519
ssh-keyscan github.com >> ~/.ssh/known_hosts 2>/dev/null
git remote add github git@github.com:manawenuz/wzp.git
# Push the current branch
BRANCH="${GITHUB_REF#refs/heads/}"
TAG="${GITHUB_REF#refs/tags/}"
if [ "${GITHUB_REF}" != "${GITHUB_REF#refs/tags/}" ]; then
echo "Pushing tag: ${TAG}"
git push github "refs/tags/${TAG}" --force
else
echo "Pushing branch: ${BRANCH}"
git push github "HEAD:refs/heads/${BRANCH}" --force
fi

25
.gitignore vendored
View File

@@ -4,3 +4,28 @@
*.swp
*.swo
*~
# Logs
logs
*.log
npm-debug.log*
yarn-debug.log*
yarn-error.log*
dev-debug.log
# Dependency directories
node_modules/
# Environment variables
.env
# Editor directories and files
.idea
.vscode
*.suo
*.ntvs*
*.njsproj
*.sln
*.sw?
# OS specific
# Taskmaster (local workflow tool)
.taskmaster/
.env.example

3247
Cargo.lock generated

File diff suppressed because it is too large Load Diff

View File

@@ -9,6 +9,9 @@ members = [
"crates/wzp-relay",
"crates/wzp-client",
"crates/wzp-web",
"crates/wzp-android",
"crates/wzp-native",
"desktop/src-tauri",
]
[workspace.package]
@@ -39,7 +42,7 @@ codec2 = "0.3"
# Crypto
x25519-dalek = { version = "2", features = ["static_secrets"] }
ed25519-dalek = { version = "2", features = ["rand_core"] }
ed25519-dalek = { version = "2", features = ["rand_core", "pkcs8"] }
chacha20poly1305 = "0.10"
hkdf = "0.12"
sha2 = "0.10"
@@ -52,3 +55,37 @@ wzp-fec = { path = "crates/wzp-fec" }
wzp-crypto = { path = "crates/wzp-crypto" }
wzp-transport = { path = "crates/wzp-transport" }
wzp-client = { path = "crates/wzp-client" }
# Fast dev profile: optimized but with debug info and incremental compilation.
# Use with: cargo run --profile dev-fast
[profile.dev-fast]
inherits = "dev"
opt-level = 2
# Optimize heavy compute deps even in debug builds —
# real-time audio needs < 20ms per frame, impossible unoptimized.
[profile.dev.package.nnnoiseless]
opt-level = 3
[profile.dev.package.audiopus_sys]
opt-level = 3
[profile.dev.package.audiopus]
opt-level = 3
[profile.dev.package.raptorq]
opt-level = 3
[profile.dev.package.wzp-codec]
opt-level = 3
[profile.dev.package.wzp-fec]
opt-level = 3
# Vendored audiopus_sys with a patched opus/CMakeLists.txt that distinguishes
# real cl.exe (MSVC) from clang-cl (used by cargo-xwin for Windows cross-
# compiles). Upstream libopus 1.3.1 gates its `-msse4.1` per-file compile
# flags on `if(NOT MSVC)`, which is false under clang-cl because CMake sets
# MSVC=1 for both compilers — resulting in SSE4.1 source files compiled
# without the required target feature and hard failures in silk/NSQ_sse4_1.c.
# The vendored copy introduces an `MSVC_CL` var (true only for real cl.exe)
# and flips the SIMD guards to use it, restoring per-file SIMD flags for
# clang-cl. See vendor/audiopus_sys/opus/CMakeLists.txt for the full diff
# and rationale, plus xiph/opus#256 / xiph/opus PR #257 upstream.
[patch.crates-io]
audiopus_sys = { path = "vendor/audiopus_sys" }

6
android/.gitignore vendored Normal file
View File

@@ -0,0 +1,6 @@
.gradle/
build/
app/build/
app/src/main/jniLibs/
local.properties
keystore/*.jks

View File

@@ -0,0 +1,85 @@
plugins {
id("com.android.application")
id("org.jetbrains.kotlin.android")
}
android {
namespace = "com.wzp.phone"
compileSdk = 34
defaultConfig {
applicationId = "com.wzp.phone"
minSdk = 26 // AAudio requires API 26
targetSdk = 34
versionCode = 1
versionName = "0.1.0"
ndk { abiFilters += listOf("arm64-v8a") }
}
signingConfigs {
create("release") {
storeFile = file("${project.rootDir}/keystore/wzp-release.jks")
storePassword = "wzphone2024"
keyAlias = "wzp-release"
keyPassword = "wzphone2024"
}
getByName("debug") {
storeFile = file("${project.rootDir}/keystore/wzp-debug.jks")
storePassword = "android"
keyAlias = "wzp-debug"
keyPassword = "android"
}
}
buildTypes {
debug {
signingConfig = signingConfigs.getByName("debug")
isDebuggable = true
}
release {
signingConfig = signingConfigs.getByName("release")
isMinifyEnabled = false
proguardFiles(
getDefaultProguardFile("proguard-android-optimize.txt"),
"proguard-rules.pro"
)
}
}
compileOptions {
sourceCompatibility = JavaVersion.VERSION_1_8
targetCompatibility = JavaVersion.VERSION_1_8
}
kotlinOptions {
jvmTarget = "1.8"
}
buildFeatures { compose = true }
composeOptions { kotlinCompilerExtensionVersion = "1.5.8" }
ndkVersion = "26.1.10909125"
}
// cargo-ndk integration: build the Rust native library for Android targets
tasks.register<Exec>("cargoNdkBuild") {
workingDir = file("${project.rootDir}/..")
commandLine(
"cargo", "ndk",
"-t", "arm64-v8a",
"-o", "${project.projectDir}/src/main/jniLibs",
"build", "--release", "-p", "wzp-android"
)
}
// Skip cargo-ndk in CI/Docker — .so is pre-built into jniLibs
// tasks.named("preBuild") { dependsOn("cargoNdkBuild") }
dependencies {
implementation("androidx.core:core-ktx:1.12.0")
implementation("androidx.lifecycle:lifecycle-runtime-ktx:2.7.0")
implementation("androidx.activity:activity-compose:1.8.2")
implementation(platform("androidx.compose:compose-bom:2024.01.00"))
implementation("androidx.compose.ui:ui")
implementation("androidx.compose.material3:material3")
}

9
android/app/proguard-rules.pro vendored Normal file
View File

@@ -0,0 +1,9 @@
# WZPhone ProGuard rules
# Keep JNI native methods
-keepclasseswithmembernames class * {
native <methods>;
}
# Keep the WZP engine bridge class
-keep class com.wzp.phone.engine.** { *; }

View File

@@ -0,0 +1,43 @@
<?xml version="1.0" encoding="utf-8"?>
<manifest xmlns:android="http://schemas.android.com/apk/res/android">
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
<uses-permission android:name="android.permission.FOREGROUND_SERVICE" />
<uses-permission android:name="android.permission.FOREGROUND_SERVICE_MICROPHONE" />
<uses-permission android:name="android.permission.WAKE_LOCK" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
<uses-permission android:name="android.permission.BLUETOOTH_CONNECT" />
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<application
android:name="com.wzp.WzpApplication"
android:label="WZ Phone"
android:supportsRtl="true"
android:theme="@android:style/Theme.Material.Light.NoActionBar">
<activity
android:name="com.wzp.ui.call.CallActivity"
android:exported="true"
android:launchMode="singleTask">
<intent-filter>
<action android:name="android.intent.action.MAIN" />
<category android:name="android.intent.category.LAUNCHER" />
</intent-filter>
</activity>
<service
android:name="com.wzp.service.CallService"
android:foregroundServiceType="microphone"
android:exported="false" />
<provider
android:name="androidx.core.content.FileProvider"
android:authorities="${applicationId}.fileprovider"
android:exported="false"
android:grantUriPermissions="true">
<meta-data
android:name="android.support.FILE_PROVIDER_PATHS"
android:resource="@xml/file_paths" />
</provider>
</application>
</manifest>

View File

@@ -0,0 +1,38 @@
package com.wzp
import android.app.Application
import android.app.NotificationChannel
import android.app.NotificationManager
import android.os.Build
/**
* Application entry point for WarzonePhone.
*
* Creates the notification channel required for the foreground [com.wzp.service.CallService].
*/
class WzpApplication : Application() {
override fun onCreate() {
super.onCreate()
createNotificationChannel()
}
private fun createNotificationChannel() {
if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.O) {
val channel = NotificationChannel(
CHANNEL_ID,
"Active Call",
NotificationManager.IMPORTANCE_LOW
).apply {
description = "Shown while a VoIP call is in progress"
setShowBadge(false)
}
val nm = getSystemService(NotificationManager::class.java)
nm.createNotificationChannel(channel)
}
}
companion object {
const val CHANNEL_ID = "wzp_call_channel"
}
}

View File

@@ -0,0 +1,359 @@
package com.wzp.audio
import android.Manifest
import android.content.Context
import android.content.pm.PackageManager
import android.media.AudioAttributes
import android.media.AudioFormat
import android.media.AudioRecord
import android.media.AudioTrack
import android.media.MediaRecorder
import android.media.audiofx.AcousticEchoCanceler
import android.media.audiofx.NoiseSuppressor
import android.util.Log
import androidx.core.content.ContextCompat
import com.wzp.engine.WzpEngine
import java.io.BufferedOutputStream
import java.io.File
import java.io.FileOutputStream
import java.io.OutputStreamWriter
import java.nio.ByteBuffer
import java.nio.ByteOrder
import java.util.concurrent.CountDownLatch
import java.util.concurrent.TimeUnit
import kotlin.math.pow
import kotlin.math.sqrt
/**
* Audio pipeline that captures mic audio and plays received audio using
* Android AudioRecord/AudioTrack APIs running on JVM threads.
*
* PCM samples are shuttled to/from the Rust engine via JNI ring buffers:
* - Capture: AudioRecord → WzpEngine.writeAudio() → Rust encoder → network
* - Playout: network → Rust decoder → WzpEngine.readAudio() → AudioTrack
*
* All audio is 48kHz, mono, 16-bit PCM (matching Opus codec requirements).
*/
class AudioPipeline(private val context: Context) {
companion object {
private const val TAG = "AudioPipeline"
private const val SAMPLE_RATE = 48000
private const val CHANNEL_IN = AudioFormat.CHANNEL_IN_MONO
private const val CHANNEL_OUT = AudioFormat.CHANNEL_OUT_MONO
private const val ENCODING = AudioFormat.ENCODING_PCM_16BIT
/** 20ms frame at 48kHz = 960 samples */
private const val FRAME_SAMPLES = 960
}
@Volatile
private var running = false
/** Playout (incoming voice) gain in dB. 0 = unity. */
@Volatile
var playoutGainDb: Float = 0f
/** Capture (mic) gain in dB. 0 = unity. */
@Volatile
var captureGainDb: Float = 0f
/** Whether to attach hardware AEC. Must be set before start(). */
var aecEnabled: Boolean = true
/** Enable debug recording of PCM + RMS histogram to cache dir. */
var debugRecording: Boolean = false
private var captureThread: Thread? = null
private var playoutThread: Thread? = null
// DirectByteBuffers for zero-copy JNI audio transfer.
// Allocated as class fields (NOT locals) because ART's JIT OSR
// can null local variables when it replaces the stack frame mid-loop.
// These survive OSR because they're on the heap.
private val captureDirectBuf: ByteBuffer =
ByteBuffer.allocateDirect(FRAME_SAMPLES * 2).order(ByteOrder.LITTLE_ENDIAN)
private val playoutDirectBuf: ByteBuffer =
ByteBuffer.allocateDirect(FRAME_SAMPLES * 2).order(ByteOrder.LITTLE_ENDIAN)
/** Latch counted down by each audio thread after exiting its loop.
* stop() does NOT wait on this — teardown waits via awaitDrain(). */
private var drainLatch: CountDownLatch? = null
private val debugDir: File by lazy {
File(context.cacheDir, "wzp_debug").also { it.mkdirs() }
}
fun start(engine: WzpEngine) {
if (running) return
running = true
drainLatch = CountDownLatch(2) // one for capture, one for playout
captureThread = Thread({
runCapture(engine)
drainLatch?.countDown() // signal: capture loop exited, no more JNI calls
// Park thread forever — exiting triggers a libcrypto TLS destructor
// crash (SIGSEGV in OPENSSL_free) on Android when a JNI-calling thread exits.
parkThread()
}, "wzp-capture").apply {
isDaemon = true
priority = Thread.MAX_PRIORITY
start()
}
playoutThread = Thread({
runPlayout(engine)
drainLatch?.countDown() // signal: playout loop exited
parkThread()
}, "wzp-playout").apply {
isDaemon = true
priority = Thread.MAX_PRIORITY
start()
}
Log.i(TAG, "audio pipeline started")
}
fun stop() {
running = false
// Don't join threads — they are parked as daemons to avoid native TLS crash.
// Don't null thread refs or drainLatch — teardown() needs awaitDrain().
Log.i(TAG, "audio pipeline stopped (running=false)")
}
/** Block until both audio threads have exited their loops (max 200ms).
* After this returns, no more JNI calls to the engine will be made. */
fun awaitDrain(): Boolean {
val ok = drainLatch?.await(200, TimeUnit.MILLISECONDS) ?: true
if (!ok) Log.w(TAG, "awaitDrain: audio threads did not drain in 200ms")
captureThread = null
playoutThread = null
drainLatch = null
return ok
}
private fun applyGain(pcm: ShortArray, count: Int, db: Float) {
if (db == 0f) return
val linear = 10f.pow(db / 20f)
for (i in 0 until count) {
pcm[i] = (pcm[i] * linear).toInt().coerceIn(-32000, 32000).toShort()
}
}
private fun computeRms(pcm: ShortArray, count: Int): Int {
var sumSq = 0.0
for (i in 0 until count) {
val s = pcm[i].toDouble()
sumSq += s * s
}
return sqrt(sumSq / count).toInt()
}
private fun parkThread() {
try {
Thread.sleep(Long.MAX_VALUE)
} catch (_: InterruptedException) {
// process exiting
}
}
private fun runCapture(engine: WzpEngine) {
if (ContextCompat.checkSelfPermission(context, Manifest.permission.RECORD_AUDIO)
!= PackageManager.PERMISSION_GRANTED
) {
Log.e(TAG, "RECORD_AUDIO permission not granted, capture disabled")
return
}
val minBuf = AudioRecord.getMinBufferSize(SAMPLE_RATE, CHANNEL_IN, ENCODING)
val bufSize = maxOf(minBuf, FRAME_SAMPLES * 2 * 4) // at least 4 frames
val recorder = try {
AudioRecord(
MediaRecorder.AudioSource.VOICE_COMMUNICATION,
SAMPLE_RATE,
CHANNEL_IN,
ENCODING,
bufSize
)
} catch (e: SecurityException) {
Log.e(TAG, "AudioRecord SecurityException: ${e.message}")
return
}
if (recorder.state != AudioRecord.STATE_INITIALIZED) {
Log.e(TAG, "AudioRecord failed to initialize")
recorder.release()
return
}
// Attach hardware AEC if available and enabled in settings
var aec: AcousticEchoCanceler? = null
var ns: NoiseSuppressor? = null
if (aecEnabled) {
if (AcousticEchoCanceler.isAvailable()) {
try {
aec = AcousticEchoCanceler.create(recorder.audioSessionId)
aec?.enabled = true
Log.i(TAG, "AEC enabled (session=${recorder.audioSessionId})")
} catch (e: Exception) {
Log.w(TAG, "AEC init failed: ${e.message}")
}
} else {
Log.w(TAG, "AEC not available on this device")
}
// Attach hardware noise suppressor if available
if (NoiseSuppressor.isAvailable()) {
try {
ns = NoiseSuppressor.create(recorder.audioSessionId)
ns?.enabled = true
Log.i(TAG, "NoiseSuppressor enabled")
} catch (e: Exception) {
Log.w(TAG, "NoiseSuppressor init failed: ${e.message}")
}
}
} else {
Log.i(TAG, "AEC disabled by user setting")
}
recorder.startRecording()
Log.i(TAG, "capture started: ${SAMPLE_RATE}Hz mono, buf=$bufSize, aec=${aec?.enabled}, ns=${ns?.enabled}")
val pcm = ShortArray(FRAME_SAMPLES)
// Debug: PCM file + RMS CSV
var pcmOut: BufferedOutputStream? = null
var rmsCsv: OutputStreamWriter? = null
val byteConv = ByteBuffer.allocate(FRAME_SAMPLES * 2).order(ByteOrder.LITTLE_ENDIAN)
var frameIdx = 0L
if (debugRecording) {
try {
pcmOut = BufferedOutputStream(FileOutputStream(File(debugDir, "capture.pcm")), 65536)
rmsCsv = OutputStreamWriter(FileOutputStream(File(debugDir, "capture_rms.csv")))
rmsCsv.write("frame,time_ms,rms\n")
} catch (e: Exception) {
Log.w(TAG, "debug recording init failed: ${e.message}")
}
}
try {
while (running) {
val read = recorder.read(pcm, 0, FRAME_SAMPLES)
if (read > 0) {
applyGain(pcm, read, captureGainDb)
// Zero-copy write via DirectByteBuffer (class field, survives JIT OSR)
captureDirectBuf.clear()
captureDirectBuf.asShortBuffer().put(pcm, 0, read)
engine.writeAudioDirect(captureDirectBuf, read)
// Debug: write raw PCM + RMS
if (pcmOut != null) {
byteConv.clear()
for (i in 0 until read) byteConv.putShort(pcm[i])
pcmOut.write(byteConv.array(), 0, read * 2)
}
if (rmsCsv != null) {
val rms = computeRms(pcm, read)
val timeMs = frameIdx * FRAME_SAMPLES * 1000L / SAMPLE_RATE
rmsCsv.write("$frameIdx,$timeMs,$rms\n")
}
frameIdx++
} else if (read < 0) {
Log.e(TAG, "AudioRecord.read error: $read")
break
}
}
} finally {
pcmOut?.close()
rmsCsv?.close()
recorder.stop()
aec?.release()
ns?.release()
recorder.release()
Log.i(TAG, "capture stopped (frames=$frameIdx)")
}
}
private fun runPlayout(engine: WzpEngine) {
val minBuf = AudioTrack.getMinBufferSize(SAMPLE_RATE, CHANNEL_OUT, ENCODING)
val bufSize = maxOf(minBuf, FRAME_SAMPLES * 2 * 4)
val track = AudioTrack.Builder()
.setAudioAttributes(
AudioAttributes.Builder()
.setUsage(AudioAttributes.USAGE_VOICE_COMMUNICATION)
.setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)
.build()
)
.setAudioFormat(
AudioFormat.Builder()
.setSampleRate(SAMPLE_RATE)
.setChannelMask(CHANNEL_OUT)
.setEncoding(ENCODING)
.build()
)
.setBufferSizeInBytes(bufSize)
.setTransferMode(AudioTrack.MODE_STREAM)
.build()
if (track.state != AudioTrack.STATE_INITIALIZED) {
Log.e(TAG, "AudioTrack failed to initialize")
track.release()
return
}
track.play()
Log.i(TAG, "playout started: ${SAMPLE_RATE}Hz mono, buf=$bufSize")
val pcm = ShortArray(FRAME_SAMPLES)
val silence = ShortArray(FRAME_SAMPLES)
// Debug: PCM file + RMS CSV for playout
var pcmOut: BufferedOutputStream? = null
var rmsCsv: OutputStreamWriter? = null
val byteConv = ByteBuffer.allocate(FRAME_SAMPLES * 2).order(ByteOrder.LITTLE_ENDIAN)
var frameIdx = 0L
if (debugRecording) {
try {
pcmOut = BufferedOutputStream(FileOutputStream(File(debugDir, "playout.pcm")), 65536)
rmsCsv = OutputStreamWriter(FileOutputStream(File(debugDir, "playout_rms.csv")))
rmsCsv.write("frame,time_ms,rms\n")
} catch (e: Exception) {
Log.w(TAG, "debug playout recording init failed: ${e.message}")
}
}
try {
while (running) {
// Zero-copy read via DirectByteBuffer (class field, survives JIT OSR)
playoutDirectBuf.clear()
val read = engine.readAudioDirect(playoutDirectBuf, FRAME_SAMPLES)
if (read >= FRAME_SAMPLES) {
playoutDirectBuf.rewind()
playoutDirectBuf.asShortBuffer().get(pcm, 0, read)
applyGain(pcm, read, playoutGainDb)
track.write(pcm, 0, read)
// Debug: write raw PCM + RMS
if (pcmOut != null) {
byteConv.clear()
for (i in 0 until read) byteConv.putShort(pcm[i])
pcmOut.write(byteConv.array(), 0, read * 2)
}
if (rmsCsv != null) {
val rms = computeRms(pcm, read)
val timeMs = frameIdx * FRAME_SAMPLES * 1000L / SAMPLE_RATE
rmsCsv.write("$frameIdx,$timeMs,$rms\n")
}
frameIdx++
} else {
track.write(silence, 0, FRAME_SAMPLES)
// Log silence frames to RMS as 0
if (rmsCsv != null) {
val timeMs = frameIdx * FRAME_SAMPLES * 1000L / SAMPLE_RATE
rmsCsv.write("$frameIdx,$timeMs,0\n")
}
frameIdx++
Thread.sleep(5)
}
}
} finally {
pcmOut?.close()
rmsCsv?.close()
track.stop()
track.release()
Log.i(TAG, "playout stopped (frames=$frameIdx)")
}
}
}

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package com.wzp.audio
import android.content.Context
import android.media.AudioDeviceCallback
import android.media.AudioDeviceInfo
import android.media.AudioManager
import android.os.Handler
import android.os.Looper
/**
* Manages audio routing between earpiece, speaker, and Bluetooth devices.
*
* Wraps [AudioManager] operations and listens for device connection changes
* via [AudioDeviceCallback] (API 23+).
*
* Usage:
* 1. Call [register] when the call starts
* 2. Use [setSpeaker] and [setBluetoothSco] to switch routes
* 3. Call [unregister] when the call ends
*/
class AudioRouteManager(context: Context) {
private val audioManager = context.getSystemService(Context.AUDIO_SERVICE) as AudioManager
private val mainHandler = Handler(Looper.getMainLooper())
/** Listener for audio route changes. */
var onRouteChanged: ((AudioRoute) -> Unit)? = null
/** Current active route. */
var currentRoute: AudioRoute = AudioRoute.EARPIECE
private set
// -- Device callback (API 23+) -------------------------------------------
private val deviceCallback = object : AudioDeviceCallback() {
override fun onAudioDevicesAdded(addedDevices: Array<out AudioDeviceInfo>) {
for (device in addedDevices) {
if (device.type == AudioDeviceInfo.TYPE_BLUETOOTH_SCO) {
// A Bluetooth headset was connected — optionally auto-switch
onRouteChanged?.invoke(AudioRoute.BLUETOOTH)
}
}
}
override fun onAudioDevicesRemoved(removedDevices: Array<out AudioDeviceInfo>) {
for (device in removedDevices) {
if (device.type == AudioDeviceInfo.TYPE_BLUETOOTH_SCO) {
// Bluetooth disconnected — fall back to earpiece or speaker
val fallback = if (audioManager.isSpeakerphoneOn) {
AudioRoute.SPEAKER
} else {
AudioRoute.EARPIECE
}
currentRoute = fallback
onRouteChanged?.invoke(fallback)
}
}
}
}
// -- Public API -----------------------------------------------------------
/** Register the device callback. Call when a call starts. */
fun register() {
audioManager.registerAudioDeviceCallback(deviceCallback, mainHandler)
}
/** Unregister the device callback and release Bluetooth SCO. Call when the call ends. */
fun unregister() {
audioManager.unregisterAudioDeviceCallback(deviceCallback)
stopBluetoothSco()
}
/**
* Enable or disable the loudspeaker.
*
* When enabling speaker, Bluetooth SCO is disconnected.
*/
@Suppress("DEPRECATION")
fun setSpeaker(enabled: Boolean) {
if (enabled) {
stopBluetoothSco()
}
audioManager.isSpeakerphoneOn = enabled
currentRoute = if (enabled) AudioRoute.SPEAKER else AudioRoute.EARPIECE
onRouteChanged?.invoke(currentRoute)
}
/**
* Enable or disable Bluetooth SCO (Synchronous Connection Oriented) audio.
*
* When enabling Bluetooth, the speaker is turned off.
*/
@Suppress("DEPRECATION")
fun setBluetoothSco(enabled: Boolean) {
if (enabled) {
audioManager.isSpeakerphoneOn = false
audioManager.startBluetoothSco()
audioManager.isBluetoothScoOn = true
currentRoute = AudioRoute.BLUETOOTH
} else {
stopBluetoothSco()
currentRoute = AudioRoute.EARPIECE
}
onRouteChanged?.invoke(currentRoute)
}
/** Check whether a Bluetooth SCO device is currently connected. */
fun isBluetoothAvailable(): Boolean {
val devices = audioManager.getDevices(AudioManager.GET_DEVICES_OUTPUTS)
return devices.any { it.type == AudioDeviceInfo.TYPE_BLUETOOTH_SCO }
}
/** List available output audio routes. */
fun availableRoutes(): List<AudioRoute> {
val routes = mutableListOf(AudioRoute.EARPIECE, AudioRoute.SPEAKER)
if (isBluetoothAvailable()) {
routes.add(AudioRoute.BLUETOOTH)
}
return routes
}
// -- Internal -------------------------------------------------------------
@Suppress("DEPRECATION")
private fun stopBluetoothSco() {
if (audioManager.isBluetoothScoOn) {
audioManager.isBluetoothScoOn = false
audioManager.stopBluetoothSco()
}
}
}
/** Audio output route. */
enum class AudioRoute {
/** Phone earpiece (default for calls). */
EARPIECE,
/** Built-in loudspeaker. */
SPEAKER,
/** Bluetooth SCO headset/headphones. */
BLUETOOTH
}

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package com.wzp.data
import android.content.Context
import android.content.SharedPreferences
import com.wzp.ui.call.ServerEntry
import org.json.JSONArray
import org.json.JSONObject
import java.security.SecureRandom
/**
* Persists user settings via SharedPreferences.
*
* Stores: servers, default server index, room name, alias, gain values,
* IPv6 preference, and the identity seed (hex-encoded 32 bytes).
*/
class SettingsRepository(context: Context) {
private val prefs: SharedPreferences =
context.applicationContext.getSharedPreferences("wzp_settings", Context.MODE_PRIVATE)
companion object {
private const val KEY_SERVERS = "servers_json"
private const val KEY_SELECTED_SERVER = "selected_server"
private const val KEY_ROOM = "room_name"
private const val KEY_ALIAS = "alias"
private const val KEY_PLAYOUT_GAIN = "playout_gain_db"
private const val KEY_CAPTURE_GAIN = "capture_gain_db"
private const val KEY_PREFER_IPV6 = "prefer_ipv6"
private const val KEY_IDENTITY_SEED = "identity_seed_hex"
private const val KEY_AEC_ENABLED = "aec_enabled"
private const val KEY_DEBUG_RECORDING = "debug_recording"
private const val KEY_RECENT_ROOMS = "recent_rooms"
private const val TOFU_PREFIX = "tofu_"
}
// --- Servers ---
fun saveServers(servers: List<ServerEntry>) {
val arr = JSONArray()
servers.forEach { entry ->
arr.put(JSONObject().apply {
put("address", entry.address)
put("label", entry.label)
})
}
prefs.edit().putString(KEY_SERVERS, arr.toString()).apply()
}
fun loadServers(): List<ServerEntry>? {
val json = prefs.getString(KEY_SERVERS, null) ?: return null
return try {
val arr = JSONArray(json)
(0 until arr.length()).map { i ->
val obj = arr.getJSONObject(i)
ServerEntry(obj.getString("address"), obj.getString("label"))
}
} catch (_: Exception) { null }
}
fun saveSelectedServer(index: Int) {
prefs.edit().putInt(KEY_SELECTED_SERVER, index).apply()
}
fun loadSelectedServer(): Int = prefs.getInt(KEY_SELECTED_SERVER, 0)
// --- Room ---
fun saveRoom(name: String) { prefs.edit().putString(KEY_ROOM, name).apply() }
fun loadRoom(): String = prefs.getString(KEY_ROOM, "android") ?: "android"
// --- Alias ---
fun saveAlias(alias: String) { prefs.edit().putString(KEY_ALIAS, alias).apply() }
/**
* Load alias, generating a random name on first launch.
*/
fun getOrCreateAlias(): String {
val existing = prefs.getString(KEY_ALIAS, null)
if (!existing.isNullOrEmpty()) return existing
val name = generateRandomName()
prefs.edit().putString(KEY_ALIAS, name).apply()
return name
}
private fun generateRandomName(): String {
val adjectives = listOf(
"Swift", "Silent", "Brave", "Calm", "Dark", "Fierce", "Ghost",
"Iron", "Lucky", "Noble", "Quick", "Sharp", "Storm", "Wild",
"Cold", "Bright", "Lone", "Red", "Grey", "Frosty", "Dusty",
"Rusty", "Neon", "Void", "Solar", "Lunar", "Cyber", "Pixel",
"Sonic", "Hyper", "Turbo", "Nano", "Mega", "Ultra", "Zinc"
)
val nouns = listOf(
"Wolf", "Hawk", "Fox", "Bear", "Lynx", "Crow", "Viper",
"Cobra", "Tiger", "Eagle", "Shark", "Raven", "Falcon", "Otter",
"Mantis", "Panda", "Jackal", "Badger", "Heron", "Bison",
"Condor", "Coyote", "Gecko", "Hornet", "Marten", "Osprey",
"Parrot", "Puma", "Raptor", "Stork", "Toucan", "Walrus"
)
val adj = adjectives.random()
val noun = nouns.random()
return "$adj $noun"
}
// --- Gain ---
fun savePlayoutGain(db: Float) { prefs.edit().putFloat(KEY_PLAYOUT_GAIN, db).apply() }
fun loadPlayoutGain(): Float = prefs.getFloat(KEY_PLAYOUT_GAIN, 0f)
fun saveCaptureGain(db: Float) { prefs.edit().putFloat(KEY_CAPTURE_GAIN, db).apply() }
fun loadCaptureGain(): Float = prefs.getFloat(KEY_CAPTURE_GAIN, 0f)
// --- IPv6 ---
fun savePreferIPv6(prefer: Boolean) { prefs.edit().putBoolean(KEY_PREFER_IPV6, prefer).apply() }
fun loadPreferIPv6(): Boolean = prefs.getBoolean(KEY_PREFER_IPV6, false)
// --- AEC ---
fun saveAecEnabled(enabled: Boolean) { prefs.edit().putBoolean(KEY_AEC_ENABLED, enabled).apply() }
fun loadAecEnabled(): Boolean = prefs.getBoolean(KEY_AEC_ENABLED, true)
// --- Debug recording ---
fun saveDebugRecording(enabled: Boolean) { prefs.edit().putBoolean(KEY_DEBUG_RECORDING, enabled).apply() }
fun loadDebugRecording(): Boolean = prefs.getBoolean(KEY_DEBUG_RECORDING, false)
// --- Codec choice ---
// 0 = Opus (GOOD), 1 = Opus Low (DEGRADED), 2 = Codec2 (CATASTROPHIC)
fun saveCodecChoice(choice: Int) { prefs.edit().putInt("codec_choice", choice).apply() }
fun loadCodecChoice(): Int = prefs.getInt("codec_choice", 0)
// --- Identity seed ---
/**
* Get or generate the identity seed. On first call, generates a random
* 32-byte seed and persists it. Subsequent calls return the same seed.
*/
fun getOrCreateSeedHex(): String {
val existing = prefs.getString(KEY_IDENTITY_SEED, null)
if (!existing.isNullOrEmpty()) return existing
val seed = ByteArray(32).also { SecureRandom().nextBytes(it) }
val hex = seed.joinToString("") { "%02x".format(it) }
prefs.edit().putString(KEY_IDENTITY_SEED, hex).apply()
return hex
}
fun loadSeedHex(): String = prefs.getString(KEY_IDENTITY_SEED, "") ?: ""
fun saveSeedHex(hex: String) {
prefs.edit().putString(KEY_IDENTITY_SEED, hex).apply()
}
// --- Recent rooms ---
data class RecentRoom(val relay: String, val room: String)
fun addRecentRoom(relay: String, room: String) {
val rooms = loadRecentRooms().toMutableList()
rooms.removeAll { it.relay == relay && it.room == room }
rooms.add(0, RecentRoom(relay, room))
if (rooms.size > 5) rooms.subList(5, rooms.size).clear()
val arr = JSONArray()
rooms.forEach { arr.put(JSONObject().apply { put("relay", it.relay); put("room", it.room) }) }
prefs.edit().putString(KEY_RECENT_ROOMS, arr.toString()).apply()
}
fun loadRecentRooms(): List<RecentRoom> {
val json = prefs.getString(KEY_RECENT_ROOMS, null) ?: return emptyList()
return try {
val arr = JSONArray(json)
(0 until arr.length()).map { i ->
val o = arr.getJSONObject(i)
RecentRoom(o.getString("relay"), o.getString("room"))
}
} catch (_: Exception) { emptyList() }
}
fun clearRecentRooms() {
prefs.edit().remove(KEY_RECENT_ROOMS).apply()
}
// --- Server fingerprint TOFU ---
fun saveServerFingerprint(address: String, fingerprint: String) {
prefs.edit().putString("$TOFU_PREFIX$address", fingerprint).apply()
}
fun loadServerFingerprint(address: String): String? {
return prefs.getString("$TOFU_PREFIX$address", null)
}
// --- Ping RTT cache ---
fun savePingRtt(address: String, rttMs: Int) {
prefs.edit().putInt("ping_rtt_$address", rttMs).apply()
}
fun loadPingRtt(address: String): Int {
return prefs.getInt("ping_rtt_$address", -1)
}
}

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package com.wzp.debug
import android.content.Context
import android.util.Log
import kotlinx.coroutines.Dispatchers
import kotlinx.coroutines.withContext
import java.io.BufferedOutputStream
import java.io.ByteArrayOutputStream
import java.io.File
import java.io.FileInputStream
import java.io.FileOutputStream
import java.nio.ByteBuffer
import java.nio.ByteOrder
import java.text.SimpleDateFormat
import java.util.Date
import java.util.Locale
import java.util.zip.ZipEntry
import java.util.zip.ZipOutputStream
/**
* Collects call debug data (audio recordings, logs, histograms, stats)
* into a zip file for email sharing.
*/
class DebugReporter(private val context: Context) {
companion object {
private const val TAG = "DebugReporter"
private const val SAMPLE_RATE = 48000
}
/**
* Build a zip with all debug data.
* Returns the zip File on success, or null on failure.
*/
suspend fun collectZip(
callDurationSecs: Double,
finalStatsJson: String,
aecEnabled: Boolean,
alias: String,
server: String,
room: String
): File? = withContext(Dispatchers.IO) {
try {
val debugDir = File(context.cacheDir, "wzp_debug")
val timestamp = SimpleDateFormat("yyyyMMdd_HHmmss", Locale.US).format(Date())
val zipFile = File(context.cacheDir, "wzp_debug_${timestamp}.zip")
ZipOutputStream(BufferedOutputStream(FileOutputStream(zipFile))).use { zos ->
// 1. Call metadata
val meta = buildString {
appendLine("=== WZ Phone Debug Report ===")
appendLine("Timestamp: $timestamp")
appendLine("Alias: $alias")
appendLine("Server: $server")
appendLine("Room: $room")
appendLine("Duration: ${"%.1f".format(callDurationSecs)}s")
appendLine("AEC: ${if (aecEnabled) "ON" else "OFF"}")
appendLine("Device: ${android.os.Build.MANUFACTURER} ${android.os.Build.MODEL}")
appendLine("Android: ${android.os.Build.VERSION.RELEASE} (API ${android.os.Build.VERSION.SDK_INT})")
appendLine()
appendLine("=== Final Stats ===")
appendLine(finalStatsJson)
}
addTextEntry(zos, "meta.txt", meta)
// 2. Logcat — WZP-related tags
val logcat = collectLogcat()
addTextEntry(zos, "logcat.txt", logcat)
// 3. Capture audio (mic) → WAV
val captureRaw = File(debugDir, "capture.pcm")
if (captureRaw.exists() && captureRaw.length() > 0) {
addWavEntry(zos, "capture.wav", captureRaw)
Log.i(TAG, "capture.pcm: ${captureRaw.length()} bytes -> WAV")
}
// 4. Playout audio (speaker) → WAV
val playoutRaw = File(debugDir, "playout.pcm")
if (playoutRaw.exists() && playoutRaw.length() > 0) {
addWavEntry(zos, "playout.wav", playoutRaw)
Log.i(TAG, "playout.pcm: ${playoutRaw.length()} bytes -> WAV")
}
// 5. RMS histogram CSV
val captureHist = File(debugDir, "capture_rms.csv")
if (captureHist.exists()) addFileEntry(zos, "capture_rms.csv", captureHist)
val playoutHist = File(debugDir, "playout_rms.csv")
if (playoutHist.exists()) addFileEntry(zos, "playout_rms.csv", playoutHist)
}
Log.i(TAG, "zip created: ${zipFile.length()} bytes (${zipFile.length() / 1024}KB)")
// Clean up raw debug files (keep zip)
debugDir.listFiles()?.forEach { it.delete() }
zipFile
} catch (e: Exception) {
Log.e(TAG, "debug report failed", e)
null
}
}
/** Clean up any leftover debug files from a previous session. */
fun prepareForCall() {
val debugDir = File(context.cacheDir, "wzp_debug")
if (debugDir.exists()) {
debugDir.listFiles()?.forEach { it.delete() }
}
debugDir.mkdirs()
// Also clean up old zip files
context.cacheDir.listFiles()?.filter { it.name.startsWith("wzp_debug_") }?.forEach { it.delete() }
}
private fun collectLogcat(): String {
return try {
val process = Runtime.getRuntime().exec(
arrayOf(
"logcat", "-d",
"-t", "5000",
"--format", "threadtime"
)
)
val output = process.inputStream.bufferedReader().readText()
process.waitFor()
output.lines()
.filter { line ->
line.contains("wzp", ignoreCase = true) ||
line.contains("WzpEngine") ||
line.contains("AudioPipeline") ||
line.contains("WzpCall") ||
line.contains("CallService") ||
line.contains("AudioTrack") ||
line.contains("AudioRecord") ||
line.contains("AcousticEchoCanceler") ||
line.contains("NoiseSuppressor") ||
line.contains("FATAL") ||
line.contains("ANR") ||
line.contains("AudioFlinger") ||
line.contains("DebugReporter") ||
line.contains("QUIC") ||
line.contains("quinn") ||
line.contains("send task") ||
line.contains("recv task") ||
line.contains("send stats") ||
line.contains("recv stats") ||
line.contains("send_media") ||
line.contains("FEC block") ||
line.contains("recv gap") ||
line.contains("frames_dropped") ||
line.contains("opus")
}
.joinToString("\n")
} catch (e: Exception) {
"Failed to collect logcat: ${e.message}"
}
}
private fun addWavEntry(zos: ZipOutputStream, name: String, pcmFile: File) {
val dataSize = pcmFile.length().toInt()
val byteRate = SAMPLE_RATE * 1 * 16 / 8
val blockAlign = 1 * 16 / 8
zos.putNextEntry(ZipEntry(name))
// Write WAV header (44 bytes)
val header = ByteBuffer.allocate(44).order(ByteOrder.LITTLE_ENDIAN)
header.put("RIFF".toByteArray())
header.putInt(36 + dataSize)
header.put("WAVE".toByteArray())
header.put("fmt ".toByteArray())
header.putInt(16)
header.putShort(1) // PCM
header.putShort(1) // mono
header.putInt(SAMPLE_RATE)
header.putInt(byteRate)
header.putShort(blockAlign.toShort())
header.putShort(16) // bits per sample
header.put("data".toByteArray())
header.putInt(dataSize)
zos.write(header.array())
// Stream PCM data directly (avoids loading entire file into memory)
FileInputStream(pcmFile).use { it.copyTo(zos) }
zos.closeEntry()
}
private fun addTextEntry(zos: ZipOutputStream, name: String, content: String) {
zos.putNextEntry(ZipEntry(name))
zos.write(content.toByteArray())
zos.closeEntry()
}
private fun addFileEntry(zos: ZipOutputStream, name: String, file: File) {
zos.putNextEntry(ZipEntry(name))
FileInputStream(file).use { it.copyTo(zos) }
zos.closeEntry()
}
}

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package com.wzp.engine
import org.json.JSONArray
import org.json.JSONObject
/**
* Snapshot of call statistics, mirroring the Rust `CallStats` struct.
*
* Constructed from the JSON string returned by [WzpEngine.getStats].
*/
data class CallStats(
/** Current call state ordinal (see [CallStateConstants]). */
val state: Int = 0,
/** Call duration in seconds. */
val durationSecs: Double = 0.0,
/** Quality tier: 0 = Good, 1 = Degraded, 2 = Catastrophic. */
val qualityTier: Int = 0,
/** Observed packet loss percentage (0..100). */
val lossPct: Float = 0f,
/** Smoothed round-trip time in milliseconds. */
val rttMs: Int = 0,
/** Jitter in milliseconds. */
val jitterMs: Int = 0,
/** Current jitter buffer depth in packets. */
val jitterBufferDepth: Int = 0,
/** Total frames encoded since call start. */
val framesEncoded: Long = 0,
/** Total frames decoded since call start. */
val framesDecoded: Long = 0,
/** Number of playout underruns (buffer empty when audio was needed). */
val underruns: Long = 0,
/** Frames recovered by FEC. */
val fecRecovered: Long = 0,
/** Current mic audio level (RMS, 0-32767). */
val audioLevel: Int = 0,
/** Our current outgoing codec (e.g. "Opus24k"). */
val currentCodec: String = "",
/** Last seen incoming codec from peers. */
val peerCodec: String = "",
/** Whether auto quality mode is active. */
val autoMode: Boolean = false,
/** Number of participants in the room. */
val roomParticipantCount: Int = 0,
/** Participants in the room (fingerprint + optional alias). */
val roomParticipants: List<RoomMember> = emptyList(),
/** SAS verification code (4-digit, null if not in a call). */
val sasCode: Int? = null,
/** Incoming call ID (or "relay|room" for CallSetup). */
val incomingCallId: String? = null,
/** Incoming caller's fingerprint. */
val incomingCallerFp: String? = null,
/** Incoming caller's alias. */
val incomingCallerAlias: String? = null,
) {
/** Human-readable quality label. */
val qualityLabel: String
get() = when (qualityTier) {
0 -> "Good"
1 -> "Degraded"
2 -> "Catastrophic"
else -> "Unknown"
}
companion object {
private fun parseParticipants(arr: JSONArray?): List<RoomMember> {
if (arr == null) return emptyList()
return (0 until arr.length()).map { i ->
val o = arr.getJSONObject(i)
RoomMember(
fingerprint = o.optString("fingerprint", ""),
alias = if (o.isNull("alias")) null else o.optString("alias", null),
relayLabel = if (o.isNull("relay_label")) null else o.optString("relay_label", null)
)
}
}
/** Deserialise from the JSON string produced by the native engine. */
fun fromJson(json: String): CallStats {
return try {
val obj = JSONObject(json)
CallStats(
state = obj.optInt("state", 0),
durationSecs = obj.optDouble("duration_secs", 0.0),
qualityTier = obj.optInt("quality_tier", 0),
lossPct = obj.optDouble("loss_pct", 0.0).toFloat(),
rttMs = obj.optInt("rtt_ms", 0),
jitterMs = obj.optInt("jitter_ms", 0),
jitterBufferDepth = obj.optInt("jitter_buffer_depth", 0),
framesEncoded = obj.optLong("frames_encoded", 0),
framesDecoded = obj.optLong("frames_decoded", 0),
underruns = obj.optLong("underruns", 0),
fecRecovered = obj.optLong("fec_recovered", 0),
audioLevel = obj.optInt("audio_level", 0),
currentCodec = obj.optString("current_codec", ""),
peerCodec = obj.optString("peer_codec", ""),
autoMode = obj.optBoolean("auto_mode", false),
roomParticipantCount = obj.optInt("room_participant_count", 0),
roomParticipants = parseParticipants(obj.optJSONArray("room_participants")),
sasCode = if (obj.has("sas_code")) obj.optInt("sas_code") else null,
incomingCallId = if (obj.isNull("incoming_call_id")) null else obj.optString("incoming_call_id", null),
incomingCallerFp = if (obj.isNull("incoming_caller_fp")) null else obj.optString("incoming_caller_fp", null),
incomingCallerAlias = if (obj.isNull("incoming_caller_alias")) null else obj.optString("incoming_caller_alias", null),
)
} catch (e: Exception) {
CallStats()
}
}
}
}
data class RoomMember(
val fingerprint: String,
val alias: String? = null,
val relayLabel: String? = null
) {
/** Short display name: alias if set, otherwise first 8 chars of fingerprint. */
val displayName: String
get() = alias?.takeIf { it.isNotBlank() }
?: fingerprint.take(8).ifEmpty { "unknown" }
}

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package com.wzp.engine
/**
* Callback interface for VoIP engine events.
*
* All callbacks are invoked on the main/UI thread.
*/
interface WzpCallback {
/**
* Called when the call state changes.
*
* @param state one of [CallStateConstants]: IDLE(0), CONNECTING(1), ACTIVE(2),
* RECONNECTING(3), CLOSED(4)
*/
fun onCallStateChanged(state: Int)
/**
* Called when the network quality tier changes.
*
* @param tier 0 = Good, 1 = Degraded, 2 = Catastrophic
*/
fun onQualityTierChanged(tier: Int)
/**
* Called when an error occurs in the native engine.
*
* @param code numeric error code (negative)
* @param message human-readable description
*/
fun onError(code: Int, message: String)
}

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package com.wzp.engine
/**
* Native VoIP engine wrapper. Delegates all work to libwzp_android.so via JNI.
*
* Lifecycle:
* 1. Construct with a [WzpCallback]
* 2. Call [init] to create the native engine
* 3. Call [startCall] to begin a VoIP session
* 4. Use [setMute], [setSpeaker], [getStats], [forceProfile] during the call
* 5. Call [stopCall] to end the session
* 6. Call [destroy] when the engine is no longer needed
*
* Thread safety: all methods must be called from the same thread (typically main).
*/
class WzpEngine(private val callback: WzpCallback) {
/** Opaque pointer to the native EngineHandle. 0 means not initialised. */
private var nativeHandle: Long = 0L
/** Whether the engine has been initialised. */
val isInitialized: Boolean get() = nativeHandle != 0L
/** Create the native engine. Must be called before any other method. */
fun init() {
check(nativeHandle == 0L) { "Engine already initialized" }
nativeHandle = nativeInit()
check(nativeHandle != 0L) { "Native engine creation failed" }
}
/**
* Start a call.
*
* @param relayAddr relay server address (host:port)
* @param room room identifier (used as QUIC SNI)
* @param seedHex 64-char hex-encoded 32-byte identity seed (empty = random)
* @param token authentication token (empty = no auth)
* @param alias display name sent to relay for room participant list
* @return 0 on success, negative error code on failure
*/
/**
* @param profile 0 = Opus GOOD, 1 = Opus DEGRADED, 2 = Codec2 CATASTROPHIC
*/
fun startCall(relayAddr: String, room: String, seedHex: String = "", token: String = "", alias: String = "", profile: Int = 0): Int {
check(nativeHandle != 0L) { "Engine not initialized" }
val result = nativeStartCall(nativeHandle, relayAddr, room, seedHex, token, alias, profile)
if (result == 0) {
callback.onCallStateChanged(CallStateConstants.CONNECTING)
} else {
callback.onError(result, "Failed to start call")
}
return result
}
/** Stop the active call. Safe to call when no call is active. */
@Synchronized
fun stopCall() {
if (nativeHandle != 0L) {
nativeStopCall(nativeHandle)
callback.onCallStateChanged(CallStateConstants.CLOSED)
}
}
/** Mute or unmute the microphone. */
fun setMute(muted: Boolean) {
if (nativeHandle != 0L) nativeSetMute(nativeHandle, muted)
}
/** Enable or disable loudspeaker mode. */
fun setSpeaker(speaker: Boolean) {
if (nativeHandle != 0L) nativeSetSpeaker(nativeHandle, speaker)
}
/**
* Get current call statistics as a JSON string.
*
* @return JSON-serialised [CallStats], or `"{}"` if the engine is not initialised.
*/
@Synchronized
fun getStats(): String {
if (nativeHandle == 0L) return "{}"
return try {
nativeGetStats(nativeHandle) ?: "{}"
} catch (_: Exception) {
"{}"
}
}
/**
* Force a quality profile, overriding adaptive selection.
*
* @param profile 0 = GOOD, 1 = DEGRADED, 2 = CATASTROPHIC
*/
fun forceProfile(profile: Int) {
if (nativeHandle != 0L) nativeForceProfile(nativeHandle, profile)
}
/** Destroy the native engine and free all resources. The instance must not be reused. */
@Synchronized
fun destroy() {
if (nativeHandle != 0L) {
nativeDestroy(nativeHandle)
nativeHandle = 0L
}
}
/**
* Write captured PCM samples into the engine's capture ring buffer.
* Called from the AudioRecord capture thread.
*/
fun writeAudio(pcm: ShortArray): Int {
if (nativeHandle == 0L) return 0
return nativeWriteAudio(nativeHandle, pcm)
}
/**
* Read decoded PCM samples from the engine's playout ring buffer.
* Called from the AudioTrack playout thread.
*/
fun readAudio(pcm: ShortArray): Int {
if (nativeHandle == 0L) return 0
return nativeReadAudio(nativeHandle, pcm)
}
/**
* Write captured PCM from a DirectByteBuffer — zero JNI array copy.
* The buffer must be a direct ByteBuffer with native byte order containing i16 samples.
* Called from the AudioRecord capture thread.
*/
fun writeAudioDirect(buffer: java.nio.ByteBuffer, sampleCount: Int): Int {
if (nativeHandle == 0L) return 0
return nativeWriteAudioDirect(nativeHandle, buffer, sampleCount)
}
/**
* Read decoded PCM into a DirectByteBuffer — zero JNI array copy.
* The buffer must be a direct ByteBuffer with native byte order.
* Called from the AudioTrack playout thread.
*/
fun readAudioDirect(buffer: java.nio.ByteBuffer, maxSamples: Int): Int {
if (nativeHandle == 0L) return 0
return nativeReadAudioDirect(nativeHandle, buffer, maxSamples)
}
// -- JNI native methods --------------------------------------------------
private external fun nativeInit(): Long
private external fun nativeStartCall(
handle: Long, relay: String, room: String, seed: String, token: String, alias: String, profile: Int
): Int
private external fun nativeStopCall(handle: Long)
private external fun nativeSetMute(handle: Long, muted: Boolean)
private external fun nativeSetSpeaker(handle: Long, speaker: Boolean)
private external fun nativeGetStats(handle: Long): String?
private external fun nativeForceProfile(handle: Long, profile: Int)
private external fun nativeWriteAudio(handle: Long, pcm: ShortArray): Int
private external fun nativeReadAudio(handle: Long, pcm: ShortArray): Int
private external fun nativeWriteAudioDirect(handle: Long, buffer: java.nio.ByteBuffer, sampleCount: Int): Int
private external fun nativeReadAudioDirect(handle: Long, buffer: java.nio.ByteBuffer, maxSamples: Int): Int
private external fun nativeDestroy(handle: Long)
private external fun nativePingRelay(handle: Long, relay: String): String?
private external fun nativeStartSignaling(handle: Long, relay: String, seed: String, token: String, alias: String): Int
private external fun nativePlaceCall(handle: Long, targetFp: String): Int
private external fun nativeAnswerCall(handle: Long, callId: String, mode: Int): Int
/**
* Ping a relay server. Requires engine to be initialized.
* Returns JSON `{"rtt_ms":N,"server_fingerprint":"hex"}` or null.
*/
fun pingRelay(address: String): String? {
if (nativeHandle == 0L) return null
return nativePingRelay(nativeHandle, address)
}
/**
* Start persistent signaling connection for direct 1:1 calls.
* The engine registers on the relay and listens for incoming calls.
* Call state updates are available via [getStats].
*
* @return 0 on success, -1 on error
*/
fun startSignaling(relay: String, seed: String = "", token: String = "", alias: String = ""): Int {
check(nativeHandle != 0L) { "Engine not initialized" }
return nativeStartSignaling(nativeHandle, relay, seed, token, alias)
}
/**
* Place a direct call to a peer by fingerprint.
* Requires [startSignaling] to have been called first.
*
* @return 0 on success, -1 on error
*/
fun placeCall(targetFingerprint: String): Int {
check(nativeHandle != 0L) { "Engine not initialized" }
return nativePlaceCall(nativeHandle, targetFingerprint)
}
/**
* Answer an incoming direct call.
*
* @param callId The call ID from the incoming call (available in stats.incoming_call_id)
* @param mode 0=Reject, 1=AcceptTrusted (P2P in Phase 2), 2=AcceptGeneric (relay-mediated)
* @return 0 on success, -1 on error
*/
fun answerCall(callId: String, mode: Int = 2): Int {
check(nativeHandle != 0L) { "Engine not initialized" }
return nativeAnswerCall(nativeHandle, callId, mode)
}
companion object {
init {
System.loadLibrary("wzp_android")
}
}
}
/** Integer constants matching the Rust [CallState] enum ordinals. */
object CallStateConstants {
const val IDLE = 0
const val CONNECTING = 1
const val ACTIVE = 2
const val RECONNECTING = 3
const val CLOSED = 4
}

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package com.wzp.net
// Relay pinging is now done via WzpEngine.pingRelay() (instance method).
// This file kept for the data class only.
object RelayPinger {
data class PingResult(
val rttMs: Int,
val reachable: Boolean,
val serverFingerprint: String = "",
)
}

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package com.wzp.service
import android.app.Notification
import android.app.PendingIntent
import android.app.Service
import android.content.Context
import android.content.Intent
import android.media.AudioManager
import android.net.wifi.WifiManager
import android.os.IBinder
import android.os.PowerManager
import androidx.core.app.NotificationCompat
import com.wzp.WzpApplication
import com.wzp.ui.call.CallActivity
/**
* Foreground service that keeps the VoIP call alive when the app is backgrounded.
*
* Responsibilities:
* - Shows a persistent notification during the call
* - Acquires a partial wake lock so the CPU stays on
* - Acquires a Wi-Fi lock to prevent Wi-Fi from going to sleep
* - Sets [AudioManager] mode to [AudioManager.MODE_IN_COMMUNICATION]
* - Releases all resources when the call ends
*/
class CallService : Service() {
private var wakeLock: PowerManager.WakeLock? = null
private var wifiLock: WifiManager.WifiLock? = null
private var previousAudioMode: Int = AudioManager.MODE_NORMAL
// -- Lifecycle ------------------------------------------------------------
override fun onCreate() {
super.onCreate()
acquireWakeLock()
acquireWifiLock()
setAudioMode()
}
override fun onStartCommand(intent: Intent?, flags: Int, startId: Int): Int {
when (intent?.action) {
ACTION_STOP -> {
onStopFromNotification?.invoke()
stopSelf()
return START_NOT_STICKY
}
}
startForeground(NOTIFICATION_ID, buildNotification())
return START_STICKY
}
override fun onDestroy() {
restoreAudioMode()
releaseWifiLock()
releaseWakeLock()
super.onDestroy()
}
override fun onBind(intent: Intent?): IBinder? = null
// -- Notification ---------------------------------------------------------
private fun buildNotification(): Notification {
// Tapping the notification returns to the call screen
val contentIntent = PendingIntent.getActivity(
this,
0,
Intent(this, CallActivity::class.java).apply {
flags = Intent.FLAG_ACTIVITY_SINGLE_TOP
},
PendingIntent.FLAG_IMMUTABLE or PendingIntent.FLAG_UPDATE_CURRENT
)
// "End call" action button
val stopIntent = PendingIntent.getService(
this,
1,
Intent(this, CallService::class.java).apply { action = ACTION_STOP },
PendingIntent.FLAG_IMMUTABLE or PendingIntent.FLAG_UPDATE_CURRENT
)
return NotificationCompat.Builder(this, WzpApplication.CHANNEL_ID)
.setContentTitle("WZ Phone")
.setContentText("Call in progress")
.setSmallIcon(android.R.drawable.ic_menu_call)
.setOngoing(true)
.setContentIntent(contentIntent)
.addAction(android.R.drawable.ic_menu_close_clear_cancel, "End Call", stopIntent)
.setCategory(NotificationCompat.CATEGORY_CALL)
.setPriority(NotificationCompat.PRIORITY_LOW)
.build()
}
// -- Wake lock ------------------------------------------------------------
private fun acquireWakeLock() {
val pm = getSystemService(Context.POWER_SERVICE) as PowerManager
wakeLock = pm.newWakeLock(
PowerManager.PARTIAL_WAKE_LOCK,
"wzp:call_wake_lock"
).apply {
acquire(MAX_CALL_DURATION_MS)
}
}
private fun releaseWakeLock() {
wakeLock?.let {
if (it.isHeld) it.release()
}
wakeLock = null
}
// -- Wi-Fi lock -----------------------------------------------------------
@Suppress("DEPRECATION")
private fun acquireWifiLock() {
val wm = applicationContext.getSystemService(Context.WIFI_SERVICE) as WifiManager
wifiLock = wm.createWifiLock(
WifiManager.WIFI_MODE_FULL_HIGH_PERF,
"wzp:call_wifi_lock"
).apply {
acquire()
}
}
private fun releaseWifiLock() {
wifiLock?.let {
if (it.isHeld) it.release()
}
wifiLock = null
}
// -- Audio mode -----------------------------------------------------------
private fun setAudioMode() {
val am = getSystemService(Context.AUDIO_SERVICE) as AudioManager
previousAudioMode = am.mode
am.mode = AudioManager.MODE_IN_COMMUNICATION
}
private fun restoreAudioMode() {
val am = getSystemService(Context.AUDIO_SERVICE) as AudioManager
am.mode = previousAudioMode
}
// -- Static helpers -------------------------------------------------------
companion object {
private const val NOTIFICATION_ID = 1001
private const val ACTION_STOP = "com.wzp.service.STOP"
private const val MAX_CALL_DURATION_MS = 4L * 60 * 60 * 1000 // 4 hours
/** Called when the user taps "End Call" in the notification. */
var onStopFromNotification: (() -> Unit)? = null
/** Start the foreground call service. */
fun start(context: Context) {
val intent = Intent(context, CallService::class.java)
context.startForegroundService(intent)
}
/** Stop the foreground call service. */
fun stop(context: Context) {
val intent = Intent(context, CallService::class.java).apply {
action = ACTION_STOP
}
context.startService(intent)
}
}
}

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package com.wzp.ui.call
import android.Manifest
import android.content.Intent
import android.content.pm.PackageManager
import android.os.Bundle
import android.util.Log
import android.widget.Toast
import androidx.activity.ComponentActivity
import androidx.activity.compose.setContent
import androidx.activity.result.contract.ActivityResultContracts
import androidx.activity.viewModels
import androidx.compose.material3.MaterialTheme
import androidx.compose.material3.darkColorScheme
import androidx.compose.material3.dynamicDarkColorScheme
import androidx.compose.material3.dynamicLightColorScheme
import androidx.compose.material3.lightColorScheme
import androidx.compose.foundation.isSystemInDarkTheme
import androidx.compose.runtime.Composable
import androidx.compose.runtime.getValue
import androidx.compose.runtime.mutableStateOf
import androidx.compose.runtime.remember
import androidx.compose.runtime.setValue
import androidx.compose.ui.platform.LocalContext
import androidx.core.content.ContextCompat
import androidx.core.content.FileProvider
import androidx.lifecycle.Lifecycle
import androidx.lifecycle.lifecycleScope
import androidx.lifecycle.repeatOnLifecycle
import com.wzp.ui.settings.SettingsScreen
import kotlinx.coroutines.launch
/**
* Main activity hosting the in-call Compose UI.
*
* Call lifecycle (wake lock, Wi-Fi lock, audio mode, notification)
* is managed by [com.wzp.service.CallService] foreground service.
*/
class CallActivity : ComponentActivity() {
companion object {
private const val TAG = "CallActivity"
}
private val viewModel: CallViewModel by viewModels()
private val audioPermissionLauncher = registerForActivityResult(
ActivityResultContracts.RequestPermission()
) { granted ->
if (!granted) {
Toast.makeText(this, "Microphone permission is required for calls", Toast.LENGTH_LONG).show()
}
}
override fun onCreate(savedInstanceState: Bundle?) {
super.onCreate(savedInstanceState)
viewModel.setContext(this)
setContent {
WzpTheme {
var showSettings by remember { mutableStateOf(false) }
if (showSettings) {
SettingsScreen(
viewModel = viewModel,
onBack = { showSettings = false }
)
} else {
InCallScreen(
viewModel = viewModel,
onHangUp = { viewModel.stopCall() },
onOpenSettings = { showSettings = true }
)
}
}
}
if (ContextCompat.checkSelfPermission(this, Manifest.permission.RECORD_AUDIO)
!= PackageManager.PERMISSION_GRANTED
) {
audioPermissionLauncher.launch(Manifest.permission.RECORD_AUDIO)
}
// Watch for debug zip ready → launch email intent
lifecycleScope.launch {
repeatOnLifecycle(Lifecycle.State.STARTED) {
viewModel.debugZipReady.collect { zipFile ->
if (zipFile != null && zipFile.exists()) {
Log.i(TAG, "debug zip ready: ${zipFile.absolutePath} (${zipFile.length()} bytes)")
launchEmailIntent(zipFile)
viewModel.onDebugReportSent()
}
}
}
}
}
private fun launchEmailIntent(zipFile: java.io.File) {
try {
val authority = "${applicationContext.packageName}.fileprovider"
Log.i(TAG, "FileProvider authority: $authority, file: ${zipFile.absolutePath}")
val uri = FileProvider.getUriForFile(this, authority, zipFile)
Log.i(TAG, "FileProvider URI: $uri")
val intent = Intent(Intent.ACTION_SEND).apply {
type = "message/rfc822"
putExtra(Intent.EXTRA_EMAIL, arrayOf("manwefarm@gmail.com"))
putExtra(Intent.EXTRA_SUBJECT, "WZ Phone Debug Report - ${zipFile.name}")
putExtra(
Intent.EXTRA_TEXT,
"Debug report attached.\n\nContains: call recordings (WAV), RMS histograms (CSV), logcat, stats."
)
putExtra(Intent.EXTRA_STREAM, uri)
addFlags(Intent.FLAG_GRANT_READ_URI_PERMISSION)
}
startActivity(Intent.createChooser(intent, "Send debug report"))
Log.i(TAG, "email intent launched")
} catch (e: Exception) {
Log.e(TAG, "email intent failed", e)
Toast.makeText(this, "Failed to launch email: ${e.message}", Toast.LENGTH_LONG).show()
}
}
override fun onDestroy() {
super.onDestroy()
if (isFinishing) {
viewModel.stopCall()
}
}
}
@Composable
fun WzpTheme(content: @Composable () -> Unit) {
val darkTheme = isSystemInDarkTheme()
val context = LocalContext.current
val colorScheme = when {
android.os.Build.VERSION.SDK_INT >= android.os.Build.VERSION_CODES.S -> {
if (darkTheme) dynamicDarkColorScheme(context) else dynamicLightColorScheme(context)
}
darkTheme -> darkColorScheme()
else -> lightColorScheme()
}
MaterialTheme(
colorScheme = colorScheme,
content = content
)
}

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package com.wzp.ui.call
import android.content.Context
import android.util.Log
import androidx.lifecycle.ViewModel
import androidx.lifecycle.viewModelScope
import com.wzp.audio.AudioPipeline
import com.wzp.audio.AudioRouteManager
import com.wzp.data.SettingsRepository
import com.wzp.debug.DebugReporter
import com.wzp.engine.CallStats
import com.wzp.service.CallService
import com.wzp.engine.WzpCallback
import com.wzp.engine.WzpEngine
import kotlinx.coroutines.Dispatchers
import kotlinx.coroutines.Job
import kotlinx.coroutines.delay
import kotlinx.coroutines.flow.MutableStateFlow
import kotlinx.coroutines.flow.StateFlow
import kotlinx.coroutines.flow.asStateFlow
import kotlinx.coroutines.isActive
import kotlinx.coroutines.launch
import kotlinx.coroutines.withContext
import org.json.JSONObject
import java.io.File
import java.net.Inet4Address
import java.net.Inet6Address
import java.net.InetAddress
data class ServerEntry(val address: String, val label: String)
data class PingResult(
val rttMs: Int,
val serverFingerprint: String = "",
val reachable: Boolean = rttMs > 0,
)
enum class LockStatus { UNKNOWN, OFFLINE, NEW, VERIFIED, CHANGED }
class CallViewModel : ViewModel(), WzpCallback {
private var engine: WzpEngine? = null
private var engineInitialized = false
private var audioPipeline: AudioPipeline? = null
private var audioRouteManager: AudioRouteManager? = null
private var audioStarted = false
private var appContext: Context? = null
private var settings: SettingsRepository? = null
private var debugReporter: DebugReporter? = null
private var lastStatsJson: String = "{}"
private var lastCallDuration: Double = 0.0
private var lastCallServer: String = ""
private val _callState = MutableStateFlow(0)
val callState: StateFlow<Int> get() = _callState.asStateFlow()
private val _isMuted = MutableStateFlow(false)
val isMuted: StateFlow<Boolean> = _isMuted.asStateFlow()
private val _isSpeaker = MutableStateFlow(false)
val isSpeaker: StateFlow<Boolean> = _isSpeaker.asStateFlow()
private val _stats = MutableStateFlow(CallStats())
val stats: StateFlow<CallStats> = _stats.asStateFlow()
private val _qualityTier = MutableStateFlow(0)
val qualityTier: StateFlow<Int> = _qualityTier.asStateFlow()
private val _errorMessage = MutableStateFlow<String?>(null)
val errorMessage: StateFlow<String?> = _errorMessage.asStateFlow()
private val _roomName = MutableStateFlow(DEFAULT_ROOM)
val roomName: StateFlow<String> = _roomName.asStateFlow()
private val _selectedServer = MutableStateFlow(0)
val selectedServer: StateFlow<Int> = _selectedServer.asStateFlow()
private val _servers = MutableStateFlow(DEFAULT_SERVERS.toList())
val servers: StateFlow<List<ServerEntry>> = _servers.asStateFlow()
private val _preferIPv6 = MutableStateFlow(false)
val preferIPv6: StateFlow<Boolean> = _preferIPv6.asStateFlow()
private val _recentRooms = MutableStateFlow<List<com.wzp.data.SettingsRepository.RecentRoom>>(emptyList())
val recentRooms: StateFlow<List<com.wzp.data.SettingsRepository.RecentRoom>> = _recentRooms.asStateFlow()
/** Ping results keyed by server address. */
private val _pingResults = MutableStateFlow<Map<String, PingResult>>(emptyMap())
val pingResults: StateFlow<Map<String, PingResult>> = _pingResults.asStateFlow()
/** Known server fingerprints (TOFU). */
private val _knownFingerprints = MutableStateFlow<Map<String, String>>(emptyMap())
private val _playoutGainDb = MutableStateFlow(0f)
val playoutGainDb: StateFlow<Float> = _playoutGainDb.asStateFlow()
private val _captureGainDb = MutableStateFlow(0f)
val captureGainDb: StateFlow<Float> = _captureGainDb.asStateFlow()
private val _alias = MutableStateFlow("")
val alias: StateFlow<String> = _alias.asStateFlow()
private val _seedHex = MutableStateFlow("")
val seedHex: StateFlow<String> = _seedHex.asStateFlow()
private val _aecEnabled = MutableStateFlow(true)
val aecEnabled: StateFlow<Boolean> = _aecEnabled.asStateFlow()
private val _debugRecording = MutableStateFlow(false)
val debugRecording: StateFlow<Boolean> = _debugRecording.asStateFlow()
// Quality profile index (matches JNI bridge profile_from_int)
private val _codecChoice = MutableStateFlow(0)
val codecChoice: StateFlow<Int> = _codecChoice.asStateFlow()
/** Key-change warning dialog state. */
data class KeyWarningInfo(val address: String, val oldFp: String, val newFp: String)
private val _keyWarning = MutableStateFlow<KeyWarningInfo?>(null)
val keyWarning: StateFlow<KeyWarningInfo?> = _keyWarning.asStateFlow()
/** True when a call just ended and debug report can be sent. */
private val _debugReportAvailable = MutableStateFlow(false)
val debugReportAvailable: StateFlow<Boolean> = _debugReportAvailable.asStateFlow()
/** Status: null=idle, "Preparing..."=in progress, "ready"=zip ready, "Error:..."=failed */
private val _debugReportStatus = MutableStateFlow<String?>(null)
val debugReportStatus: StateFlow<String?> = _debugReportStatus.asStateFlow()
/** The zip file ready to be emailed. Set by sendDebugReport, consumed by Activity. */
private val _debugZipReady = MutableStateFlow<File?>(null)
val debugZipReady: StateFlow<File?> = _debugZipReady.asStateFlow()
private var statsJob: Job? = null
// ── Direct calling state ──
/** 0=room mode, 1=direct call mode */
private val _callMode = MutableStateFlow(0)
val callMode: StateFlow<Int> = _callMode.asStateFlow()
/** Target fingerprint for direct call */
private val _targetFingerprint = MutableStateFlow("")
val targetFingerprint: StateFlow<String> = _targetFingerprint.asStateFlow()
/** Signal connection state: 0=idle, 5=registered, 6=ringing, 7=incoming */
private val _signalState = MutableStateFlow(0)
val signalState: StateFlow<Int> = _signalState.asStateFlow()
/** Incoming call info */
private val _incomingCallId = MutableStateFlow<String?>(null)
val incomingCallId: StateFlow<String?> = _incomingCallId.asStateFlow()
private val _incomingCallerFp = MutableStateFlow<String?>(null)
val incomingCallerFp: StateFlow<String?> = _incomingCallerFp.asStateFlow()
private val _incomingCallerAlias = MutableStateFlow<String?>(null)
val incomingCallerAlias: StateFlow<String?> = _incomingCallerAlias.asStateFlow()
fun setCallMode(mode: Int) { _callMode.value = mode }
fun setTargetFingerprint(fp: String) { _targetFingerprint.value = fp }
/** Register on relay for direct calls */
fun registerForCalls() {
if (engine == null) {
engine = WzpEngine(this).also { it.init() }
}
val serverIdx = _selectedServer.value
val serverList = _servers.value
if (serverIdx >= serverList.size) return
val relay = serverList[serverIdx].address
val seed = _seedHex.value
val alias = _alias.value
viewModelScope.launch(Dispatchers.IO) {
val resolvedRelay = resolveToIp(relay) ?: relay
val result = engine?.startSignaling(resolvedRelay, seed, "", alias)
if (result == 0) {
_signalState.value = 5 // Registered
startStatsPolling()
} else {
_errorMessage.value = "Failed to register on relay"
}
}
}
/** Place a direct call to the target fingerprint */
fun placeDirectCall() {
val target = _targetFingerprint.value.trim()
if (target.isEmpty()) {
_errorMessage.value = "Enter a fingerprint to call"
return
}
engine?.placeCall(target)
_signalState.value = 6 // Ringing
}
/** Answer an incoming direct call */
fun answerIncomingCall(mode: Int = 2) {
val callId = _incomingCallId.value ?: return
engine?.answerCall(callId, mode)
}
/** Reject an incoming direct call */
fun rejectIncomingCall() {
val callId = _incomingCallId.value ?: return
engine?.answerCall(callId, 0) // 0 = Reject
_signalState.value = 5 // Back to registered
_incomingCallId.value = null
_incomingCallerFp.value = null
_incomingCallerAlias.value = null
}
companion object {
private const val TAG = "WzpCall"
val DEFAULT_SERVERS = listOf(
ServerEntry("172.16.81.175:4433", "LAN (172.16.81.175)"),
ServerEntry("193.180.213.68:4433", "Pangolin (IP)"),
)
const val DEFAULT_ROOM = "general"
}
fun setContext(context: Context) {
val appCtx = context.applicationContext
appContext = appCtx
if (audioPipeline == null) {
audioPipeline = AudioPipeline(appCtx)
}
if (audioRouteManager == null) {
audioRouteManager = AudioRouteManager(appCtx)
}
if (debugReporter == null) {
debugReporter = DebugReporter(appCtx)
}
if (settings == null) {
settings = SettingsRepository(appCtx)
loadSettings()
}
}
private fun loadSettings() {
val s = settings ?: return
s.loadServers()?.let { saved ->
if (saved.isNotEmpty()) _servers.value = saved
}
_selectedServer.value = s.loadSelectedServer().coerceIn(0, _servers.value.lastIndex)
_roomName.value = s.loadRoom()
_alias.value = s.getOrCreateAlias()
_preferIPv6.value = s.loadPreferIPv6()
_playoutGainDb.value = s.loadPlayoutGain()
_captureGainDb.value = s.loadCaptureGain()
_seedHex.value = s.getOrCreateSeedHex()
_aecEnabled.value = s.loadAecEnabled()
_debugRecording.value = s.loadDebugRecording()
_codecChoice.value = s.loadCodecChoice()
_recentRooms.value = s.loadRecentRooms()
}
fun selectServer(index: Int) {
if (index in _servers.value.indices) {
_selectedServer.value = index
settings?.saveSelectedServer(index)
}
}
fun setPreferIPv6(prefer: Boolean) {
_preferIPv6.value = prefer
settings?.savePreferIPv6(prefer)
}
fun addServer(hostPort: String, label: String) {
val current = _servers.value.toMutableList()
current.add(ServerEntry(hostPort, label))
_servers.value = current
settings?.saveServers(current)
}
fun removeServer(index: Int) {
if (index < DEFAULT_SERVERS.size) return // don't remove built-in servers
val current = _servers.value.toMutableList()
if (index in current.indices) {
current.removeAt(index)
_servers.value = current
if (_selectedServer.value >= current.size) {
_selectedServer.value = 0
}
settings?.saveServers(current)
settings?.saveSelectedServer(_selectedServer.value)
}
}
/** Batch-apply servers and selection from Settings draft state. */
fun applyServers(servers: List<ServerEntry>, selected: Int) {
_servers.value = servers
_selectedServer.value = selected.coerceIn(0, servers.lastIndex)
settings?.saveServers(servers)
settings?.saveSelectedServer(_selectedServer.value)
}
/**
* Ping all servers via native QUIC. Requires engine to be initialized.
* Creates engine if needed, pings, keeps engine alive for subsequent Connect.
*/
fun pingAllServers() {
viewModelScope.launch {
// Ensure engine exists
if (engine == null || engine?.isInitialized != true) {
try {
engine = WzpEngine(this@CallViewModel).also { it.init() }
engineInitialized = true
} catch (e: Exception) {
Log.w(TAG, "engine init for ping failed: $e")
return@launch
}
}
val eng = engine ?: return@launch
val results = mutableMapOf<String, PingResult>()
val known = mutableMapOf<String, String>()
_servers.value.forEach { server ->
val json = withContext(Dispatchers.IO) {
eng.pingRelay(server.address)
}
if (json != null) {
try {
val obj = JSONObject(json)
val rtt = obj.getInt("rtt_ms")
val fp = obj.optString("server_fingerprint", "")
results[server.address] = PingResult(rttMs = rtt, serverFingerprint = fp)
// TOFU
if (fp.isNotEmpty()) {
val saved = settings?.loadServerFingerprint(server.address)
if (saved == null) settings?.saveServerFingerprint(server.address, fp)
known[server.address] = saved ?: fp
}
} catch (_: Exception) {}
}
}
_pingResults.value = results
_knownFingerprints.value = known
}
}
/** Load saved TOFU fingerprints. */
fun loadSavedFingerprints() {
val known = mutableMapOf<String, String>()
_servers.value.forEach { server ->
settings?.loadServerFingerprint(server.address)?.let {
known[server.address] = it
}
}
_knownFingerprints.value = known
}
/** Get lock status for a server. */
fun lockStatus(address: String): LockStatus {
val pr = _pingResults.value[address] ?: return LockStatus.UNKNOWN
if (!pr.reachable) return LockStatus.OFFLINE
val known = _knownFingerprints.value[address] ?: return LockStatus.NEW
if (pr.serverFingerprint.isEmpty()) return LockStatus.NEW
return if (pr.serverFingerprint == known) LockStatus.VERIFIED else LockStatus.CHANGED
}
fun setRoomName(name: String) {
_roomName.value = name
settings?.saveRoom(name)
}
fun setPlayoutGainDb(db: Float) {
_playoutGainDb.value = db
audioPipeline?.playoutGainDb = db
settings?.savePlayoutGain(db)
}
fun setCaptureGainDb(db: Float) {
_captureGainDb.value = db
audioPipeline?.captureGainDb = db
settings?.saveCaptureGain(db)
}
fun setAlias(alias: String) {
_alias.value = alias
settings?.saveAlias(alias)
}
fun restoreSeed(hex: String) {
_seedHex.value = hex
settings?.saveSeedHex(hex)
}
fun setAecEnabled(enabled: Boolean) {
_aecEnabled.value = enabled
settings?.saveAecEnabled(enabled)
}
fun setDebugRecording(enabled: Boolean) {
_debugRecording.value = enabled
settings?.saveDebugRecording(enabled)
}
fun setCodecChoice(choice: Int) {
_codecChoice.value = choice
settings?.saveCodecChoice(choice)
}
/**
* Resolve DNS hostname to IP address on the Kotlin/Android side,
* since Rust's DNS resolution may not work on Android.
* Returns "ip:port" string.
*/
private fun resolveToIp(hostPort: String): String {
val parts = hostPort.split(":")
if (parts.size != 2) return hostPort
val host = parts[0]
val port = parts[1]
// Already an IP address — return as-is
if (host.matches(Regex("""\d+\.\d+\.\d+\.\d+"""))) return hostPort
if (host.contains(":")) return hostPort // IPv6 literal
return try {
val addresses = InetAddress.getAllByName(host)
val preferV6 = _preferIPv6.value
val picked = if (preferV6) {
addresses.firstOrNull { it is Inet6Address } ?: addresses.firstOrNull { it is Inet4Address }
} else {
addresses.firstOrNull { it is Inet4Address } ?: addresses.firstOrNull { it is Inet6Address }
}
if (picked != null) {
val ip = picked.hostAddress ?: host
val formatted = if (picked is Inet6Address) "[$ip]:$port" else "$ip:$port"
formatted
} else {
hostPort
}
} catch (_: Exception) {
hostPort // resolution failed — pass through and let Rust try
}
}
/** Tear down engine and audio. Pass stopService=true to also stop the foreground service. */
private fun teardown(stopService: Boolean = true) {
Log.i(TAG, "teardown: stopping audio, stopService=$stopService")
val hadCall = audioStarted
CallService.onStopFromNotification = null
stopAudio() // sets running=false (non-blocking)
stopStatsPolling()
// Wait for audio threads to exit their loops before destroying the engine.
// This guarantees no in-flight JNI calls to writeAudio/readAudio.
val drained = audioPipeline?.awaitDrain() ?: true
if (!drained) {
Log.w(TAG, "teardown: audio threads did not drain in time")
}
audioPipeline = null
Log.i(TAG, "teardown: stopping engine")
try { engine?.stopCall() } catch (e: Exception) { Log.w(TAG, "stopCall err: $e") }
try { engine?.destroy() } catch (e: Exception) { Log.w(TAG, "destroy err: $e") }
engine = null
engineInitialized = false
_callState.value = 0
if (hadCall) {
_debugReportAvailable.value = true
}
if (stopService) {
try { appContext?.let { CallService.stop(it) } } catch (_: Exception) {}
}
Log.i(TAG, "teardown: done")
}
/** Accept the new server key and proceed with the call. */
fun acceptNewFingerprint() {
val info = _keyWarning.value ?: return
_knownFingerprints.value = _knownFingerprints.value.toMutableMap().also {
it[info.address] = info.newFp
}
settings?.saveServerFingerprint(info.address, info.newFp)
_keyWarning.value = null
startCallInternal()
}
fun dismissKeyWarning() {
_keyWarning.value = null
}
fun startCall() {
val serverEntry = _servers.value[_selectedServer.value]
// Check for key change before connecting
val ls = lockStatus(serverEntry.address)
if (ls == LockStatus.CHANGED) {
val known = _knownFingerprints.value[serverEntry.address] ?: ""
val current = _pingResults.value[serverEntry.address]?.serverFingerprint ?: ""
_keyWarning.value = KeyWarningInfo(serverEntry.address, known, current)
return
}
startCallInternal()
}
/** Start a call to a specific relay + room (used by direct call setup). */
private fun startCallInternal(relay: String, room: String) {
Log.i(TAG, "startCallDirect: relay=$relay room=$room")
try {
// Don't teardown — keep the signal connection alive
engine = WzpEngine(this)
engine!!.init()
engineInitialized = true
_callState.value = 1
_errorMessage.value = null
try { appContext?.let { CallService.start(it) } } catch (e: Exception) {
Log.w(TAG, "service start err: $e")
}
startStatsPolling()
viewModelScope.launch(kotlinx.coroutines.Dispatchers.IO) {
try {
val seed = _seedHex.value
val name = _alias.value
val result = engine?.startCall(relay, room, seedHex = seed, alias = name, profile = _codecChoice.value) ?: -1
CallService.onStopFromNotification = { stopCall() }
if (result != 0) {
_callState.value = 0
_errorMessage.value = "Failed to connect to call room (code $result)"
appContext?.let { CallService.stop(it) }
}
} catch (e: Exception) {
Log.e(TAG, "startCallDirect error", e)
_callState.value = 0
_errorMessage.value = "Engine error: ${e.message}"
appContext?.let { CallService.stop(it) }
}
}
} catch (e: Exception) {
Log.e(TAG, "startCallDirect error", e)
_callState.value = 0
_errorMessage.value = "Engine error: ${e.message}"
}
}
private fun startCallInternal() {
val serverEntry = _servers.value[_selectedServer.value]
val room = _roomName.value
Log.i(TAG, "startCall: server=${serverEntry.address} room=$room")
_debugReportAvailable.value = false
_debugReportStatus.value = null
lastCallServer = serverEntry.address
settings?.addRecentRoom(serverEntry.address, room)
_recentRooms.value = settings?.loadRecentRooms() ?: emptyList()
debugReporter?.prepareForCall()
try {
// Teardown previous call but don't stop the service (we're about to restart it)
teardown(stopService = false)
Log.i(TAG, "startCall: creating engine")
engine = WzpEngine(this)
engine!!.init()
engineInitialized = true
_callState.value = 1
_errorMessage.value = null
try { appContext?.let { CallService.start(it) } } catch (e: Exception) {
Log.w(TAG, "service start err: $e")
}
startStatsPolling()
viewModelScope.launch(kotlinx.coroutines.Dispatchers.IO) {
try {
val relay = resolveToIp(serverEntry.address)
val seed = _seedHex.value
val name = _alias.value
Log.i(TAG, "startCall: resolved=$relay, alias=$name, calling engine.startCall")
val result = engine?.startCall(relay, room, seedHex = seed, alias = name, profile = _codecChoice.value) ?: -1
Log.i(TAG, "startCall: engine returned $result")
// Only wire up notification callback after engine is running
CallService.onStopFromNotification = { stopCall() }
if (result != 0) {
_callState.value = 0
_errorMessage.value = "Failed to start call (code $result)"
appContext?.let { CallService.stop(it) }
}
} catch (e: Exception) {
Log.e(TAG, "startCall IO error", e)
_callState.value = 0
_errorMessage.value = "Engine error: ${e.message}"
appContext?.let { CallService.stop(it) }
}
}
} catch (e: Exception) {
Log.e(TAG, "startCall error", e)
_callState.value = 0
_errorMessage.value = "Engine error: ${e.message}"
appContext?.let { CallService.stop(it) }
}
}
fun stopCall() {
Log.i(TAG, "stopCall")
teardown()
}
fun toggleMute() {
val newMuted = !_isMuted.value
_isMuted.value = newMuted
try { engine?.setMute(newMuted) } catch (_: Exception) {}
}
fun toggleSpeaker() {
val newSpeaker = !_isSpeaker.value
_isSpeaker.value = newSpeaker
audioRouteManager?.setSpeaker(newSpeaker)
}
fun clearError() { _errorMessage.value = null }
fun sendDebugReport() {
val reporter = debugReporter ?: return
_debugReportStatus.value = "Preparing debug report..."
viewModelScope.launch(kotlinx.coroutines.Dispatchers.IO) {
val zipFile = reporter.collectZip(
callDurationSecs = lastCallDuration,
finalStatsJson = lastStatsJson,
aecEnabled = _aecEnabled.value,
alias = _alias.value,
server = lastCallServer,
room = _roomName.value
)
if (zipFile != null) {
_debugZipReady.value = zipFile
_debugReportStatus.value = "ready"
} else {
_debugReportStatus.value = "Error: failed to create zip"
}
_debugReportAvailable.value = false
}
}
/** Called by Activity after email intent is launched. */
fun onDebugReportSent() {
_debugZipReady.value = null
_debugReportStatus.value = null
}
fun dismissDebugReport() {
_debugReportAvailable.value = false
_debugReportStatus.value = null
_debugZipReady.value = null
}
// WzpCallback
override fun onCallStateChanged(state: Int) { _callState.value = state }
override fun onQualityTierChanged(tier: Int) { _qualityTier.value = tier }
override fun onError(code: Int, message: String) { _errorMessage.value = "Error $code: $message" }
private fun startAudio() {
if (audioStarted) return
val e = engine ?: return
val ctx = appContext ?: return
// Create a fresh pipeline each call to avoid stale threads
audioPipeline = AudioPipeline(ctx).also {
it.playoutGainDb = _playoutGainDb.value
it.captureGainDb = _captureGainDb.value
it.aecEnabled = _aecEnabled.value
it.debugRecording = _debugRecording.value
it.start(e)
}
audioRouteManager?.register()
audioStarted = true
}
private fun stopAudio() {
if (!audioStarted) return
audioPipeline?.stop() // sets running=false; DON'T null — teardown needs awaitDrain()
audioRouteManager?.unregister()
audioRouteManager?.setSpeaker(false)
_isSpeaker.value = false
audioStarted = false
}
private fun startStatsPolling() {
statsJob?.cancel()
statsJob = viewModelScope.launch {
while (isActive) {
try {
val json = engine?.getStats() ?: "{}"
if (json.isNotEmpty()) {
Log.d(TAG, "raw: $json")
lastStatsJson = json
val s = CallStats.fromJson(json)
lastCallDuration = s.durationSecs
_stats.value = s
if (s.state != 0) {
_callState.value = s.state
}
// Track signal state changes for direct calling
if (s.state in 5..7) {
_signalState.value = s.state
}
// Incoming call detection
if (s.state == 7) { // IncomingCall
_incomingCallId.value = s.incomingCallId
_incomingCallerFp.value = s.incomingCallerFp
_incomingCallerAlias.value = s.incomingCallerAlias
}
// CallSetup: auto-connect to media room
if (s.state == 1 && s.incomingCallId != null && s.incomingCallId.contains("|")) {
// Format: "relay_addr|room_name"
val parts = s.incomingCallId.split("|", limit = 2)
if (parts.size == 2) {
val mediaRelay = parts[0]
val mediaRoom = parts[1]
Log.i(TAG, "CallSetup: connecting to $mediaRelay room $mediaRoom")
startCallInternal(mediaRelay, mediaRoom)
}
}
if (s.state == 2 && !audioStarted) {
startAudio()
}
}
} catch (_: Exception) {}
delay(500L)
}
}
}
private fun stopStatsPolling() {
statsJob?.cancel()
statsJob = null
}
override fun onCleared() {
super.onCleared()
Log.i(TAG, "onCleared")
teardown()
}
}

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package com.wzp.ui.components
import android.widget.Toast
import androidx.compose.foundation.Canvas
import androidx.compose.foundation.clickable
import androidx.compose.foundation.layout.size
import androidx.compose.foundation.shape.RoundedCornerShape
import androidx.compose.runtime.Composable
import androidx.compose.ui.Modifier
import androidx.compose.ui.draw.clip
import androidx.compose.ui.geometry.Offset
import androidx.compose.ui.geometry.Size
import androidx.compose.ui.graphics.Color
import androidx.compose.ui.platform.LocalClipboardManager
import androidx.compose.ui.platform.LocalContext
import androidx.compose.ui.text.AnnotatedString
import androidx.compose.ui.unit.Dp
import androidx.compose.ui.unit.dp
import kotlin.math.min
/**
* Deterministic identicon — generates a unique 5x5 symmetric pattern
* from a hex fingerprint string. Identical algorithm to the desktop
* TypeScript implementation in identicon.ts.
*/
@Composable
fun Identicon(
fingerprint: String,
size: Dp = 36.dp,
clickToCopy: Boolean = true,
modifier: Modifier = Modifier,
) {
val clipboard = LocalClipboardManager.current
val context = LocalContext.current
val bytes = hashBytes(fingerprint)
val (bg, fg) = deriveColors(bytes)
val grid = buildGrid(bytes)
Canvas(
modifier = modifier
.size(size)
.clip(RoundedCornerShape(size * 0.12f))
.then(
if (clickToCopy && fingerprint.isNotEmpty()) {
Modifier.clickable {
clipboard.setText(AnnotatedString(fingerprint))
Toast.makeText(context, "Copied", Toast.LENGTH_SHORT).show()
}
} else Modifier
)
) {
val cellW = this.size.width / 5f
val cellH = this.size.height / 5f
// Background
drawRect(color = bg, size = this.size)
// Foreground cells
for (y in 0 until 5) {
for (x in 0 until 5) {
if (grid[y][x]) {
drawRect(
color = fg,
topLeft = Offset(x * cellW, y * cellH),
size = Size(cellW, cellH),
)
}
}
}
}
}
/**
* Fingerprint text that copies to clipboard on tap.
*/
@Composable
fun CopyableFingerprint(
fingerprint: String,
modifier: Modifier = Modifier,
style: androidx.compose.ui.text.TextStyle = androidx.compose.material3.MaterialTheme.typography.bodySmall,
color: Color = Color.Unspecified,
) {
val clipboard = LocalClipboardManager.current
val context = LocalContext.current
androidx.compose.material3.Text(
text = fingerprint,
style = style,
color = color,
modifier = modifier.clickable {
if (fingerprint.isNotEmpty()) {
clipboard.setText(AnnotatedString(fingerprint))
Toast.makeText(context, "Fingerprint copied", Toast.LENGTH_SHORT).show()
}
}
)
}
// --- Internal helpers (matching desktop identicon.ts) ---
private fun hashBytes(hex: String): List<Int> {
val clean = hex.filter { it.isLetterOrDigit() }
val bytes = mutableListOf<Int>()
var i = 0
while (i + 1 < clean.length) {
val b = clean.substring(i, i + 2).toIntOrNull(16) ?: 0
bytes.add(b)
i += 2
}
// Pad to at least 16 bytes
while (bytes.size < 16) bytes.add(0)
return bytes
}
private fun deriveColors(bytes: List<Int>): Pair<Color, Color> {
val hue1 = bytes[0] * 360f / 256f
val hue2 = (bytes[1] * 360f / 256f + 120f) % 360f
val bg = hslToColor(hue1, 0.65f, 0.35f)
val fg = hslToColor(hue2, 0.70f, 0.55f)
return bg to fg
}
private fun buildGrid(bytes: List<Int>): List<List<Boolean>> {
return (0 until 5).map { y ->
val left = (0 until 3).map { x ->
val idx = 2 + y * 3 + x
bytes[idx % bytes.size] > 128
}
// Mirror: col3 = col1, col4 = col0
listOf(left[0], left[1], left[2], left[1], left[0])
}
}
private fun hslToColor(h: Float, s: Float, l: Float): Color {
val k = { n: Float -> (n + h / 30f) % 12f }
val a = s * min(l, 1f - l)
val f = { n: Float ->
l - a * maxOf(-1f, minOf(k(n) - 3f, minOf(9f - k(n), 1f)))
}
return Color(f(0f), f(8f), f(4f))
}

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package com.wzp.ui.settings
import androidx.compose.foundation.clickable
import android.content.ClipData
import android.content.ClipboardManager
import android.content.Context
import android.widget.Toast
import androidx.compose.foundation.layout.Arrangement
import androidx.compose.foundation.layout.Column
import androidx.compose.foundation.layout.ExperimentalLayoutApi
import androidx.compose.foundation.layout.FlowRow
import androidx.compose.foundation.layout.Row
import androidx.compose.foundation.layout.Spacer
import androidx.compose.foundation.layout.fillMaxSize
import androidx.compose.foundation.layout.fillMaxWidth
import androidx.compose.foundation.layout.height
import androidx.compose.foundation.layout.padding
import androidx.compose.foundation.layout.width
import androidx.compose.foundation.rememberScrollState
import androidx.compose.foundation.shape.RoundedCornerShape
import androidx.compose.foundation.verticalScroll
import androidx.compose.material3.AlertDialog
import androidx.compose.material3.Button
import androidx.compose.material3.ButtonDefaults
import androidx.compose.material3.Divider
import androidx.compose.material3.RadioButton
import androidx.compose.material3.FilledTonalButton
import androidx.compose.material3.FilledTonalIconButton
import androidx.compose.material3.IconButtonDefaults
import androidx.compose.material3.MaterialTheme
import androidx.compose.material3.OutlinedButton
import androidx.compose.material3.OutlinedTextField
import androidx.compose.material3.Slider
import androidx.compose.material3.Surface
import androidx.compose.material3.Switch
import androidx.compose.material3.Text
import androidx.compose.material3.TextButton
import androidx.compose.runtime.Composable
import androidx.compose.runtime.collectAsState
import androidx.compose.runtime.getValue
import androidx.compose.runtime.mutableFloatStateOf
import androidx.compose.runtime.mutableIntStateOf
import androidx.compose.runtime.mutableStateOf
import androidx.compose.runtime.remember
import androidx.compose.runtime.setValue
import androidx.compose.runtime.toMutableStateList
import androidx.compose.ui.Alignment
import androidx.compose.ui.Modifier
import androidx.compose.ui.graphics.Color
import androidx.compose.ui.platform.LocalContext
import androidx.compose.ui.text.font.FontFamily
import androidx.compose.ui.text.font.FontWeight
import androidx.compose.ui.unit.dp
import com.wzp.ui.call.CallViewModel
import com.wzp.ui.call.ServerEntry
@OptIn(ExperimentalLayoutApi::class)
@Composable
fun SettingsScreen(
viewModel: CallViewModel,
onBack: () -> Unit
) {
val context = LocalContext.current
// Snapshot current values into local draft state
val currentAlias by viewModel.alias.collectAsState()
val currentSeedHex by viewModel.seedHex.collectAsState()
val currentServers by viewModel.servers.collectAsState()
val currentSelectedServer by viewModel.selectedServer.collectAsState()
val currentRoomName by viewModel.roomName.collectAsState()
val currentPreferIPv6 by viewModel.preferIPv6.collectAsState()
val currentPlayoutGain by viewModel.playoutGainDb.collectAsState()
val currentCaptureGain by viewModel.captureGainDb.collectAsState()
val currentAecEnabled by viewModel.aecEnabled.collectAsState()
// Draft state — initialized from current values
var draftAlias by remember { mutableStateOf(currentAlias) }
var draftSeedHex by remember { mutableStateOf(currentSeedHex) }
val draftServers = remember { currentServers.toMutableStateList() }
var draftSelectedServer by remember { mutableIntStateOf(currentSelectedServer) }
var draftRoomName by remember { mutableStateOf(currentRoomName) }
var draftPreferIPv6 by remember { mutableStateOf(currentPreferIPv6) }
var draftPlayoutGain by remember { mutableFloatStateOf(currentPlayoutGain) }
var draftCaptureGain by remember { mutableFloatStateOf(currentCaptureGain) }
var draftAecEnabled by remember { mutableStateOf(currentAecEnabled) }
// Track if anything changed
val hasChanges = draftAlias != currentAlias ||
draftSeedHex != currentSeedHex ||
draftServers.toList() != currentServers ||
draftSelectedServer != currentSelectedServer ||
draftRoomName != currentRoomName ||
draftPreferIPv6 != currentPreferIPv6 ||
draftPlayoutGain != currentPlayoutGain ||
draftCaptureGain != currentCaptureGain ||
draftAecEnabled != currentAecEnabled
var showAddServerDialog by remember { mutableStateOf(false) }
var showRestoreKeyDialog by remember { mutableStateOf(false) }
Surface(
modifier = Modifier.fillMaxSize(),
color = MaterialTheme.colorScheme.background
) {
Column(
modifier = Modifier
.fillMaxSize()
.padding(24.dp)
.verticalScroll(rememberScrollState())
) {
// Header
Row(
modifier = Modifier.fillMaxWidth(),
verticalAlignment = Alignment.CenterVertically
) {
TextButton(onClick = onBack) {
Text("< Back")
}
Spacer(modifier = Modifier.weight(1f))
Text(
text = "Settings",
style = MaterialTheme.typography.headlineSmall.copy(
fontWeight = FontWeight.Bold
),
color = MaterialTheme.colorScheme.primary
)
Spacer(modifier = Modifier.weight(1f))
// Save button — only enabled when changes exist
Button(
onClick = {
viewModel.setAlias(draftAlias)
if (draftSeedHex != currentSeedHex) viewModel.restoreSeed(draftSeedHex)
viewModel.applyServers(draftServers.toList(), draftSelectedServer)
viewModel.setRoomName(draftRoomName)
viewModel.setPreferIPv6(draftPreferIPv6)
viewModel.setPlayoutGainDb(draftPlayoutGain)
viewModel.setCaptureGainDb(draftCaptureGain)
viewModel.setAecEnabled(draftAecEnabled)
Toast.makeText(context, "Settings saved", Toast.LENGTH_SHORT).show()
onBack()
},
enabled = hasChanges
) {
Text("Save")
}
}
Spacer(modifier = Modifier.height(24.dp))
// --- Identity ---
SectionHeader("Identity")
OutlinedTextField(
value = draftAlias,
onValueChange = { draftAlias = it },
label = { Text("Display Name") },
singleLine = true,
modifier = Modifier.fillMaxWidth()
)
Spacer(modifier = Modifier.height(16.dp))
// Fingerprint display with identicon
val fingerprint = if (draftSeedHex.length >= 16) draftSeedHex.take(16).uppercase() else "Not generated"
Text(
text = "Fingerprint",
style = MaterialTheme.typography.labelSmall,
color = MaterialTheme.colorScheme.onSurfaceVariant
)
Row(
verticalAlignment = Alignment.CenterVertically,
modifier = Modifier.padding(vertical = 4.dp)
) {
com.wzp.ui.components.Identicon(
fingerprint = draftSeedHex,
size = 40.dp,
)
Spacer(modifier = Modifier.width(12.dp))
com.wzp.ui.components.CopyableFingerprint(
fingerprint = fingerprint.chunked(4).joinToString(" "),
style = MaterialTheme.typography.bodyMedium.copy(
fontFamily = FontFamily.Monospace
),
color = MaterialTheme.colorScheme.onSurface,
)
}
Spacer(modifier = Modifier.height(12.dp))
// Key backup/restore
Row(horizontalArrangement = Arrangement.spacedBy(8.dp)) {
FilledTonalButton(onClick = {
val clipboard = context.getSystemService(Context.CLIPBOARD_SERVICE) as ClipboardManager
clipboard.setPrimaryClip(ClipData.newPlainText("WZP Key", draftSeedHex))
Toast.makeText(context, "Key copied to clipboard", Toast.LENGTH_SHORT).show()
}) {
Text("Copy Key")
}
OutlinedButton(onClick = { showRestoreKeyDialog = true }) {
Text("Restore Key")
}
}
Spacer(modifier = Modifier.height(24.dp))
Divider()
Spacer(modifier = Modifier.height(16.dp))
// --- Audio ---
SectionHeader("Audio Defaults")
GainSlider(
label = "Voice Volume",
gainDb = draftPlayoutGain,
onGainChange = { draftPlayoutGain = Math.round(it).toFloat() }
)
Spacer(modifier = Modifier.height(4.dp))
GainSlider(
label = "Mic Gain",
gainDb = draftCaptureGain,
onGainChange = { draftCaptureGain = Math.round(it).toFloat() }
)
Spacer(modifier = Modifier.height(12.dp))
Row(
verticalAlignment = Alignment.CenterVertically,
modifier = Modifier.fillMaxWidth()
) {
Column(modifier = Modifier.weight(1f)) {
Text(
text = "Echo Cancellation (AEC)",
style = MaterialTheme.typography.bodyMedium
)
Text(
text = "Disable if audio sounds distorted",
style = MaterialTheme.typography.bodySmall,
color = MaterialTheme.colorScheme.onSurfaceVariant
)
}
Switch(
checked = draftAecEnabled,
onCheckedChange = { draftAecEnabled = it }
)
}
Spacer(modifier = Modifier.height(12.dp))
// Quality selection — slider from best (studio 64k) to worst (codec2 1.2k) + auto
val qualityLabels = listOf(
"Studio 64k", "Studio 48k", "Studio 32k", "Auto",
"Opus 24k", "Opus 6k", "Codec2 3.2k", "Codec2 1.2k"
)
// Map slider position to JNI profile int:
// 0=Studio64k(6), 1=Studio48k(5), 2=Studio32k(4), 3=Auto(7),
// 4=Opus24k(0), 5=Opus6k(1), 6=Codec2_3.2k(3), 7=Codec2_1.2k(2)
val sliderToProfile = intArrayOf(6, 5, 4, 7, 0, 1, 3, 2)
val profileToSlider = mapOf(6 to 0, 5 to 1, 4 to 2, 7 to 3, 0 to 4, 1 to 5, 3 to 6, 2 to 7)
val qualityColors = listOf(
Color(0xFF22C55E), Color(0xFF4ADE80), Color(0xFF86EFAC), Color(0xFFA3E635),
Color(0xFFA3E635), Color(0xFFFACC15), Color(0xFFE97320), Color(0xFF991B1B)
)
val currentCodec by viewModel.codecChoice.collectAsState()
val sliderPos = profileToSlider[currentCodec] ?: 3
Text("Quality", style = MaterialTheme.typography.bodyMedium)
Text(
text = "Decode always accepts all codecs",
style = MaterialTheme.typography.bodySmall,
color = MaterialTheme.colorScheme.onSurfaceVariant
)
Spacer(modifier = Modifier.height(4.dp))
Text(
text = qualityLabels[sliderPos],
style = MaterialTheme.typography.titleMedium.copy(fontWeight = FontWeight.Bold),
color = qualityColors[sliderPos]
)
Slider(
value = sliderPos.toFloat(),
onValueChange = { viewModel.setCodecChoice(sliderToProfile[it.toInt()]) },
valueRange = 0f..7f,
steps = 6,
modifier = Modifier.fillMaxWidth()
)
Row(
modifier = Modifier.fillMaxWidth(),
horizontalArrangement = Arrangement.SpaceBetween
) {
Text("Best", style = MaterialTheme.typography.labelSmall, color = Color(0xFF22C55E))
Text("Lowest", style = MaterialTheme.typography.labelSmall, color = Color(0xFF991B1B))
}
Spacer(modifier = Modifier.height(24.dp))
Divider()
Spacer(modifier = Modifier.height(16.dp))
// --- Servers ---
SectionHeader("Servers")
FlowRow(
modifier = Modifier.fillMaxWidth(),
horizontalArrangement = Arrangement.Start,
verticalArrangement = Arrangement.spacedBy(4.dp)
) {
draftServers.forEachIndexed { idx, entry ->
val isSelected = draftSelectedServer == idx
Row(verticalAlignment = Alignment.CenterVertically) {
FilledTonalIconButton(
onClick = { draftSelectedServer = idx },
modifier = Modifier
.padding(end = 2.dp)
.height(36.dp)
.width(140.dp),
shape = RoundedCornerShape(8.dp),
colors = if (isSelected) {
IconButtonDefaults.filledTonalIconButtonColors(
containerColor = MaterialTheme.colorScheme.primaryContainer,
contentColor = MaterialTheme.colorScheme.onPrimaryContainer
)
} else {
IconButtonDefaults.filledTonalIconButtonColors()
}
) {
Text(
text = entry.label,
style = MaterialTheme.typography.labelSmall,
maxLines = 1
)
}
// Show remove button for non-default servers
if (idx >= 2) {
TextButton(
onClick = {
draftServers.removeAt(idx)
if (draftSelectedServer >= draftServers.size) {
draftSelectedServer = 0
}
},
modifier = Modifier.height(36.dp)
) {
Text("X", color = MaterialTheme.colorScheme.error)
}
}
}
}
}
Spacer(modifier = Modifier.height(8.dp))
OutlinedButton(
onClick = { showAddServerDialog = true },
shape = RoundedCornerShape(8.dp)
) {
Text("+ Add Server")
}
// Show selected server address
Spacer(modifier = Modifier.height(8.dp))
Text(
text = "Default: ${draftServers.getOrNull(draftSelectedServer)?.address ?: "none"}",
style = MaterialTheme.typography.bodySmall,
color = MaterialTheme.colorScheme.onSurfaceVariant
)
Spacer(modifier = Modifier.height(24.dp))
Divider()
Spacer(modifier = Modifier.height(16.dp))
// --- Network ---
SectionHeader("Network")
Row(
verticalAlignment = Alignment.CenterVertically,
modifier = Modifier.fillMaxWidth()
) {
Text(
text = "Prefer IPv6",
style = MaterialTheme.typography.bodyMedium,
modifier = Modifier.weight(1f)
)
Switch(
checked = draftPreferIPv6,
onCheckedChange = { draftPreferIPv6 = it }
)
}
Spacer(modifier = Modifier.height(24.dp))
Divider()
Spacer(modifier = Modifier.height(16.dp))
// --- Room ---
SectionHeader("Room")
OutlinedTextField(
value = draftRoomName,
onValueChange = { draftRoomName = it },
label = { Text("Default Room") },
singleLine = true,
modifier = Modifier.fillMaxWidth()
)
Spacer(modifier = Modifier.height(32.dp))
}
}
if (showAddServerDialog) {
AddServerDialog(
onDismiss = { showAddServerDialog = false },
onAdd = { host, port, label ->
draftServers.add(ServerEntry("$host:$port", label))
showAddServerDialog = false
}
)
}
if (showRestoreKeyDialog) {
RestoreKeyDialog(
onDismiss = { showRestoreKeyDialog = false },
onRestore = { hex ->
draftSeedHex = hex
showRestoreKeyDialog = false
Toast.makeText(context, "Key staged — press Save to apply", Toast.LENGTH_SHORT).show()
}
)
}
}
@Composable
private fun SectionHeader(title: String) {
Text(
text = title,
style = MaterialTheme.typography.titleMedium.copy(fontWeight = FontWeight.Bold),
color = MaterialTheme.colorScheme.primary
)
Spacer(modifier = Modifier.height(8.dp))
}
@Composable
private fun GainSlider(label: String, gainDb: Float, onGainChange: (Float) -> Unit) {
Column(
modifier = Modifier.fillMaxWidth(),
horizontalAlignment = Alignment.CenterHorizontally
) {
val sign = if (gainDb >= 0) "+" else ""
Text(
text = "$label: ${sign}${"%.0f".format(gainDb)} dB",
style = MaterialTheme.typography.labelSmall,
color = MaterialTheme.colorScheme.onSurfaceVariant
)
Slider(
value = gainDb,
onValueChange = onGainChange,
valueRange = -20f..20f,
steps = 0,
modifier = Modifier.fillMaxWidth()
)
}
}
@Composable
private fun AddServerDialog(
onDismiss: () -> Unit,
onAdd: (host: String, port: String, label: String) -> Unit
) {
var host by remember { mutableStateOf("") }
var port by remember { mutableStateOf("4433") }
var label by remember { mutableStateOf("") }
AlertDialog(
onDismissRequest = onDismiss,
title = { Text("Add Server") },
text = {
Column {
OutlinedTextField(
value = host,
onValueChange = { host = it },
label = { Text("Host (IP or domain)") },
singleLine = true,
modifier = Modifier.fillMaxWidth()
)
Spacer(modifier = Modifier.height(8.dp))
OutlinedTextField(
value = port,
onValueChange = { port = it },
label = { Text("Port") },
singleLine = true,
modifier = Modifier.fillMaxWidth()
)
Spacer(modifier = Modifier.height(8.dp))
OutlinedTextField(
value = label,
onValueChange = { label = it },
label = { Text("Label (optional)") },
singleLine = true,
modifier = Modifier.fillMaxWidth()
)
}
},
confirmButton = {
TextButton(
onClick = {
if (host.isNotBlank()) {
val displayLabel = label.ifBlank { host }
onAdd(host.trim(), port.trim(), displayLabel)
}
}
) { Text("Add") }
},
dismissButton = {
TextButton(onClick = onDismiss) { Text("Cancel") }
}
)
}
@Composable
private fun RestoreKeyDialog(
onDismiss: () -> Unit,
onRestore: (hex: String) -> Unit
) {
var keyInput by remember { mutableStateOf("") }
var error by remember { mutableStateOf<String?>(null) }
AlertDialog(
onDismissRequest = onDismiss,
title = { Text("Restore Identity Key") },
text = {
Column {
Text(
text = "Paste your 64-character hex key below. This will replace your current identity.",
style = MaterialTheme.typography.bodySmall,
color = MaterialTheme.colorScheme.onSurfaceVariant
)
Spacer(modifier = Modifier.height(8.dp))
OutlinedTextField(
value = keyInput,
onValueChange = {
keyInput = it.trim().lowercase()
error = null
},
label = { Text("Identity Key (hex)") },
singleLine = true,
modifier = Modifier.fillMaxWidth(),
isError = error != null
)
error?.let {
Text(
text = it,
style = MaterialTheme.typography.bodySmall,
color = MaterialTheme.colorScheme.error
)
}
}
},
confirmButton = {
TextButton(
onClick = {
val cleaned = keyInput.replace("\\s".toRegex(), "")
if (cleaned.length != 64 || !cleaned.all { it in '0'..'9' || it in 'a'..'f' }) {
error = "Key must be exactly 64 hex characters"
} else {
onRestore(cleaned)
}
}
) { Text("Restore") }
},
dismissButton = {
TextButton(onClick = onDismiss) { Text("Cancel") }
}
)
}

View File

@@ -0,0 +1,4 @@
<?xml version="1.0" encoding="utf-8"?>
<paths>
<cache-path name="debug" path="." />
</paths>

4
android/build.gradle.kts Normal file
View File

@@ -0,0 +1,4 @@
plugins {
id("com.android.application") version "8.2.0" apply false
id("org.jetbrains.kotlin.android") version "1.9.22" apply false
}

View File

@@ -0,0 +1,4 @@
org.gradle.jvmargs=-Xmx2048m -Dfile.encoding=UTF-8
android.useAndroidX=true
kotlin.code.style=official
android.nonTransitiveRClass=true

Binary file not shown.

View File

@@ -0,0 +1,6 @@
distributionBase=GRADLE_USER_HOME
distributionPath=wrapper/dists
distributionUrl=https\://services.gradle.org/distributions/gradle-8.5-bin.zip
networkTimeout=10000
zipStoreBase=GRADLE_USER_HOME
zipStorePath=wrapper/dists

5
android/gradlew vendored Executable file
View File

@@ -0,0 +1,5 @@
#!/bin/sh
# Gradle wrapper script
APP_HOME=$(cd "$(dirname "$0")" && pwd)
CLASSPATH="$APP_HOME/gradle/wrapper/gradle-wrapper.jar"
exec java -classpath "$CLASSPATH" org.gradle.wrapper.GradleWrapperMain "$@"

View File

@@ -0,0 +1,18 @@
pluginManagement {
repositories {
google()
mavenCentral()
gradlePluginPortal()
}
}
dependencyResolutionManagement {
repositoriesMode.set(RepositoriesMode.FAIL_ON_PROJECT_REPOS)
repositories {
google()
mavenCentral()
}
}
rootProject.name = "WZPhone"
include(":app")

View File

@@ -0,0 +1,34 @@
[package]
name = "wzp-android"
version.workspace = true
edition.workspace = true
license.workspace = true
rust-version.workspace = true
description = "WarzonePhone Android native VoIP engine — Oboe audio, JNI bridge, call pipeline"
[lib]
crate-type = ["cdylib", "rlib"]
[dependencies]
wzp-proto = { workspace = true }
wzp-codec = { workspace = true }
wzp-fec = { workspace = true }
wzp-crypto = { workspace = true }
wzp-transport = { workspace = true }
tokio = { workspace = true }
tracing = { workspace = true }
tracing-subscriber = { workspace = true, features = ["env-filter"] }
bytes = { workspace = true }
serde = { workspace = true }
serde_json = "1"
thiserror = { workspace = true }
async-trait = { workspace = true }
anyhow = "1"
libc = "0.2"
jni = { version = "0.21", default-features = false }
rand = { workspace = true }
rustls = { version = "0.23", default-features = false, features = ["ring"] }
tracing-android = "0.2"
[build-dependencies]
cc = "1"

154
crates/wzp-android/build.rs Normal file
View File

@@ -0,0 +1,154 @@
use std::path::PathBuf;
fn main() {
let target = std::env::var("TARGET").unwrap_or_default();
if target.contains("android") {
// Override broken static getauxval from compiler-rt that crashes
// in shared libraries. Must be compiled first to take link priority.
cc::Build::new()
.file("cpp/getauxval_fix.c")
.compile("getauxval_fix");
let oboe_dir = fetch_oboe();
match oboe_dir {
Some(oboe_path) => {
println!("cargo:warning=Building with Oboe from {:?}", oboe_path);
let mut build = cc::Build::new();
build
.cpp(true)
.std("c++17")
// Use shared libc++ — avoids pulling in static libc stubs
// that crash in shared libraries (getauxval, pthread_create, etc.)
.cpp_link_stdlib(Some("c++_shared"))
.include("cpp")
.include(oboe_path.join("include"))
.include(oboe_path.join("src"))
.define("WZP_HAS_OBOE", None)
.file("cpp/oboe_bridge.cpp");
// Compile all Oboe source files
let src_dir = oboe_path.join("src");
add_cpp_files_recursive(&mut build, &src_dir);
build.compile("oboe_bridge");
}
None => {
println!("cargo:warning=Oboe not found, building with stub");
cc::Build::new()
.cpp(true)
.std("c++17")
.cpp_link_stdlib(Some("c++_shared"))
.file("cpp/oboe_stub.cpp")
.include("cpp")
.compile("oboe_bridge");
}
}
// Dynamic C++ runtime — libc++_shared.so must be in jniLibs alongside
// libwzp_android.so. We copy it there from the NDK sysroot.
//
// WHY NOT STATIC: libc++_static.a + libc++abi.a transitively pull in
// object files from libc.a (static libc) which contain broken stubs for
// getauxval, __init_tcb, pthread_create, etc. These stubs only work in
// statically-linked executables. In shared libraries loaded by dlopen(),
// they SIGSEGV because the static libc init hasn't run.
// Google's official recommendation: use libc++_shared.so for native libs.
if let Ok(ndk) = std::env::var("ANDROID_NDK_HOME") {
let arch = if target.contains("aarch64") {
"aarch64-linux-android"
} else if target.contains("armv7") {
"arm-linux-androideabi"
} else if target.contains("x86_64") {
"x86_64-linux-android"
} else {
"aarch64-linux-android"
};
let lib_dir = format!(
"{ndk}/toolchains/llvm/prebuilt/linux-x86_64/sysroot/usr/lib/{arch}"
);
println!("cargo:rustc-link-search=native={lib_dir}");
// Copy libc++_shared.so to the jniLibs directory
let shared_so = format!("{lib_dir}/libc++_shared.so");
if std::path::Path::new(&shared_so).exists() {
let jni_abi = if target.contains("aarch64") {
"arm64-v8a"
} else if target.contains("armv7") {
"armeabi-v7a"
} else {
"arm64-v8a"
};
// Try to copy to the Gradle jniLibs directory
let manifest = std::env::var("CARGO_MANIFEST_DIR").unwrap_or_default();
let jni_dir = format!(
"{manifest}/../../android/app/src/main/jniLibs/{jni_abi}"
);
if let Ok(_) = std::fs::create_dir_all(&jni_dir) {
let _ = std::fs::copy(&shared_so, format!("{jni_dir}/libc++_shared.so"));
println!("cargo:warning=Copied libc++_shared.so to {jni_dir}");
}
}
}
// Oboe needs liblog and libOpenSLES from Android
println!("cargo:rustc-link-lib=log");
println!("cargo:rustc-link-lib=OpenSLES");
} else {
// Non-Android: always use stub
cc::Build::new()
.cpp(true)
.std("c++17")
.file("cpp/oboe_stub.cpp")
.include("cpp")
.compile("oboe_bridge");
}
}
/// Recursively add all .cpp files from a directory to a cc::Build.
fn add_cpp_files_recursive(build: &mut cc::Build, dir: &std::path::Path) {
if !dir.is_dir() {
return;
}
for entry in std::fs::read_dir(dir).unwrap() {
let entry = entry.unwrap();
let path = entry.path();
if path.is_dir() {
add_cpp_files_recursive(build, &path);
} else if path.extension().map_or(false, |e| e == "cpp") {
build.file(&path);
}
}
}
/// Try to find or fetch Oboe headers + source.
fn fetch_oboe() -> Option<PathBuf> {
let out_dir = PathBuf::from(std::env::var("OUT_DIR").unwrap());
let oboe_dir = out_dir.join("oboe");
if oboe_dir.join("include").join("oboe").join("Oboe.h").exists() {
return Some(oboe_dir);
}
let status = std::process::Command::new("git")
.args([
"clone",
"--depth=1",
"--branch=1.8.1",
"https://github.com/google/oboe.git",
oboe_dir.to_str().unwrap(),
])
.status();
match status {
Ok(s) if s.success() => {
if oboe_dir.join("include").join("oboe").join("Oboe.h").exists() {
Some(oboe_dir)
} else {
None
}
}
_ => None,
}
}

View File

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// Override the broken static getauxval from compiler-rt/CRT.
// The static version reads from __libc_auxv which is NULL in shared libs
// loaded via dlopen, causing SIGSEGV in init_have_lse_atomics at load time.
// This version calls the real bionic getauxval via dlsym.
#ifdef __ANDROID__
#include <dlfcn.h>
#include <stdint.h>
typedef unsigned long (*getauxval_fn)(unsigned long);
unsigned long getauxval(unsigned long type) {
static getauxval_fn real_getauxval = (getauxval_fn)0;
if (!real_getauxval) {
real_getauxval = (getauxval_fn)dlsym((void*)-1L /* RTLD_DEFAULT */, "getauxval");
if (!real_getauxval) {
return 0;
}
}
return real_getauxval(type);
}
#endif

View File

@@ -0,0 +1,278 @@
// Full Oboe implementation for Android
// This file is compiled only when targeting Android
#include "oboe_bridge.h"
#ifdef __ANDROID__
#include <oboe/Oboe.h>
#include <android/log.h>
#include <cstring>
#include <atomic>
#define LOG_TAG "wzp-oboe"
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
#define LOGW(...) __android_log_print(ANDROID_LOG_WARN, LOG_TAG, __VA_ARGS__)
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
// ---------------------------------------------------------------------------
// Ring buffer helpers (SPSC, lock-free)
// ---------------------------------------------------------------------------
static inline int32_t ring_available_read(const wzp_atomic_int* write_idx,
const wzp_atomic_int* read_idx,
int32_t capacity) {
int32_t w = std::atomic_load_explicit(write_idx, std::memory_order_acquire);
int32_t r = std::atomic_load_explicit(read_idx, std::memory_order_relaxed);
int32_t avail = w - r;
if (avail < 0) avail += capacity;
return avail;
}
static inline int32_t ring_available_write(const wzp_atomic_int* write_idx,
const wzp_atomic_int* read_idx,
int32_t capacity) {
return capacity - 1 - ring_available_read(write_idx, read_idx, capacity);
}
static inline void ring_write(int16_t* buf, int32_t capacity,
wzp_atomic_int* write_idx, const wzp_atomic_int* read_idx,
const int16_t* src, int32_t count) {
int32_t w = std::atomic_load_explicit(write_idx, std::memory_order_relaxed);
for (int32_t i = 0; i < count; i++) {
buf[w] = src[i];
w++;
if (w >= capacity) w = 0;
}
std::atomic_store_explicit(write_idx, w, std::memory_order_release);
}
static inline void ring_read(int16_t* buf, int32_t capacity,
const wzp_atomic_int* write_idx, wzp_atomic_int* read_idx,
int16_t* dst, int32_t count) {
int32_t r = std::atomic_load_explicit(read_idx, std::memory_order_relaxed);
for (int32_t i = 0; i < count; i++) {
dst[i] = buf[r];
r++;
if (r >= capacity) r = 0;
}
std::atomic_store_explicit(read_idx, r, std::memory_order_release);
}
// ---------------------------------------------------------------------------
// Global state
// ---------------------------------------------------------------------------
static std::shared_ptr<oboe::AudioStream> g_capture_stream;
static std::shared_ptr<oboe::AudioStream> g_playout_stream;
static const WzpOboeRings* g_rings = nullptr;
static std::atomic<bool> g_running{false};
static std::atomic<float> g_capture_latency_ms{0.0f};
static std::atomic<float> g_playout_latency_ms{0.0f};
// ---------------------------------------------------------------------------
// Capture callback
// ---------------------------------------------------------------------------
class CaptureCallback : public oboe::AudioStreamDataCallback {
public:
oboe::DataCallbackResult onAudioReady(
oboe::AudioStream* stream,
void* audioData,
int32_t numFrames) override {
if (!g_running.load(std::memory_order_relaxed) || !g_rings) {
return oboe::DataCallbackResult::Stop;
}
const int16_t* src = static_cast<const int16_t*>(audioData);
int32_t avail = ring_available_write(g_rings->capture_write_idx,
g_rings->capture_read_idx,
g_rings->capture_capacity);
int32_t to_write = (numFrames < avail) ? numFrames : avail;
if (to_write > 0) {
ring_write(g_rings->capture_buf, g_rings->capture_capacity,
g_rings->capture_write_idx, g_rings->capture_read_idx,
src, to_write);
}
// Update latency estimate
auto result = stream->calculateLatencyMillis();
if (result) {
g_capture_latency_ms.store(static_cast<float>(result.value()),
std::memory_order_relaxed);
}
return oboe::DataCallbackResult::Continue;
}
};
// ---------------------------------------------------------------------------
// Playout callback
// ---------------------------------------------------------------------------
class PlayoutCallback : public oboe::AudioStreamDataCallback {
public:
oboe::DataCallbackResult onAudioReady(
oboe::AudioStream* stream,
void* audioData,
int32_t numFrames) override {
if (!g_running.load(std::memory_order_relaxed) || !g_rings) {
memset(audioData, 0, numFrames * sizeof(int16_t));
return oboe::DataCallbackResult::Stop;
}
int16_t* dst = static_cast<int16_t*>(audioData);
int32_t avail = ring_available_read(g_rings->playout_write_idx,
g_rings->playout_read_idx,
g_rings->playout_capacity);
int32_t to_read = (numFrames < avail) ? numFrames : avail;
if (to_read > 0) {
ring_read(g_rings->playout_buf, g_rings->playout_capacity,
g_rings->playout_write_idx, g_rings->playout_read_idx,
dst, to_read);
}
// Fill remainder with silence on underrun
if (to_read < numFrames) {
memset(dst + to_read, 0, (numFrames - to_read) * sizeof(int16_t));
}
// Update latency estimate
auto result = stream->calculateLatencyMillis();
if (result) {
g_playout_latency_ms.store(static_cast<float>(result.value()),
std::memory_order_relaxed);
}
return oboe::DataCallbackResult::Continue;
}
};
static CaptureCallback g_capture_cb;
static PlayoutCallback g_playout_cb;
// ---------------------------------------------------------------------------
// Public C API
// ---------------------------------------------------------------------------
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
if (g_running.load(std::memory_order_relaxed)) {
LOGW("wzp_oboe_start: already running");
return -1;
}
g_rings = rings;
// Build capture stream
oboe::AudioStreamBuilder captureBuilder;
captureBuilder.setDirection(oboe::Direction::Input)
->setPerformanceMode(oboe::PerformanceMode::LowLatency)
->setSharingMode(oboe::SharingMode::Exclusive)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setInputPreset(oboe::InputPreset::VoiceCommunication)
->setDataCallback(&g_capture_cb);
oboe::Result result = captureBuilder.openStream(g_capture_stream);
if (result != oboe::Result::OK) {
LOGE("Failed to open capture stream: %s", oboe::convertToText(result));
return -2;
}
// Build playout stream
oboe::AudioStreamBuilder playoutBuilder;
playoutBuilder.setDirection(oboe::Direction::Output)
->setPerformanceMode(oboe::PerformanceMode::LowLatency)
->setSharingMode(oboe::SharingMode::Exclusive)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setUsage(oboe::Usage::VoiceCommunication)
->setDataCallback(&g_playout_cb);
result = playoutBuilder.openStream(g_playout_stream);
if (result != oboe::Result::OK) {
LOGE("Failed to open playout stream: %s", oboe::convertToText(result));
g_capture_stream->close();
g_capture_stream.reset();
return -3;
}
g_running.store(true, std::memory_order_release);
// Start both streams
result = g_capture_stream->requestStart();
if (result != oboe::Result::OK) {
LOGE("Failed to start capture: %s", oboe::convertToText(result));
g_running.store(false, std::memory_order_release);
g_capture_stream->close();
g_playout_stream->close();
g_capture_stream.reset();
g_playout_stream.reset();
return -4;
}
result = g_playout_stream->requestStart();
if (result != oboe::Result::OK) {
LOGE("Failed to start playout: %s", oboe::convertToText(result));
g_running.store(false, std::memory_order_release);
g_capture_stream->requestStop();
g_capture_stream->close();
g_playout_stream->close();
g_capture_stream.reset();
g_playout_stream.reset();
return -5;
}
LOGI("Oboe started: sr=%d burst=%d ch=%d",
config->sample_rate, config->frames_per_burst, config->channel_count);
return 0;
}
void wzp_oboe_stop(void) {
g_running.store(false, std::memory_order_release);
if (g_capture_stream) {
g_capture_stream->requestStop();
g_capture_stream->close();
g_capture_stream.reset();
}
if (g_playout_stream) {
g_playout_stream->requestStop();
g_playout_stream->close();
g_playout_stream.reset();
}
g_rings = nullptr;
LOGI("Oboe stopped");
}
float wzp_oboe_capture_latency_ms(void) {
return g_capture_latency_ms.load(std::memory_order_relaxed);
}
float wzp_oboe_playout_latency_ms(void) {
return g_playout_latency_ms.load(std::memory_order_relaxed);
}
int wzp_oboe_is_running(void) {
return g_running.load(std::memory_order_relaxed) ? 1 : 0;
}
#else
// Non-Android fallback — should not be reached; oboe_stub.cpp is used instead.
// Provide empty implementations just in case.
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
(void)config; (void)rings;
return -99;
}
void wzp_oboe_stop(void) {}
float wzp_oboe_capture_latency_ms(void) { return 0.0f; }
float wzp_oboe_playout_latency_ms(void) { return 0.0f; }
int wzp_oboe_is_running(void) { return 0; }
#endif // __ANDROID__

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#ifndef WZP_OBOE_BRIDGE_H
#define WZP_OBOE_BRIDGE_H
#include <stdint.h>
#ifdef __cplusplus
#include <atomic>
typedef std::atomic<int32_t> wzp_atomic_int;
extern "C" {
#else
#include <stdatomic.h>
typedef atomic_int wzp_atomic_int;
#endif
typedef struct {
int32_t sample_rate;
int32_t frames_per_burst;
int32_t channel_count;
} WzpOboeConfig;
typedef struct {
int16_t* capture_buf;
int32_t capture_capacity;
wzp_atomic_int* capture_write_idx;
wzp_atomic_int* capture_read_idx;
int16_t* playout_buf;
int32_t playout_capacity;
wzp_atomic_int* playout_write_idx;
wzp_atomic_int* playout_read_idx;
} WzpOboeRings;
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings);
void wzp_oboe_stop(void);
float wzp_oboe_capture_latency_ms(void);
float wzp_oboe_playout_latency_ms(void);
int wzp_oboe_is_running(void);
#ifdef __cplusplus
}
#endif
#endif // WZP_OBOE_BRIDGE_H

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// Stub implementation for non-Android host builds (testing, cargo check, etc.)
#include "oboe_bridge.h"
#include <stdio.h>
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
(void)config;
(void)rings;
fprintf(stderr, "wzp_oboe_start: stub (not on Android)\n");
return 0;
}
void wzp_oboe_stop(void) {
fprintf(stderr, "wzp_oboe_stop: stub (not on Android)\n");
}
float wzp_oboe_capture_latency_ms(void) {
return 0.0f;
}
float wzp_oboe_playout_latency_ms(void) {
return 0.0f;
}
int wzp_oboe_is_running(void) {
return 0;
}

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@@ -0,0 +1,424 @@
//! Lock-free SPSC ring buffer audio backend for Android (Oboe).
//!
//! The ring buffers are shared between Rust and C++: the Oboe callbacks
//! (running on a high-priority audio thread) read/write directly into
//! the buffers via atomic indices, while the Rust codec thread on the
//! other side does the same.
use std::sync::atomic::{AtomicI32, Ordering};
use tracing::info;
#[allow(unused_imports)]
use tracing::warn;
/// Number of samples per 20 ms frame at 48 kHz mono.
pub const FRAME_SAMPLES: usize = 960;
/// Default ring buffer capacity: 8 frames = 160 ms at 48 kHz.
const RING_CAPACITY: usize = 7680;
// ---------------------------------------------------------------------------
// FFI declarations matching oboe_bridge.h
// ---------------------------------------------------------------------------
#[repr(C)]
#[allow(non_snake_case)]
struct WzpOboeConfig {
sample_rate: i32,
frames_per_burst: i32,
channel_count: i32,
}
#[repr(C)]
#[allow(non_snake_case)]
struct WzpOboeRings {
capture_buf: *mut i16,
capture_capacity: i32,
capture_write_idx: *mut AtomicI32,
capture_read_idx: *mut AtomicI32,
playout_buf: *mut i16,
playout_capacity: i32,
playout_write_idx: *mut AtomicI32,
playout_read_idx: *mut AtomicI32,
}
unsafe impl Send for WzpOboeRings {}
unsafe impl Sync for WzpOboeRings {}
unsafe extern "C" {
fn wzp_oboe_start(config: *const WzpOboeConfig, rings: *const WzpOboeRings) -> i32;
fn wzp_oboe_stop();
fn wzp_oboe_capture_latency_ms() -> f32;
fn wzp_oboe_playout_latency_ms() -> f32;
fn wzp_oboe_is_running() -> i32;
}
// ---------------------------------------------------------------------------
// SPSC Ring Buffer
// ---------------------------------------------------------------------------
/// Single-producer single-consumer lock-free ring buffer.
///
/// The producer calls `write()` and the consumer calls `read()`.
/// Atomics use acquire/release ordering to ensure correct visibility
/// across the Oboe audio thread and the Rust codec thread.
pub struct RingBuffer {
buf: Vec<i16>,
capacity: usize,
write_idx: AtomicI32,
read_idx: AtomicI32,
}
impl RingBuffer {
/// Create a new ring buffer with the given capacity (in samples).
///
/// The actual usable capacity is `capacity - 1` to distinguish
/// full from empty.
pub fn new(capacity: usize) -> Self {
Self {
buf: vec![0i16; capacity],
capacity,
write_idx: AtomicI32::new(0),
read_idx: AtomicI32::new(0),
}
}
/// Number of samples available to read.
pub fn available_read(&self) -> usize {
let w = self.write_idx.load(Ordering::Acquire);
let r = self.read_idx.load(Ordering::Relaxed);
let avail = w - r;
if avail < 0 {
(avail + self.capacity as i32) as usize
} else {
avail as usize
}
}
/// Number of samples that can be written before the buffer is full.
pub fn available_write(&self) -> usize {
self.capacity - 1 - self.available_read()
}
/// Write samples into the ring buffer (producer side).
///
/// Returns the number of samples actually written (may be less than
/// `data.len()` if the buffer is nearly full).
pub fn write(&self, data: &[i16]) -> usize {
let avail = self.available_write();
let count = data.len().min(avail);
if count == 0 {
return 0;
}
let mut w = self.write_idx.load(Ordering::Relaxed) as usize;
let cap = self.capacity;
let buf_ptr = self.buf.as_ptr() as *mut i16;
for i in 0..count {
// SAFETY: w is always in [0, capacity) and we are the sole producer.
unsafe {
*buf_ptr.add(w) = data[i];
}
w += 1;
if w >= cap {
w = 0;
}
}
self.write_idx.store(w as i32, Ordering::Release);
count
}
/// Read samples from the ring buffer (consumer side).
///
/// Returns the number of samples actually read (may be less than
/// `out.len()` if the buffer doesn't have enough data).
pub fn read(&self, out: &mut [i16]) -> usize {
let avail = self.available_read();
let count = out.len().min(avail);
if count == 0 {
return 0;
}
let mut r = self.read_idx.load(Ordering::Relaxed) as usize;
let cap = self.capacity;
let buf_ptr = self.buf.as_ptr();
for i in 0..count {
// SAFETY: r is always in [0, capacity) and we are the sole consumer.
unsafe {
out[i] = *buf_ptr.add(r);
}
r += 1;
if r >= cap {
r = 0;
}
}
self.read_idx.store(r as i32, Ordering::Release);
count
}
/// Get a raw pointer to the buffer data (for FFI).
fn buf_ptr(&self) -> *mut i16 {
self.buf.as_ptr() as *mut i16
}
/// Get a raw pointer to the write index atomic (for FFI).
fn write_idx_ptr(&self) -> *mut AtomicI32 {
&self.write_idx as *const AtomicI32 as *mut AtomicI32
}
/// Get a raw pointer to the read index atomic (for FFI).
fn read_idx_ptr(&self) -> *mut AtomicI32 {
&self.read_idx as *const AtomicI32 as *mut AtomicI32
}
}
// SAFETY: The ring buffer is designed for SPSC use where producer and consumer
// are on different threads. The atomic indices provide the synchronization.
unsafe impl Send for RingBuffer {}
unsafe impl Sync for RingBuffer {}
// ---------------------------------------------------------------------------
// Oboe Backend
// ---------------------------------------------------------------------------
/// Oboe-based audio backend for Android.
///
/// Owns two SPSC ring buffers (capture and playout) that are shared with
/// the C++ Oboe callbacks via raw pointers. The Oboe callbacks run on
/// high-priority audio threads managed by the Android audio system.
pub struct OboeBackend {
capture_ring: RingBuffer,
playout_ring: RingBuffer,
started: bool,
}
impl OboeBackend {
/// Create a new backend with default ring buffer sizes (160 ms each).
pub fn new() -> Self {
Self {
capture_ring: RingBuffer::new(RING_CAPACITY),
playout_ring: RingBuffer::new(RING_CAPACITY),
started: false,
}
}
/// Start Oboe audio streams.
///
/// This sets up the ring buffer pointers and calls into the C++ layer
/// to open and start the capture and playout Oboe streams.
pub fn start(&mut self) -> Result<(), anyhow::Error> {
if self.started {
return Ok(());
}
let config = WzpOboeConfig {
sample_rate: 48_000,
frames_per_burst: FRAME_SAMPLES as i32,
channel_count: 1,
};
let rings = WzpOboeRings {
capture_buf: self.capture_ring.buf_ptr(),
capture_capacity: self.capture_ring.capacity as i32,
capture_write_idx: self.capture_ring.write_idx_ptr(),
capture_read_idx: self.capture_ring.read_idx_ptr(),
playout_buf: self.playout_ring.buf_ptr(),
playout_capacity: self.playout_ring.capacity as i32,
playout_write_idx: self.playout_ring.write_idx_ptr(),
playout_read_idx: self.playout_ring.read_idx_ptr(),
};
let ret = unsafe { wzp_oboe_start(&config, &rings) };
if ret != 0 {
return Err(anyhow::anyhow!("wzp_oboe_start failed with code {}", ret));
}
self.started = true;
info!("Oboe backend started");
Ok(())
}
/// Stop Oboe audio streams.
pub fn stop(&mut self) {
if !self.started {
return;
}
unsafe { wzp_oboe_stop() };
self.started = false;
info!("Oboe backend stopped");
}
/// Read captured audio samples from the capture ring buffer.
///
/// Returns the number of samples actually read. The caller should
/// provide a buffer of at least `FRAME_SAMPLES` (960) samples.
pub fn read_capture(&self, out: &mut [i16]) -> usize {
self.capture_ring.read(out)
}
/// Write audio samples to the playout ring buffer.
///
/// Returns the number of samples actually written.
pub fn write_playout(&self, samples: &[i16]) -> usize {
self.playout_ring.write(samples)
}
/// Get the current capture latency in milliseconds (from Oboe).
#[allow(unused)]
pub fn capture_latency_ms(&self) -> f32 {
unsafe { wzp_oboe_capture_latency_ms() }
}
/// Get the current playout latency in milliseconds (from Oboe).
#[allow(unused)]
pub fn playout_latency_ms(&self) -> f32 {
unsafe { wzp_oboe_playout_latency_ms() }
}
/// Check if the Oboe streams are currently running.
#[allow(unused)]
pub fn is_running(&self) -> bool {
unsafe { wzp_oboe_is_running() != 0 }
}
}
impl Drop for OboeBackend {
fn drop(&mut self) {
self.stop();
}
}
// ---------------------------------------------------------------------------
// Thread affinity / priority helpers
// ---------------------------------------------------------------------------
/// Pin the current thread to the highest-numbered CPU cores (big cores on
/// ARM big.LITTLE architectures). Falls back silently on failure.
#[allow(unused)]
pub fn pin_to_big_core() {
#[cfg(target_os = "android")]
{
unsafe {
let num_cpus = libc::sysconf(libc::_SC_NPROCESSORS_ONLN);
if num_cpus <= 0 {
warn!("pin_to_big_core: could not determine CPU count");
return;
}
let num_cpus = num_cpus as usize;
// Target the upper half of CPUs (big cores on most big.LITTLE SoCs)
let start = num_cpus / 2;
let mut set: libc::cpu_set_t = std::mem::zeroed();
libc::CPU_ZERO(&mut set);
for cpu in start..num_cpus {
libc::CPU_SET(cpu, &mut set);
}
let ret = libc::sched_setaffinity(
0, // current thread
std::mem::size_of::<libc::cpu_set_t>(),
&set,
);
if ret != 0 {
warn!("sched_setaffinity failed: {}", std::io::Error::last_os_error());
} else {
info!(start, num_cpus, "pinned to big cores");
}
}
}
#[cfg(not(target_os = "android"))]
{
// No-op on non-Android
}
}
/// Attempt to set SCHED_FIFO real-time priority for the current thread.
/// Falls back silently on failure (requires appropriate permissions on Android).
#[allow(unused)]
pub fn set_realtime_priority() {
#[cfg(target_os = "android")]
{
unsafe {
let param = libc::sched_param {
sched_priority: 2, // Low RT priority — enough for audio, safe
};
let ret = libc::sched_setscheduler(0, libc::SCHED_FIFO, &param);
if ret != 0 {
warn!(
"sched_setscheduler(SCHED_FIFO) failed: {}",
std::io::Error::last_os_error()
);
} else {
info!("set SCHED_FIFO priority 2");
}
}
}
#[cfg(not(target_os = "android"))]
{
// No-op on non-Android
}
}
// ---------------------------------------------------------------------------
// Tests
// ---------------------------------------------------------------------------
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn ring_buffer_write_read() {
let ring = RingBuffer::new(16);
let data = [1i16, 2, 3, 4, 5];
assert_eq!(ring.write(&data), 5);
assert_eq!(ring.available_read(), 5);
let mut out = [0i16; 5];
assert_eq!(ring.read(&mut out), 5);
assert_eq!(out, [1, 2, 3, 4, 5]);
assert_eq!(ring.available_read(), 0);
}
#[test]
fn ring_buffer_wraparound() {
let ring = RingBuffer::new(8);
let data = [10i16, 20, 30, 40, 50, 60]; // 6 samples, capacity 8 (usable 7)
assert_eq!(ring.write(&data), 6);
let mut out = [0i16; 4];
assert_eq!(ring.read(&mut out), 4);
assert_eq!(out, [10, 20, 30, 40]);
// Now write more, which should wrap around
let data2 = [70i16, 80, 90, 100];
assert_eq!(ring.write(&data2), 4);
let mut out2 = [0i16; 6];
assert_eq!(ring.read(&mut out2), 6);
assert_eq!(out2, [50, 60, 70, 80, 90, 100]);
}
#[test]
fn ring_buffer_full() {
let ring = RingBuffer::new(4); // usable capacity = 3
let data = [1i16, 2, 3, 4, 5];
assert_eq!(ring.write(&data), 3); // Only 3 fit
assert_eq!(ring.available_write(), 0);
}
#[test]
fn oboe_backend_stub_start_stop() {
let mut backend = OboeBackend::new();
backend.start().expect("stub start should succeed");
assert!(backend.started);
backend.stop();
assert!(!backend.started);
}
}

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//! Lock-free SPSC ring buffer — "Reader-Detects-Lap" architecture.
//!
//! SPSC invariant: the producer ONLY writes `write_pos`, the consumer
//! ONLY writes `read_pos`. Neither thread touches the other's cursor.
//!
//! On overflow (writer laps the reader), the writer simply overwrites
//! old buffer data. The reader detects the lap via `available() >
//! RING_CAPACITY` and snaps its own `read_pos` forward.
//!
//! Capacity is a power of 2 for bitmask indexing (no modulo).
use std::sync::atomic::{AtomicU64, AtomicUsize, Ordering};
/// Ring buffer capacity — power of 2 for bitmask indexing.
/// 16384 samples = 341.3ms at 48kHz mono. 70% more headroom
/// than the previous 9600 (200ms) for surviving Android GC pauses.
const RING_CAPACITY: usize = 16384; // 2^14
const RING_MASK: usize = RING_CAPACITY - 1;
/// Lock-free single-producer single-consumer ring buffer for i16 PCM samples.
pub struct AudioRing {
buf: Box<[i16]>,
/// Monotonically increasing write cursor. ONLY written by producer.
write_pos: AtomicUsize,
/// Monotonically increasing read cursor. ONLY written by consumer.
read_pos: AtomicUsize,
/// Incremented by reader when it detects it was lapped (overflow).
overflow_count: AtomicU64,
/// Incremented by reader when ring is empty (underrun).
underrun_count: AtomicU64,
}
// SAFETY: AudioRing is SPSC — one thread writes (producer), one reads (consumer).
// The producer only writes write_pos. The consumer only writes read_pos.
// Neither thread writes the other's cursor. Buffer indices are derived from
// the owning thread's cursor, ensuring no concurrent access to the same index.
unsafe impl Send for AudioRing {}
unsafe impl Sync for AudioRing {}
impl AudioRing {
pub fn new() -> Self {
debug_assert!(RING_CAPACITY.is_power_of_two());
Self {
buf: vec![0i16; RING_CAPACITY].into_boxed_slice(),
write_pos: AtomicUsize::new(0),
read_pos: AtomicUsize::new(0),
overflow_count: AtomicU64::new(0),
underrun_count: AtomicU64::new(0),
}
}
/// Number of samples available to read (clamped to capacity).
pub fn available(&self) -> usize {
let w = self.write_pos.load(Ordering::Acquire);
let r = self.read_pos.load(Ordering::Relaxed);
w.wrapping_sub(r).min(RING_CAPACITY)
}
/// Number of samples that can be written without overwriting unread data.
pub fn free_space(&self) -> usize {
RING_CAPACITY.saturating_sub(self.available())
}
/// Write samples into the ring. Returns number of samples written.
///
/// If the ring is full, old data is silently overwritten. The reader
/// will detect the lap and self-correct. The writer NEVER touches
/// `read_pos` — this is the key invariant that prevents cursor desync.
pub fn write(&self, samples: &[i16]) -> usize {
let count = samples.len().min(RING_CAPACITY);
let w = self.write_pos.load(Ordering::Relaxed);
for i in 0..count {
unsafe {
let ptr = self.buf.as_ptr() as *mut i16;
*ptr.add((w + i) & RING_MASK) = samples[i];
}
}
self.write_pos.store(w.wrapping_add(count), Ordering::Release);
count
}
/// Read samples from the ring into `out`. Returns number of samples read.
///
/// If the writer has lapped the reader (overflow), `read_pos` is snapped
/// forward to the oldest valid data. This is safe because only the
/// reader thread writes `read_pos`.
pub fn read(&self, out: &mut [i16]) -> usize {
let w = self.write_pos.load(Ordering::Acquire);
let mut r = self.read_pos.load(Ordering::Relaxed);
let mut avail = w.wrapping_sub(r);
// Lap detection: writer has overwritten our unread data.
// Snap read_pos forward to oldest valid data in the buffer.
if avail > RING_CAPACITY {
r = w.wrapping_sub(RING_CAPACITY);
avail = RING_CAPACITY;
self.overflow_count.fetch_add(1, Ordering::Relaxed);
}
let count = out.len().min(avail);
if count == 0 {
if w == r {
self.underrun_count.fetch_add(1, Ordering::Relaxed);
}
return 0;
}
for i in 0..count {
out[i] = unsafe { *self.buf.as_ptr().add((r + i) & RING_MASK) };
}
self.read_pos.store(r.wrapping_add(count), Ordering::Release);
count
}
/// Number of overflow events (reader was lapped by writer).
pub fn overflow_count(&self) -> u64 {
self.overflow_count.load(Ordering::Relaxed)
}
/// Number of underrun events (reader found empty buffer).
pub fn underrun_count(&self) -> u64 {
self.underrun_count.load(Ordering::Relaxed)
}
}

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//! Engine commands sent from the JNI/UI thread to the engine.
use wzp_proto::QualityProfile;
/// Commands that can be sent to the running engine.
pub enum EngineCommand {
/// Mute or unmute the microphone.
SetMute(bool),
/// Enable or disable speaker (loudspeaker) mode.
SetSpeaker(bool),
/// Force a specific quality profile (overrides adaptive logic).
ForceProfile(QualityProfile),
/// Stop the call and shut down the engine.
Stop,
/// Place a direct call to a fingerprint (requires signal connection).
PlaceCall { target_fingerprint: String },
/// Answer an incoming direct call.
AnswerCall {
call_id: String,
accept_mode: wzp_proto::CallAcceptMode,
},
/// Reject an incoming direct call.
RejectCall { call_id: String },
}

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//! JNI bridge for Android — thin layer between Kotlin and the WzpEngine.
use std::panic;
use std::sync::Once;
use jni::objects::{JClass, JObject, JString};
use jni::sys::{jboolean, jint, jlong, jstring};
use jni::JNIEnv;
use tracing::{error, info};
use wzp_proto::QualityProfile;
use crate::engine::{CallStartConfig, WzpEngine};
/// Opaque engine handle passed to/from Kotlin as a `jlong`.
struct EngineHandle {
engine: WzpEngine,
}
/// Recover the `EngineHandle` from a raw handle value.
unsafe fn handle_ref(handle: jlong) -> &'static mut EngineHandle {
unsafe { &mut *(handle as *mut EngineHandle) }
}
/// 7 = auto (use relay's chosen profile)
const PROFILE_AUTO: jint = 7;
fn profile_from_int(value: jint) -> QualityProfile {
match value {
0 => QualityProfile::GOOD, // Opus 24k
1 => QualityProfile::DEGRADED, // Opus 6k
2 => QualityProfile::CATASTROPHIC, // Codec2 1.2k
3 => QualityProfile { // Codec2 3.2k
codec: wzp_proto::CodecId::Codec2_3200,
fec_ratio: 0.5,
frame_duration_ms: 20,
frames_per_block: 5,
},
4 => QualityProfile::STUDIO_32K, // Opus 32k
5 => QualityProfile::STUDIO_48K, // Opus 48k
6 => QualityProfile::STUDIO_64K, // Opus 64k
_ => QualityProfile::GOOD, // auto falls back to GOOD
}
}
static INIT_LOGGING: Once = Once::new();
/// Initialize tracing → Android logcat (tag "wzp_android").
/// Safe to call multiple times — only the first call takes effect.
fn init_logging() {
INIT_LOGGING.call_once(|| {
// Wrap in catch_unwind — sharded_slab allocation inside
// tracing_subscriber::registry() can crash on some Android
// devices if scudo malloc fails during early initialization.
let _ = std::panic::catch_unwind(|| {
use tracing_subscriber::layer::SubscriberExt;
use tracing_subscriber::util::SubscriberInitExt;
use tracing_subscriber::EnvFilter;
if let Ok(layer) = tracing_android::layer("wzp_android") {
// Filter: INFO for our crates, WARN for everything else.
// The jni crate emits VERBOSE logs for every method lookup
// (~10 lines per JNI call, 100+ calls/sec) which floods logcat
// and causes the system to kill the app.
let filter = EnvFilter::new("warn,wzp_android=info,wzp_proto=info,wzp_transport=info,wzp_codec=info,wzp_fec=info,wzp_crypto=info");
let _ = tracing_subscriber::registry()
.with(layer)
.with(filter)
.try_init();
}
});
});
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeInit(
_env: JNIEnv,
_class: JClass,
) -> jlong {
let result = panic::catch_unwind(|| {
init_logging();
let handle = Box::new(EngineHandle {
engine: WzpEngine::new(),
});
Box::into_raw(handle) as jlong
});
match result {
Ok(h) => h,
Err(_) => 0,
}
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeStartCall(
mut env: JNIEnv,
_class: JClass,
handle: jlong,
relay_addr_j: JString,
room_j: JString,
seed_hex_j: JString,
token_j: JString,
alias_j: JString,
profile_j: jint,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let relay_addr: String = env.get_string(&relay_addr_j).map(|s| s.into()).unwrap_or_default();
let room: String = env.get_string(&room_j).map(|s| s.into()).unwrap_or_default();
let seed_hex: String = env.get_string(&seed_hex_j).map(|s| s.into()).unwrap_or_default();
let token: String = env.get_string(&token_j).map(|s| s.into()).unwrap_or_default();
let alias: String = env.get_string(&alias_j).map(|s| s.into()).unwrap_or_default();
let h = unsafe { handle_ref(handle) };
// Parse hex seed
let mut identity_seed = [0u8; 32];
if seed_hex.len() == 64 {
for i in 0..32 {
if let Ok(byte) = u8::from_str_radix(&seed_hex[i * 2..i * 2 + 2], 16) {
identity_seed[i] = byte;
}
}
} else {
// Generate random seed if not provided
use rand::RngCore;
rand::thread_rng().fill_bytes(&mut identity_seed);
}
let config = CallStartConfig {
profile: profile_from_int(profile_j),
auto_profile: profile_j == PROFILE_AUTO,
relay_addr,
room,
auth_token: if token.is_empty() { Vec::new() } else { token.into_bytes() },
identity_seed,
alias: if alias.is_empty() { None } else { Some(alias) },
};
match h.engine.start_call(config) {
Ok(()) => 0,
Err(e) => {
error!("start_call failed: {e}");
-1
}
}
}));
match result {
Ok(code) => code,
Err(_) => -1,
}
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeStopCall(
_env: JNIEnv,
_class: JClass,
handle: jlong,
) {
let _ = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
h.engine.stop_call();
}));
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeSetMute(
_env: JNIEnv,
_class: JClass,
handle: jlong,
muted: jboolean,
) {
let _ = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
h.engine.set_mute(muted != 0);
}));
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeSetSpeaker(
_env: JNIEnv,
_class: JClass,
handle: jlong,
speaker: jboolean,
) {
let _ = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
h.engine.set_speaker(speaker != 0);
}));
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeGetStats<'a>(
mut env: JNIEnv<'a>,
_class: JClass,
handle: jlong,
) -> jstring {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let stats = h.engine.get_stats();
serde_json::to_string(&stats).unwrap_or_else(|_| "{}".to_string())
}));
let json = match result {
Ok(s) => s,
Err(_) => "{}".to_string(),
};
env.new_string(&json)
.map(|s| s.into_raw())
.unwrap_or(JObject::null().into_raw())
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeForceProfile(
_env: JNIEnv,
_class: JClass,
handle: jlong,
profile: jint,
) {
let _ = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let qp = profile_from_int(profile);
h.engine.force_profile(qp);
}));
}
/// Write captured PCM samples from Kotlin AudioRecord into the engine's capture ring.
/// pcm is a Java short[] array.
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeWriteAudio(
env: JNIEnv,
_class: JClass,
handle: jlong,
pcm: jni::objects::JShortArray,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let len = env.get_array_length(&pcm).unwrap_or(0) as usize;
if len == 0 {
return 0;
}
let mut buf = vec![0i16; len];
if env.get_short_array_region(&pcm, 0, &mut buf).is_err() {
return 0;
}
h.engine.write_audio(&buf) as jint
}));
result.unwrap_or(0)
}
/// Read decoded PCM samples from the engine's playout ring for Kotlin AudioTrack.
/// pcm is a Java short[] array to fill. Returns number of samples actually read.
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeReadAudio(
env: JNIEnv,
_class: JClass,
handle: jlong,
pcm: jni::objects::JShortArray,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let len = env.get_array_length(&pcm).unwrap_or(0) as usize;
if len == 0 {
return 0;
}
let mut buf = vec![0i16; len];
let read = h.engine.read_audio(&mut buf);
if read > 0 {
let _ = env.set_short_array_region(&pcm, 0, &buf[..read]);
}
read as jint
}));
result.unwrap_or(0)
}
/// Write captured PCM from a DirectByteBuffer — zero JNI array copies.
/// The ByteBuffer must contain little-endian i16 samples.
/// Called from the AudioRecord capture thread.
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeWriteAudioDirect(
env: JNIEnv,
_class: JClass,
handle: jlong,
buffer: jni::objects::JByteBuffer,
sample_count: jint,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let ptr = env.get_direct_buffer_address(&buffer).unwrap_or(std::ptr::null_mut());
if ptr.is_null() || sample_count <= 0 {
return 0;
}
let samples = unsafe {
std::slice::from_raw_parts(ptr as *const i16, sample_count as usize)
};
h.engine.write_audio(samples) as jint
}));
result.unwrap_or(0)
}
/// Read decoded PCM into a DirectByteBuffer — zero JNI array copies.
/// The ByteBuffer will be filled with little-endian i16 samples.
/// Called from the AudioTrack playout thread.
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeReadAudioDirect(
env: JNIEnv,
_class: JClass,
handle: jlong,
buffer: jni::objects::JByteBuffer,
max_samples: jint,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let ptr = env.get_direct_buffer_address(&buffer).unwrap_or(std::ptr::null_mut());
if ptr.is_null() || max_samples <= 0 {
return 0;
}
let samples = unsafe {
std::slice::from_raw_parts_mut(ptr as *mut i16, max_samples as usize)
};
h.engine.read_audio(samples) as jint
}));
result.unwrap_or(0)
}
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeDestroy(
_env: JNIEnv,
_class: JClass,
handle: jlong,
) {
let _ = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { Box::from_raw(handle as *mut EngineHandle) };
drop(h);
}));
}
/// Ping a relay server — instance method, requires engine handle.
/// Returns JSON `{"rtt_ms":N,"server_fingerprint":"hex"}` or null on failure.
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativePingRelay<'a>(
mut env: JNIEnv<'a>,
_class: JClass,
handle: jlong,
relay_j: JString,
) -> jstring {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let relay: String = env.get_string(&relay_j).map(|s| s.into()).unwrap_or_default();
match h.engine.ping_relay(&relay) {
Ok(json) => Some(json),
Err(_) => None,
}
}));
let json = match result {
Ok(Some(s)) => s,
_ => return JObject::null().into_raw(),
};
env.new_string(&json)
.map(|s| s.into_raw())
.unwrap_or(JObject::null().into_raw())
}
// ── Direct calling JNI functions ──
/// Start persistent signaling connection to relay for direct calls.
/// Returns 0 on success, -1 on error.
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeStartSignaling<'a>(
mut env: JNIEnv<'a>,
_class: JClass,
handle: jlong,
relay_addr_j: JString,
seed_hex_j: JString,
token_j: JString,
alias_j: JString,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let relay_addr: String = env.get_string(&relay_addr_j).map(|s| s.into()).unwrap_or_default();
let seed_hex: String = env.get_string(&seed_hex_j).map(|s| s.into()).unwrap_or_default();
let token: String = env.get_string(&token_j).map(|s| s.into()).unwrap_or_default();
let alias: String = env.get_string(&alias_j).map(|s| s.into()).unwrap_or_default();
h.engine.start_signaling(
&relay_addr,
&seed_hex,
if token.is_empty() { None } else { Some(&token) },
if alias.is_empty() { None } else { Some(&alias) },
)
}));
match result {
Ok(Ok(())) => 0,
Ok(Err(e)) => { error!("start_signaling failed: {e}"); -1 }
Err(_) => { error!("start_signaling panicked"); -1 }
}
}
/// Place a direct call to a target fingerprint.
/// Returns 0 on success, -1 on error.
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativePlaceCall<'a>(
mut env: JNIEnv<'a>,
_class: JClass,
handle: jlong,
target_fp_j: JString,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let target: String = env.get_string(&target_fp_j).map(|s| s.into()).unwrap_or_default();
h.engine.place_call(&target)
}));
match result {
Ok(Ok(())) => 0,
Ok(Err(e)) => { error!("place_call failed: {e}"); -1 }
Err(_) => { error!("place_call panicked"); -1 }
}
}
/// Answer an incoming direct call.
/// mode: 0=Reject, 1=AcceptTrusted, 2=AcceptGeneric
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeAnswerCall<'a>(
mut env: JNIEnv<'a>,
_class: JClass,
handle: jlong,
call_id_j: JString,
mode: jint,
) -> jint {
let result = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
let call_id: String = env.get_string(&call_id_j).map(|s| s.into()).unwrap_or_default();
let accept_mode = match mode {
0 => wzp_proto::CallAcceptMode::Reject,
1 => wzp_proto::CallAcceptMode::AcceptTrusted,
_ => wzp_proto::CallAcceptMode::AcceptGeneric,
};
h.engine.answer_call(&call_id, accept_mode)
}));
match result {
Ok(Ok(())) => 0,
Ok(Err(e)) => { error!("answer_call failed: {e}"); -1 }
Err(_) => { error!("answer_call panicked"); -1 }
}
}

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//! WarzonePhone Android native VoIP engine.
//!
//! Provides:
//! - Oboe audio backend with lock-free SPSC ring buffers
//! - Engine orchestrator managing call lifecycle
//! - Codec pipeline thread (encode/decode/FEC/jitter)
//! - Call statistics and command interface
//!
//! On non-Android targets, the Oboe C++ layer compiles as a stub,
//! allowing `cargo check` and unit tests on the host.
pub mod audio_android;
pub mod audio_ring;
pub mod commands;
pub mod engine;
pub mod pipeline;
pub mod stats;
pub mod jni_bridge;

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//! Codec pipeline — encode/decode with FEC and jitter buffer.
//!
//! Runs on a dedicated thread, processing 20 ms frames at 48 kHz.
//! The pipeline is NOT Send/Sync (Opus encoder state) — it is owned
//! exclusively by the codec thread.
use tracing::{debug, warn};
use wzp_codec::{AdaptiveDecoder, AdaptiveEncoder, AutoGainControl, EchoCanceller};
use wzp_fec::{RaptorQFecDecoder, RaptorQFecEncoder};
use wzp_proto::jitter::{JitterBuffer, PlayoutResult};
use wzp_proto::quality::AdaptiveQualityController;
use wzp_proto::traits::{AudioDecoder, AudioEncoder, FecDecoder, FecEncoder};
use wzp_proto::traits::QualityController;
use wzp_proto::{MediaPacket, QualityProfile};
use crate::audio_android::FRAME_SAMPLES;
/// Maximum encoded frame size (Opus worst case at highest bitrate).
const MAX_ENCODED_BYTES: usize = 1275;
/// Pipeline statistics snapshot.
#[derive(Clone, Debug, Default)]
pub struct PipelineStats {
pub frames_encoded: u64,
pub frames_decoded: u64,
pub underruns: u64,
pub jitter_depth: usize,
pub quality_tier: u8,
}
/// The codec pipeline: encode, FEC, jitter buffer, decode.
///
/// This struct is owned by the codec thread and not shared.
pub struct Pipeline {
encoder: AdaptiveEncoder,
decoder: AdaptiveDecoder,
fec_encoder: RaptorQFecEncoder,
fec_decoder: RaptorQFecDecoder,
jitter_buffer: JitterBuffer,
quality_ctrl: AdaptiveQualityController,
/// Acoustic echo canceller applied before encoding.
aec: EchoCanceller,
/// Automatic gain control applied before encoding.
agc: AutoGainControl,
/// Last decoded PCM frame, used as the AEC far-end reference.
last_decoded_farend: Option<Vec<i16>>,
// Pre-allocated scratch buffers
capture_buf: Vec<i16>,
#[allow(dead_code)]
playout_buf: Vec<i16>,
encode_out: Vec<u8>,
// Stats counters
frames_encoded: u64,
frames_decoded: u64,
underruns: u64,
}
impl Pipeline {
/// Create a new pipeline configured for the given quality profile.
pub fn new(profile: QualityProfile) -> Result<Self, anyhow::Error> {
let encoder = AdaptiveEncoder::new(profile)
.map_err(|e| anyhow::anyhow!("encoder init: {e}"))?;
let decoder = AdaptiveDecoder::new(profile)
.map_err(|e| anyhow::anyhow!("decoder init: {e}"))?;
let fec_encoder =
RaptorQFecEncoder::with_defaults(profile.frames_per_block as usize);
let fec_decoder =
RaptorQFecDecoder::with_defaults(profile.frames_per_block as usize);
let jitter_buffer = JitterBuffer::new(10, 250, 3);
let quality_ctrl = AdaptiveQualityController::new();
Ok(Self {
encoder,
decoder,
fec_encoder,
fec_decoder,
jitter_buffer,
quality_ctrl,
aec: EchoCanceller::new(48000, 100), // 100 ms echo tail
agc: AutoGainControl::new(),
last_decoded_farend: None,
capture_buf: vec![0i16; FRAME_SAMPLES],
playout_buf: vec![0i16; FRAME_SAMPLES],
encode_out: vec![0u8; MAX_ENCODED_BYTES],
frames_encoded: 0,
frames_decoded: 0,
underruns: 0,
})
}
/// Encode a PCM frame into a compressed packet.
///
/// If `muted` is true, a silence frame is encoded (all zeros).
/// Returns the encoded bytes, or `None` on encoder error.
pub fn encode_frame(&mut self, pcm: &[i16], muted: bool) -> Option<Vec<u8>> {
let input = if muted {
// Zero the capture buffer for silence
for s in self.capture_buf.iter_mut() {
*s = 0;
}
&self.capture_buf[..]
} else {
// Feed the last decoded playout as AEC far-end reference.
if let Some(ref farend) = self.last_decoded_farend {
self.aec.feed_farend(farend);
}
// Apply AEC + AGC to the captured PCM.
let len = pcm.len().min(self.capture_buf.len());
self.capture_buf[..len].copy_from_slice(&pcm[..len]);
self.aec.process_frame(&mut self.capture_buf[..len]);
self.agc.process_frame(&mut self.capture_buf[..len]);
&self.capture_buf[..len]
};
match self.encoder.encode(input, &mut self.encode_out) {
Ok(n) => {
self.frames_encoded += 1;
let encoded = self.encode_out[..n].to_vec();
// Feed into FEC encoder
if let Err(e) = self.fec_encoder.add_source_symbol(&encoded) {
warn!("FEC encode error: {e}");
}
Some(encoded)
}
Err(e) => {
warn!("encode error: {e}");
None
}
}
}
/// Feed a received media packet into the jitter buffer.
pub fn feed_packet(&mut self, packet: MediaPacket) {
// Feed FEC symbols if present
let header = &packet.header;
if header.fec_block != 0 || header.fec_symbol != 0 {
let is_repair = header.is_repair;
if let Err(e) = self.fec_decoder.add_symbol(
header.fec_block,
header.fec_symbol,
is_repair,
&packet.payload,
) {
debug!("FEC symbol feed error: {e}");
}
}
self.jitter_buffer.push(packet);
}
/// Decode the next frame from the jitter buffer.
///
/// Returns decoded PCM samples, or `None` if the buffer is not ready.
/// Decoded PCM is also stored as the AEC far-end reference for the next
/// encode cycle.
pub fn decode_frame(&mut self) -> Option<Vec<i16>> {
let result = match self.jitter_buffer.pop() {
PlayoutResult::Packet(pkt) => {
let mut pcm = vec![0i16; FRAME_SAMPLES];
match self.decoder.decode(&pkt.payload, &mut pcm) {
Ok(n) => {
self.frames_decoded += 1;
pcm.truncate(n);
Some(pcm)
}
Err(e) => {
warn!("decode error: {e}");
// Attempt PLC
self.generate_plc()
}
}
}
PlayoutResult::Missing { seq } => {
debug!(seq, "jitter buffer: missing packet, generating PLC");
self.generate_plc()
}
PlayoutResult::NotReady => {
self.underruns += 1;
None
}
};
// Save decoded PCM as far-end reference for AEC.
if let Some(ref pcm) = result {
self.last_decoded_farend = Some(pcm.clone());
}
result
}
/// Generate packet loss concealment output.
fn generate_plc(&mut self) -> Option<Vec<i16>> {
let mut pcm = vec![0i16; FRAME_SAMPLES];
match self.decoder.decode_lost(&mut pcm) {
Ok(n) => {
self.frames_decoded += 1;
pcm.truncate(n);
Some(pcm)
}
Err(e) => {
warn!("PLC error: {e}");
None
}
}
}
/// Feed a quality report into the adaptive quality controller.
///
/// Returns a new profile if a tier transition occurred.
#[allow(unused)]
pub fn observe_quality(
&mut self,
report: &wzp_proto::QualityReport,
) -> Option<QualityProfile> {
let new_profile = self.quality_ctrl.observe(report);
if let Some(ref profile) = new_profile {
if let Err(e) = self.encoder.set_profile(*profile) {
warn!("encoder set_profile error: {e}");
}
if let Err(e) = self.decoder.set_profile(*profile) {
warn!("decoder set_profile error: {e}");
}
}
new_profile
}
/// Force a specific quality profile.
#[allow(unused)]
pub fn force_profile(&mut self, profile: QualityProfile) {
self.quality_ctrl.force_profile(profile);
if let Err(e) = self.encoder.set_profile(profile) {
warn!("encoder set_profile error: {e}");
}
if let Err(e) = self.decoder.set_profile(profile) {
warn!("decoder set_profile error: {e}");
}
}
/// Get current pipeline statistics.
pub fn stats(&self) -> PipelineStats {
PipelineStats {
frames_encoded: self.frames_encoded,
frames_decoded: self.frames_decoded,
underruns: self.underruns,
jitter_depth: self.jitter_buffer.stats().current_depth,
quality_tier: self.quality_ctrl.tier() as u8,
}
}
/// Enable or disable acoustic echo cancellation.
pub fn set_aec_enabled(&mut self, enabled: bool) {
self.aec.set_enabled(enabled);
}
/// Enable or disable automatic gain control.
pub fn set_agc_enabled(&mut self, enabled: bool) {
self.agc.set_enabled(enabled);
}
}

View File

@@ -0,0 +1,101 @@
//! Call statistics for the Android engine.
/// State of the call.
/// Serializes as integer for easy parsing on the Kotlin side:
/// 0=Idle, 1=Connecting, 2=Active, 3=Reconnecting, 4=Closed
#[derive(Clone, Debug, Default, PartialEq, Eq)]
pub enum CallState {
#[default]
Idle,
Connecting,
Active,
Reconnecting,
Closed,
/// Connected to relay signal channel, registered for direct calls.
Registered,
/// Outgoing call ringing on callee's side.
Ringing,
/// Incoming call received, waiting for user to accept/reject.
IncomingCall,
}
impl serde::Serialize for CallState {
fn serialize<S: serde::Serializer>(&self, serializer: S) -> Result<S::Ok, S::Error> {
let n: u8 = match self {
CallState::Idle => 0,
CallState::Connecting => 1,
CallState::Active => 2,
CallState::Reconnecting => 3,
CallState::Closed => 4,
CallState::Registered => 5,
CallState::Ringing => 6,
CallState::IncomingCall => 7,
};
serializer.serialize_u8(n)
}
}
/// Aggregated call statistics, serializable for JNI bridge.
#[derive(Clone, Debug, Default, serde::Serialize)]
pub struct CallStats {
/// Current call state.
pub state: CallState,
/// Call duration in seconds.
pub duration_secs: f64,
/// Current quality tier (0=GOOD, 1=DEGRADED, 2=CATASTROPHIC).
pub quality_tier: u8,
/// Observed packet loss percentage.
pub loss_pct: f32,
/// Smoothed round-trip time in milliseconds.
pub rtt_ms: u32,
/// Jitter in milliseconds.
pub jitter_ms: u32,
/// Current jitter buffer depth in packets.
pub jitter_buffer_depth: usize,
/// Total frames encoded since call start.
pub frames_encoded: u64,
/// Total frames decoded since call start.
pub frames_decoded: u64,
/// Number of playout underruns (buffer empty when audio needed).
pub underruns: u64,
/// Frames recovered by FEC.
pub fec_recovered: u64,
/// Playout ring overflow count (reader was lapped by writer).
pub playout_overflows: u64,
/// Playout ring underrun count (reader found empty buffer).
pub playout_underruns: u64,
/// Capture ring overflow count.
pub capture_overflows: u64,
/// Current mic audio level (RMS of i16 samples, 0-32767).
pub audio_level: u32,
/// Our current outgoing codec name (e.g. "Opus24k", "Codec2_1200").
pub current_codec: String,
/// Last seen incoming codec from other participants.
pub peer_codec: String,
/// Whether auto quality mode is active.
pub auto_mode: bool,
/// Number of participants in the room (from last RoomUpdate).
pub room_participant_count: u32,
/// Participant list (fingerprint + optional alias) serialized as JSON array.
pub room_participants: Vec<RoomMember>,
/// SAS code for verbal verification (None if not in a call).
#[serde(skip_serializing_if = "Option::is_none")]
pub sas_code: Option<u32>,
/// Incoming call info (present when state == IncomingCall).
#[serde(skip_serializing_if = "Option::is_none")]
pub incoming_call_id: Option<String>,
/// Fingerprint of the caller (present when state == IncomingCall).
#[serde(skip_serializing_if = "Option::is_none")]
pub incoming_caller_fp: Option<String>,
/// Alias of the caller (present when state == IncomingCall).
#[serde(skip_serializing_if = "Option::is_none")]
pub incoming_caller_alias: Option<String>,
}
/// A room member entry, serialized into the stats JSON.
#[derive(Clone, Debug, Default, serde::Serialize)]
pub struct RoomMember {
pub fingerprint: String,
pub alias: Option<String>,
pub relay_label: Option<String>,
}

View File

@@ -23,10 +23,71 @@ serde_json = "1"
chrono = "0.4"
rustls = { version = "0.23", default-features = false, features = ["ring", "std"] }
cpal = { version = "0.15", optional = true }
libc = "0.2"
# coreaudio-rs is Apple-framework-only; gate it to macOS so enabling
# the `vpio` feature from a non-macOS target builds cleanly instead of
# pulling in a crate that can only link against Apple frameworks.
[target.'cfg(target_os = "macos")'.dependencies]
coreaudio-rs = { version = "0.11", optional = true }
# Windows-only: direct WASAPI bindings for the `windows-aec` feature.
# `windows` is Microsoft's official Rust COM bindings crate. We pull in
# only the audio + COM subfeatures we need — the crate is organized as
# a massive optional-feature tree, so enabling just these keeps compile
# times reasonable (~5s for these features vs ~60s for the full crate).
[target.'cfg(target_os = "windows")'.dependencies]
windows = { version = "0.58", optional = true, features = [
"Win32_Foundation",
"Win32_Media_Audio",
"Win32_Security",
"Win32_System_Com",
"Win32_System_Com_StructuredStorage",
"Win32_System_Threading",
"Win32_System_Variant",
] }
# Linux-only: WebRTC AEC (Audio Processing Module) bindings for the
# `linux-aec` feature. This is the 0.3.x line of the `tonarino/
# webrtc-audio-processing` crate, which links against Debian's
# `libwebrtc-audio-processing-dev` apt package (0.3-1+b1 on Bookworm).
#
# Note: we attempted the 2.x line with its `bundled` sub-feature first
# (which would give us AEC3 instead of AEC2), but both the crates.io
# tarball AND the upstream git `main` branch of webrtc-audio-processing-sys
# 2.0.3 hit a `meson setup --reconfigure` bug where the build.rs passes
# --reconfigure unconditionally even on first-run empty build dirs,
# causing the bundled build to fail with "Directory does not contain a
# valid build tree". The 0.x line doesn't use bundled mode and sidesteps
# this entirely by linking the apt-provided library. AEC2 is older than
# AEC3 but still the same algorithm family — this is what PulseAudio's
# module-echo-cancel and PipeWire's filter-chain use by default on
# current Debian-family distros.
[target.'cfg(target_os = "linux")'.dependencies]
webrtc-audio-processing = { version = "0.3", optional = true }
[features]
default = []
audio = ["cpal"]
# vpio enables coreaudio-rs but that dep is itself gated to macOS above,
# so enabling this feature on Windows/Linux is a no-op (the audio_vpio
# module is also #[cfg(target_os = "macos")] in lib.rs).
vpio = ["dep:coreaudio-rs"]
# windows-aec enables a direct WASAPI capture backend that opens the
# microphone under AudioCategory_Communications, turning on Windows's
# OS-level communications audio processing (AEC + noise suppression +
# AGC). The `windows` dep is itself target-gated to Windows above, so
# enabling this feature on non-Windows targets is a no-op (the
# audio_wasapi module is also #[cfg(target_os = "windows")] in lib.rs).
windows-aec = ["dep:windows"]
# linux-aec enables a CPAL + WebRTC AEC3 capture/playback backend that
# runs the WebRTC Audio Processing Module (same algo as Chrome / Zoom /
# Teams) in-process, using the playback PCM as the reference signal for
# echo cancellation. The webrtc-audio-processing dep is target-gated to
# Linux above, so enabling this feature on non-Linux targets is a no-op
# (the audio_linux_aec module is also #[cfg(target_os = "linux")] in
# lib.rs).
linux-aec = ["dep:webrtc-audio-processing"]
[[bin]]
name = "wzp-client"

View File

@@ -3,12 +3,10 @@
//! Both structs use 48 kHz, mono, i16 format to match the WarzonePhone codec
//! pipeline. Frames are 960 samples (20 ms at 48 kHz).
//!
//! The cpal `Stream` type is not `Send`, so each struct spawns a dedicated OS
//! thread that owns the stream. The public API exposes only `Send + Sync`
//! channel handles.
//! Audio callbacks are **lock-free**: they read/write directly to an `AudioRing`
//! (atomic SPSC ring buffer). No Mutex, no channel, no allocation on the hot path.
use std::sync::atomic::{AtomicBool, Ordering};
use std::sync::mpsc;
use std::sync::Arc;
use anyhow::{anyhow, Context};
@@ -16,6 +14,8 @@ use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
use cpal::{SampleFormat, SampleRate, StreamConfig};
use tracing::{info, warn};
use crate::audio_ring::AudioRing;
/// Number of samples per 20 ms frame at 48 kHz mono.
pub const FRAME_SAMPLES: usize = 960;
@@ -23,22 +23,24 @@ pub const FRAME_SAMPLES: usize = 960;
// AudioCapture
// ---------------------------------------------------------------------------
/// Captures microphone input and yields 960-sample PCM frames.
/// Captures microphone input via CPAL and writes PCM into a lock-free ring buffer.
///
/// The cpal stream lives on a dedicated OS thread; this handle is `Send + Sync`.
pub struct AudioCapture {
rx: mpsc::Receiver<Vec<i16>>,
ring: Arc<AudioRing>,
running: Arc<AtomicBool>,
}
impl AudioCapture {
/// Create and start capturing from the default input device at 48 kHz mono.
pub fn start() -> Result<Self, anyhow::Error> {
let (tx, rx) = mpsc::sync_channel::<Vec<i16>>(64);
let ring = Arc::new(AudioRing::new());
let running = Arc::new(AtomicBool::new(true));
let running_clone = running.clone();
let (init_tx, init_rx) = mpsc::sync_channel::<Result<(), String>>(1);
let (init_tx, init_rx) = std::sync::mpsc::sync_channel::<Result<(), String>>(1);
let ring_cb = ring.clone();
let running_clone = running.clone();
std::thread::Builder::new()
.name("wzp-audio-capture".into())
@@ -59,53 +61,51 @@ impl AudioCapture {
let use_f32 = !supports_i16_input(&device)?;
let buf = Arc::new(std::sync::Mutex::new(
Vec::<i16>::with_capacity(FRAME_SAMPLES),
));
let err_cb = |e: cpal::StreamError| {
warn!("input stream error: {e}");
};
let logged_cb_size = Arc::new(AtomicBool::new(false));
let stream = if use_f32 {
let buf = buf.clone();
let tx = tx.clone();
let ring = ring_cb.clone();
let running = running_clone.clone();
let logged = logged_cb_size.clone();
device.build_input_stream(
&config,
move |data: &[f32], _: &cpal::InputCallbackInfo| {
if !running.load(Ordering::Relaxed) {
return;
}
let mut lock = buf.lock().unwrap();
for &s in data {
lock.push(f32_to_i16(s));
if lock.len() == FRAME_SAMPLES {
let frame = lock.drain(..).collect();
let _ = tx.try_send(frame);
if !logged.swap(true, Ordering::Relaxed) {
eprintln!("[audio] capture callback: {} f32 samples", data.len());
}
let mut tmp = [0i16; FRAME_SAMPLES];
for chunk in data.chunks(FRAME_SAMPLES) {
let n = chunk.len();
for i in 0..n {
tmp[i] = f32_to_i16(chunk[i]);
}
ring.write(&tmp[..n]);
}
},
err_cb,
None,
)?
} else {
let buf = buf.clone();
let tx = tx.clone();
let ring = ring_cb.clone();
let running = running_clone.clone();
let logged = logged_cb_size.clone();
device.build_input_stream(
&config,
move |data: &[i16], _: &cpal::InputCallbackInfo| {
if !running.load(Ordering::Relaxed) {
return;
}
let mut lock = buf.lock().unwrap();
for &s in data {
lock.push(s);
if lock.len() == FRAME_SAMPLES {
let frame = lock.drain(..).collect();
let _ = tx.try_send(frame);
}
if !logged.swap(true, Ordering::Relaxed) {
eprintln!("[audio] capture callback: {} i16 samples", data.len());
}
ring.write(data);
},
err_cb,
None,
@@ -114,7 +114,6 @@ impl AudioCapture {
stream.play().context("failed to start input stream")?;
// Signal success to the caller before parking.
let _ = init_tx.send(Ok(()));
// Keep stream alive until stopped.
@@ -135,15 +134,12 @@ impl AudioCapture {
.map_err(|_| anyhow!("capture thread exited before signaling"))?
.map_err(|e| anyhow!("{e}"))?;
Ok(Self { rx, running })
Ok(Self { ring, running })
}
/// Read the next frame of 960 PCM samples (blocking until available).
///
/// Returns `None` when the stream has been stopped or the channel is
/// disconnected.
pub fn read_frame(&self) -> Option<Vec<i16>> {
self.rx.recv().ok()
/// Get a reference to the capture ring buffer for direct polling.
pub fn ring(&self) -> &Arc<AudioRing> {
&self.ring
}
/// Stop capturing.
@@ -152,26 +148,34 @@ impl AudioCapture {
}
}
impl Drop for AudioCapture {
fn drop(&mut self) {
self.stop();
}
}
// ---------------------------------------------------------------------------
// AudioPlayback
// ---------------------------------------------------------------------------
/// Plays PCM frames through the default output device at 48 kHz mono.
/// Plays PCM through the default output device, reading from a lock-free ring buffer.
///
/// The cpal stream lives on a dedicated OS thread; this handle is `Send + Sync`.
pub struct AudioPlayback {
tx: mpsc::SyncSender<Vec<i16>>,
ring: Arc<AudioRing>,
running: Arc<AtomicBool>,
}
impl AudioPlayback {
/// Create and start playback on the default output device at 48 kHz mono.
pub fn start() -> Result<Self, anyhow::Error> {
let (tx, rx) = mpsc::sync_channel::<Vec<i16>>(64);
let ring = Arc::new(AudioRing::new());
let running = Arc::new(AtomicBool::new(true));
let running_clone = running.clone();
let (init_tx, init_rx) = mpsc::sync_channel::<Result<(), String>>(1);
let (init_tx, init_rx) = std::sync::mpsc::sync_channel::<Result<(), String>>(1);
let ring_cb = ring.clone();
let running_clone = running.clone();
std::thread::Builder::new()
.name("wzp-audio-playback".into())
@@ -192,62 +196,40 @@ impl AudioPlayback {
let use_f32 = !supports_i16_output(&device)?;
// Shared ring of samples the cpal callback drains from.
let ring = Arc::new(std::sync::Mutex::new(
std::collections::VecDeque::<i16>::with_capacity(FRAME_SAMPLES * 8),
));
// Background drainer: moves frames from the mpsc channel into the ring.
{
let ring = ring.clone();
let running = running_clone.clone();
std::thread::Builder::new()
.name("wzp-playback-drain".into())
.spawn(move || {
while running.load(Ordering::Relaxed) {
match rx.recv_timeout(std::time::Duration::from_millis(100)) {
Ok(frame) => {
let mut lock = ring.lock().unwrap();
lock.extend(frame);
while lock.len() > FRAME_SAMPLES * 16 {
lock.pop_front();
}
}
Err(mpsc::RecvTimeoutError::Timeout) => {}
Err(mpsc::RecvTimeoutError::Disconnected) => break,
}
}
})?;
}
let err_cb = |e: cpal::StreamError| {
warn!("output stream error: {e}");
};
let stream = if use_f32 {
let ring = ring.clone();
let ring = ring_cb.clone();
device.build_output_stream(
&config,
move |data: &mut [f32], _: &cpal::OutputCallbackInfo| {
let mut lock = ring.lock().unwrap();
for sample in data.iter_mut() {
*sample = match lock.pop_front() {
Some(s) => i16_to_f32(s),
None => 0.0,
};
let mut tmp = [0i16; FRAME_SAMPLES];
for chunk in data.chunks_mut(FRAME_SAMPLES) {
let n = chunk.len();
let read = ring.read(&mut tmp[..n]);
for i in 0..read {
chunk[i] = i16_to_f32(tmp[i]);
}
// Fill remainder with silence if ring underran
for i in read..n {
chunk[i] = 0.0;
}
}
},
err_cb,
None,
)?
} else {
let ring = ring.clone();
let ring = ring_cb.clone();
device.build_output_stream(
&config,
move |data: &mut [i16], _: &cpal::OutputCallbackInfo| {
let mut lock = ring.lock().unwrap();
for sample in data.iter_mut() {
*sample = lock.pop_front().unwrap_or(0);
let read = ring.read(data);
// Fill remainder with silence if ring underran
for sample in &mut data[read..] {
*sample = 0;
}
},
err_cb,
@@ -257,7 +239,6 @@ impl AudioPlayback {
stream.play().context("failed to start output stream")?;
// Signal success to the caller before parking.
let _ = init_tx.send(Ok(()));
// Keep stream alive until stopped.
@@ -278,12 +259,12 @@ impl AudioPlayback {
.map_err(|_| anyhow!("playback thread exited before signaling"))?
.map_err(|e| anyhow!("{e}"))?;
Ok(Self { tx, running })
Ok(Self { ring, running })
}
/// Write a frame of PCM samples for playback.
pub fn write_frame(&self, pcm: &[i16]) {
let _ = self.tx.try_send(pcm.to_vec());
/// Get a reference to the playout ring buffer for direct writing.
pub fn ring(&self) -> &Arc<AudioRing> {
&self.ring
}
/// Stop playback.
@@ -292,11 +273,16 @@ impl AudioPlayback {
}
}
impl Drop for AudioPlayback {
fn drop(&mut self) {
self.stop();
}
}
// ---------------------------------------------------------------------------
// Helpers
// ---------------------------------------------------------------------------
/// Check if the input device supports i16 at 48 kHz mono.
fn supports_i16_input(device: &cpal::Device) -> Result<bool, anyhow::Error> {
let supported = device
.supported_input_configs()
@@ -313,7 +299,6 @@ fn supports_i16_input(device: &cpal::Device) -> Result<bool, anyhow::Error> {
Ok(false)
}
/// Check if the output device supports i16 at 48 kHz mono.
fn supports_i16_output(device: &cpal::Device) -> Result<bool, anyhow::Error> {
let supported = device
.supported_output_configs()

View File

@@ -0,0 +1,537 @@
//! Linux AEC backend: CPAL capture + playback wired through the WebRTC Audio
//! Processing Module (AEC3 + noise suppression + high-pass filter).
//!
//! This is the same algorithm used by Chrome WebRTC, Zoom, Teams, Jitsi, and
//! any other "serious" Linux VoIP app. It runs in-process — no dependency on
//! PulseAudio's module-echo-cancel or PipeWire's filter-chain, so it works
//! identically on ALSA / PulseAudio / PipeWire systems.
//!
//! ## Architecture
//!
//! A single module-level `Arc<Mutex<Processor>>` is shared between the
//! capture and playback paths. On each 20 ms frame (960 samples @ 48 kHz
//! mono):
//!
//! - **Playback path**: `LinuxAecPlayback::start` spawns the usual CPAL
//! output thread, but wraps each chunk in a call to
//! `Processor::process_render_frame` **before** handing it to CPAL. That
//! gives APM an authoritative reference of exactly what's going out to
//! the speakers (same approach Zoom/Teams/Jitsi use). The AEC then knows
//! what to cancel when it sees echo in the capture stream.
//!
//! - **Capture path**: `LinuxAecCapture::start` spawns the usual CPAL
//! input thread, and runs `Processor::process_capture_frame` on each
//! incoming mic chunk **in place** before pushing it into the ring
//! buffer. The AEC subtracts the echo using the render reference it
//! saw on the playback side.
//!
//! APM is strict about frame size: it requires exactly 10 ms = 480 samples
//! per call at 48 kHz. Our pipeline uses 20 ms = 960 samples, so each 20 ms
//! frame is split into two 480-sample halves, APM is called twice, and the
//! halves are stitched back together.
//!
//! APM only accepts f32 samples in `[-1.0, 1.0]`, so we convert i16 → f32
//! before the call and f32 → i16 after (with clamping on the return path).
//!
//! ## Stream delay
//!
//! AEC needs to know roughly how long it takes between a sample being passed
//! to `process_render_frame` and its echo showing up at `process_capture_frame`
//! — i.e. the round trip through CPAL playback → speaker → air → microphone
//! → CPAL capture. AEC3's internal estimator tracks this within a window
//! around whatever hint we give it. We hardcode 60 ms as a reasonable
//! starting point for typical Linux audio stacks; the delay estimator does
//! the fine-tuning automatically.
//!
//! ## Thread safety
//!
//! The 0.3.x line of `webrtc-audio-processing` takes `&mut self` on both
//! `process_capture_frame` and `process_render_frame`, so the `Processor`
//! needs a `Mutex` around it for cross-thread sharing. The capture and
//! playback threads each acquire the lock briefly (sub-millisecond per
//! 10 ms frame) so contention is minimal at our frame rates.
use std::sync::atomic::{AtomicBool, Ordering};
use std::sync::{Arc, Mutex, OnceLock};
use anyhow::{anyhow, Context};
use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
use cpal::{SampleFormat, SampleRate, StreamConfig};
use tracing::{info, warn};
use webrtc_audio_processing::{
Config, EchoCancellation, EchoCancellationSuppressionLevel, InitializationConfig,
NoiseSuppression, NoiseSuppressionLevel, Processor, NUM_SAMPLES_PER_FRAME,
};
use crate::audio_ring::AudioRing;
/// 20 ms at 48 kHz, mono — matches the rest of the pipeline and the codec.
pub const FRAME_SAMPLES: usize = 960;
/// APM requires strict 10 ms frames at 48 kHz = 480 samples per call.
/// Imported from the webrtc-audio-processing crate so we can't drift out
/// of sync with whatever sample rate / frame length the C++ lib is using.
const APM_FRAME_SAMPLES: usize = NUM_SAMPLES_PER_FRAME as usize;
const APM_NUM_CHANNELS: usize = 1;
/// Round-trip delay hint passed to APM; the estimator refines from here.
/// 60 ms is a reasonable default for CPAL on ALSA / PulseAudio / PipeWire.
#[allow(dead_code)]
const STREAM_DELAY_MS: i32 = 60;
// ---------------------------------------------------------------------------
// Shared APM instance
// ---------------------------------------------------------------------------
/// Module-level lazily-initialized APM. Shared between capture and playback
/// so they operate on the same echo-cancellation state — the render frames
/// pushed by playback are what the capture path subtracts from the mic input.
/// Wrapped in a Mutex because the 0.3.x Processor takes `&mut self` on both
/// process_capture_frame and process_render_frame.
static PROCESSOR: OnceLock<Arc<Mutex<Processor>>> = OnceLock::new();
fn get_or_init_processor() -> anyhow::Result<Arc<Mutex<Processor>>> {
if let Some(p) = PROCESSOR.get() {
return Ok(p.clone());
}
let init_config = InitializationConfig {
num_capture_channels: APM_NUM_CHANNELS as i32,
num_render_channels: APM_NUM_CHANNELS as i32,
..Default::default()
};
let mut processor = Processor::new(&init_config)
.map_err(|e| anyhow!("webrtc APM init failed: {e:?}"))?;
let config = Config {
echo_cancellation: Some(EchoCancellation {
suppression_level: EchoCancellationSuppressionLevel::High,
stream_delay_ms: Some(STREAM_DELAY_MS),
enable_delay_agnostic: true,
enable_extended_filter: true,
}),
noise_suppression: Some(NoiseSuppression {
suppression_level: NoiseSuppressionLevel::High,
}),
enable_high_pass_filter: true,
// AGC left off for now — it can fight the Opus encoder's own gain
// staging and the adaptive-quality controller. Add later if users
// report low mic levels.
..Default::default()
};
processor.set_config(config);
let arc = Arc::new(Mutex::new(processor));
let _ = PROCESSOR.set(arc.clone());
info!(
stream_delay_ms = STREAM_DELAY_MS,
"webrtc APM initialized (AEC High + NS High + HPF, AGC off)"
);
Ok(arc)
}
// ---------------------------------------------------------------------------
// Helpers: i16 ↔ f32 and APM frame processing
// ---------------------------------------------------------------------------
#[inline]
fn i16_to_f32(s: i16) -> f32 {
s as f32 / 32768.0
}
#[inline]
fn f32_to_i16(s: f32) -> i16 {
(s.clamp(-1.0, 1.0) * 32767.0) as i16
}
/// Feed a 20 ms (960-sample) playback frame to APM as the render reference.
/// Splits into two 10 ms halves because APM is strict about frame size.
/// Takes the Mutex-wrapped Processor and locks briefly around each call.
fn push_render_frame_20ms(apm: &Mutex<Processor>, pcm: &[i16]) {
debug_assert_eq!(pcm.len(), FRAME_SAMPLES);
let mut buf = [0f32; APM_FRAME_SAMPLES];
for half in pcm.chunks_exact(APM_FRAME_SAMPLES) {
for (i, &s) in half.iter().enumerate() {
buf[i] = i16_to_f32(s);
}
match apm.lock() {
Ok(mut p) => {
if let Err(e) = p.process_render_frame(&mut buf) {
warn!("webrtc APM process_render_frame failed: {e:?}");
}
}
Err(_) => {
warn!("webrtc APM mutex poisoned in render path");
return;
}
}
}
}
/// Run a 20 ms (960-sample) capture frame through APM's echo cancellation
/// in place. Splits into two 10 ms halves, runs APM on each, stitches
/// results back into the caller's buffer. Briefly holds the Mutex once
/// per 10 ms half.
fn process_capture_frame_20ms(apm: &Mutex<Processor>, pcm: &mut [i16]) {
debug_assert_eq!(pcm.len(), FRAME_SAMPLES);
let mut buf = [0f32; APM_FRAME_SAMPLES];
for half in pcm.chunks_exact_mut(APM_FRAME_SAMPLES) {
for (i, &s) in half.iter().enumerate() {
buf[i] = i16_to_f32(s);
}
match apm.lock() {
Ok(mut p) => {
if let Err(e) = p.process_capture_frame(&mut buf) {
warn!("webrtc APM process_capture_frame failed: {e:?}");
}
}
Err(_) => {
warn!("webrtc APM mutex poisoned in capture path");
return;
}
}
for (i, d) in half.iter_mut().enumerate() {
*d = f32_to_i16(buf[i]);
}
}
}
// ---------------------------------------------------------------------------
// LinuxAecCapture — CPAL mic + WebRTC AEC capture-side processing
// ---------------------------------------------------------------------------
/// Microphone capture with WebRTC AEC3 applied in place before the codec
/// sees the samples. Mirrors the public API of `audio_io::AudioCapture` so
/// downstream code doesn't change.
pub struct LinuxAecCapture {
ring: Arc<AudioRing>,
running: Arc<AtomicBool>,
}
impl LinuxAecCapture {
pub fn start() -> Result<Self, anyhow::Error> {
// Eagerly init the APM so the playback side can find it already
// configured, and so init errors surface on the caller thread
// instead of silently failing inside the capture thread.
let apm = get_or_init_processor()?;
let ring = Arc::new(AudioRing::new());
let running = Arc::new(AtomicBool::new(true));
let (init_tx, init_rx) = std::sync::mpsc::sync_channel::<Result<(), String>>(1);
let ring_cb = ring.clone();
let running_clone = running.clone();
let apm_capture = apm.clone();
std::thread::Builder::new()
.name("wzp-audio-capture-linuxaec".into())
.spawn(move || {
let result = (|| -> Result<(), anyhow::Error> {
let host = cpal::default_host();
let device = host
.default_input_device()
.ok_or_else(|| anyhow!("no default input audio device found"))?;
info!(device = %device.name().unwrap_or_default(), "LinuxAEC: using input device");
let config = StreamConfig {
channels: 1,
sample_rate: SampleRate(48_000),
buffer_size: cpal::BufferSize::Default,
};
let use_f32 = !supports_i16_input(&device)?;
let err_cb = |e: cpal::StreamError| {
warn!("LinuxAEC input stream error: {e}");
};
// Leftover buffer for when CPAL gives us partial frames.
// We need exactly 960-sample chunks to feed APM.
let leftover = std::sync::Mutex::new(Vec::<i16>::with_capacity(FRAME_SAMPLES * 4));
let stream = if use_f32 {
let ring = ring_cb.clone();
let running = running_clone.clone();
let apm = apm_capture.clone();
device.build_input_stream(
&config,
move |data: &[f32], _: &cpal::InputCallbackInfo| {
if !running.load(Ordering::Relaxed) {
return;
}
let mut lv = leftover.lock().unwrap();
lv.reserve(data.len());
for &s in data {
lv.push(f32_to_i16(s));
}
drain_frames_through_apm(&mut lv, &apm, &ring);
},
err_cb,
None,
)?
} else {
let ring = ring_cb.clone();
let running = running_clone.clone();
let apm = apm_capture.clone();
device.build_input_stream(
&config,
move |data: &[i16], _: &cpal::InputCallbackInfo| {
if !running.load(Ordering::Relaxed) {
return;
}
let mut lv = leftover.lock().unwrap();
lv.extend_from_slice(data);
drain_frames_through_apm(&mut lv, &apm, &ring);
},
err_cb,
None,
)?
};
stream.play().context("failed to start LinuxAEC input stream")?;
let _ = init_tx.send(Ok(()));
info!("LinuxAEC capture started (AEC3 active)");
while running_clone.load(Ordering::Relaxed) {
std::thread::park_timeout(std::time::Duration::from_millis(200));
}
drop(stream);
Ok(())
})();
if let Err(e) = result {
let _ = init_tx.send(Err(e.to_string()));
}
})?;
init_rx
.recv()
.map_err(|_| anyhow!("LinuxAEC capture thread exited before signaling"))?
.map_err(|e| anyhow!("{e}"))?;
Ok(Self { ring, running })
}
pub fn ring(&self) -> &Arc<AudioRing> {
&self.ring
}
pub fn stop(&self) {
self.running.store(false, Ordering::Relaxed);
}
}
impl Drop for LinuxAecCapture {
fn drop(&mut self) {
self.stop();
}
}
/// Pull whole 960-sample frames out of the leftover buffer, run them through
/// APM's capture-side processing, and push to the ring. Leaves any partial
/// sub-960 remainder in `leftover` for the next callback.
fn drain_frames_through_apm(leftover: &mut Vec<i16>, apm: &Mutex<Processor>, ring: &AudioRing) {
let mut frame = [0i16; FRAME_SAMPLES];
while leftover.len() >= FRAME_SAMPLES {
frame.copy_from_slice(&leftover[..FRAME_SAMPLES]);
process_capture_frame_20ms(apm, &mut frame);
ring.write(&frame);
leftover.drain(..FRAME_SAMPLES);
}
}
// ---------------------------------------------------------------------------
// LinuxAecPlayback — CPAL speaker output + WebRTC AEC render-side tee
// ---------------------------------------------------------------------------
/// Speaker playback with a render-side tee: each frame written to CPAL is
/// ALSO fed to APM via `process_render_frame` as the echo-cancellation
/// reference signal. This is the "tee the playback ring" approach (Zoom,
/// Teams, Jitsi) — deterministic, does not depend on PulseAudio loopback or
/// PipeWire monitor sources.
pub struct LinuxAecPlayback {
ring: Arc<AudioRing>,
running: Arc<AtomicBool>,
}
impl LinuxAecPlayback {
pub fn start() -> Result<Self, anyhow::Error> {
let apm = get_or_init_processor()?;
let ring = Arc::new(AudioRing::new());
let running = Arc::new(AtomicBool::new(true));
let (init_tx, init_rx) = std::sync::mpsc::sync_channel::<Result<(), String>>(1);
let ring_cb = ring.clone();
let running_clone = running.clone();
let apm_render = apm.clone();
std::thread::Builder::new()
.name("wzp-audio-playback-linuxaec".into())
.spawn(move || {
let result = (|| -> Result<(), anyhow::Error> {
let host = cpal::default_host();
let device = host
.default_output_device()
.ok_or_else(|| anyhow!("no default output audio device found"))?;
info!(device = %device.name().unwrap_or_default(), "LinuxAEC: using output device");
let config = StreamConfig {
channels: 1,
sample_rate: SampleRate(48_000),
buffer_size: cpal::BufferSize::Default,
};
let use_f32 = !supports_i16_output(&device)?;
let err_cb = |e: cpal::StreamError| {
warn!("LinuxAEC output stream error: {e}");
};
// Same 960-sample batching approach as the capture side:
// CPAL may ask for N samples in a callback where N doesn't
// divide 960. We accumulate partial frames in a Vec and
// feed APM as soon as we have a whole 20 ms frame.
let carry = std::sync::Mutex::new(Vec::<i16>::with_capacity(FRAME_SAMPLES * 4));
let stream = if use_f32 {
let ring = ring_cb.clone();
let apm = apm_render.clone();
device.build_output_stream(
&config,
move |data: &mut [f32], _: &cpal::OutputCallbackInfo| {
fill_output_and_tee_f32(data, &ring, &apm, &carry);
},
err_cb,
None,
)?
} else {
let ring = ring_cb.clone();
let apm = apm_render.clone();
device.build_output_stream(
&config,
move |data: &mut [i16], _: &cpal::OutputCallbackInfo| {
fill_output_and_tee_i16(data, &ring, &apm, &carry);
},
err_cb,
None,
)?
};
stream.play().context("failed to start LinuxAEC output stream")?;
let _ = init_tx.send(Ok(()));
info!("LinuxAEC playback started (render tee active)");
while running_clone.load(Ordering::Relaxed) {
std::thread::park_timeout(std::time::Duration::from_millis(200));
}
drop(stream);
Ok(())
})();
if let Err(e) = result {
let _ = init_tx.send(Err(e.to_string()));
}
})?;
init_rx
.recv()
.map_err(|_| anyhow!("LinuxAEC playback thread exited before signaling"))?
.map_err(|e| anyhow!("{e}"))?;
Ok(Self { ring, running })
}
pub fn ring(&self) -> &Arc<AudioRing> {
&self.ring
}
pub fn stop(&self) {
self.running.store(false, Ordering::Relaxed);
}
}
impl Drop for LinuxAecPlayback {
fn drop(&mut self) {
self.stop();
}
}
fn fill_output_and_tee_i16(
data: &mut [i16],
ring: &AudioRing,
apm: &Mutex<Processor>,
carry: &std::sync::Mutex<Vec<i16>>,
) {
let read = ring.read(data);
for s in &mut data[read..] {
*s = 0;
}
tee_render_samples(data, apm, carry);
}
fn fill_output_and_tee_f32(
data: &mut [f32],
ring: &AudioRing,
apm: &Mutex<Processor>,
carry: &std::sync::Mutex<Vec<i16>>,
) {
let mut tmp = vec![0i16; data.len()];
let read = ring.read(&mut tmp);
for s in &mut tmp[read..] {
*s = 0;
}
for (d, &s) in data.iter_mut().zip(tmp.iter()) {
*d = i16_to_f32(s);
}
tee_render_samples(&tmp, apm, carry);
}
/// Push CPAL-bound samples into APM's render-side input for echo cancellation.
/// Uses a carry buffer to batch into exact 960-sample (20 ms) frames.
fn tee_render_samples(samples: &[i16], apm: &Mutex<Processor>, carry: &std::sync::Mutex<Vec<i16>>) {
let mut lv = carry.lock().unwrap();
lv.extend_from_slice(samples);
while lv.len() >= FRAME_SAMPLES {
let mut frame = [0i16; FRAME_SAMPLES];
frame.copy_from_slice(&lv[..FRAME_SAMPLES]);
push_render_frame_20ms(apm, &frame);
lv.drain(..FRAME_SAMPLES);
}
}
// ---------------------------------------------------------------------------
// CPAL format helpers (duplicated from audio_io.rs to keep the modules
// independent — each backend file is a self-contained unit)
// ---------------------------------------------------------------------------
fn supports_i16_input(device: &cpal::Device) -> Result<bool, anyhow::Error> {
let supported = device
.supported_input_configs()
.context("failed to query input configs")?;
for cfg in supported {
if cfg.sample_format() == SampleFormat::I16
&& cfg.min_sample_rate() <= SampleRate(48_000)
&& cfg.max_sample_rate() >= SampleRate(48_000)
&& cfg.channels() >= 1
{
return Ok(true);
}
}
Ok(false)
}
fn supports_i16_output(device: &cpal::Device) -> Result<bool, anyhow::Error> {
let supported = device
.supported_output_configs()
.context("failed to query output configs")?;
for cfg in supported {
if cfg.sample_format() == SampleFormat::I16
&& cfg.min_sample_rate() <= SampleRate(48_000)
&& cfg.max_sample_rate() >= SampleRate(48_000)
&& cfg.channels() >= 1
{
return Ok(true);
}
}
Ok(false)
}

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@@ -0,0 +1,122 @@
//! Lock-free SPSC ring buffer — "Reader-Detects-Lap" architecture.
//!
//! SPSC invariant: the producer ONLY writes `write_pos`, the consumer
//! ONLY writes `read_pos`. Neither thread touches the other's cursor.
//!
//! On overflow (writer laps the reader), the writer simply overwrites
//! old buffer data. The reader detects the lap via `available() >
//! RING_CAPACITY` and snaps its own `read_pos` forward.
//!
//! Capacity is a power of 2 for bitmask indexing (no modulo).
use std::sync::atomic::{AtomicU64, AtomicUsize, Ordering};
/// Ring buffer capacity — power of 2 for bitmask indexing.
/// 16384 samples = 341.3ms at 48kHz mono.
const RING_CAPACITY: usize = 16384; // 2^14
const RING_MASK: usize = RING_CAPACITY - 1;
/// Lock-free single-producer single-consumer ring buffer for i16 PCM samples.
pub struct AudioRing {
buf: Box<[i16]>,
/// Monotonically increasing write cursor. ONLY written by producer.
write_pos: AtomicUsize,
/// Monotonically increasing read cursor. ONLY written by consumer.
read_pos: AtomicUsize,
/// Incremented by reader when it detects it was lapped (overflow).
overflow_count: AtomicU64,
/// Incremented by reader when ring is empty (underrun).
underrun_count: AtomicU64,
}
// SAFETY: AudioRing is SPSC — one thread writes (producer), one reads (consumer).
// The producer only writes write_pos. The consumer only writes read_pos.
// Neither thread writes the other's cursor. Buffer indices are derived from
// the owning thread's cursor, ensuring no concurrent access to the same index.
unsafe impl Send for AudioRing {}
unsafe impl Sync for AudioRing {}
impl AudioRing {
pub fn new() -> Self {
debug_assert!(RING_CAPACITY.is_power_of_two());
Self {
buf: vec![0i16; RING_CAPACITY].into_boxed_slice(),
write_pos: AtomicUsize::new(0),
read_pos: AtomicUsize::new(0),
overflow_count: AtomicU64::new(0),
underrun_count: AtomicU64::new(0),
}
}
/// Number of samples available to read (clamped to capacity).
pub fn available(&self) -> usize {
let w = self.write_pos.load(Ordering::Acquire);
let r = self.read_pos.load(Ordering::Relaxed);
w.wrapping_sub(r).min(RING_CAPACITY)
}
/// Write samples into the ring. Returns number of samples written.
///
/// If the ring is full, old data is silently overwritten. The reader
/// will detect the lap and self-correct. The writer NEVER touches
/// `read_pos`.
pub fn write(&self, samples: &[i16]) -> usize {
let count = samples.len().min(RING_CAPACITY);
let w = self.write_pos.load(Ordering::Relaxed);
for i in 0..count {
unsafe {
let ptr = self.buf.as_ptr() as *mut i16;
*ptr.add((w + i) & RING_MASK) = samples[i];
}
}
self.write_pos
.store(w.wrapping_add(count), Ordering::Release);
count
}
/// Read samples from the ring into `out`. Returns number of samples read.
///
/// If the writer has lapped the reader (overflow), `read_pos` is snapped
/// forward to the oldest valid data.
pub fn read(&self, out: &mut [i16]) -> usize {
let w = self.write_pos.load(Ordering::Acquire);
let mut r = self.read_pos.load(Ordering::Relaxed);
let mut avail = w.wrapping_sub(r);
// Lap detection: writer has overwritten our unread data.
if avail > RING_CAPACITY {
r = w.wrapping_sub(RING_CAPACITY);
avail = RING_CAPACITY;
self.overflow_count.fetch_add(1, Ordering::Relaxed);
}
let count = out.len().min(avail);
if count == 0 {
if w == r {
self.underrun_count.fetch_add(1, Ordering::Relaxed);
}
return 0;
}
for i in 0..count {
out[i] = unsafe { *self.buf.as_ptr().add((r + i) & RING_MASK) };
}
self.read_pos
.store(r.wrapping_add(count), Ordering::Release);
count
}
/// Number of overflow events (reader was lapped by writer).
pub fn overflow_count(&self) -> u64 {
self.overflow_count.load(Ordering::Relaxed)
}
/// Number of underrun events (reader found empty buffer).
pub fn underrun_count(&self) -> u64 {
self.underrun_count.load(Ordering::Relaxed)
}
}

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@@ -0,0 +1,179 @@
//! macOS Voice Processing I/O — uses Apple's VoiceProcessingIO audio unit
//! for hardware-accelerated echo cancellation, AGC, and noise suppression.
//!
//! VoiceProcessingIO is a combined input+output unit that knows what's going
//! to the speaker, so it can cancel the echo from the mic signal internally.
//! This is the same engine FaceTime and other Apple apps use.
use std::sync::atomic::{AtomicBool, Ordering};
use std::sync::Arc;
use anyhow::Context;
use coreaudio::audio_unit::audio_format::LinearPcmFlags;
use coreaudio::audio_unit::render_callback::{self, data};
use coreaudio::audio_unit::{AudioUnit, Element, IOType, SampleFormat, Scope, StreamFormat};
use coreaudio::sys;
use tracing::info;
use crate::audio_ring::AudioRing;
/// Number of samples per 20 ms frame at 48 kHz mono.
pub const FRAME_SAMPLES: usize = 960;
/// Combined capture + playback via macOS VoiceProcessingIO.
///
/// The OS handles AEC internally — no manual far-end feeding needed.
pub struct VpioAudio {
capture_ring: Arc<AudioRing>,
playout_ring: Arc<AudioRing>,
_audio_unit: AudioUnit,
running: Arc<AtomicBool>,
}
impl VpioAudio {
/// Start VoiceProcessingIO with AEC enabled.
pub fn start() -> Result<Self, anyhow::Error> {
let capture_ring = Arc::new(AudioRing::new());
let playout_ring = Arc::new(AudioRing::new());
let running = Arc::new(AtomicBool::new(true));
let mut au = AudioUnit::new(IOType::VoiceProcessingIO)
.context("failed to create VoiceProcessingIO audio unit")?;
// Must uninitialize before configuring properties.
au.uninitialize()
.context("failed to uninitialize VPIO for configuration")?;
// Enable input (mic) on Element::Input (bus 1).
let enable: u32 = 1;
au.set_property(
sys::kAudioOutputUnitProperty_EnableIO,
Scope::Input,
Element::Input,
Some(&enable),
)
.context("failed to enable VPIO input")?;
// Output (speaker) is enabled by default on VPIO, but be explicit.
au.set_property(
sys::kAudioOutputUnitProperty_EnableIO,
Scope::Output,
Element::Output,
Some(&enable),
)
.context("failed to enable VPIO output")?;
// Configure stream format: 48kHz mono f32 non-interleaved
let stream_format = StreamFormat {
sample_rate: 48_000.0,
sample_format: SampleFormat::F32,
flags: LinearPcmFlags::IS_FLOAT
| LinearPcmFlags::IS_PACKED
| LinearPcmFlags::IS_NON_INTERLEAVED,
channels: 1,
};
let asbd = stream_format.to_asbd();
// Input: set format on Output scope of Input element
// (= the format the AU delivers to us from the mic)
au.set_property(
sys::kAudioUnitProperty_StreamFormat,
Scope::Output,
Element::Input,
Some(&asbd),
)
.context("failed to set input stream format")?;
// Output: set format on Input scope of Output element
// (= the format we feed to the AU for the speaker)
au.set_property(
sys::kAudioUnitProperty_StreamFormat,
Scope::Input,
Element::Output,
Some(&asbd),
)
.context("failed to set output stream format")?;
// Set up input callback (mic capture with AEC applied)
let cap_ring = capture_ring.clone();
let cap_running = running.clone();
let logged = Arc::new(AtomicBool::new(false));
au.set_input_callback(
move |args: render_callback::Args<data::NonInterleaved<f32>>| {
if !cap_running.load(Ordering::Relaxed) {
return Ok(());
}
let mut buffers = args.data.channels();
if let Some(ch) = buffers.next() {
if !logged.swap(true, Ordering::Relaxed) {
eprintln!("[vpio] capture callback: {} f32 samples", ch.len());
}
let mut tmp = [0i16; FRAME_SAMPLES];
for chunk in ch.chunks(FRAME_SAMPLES) {
let n = chunk.len();
for i in 0..n {
tmp[i] = (chunk[i].clamp(-1.0, 1.0) * i16::MAX as f32) as i16;
}
cap_ring.write(&tmp[..n]);
}
}
Ok(())
},
)
.context("failed to set input callback")?;
// Set up output callback (speaker playback — AEC uses this as reference)
let play_ring = playout_ring.clone();
au.set_render_callback(
move |mut args: render_callback::Args<data::NonInterleaved<f32>>| {
let mut buffers = args.data.channels_mut();
if let Some(ch) = buffers.next() {
let mut tmp = [0i16; FRAME_SAMPLES];
for chunk in ch.chunks_mut(FRAME_SAMPLES) {
let n = chunk.len();
let read = play_ring.read(&mut tmp[..n]);
for i in 0..read {
chunk[i] = tmp[i] as f32 / i16::MAX as f32;
}
for i in read..n {
chunk[i] = 0.0;
}
}
}
Ok(())
},
)
.context("failed to set render callback")?;
au.initialize().context("failed to initialize VoiceProcessingIO")?;
au.start().context("failed to start VoiceProcessingIO")?;
info!("VoiceProcessingIO started (OS-level AEC enabled)");
Ok(Self {
capture_ring,
playout_ring,
_audio_unit: au,
running,
})
}
pub fn capture_ring(&self) -> &Arc<AudioRing> {
&self.capture_ring
}
pub fn playout_ring(&self) -> &Arc<AudioRing> {
&self.playout_ring
}
pub fn stop(&self) {
self.running.store(false, Ordering::Relaxed);
}
}
impl Drop for VpioAudio {
fn drop(&mut self) {
self.stop();
}
}

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@@ -0,0 +1,332 @@
//! Direct WASAPI microphone capture with Windows's OS-level AEC enabled.
//!
//! Bypasses CPAL and opens the default capture endpoint directly via
//! `IMMDeviceEnumerator` + `IAudioClient2::SetClientProperties`, setting
//! `AudioClientProperties.eCategory = AudioCategory_Communications`. That's
//! the switch that tells Windows "this is a VoIP call" — the OS then
//! enables its communications audio processing chain (AEC, noise
//! suppression, automatic gain control) for the stream. AEC operates at
//! the OS level using the currently-playing audio as the reference
//! signal, so it cancels echo from our CPAL playback (and any other app's
//! audio) without us having to plumb a reference signal ourselves.
//!
//! Platform: Windows only, compiled only when the `windows-aec` feature
//! is enabled. Mirrors the public API of `audio_io::AudioCapture` so
//! `wzp-client`'s lib.rs can transparently re-export either one as
//! `AudioCapture`.
use std::sync::atomic::{AtomicBool, Ordering};
use std::sync::Arc;
use anyhow::{anyhow, Context};
use tracing::{info, warn};
use windows::core::{Interface, GUID};
use windows::Win32::Foundation::{CloseHandle, BOOL, WAIT_OBJECT_0};
use windows::Win32::Media::Audio::{
eCapture, eCommunications, AudioCategory_Communications, AudioClientProperties,
IAudioCaptureClient, IAudioClient, IAudioClient2, IMMDeviceEnumerator, MMDeviceEnumerator,
AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK, AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY, WAVEFORMATEX,
WAVE_FORMAT_PCM,
};
use windows::Win32::System::Com::{
CoCreateInstance, CoInitializeEx, CoUninitialize, CLSCTX_ALL, COINIT_MULTITHREADED,
};
use windows::Win32::System::Threading::{CreateEventW, WaitForSingleObject, INFINITE};
use crate::audio_ring::AudioRing;
/// 20 ms at 48 kHz, mono. Matches the rest of the audio pipeline.
pub const FRAME_SAMPLES: usize = 960;
/// Microphone capture via WASAPI with Windows's communications AEC enabled.
///
/// The WASAPI capture stream runs on a dedicated OS thread. This handle is
/// `Send + Sync`. Dropping it stops the stream and joins the thread.
pub struct WasapiAudioCapture {
ring: Arc<AudioRing>,
running: Arc<AtomicBool>,
thread: Option<std::thread::JoinHandle<()>>,
}
impl WasapiAudioCapture {
/// Open the default communications microphone, enable OS AEC, and start
/// streaming PCM into a lock-free ring buffer.
///
/// Returns only after the capture thread has successfully initialized
/// the stream, or propagates the error back to the caller.
pub fn start() -> Result<Self, anyhow::Error> {
let ring = Arc::new(AudioRing::new());
let running = Arc::new(AtomicBool::new(true));
let (init_tx, init_rx) = std::sync::mpsc::sync_channel::<Result<(), String>>(1);
let ring_cb = ring.clone();
let running_cb = running.clone();
let thread = std::thread::Builder::new()
.name("wzp-audio-capture-wasapi".into())
.spawn(move || {
let result = unsafe { capture_thread_main(ring_cb, running_cb.clone(), &init_tx) };
if let Err(e) = result {
warn!("wasapi capture thread exited with error: {e}");
// If we failed before signaling init, signal now so the
// caller unblocks. Double-send is harmless (channel is
// bounded to 1 and we only hit the second send path on
// late errors).
let _ = init_tx.send(Err(e.to_string()));
}
})
.context("failed to spawn WASAPI capture thread")?;
init_rx
.recv()
.map_err(|_| anyhow!("WASAPI capture thread exited before signaling init"))?
.map_err(|e| anyhow!("{e}"))?;
Ok(Self {
ring,
running,
thread: Some(thread),
})
}
/// Get a reference to the capture ring buffer for direct polling.
pub fn ring(&self) -> &Arc<AudioRing> {
&self.ring
}
/// Stop capturing.
pub fn stop(&self) {
self.running.store(false, Ordering::Relaxed);
}
}
impl Drop for WasapiAudioCapture {
fn drop(&mut self) {
self.stop();
if let Some(handle) = self.thread.take() {
// Join best-effort. The thread loop polls `running` every 200ms
// via a short WaitForSingleObject timeout, so it should exit
// within ~200ms of `stop()`.
let _ = handle.join();
}
}
}
// ---------------------------------------------------------------------------
// WASAPI thread entry point — everything below this line runs on the
// dedicated wzp-audio-capture-wasapi thread.
// ---------------------------------------------------------------------------
unsafe fn capture_thread_main(
ring: Arc<AudioRing>,
running: Arc<AtomicBool>,
init_tx: &std::sync::mpsc::SyncSender<Result<(), String>>,
) -> Result<(), anyhow::Error> {
// COM init for the capture thread. MULTITHREADED because we're not
// running a message pump. Must be balanced by CoUninitialize on exit.
CoInitializeEx(None, COINIT_MULTITHREADED)
.ok()
.context("CoInitializeEx failed")?;
// Use a guard struct so CoUninitialize runs even on early returns.
struct ComGuard;
impl Drop for ComGuard {
fn drop(&mut self) {
unsafe { CoUninitialize() };
}
}
let _com_guard = ComGuard;
let enumerator: IMMDeviceEnumerator =
CoCreateInstance(&MMDeviceEnumerator, None, CLSCTX_ALL)
.context("CoCreateInstance(MMDeviceEnumerator) failed")?;
// eCommunications role (not eConsole) — this picks the device the user
// has designated for communications in Sound Settings. It's the one
// Windows's AEC is actually tuned for and the one Teams/Zoom use.
let device = enumerator
.GetDefaultAudioEndpoint(eCapture, eCommunications)
.context("GetDefaultAudioEndpoint(eCapture, eCommunications) failed")?;
if let Ok(name) = device_name(&device) {
info!(device = %name, "opening WASAPI communications capture endpoint");
}
let audio_client: IAudioClient = device
.Activate(CLSCTX_ALL, None)
.context("IMMDevice::Activate(IAudioClient) failed")?;
// IAudioClient2 exposes SetClientProperties, which is the ONLY way to
// set AudioCategory_Communications pre-Initialize. Calling it on the
// base IAudioClient would not compile, and setting it after Initialize
// is a no-op.
let audio_client2: IAudioClient2 = audio_client
.cast()
.context("QueryInterface IAudioClient2 failed")?;
let mut props = AudioClientProperties {
cbSize: std::mem::size_of::<AudioClientProperties>() as u32,
bIsOffload: BOOL(0),
eCategory: AudioCategory_Communications,
// 0 = AUDCLNT_STREAMOPTIONS_NONE. The `windows` crate doesn't
// export the enum constant in all versions, so use 0 directly.
Options: Default::default(),
};
audio_client2
.SetClientProperties(&mut props as *mut _)
.context("SetClientProperties(AudioCategory_Communications) failed")?;
// Request 48 kHz mono i16 directly. AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM
// tells Windows to do any needed format conversion inside the audio
// engine rather than rejecting our format. SRC_DEFAULT_QUALITY picks
// the standard Windows resampler quality (fine for voice).
let wave_format = WAVEFORMATEX {
wFormatTag: WAVE_FORMAT_PCM as u16,
nChannels: 1,
nSamplesPerSec: 48_000,
nAvgBytesPerSec: 48_000 * 2, // 1 ch * 2 bytes/sample * 48000 Hz
nBlockAlign: 2, // 1 ch * 2 bytes/sample
wBitsPerSample: 16,
cbSize: 0,
};
// 1,000,000 hns = 100 ms buffer (hns = 100-nanosecond units). Windows
// treats this as the minimum; the engine may give us a larger one.
const BUFFER_DURATION_HNS: i64 = 1_000_000;
audio_client
.Initialize(
AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK
| AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM
| AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY,
BUFFER_DURATION_HNS,
0,
&wave_format,
Some(&GUID::zeroed()),
)
.context("IAudioClient::Initialize failed — Windows rejected communications-mode 48k mono i16")?;
// Event-driven capture: Windows signals this handle each time a new
// audio packet is available. We wait on it from the loop below.
let event = CreateEventW(None, false, false, None)
.context("CreateEventW failed")?;
audio_client
.SetEventHandle(event)
.context("SetEventHandle failed")?;
let capture_client: IAudioCaptureClient = audio_client
.GetService()
.context("IAudioClient::GetService(IAudioCaptureClient) failed")?;
audio_client.Start().context("IAudioClient::Start failed")?;
// Signal to the parent thread that init succeeded before entering the
// hot loop. From this point on, errors get logged but don't propagate
// back to the caller (they'd just cause the ring buffer to stop
// filling, which the main thread detects as underruns).
let _ = init_tx.send(Ok(()));
info!("WASAPI communications-mode capture started with OS AEC enabled");
let mut logged_first_packet = false;
// Main capture loop. Exit when `running` goes false (from Drop or an
// explicit stop() call).
while running.load(Ordering::Relaxed) {
// 200 ms timeout so we check `running` regularly even if the audio
// engine stops delivering packets (e.g. device unplugged).
let wait = WaitForSingleObject(event, 200);
if wait.0 != WAIT_OBJECT_0.0 {
// Timeout or failure — just loop and re-check running.
continue;
}
// Drain all available packets. Windows may have queued more than
// one since we were last scheduled.
loop {
let packet_length = match capture_client.GetNextPacketSize() {
Ok(n) => n,
Err(e) => {
warn!("GetNextPacketSize failed: {e}");
break;
}
};
if packet_length == 0 {
break;
}
let mut buffer_ptr: *mut u8 = std::ptr::null_mut();
let mut num_frames: u32 = 0;
let mut flags: u32 = 0;
let mut device_position: u64 = 0;
let mut qpc_position: u64 = 0;
if let Err(e) = capture_client.GetBuffer(
&mut buffer_ptr,
&mut num_frames,
&mut flags,
Some(&mut device_position),
Some(&mut qpc_position),
) {
warn!("GetBuffer failed: {e}");
break;
}
if num_frames > 0 && !buffer_ptr.is_null() {
if !logged_first_packet {
info!(
frames = num_frames,
flags, "WASAPI capture: first packet received"
);
logged_first_packet = true;
}
// Because we asked for 48 kHz mono i16, each frame is
// exactly one i16. Windows's AUTOCONVERTPCM handles the
// conversion from whatever the engine mix format is.
let samples = std::slice::from_raw_parts(
buffer_ptr as *const i16,
num_frames as usize,
);
ring.write(samples);
}
if let Err(e) = capture_client.ReleaseBuffer(num_frames) {
warn!("ReleaseBuffer failed: {e}");
break;
}
}
}
info!("WASAPI capture thread stopping");
let _ = audio_client.Stop();
let _ = CloseHandle(event);
// _com_guard drops here, calling CoUninitialize.
// Silence INFINITE unused-import warning — it's referenced by the
// `windows` crate's WaitForSingleObject alternative but we use the
// 200 ms timeout variant instead. Explicit suppression for clarity.
let _ = INFINITE;
Ok(())
}
// ---------------------------------------------------------------------------
// Helpers
// ---------------------------------------------------------------------------
/// Best-effort device ID string for logging. Grabbing the friendly name via
/// PKEY_Device_FriendlyName requires IPropertyStore + PROPVARIANT plumbing
/// that's far more ceremony than a log line justifies; the ID is already
/// sufficient to confirm we opened the right endpoint.
///
/// Rust 2024 edition's `unsafe_op_in_unsafe_fn` lint requires explicit
/// `unsafe { ... }` blocks inside `unsafe fn` bodies for each unsafe call,
/// even though the whole function is already marked unsafe.
unsafe fn device_name(
device: &windows::Win32::Media::Audio::IMMDevice,
) -> Result<String, anyhow::Error> {
let id = unsafe { device.GetId() }.context("IMMDevice::GetId failed")?;
Ok(unsafe { id.to_string() }.unwrap_or_else(|_| "<non-utf16>".to_string()))
}

View File

@@ -7,7 +7,7 @@ use std::time::{Duration, Instant};
use bytes::Bytes;
use tracing::{debug, info, warn};
use wzp_codec::{ComfortNoise, NoiseSupressor, SilenceDetector};
use wzp_codec::{AutoGainControl, ComfortNoise, EchoCanceller, NoiseSupressor, SilenceDetector};
use wzp_fec::{RaptorQFecDecoder, RaptorQFecEncoder};
use wzp_proto::jitter::{JitterBuffer, PlayoutResult};
use wzp_proto::packet::{MediaHeader, MediaPacket, MiniFrameContext};
@@ -42,6 +42,9 @@ pub struct CallConfig {
/// When enabled, only every 50th frame carries a full 12-byte MediaHeader;
/// intermediate frames use a compact 4-byte MiniHeader.
pub mini_frames_enabled: bool,
/// AEC far-end delay compensation in milliseconds (default: 40).
/// Compensates for the round-trip audio latency from playout to mic capture.
pub aec_delay_ms: u32,
/// Enable adaptive jitter buffer (default: true).
///
/// When true, the jitter buffer target depth is automatically adjusted
@@ -63,6 +66,7 @@ impl Default for CallConfig {
noise_suppression: true,
mini_frames_enabled: true,
adaptive_jitter: true,
aec_delay_ms: 40,
}
}
}
@@ -207,6 +211,10 @@ pub struct CallEncoder {
frame_in_block: u8,
/// Timestamp counter (ms).
timestamp_ms: u32,
/// Acoustic echo canceller (removes speaker echo from mic signal).
aec: EchoCanceller,
/// Automatic gain control (normalises mic level).
agc: AutoGainControl,
/// Silence detector for suppression.
silence_detector: SilenceDetector,
/// Whether silence suppression is enabled.
@@ -237,6 +245,8 @@ impl CallEncoder {
block_id: 0,
frame_in_block: 0,
timestamp_ms: 0,
aec: EchoCanceller::with_delay(48000, 60, config.aec_delay_ms),
agc: AutoGainControl::new(),
silence_detector: SilenceDetector::new(
config.silence_threshold_rms,
config.silence_hangover_frames,
@@ -274,15 +284,21 @@ impl CallEncoder {
/// Input: 48kHz mono PCM, frame size depends on profile (960 for 20ms, 1920 for 40ms).
/// Output: one or more MediaPackets to send.
pub fn encode_frame(&mut self, pcm: &[i16]) -> Result<Vec<MediaPacket>, anyhow::Error> {
// Noise suppression: denoise the PCM before silence detection and encoding.
let pcm = if self.denoiser.is_enabled() {
let mut buf = pcm.to_vec();
self.denoiser.process(&mut buf);
buf
} else {
pcm.to_vec()
};
let pcm = &pcm[..];
// Copy PCM into a mutable buffer for the processing pipeline.
let mut pcm_buf = pcm.to_vec();
// Step 1: Echo cancellation (far-end reference must have been fed already).
self.aec.process_frame(&mut pcm_buf);
// Step 2: Automatic gain control (normalise mic level).
self.agc.process_frame(&mut pcm_buf);
// Step 3: Noise suppression (RNNoise).
if self.denoiser.is_enabled() {
self.denoiser.process(&mut pcm_buf);
}
let pcm = &pcm_buf[..];
// Silence suppression: skip encoding silent frames, periodically send CN.
if self.suppression_enabled && self.silence_detector.is_silent(pcm) {
@@ -400,6 +416,24 @@ impl CallEncoder {
self.frame_in_block = 0;
Ok(())
}
/// Feed decoded playout audio as the echo reference signal.
///
/// Must be called with each decoded frame BEFORE the corresponding
/// microphone frame is processed.
pub fn feed_aec_farend(&mut self, farend: &[i16]) {
self.aec.feed_farend(farend);
}
/// Enable or disable acoustic echo cancellation.
pub fn set_aec_enabled(&mut self, enabled: bool) {
self.aec.set_enabled(enabled);
}
/// Enable or disable automatic gain control.
pub fn set_agc_enabled(&mut self, enabled: bool) {
self.agc.set_enabled(enabled);
}
}
/// Manages the recv/decode side of a call.
@@ -466,6 +500,52 @@ impl CallDecoder {
}
}
/// Switch the decoder to match an incoming packet's codec if it differs
/// from the current profile. This enables cross-codec interop (e.g. one
/// client sends Opus, the other sends Codec2).
fn switch_decoder_if_needed(&mut self, incoming_codec: CodecId) {
if incoming_codec == self.profile.codec || incoming_codec == CodecId::ComfortNoise {
return;
}
let new_profile = Self::profile_for_codec(incoming_codec);
info!(
from = ?self.profile.codec,
to = ?incoming_codec,
"decoder switching codec to match incoming packet"
);
if let Err(e) = self.audio_dec.set_profile(new_profile) {
warn!("failed to switch decoder profile: {e}");
return;
}
self.fec_dec = wzp_fec::create_decoder(&new_profile);
self.profile = new_profile;
}
/// Map a `CodecId` to a reasonable `QualityProfile` for decoding.
fn profile_for_codec(codec: CodecId) -> QualityProfile {
match codec {
CodecId::Opus24k => QualityProfile::GOOD,
CodecId::Opus16k => QualityProfile {
codec: CodecId::Opus16k,
fec_ratio: 0.3,
frame_duration_ms: 20,
frames_per_block: 5,
},
CodecId::Opus6k => QualityProfile::DEGRADED,
CodecId::Opus32k => QualityProfile::STUDIO_32K,
CodecId::Opus48k => QualityProfile::STUDIO_48K,
CodecId::Opus64k => QualityProfile::STUDIO_64K,
CodecId::Codec2_3200 => QualityProfile {
codec: CodecId::Codec2_3200,
fec_ratio: 0.5,
frame_duration_ms: 20,
frames_per_block: 5,
},
CodecId::Codec2_1200 => QualityProfile::CATASTROPHIC,
CodecId::ComfortNoise => QualityProfile::GOOD,
}
}
/// Decode the next audio frame from the jitter buffer.
///
/// Returns PCM samples (48kHz mono) or None if not ready.
@@ -480,6 +560,9 @@ impl CallDecoder {
return Some(pcm.len());
}
// Auto-switch decoder if incoming codec differs from current.
self.switch_decoder_if_needed(pkt.header.codec_id);
self.last_was_cn = false;
let result = match self.audio_dec.decode(&pkt.payload, pcm) {
Ok(n) => Some(n),

View File

@@ -47,6 +47,11 @@ struct CliArgs {
room: Option<String>,
token: Option<String>,
_metrics_file: Option<String>,
version_check: bool,
/// Connect to relay for persistent signaling (direct calls).
signal: bool,
/// Place a direct call to a fingerprint (requires --signal).
call_target: Option<String>,
}
impl CliArgs {
@@ -88,12 +93,20 @@ fn parse_args() -> CliArgs {
let mut room = None;
let mut token = None;
let mut metrics_file = None;
let mut version_check = false;
let mut relay_str = None;
let mut signal = false;
let mut call_target = None;
let mut i = 1;
while i < args.len() {
match args[i].as_str() {
"--live" => live = true,
"--signal" => signal = true,
"--call" => {
i += 1;
call_target = Some(args.get(i).expect("--call requires a fingerprint").to_string());
}
"--send-tone" => {
i += 1;
send_tone_secs = Some(
@@ -169,6 +182,7 @@ fn parse_args() -> CliArgs {
);
}
"--sweep" => sweep = true,
"--version-check" => { version_check = true; }
"--help" | "-h" => {
eprintln!("Usage: wzp-client [options] [relay-addr]");
eprintln!();
@@ -221,6 +235,9 @@ fn parse_args() -> CliArgs {
room,
token,
_metrics_file: metrics_file,
version_check,
signal,
call_target,
}
}
@@ -239,6 +256,32 @@ async fn main() -> anyhow::Result<()> {
return Ok(());
}
// --version-check: query relay version over QUIC and exit
if cli.version_check {
let client_config = wzp_transport::client_config();
let bind_addr: SocketAddr = "0.0.0.0:0".parse()?;
let endpoint = wzp_transport::create_endpoint(bind_addr, None)?;
let conn = wzp_transport::connect(&endpoint, cli.relay_addr, "version", client_config).await?;
match conn.accept_uni().await {
Ok(mut recv) => {
let data = recv.read_to_end(256).await.unwrap_or_default();
let version = String::from_utf8_lossy(&data);
println!("{} {}", cli.relay_addr, version.trim());
}
Err(e) => {
eprintln!("relay {} does not support version query: {e}", cli.relay_addr);
}
}
endpoint.close(0u32.into(), b"done");
return Ok(());
}
// --signal mode: persistent signaling for direct calls
if cli.signal {
let seed = cli.resolve_seed();
return run_signal_mode(cli.relay_addr, seed, cli.token, cli.call_target).await;
}
let seed = cli.resolve_seed();
info!(
@@ -250,12 +293,11 @@ async fn main() -> anyhow::Result<()> {
"WarzonePhone client"
);
// Hash room name for SNI privacy (or "default" if none specified)
// Use raw room name as SNI (consistent with Android + Desktop clients for federation)
let sni = match &cli.room {
Some(name) => {
let hashed = wzp_crypto::hash_room_name(name);
info!(room = %name, hashed = %hashed, "room name hashed for SNI");
hashed
info!(room = %name, "using room name as SNI");
name.clone()
}
None => "default".to_string(),
};
@@ -274,6 +316,26 @@ async fn main() -> anyhow::Result<()> {
let transport = Arc::new(wzp_transport::QuinnTransport::new(connection));
// Register shutdown handler so SIGTERM/SIGINT always closes QUIC cleanly.
// Without this, killed clients leave zombie connections on the relay for ~30s.
{
let shutdown_transport = transport.clone();
tokio::spawn(async move {
let mut sigterm = tokio::signal::unix::signal(tokio::signal::unix::SignalKind::terminate())
.expect("failed to register SIGTERM handler");
let mut sigint = tokio::signal::unix::signal(tokio::signal::unix::SignalKind::interrupt())
.expect("failed to register SIGINT handler");
tokio::select! {
_ = sigterm.recv() => { info!("SIGTERM received, closing connection..."); }
_ = sigint.recv() => { info!("SIGINT received, closing connection..."); }
}
// Close the QUIC connection immediately (APPLICATION_CLOSE frame).
// Don't call process::exit — let the main task detect the closed
// connection and perform clean shutdown (e.g., save recordings).
shutdown_transport.connection().close(0u32.into(), b"shutdown");
});
}
// Send auth token if provided (relay with --auth-url expects this first)
if let Some(ref token) = cli.token {
let auth = wzp_proto::SignalMessage::AuthToken {
@@ -287,6 +349,7 @@ async fn main() -> anyhow::Result<()> {
let _crypto_session = wzp_client::handshake::perform_handshake(
&*transport,
&seed.0,
None, // alias — desktop client doesn't set one yet
).await?;
info!("crypto handshake complete");
@@ -623,3 +686,195 @@ async fn run_live(transport: Arc<wzp_transport::QuinnTransport>) -> anyhow::Resu
info!("done");
Ok(())
}
/// Persistent signaling mode for direct 1:1 calls.
async fn run_signal_mode(
relay_addr: SocketAddr,
seed: wzp_crypto::Seed,
token: Option<String>,
call_target: Option<String>,
) -> anyhow::Result<()> {
use wzp_proto::SignalMessage;
let identity = seed.derive_identity();
let pub_id = identity.public_identity();
let fp = pub_id.fingerprint.to_string();
let identity_pub = *pub_id.signing.as_bytes();
info!(fingerprint = %fp, "signal mode");
// Connect to relay with SNI "_signal"
let client_config = wzp_transport::client_config();
let bind_addr: SocketAddr = if relay_addr.is_ipv6() {
"[::]:0".parse()?
} else {
"0.0.0.0:0".parse()?
};
let endpoint = wzp_transport::create_endpoint(bind_addr, None)?;
let conn = wzp_transport::connect(&endpoint, relay_addr, "_signal", client_config).await?;
let transport = Arc::new(wzp_transport::QuinnTransport::new(conn));
info!("connected to relay (signal channel)");
// Auth if token provided
if let Some(ref tok) = token {
transport.send_signal(&SignalMessage::AuthToken { token: tok.clone() }).await?;
}
// Register presence (signature not verified in Phase 1)
transport.send_signal(&SignalMessage::RegisterPresence {
identity_pub,
signature: vec![], // Phase 1: not verified
alias: None,
}).await?;
// Wait for ack
match transport.recv_signal().await? {
Some(SignalMessage::RegisterPresenceAck { success: true, .. }) => {
info!(fingerprint = %fp, "registered on relay — waiting for calls");
}
Some(SignalMessage::RegisterPresenceAck { success: false, error }) => {
anyhow::bail!("registration failed: {}", error.unwrap_or_default());
}
other => {
anyhow::bail!("unexpected response: {other:?}");
}
}
// If --call specified, place the call
if let Some(ref target) = call_target {
info!(target = %target, "placing direct call...");
let call_id = format!("{:016x}", std::time::SystemTime::now()
.duration_since(std::time::UNIX_EPOCH).unwrap().as_nanos());
transport.send_signal(&SignalMessage::DirectCallOffer {
caller_fingerprint: fp.clone(),
caller_alias: None,
target_fingerprint: target.clone(),
call_id: call_id.clone(),
identity_pub,
ephemeral_pub: [0u8; 32], // Phase 1: not used for key exchange
signature: vec![],
supported_profiles: vec![wzp_proto::QualityProfile::GOOD],
}).await?;
}
// Signal recv loop — handle incoming signals
let signal_transport = transport.clone();
let relay = relay_addr;
let my_fp = fp.clone();
let my_seed = seed.0;
loop {
match signal_transport.recv_signal().await {
Ok(Some(msg)) => match msg {
SignalMessage::CallRinging { call_id } => {
info!(call_id = %call_id, "ringing...");
}
SignalMessage::DirectCallOffer { caller_fingerprint, caller_alias, call_id, .. } => {
info!(
from = %caller_fingerprint,
alias = ?caller_alias,
call_id = %call_id,
"incoming call — auto-accepting (generic)"
);
// Auto-accept for CLI testing
let _ = signal_transport.send_signal(&SignalMessage::DirectCallAnswer {
call_id,
accept_mode: wzp_proto::CallAcceptMode::AcceptGeneric,
identity_pub: Some(identity_pub),
ephemeral_pub: None,
signature: None,
chosen_profile: Some(wzp_proto::QualityProfile::GOOD),
}).await;
}
SignalMessage::DirectCallAnswer { call_id, accept_mode, .. } => {
info!(call_id = %call_id, mode = ?accept_mode, "call answered");
}
SignalMessage::CallSetup { call_id, room, relay_addr: setup_relay } => {
info!(call_id = %call_id, room = %room, relay = %setup_relay, "call setup — connecting to media room");
// Connect to the media room
let media_relay: SocketAddr = setup_relay.parse().unwrap_or(relay);
let media_cfg = wzp_transport::client_config();
match wzp_transport::connect(&endpoint, media_relay, &room, media_cfg).await {
Ok(media_conn) => {
let media_transport = Arc::new(wzp_transport::QuinnTransport::new(media_conn));
// Crypto handshake
match wzp_client::handshake::perform_handshake(&*media_transport, &my_seed, None).await {
Ok(_session) => {
info!("media connected — sending tone (press Ctrl+C to hang up)");
// Simple tone sender for testing
let mt = media_transport.clone();
let send_task = tokio::spawn(async move {
let config = wzp_client::call::CallConfig::default();
let mut encoder = wzp_client::call::CallEncoder::new(&config);
let duration = tokio::time::Duration::from_millis(20);
loop {
let pcm: Vec<i16> = (0..FRAME_SAMPLES)
.map(|_| 0i16) // silence — could be tone
.collect();
if let Ok(pkts) = encoder.encode_frame(&pcm) {
for pkt in &pkts {
if mt.send_media(pkt).await.is_err() { return; }
}
}
tokio::time::sleep(duration).await;
}
});
// Wait for hangup or ctrl+c
loop {
tokio::select! {
sig = signal_transport.recv_signal() => {
match sig {
Ok(Some(SignalMessage::Hangup { .. })) => {
info!("remote hung up");
break;
}
Ok(None) | Err(_) => break,
_ => {}
}
}
_ = tokio::signal::ctrl_c() => {
info!("hanging up...");
let _ = signal_transport.send_signal(&SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
}).await;
break;
}
}
}
send_task.abort();
media_transport.close().await.ok();
info!("call ended");
}
Err(e) => error!("media handshake failed: {e}"),
}
}
Err(e) => error!("media connect failed: {e}"),
}
}
SignalMessage::Hangup { reason } => {
info!(reason = ?reason, "call ended by remote");
}
SignalMessage::Pong { .. } => {}
other => {
info!("signal: {:?}", std::mem::discriminant(&other));
}
},
Ok(None) => {
info!("signal connection closed");
break;
}
Err(e) => {
error!("signal error: {e}");
break;
}
}
}
transport.close().await.ok();
Ok(())
}

View File

@@ -109,12 +109,23 @@ pub fn signal_to_call_type(signal: &SignalMessage) -> CallSignalType {
SignalMessage::RouteResponse { .. } => CallSignalType::Offer, // reuse
SignalMessage::SessionForward { .. } => CallSignalType::Offer, // reuse
SignalMessage::SessionForwardAck { .. } => CallSignalType::Offer, // reuse
SignalMessage::RoomUpdate { .. } => CallSignalType::Offer, // reuse
SignalMessage::FederationHello { .. }
| SignalMessage::GlobalRoomActive { .. }
| SignalMessage::GlobalRoomInactive { .. } => CallSignalType::Offer, // relay-only
SignalMessage::DirectCallOffer { .. } => CallSignalType::Offer,
SignalMessage::DirectCallAnswer { .. } => CallSignalType::Answer,
SignalMessage::CallSetup { .. } => CallSignalType::Offer, // relay-only
SignalMessage::CallRinging { .. } => CallSignalType::Ringing,
SignalMessage::RegisterPresence { .. }
| SignalMessage::RegisterPresenceAck { .. } => CallSignalType::Offer, // relay-only
}
}
#[cfg(test)]
mod tests {
use super::*;
use wzp_proto::QualityProfile;
#[test]
fn payload_roundtrip() {
@@ -123,6 +134,7 @@ mod tests {
ephemeral_pub: [2u8; 32],
signature: vec![3u8; 64],
supported_profiles: vec![QualityProfile::GOOD],
alias: None,
};
let encoded = encode_call_payload(&signal, Some("relay.example.com:4433"), Some("myroom"));
@@ -140,6 +152,7 @@ mod tests {
ephemeral_pub: [0; 32],
signature: vec![],
supported_profiles: vec![],
alias: None,
};
assert!(matches!(signal_to_call_type(&offer), CallSignalType::Offer));

View File

@@ -17,6 +17,7 @@ use wzp_proto::{MediaTransport, QualityProfile, SignalMessage};
pub async fn perform_handshake(
transport: &dyn MediaTransport,
seed: &[u8; 32],
alias: Option<&str>,
) -> Result<Box<dyn CryptoSession>, anyhow::Error> {
// 1. Create key exchange from identity seed
let mut kx = WarzoneKeyExchange::from_identity_seed(seed);
@@ -37,10 +38,14 @@ pub async fn perform_handshake(
ephemeral_pub,
signature,
supported_profiles: vec![
QualityProfile::STUDIO_64K,
QualityProfile::STUDIO_48K,
QualityProfile::STUDIO_32K,
QualityProfile::GOOD,
QualityProfile::DEGRADED,
QualityProfile::CATASTROPHIC,
],
alias: alias.map(|s| s.to_string()),
};
transport.send_signal(&offer).await?;

View File

@@ -8,6 +8,24 @@
#[cfg(feature = "audio")]
pub mod audio_io;
#[cfg(feature = "audio")]
pub mod audio_ring;
// VoiceProcessingIO is an Apple Core Audio API — only compile the module
// when the `vpio` feature is on AND we're targeting macOS. Enabling the
// feature on Windows/Linux was previously silently broken.
#[cfg(all(feature = "vpio", target_os = "macos"))]
pub mod audio_vpio;
// WASAPI-direct capture with Windows's OS-level AEC (AudioCategory_Communications).
// Only compiled when `windows-aec` feature is on AND target is Windows. The
// `windows` dependency is itself gated to Windows in Cargo.toml, so enabling
// this feature on non-Windows targets is a no-op.
#[cfg(all(feature = "windows-aec", target_os = "windows"))]
pub mod audio_wasapi;
// WebRTC AEC3 (Audio Processing Module) wrapper around CPAL capture + playback
// on Linux. Only compiled when `linux-aec` feature is on AND target is Linux.
// The webrtc-audio-processing dep is itself gated to Linux in Cargo.toml.
#[cfg(all(feature = "linux-aec", target_os = "linux"))]
pub mod audio_linux_aec;
pub mod bench;
pub mod call;
pub mod drift_test;
@@ -17,7 +35,48 @@ pub mod handshake;
pub mod metrics;
pub mod sweep;
#[cfg(feature = "audio")]
pub use audio_io::{AudioCapture, AudioPlayback};
// AudioPlayback: three possible backends depending on feature flags.
// 1. Default CPAL (`audio_io::AudioPlayback`) — baseline on every platform.
// 2. Linux AEC (`audio_linux_aec::LinuxAecPlayback`) — CPAL + WebRTC APM
// render-side tee, so echo from speakers gets cancelled from the mic.
//
// On macOS and Windows we always use the default CPAL playback because:
// - macOS: VoiceProcessingIO handles AEC at the capture side (Apple's
// native hardware AEC uses its own reference signal handling).
// - Windows: WASAPI AudioCategory_Communications AEC uses the system
// render mix as reference — no per-process plumbing needed.
//
// Linux is the only platform where the in-app approach is necessary, so
// the AEC playback path is gated to target_os = "linux".
#[cfg(all(
feature = "audio",
any(not(feature = "linux-aec"), not(target_os = "linux"))
))]
pub use audio_io::AudioPlayback;
#[cfg(all(feature = "linux-aec", target_os = "linux"))]
pub use audio_linux_aec::LinuxAecPlayback as AudioPlayback;
// AudioCapture: three possible backends depending on feature flags.
// 1. Default CPAL (`audio_io::AudioCapture`) — baseline on every platform.
// 2. Windows AEC (`audio_wasapi::WasapiAudioCapture`) — direct WASAPI
// with AudioCategory_Communications, OS APO chain does AEC.
// 3. Linux AEC (`audio_linux_aec::LinuxAecCapture`) — CPAL + WebRTC APM
// capture-side echo cancellation using the playback tee as reference.
// All three expose the same public API (`start`, `ring`, `stop`, `Drop`).
#[cfg(all(
feature = "audio",
any(not(feature = "windows-aec"), not(target_os = "windows")),
any(not(feature = "linux-aec"), not(target_os = "linux"))
))]
pub use audio_io::AudioCapture;
#[cfg(all(feature = "windows-aec", target_os = "windows"))]
pub use audio_wasapi::WasapiAudioCapture as AudioCapture;
#[cfg(all(feature = "linux-aec", target_os = "linux"))]
pub use audio_linux_aec::LinuxAecCapture as AudioCapture;
pub use call::{CallConfig, CallDecoder, CallEncoder};
pub use handshake::perform_handshake;

View File

@@ -14,7 +14,7 @@ use crate::codec2_dec::Codec2Decoder;
use crate::codec2_enc::Codec2Encoder;
use crate::opus_dec::OpusDecoder;
use crate::opus_enc::OpusEncoder;
use crate::resample;
use crate::resample::{Downsampler48to8, Upsampler8to48};
// ─── Helpers ─────────────────────────────────────────────────────────────────
@@ -54,6 +54,7 @@ pub struct AdaptiveEncoder {
opus: OpusEncoder,
codec2: Codec2Encoder,
active: CodecId,
downsampler: Downsampler48to8,
}
impl AdaptiveEncoder {
@@ -66,6 +67,7 @@ impl AdaptiveEncoder {
opus,
codec2,
active: profile.codec,
downsampler: Downsampler48to8::new(),
})
}
}
@@ -74,7 +76,7 @@ impl AudioEncoder for AdaptiveEncoder {
fn encode(&mut self, pcm: &[i16], out: &mut [u8]) -> Result<usize, CodecError> {
if is_codec2(self.active) {
// Downsample 48 kHz → 8 kHz then encode via Codec2.
let pcm_8k = resample::resample_48k_to_8k(pcm);
let pcm_8k = self.downsampler.process(pcm);
self.codec2.encode(&pcm_8k, out)
} else {
self.opus.encode(pcm, out)
@@ -126,6 +128,7 @@ pub struct AdaptiveDecoder {
opus: OpusDecoder,
codec2: Codec2Decoder,
active: CodecId,
upsampler: Upsampler8to48,
}
impl AdaptiveDecoder {
@@ -138,6 +141,7 @@ impl AdaptiveDecoder {
opus,
codec2,
active: profile.codec,
upsampler: Upsampler8to48::new(),
})
}
}
@@ -149,7 +153,7 @@ impl AudioDecoder for AdaptiveDecoder {
let c2_samples = self.codec2_frame_samples();
let mut buf_8k = vec![0i16; c2_samples];
let n = self.codec2.decode(encoded, &mut buf_8k)?;
let pcm_48k = resample::resample_8k_to_48k(&buf_8k[..n]);
let pcm_48k = self.upsampler.process(&buf_8k[..n]);
let out_len = pcm_48k.len().min(pcm.len());
pcm[..out_len].copy_from_slice(&pcm_48k[..out_len]);
Ok(out_len)
@@ -163,7 +167,7 @@ impl AudioDecoder for AdaptiveDecoder {
let c2_samples = self.codec2_frame_samples();
let mut buf_8k = vec![0i16; c2_samples];
let n = self.codec2.decode_lost(&mut buf_8k)?;
let pcm_48k = resample::resample_8k_to_48k(&buf_8k[..n]);
let pcm_48k = self.upsampler.process(&buf_8k[..n]);
let out_len = pcm_48k.len().min(pcm.len());
pcm[..out_len].copy_from_slice(&pcm_48k[..out_len]);
Ok(out_len)

335
crates/wzp-codec/src/aec.rs Normal file
View File

@@ -0,0 +1,335 @@
//! Acoustic Echo Cancellation — delay-compensated leaky NLMS with
//! Geigel double-talk detection.
//!
//! Key insight: on a laptop, the round-trip audio latency (playout → speaker
//! → air → mic → capture) is 3050ms. The far-end reference must be delayed
//! by this amount so the adaptive filter models the *echo path*, not the
//! *system delay + echo path*.
//!
//! The leaky coefficient decay prevents the filter from diverging when the
//! echo path changes (e.g. hand near laptop) or when the delay estimate
//! is slightly off.
/// Delay-compensated leaky NLMS echo canceller with Geigel DTD.
pub struct EchoCanceller {
// --- Adaptive filter ---
filter: Vec<f32>,
filter_len: usize,
/// Circular buffer of far-end reference samples (after delay).
far_buf: Vec<f32>,
far_pos: usize,
/// NLMS step size.
mu: f32,
/// Leakage factor: coefficients are multiplied by (1 - leak) each frame.
/// Prevents unbounded growth / divergence. 0.0001 is gentle.
leak: f32,
enabled: bool,
// --- Delay buffer ---
/// Raw far-end samples before delay compensation.
delay_ring: Vec<f32>,
delay_write: usize,
delay_read: usize,
/// Delay in samples (e.g. 1920 = 40ms at 48kHz).
delay_samples: usize,
/// Capacity of the delay ring.
delay_cap: usize,
// --- Double-talk detection (Geigel) ---
/// Peak far-end level over the last filter_len samples.
far_peak: f32,
/// Geigel threshold: if |near| > threshold * far_peak, assume double-talk.
geigel_threshold: f32,
/// Holdover counter: keep DTD active for a few frames after detection.
dtd_holdover: u32,
dtd_hold_frames: u32,
}
impl EchoCanceller {
/// Create a new echo canceller.
///
/// * `sample_rate` — typically 48000
/// * `filter_ms` — echo-tail length in milliseconds (60ms recommended)
/// * `delay_ms` — far-end delay compensation in milliseconds (40ms for laptops)
pub fn new(sample_rate: u32, filter_ms: u32) -> Self {
Self::with_delay(sample_rate, filter_ms, 40)
}
pub fn with_delay(sample_rate: u32, filter_ms: u32, delay_ms: u32) -> Self {
let filter_len = (sample_rate as usize) * (filter_ms as usize) / 1000;
let delay_samples = (sample_rate as usize) * (delay_ms as usize) / 1000;
// Delay ring must hold at least delay_samples + one frame (960) of headroom.
let delay_cap = delay_samples + (sample_rate as usize / 10); // +100ms headroom
Self {
filter: vec![0.0; filter_len],
filter_len,
far_buf: vec![0.0; filter_len],
far_pos: 0,
mu: 0.01,
leak: 0.0001,
enabled: true,
delay_ring: vec![0.0; delay_cap],
delay_write: 0,
delay_read: 0,
delay_samples,
delay_cap,
far_peak: 0.0,
geigel_threshold: 0.7,
dtd_holdover: 0,
dtd_hold_frames: 5,
}
}
/// Feed far-end (speaker) samples. These go into the delay buffer first;
/// once enough samples have accumulated, they are released to the filter's
/// circular buffer with the correct delay offset.
pub fn feed_farend(&mut self, farend: &[i16]) {
// Write raw samples into the delay ring.
for &s in farend {
self.delay_ring[self.delay_write % self.delay_cap] = s as f32;
self.delay_write += 1;
}
// Release delayed samples to the filter's far-end buffer.
while self.delay_available() >= 1 {
let sample = self.delay_ring[self.delay_read % self.delay_cap];
self.delay_read += 1;
self.far_buf[self.far_pos] = sample;
self.far_pos = (self.far_pos + 1) % self.filter_len;
// Track peak far-end level for Geigel DTD.
let abs_s = sample.abs();
if abs_s > self.far_peak {
self.far_peak = abs_s;
}
}
// Decay far_peak slowly (avoids stale peak from a loud burst long ago).
self.far_peak *= 0.9995;
}
/// Number of delayed samples available to release.
fn delay_available(&self) -> usize {
let buffered = self.delay_write - self.delay_read;
if buffered > self.delay_samples {
buffered - self.delay_samples
} else {
0
}
}
/// Process a near-end (microphone) frame, removing the estimated echo.
pub fn process_frame(&mut self, nearend: &mut [i16]) -> f32 {
if !self.enabled {
return 1.0;
}
let n = nearend.len();
let fl = self.filter_len;
// --- Geigel double-talk detection ---
// If any near-end sample exceeds threshold * far_peak, assume
// the local speaker is active and freeze adaptation.
let mut is_doubletalk = self.dtd_holdover > 0;
if !is_doubletalk {
let threshold_level = self.geigel_threshold * self.far_peak;
for &s in nearend.iter() {
if (s as f32).abs() > threshold_level && self.far_peak > 100.0 {
is_doubletalk = true;
self.dtd_holdover = self.dtd_hold_frames;
break;
}
}
}
if self.dtd_holdover > 0 {
self.dtd_holdover -= 1;
}
// Check if far-end is active (otherwise nothing to cancel).
let far_active = self.far_peak > 100.0;
// --- Leaky coefficient decay ---
// Applied once per frame for efficiency.
let decay = 1.0 - self.leak;
for c in self.filter.iter_mut() {
*c *= decay;
}
let mut sum_near_sq: f64 = 0.0;
let mut sum_err_sq: f64 = 0.0;
for i in 0..n {
let near_f = nearend[i] as f32;
// Position of far-end "now" for this near-end sample.
let base = (self.far_pos + fl * ((n / fl) + 2) + i - n) % fl;
// --- Echo estimation: dot(filter, far_end_window) ---
let mut echo_est: f32 = 0.0;
let mut power: f32 = 0.0;
for k in 0..fl {
let fe_idx = (base + fl - k) % fl;
let fe = self.far_buf[fe_idx];
echo_est += self.filter[k] * fe;
power += fe * fe;
}
let error = near_f - echo_est;
// --- NLMS adaptation (only when far-end active & no double-talk) ---
if far_active && !is_doubletalk && power > 10.0 {
let step = self.mu * error / (power + 1.0);
for k in 0..fl {
let fe_idx = (base + fl - k) % fl;
self.filter[k] += step * self.far_buf[fe_idx];
}
}
let out = error.clamp(-32768.0, 32767.0);
nearend[i] = out as i16;
sum_near_sq += (near_f as f64).powi(2);
sum_err_sq += (out as f64).powi(2);
}
if sum_err_sq < 1.0 {
100.0
} else {
(sum_near_sq / sum_err_sq).sqrt() as f32
}
}
pub fn set_enabled(&mut self, enabled: bool) {
self.enabled = enabled;
}
pub fn is_enabled(&self) -> bool {
self.enabled
}
pub fn reset(&mut self) {
self.filter.iter_mut().for_each(|c| *c = 0.0);
self.far_buf.iter_mut().for_each(|s| *s = 0.0);
self.far_pos = 0;
self.far_peak = 0.0;
self.delay_ring.iter_mut().for_each(|s| *s = 0.0);
self.delay_write = 0;
self.delay_read = 0;
self.dtd_holdover = 0;
}
}
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn creates_with_correct_sizes() {
let aec = EchoCanceller::with_delay(48000, 60, 40);
assert_eq!(aec.filter_len, 2880); // 60ms @ 48kHz
assert_eq!(aec.delay_samples, 1920); // 40ms @ 48kHz
}
#[test]
fn passthrough_when_disabled() {
let mut aec = EchoCanceller::new(48000, 60);
aec.set_enabled(false);
let original: Vec<i16> = (0..960).map(|i| (i * 10) as i16).collect();
let mut frame = original.clone();
aec.process_frame(&mut frame);
assert_eq!(frame, original);
}
#[test]
fn silence_passthrough() {
let mut aec = EchoCanceller::with_delay(48000, 30, 0);
aec.feed_farend(&vec![0i16; 960]);
let mut frame = vec![0i16; 960];
aec.process_frame(&mut frame);
assert!(frame.iter().all(|&s| s == 0));
}
#[test]
fn reduces_echo_with_no_delay() {
// Simulate: far-end plays, echo arrives at mic attenuated by ~50%
// (realistic — speaker to mic on laptop loses volume).
let mut aec = EchoCanceller::with_delay(48000, 10, 0);
let frame_len = 480;
let make_tone = |offset: usize| -> Vec<i16> {
(0..frame_len)
.map(|i| {
let t = (offset + i) as f64 / 48000.0;
(5000.0 * (2.0 * std::f64::consts::PI * 300.0 * t).sin()) as i16
})
.collect()
};
let mut last_erle = 1.0f32;
for frame_idx in 0..100 {
let farend = make_tone(frame_idx * frame_len);
aec.feed_farend(&farend);
// Near-end = attenuated copy of far-end (echo at ~50% volume).
let mut nearend: Vec<i16> = farend.iter().map(|&s| s / 2).collect();
last_erle = aec.process_frame(&mut nearend);
}
assert!(
last_erle > 1.0,
"expected ERLE > 1.0 after adaptation, got {last_erle}"
);
}
#[test]
fn preserves_nearend_during_doubletalk() {
let mut aec = EchoCanceller::with_delay(48000, 30, 0);
let frame_len = 960;
let nearend: Vec<i16> = (0..frame_len)
.map(|i| {
let t = i as f64 / 48000.0;
(10000.0 * (2.0 * std::f64::consts::PI * 440.0 * t).sin()) as i16
})
.collect();
// Feed silence as far-end (no echo source).
aec.feed_farend(&vec![0i16; frame_len]);
let mut frame = nearend.clone();
aec.process_frame(&mut frame);
let input_energy: f64 = nearend.iter().map(|&s| (s as f64).powi(2)).sum();
let output_energy: f64 = frame.iter().map(|&s| (s as f64).powi(2)).sum();
let ratio = output_energy / input_energy;
assert!(
ratio > 0.8,
"near-end speech should be preserved, energy ratio = {ratio:.3}"
);
}
#[test]
fn delay_buffer_holds_samples() {
let mut aec = EchoCanceller::with_delay(48000, 10, 20);
// 20ms delay = 960 samples @ 48kHz.
// After feeding, feed_farend auto-drains available samples to far_buf.
// So delay_available() is always 0 after feed_farend returns.
// Instead, verify far_pos advances only after the delay is filled.
// Feed 960 samples (= delay amount). No samples released yet.
aec.feed_farend(&vec![1i16; 960]);
// far_buf should still be all zeros (nothing released).
assert!(aec.far_buf.iter().all(|&s| s == 0.0), "nothing should be released yet");
// Feed 480 more. 480 should be released to far_buf.
aec.feed_farend(&vec![2i16; 480]);
let non_zero = aec.far_buf.iter().filter(|&&s| s != 0.0).count();
assert!(non_zero > 0, "samples should have been released to far_buf");
}
}

219
crates/wzp-codec/src/agc.rs Normal file
View File

@@ -0,0 +1,219 @@
//! Automatic Gain Control (AGC) with two-stage smoothing.
//!
//! Uses a fast attack / slow release envelope follower to keep the
//! output signal near a configurable target RMS level. This prevents
//! both clipping (when the speaker is too loud) and inaudibility (when
//! the speaker is too quiet or far from the mic).
/// Two-stage automatic gain control.
///
/// The gain is adjusted per-frame based on the measured RMS energy,
/// with a fast attack (gain decreases quickly when signal gets louder)
/// and a slow release (gain increases gradually when signal gets quieter).
pub struct AutoGainControl {
target_rms: f64,
current_gain: f64,
min_gain: f64,
max_gain: f64,
attack_alpha: f64,
release_alpha: f64,
enabled: bool,
}
impl AutoGainControl {
/// Create a new AGC with sensible VoIP defaults.
pub fn new() -> Self {
Self {
target_rms: 3000.0, // ~-20 dBFS for i16
current_gain: 1.0,
min_gain: 0.5,
max_gain: 32.0,
attack_alpha: 0.3, // fast attack
release_alpha: 0.02, // slow release
enabled: true,
}
}
/// Process a frame of PCM audio in-place, applying gain adjustment.
pub fn process_frame(&mut self, pcm: &mut [i16]) {
if !self.enabled {
return;
}
// Compute RMS of the frame.
let rms = Self::compute_rms(pcm);
// Don't amplify near-silence — it would just boost noise.
if rms < 10.0 {
return;
}
// Desired instantaneous gain.
let desired_gain = (self.target_rms / rms).clamp(self.min_gain, self.max_gain);
// Smooth the gain transition.
let alpha = if desired_gain < self.current_gain {
// Signal is louder than target → reduce gain quickly (attack).
self.attack_alpha
} else {
// Signal is quieter than target → raise gain slowly (release).
self.release_alpha
};
self.current_gain = self.current_gain * (1.0 - alpha) + desired_gain * alpha;
// Apply gain to each sample with hard limiting at ±31000 (~0.946 * i16::MAX).
const LIMIT: f64 = 31000.0;
let gain = self.current_gain;
for sample in pcm.iter_mut() {
let amplified = (*sample as f64) * gain;
let clamped = amplified.clamp(-LIMIT, LIMIT);
*sample = clamped as i16;
}
}
/// Enable or disable the AGC.
pub fn set_enabled(&mut self, enabled: bool) {
self.enabled = enabled;
}
/// Returns whether the AGC is currently enabled.
pub fn is_enabled(&self) -> bool {
self.enabled
}
/// Current gain expressed in dB.
pub fn current_gain_db(&self) -> f64 {
20.0 * self.current_gain.log10()
}
/// Compute the RMS (root mean square) of a PCM buffer.
fn compute_rms(pcm: &[i16]) -> f64 {
if pcm.is_empty() {
return 0.0;
}
let sum_sq: f64 = pcm.iter().map(|&s| (s as f64) * (s as f64)).sum();
(sum_sq / pcm.len() as f64).sqrt()
}
}
impl Default for AutoGainControl {
fn default() -> Self {
Self::new()
}
}
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn agc_creates_with_defaults() {
let agc = AutoGainControl::new();
assert!(agc.is_enabled());
assert!((agc.current_gain - 1.0).abs() < f64::EPSILON);
}
#[test]
fn agc_passthrough_when_disabled() {
let mut agc = AutoGainControl::new();
agc.set_enabled(false);
let original: Vec<i16> = (0..960).map(|i| (i * 5) as i16).collect();
let mut frame = original.clone();
agc.process_frame(&mut frame);
assert_eq!(frame, original);
}
#[test]
fn agc_does_not_amplify_silence() {
let mut agc = AutoGainControl::new();
let mut frame = vec![0i16; 960];
agc.process_frame(&mut frame);
assert!(frame.iter().all(|&s| s == 0));
// Gain should remain at initial value.
assert!((agc.current_gain - 1.0).abs() < f64::EPSILON);
}
#[test]
fn agc_amplifies_quiet_signal() {
let mut agc = AutoGainControl::new();
// Very quiet signal (RMS ~ 50).
let mut frame: Vec<i16> = (0..960)
.map(|i| {
let t = i as f64 / 48000.0;
(50.0 * (2.0 * std::f64::consts::PI * 440.0 * t).sin()) as i16
})
.collect();
// Process several frames to let the gain ramp up.
for _ in 0..50 {
let mut f = frame.clone();
agc.process_frame(&mut f);
frame = f;
}
// Gain should have increased past 1.0.
assert!(
agc.current_gain > 1.05,
"expected gain > 1.05 for quiet signal, got {}",
agc.current_gain
);
}
#[test]
fn agc_attenuates_loud_signal() {
let mut agc = AutoGainControl::new();
// Loud signal (RMS ~ 20000).
let frame: Vec<i16> = (0..960)
.map(|i| {
let t = i as f64 / 48000.0;
(28000.0 * (2.0 * std::f64::consts::PI * 440.0 * t).sin()) as i16
})
.collect();
// Process several frames.
for _ in 0..20 {
let mut f = frame.clone();
agc.process_frame(&mut f);
}
// Gain should have decreased below 1.0.
assert!(
agc.current_gain < 1.0,
"expected gain < 1.0 for loud signal, got {}",
agc.current_gain
);
}
#[test]
fn agc_output_within_limits() {
let mut agc = AutoGainControl::new();
// Force a high gain by processing many quiet frames first.
for _ in 0..100 {
let mut f: Vec<i16> = vec![100; 960];
agc.process_frame(&mut f);
}
// Now send a louder frame — output should still be within ±31000.
let mut frame: Vec<i16> = vec![20000; 960];
agc.process_frame(&mut frame);
assert!(
frame.iter().all(|&s| s.abs() <= 31000),
"output samples must be within ±31000"
);
}
#[test]
fn agc_gain_db_at_unity() {
let agc = AutoGainControl::new();
let db = agc.current_gain_db();
assert!(
db.abs() < 0.01,
"expected ~0 dB at unity gain, got {db}"
);
}
}

View File

@@ -10,6 +10,8 @@
//! trait-object encoders/decoders that handle adaptive switching internally.
pub mod adaptive;
pub mod aec;
pub mod agc;
pub mod codec2_dec;
pub mod codec2_enc;
pub mod denoise;
@@ -19,6 +21,8 @@ pub mod resample;
pub mod silence;
pub use adaptive::{AdaptiveDecoder, AdaptiveEncoder};
pub use aec::EchoCanceller;
pub use agc::AutoGainControl;
pub use denoise::NoiseSupressor;
pub use silence::{ComfortNoise, SilenceDetector};
pub use wzp_proto::{AudioDecoder, AudioEncoder, CodecId, QualityProfile};

View File

@@ -79,7 +79,7 @@ impl AudioDecoder for OpusDecoder {
fn set_profile(&mut self, profile: QualityProfile) -> Result<(), CodecError> {
match profile.codec {
CodecId::Opus24k | CodecId::Opus16k | CodecId::Opus6k => {
c if c.is_opus() => {
self.codec_id = profile.codec;
self.frame_duration_ms = profile.frame_duration_ms;
Ok(())

View File

@@ -40,6 +40,11 @@ impl OpusEncoder {
.set_signal(Signal::Voice)
.map_err(|e| CodecError::EncodeFailed(format!("set signal: {e}")))?;
// Default complexity 7 — good quality/CPU trade-off for VoIP
enc.inner
.set_complexity(7)
.map_err(|e| CodecError::EncodeFailed(format!("set complexity: {e}")))?;
Ok(enc)
}
@@ -56,6 +61,21 @@ impl OpusEncoder {
pub fn frame_samples(&self) -> usize {
(48_000 * self.frame_duration_ms as usize) / 1000
}
/// Set the encoder complexity (0-10). Higher values produce better quality
/// at the cost of more CPU. Default is 7.
pub fn set_complexity(&mut self, complexity: i32) {
let c = (complexity as u8).min(10);
let _ = self.inner.set_complexity(c);
}
/// Hint the encoder about expected packet loss percentage (0-100).
///
/// Higher values cause the encoder to use more redundancy to survive
/// packet loss, at the expense of slightly higher bitrate.
pub fn set_expected_loss(&mut self, loss_pct: u8) {
let _ = self.inner.set_packet_loss_perc(loss_pct.min(100));
}
}
impl AudioEncoder for OpusEncoder {
@@ -80,7 +100,7 @@ impl AudioEncoder for OpusEncoder {
fn set_profile(&mut self, profile: QualityProfile) -> Result<(), CodecError> {
match profile.codec {
CodecId::Opus24k | CodecId::Opus16k | CodecId::Opus6k => {
c if c.is_opus() => {
self.codec_id = profile.codec;
self.frame_duration_ms = profile.frame_duration_ms;
self.apply_bitrate(profile.codec)?;

View File

@@ -1,55 +1,258 @@
//! Simple linear resampler for 48 kHz <-> 8 kHz conversion.
//! Windowed-sinc FIR resampler for 48 kHz <-> 8 kHz conversion.
//!
//! These are basic implementations suitable for voice. For higher quality,
//! replace with the `rubato` crate later.
//! Provides both stateless free functions (backward-compatible) and stateful
//! `Downsampler48to8` / `Upsampler8to48` structs that maintain overlap history
//! between frames for glitch-free streaming.
/// Downsample from 48 kHz to 8 kHz (6:1 decimation with averaging).
use std::f64::consts::PI;
// ─── FIR kernel parameters ─────────────────────────────────────────────────
/// Number of FIR taps in the anti-alias / interpolation filter.
const FIR_TAPS: usize = 48;
/// Kaiser window beta parameter — controls sidelobe attenuation.
const KAISER_BETA: f64 = 8.0;
/// Cutoff frequency in Hz for the low-pass filter (just below 4 kHz Nyquist of 8 kHz).
const CUTOFF_HZ: f64 = 3800.0;
/// Working sample rate in Hz.
const SAMPLE_RATE: f64 = 48000.0;
/// Decimation / interpolation ratio between 48 kHz and 8 kHz.
const RATIO: usize = 6;
// ─── Kaiser window helpers ─────────────────────────────────────────────────
/// Zeroth-order modified Bessel function of the first kind, I₀(x).
///
/// Each output sample is the average of 6 consecutive input samples,
/// providing basic anti-aliasing via a box filter.
pub fn resample_48k_to_8k(input: &[i16]) -> Vec<i16> {
const RATIO: usize = 6;
let out_len = input.len() / RATIO;
let mut output = Vec::with_capacity(out_len);
for chunk in input.chunks_exact(RATIO) {
let sum: i32 = chunk.iter().map(|&s| s as i32).sum();
output.push((sum / RATIO as i32) as i16);
/// Computed via the well-known power-series expansion, converging rapidly
/// for the moderate values of x used in Kaiser window design.
fn bessel_i0(x: f64) -> f64 {
let mut sum = 1.0f64;
let mut term = 1.0f64;
let half_x = x / 2.0;
for k in 1..=25 {
term *= (half_x / k as f64) * (half_x / k as f64);
sum += term;
if term < 1e-12 * sum {
break;
}
}
output
sum
}
/// Upsample from 8 kHz to 48 kHz (1:6 interpolation with linear interp).
/// Build a windowed-sinc low-pass FIR kernel.
///
/// Linearly interpolates between each pair of input samples to produce
/// 6 output samples per input sample.
pub fn resample_8k_to_48k(input: &[i16]) -> Vec<i16> {
const RATIO: usize = 6;
if input.is_empty() {
return Vec::new();
}
/// Returns `FIR_TAPS` coefficients normalised so that the DC gain is exactly 1.0.
fn build_fir_kernel() -> [f64; FIR_TAPS] {
let mut kernel = [0.0f64; FIR_TAPS];
let m = (FIR_TAPS - 1) as f64;
let fc = CUTOFF_HZ / SAMPLE_RATE; // normalised cutoff (0..0.5)
let beta_denom = bessel_i0(KAISER_BETA);
let out_len = input.len() * RATIO;
let mut output = Vec::with_capacity(out_len);
for i in 0..input.len() {
let current = input[i] as i32;
let next = if i + 1 < input.len() {
input[i + 1] as i32
for i in 0..FIR_TAPS {
// Sinc
let n = i as f64 - m / 2.0;
let sinc = if n.abs() < 1e-12 {
2.0 * fc
} else {
current // hold last sample
(2.0 * PI * fc * n).sin() / (PI * n)
};
for j in 0..RATIO {
let interp = current + (next - current) * j as i32 / RATIO as i32;
output.push(interp as i16);
// Kaiser window
let t = 2.0 * i as f64 / m - 1.0; // range [-1, 1]
let kaiser = bessel_i0(KAISER_BETA * (1.0 - t * t).max(0.0).sqrt()) / beta_denom;
kernel[i] = sinc * kaiser;
}
// Normalise to unity DC gain.
let sum: f64 = kernel.iter().sum();
if sum.abs() > 1e-15 {
for k in kernel.iter_mut() {
*k /= sum;
}
}
output
kernel
}
// ─── Stateful Downsampler 48→8 ─────────────────────────────────────────────
/// Stateful polyphase FIR downsampler from 48 kHz to 8 kHz.
///
/// Maintains `FIR_TAPS - 1` samples of history between successive calls to
/// `process()` for seamless frame boundaries.
pub struct Downsampler48to8 {
kernel: [f64; FIR_TAPS],
history: Vec<f64>,
}
impl Downsampler48to8 {
pub fn new() -> Self {
Self {
kernel: build_fir_kernel(),
history: vec![0.0; FIR_TAPS - 1],
}
}
/// Downsample a block of 48 kHz samples to 8 kHz.
///
/// The input length should be a multiple of 6; any trailing samples that
/// don't form a complete output sample are consumed into the history.
pub fn process(&mut self, input: &[i16]) -> Vec<i16> {
let hist_len = self.history.len(); // FIR_TAPS - 1
let total_len = hist_len + input.len();
// Build a working buffer: history ++ input (as f64).
let mut work = Vec::with_capacity(total_len);
work.extend_from_slice(&self.history);
work.extend(input.iter().map(|&s| s as f64));
let out_len = input.len() / RATIO;
let mut output = Vec::with_capacity(out_len);
for i in 0..out_len {
// The centre of the filter for output sample i sits at
// position hist_len + i*RATIO in the work buffer (aligning
// with the first new input sample at decimation phase 0).
let centre = hist_len + i * RATIO;
let start = centre + 1 - FIR_TAPS; // may be 0 for the first few
let mut acc = 0.0f64;
for k in 0..FIR_TAPS {
let idx = start + k;
if idx < work.len() {
acc += work[idx] * self.kernel[k];
}
}
output.push(acc.round().clamp(-32768.0, 32767.0) as i16);
}
// Update history: keep the last (FIR_TAPS - 1) samples from work.
if work.len() >= hist_len {
self.history
.copy_from_slice(&work[work.len() - hist_len..]);
} else {
// Input was shorter than history — shift.
let shift = hist_len - work.len();
self.history.copy_within(shift.., 0);
for (i, &v) in work.iter().enumerate() {
self.history[hist_len - work.len() + i] = v;
}
}
output
}
}
impl Default for Downsampler48to8 {
fn default() -> Self {
Self::new()
}
}
// ─── Stateful Upsampler 8→48 ───────────────────────────────────────────────
/// Stateful FIR upsampler from 8 kHz to 48 kHz.
///
/// Inserts zeros between input samples (zero-stuffing), then applies the
/// low-pass FIR to remove imaging, with gain compensation of `RATIO`.
pub struct Upsampler8to48 {
kernel: [f64; FIR_TAPS],
history: Vec<f64>,
}
impl Upsampler8to48 {
pub fn new() -> Self {
Self {
kernel: build_fir_kernel(),
history: vec![0.0; FIR_TAPS - 1],
}
}
/// Upsample a block of 8 kHz samples to 48 kHz.
pub fn process(&mut self, input: &[i16]) -> Vec<i16> {
let hist_len = self.history.len(); // FIR_TAPS - 1
// Zero-stuff: insert RATIO-1 zeros between each input sample.
let stuffed_len = input.len() * RATIO;
let total_len = hist_len + stuffed_len;
let mut work = Vec::with_capacity(total_len);
work.extend_from_slice(&self.history);
for &s in input {
work.push(s as f64);
for _ in 1..RATIO {
work.push(0.0);
}
}
let out_len = stuffed_len;
let mut output = Vec::with_capacity(out_len);
// The gain factor compensates for the zeros introduced by stuffing.
let gain = RATIO as f64;
for i in 0..out_len {
let centre = hist_len + i;
let start = centre + 1 - FIR_TAPS;
let mut acc = 0.0f64;
for k in 0..FIR_TAPS {
let idx = start + k;
if idx < work.len() {
acc += work[idx] * self.kernel[k];
}
}
acc *= gain;
output.push(acc.round().clamp(-32768.0, 32767.0) as i16);
}
// Update history.
if work.len() >= hist_len {
self.history
.copy_from_slice(&work[work.len() - hist_len..]);
} else {
let shift = hist_len - work.len();
self.history.copy_within(shift.., 0);
for (i, &v) in work.iter().enumerate() {
self.history[hist_len - work.len() + i] = v;
}
}
output
}
}
impl Default for Upsampler8to48 {
fn default() -> Self {
Self::new()
}
}
// ─── Backward-compatible free functions ─────────────────────────────────────
/// Downsample from 48 kHz to 8 kHz (6:1 decimation with FIR anti-alias filter).
///
/// This is a convenience wrapper that creates a temporary [`Downsampler48to8`].
/// For streaming use, prefer the stateful struct to avoid edge artefacts between
/// frames.
pub fn resample_48k_to_8k(input: &[i16]) -> Vec<i16> {
let mut ds = Downsampler48to8::new();
ds.process(input)
}
/// Upsample from 8 kHz to 48 kHz (1:6 interpolation with FIR imaging filter).
///
/// This is a convenience wrapper that creates a temporary [`Upsampler8to48`].
/// For streaming use, prefer the stateful struct to avoid edge artefacts between
/// frames.
pub fn resample_8k_to_48k(input: &[i16]) -> Vec<i16> {
let mut us = Upsampler8to48::new();
us.process(input)
}
// ─── Tests ──────────────────────────────────────────────────────────────────
#[cfg(test)]
mod tests {
use super::*;
@@ -66,12 +269,28 @@ mod tests {
#[test]
fn dc_signal_preserved() {
// A constant signal should survive resampling
// A constant signal should survive resampling (approximately).
let input = vec![1000i16; 960];
let down = resample_48k_to_8k(&input);
assert!(down.iter().all(|&s| s == 1000));
// Allow some edge transient — check that the middle samples are close.
let mid_start = down.len() / 4;
let mid_end = 3 * down.len() / 4;
for &s in &down[mid_start..mid_end] {
assert!(
(s - 1000).abs() < 50,
"DC downsampled sample {s} too far from 1000"
);
}
let up = resample_8k_to_48k(&down);
assert!(up.iter().all(|&s| s == 1000));
let mid_start_up = up.len() / 4;
let mid_end_up = 3 * up.len() / 4;
for &s in &up[mid_start_up..mid_end_up] {
assert!(
(s - 1000).abs() < 100,
"DC upsampled sample {s} too far from 1000"
);
}
}
#[test]
@@ -79,4 +298,40 @@ mod tests {
assert!(resample_48k_to_8k(&[]).is_empty());
assert!(resample_8k_to_48k(&[]).is_empty());
}
#[test]
fn stateful_downsampler_produces_correct_length() {
let mut ds = Downsampler48to8::new();
let out = ds.process(&vec![0i16; 960]);
assert_eq!(out.len(), 160);
let out2 = ds.process(&vec![0i16; 960]);
assert_eq!(out2.len(), 160);
}
#[test]
fn stateful_upsampler_produces_correct_length() {
let mut us = Upsampler8to48::new();
let out = us.process(&vec![0i16; 160]);
assert_eq!(out.len(), 960);
let out2 = us.process(&vec![0i16; 160]);
assert_eq!(out2.len(), 960);
}
#[test]
fn fir_kernel_has_unity_dc_gain() {
let kernel = build_fir_kernel();
let sum: f64 = kernel.iter().sum();
assert!(
(sum - 1.0).abs() < 1e-10,
"FIR kernel DC gain should be 1.0, got {sum}"
);
}
#[test]
fn bessel_i0_known_values() {
// I₀(0) = 1
assert!((bessel_i0(0.0) - 1.0).abs() < 1e-12);
// I₀(1) ≈ 1.2660658
assert!((bessel_i0(1.0) - 1.2660658).abs() < 1e-5);
}
}

View File

@@ -110,7 +110,18 @@ impl KeyExchange for WarzoneKeyExchange {
hk.expand(b"warzone-session-key", &mut session_key)
.expect("HKDF expand for session key should not fail");
Ok(Box::new(ChaChaSession::new(session_key)))
// Derive SAS (Short Authentication String) from shared secret only.
// The shared secret is identical on both sides (X25519 DH property).
// A MITM would produce a different shared secret → different SAS.
// We use a dedicated HKDF label so SAS is independent of the session key.
let mut sas_key = [0u8; 4];
hk.expand(b"warzone-sas-code", &mut sas_key)
.expect("HKDF expand for SAS should not fail");
let sas_code = u32::from_be_bytes(sas_key) % 10000;
let mut session = ChaChaSession::new(session_key);
session.set_sas(sas_code);
Ok(Box::new(session))
}
}
@@ -211,4 +222,47 @@ mod tests {
assert_eq!(&decrypted, plaintext);
}
#[test]
fn sas_codes_match_between_peers() {
let mut alice = WarzoneKeyExchange::from_identity_seed(&[0xAA; 32]);
let mut bob = WarzoneKeyExchange::from_identity_seed(&[0xBB; 32]);
let alice_eph_pub = alice.generate_ephemeral();
let bob_eph_pub = bob.generate_ephemeral();
let alice_session = alice.derive_session(&bob_eph_pub).unwrap();
let bob_session = bob.derive_session(&alice_eph_pub).unwrap();
let alice_sas = alice_session.sas_code();
let bob_sas = bob_session.sas_code();
assert!(alice_sas.is_some(), "Alice should have SAS");
assert!(bob_sas.is_some(), "Bob should have SAS");
assert_eq!(alice_sas, bob_sas, "SAS codes must match between peers");
assert!(alice_sas.unwrap() < 10000, "SAS should be 4 digits");
}
#[test]
fn sas_differs_for_different_peers() {
let mut alice = WarzoneKeyExchange::from_identity_seed(&[0xAA; 32]);
let mut bob = WarzoneKeyExchange::from_identity_seed(&[0xBB; 32]);
let mut eve = WarzoneKeyExchange::from_identity_seed(&[0xEE; 32]);
let alice_eph = alice.generate_ephemeral();
let bob_eph = bob.generate_ephemeral();
let eve_eph = eve.generate_ephemeral();
let alice_bob_session = alice.derive_session(&bob_eph).unwrap();
// Eve does separate handshake with Bob (MITM scenario)
let eve_bob_session = eve.derive_session(&bob_eph).unwrap();
// SAS codes should differ — Eve's session has different shared secret
assert_ne!(
alice_bob_session.sas_code(),
eve_bob_session.sas_code(),
"MITM session should produce different SAS"
);
}
}

View File

@@ -26,6 +26,8 @@ pub struct ChaChaSession {
rekey_mgr: RekeyManager,
/// Pending ephemeral secret for rekey (stored until peer responds).
pending_rekey_secret: Option<StaticSecret>,
/// Short Authentication String (4-digit code for verbal verification).
sas_code: Option<u32>,
}
impl ChaChaSession {
@@ -46,9 +48,15 @@ impl ChaChaSession {
recv_seq: 0,
rekey_mgr: RekeyManager::new(shared_secret),
pending_rekey_secret: None,
sas_code: None,
}
}
/// Set the SAS code (called by key exchange after derivation).
pub fn set_sas(&mut self, code: u32) {
self.sas_code = Some(code);
}
/// Install a new key (after rekeying).
fn install_key(&mut self, new_key: [u8; 32]) {
use sha2::Digest;
@@ -136,6 +144,10 @@ impl CryptoSession for ChaChaSession {
Ok(())
}
fn sas_code(&self) -> Option<u32> {
self.sas_code
}
}
#[cfg(test)]

View File

@@ -1,6 +1,7 @@
//! RaptorQ FEC decoder — reassembles source blocks from received source and repair symbols.
use std::collections::HashMap;
use std::time::Instant;
use raptorq::{EncodingPacket, ObjectTransmissionInformation, PayloadId, SourceBlockDecoder};
use wzp_proto::error::FecError;
@@ -9,6 +10,9 @@ use wzp_proto::FecDecoder;
/// Length prefix size (u16 little-endian), must match encoder.
const LEN_PREFIX: usize = 2;
/// Decoded blocks older than this are eligible for reuse by a new sender.
const BLOCK_STALE_SECS: u64 = 2;
/// State for one in-flight block being decoded.
struct BlockState {
/// Number of source symbols expected.
@@ -21,6 +25,8 @@ struct BlockState {
decoded: bool,
/// Cached decoded result.
result: Option<Vec<Vec<u8>>>,
/// When this block was last decoded (for staleness check).
decoded_at: Option<Instant>,
}
/// RaptorQ-based FEC decoder that handles multiple concurrent blocks.
@@ -58,6 +64,7 @@ impl RaptorQFecDecoder {
symbol_size: self.symbol_size,
decoded: false,
result: None,
decoded_at: None,
})
}
}
@@ -74,8 +81,20 @@ impl FecDecoder for RaptorQFecDecoder {
let block = self.get_or_create_block(block_id);
if block.decoded {
// Already decoded, ignore additional symbols.
return Ok(());
// If the block was decoded recently, skip (normal duplicate).
// If it's stale (>2s), a new sender is reusing this block_id — reset it.
if let Some(at) = block.decoded_at {
if at.elapsed().as_secs() >= BLOCK_STALE_SECS {
block.decoded = false;
block.result = None;
block.decoded_at = None;
block.packets.clear();
} else {
return Ok(());
}
} else {
return Ok(());
}
}
// Data should already be at symbol_size (length-prefixed and padded by the encoder).
@@ -132,6 +151,7 @@ impl FecDecoder for RaptorQFecDecoder {
let block = self.blocks.get_mut(&block_id).unwrap();
block.decoded = true;
block.decoded_at = Some(Instant::now());
block.result = Some(frames.clone());
Ok(Some(frames))
}

View File

@@ -0,0 +1,29 @@
[package]
name = "wzp-native"
version = "0.1.0"
edition = "2024"
description = "WarzonePhone native audio library — standalone Android cdylib that eventually owns all C++ (Oboe bridge) and exposes a pure-C FFI. Built with cargo-ndk, loaded at runtime by the Tauri desktop cdylib via libloading."
# Crate-type is DELIBERATELY only cdylib (no rlib, no staticlib). This crate
# is built with `cargo ndk -t arm64-v8a build --release -p wzp-native` as a
# standalone .so, which is the same path the legacy wzp-android crate uses
# successfully on the same phone / same NDK. Keeping the crate-type single
# avoids the rust-lang/rust#104707 symbol leak that bit us when Tauri's
# desktop crate had ["staticlib", "cdylib", "rlib"] and any C++ static
# archive pulled bionic's internal pthread_create into the final .so.
[lib]
name = "wzp_native"
crate-type = ["cdylib"]
[build-dependencies]
# cc is SAFE to use here because this crate is a single-cdylib: no
# staticlib in crate-type → no rust-lang/rust#104707 symbol leak. The
# legacy wzp-android crate uses the same setup and works.
cc = "1"
[dependencies]
# Phase 2: Oboe C++ audio bridge. Still no Rust deps — we do the whole
# audio pipeline via extern "C" into the bundled C++ and expose our own
# narrow extern "C" API for wzp-desktop to dlopen via libloading.
# Phase 3 can add wzp-proto/wzp-codec if we want to share codec logic
# instead of calling back into wzp-desktop via callbacks.

119
crates/wzp-native/build.rs Normal file
View File

@@ -0,0 +1,119 @@
//! wzp-native build.rs — Oboe C++ bridge compile on Android.
//!
//! Near-verbatim copy of crates/wzp-android/build.rs (which is known to
//! work). The crucial distinction: this crate is a single-cdylib (no
//! staticlib, no rlib in crate-type) so rust-lang/rust#104707 doesn't
//! apply — bionic's internal pthread_create / __init_tcb symbols stay
//! UND and resolve against libc.so at runtime, as they should.
//!
//! On non-Android hosts we compile `cpp/oboe_stub.cpp` (empty stubs) so
//! `cargo check --target <host>` still works for IDEs and CI.
use std::path::PathBuf;
fn main() {
let target = std::env::var("TARGET").unwrap_or_default();
if target.contains("android") {
// getauxval_fix: override compiler-rt's broken static getauxval
// stub that SIGSEGVs in shared libraries.
cc::Build::new()
.file("cpp/getauxval_fix.c")
.compile("wzp_native_getauxval_fix");
let oboe_dir = fetch_oboe();
match oboe_dir {
Some(oboe_path) => {
println!("cargo:warning=wzp-native: building with Oboe from {:?}", oboe_path);
let mut build = cc::Build::new();
build
.cpp(true)
.std("c++17")
// Shared libc++ — matches legacy wzp-android setup.
.cpp_link_stdlib(Some("c++_shared"))
.include("cpp")
.include(oboe_path.join("include"))
.include(oboe_path.join("src"))
.define("WZP_HAS_OBOE", None)
.file("cpp/oboe_bridge.cpp");
add_cpp_files_recursive(&mut build, &oboe_path.join("src"));
build.compile("wzp_native_oboe_bridge");
}
None => {
println!("cargo:warning=wzp-native: Oboe not found, building stub");
cc::Build::new()
.cpp(true)
.std("c++17")
.cpp_link_stdlib(Some("c++_shared"))
.file("cpp/oboe_stub.cpp")
.include("cpp")
.compile("wzp_native_oboe_bridge");
}
}
// Oboe needs log + OpenSLES backends at runtime.
println!("cargo:rustc-link-lib=log");
println!("cargo:rustc-link-lib=OpenSLES");
// Re-run if any cpp file changes
println!("cargo:rerun-if-changed=cpp/oboe_bridge.cpp");
println!("cargo:rerun-if-changed=cpp/oboe_bridge.h");
println!("cargo:rerun-if-changed=cpp/oboe_stub.cpp");
println!("cargo:rerun-if-changed=cpp/getauxval_fix.c");
} else {
// Non-Android hosts: compile the empty stub so lib.rs's extern
// declarations resolve when someone runs `cargo check` on macOS
// or Linux without an NDK.
cc::Build::new()
.cpp(true)
.std("c++17")
.file("cpp/oboe_stub.cpp")
.include("cpp")
.compile("wzp_native_oboe_bridge");
println!("cargo:rerun-if-changed=cpp/oboe_stub.cpp");
}
}
/// Recursively add all `.cpp` files from a directory to a cc::Build.
fn add_cpp_files_recursive(build: &mut cc::Build, dir: &std::path::Path) {
if !dir.is_dir() {
return;
}
for entry in std::fs::read_dir(dir).unwrap() {
let entry = entry.unwrap();
let path = entry.path();
if path.is_dir() {
add_cpp_files_recursive(build, &path);
} else if path.extension().map_or(false, |e| e == "cpp") {
build.file(&path);
}
}
}
/// Fetch or find Oboe headers + sources (v1.8.1). Same logic as the
/// legacy wzp-android crate's build.rs.
fn fetch_oboe() -> Option<PathBuf> {
let out_dir = PathBuf::from(std::env::var("OUT_DIR").unwrap());
let oboe_dir = out_dir.join("oboe");
if oboe_dir.join("include").join("oboe").join("Oboe.h").exists() {
return Some(oboe_dir);
}
let status = std::process::Command::new("git")
.args([
"clone",
"--depth=1",
"--branch=1.8.1",
"https://github.com/google/oboe.git",
oboe_dir.to_str().unwrap(),
])
.status();
match status {
Ok(s) if s.success() && oboe_dir.join("include").join("oboe").join("Oboe.h").exists() => {
Some(oboe_dir)
}
_ => None,
}
}

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@@ -0,0 +1,21 @@
// Override the broken static getauxval from compiler-rt/CRT.
// The static version reads from __libc_auxv which is NULL in shared libs
// loaded via dlopen, causing SIGSEGV in init_have_lse_atomics at load time.
// This version calls the real bionic getauxval via dlsym.
#ifdef __ANDROID__
#include <dlfcn.h>
#include <stdint.h>
typedef unsigned long (*getauxval_fn)(unsigned long);
unsigned long getauxval(unsigned long type) {
static getauxval_fn real_getauxval = (getauxval_fn)0;
if (!real_getauxval) {
real_getauxval = (getauxval_fn)dlsym((void*)-1L /* RTLD_DEFAULT */, "getauxval");
if (!real_getauxval) {
return 0;
}
}
return real_getauxval(type);
}
#endif

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@@ -0,0 +1,420 @@
// Full Oboe implementation for Android
// This file is compiled only when targeting Android
#include "oboe_bridge.h"
#ifdef __ANDROID__
#include <oboe/Oboe.h>
#include <android/log.h>
#include <cstring>
#include <atomic>
#define LOG_TAG "wzp-oboe"
#define LOGI(...) __android_log_print(ANDROID_LOG_INFO, LOG_TAG, __VA_ARGS__)
#define LOGW(...) __android_log_print(ANDROID_LOG_WARN, LOG_TAG, __VA_ARGS__)
#define LOGE(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
// ---------------------------------------------------------------------------
// Ring buffer helpers (SPSC, lock-free)
// ---------------------------------------------------------------------------
static inline int32_t ring_available_read(const wzp_atomic_int* write_idx,
const wzp_atomic_int* read_idx,
int32_t capacity) {
int32_t w = std::atomic_load_explicit(write_idx, std::memory_order_acquire);
int32_t r = std::atomic_load_explicit(read_idx, std::memory_order_relaxed);
int32_t avail = w - r;
if (avail < 0) avail += capacity;
return avail;
}
static inline int32_t ring_available_write(const wzp_atomic_int* write_idx,
const wzp_atomic_int* read_idx,
int32_t capacity) {
return capacity - 1 - ring_available_read(write_idx, read_idx, capacity);
}
static inline void ring_write(int16_t* buf, int32_t capacity,
wzp_atomic_int* write_idx, const wzp_atomic_int* read_idx,
const int16_t* src, int32_t count) {
int32_t w = std::atomic_load_explicit(write_idx, std::memory_order_relaxed);
for (int32_t i = 0; i < count; i++) {
buf[w] = src[i];
w++;
if (w >= capacity) w = 0;
}
std::atomic_store_explicit(write_idx, w, std::memory_order_release);
}
static inline void ring_read(int16_t* buf, int32_t capacity,
const wzp_atomic_int* write_idx, wzp_atomic_int* read_idx,
int16_t* dst, int32_t count) {
int32_t r = std::atomic_load_explicit(read_idx, std::memory_order_relaxed);
for (int32_t i = 0; i < count; i++) {
dst[i] = buf[r];
r++;
if (r >= capacity) r = 0;
}
std::atomic_store_explicit(read_idx, r, std::memory_order_release);
}
// ---------------------------------------------------------------------------
// Global state
// ---------------------------------------------------------------------------
static std::shared_ptr<oboe::AudioStream> g_capture_stream;
static std::shared_ptr<oboe::AudioStream> g_playout_stream;
// Value copy — the WzpOboeRings the Rust side passes us lives on the caller's
// stack frame and goes away as soon as wzp_oboe_start returns. The raw
// int16/atomic pointers INSIDE the struct point into the Rust-owned, leaked-
// for-the-lifetime-of-the-process AudioBackend singleton, so copying the
// struct by value is safe and keeps the inner pointers valid indefinitely.
// g_rings_valid guards the audio-callback-side read; clearing it in stop()
// signals "no backend" to the callbacks which then return silence + Stop.
static WzpOboeRings g_rings{};
static std::atomic<bool> g_rings_valid{false};
static std::atomic<bool> g_running{false};
static std::atomic<float> g_capture_latency_ms{0.0f};
static std::atomic<float> g_playout_latency_ms{0.0f};
// ---------------------------------------------------------------------------
// Capture callback
// ---------------------------------------------------------------------------
class CaptureCallback : public oboe::AudioStreamDataCallback {
public:
uint64_t calls = 0;
uint64_t total_frames = 0;
uint64_t total_written = 0;
uint64_t ring_full_drops = 0;
oboe::DataCallbackResult onAudioReady(
oboe::AudioStream* stream,
void* audioData,
int32_t numFrames) override {
if (!g_running.load(std::memory_order_relaxed) ||
!g_rings_valid.load(std::memory_order_acquire)) {
return oboe::DataCallbackResult::Stop;
}
const int16_t* src = static_cast<const int16_t*>(audioData);
int32_t avail = ring_available_write(g_rings.capture_write_idx,
g_rings.capture_read_idx,
g_rings.capture_capacity);
int32_t to_write = (numFrames < avail) ? numFrames : avail;
if (to_write > 0) {
ring_write(g_rings.capture_buf, g_rings.capture_capacity,
g_rings.capture_write_idx, g_rings.capture_read_idx,
src, to_write);
}
total_frames += numFrames;
total_written += to_write;
if (to_write < numFrames) {
ring_full_drops += (numFrames - to_write);
}
// Sample-range probe on the FIRST callback to prove we get real audio
if (calls == 0 && numFrames > 0) {
int16_t lo = src[0], hi = src[0];
int32_t sumsq = 0;
for (int32_t i = 0; i < numFrames; i++) {
if (src[i] < lo) lo = src[i];
if (src[i] > hi) hi = src[i];
sumsq += (int32_t)src[i] * (int32_t)src[i];
}
int32_t rms = (int32_t) (numFrames > 0 ? (int32_t)__builtin_sqrt((double)sumsq / (double)numFrames) : 0);
LOGI("capture cb#0: numFrames=%d sample_range=[%d..%d] rms=%d to_write=%d",
numFrames, lo, hi, rms, to_write);
}
// Heartbeat every 50 callbacks (~1s at 20ms/burst)
calls++;
if ((calls % 50) == 0) {
LOGI("capture heartbeat: calls=%llu numFrames=%d ring_avail_write=%d to_write=%d full_drops=%llu total_written=%llu",
(unsigned long long)calls, numFrames, avail, to_write,
(unsigned long long)ring_full_drops, (unsigned long long)total_written);
}
// Update latency estimate
auto result = stream->calculateLatencyMillis();
if (result) {
g_capture_latency_ms.store(static_cast<float>(result.value()),
std::memory_order_relaxed);
}
return oboe::DataCallbackResult::Continue;
}
};
// ---------------------------------------------------------------------------
// Playout callback
// ---------------------------------------------------------------------------
class PlayoutCallback : public oboe::AudioStreamDataCallback {
public:
uint64_t calls = 0;
uint64_t total_frames = 0;
uint64_t total_played_real = 0;
uint64_t underrun_frames = 0;
uint64_t nonempty_calls = 0;
oboe::DataCallbackResult onAudioReady(
oboe::AudioStream* stream,
void* audioData,
int32_t numFrames) override {
if (!g_running.load(std::memory_order_relaxed) ||
!g_rings_valid.load(std::memory_order_acquire)) {
memset(audioData, 0, numFrames * sizeof(int16_t));
return oboe::DataCallbackResult::Stop;
}
int16_t* dst = static_cast<int16_t*>(audioData);
int32_t avail = ring_available_read(g_rings.playout_write_idx,
g_rings.playout_read_idx,
g_rings.playout_capacity);
int32_t to_read = (numFrames < avail) ? numFrames : avail;
if (to_read > 0) {
ring_read(g_rings.playout_buf, g_rings.playout_capacity,
g_rings.playout_write_idx, g_rings.playout_read_idx,
dst, to_read);
nonempty_calls++;
}
// Fill remainder with silence on underrun
if (to_read < numFrames) {
memset(dst + to_read, 0, (numFrames - to_read) * sizeof(int16_t));
underrun_frames += (numFrames - to_read);
}
total_frames += numFrames;
total_played_real += to_read;
// First callback: log requested config + prove we're being called
if (calls == 0) {
LOGI("playout cb#0: numFrames=%d ring_avail_read=%d to_read=%d",
numFrames, avail, to_read);
}
// On the first callback that actually has data, log the sample range
// so we can tell if the samples coming out of the ring look like real
// audio vs constant-zeroes vs garbage.
if (to_read > 0 && nonempty_calls == 1) {
int16_t lo = dst[0], hi = dst[0];
int32_t sumsq = 0;
for (int32_t i = 0; i < to_read; i++) {
if (dst[i] < lo) lo = dst[i];
if (dst[i] > hi) hi = dst[i];
sumsq += (int32_t)dst[i] * (int32_t)dst[i];
}
int32_t rms = (to_read > 0) ? (int32_t)__builtin_sqrt((double)sumsq / (double)to_read) : 0;
LOGI("playout FIRST nonempty read: to_read=%d sample_range=[%d..%d] rms=%d",
to_read, lo, hi, rms);
}
// Heartbeat every 50 callbacks (~1s at 20ms/burst)
calls++;
if ((calls % 50) == 0) {
int state = (int)stream->getState();
auto xrunRes = stream->getXRunCount();
int xruns = xrunRes ? xrunRes.value() : -1;
LOGI("playout heartbeat: calls=%llu nonempty=%llu numFrames=%d ring_avail_read=%d to_read=%d underrun_frames=%llu total_played_real=%llu state=%d xruns=%d",
(unsigned long long)calls, (unsigned long long)nonempty_calls,
numFrames, avail, to_read,
(unsigned long long)underrun_frames, (unsigned long long)total_played_real,
state, xruns);
}
// Update latency estimate
auto result = stream->calculateLatencyMillis();
if (result) {
g_playout_latency_ms.store(static_cast<float>(result.value()),
std::memory_order_relaxed);
}
return oboe::DataCallbackResult::Continue;
}
};
static CaptureCallback g_capture_cb;
static PlayoutCallback g_playout_cb;
// ---------------------------------------------------------------------------
// Public C API
// ---------------------------------------------------------------------------
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
if (g_running.load(std::memory_order_relaxed)) {
LOGW("wzp_oboe_start: already running");
return -1;
}
// Deep-copy the rings struct into static storage BEFORE we publish it to
// the audio callbacks — `rings` points at the caller's stack frame and
// goes away as soon as this function returns.
g_rings = *rings;
g_rings_valid.store(true, std::memory_order_release);
// Build capture stream
oboe::AudioStreamBuilder captureBuilder;
captureBuilder.setDirection(oboe::Direction::Input)
->setPerformanceMode(oboe::PerformanceMode::LowLatency)
->setSharingMode(oboe::SharingMode::Exclusive)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setInputPreset(oboe::InputPreset::VoiceCommunication)
->setDataCallback(&g_capture_cb);
oboe::Result result = captureBuilder.openStream(g_capture_stream);
if (result != oboe::Result::OK) {
LOGE("Failed to open capture stream: %s", oboe::convertToText(result));
return -2;
}
LOGI("capture stream opened: actualSR=%d actualCh=%d actualFormat=%d actualFramesPerBurst=%d actualFramesPerDataCallback=%d bufferCapacityInFrames=%d sharing=%d perfMode=%d",
g_capture_stream->getSampleRate(),
g_capture_stream->getChannelCount(),
(int)g_capture_stream->getFormat(),
g_capture_stream->getFramesPerBurst(),
g_capture_stream->getFramesPerDataCallback(),
g_capture_stream->getBufferCapacityInFrames(),
(int)g_capture_stream->getSharingMode(),
(int)g_capture_stream->getPerformanceMode());
// Build playout stream.
//
// Regression triangulation between builds:
// 96be740 (Usage::Media, default API): playout callback DID drain
// the ring at steady 50Hz (playout heartbeat: calls=1100,
// total_played_real=1055040). Audio not audible because OS routing
// sent it to a silent output.
//
// 8c36fb5 (Usage::VoiceCommunication + setAudioApi(AAudio) +
// ContentType::Speech): playout callback fired cb#0 once then
// stopped draining the ring entirely. written_samples stuck at
// ring capacity (7679) across all subsequent heartbeats, so Oboe
// accepted zero samples after startup. Still inaudible.
//
// Hypothesis: forcing setAudioApi(AAudio) + VoiceCommunication on
// Pixel 6 / Android 15 opens a stream that succeeds at cb#0 but
// then detaches from the real audio driver. Reverting to the
// config that at least drove callbacks correctly, plus the
// Kotlin-side MODE_IN_COMMUNICATION + setSpeakerphoneOn(true)
// handled in MainActivity.kt to route audio to the loud speaker.
// Usage::VoiceCommunication is the correct Oboe usage for a VoIP app
// — it respects Android's in-call audio routing and lets
// AudioManager.setSpeakerphoneOn/setBluetoothScoOn actually switch
// between earpiece, loudspeaker, and Bluetooth headset. Combined with
// MODE_IN_COMMUNICATION set from MainActivity.kt and
// speakerphoneOn=false by default, this produces handset/earpiece as
// the default output.
//
// IMPORTANT: do NOT add setAudioApi(AAudio) here. Build 8c36fb5 proved
// forcing AAudio with Usage::VoiceCommunication makes the playout
// callback stop draining the ring after cb#0, even though the stream
// opens successfully. Letting Oboe pick the API (which will be AAudio
// on API ≥ 27 but via a different codepath) kept callbacks firing in
// every other build.
oboe::AudioStreamBuilder playoutBuilder;
playoutBuilder.setDirection(oboe::Direction::Output)
->setPerformanceMode(oboe::PerformanceMode::LowLatency)
->setSharingMode(oboe::SharingMode::Exclusive)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setUsage(oboe::Usage::VoiceCommunication)
->setDataCallback(&g_playout_cb);
result = playoutBuilder.openStream(g_playout_stream);
if (result != oboe::Result::OK) {
LOGE("Failed to open playout stream: %s", oboe::convertToText(result));
g_capture_stream->close();
g_capture_stream.reset();
return -3;
}
LOGI("playout stream opened: actualSR=%d actualCh=%d actualFormat=%d actualFramesPerBurst=%d actualFramesPerDataCallback=%d bufferCapacityInFrames=%d sharing=%d perfMode=%d",
g_playout_stream->getSampleRate(),
g_playout_stream->getChannelCount(),
(int)g_playout_stream->getFormat(),
g_playout_stream->getFramesPerBurst(),
g_playout_stream->getFramesPerDataCallback(),
g_playout_stream->getBufferCapacityInFrames(),
(int)g_playout_stream->getSharingMode(),
(int)g_playout_stream->getPerformanceMode());
g_running.store(true, std::memory_order_release);
// Start both streams
result = g_capture_stream->requestStart();
if (result != oboe::Result::OK) {
LOGE("Failed to start capture: %s", oboe::convertToText(result));
g_running.store(false, std::memory_order_release);
g_capture_stream->close();
g_playout_stream->close();
g_capture_stream.reset();
g_playout_stream.reset();
return -4;
}
result = g_playout_stream->requestStart();
if (result != oboe::Result::OK) {
LOGE("Failed to start playout: %s", oboe::convertToText(result));
g_running.store(false, std::memory_order_release);
g_capture_stream->requestStop();
g_capture_stream->close();
g_playout_stream->close();
g_capture_stream.reset();
g_playout_stream.reset();
return -5;
}
LOGI("Oboe started: sr=%d burst=%d ch=%d",
config->sample_rate, config->frames_per_burst, config->channel_count);
return 0;
}
void wzp_oboe_stop(void) {
g_running.store(false, std::memory_order_release);
// Tell the audio callbacks to stop touching g_rings BEFORE we tear down
// the streams, so any in-flight callback returns Stop instead of reading
// stale pointers.
g_rings_valid.store(false, std::memory_order_release);
if (g_capture_stream) {
g_capture_stream->requestStop();
g_capture_stream->close();
g_capture_stream.reset();
}
if (g_playout_stream) {
g_playout_stream->requestStop();
g_playout_stream->close();
g_playout_stream.reset();
}
LOGI("Oboe stopped");
}
float wzp_oboe_capture_latency_ms(void) {
return g_capture_latency_ms.load(std::memory_order_relaxed);
}
float wzp_oboe_playout_latency_ms(void) {
return g_playout_latency_ms.load(std::memory_order_relaxed);
}
int wzp_oboe_is_running(void) {
return g_running.load(std::memory_order_relaxed) ? 1 : 0;
}
#else
// Non-Android fallback — should not be reached; oboe_stub.cpp is used instead.
// Provide empty implementations just in case.
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
(void)config; (void)rings;
return -99;
}
void wzp_oboe_stop(void) {}
float wzp_oboe_capture_latency_ms(void) { return 0.0f; }
float wzp_oboe_playout_latency_ms(void) { return 0.0f; }
int wzp_oboe_is_running(void) { return 0; }
#endif // __ANDROID__

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@@ -0,0 +1,43 @@
#ifndef WZP_OBOE_BRIDGE_H
#define WZP_OBOE_BRIDGE_H
#include <stdint.h>
#ifdef __cplusplus
#include <atomic>
typedef std::atomic<int32_t> wzp_atomic_int;
extern "C" {
#else
#include <stdatomic.h>
typedef atomic_int wzp_atomic_int;
#endif
typedef struct {
int32_t sample_rate;
int32_t frames_per_burst;
int32_t channel_count;
} WzpOboeConfig;
typedef struct {
int16_t* capture_buf;
int32_t capture_capacity;
wzp_atomic_int* capture_write_idx;
wzp_atomic_int* capture_read_idx;
int16_t* playout_buf;
int32_t playout_capacity;
wzp_atomic_int* playout_write_idx;
wzp_atomic_int* playout_read_idx;
} WzpOboeRings;
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings);
void wzp_oboe_stop(void);
float wzp_oboe_capture_latency_ms(void);
float wzp_oboe_playout_latency_ms(void);
int wzp_oboe_is_running(void);
#ifdef __cplusplus
}
#endif
#endif // WZP_OBOE_BRIDGE_H

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@@ -0,0 +1,27 @@
// Stub implementation for non-Android host builds (testing, cargo check, etc.)
#include "oboe_bridge.h"
#include <stdio.h>
int wzp_oboe_start(const WzpOboeConfig* config, const WzpOboeRings* rings) {
(void)config;
(void)rings;
fprintf(stderr, "wzp_oboe_start: stub (not on Android)\n");
return 0;
}
void wzp_oboe_stop(void) {
fprintf(stderr, "wzp_oboe_stop: stub (not on Android)\n");
}
float wzp_oboe_capture_latency_ms(void) {
return 0.0f;
}
float wzp_oboe_playout_latency_ms(void) {
return 0.0f;
}
int wzp_oboe_is_running(void) {
return 0;
}

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//! wzp-native — standalone Android cdylib for all the C++ audio code.
//!
//! Built with `cargo ndk`, NOT `cargo tauri android build`. Loaded at
//! runtime by the Tauri desktop cdylib (`wzp-desktop`) via libloading.
//! See `docs/incident-tauri-android-init-tcb.md` for why the split exists.
//!
//! Phase 2: real Oboe audio backend.
//!
//! Architecture: Oboe runs capture + playout streams on its own high-
//! priority AAudio callback threads inside the C++ bridge. Two SPSC ring
//! buffers (capture and playout) are shared between the C++ callbacks
//! and the Rust side via atomic indices — no locks on the hot path.
//! `wzp-desktop` drains the capture ring into its Opus encoder and fills
//! the playout ring with decoded PCM.
use std::sync::atomic::{AtomicI32, Ordering};
// ─── Phase 1 smoke-test exports (kept for sanity checks) ─────────────────
/// Returns 42. Used by wzp-desktop's setup() to verify dlopen + dlsym
/// work before any audio code runs.
#[unsafe(no_mangle)]
pub extern "C" fn wzp_native_version() -> i32 {
42
}
/// Writes a NUL-terminated string into `out` (capped at `cap`) and
/// returns bytes written excluding the NUL.
#[unsafe(no_mangle)]
pub unsafe extern "C" fn wzp_native_hello(out: *mut u8, cap: usize) -> usize {
const MSG: &[u8] = b"hello from wzp-native\0";
if out.is_null() || cap == 0 {
return 0;
}
let n = MSG.len().min(cap);
unsafe {
core::ptr::copy_nonoverlapping(MSG.as_ptr(), out, n);
*out.add(n - 1) = 0;
}
n - 1
}
// ─── C++ Oboe bridge FFI ─────────────────────────────────────────────────
#[repr(C)]
struct WzpOboeConfig {
sample_rate: i32,
frames_per_burst: i32,
channel_count: i32,
}
#[repr(C)]
struct WzpOboeRings {
capture_buf: *mut i16,
capture_capacity: i32,
capture_write_idx: *mut AtomicI32,
capture_read_idx: *mut AtomicI32,
playout_buf: *mut i16,
playout_capacity: i32,
playout_write_idx: *mut AtomicI32,
playout_read_idx: *mut AtomicI32,
}
// SAFETY: atomics synchronise producer/consumer; raw pointers are owned
// by the AudioBackend singleton below whose lifetime covers all calls.
unsafe impl Send for WzpOboeRings {}
unsafe impl Sync for WzpOboeRings {}
unsafe extern "C" {
fn wzp_oboe_start(config: *const WzpOboeConfig, rings: *const WzpOboeRings) -> i32;
fn wzp_oboe_stop();
fn wzp_oboe_capture_latency_ms() -> f32;
fn wzp_oboe_playout_latency_ms() -> f32;
fn wzp_oboe_is_running() -> i32;
}
// ─── SPSC ring buffer (shared with C++ via AtomicI32) ────────────────────
/// 20 ms @ 48 kHz mono = 960 samples.
const FRAME_SAMPLES: usize = 960;
/// ~160 ms headroom at 48 kHz.
const RING_CAPACITY: usize = 7680;
struct RingBuffer {
buf: Vec<i16>,
capacity: usize,
write_idx: AtomicI32,
read_idx: AtomicI32,
}
// SAFETY: SPSC with atomic read/write cursors; producer and consumer
// are always on different threads.
unsafe impl Send for RingBuffer {}
unsafe impl Sync for RingBuffer {}
impl RingBuffer {
fn new(capacity: usize) -> Self {
Self {
buf: vec![0i16; capacity],
capacity,
write_idx: AtomicI32::new(0),
read_idx: AtomicI32::new(0),
}
}
fn available_read(&self) -> usize {
let w = self.write_idx.load(Ordering::Acquire);
let r = self.read_idx.load(Ordering::Relaxed);
let avail = w - r;
if avail < 0 { (avail + self.capacity as i32) as usize } else { avail as usize }
}
fn available_write(&self) -> usize {
self.capacity - 1 - self.available_read()
}
fn write(&self, data: &[i16]) -> usize {
let count = data.len().min(self.available_write());
if count == 0 {
return 0;
}
let mut w = self.write_idx.load(Ordering::Relaxed) as usize;
let cap = self.capacity;
let buf_ptr = self.buf.as_ptr() as *mut i16;
for sample in &data[..count] {
unsafe { *buf_ptr.add(w) = *sample; }
w += 1;
if w >= cap { w = 0; }
}
self.write_idx.store(w as i32, Ordering::Release);
count
}
fn read(&self, out: &mut [i16]) -> usize {
let count = out.len().min(self.available_read());
if count == 0 {
return 0;
}
let mut r = self.read_idx.load(Ordering::Relaxed) as usize;
let cap = self.capacity;
let buf_ptr = self.buf.as_ptr();
for slot in &mut out[..count] {
unsafe { *slot = *buf_ptr.add(r); }
r += 1;
if r >= cap { r = 0; }
}
self.read_idx.store(r as i32, Ordering::Release);
count
}
fn buf_ptr(&self) -> *mut i16 {
self.buf.as_ptr() as *mut i16
}
fn write_idx_ptr(&self) -> *mut AtomicI32 {
&self.write_idx as *const AtomicI32 as *mut AtomicI32
}
fn read_idx_ptr(&self) -> *mut AtomicI32 {
&self.read_idx as *const AtomicI32 as *mut AtomicI32
}
}
// ─── AudioBackend singleton ──────────────────────────────────────────────
//
// There is one global AudioBackend instance because Oboe's C++ side
// holds its own singleton of the streams. The `Box::leak`'d statics own
// the ring buffers for the lifetime of the process — dropping them while
// Oboe is still running would cause use-after-free in the audio callback.
use std::sync::OnceLock;
struct AudioBackend {
capture: RingBuffer,
playout: RingBuffer,
started: std::sync::Mutex<bool>,
/// Per-write logging throttle counter for wzp_native_audio_write_playout.
playout_write_log_count: std::sync::atomic::AtomicU64,
}
static BACKEND: OnceLock<&'static AudioBackend> = OnceLock::new();
fn backend() -> &'static AudioBackend {
BACKEND.get_or_init(|| {
Box::leak(Box::new(AudioBackend {
capture: RingBuffer::new(RING_CAPACITY),
playout: RingBuffer::new(RING_CAPACITY),
started: std::sync::Mutex::new(false),
playout_write_log_count: std::sync::atomic::AtomicU64::new(0),
}))
})
}
// ─── C FFI for wzp-desktop ───────────────────────────────────────────────
/// Start the Oboe audio streams. Returns 0 on success, non-zero on error.
/// Idempotent — calling while already running is a no-op that returns 0.
#[unsafe(no_mangle)]
pub extern "C" fn wzp_native_audio_start() -> i32 {
let b = backend();
let mut started = match b.started.lock() {
Ok(g) => g,
Err(_) => return -1,
};
if *started {
return 0;
}
let config = WzpOboeConfig {
sample_rate: 48_000,
frames_per_burst: FRAME_SAMPLES as i32,
channel_count: 1,
};
let rings = WzpOboeRings {
capture_buf: b.capture.buf_ptr(),
capture_capacity: b.capture.capacity as i32,
capture_write_idx: b.capture.write_idx_ptr(),
capture_read_idx: b.capture.read_idx_ptr(),
playout_buf: b.playout.buf_ptr(),
playout_capacity: b.playout.capacity as i32,
playout_write_idx: b.playout.write_idx_ptr(),
playout_read_idx: b.playout.read_idx_ptr(),
};
let ret = unsafe { wzp_oboe_start(&config, &rings) };
if ret != 0 {
return ret;
}
*started = true;
0
}
/// Stop Oboe. Idempotent. Safe to call from any thread.
#[unsafe(no_mangle)]
pub extern "C" fn wzp_native_audio_stop() {
let b = backend();
if let Ok(mut started) = b.started.lock() {
if *started {
unsafe { wzp_oboe_stop() };
*started = false;
}
}
}
/// Read captured PCM samples from the capture ring. Returns the number
/// of `i16` samples actually copied into `out` (may be less than
/// `out_len` if the ring is empty).
#[unsafe(no_mangle)]
pub unsafe extern "C" fn wzp_native_audio_read_capture(out: *mut i16, out_len: usize) -> usize {
if out.is_null() || out_len == 0 {
return 0;
}
let slice = unsafe { std::slice::from_raw_parts_mut(out, out_len) };
backend().capture.read(slice)
}
/// Write PCM samples into the playout ring. Returns the number of
/// samples actually enqueued (may be less than `in_len` if the ring
/// is nearly full — in practice the caller should pace to 20 ms
/// frames and spin briefly if the ring is full).
#[unsafe(no_mangle)]
pub unsafe extern "C" fn wzp_native_audio_write_playout(input: *const i16, in_len: usize) -> usize {
if input.is_null() || in_len == 0 {
return 0;
}
let slice = unsafe { std::slice::from_raw_parts(input, in_len) };
let b = backend();
let before_w = b.playout.write_idx.load(std::sync::atomic::Ordering::Relaxed);
let before_r = b.playout.read_idx.load(std::sync::atomic::Ordering::Relaxed);
let written = b.playout.write(slice);
// First few writes: log ring state + sample range so we can compare what
// engine.rs hands us to what the C++ playout callback reads.
let first_writes = b.playout_write_log_count.fetch_add(1, std::sync::atomic::Ordering::Relaxed);
if first_writes < 3 || first_writes % 50 == 0 {
let (mut lo, mut hi, mut sumsq) = (i16::MAX, i16::MIN, 0i64);
for &s in slice.iter() {
if s < lo { lo = s; }
if s > hi { hi = s; }
sumsq += (s as i64) * (s as i64);
}
let rms = (sumsq as f64 / slice.len() as f64).sqrt() as i32;
let avail_w_after = b.playout.available_write();
let avail_r_after = b.playout.available_read();
let msg = format!(
"playout WRITE #{first_writes}: in_len={} written={} range=[{lo}..{hi}] rms={rms} before_w={before_w} before_r={before_r} avail_read_after={avail_r_after} avail_write_after={avail_w_after}",
slice.len(), written
);
unsafe {
android_log(msg.as_str());
}
}
written
}
// Minimal android logcat shim so we can print from the cdylib without pulling
// in android_logger crate (which would add another dep that has to build with
// cargo-ndk). Uses libc's __android_log_print via extern linkage.
#[cfg(target_os = "android")]
unsafe extern "C" {
fn __android_log_write(prio: i32, tag: *const u8, text: *const u8) -> i32;
}
#[cfg(target_os = "android")]
unsafe fn android_log(msg: &str) {
// ANDROID_LOG_INFO = 4. Tag and text must be NUL-terminated.
let tag = b"wzp-native\0";
let mut buf = Vec::with_capacity(msg.len() + 1);
buf.extend_from_slice(msg.as_bytes());
buf.push(0);
unsafe { __android_log_write(4, tag.as_ptr(), buf.as_ptr()); }
}
#[cfg(not(target_os = "android"))]
#[allow(dead_code)]
unsafe fn android_log(_msg: &str) {}
/// Current capture latency reported by Oboe, in milliseconds. Returns
/// NaN / 0.0 if the stream isn't running.
#[unsafe(no_mangle)]
pub extern "C" fn wzp_native_audio_capture_latency_ms() -> f32 {
unsafe { wzp_oboe_capture_latency_ms() }
}
/// Current playout latency reported by Oboe, in milliseconds.
#[unsafe(no_mangle)]
pub extern "C" fn wzp_native_audio_playout_latency_ms() -> f32 {
unsafe { wzp_oboe_playout_latency_ms() }
}
/// Non-zero if both Oboe streams are currently running.
#[unsafe(no_mangle)]
pub extern "C" fn wzp_native_audio_is_running() -> i32 {
unsafe { wzp_oboe_is_running() }
}

View File

@@ -18,6 +18,12 @@ pub enum CodecId {
Codec2_1200 = 4,
/// Comfort noise descriptor (silence suppression)
ComfortNoise = 5,
/// Opus at 32kbps (studio low)
Opus32k = 6,
/// Opus at 48kbps (studio)
Opus48k = 7,
/// Opus at 64kbps (studio high)
Opus64k = 8,
}
impl CodecId {
@@ -27,6 +33,9 @@ impl CodecId {
Self::Opus24k => 24_000,
Self::Opus16k => 16_000,
Self::Opus6k => 6_000,
Self::Opus32k => 32_000,
Self::Opus48k => 48_000,
Self::Opus64k => 64_000,
Self::Codec2_3200 => 3_200,
Self::Codec2_1200 => 1_200,
Self::ComfortNoise => 0,
@@ -36,8 +45,7 @@ impl CodecId {
/// Preferred frame duration in milliseconds.
pub const fn frame_duration_ms(self) -> u8 {
match self {
Self::Opus24k => 20,
Self::Opus16k => 20,
Self::Opus24k | Self::Opus16k | Self::Opus32k | Self::Opus48k | Self::Opus64k => 20,
Self::Opus6k => 40,
Self::Codec2_3200 => 20,
Self::Codec2_1200 => 40,
@@ -48,7 +56,8 @@ impl CodecId {
/// Sample rate expected by this codec.
pub const fn sample_rate_hz(self) -> u32 {
match self {
Self::Opus24k | Self::Opus16k | Self::Opus6k => 48_000,
Self::Opus24k | Self::Opus16k | Self::Opus6k
| Self::Opus32k | Self::Opus48k | Self::Opus64k => 48_000,
Self::Codec2_3200 | Self::Codec2_1200 => 8_000,
Self::ComfortNoise => 48_000,
}
@@ -63,6 +72,9 @@ impl CodecId {
3 => Some(Self::Codec2_3200),
4 => Some(Self::Codec2_1200),
5 => Some(Self::ComfortNoise),
6 => Some(Self::Opus32k),
7 => Some(Self::Opus48k),
8 => Some(Self::Opus64k),
_ => None,
}
}
@@ -71,6 +83,12 @@ impl CodecId {
pub const fn to_wire(self) -> u8 {
self as u8
}
/// Returns true if this is an Opus variant.
pub const fn is_opus(self) -> bool {
matches!(self, Self::Opus6k | Self::Opus16k | Self::Opus24k
| Self::Opus32k | Self::Opus48k | Self::Opus64k)
}
}
/// Describes the complete quality configuration for a call session.
@@ -111,6 +129,30 @@ impl QualityProfile {
frames_per_block: 8,
};
/// Studio low: Opus 32kbps, minimal FEC.
pub const STUDIO_32K: Self = Self {
codec: CodecId::Opus32k,
fec_ratio: 0.1,
frame_duration_ms: 20,
frames_per_block: 5,
};
/// Studio: Opus 48kbps, minimal FEC.
pub const STUDIO_48K: Self = Self {
codec: CodecId::Opus48k,
fec_ratio: 0.1,
frame_duration_ms: 20,
frames_per_block: 5,
};
/// Studio high: Opus 64kbps, minimal FEC.
pub const STUDIO_64K: Self = Self {
codec: CodecId::Opus64k,
fec_ratio: 0.1,
frame_duration_ms: 20,
frames_per_block: 5,
};
/// Estimated total bandwidth in kbps including FEC overhead.
pub fn total_bitrate_kbps(&self) -> f32 {
let base = self.codec.bitrate_bps() as f32 / 1000.0;

View File

@@ -1,4 +1,5 @@
use std::collections::BTreeMap;
use std::time::{Duration, Instant};
use crate::packet::MediaPacket;
@@ -20,19 +21,29 @@ pub struct AdaptivePlayoutDelay {
max_delay: usize,
/// Exponential moving average of inter-packet arrival jitter (ms).
jitter_ema: f64,
/// EMA smoothing factor (0.0-1.0, lower = smoother).
alpha: f64,
/// EMA smoothing factor for jitter increases (fast reaction).
alpha_up: f64,
/// EMA smoothing factor for jitter decreases (slow decay).
alpha_down: f64,
/// Last packet arrival timestamp (for computing inter-arrival jitter).
last_arrival_ms: Option<u64>,
/// Last packet expected timestamp.
last_expected_ms: Option<u64>,
/// Safety margin added to jitter-derived target (in packets).
safety_margin: f64,
/// Instant when a jitter spike was detected (handoff detection).
spike_detected_at: Option<Instant>,
/// Duration to hold max_delay after a spike is detected.
spike_cooldown: Duration,
/// Multiplier of jitter_ema that constitutes a spike.
spike_threshold_multiplier: f64,
}
/// Frame duration in milliseconds (20ms Opus/Codec2 frames).
const FRAME_DURATION_MS: f64 = 20.0;
/// Safety margin added to jitter-derived target (in packets).
const SAFETY_MARGIN_PACKETS: f64 = 2.0;
/// Default EMA smoothing factor.
/// Default safety margin in packets.
const DEFAULT_SAFETY_MARGIN: f64 = 2.0;
/// Default EMA smoothing factor (used for both up/down in non-mobile mode).
const DEFAULT_ALPHA: f64 = 0.05;
impl AdaptivePlayoutDelay {
@@ -46,9 +57,14 @@ impl AdaptivePlayoutDelay {
min_delay,
max_delay,
jitter_ema: 0.0,
alpha: DEFAULT_ALPHA,
alpha_up: DEFAULT_ALPHA,
alpha_down: DEFAULT_ALPHA,
last_arrival_ms: None,
last_expected_ms: None,
safety_margin: DEFAULT_SAFETY_MARGIN,
spike_detected_at: None,
spike_cooldown: Duration::from_secs(2),
spike_threshold_multiplier: 3.0,
}
}
@@ -64,13 +80,38 @@ impl AdaptivePlayoutDelay {
let expected_delta = expected_ms as f64 - last_expected as f64;
let jitter = (actual_delta - expected_delta).abs();
// Update EMA
self.jitter_ema = self.alpha * jitter + (1.0 - self.alpha) * self.jitter_ema;
// Spike detection: check before EMA update
if self.jitter_ema > 0.0
&& jitter > self.jitter_ema * self.spike_threshold_multiplier
{
self.spike_detected_at = Some(Instant::now());
}
// Convert jitter estimate to target delay in packets
let raw_target = (self.jitter_ema / FRAME_DURATION_MS).ceil() + SAFETY_MARGIN_PACKETS;
self.target_delay =
(raw_target as usize).clamp(self.min_delay, self.max_delay);
// Asymmetric EMA update
let alpha = if jitter > self.jitter_ema {
self.alpha_up
} else {
self.alpha_down
};
self.jitter_ema = alpha * jitter + (1.0 - alpha) * self.jitter_ema;
// Check if spike cooldown has expired
if let Some(spike_time) = self.spike_detected_at {
if spike_time.elapsed() >= self.spike_cooldown {
self.spike_detected_at = None;
}
}
// If within spike cooldown, return max_delay
if self.spike_detected_at.is_some() {
self.target_delay = self.max_delay;
} else {
// Convert jitter estimate to target delay in packets
let raw_target =
(self.jitter_ema / FRAME_DURATION_MS).ceil() + self.safety_margin;
self.target_delay =
(raw_target as usize).clamp(self.min_delay, self.max_delay);
}
}
self.last_arrival_ms = Some(arrival_ms);
@@ -87,6 +128,28 @@ impl AdaptivePlayoutDelay {
pub fn jitter_estimate_ms(&self) -> f64 {
self.jitter_ema
}
/// Enable or disable mobile mode, adjusting parameters for cellular networks.
///
/// Mobile mode uses:
/// - Asymmetric alpha (fast up=0.3, slow down=0.02) for quicker spike detection
/// - Higher safety margin (3.0 packets) to absorb handoff jitter
/// - Spike detection with 2-second cooldown at 3x threshold
pub fn set_mobile_mode(&mut self, enabled: bool) {
if enabled {
self.safety_margin = 3.0;
self.alpha_up = 0.3;
self.alpha_down = 0.02;
self.spike_threshold_multiplier = 3.0;
self.spike_cooldown = Duration::from_secs(2);
} else {
self.safety_margin = DEFAULT_SAFETY_MARGIN;
self.alpha_up = DEFAULT_ALPHA;
self.alpha_down = DEFAULT_ALPHA;
self.spike_threshold_multiplier = 3.0;
self.spike_cooldown = Duration::from_secs(2);
}
}
}
// ---------------------------------------------------------------------------
@@ -210,10 +273,21 @@ impl JitterBuffer {
return;
}
// Check if packet is too old (already played out)
// Check if packet is too old (already played out).
// A backward jump of >100 seq (~2s at 50fps) indicates a new sender in a
// federation room — reset instead of dropping.
if self.stats.packets_played > 0 && seq_before(seq, self.next_playout_seq) {
self.stats.packets_late += 1;
return;
let backward_distance = self.next_playout_seq.wrapping_sub(seq);
tracing::warn!(seq, next = self.next_playout_seq, backward_distance, "jitter: backward seq detected");
if backward_distance > 100 {
tracing::info!(seq, next = self.next_playout_seq, "jitter: RESET — new sender detected");
self.buffer.clear();
self.next_playout_seq = seq;
self.stats.packets_late = 0;
} else {
self.stats.packets_late += 1;
return;
}
}
// If we haven't started playout yet, adjust next_playout_seq to earliest known
@@ -349,10 +423,21 @@ impl JitterBuffer {
return;
}
// Check if packet is too old (already played out)
// Check if packet is too old (already played out).
// A backward jump of >100 seq (~2s at 50fps) indicates a new sender in a
// federation room — reset instead of dropping.
if self.stats.packets_played > 0 && seq_before(seq, self.next_playout_seq) {
self.stats.packets_late += 1;
return;
let backward_distance = self.next_playout_seq.wrapping_sub(seq);
tracing::warn!(seq, next = self.next_playout_seq, backward_distance, "jitter: backward seq detected");
if backward_distance > 100 {
tracing::info!(seq, next = self.next_playout_seq, "jitter: RESET — new sender detected");
self.buffer.clear();
self.next_playout_seq = seq;
self.stats.packets_late = 0;
} else {
self.stats.packets_late += 1;
return;
}
}
// If we haven't started playout yet, adjust next_playout_seq to earliest known
@@ -391,6 +476,11 @@ impl JitterBuffer {
self.adaptive.as_ref()
}
/// Get a mutable reference to the adaptive playout delay estimator.
pub fn adaptive_delay_mut(&mut self) -> Option<&mut AdaptivePlayoutDelay> {
self.adaptive.as_mut()
}
/// Adjust target depth based on observed jitter.
pub fn set_target_depth(&mut self, depth: usize) {
self.target_depth = depth.min(self.max_depth);
@@ -720,4 +810,29 @@ mod tests {
let ad = jb.adaptive_delay().unwrap();
assert_eq!(ad.target_delay(), 3);
}
// ---------------------------------------------------------------
// Mobile mode tests
// ---------------------------------------------------------------
#[test]
fn mobile_mode_increases_safety_margin() {
let mut apd = AdaptivePlayoutDelay::new(3, 50);
apd.set_mobile_mode(true);
assert_eq!(apd.safety_margin, 3.0);
assert_eq!(apd.alpha_up, 0.3);
assert_eq!(apd.alpha_down, 0.02);
apd.set_mobile_mode(false);
assert_eq!(apd.safety_margin, DEFAULT_SAFETY_MARGIN);
assert_eq!(apd.alpha_up, DEFAULT_ALPHA);
assert_eq!(apd.alpha_down, DEFAULT_ALPHA);
}
#[test]
fn mobile_mode_accessible_via_jitter_buffer() {
let mut jb = JitterBuffer::new_adaptive(3, 50);
jb.adaptive_delay_mut().unwrap().set_mobile_mode(true);
assert_eq!(jb.adaptive_delay().unwrap().safety_margin, 3.0);
}
}

View File

@@ -25,10 +25,11 @@ pub mod traits;
pub use codec_id::{CodecId, QualityProfile};
pub use error::*;
pub use packet::{
HangupReason, MediaHeader, MediaPacket, MiniFrameContext, MiniHeader, QualityReport,
SignalMessage, TrunkEntry, TrunkFrame, FRAME_TYPE_FULL, FRAME_TYPE_MINI,
CallAcceptMode, HangupReason, MediaHeader, MediaPacket, MiniFrameContext, MiniHeader,
QualityReport, RoomParticipant, SignalMessage, TrunkEntry, TrunkFrame, FRAME_TYPE_FULL,
FRAME_TYPE_MINI,
};
pub use bandwidth::{BandwidthEstimator, CongestionState};
pub use quality::{AdaptiveQualityController, Tier};
pub use quality::{AdaptiveQualityController, NetworkContext, Tier};
pub use session::{Session, SessionEvent, SessionState};
pub use traits::*;

View File

@@ -548,6 +548,9 @@ pub enum SignalMessage {
signature: Vec<u8>,
/// Supported quality profiles.
supported_profiles: Vec<crate::QualityProfile>,
/// Optional display name set by the caller.
#[serde(default)]
alias: Option<String>,
},
/// Call acceptance (analogous to Warzone's WireMessage::CallAnswer).
@@ -645,6 +648,133 @@ pub enum SignalMessage {
session_id: String,
room_name: String,
},
/// Room membership update — sent by relay to all participants when someone joins or leaves.
RoomUpdate {
/// Current participant count.
count: u32,
/// List of participants currently in the room.
participants: Vec<RoomParticipant>,
},
// ── Federation signals (relay-to-relay) ──
/// Federation: initial handshake — the connecting relay identifies itself.
FederationHello {
/// TLS certificate fingerprint of the connecting relay.
tls_fingerprint: String,
},
/// Federation: this relay now has local participants in a global room.
GlobalRoomActive {
room: String,
/// Participants on the announcing relay (for federated presence).
#[serde(default)]
participants: Vec<RoomParticipant>,
},
/// Federation: this relay's last local participant left a global room.
GlobalRoomInactive {
room: String,
},
// ── Direct calling signals (client ↔ relay signaling) ──
/// Register on relay for direct calls. Sent on `_signal` connections
/// after optional AuthToken.
RegisterPresence {
/// Client's Ed25519 identity public key.
identity_pub: [u8; 32],
/// Signature over ("register-presence" || identity_pub).
signature: Vec<u8>,
/// Optional display name.
alias: Option<String>,
},
/// Relay confirms presence registration.
RegisterPresenceAck {
success: bool,
#[serde(skip_serializing_if = "Option::is_none")]
error: Option<String>,
},
/// Direct call offer routed through the relay to a specific peer.
DirectCallOffer {
/// Caller's fingerprint.
caller_fingerprint: String,
/// Caller's display name.
caller_alias: Option<String>,
/// Target's fingerprint.
target_fingerprint: String,
/// Unique call session ID (UUID).
call_id: String,
/// Caller's Ed25519 identity pub.
identity_pub: [u8; 32],
/// Caller's ephemeral X25519 pub (for key exchange on media connect).
ephemeral_pub: [u8; 32],
/// Signature over (ephemeral_pub || target_fingerprint || call_id).
signature: Vec<u8>,
/// Supported quality profiles.
supported_profiles: Vec<crate::QualityProfile>,
},
/// Callee's response to a direct call.
DirectCallAnswer {
call_id: String,
/// How the callee accepts (or rejects).
accept_mode: CallAcceptMode,
/// Callee's identity pub (present when accepting).
#[serde(skip_serializing_if = "Option::is_none")]
identity_pub: Option<[u8; 32]>,
/// Callee's ephemeral pub (present when accepting).
#[serde(skip_serializing_if = "Option::is_none")]
ephemeral_pub: Option<[u8; 32]>,
/// Signature (present when accepting).
#[serde(skip_serializing_if = "Option::is_none")]
signature: Option<Vec<u8>>,
/// Chosen quality profile (present when accepting).
#[serde(skip_serializing_if = "Option::is_none")]
chosen_profile: Option<crate::QualityProfile>,
},
/// Relay tells both parties: media room is ready.
CallSetup {
call_id: String,
/// Room name on the relay for the media session (e.g., "_call:a1b2c3d4").
room: String,
/// Relay address for the QUIC media connection.
relay_addr: String,
},
/// Ringing notification (relay → caller, callee received the offer).
CallRinging {
call_id: String,
},
}
/// How the callee responds to a direct call.
#[derive(Clone, Copy, Debug, PartialEq, Eq, Serialize, Deserialize)]
pub enum CallAcceptMode {
/// Reject the call.
Reject,
/// Accept with trust — in Phase 2, this enables P2P (reveals IP).
/// In Phase 1, behaves the same as AcceptGeneric.
AcceptTrusted,
/// Accept with privacy — relay always mediates media.
AcceptGeneric,
}
/// A participant entry in a RoomUpdate message.
#[derive(Clone, Debug, Serialize, Deserialize)]
pub struct RoomParticipant {
/// Identity fingerprint (hex string, stable across reconnects if seed is persisted).
pub fingerprint: String,
/// Optional display name set by the client.
pub alias: Option<String>,
/// Relay label — identifies which relay this participant is connected to.
/// None for local participants, Some("Relay B") for federated.
#[serde(default)]
pub relay_label: Option<String>,
}
/// Reasons for ending a call.

View File

@@ -1,4 +1,5 @@
use std::collections::VecDeque;
use std::time::{Duration, Instant};
use crate::packet::QualityReport;
use crate::traits::QualityController;
@@ -24,24 +25,71 @@ impl Tier {
}
}
/// Determine which tier a quality report belongs to.
/// Determine which tier a quality report belongs to (default/WiFi thresholds).
pub fn classify(report: &QualityReport) -> Self {
Self::classify_with_context(report, NetworkContext::Unknown)
}
/// Classify with network-context-aware thresholds.
pub fn classify_with_context(report: &QualityReport, context: NetworkContext) -> Self {
let loss = report.loss_percent();
let rtt = report.rtt_ms();
if loss > 40.0 || rtt > 600 {
Self::Catastrophic
} else if loss > 10.0 || rtt > 400 {
Self::Degraded
} else {
Self::Good
match context {
NetworkContext::CellularLte
| NetworkContext::Cellular5g
| NetworkContext::Cellular3g => {
// Tighter thresholds for cellular networks
if loss > 25.0 || rtt > 500 {
Self::Catastrophic
} else if loss > 8.0 || rtt > 300 {
Self::Degraded
} else {
Self::Good
}
}
NetworkContext::WiFi | NetworkContext::Unknown => {
// Original thresholds
if loss > 40.0 || rtt > 600 {
Self::Catastrophic
} else if loss > 10.0 || rtt > 400 {
Self::Degraded
} else {
Self::Good
}
}
}
}
/// Return the next lower (worse) tier, or None if already at the worst.
pub fn downgrade(self) -> Option<Tier> {
match self {
Self::Good => Some(Self::Degraded),
Self::Degraded => Some(Self::Catastrophic),
Self::Catastrophic => None,
}
}
}
/// Describes the network transport type for context-aware quality decisions.
#[derive(Clone, Copy, Debug, PartialEq, Eq)]
pub enum NetworkContext {
WiFi,
CellularLte,
Cellular5g,
Cellular3g,
Unknown,
}
impl Default for NetworkContext {
fn default() -> Self {
Self::Unknown
}
}
/// Adaptive quality controller with hysteresis to prevent tier flapping.
///
/// - Downgrade: 3 consecutive reports in a worse tier
/// - Downgrade: 3 consecutive reports in a worse tier (2 on cellular)
/// - Upgrade: 10 consecutive reports in a better tier
pub struct AdaptiveQualityController {
current_tier: Tier,
@@ -54,14 +102,26 @@ pub struct AdaptiveQualityController {
history: VecDeque<QualityReport>,
/// Whether the profile was manually forced (disables adaptive logic).
forced: bool,
/// Current network context for threshold selection.
network_context: NetworkContext,
/// FEC boost expiry time (set during network handoff).
fec_boost_until: Option<Instant>,
/// FEC boost amount to add during handoff recovery window.
fec_boost_amount: f32,
}
/// Threshold for downgrading (fast reaction to degradation).
const DOWNGRADE_THRESHOLD: u32 = 3;
/// Threshold for downgrading on cellular networks (even faster).
const CELLULAR_DOWNGRADE_THRESHOLD: u32 = 2;
/// Threshold for upgrading (slow, cautious improvement).
const UPGRADE_THRESHOLD: u32 = 10;
/// Maximum history window size.
const HISTORY_SIZE: usize = 20;
/// Default FEC boost amount during handoff recovery.
const DEFAULT_FEC_BOOST: f32 = 0.2;
/// Duration of FEC boost after a network handoff.
const FEC_BOOST_DURATION_SECS: u64 = 10;
impl AdaptiveQualityController {
pub fn new() -> Self {
@@ -72,6 +132,9 @@ impl AdaptiveQualityController {
consecutive_down: 0,
history: VecDeque::with_capacity(HISTORY_SIZE),
forced: false,
network_context: NetworkContext::default(),
fec_boost_until: None,
fec_boost_amount: DEFAULT_FEC_BOOST,
}
}
@@ -80,6 +143,69 @@ impl AdaptiveQualityController {
self.current_tier
}
/// Get the current network context.
pub fn network_context(&self) -> NetworkContext {
self.network_context
}
/// Signal a network transport change (e.g., WiFi to cellular handoff).
///
/// When switching from WiFi to any cellular type, this preemptively
/// downgrades one quality tier and activates a temporary FEC boost.
pub fn signal_network_change(&mut self, new_context: NetworkContext) {
let old = self.network_context;
self.network_context = new_context;
let new_is_cellular = matches!(
new_context,
NetworkContext::CellularLte | NetworkContext::Cellular5g | NetworkContext::Cellular3g
);
// If switching from WiFi to cellular, preemptively downgrade one tier
if old == NetworkContext::WiFi && new_is_cellular {
if let Some(lower_tier) = self.current_tier.downgrade() {
self.current_tier = lower_tier;
self.current_profile = lower_tier.profile();
}
// Reset counters to avoid stale hysteresis state
self.consecutive_up = 0;
self.consecutive_down = 0;
// Un-force so adaptive logic resumes
self.forced = false;
}
// Activate FEC boost for any network change
self.fec_boost_until = Some(Instant::now() + Duration::from_secs(FEC_BOOST_DURATION_SECS));
}
/// Returns the FEC boost amount if within the handoff recovery window, 0.0 otherwise.
///
/// Callers should add this to their base FEC ratio during the boost window.
pub fn fec_boost(&self) -> f32 {
if let Some(until) = self.fec_boost_until {
if Instant::now() < until {
return self.fec_boost_amount;
}
}
0.0
}
/// Reset the hysteresis counters.
pub fn reset_counters(&mut self) {
self.consecutive_up = 0;
self.consecutive_down = 0;
}
/// Get the effective downgrade threshold based on network context.
fn downgrade_threshold(&self) -> u32 {
match self.network_context {
NetworkContext::CellularLte
| NetworkContext::Cellular5g
| NetworkContext::Cellular3g => CELLULAR_DOWNGRADE_THRESHOLD,
_ => DOWNGRADE_THRESHOLD,
}
}
fn try_transition(&mut self, observed_tier: Tier) -> Option<QualityProfile> {
if observed_tier == self.current_tier {
self.consecutive_up = 0;
@@ -96,7 +222,7 @@ impl AdaptiveQualityController {
if is_worse {
self.consecutive_up = 0;
self.consecutive_down += 1;
if self.consecutive_down >= DOWNGRADE_THRESHOLD {
if self.consecutive_down >= self.downgrade_threshold() {
self.current_tier = observed_tier;
self.current_profile = observed_tier.profile();
self.consecutive_down = 0;
@@ -142,7 +268,7 @@ impl QualityController for AdaptiveQualityController {
return None;
}
let observed = Tier::classify(report);
let observed = Tier::classify_with_context(report, self.network_context);
self.try_transition(observed)
}
@@ -246,4 +372,110 @@ mod tests {
assert_eq!(Tier::classify(&make_report(50.0, 200)), Tier::Catastrophic);
assert_eq!(Tier::classify(&make_report(5.0, 700)), Tier::Catastrophic);
}
// ---------------------------------------------------------------
// Network context tests
// ---------------------------------------------------------------
#[test]
fn cellular_tighter_thresholds() {
// 12% loss: Good on WiFi, Degraded on cellular
let report = make_report(12.0, 200);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Degraded
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::CellularLte),
Tier::Degraded
);
// 9% loss: Good on WiFi, Degraded on cellular
let report = make_report(9.0, 200);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Good
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::CellularLte),
Tier::Degraded
);
// 30% loss: Degraded on WiFi, Catastrophic on cellular
let report = make_report(30.0, 200);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Degraded
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::Cellular3g),
Tier::Catastrophic
);
}
#[test]
fn cellular_rtt_thresholds() {
// RTT 350ms: Good on WiFi, Degraded on cellular
let report = make_report(2.0, 348); // rtt_4ms rounds so use 348
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::WiFi),
Tier::Good
);
assert_eq!(
Tier::classify_with_context(&report, NetworkContext::CellularLte),
Tier::Degraded
);
}
#[test]
fn cellular_faster_downgrade() {
let mut ctrl = AdaptiveQualityController::new();
ctrl.signal_network_change(NetworkContext::CellularLte);
// Reset tier back to Good for testing downgrade threshold
ctrl.current_tier = Tier::Good;
ctrl.current_profile = Tier::Good.profile();
// On cellular, downgrade threshold is 2 instead of 3
let bad = make_report(50.0, 200);
assert!(ctrl.observe(&bad).is_none()); // 1st bad
let result = ctrl.observe(&bad); // 2nd bad — should trigger on cellular
assert!(result.is_some());
}
#[test]
fn signal_network_change_preemptive_downgrade() {
let mut ctrl = AdaptiveQualityController::new();
assert_eq!(ctrl.tier(), Tier::Good);
// Switch from WiFi to cellular
ctrl.network_context = NetworkContext::WiFi;
ctrl.signal_network_change(NetworkContext::CellularLte);
// Should have downgraded one tier: Good -> Degraded
assert_eq!(ctrl.tier(), Tier::Degraded);
}
#[test]
fn signal_network_change_fec_boost() {
let mut ctrl = AdaptiveQualityController::new();
assert_eq!(ctrl.fec_boost(), 0.0);
ctrl.signal_network_change(NetworkContext::CellularLte);
// FEC boost should be active
assert!(ctrl.fec_boost() > 0.0);
assert_eq!(ctrl.fec_boost(), DEFAULT_FEC_BOOST);
}
#[test]
fn tier_downgrade() {
assert_eq!(Tier::Good.downgrade(), Some(Tier::Degraded));
assert_eq!(Tier::Degraded.downgrade(), Some(Tier::Catastrophic));
assert_eq!(Tier::Catastrophic.downgrade(), None);
}
#[test]
fn network_context_default() {
assert_eq!(NetworkContext::default(), NetworkContext::Unknown);
}
}

View File

@@ -132,6 +132,14 @@ pub trait CryptoSession: Send + Sync {
fn overhead(&self) -> usize {
16 // ChaCha20-Poly1305 tag
}
/// Short Authentication String (SAS) — 4-digit code for verbal verification.
/// Both peers derive the same code from the shared secret + identity keys.
/// If a MITM relay is intercepting, the codes will differ.
/// Returns None if SAS was not computed (e.g., relay-side sessions).
fn sas_code(&self) -> Option<u32> {
None
}
}
/// Key exchange using the Warzone identity model.

View File

@@ -28,6 +28,9 @@ prometheus = "0.13"
axum = { version = "0.7", default-features = false, features = ["tokio", "http1", "ws"] }
tower-http = { version = "0.6", features = ["fs"] }
futures-util = "0.3"
dirs = "6"
sha2 = { workspace = true }
chrono = "0.4"
[[bin]]
name = "wzp-relay"

18
crates/wzp-relay/build.rs Normal file
View File

@@ -0,0 +1,18 @@
use std::process::Command;
fn main() {
// Get git hash at build time
let output = Command::new("git")
.args(["rev-parse", "--short", "HEAD"])
.output();
let hash = match output {
Ok(o) if o.status.success() => {
String::from_utf8_lossy(&o.stdout).trim().to_string()
}
_ => "unknown".to_string(),
};
println!("cargo:rustc-env=WZP_BUILD_HASH={hash}");
println!("cargo:rerun-if-changed=.git/HEAD");
}

View File

@@ -0,0 +1,199 @@
//! Direct call state tracking.
//!
//! Manages the lifecycle of 1:1 direct calls placed via the `_signal` channel.
//! Each call goes through: Pending → Ringing → Active → Ended.
use std::collections::HashMap;
use std::time::{Duration, Instant};
/// State of a direct call.
#[derive(Clone, Copy, Debug, PartialEq, Eq)]
pub enum DirectCallState {
/// Offer sent to callee, waiting for response.
Pending,
/// Callee acknowledged, ringing.
Ringing,
/// Call accepted, media room active.
Active,
/// Call ended (hangup, reject, timeout, or error).
Ended,
}
/// A tracked direct call between two users.
pub struct DirectCall {
pub call_id: String,
pub caller_fingerprint: String,
pub callee_fingerprint: String,
pub state: DirectCallState,
pub accept_mode: Option<wzp_proto::CallAcceptMode>,
/// Private room name (set when accepted).
pub room_name: Option<String>,
pub created_at: Instant,
pub answered_at: Option<Instant>,
pub ended_at: Option<Instant>,
}
/// Registry of active direct calls.
pub struct CallRegistry {
calls: HashMap<String, DirectCall>,
}
impl CallRegistry {
pub fn new() -> Self {
Self {
calls: HashMap::new(),
}
}
/// Create a new pending call. Returns the call_id.
pub fn create_call(&mut self, call_id: String, caller_fp: String, callee_fp: String) -> &DirectCall {
let call = DirectCall {
call_id: call_id.clone(),
caller_fingerprint: caller_fp,
callee_fingerprint: callee_fp,
state: DirectCallState::Pending,
accept_mode: None,
room_name: None,
created_at: Instant::now(),
answered_at: None,
ended_at: None,
};
self.calls.insert(call_id.clone(), call);
self.calls.get(&call_id).unwrap()
}
/// Get a call by ID.
pub fn get(&self, call_id: &str) -> Option<&DirectCall> {
self.calls.get(call_id)
}
/// Get a mutable call by ID.
pub fn get_mut(&mut self, call_id: &str) -> Option<&mut DirectCall> {
self.calls.get_mut(call_id)
}
/// Transition to Ringing state.
pub fn set_ringing(&mut self, call_id: &str) -> bool {
if let Some(call) = self.calls.get_mut(call_id) {
if call.state == DirectCallState::Pending {
call.state = DirectCallState::Ringing;
return true;
}
}
false
}
/// Transition to Active state.
pub fn set_active(&mut self, call_id: &str, mode: wzp_proto::CallAcceptMode, room: String) -> bool {
if let Some(call) = self.calls.get_mut(call_id) {
if call.state == DirectCallState::Pending || call.state == DirectCallState::Ringing {
call.state = DirectCallState::Active;
call.accept_mode = Some(mode);
call.room_name = Some(room);
call.answered_at = Some(Instant::now());
return true;
}
}
false
}
/// End a call.
pub fn end_call(&mut self, call_id: &str) -> Option<DirectCall> {
if let Some(call) = self.calls.get_mut(call_id) {
call.state = DirectCallState::Ended;
call.ended_at = Some(Instant::now());
}
self.calls.remove(call_id)
}
/// Find active/pending calls involving a fingerprint.
pub fn calls_for_fingerprint(&self, fp: &str) -> Vec<&DirectCall> {
self.calls.values()
.filter(|c| {
c.state != DirectCallState::Ended
&& (c.caller_fingerprint == fp || c.callee_fingerprint == fp)
})
.collect()
}
/// Find the peer's fingerprint in a call.
pub fn peer_fingerprint(&self, call_id: &str, my_fp: &str) -> Option<&str> {
self.calls.get(call_id).map(|c| {
if c.caller_fingerprint == my_fp {
c.callee_fingerprint.as_str()
} else {
c.caller_fingerprint.as_str()
}
})
}
/// Remove calls that have been pending longer than the timeout.
/// Returns call IDs of expired calls.
pub fn expire_stale(&mut self, timeout: Duration) -> Vec<DirectCall> {
let now = Instant::now();
let expired: Vec<String> = self.calls.iter()
.filter(|(_, c)| {
c.state == DirectCallState::Pending
&& now.duration_since(c.created_at) > timeout
})
.map(|(id, _)| id.clone())
.collect();
expired.into_iter()
.filter_map(|id| self.calls.remove(&id))
.collect()
}
/// Number of active (non-ended) calls.
pub fn active_count(&self) -> usize {
self.calls.values()
.filter(|c| c.state != DirectCallState::Ended)
.count()
}
}
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn call_lifecycle() {
let mut reg = CallRegistry::new();
reg.create_call("c1".into(), "alice".into(), "bob".into());
assert_eq!(reg.get("c1").unwrap().state, DirectCallState::Pending);
assert!(reg.set_ringing("c1"));
assert_eq!(reg.get("c1").unwrap().state, DirectCallState::Ringing);
assert!(reg.set_active("c1", wzp_proto::CallAcceptMode::AcceptGeneric, "_call:c1".into()));
assert_eq!(reg.get("c1").unwrap().state, DirectCallState::Active);
assert_eq!(reg.get("c1").unwrap().room_name.as_deref(), Some("_call:c1"));
let ended = reg.end_call("c1").unwrap();
assert_eq!(ended.state, DirectCallState::Ended);
assert_eq!(reg.active_count(), 0);
}
#[test]
fn expire_stale_calls() {
let mut reg = CallRegistry::new();
reg.create_call("c1".into(), "alice".into(), "bob".into());
// Not expired yet
let expired = reg.expire_stale(Duration::from_secs(30));
assert!(expired.is_empty());
// Force expiry with 0 timeout
let expired = reg.expire_stale(Duration::from_secs(0));
assert_eq!(expired.len(), 1);
assert_eq!(expired[0].call_id, "c1");
}
#[test]
fn peer_lookup() {
let mut reg = CallRegistry::new();
reg.create_call("c1".into(), "alice".into(), "bob".into());
assert_eq!(reg.peer_fingerprint("c1", "alice"), Some("bob"));
assert_eq!(reg.peer_fingerprint("c1", "bob"), Some("alice"));
}
}

View File

@@ -3,8 +3,41 @@
use serde::{Deserialize, Serialize};
use std::net::SocketAddr;
/// Configuration for the relay daemon.
/// A federated peer relay.
#[derive(Clone, Debug, Serialize, Deserialize)]
pub struct PeerConfig {
/// Address of the peer relay (e.g., "193.180.213.68:4433").
pub url: String,
/// Expected TLS certificate fingerprint (hex, with colons).
pub fingerprint: String,
/// Optional human-readable label.
#[serde(default)]
pub label: Option<String>,
}
/// A trusted relay — accepts inbound federation without needing the peer's address.
#[derive(Clone, Debug, Serialize, Deserialize)]
pub struct TrustedConfig {
/// Expected TLS certificate fingerprint (hex, with colons).
pub fingerprint: String,
/// Optional human-readable label.
#[serde(default)]
pub label: Option<String>,
}
/// A room declared global — bridged across all federated peers.
#[derive(Clone, Debug, Serialize, Deserialize)]
pub struct GlobalRoomConfig {
/// Room name to bridge (e.g., "android").
pub name: String,
}
/// Configuration for the relay daemon.
///
/// All fields have defaults, so a minimal TOML file only needs the
/// fields you want to override (e.g., just `[[peers]]`).
#[derive(Clone, Debug, Serialize, Deserialize)]
#[serde(default)]
pub struct RelayConfig {
/// Address to listen on for incoming connections (client-facing).
pub listen_addr: SocketAddr,
@@ -44,6 +77,22 @@ pub struct RelayConfig {
pub ws_port: Option<u16>,
/// Directory to serve static files from (HTML/JS/WASM for web clients).
pub static_dir: Option<String>,
/// Federation peer relays.
#[serde(default)]
pub peers: Vec<PeerConfig>,
/// Global rooms bridged across federation.
#[serde(default)]
pub global_rooms: Vec<GlobalRoomConfig>,
/// Trusted relay fingerprints — accept inbound federation from these relays.
/// Unlike [[peers]], no url is needed — the peer connects to us.
#[serde(default)]
pub trusted: Vec<TrustedConfig>,
/// Debug tap: log packet headers for matching rooms ("*" = all rooms).
/// Activated via --debug-tap <room> or debug_tap = "room" in TOML.
pub debug_tap: Option<String>,
/// JSONL event log path for protocol analysis (--event-log).
#[serde(skip)]
pub event_log: Option<String>,
}
impl Default for RelayConfig {
@@ -62,6 +111,100 @@ impl Default for RelayConfig {
trunking_enabled: false,
ws_port: None,
static_dir: None,
peers: Vec::new(),
global_rooms: Vec::new(),
trusted: Vec::new(),
debug_tap: None,
event_log: None,
}
}
}
/// Load relay configuration from a TOML file.
pub fn load_config(path: &str) -> Result<RelayConfig, anyhow::Error> {
let content = std::fs::read_to_string(path)?;
let config: RelayConfig = toml::from_str(&content)?;
Ok(config)
}
/// Info about this relay instance, used to generate personalized example configs.
pub struct RelayInfo {
pub listen_addr: String,
pub tls_fingerprint: String,
pub public_ip: Option<String>,
}
/// Load config from path, or create a personalized example config if it doesn't exist.
pub fn load_or_create_config(path: &str, info: Option<&RelayInfo>) -> Result<RelayConfig, anyhow::Error> {
let p = std::path::Path::new(path);
if p.exists() {
return load_config(path);
}
// Create parent directory if needed
if let Some(parent) = p.parent() {
std::fs::create_dir_all(parent)?;
}
// Generate personalized example config
let example = generate_example_config(info);
std::fs::write(p, &example)?;
eprintln!("Created example config at {path} — edit it and restart.");
let config: RelayConfig = toml::from_str(&example)?;
Ok(config)
}
/// Generate an example TOML config, personalized with this relay's info if available.
fn generate_example_config(info: Option<&RelayInfo>) -> String {
let listen = info.map(|i| i.listen_addr.as_str()).unwrap_or("0.0.0.0:4433");
let peer_example = if let Some(i) = info {
let ip = i.public_ip.as_deref().unwrap_or("this-relay-ip");
format!(
r#"# Other relays can peer with this relay using:
# [[peers]]
# url = "{ip}:{port}"
# fingerprint = "{fp}"
# label = "This Relay""#,
port = listen.rsplit(':').next().unwrap_or("4433"),
fp = i.tls_fingerprint,
)
} else {
"# To peer with another relay, add its url + fingerprint:".to_string()
};
format!(
r#"# WarzonePhone Relay Configuration
# See docs/ADMINISTRATION.md for full reference.
# Listen address for client connections
listen_addr = "{listen}"
# Maximum concurrent sessions
# max_sessions = 100
# Prometheus metrics endpoint (uncomment to enable)
# metrics_port = 9090
# featherChat auth endpoint (uncomment to enable)
# auth_url = "https://chat.example.com/v1/auth/validate"
{peer_example}
# Federation: peer relays we connect to (outbound)
# [[peers]]
# url = "other-relay.example.com:4433"
# fingerprint = "aa:bb:cc:dd:..."
# label = "Relay B"
# Federation: relays we trust inbound connections from
# [[trusted]]
# fingerprint = "ee:ff:00:11:..."
# label = "Relay X"
# Global rooms bridged across all federated peers
# [[global_rooms]]
# name = "general"
# Debug: log packet headers for a room ("*" for all)
# debug_tap = "*"
"#
)
}

View File

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//! JSONL event log for protocol analysis.
//!
//! When `--event-log <path>` is set, every media packet emits a structured
//! event at each decision point (recv, forward, drop, deliver).
//! Use `wzp-analyzer` to correlate events across multiple relays.
use std::path::PathBuf;
use std::sync::Arc;
use serde::Serialize;
use tokio::sync::mpsc;
use tracing::{error, info};
/// A single protocol event for JSONL output.
#[derive(Debug, Serialize)]
pub struct Event {
/// ISO 8601 timestamp with microseconds.
pub ts: String,
/// Event type.
pub event: &'static str,
/// Room name.
#[serde(skip_serializing_if = "Option::is_none")]
pub room: Option<String>,
/// Source address or peer label.
#[serde(skip_serializing_if = "Option::is_none")]
pub src: Option<String>,
/// Packet sequence number.
#[serde(skip_serializing_if = "Option::is_none")]
pub seq: Option<u16>,
/// Codec identifier.
#[serde(skip_serializing_if = "Option::is_none")]
pub codec: Option<String>,
/// FEC block ID.
#[serde(skip_serializing_if = "Option::is_none")]
pub fec_block: Option<u8>,
/// FEC symbol index.
#[serde(skip_serializing_if = "Option::is_none")]
pub fec_sym: Option<u8>,
/// Is FEC repair packet.
#[serde(skip_serializing_if = "Option::is_none")]
pub repair: Option<bool>,
/// Payload length in bytes.
#[serde(skip_serializing_if = "Option::is_none")]
pub len: Option<usize>,
/// Number of recipients.
#[serde(skip_serializing_if = "Option::is_none")]
pub to_count: Option<usize>,
/// Peer label (for federation events).
#[serde(skip_serializing_if = "Option::is_none")]
pub peer: Option<String>,
/// Drop/error reason.
#[serde(skip_serializing_if = "Option::is_none")]
pub reason: Option<String>,
/// Presence action (active/inactive).
#[serde(skip_serializing_if = "Option::is_none")]
pub action: Option<String>,
/// Participant count (presence events).
#[serde(skip_serializing_if = "Option::is_none")]
pub participants: Option<usize>,
}
impl Event {
fn now() -> String {
chrono::Utc::now().format("%Y-%m-%dT%H:%M:%S%.6fZ").to_string()
}
/// Create a minimal event with just type and timestamp.
pub fn new(event: &'static str) -> Self {
Self {
ts: Self::now(),
event,
room: None,
src: None,
seq: None,
codec: None,
fec_block: None,
fec_sym: None,
repair: None,
len: None,
to_count: None,
peer: None,
reason: None,
action: None,
participants: None,
}
}
/// Set room.
pub fn room(mut self, room: &str) -> Self { self.room = Some(room.to_string()); self }
/// Set source.
pub fn src(mut self, src: &str) -> Self { self.src = Some(src.to_string()); self }
/// Set packet header fields from a MediaPacket.
pub fn packet(mut self, pkt: &wzp_proto::MediaPacket) -> Self {
self.seq = Some(pkt.header.seq);
self.codec = Some(format!("{:?}", pkt.header.codec_id));
self.fec_block = Some(pkt.header.fec_block);
self.fec_sym = Some(pkt.header.fec_symbol);
self.repair = Some(pkt.header.is_repair);
self.len = Some(pkt.payload.len());
self
}
/// Set seq only (when full packet not available).
pub fn seq(mut self, seq: u16) -> Self { self.seq = Some(seq); self }
/// Set payload length.
pub fn len(mut self, len: usize) -> Self { self.len = Some(len); self }
/// Set recipient count.
pub fn to_count(mut self, n: usize) -> Self { self.to_count = Some(n); self }
/// Set peer label.
pub fn peer(mut self, peer: &str) -> Self { self.peer = Some(peer.to_string()); self }
/// Set drop reason.
pub fn reason(mut self, reason: &str) -> Self { self.reason = Some(reason.to_string()); self }
/// Set presence action.
pub fn action(mut self, action: &str) -> Self { self.action = Some(action.to_string()); self }
/// Set participant count.
pub fn participants(mut self, n: usize) -> Self { self.participants = Some(n); self }
}
/// Handle for emitting events. Cheap to clone.
#[derive(Clone)]
pub struct EventLog {
tx: mpsc::UnboundedSender<Event>,
}
impl EventLog {
/// Emit an event (non-blocking, drops if channel is full).
pub fn emit(&self, event: Event) {
let _ = self.tx.send(event);
}
}
/// No-op event log for when `--event-log` is not set.
/// All methods are no-ops that compile to nothing.
#[derive(Clone)]
pub struct NoopEventLog;
/// Unified event log handle — either real or no-op.
#[derive(Clone)]
pub enum EventLogger {
Active(EventLog),
Noop,
}
impl EventLogger {
pub fn emit(&self, event: Event) {
if let EventLogger::Active(log) = self {
log.emit(event);
}
}
pub fn is_active(&self) -> bool {
matches!(self, EventLogger::Active(_))
}
}
/// Start the event log writer. Returns an `EventLogger` handle.
pub fn start_event_log(path: Option<PathBuf>) -> EventLogger {
match path {
Some(path) => {
let (tx, rx) = mpsc::unbounded_channel();
tokio::spawn(writer_task(path, rx));
info!("event log enabled");
EventLogger::Active(EventLog { tx })
}
None => EventLogger::Noop,
}
}
/// Background task that writes events to a JSONL file.
async fn writer_task(path: PathBuf, mut rx: mpsc::UnboundedReceiver<Event>) {
use tokio::io::AsyncWriteExt;
let file = match tokio::fs::File::create(&path).await {
Ok(f) => f,
Err(e) => {
error!("failed to create event log {}: {e}", path.display());
return;
}
};
let mut writer = tokio::io::BufWriter::new(file);
let mut count: u64 = 0;
while let Some(event) = rx.recv().await {
match serde_json::to_string(&event) {
Ok(json) => {
if writer.write_all(json.as_bytes()).await.is_err() { break; }
if writer.write_all(b"\n").await.is_err() { break; }
count += 1;
// Flush every 100 events
if count % 100 == 0 {
let _ = writer.flush().await;
}
}
Err(e) => {
error!("event log serialize error: {e}");
}
}
}
let _ = writer.flush().await;
info!(events = count, "event log closed");
}

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//! Relay federation — global room routing between peer relays.
//!
//! Each relay maintains a forwarding table per global room. When a local participant
//! sends media in a global room, it's forwarded to all peer relays that have the room
//! active. Incoming federated media is delivered to local participants and optionally
//! forwarded to other active peers (multi-hop).
use std::collections::{HashMap, HashSet};
use std::net::SocketAddr;
use std::sync::Arc;
use std::time::{Duration, Instant};
use bytes::Bytes;
use sha2::{Sha256, Digest};
use tokio::sync::Mutex;
use tracing::{error, info, warn};
use wzp_proto::{MediaTransport, SignalMessage};
use wzp_transport::QuinnTransport;
use crate::config::{PeerConfig, TrustedConfig};
use crate::event_log::{Event, EventLogger};
use crate::room::{self, FederationMediaOut, RoomEvent, RoomManager};
/// Compute 8-byte room hash for federation datagram tagging.
pub fn room_hash(room_name: &str) -> [u8; 8] {
let h = Sha256::digest(room_name.as_bytes());
let mut out = [0u8; 8];
out.copy_from_slice(&h[..8]);
out
}
/// Normalize a fingerprint string (remove colons, lowercase).
fn normalize_fp(fp: &str) -> String {
fp.replace(':', "").to_lowercase()
}
/// Time-based dedup filter for federation datagrams.
/// Tracks recently seen packets and expires entries older than 2 seconds.
/// This prevents duplicate delivery when the same packet arrives via
/// multiple federation paths, while allowing new senders that happen to
/// reuse the same seq numbers.
struct Deduplicator {
/// Recently seen packet keys with insertion time.
entries: HashMap<u64, Instant>,
/// Expiry duration.
ttl: Duration,
}
impl Deduplicator {
fn new(_capacity: usize) -> Self {
Self {
entries: HashMap::with_capacity(512),
ttl: Duration::from_secs(2),
}
}
/// Returns true if this packet is a duplicate (already seen within TTL).
fn is_dup(&mut self, room_hash: &[u8; 8], seq: u16, extra: u64) -> bool {
let key = u64::from_be_bytes(*room_hash) ^ (seq as u64) ^ extra;
let now = Instant::now();
// Periodic cleanup (every ~256 packets)
if self.entries.len() > 256 {
self.entries.retain(|_, ts| now.duration_since(*ts) < self.ttl);
}
if let Some(ts) = self.entries.get(&key) {
if now.duration_since(*ts) < self.ttl {
return true; // seen recently — duplicate
}
}
self.entries.insert(key, now);
false
}
}
/// Per-room token bucket rate limiter for federation forwarding.
struct RateLimiter {
/// Max packets per second per room.
max_pps: u32,
/// Tokens remaining in current window.
tokens: u32,
/// When the current window started.
window_start: Instant,
}
impl RateLimiter {
fn new(max_pps: u32) -> Self {
Self {
max_pps,
tokens: max_pps,
window_start: Instant::now(),
}
}
/// Returns true if the packet should be allowed through.
fn allow(&mut self) -> bool {
let elapsed = self.window_start.elapsed();
if elapsed >= Duration::from_secs(1) {
self.tokens = self.max_pps;
self.window_start = Instant::now();
}
if self.tokens > 0 {
self.tokens -= 1;
true
} else {
false
}
}
}
/// Active link to a peer relay.
struct PeerLink {
transport: Arc<QuinnTransport>,
label: String,
/// Global rooms that this peer has reported as active.
active_rooms: HashSet<String>,
/// Remote participants per room (for federated presence in RoomUpdate).
remote_participants: HashMap<String, Vec<wzp_proto::packet::RoomParticipant>>,
/// Last time we received any data (signal or media) from this peer.
last_seen: Instant,
}
/// Max federation packets per second per room (0 = unlimited).
const FEDERATION_RATE_LIMIT_PPS: u32 = 500;
/// Dedup window size (number of recent packets to remember).
const DEDUP_WINDOW_SIZE: usize = 4096;
/// Remote participants are considered stale after this duration with no updates.
const REMOTE_PARTICIPANT_STALE_SECS: u64 = 15;
/// Manages federation connections and global room forwarding.
pub struct FederationManager {
peers: Vec<PeerConfig>,
trusted: Vec<TrustedConfig>,
global_rooms: HashSet<String>,
room_mgr: Arc<Mutex<RoomManager>>,
endpoint: quinn::Endpoint,
local_tls_fp: String,
metrics: Arc<crate::metrics::RelayMetrics>,
/// Active peer connections, keyed by normalized fingerprint.
peer_links: Arc<Mutex<HashMap<String, PeerLink>>>,
/// Dedup filter for incoming federation datagrams.
dedup: Mutex<Deduplicator>,
/// Per-room seq counter for federation media delivered to local clients.
/// Ensures clients see monotonically increasing seq regardless of federation sender.
local_delivery_seq: std::sync::atomic::AtomicU16,
/// JSONL event log for protocol analysis.
event_log: EventLogger,
/// Per-room rate limiters for inbound federation media.
rate_limiters: Mutex<HashMap<String, RateLimiter>>,
}
impl FederationManager {
pub fn new(
peers: Vec<PeerConfig>,
trusted: Vec<TrustedConfig>,
global_rooms: HashSet<String>,
room_mgr: Arc<Mutex<RoomManager>>,
endpoint: quinn::Endpoint,
local_tls_fp: String,
metrics: Arc<crate::metrics::RelayMetrics>,
event_log: EventLogger,
) -> Self {
Self {
peers,
trusted,
global_rooms,
room_mgr,
endpoint,
local_tls_fp,
metrics,
peer_links: Arc::new(Mutex::new(HashMap::new())),
dedup: Mutex::new(Deduplicator::new(DEDUP_WINDOW_SIZE)),
local_delivery_seq: std::sync::atomic::AtomicU16::new(0),
event_log,
rate_limiters: Mutex::new(HashMap::new()),
}
}
/// Check if a room name (which may be hashed) is a global room.
pub fn is_global_room(&self, room: &str) -> bool {
self.resolve_global_room(room).is_some()
}
/// Resolve a room name (raw or hashed) to the canonical global room name.
/// Returns the configured global room name if it matches.
pub fn resolve_global_room(&self, room: &str) -> Option<&str> {
// Direct match (raw room name, e.g. Android clients)
if self.global_rooms.contains(room) {
return Some(self.global_rooms.iter().find(|n| n.as_str() == room).unwrap());
}
// Hashed match (desktop clients hash room names for SNI privacy)
self.global_rooms.iter().find(|name| {
wzp_crypto::hash_room_name(name) == room
}).map(|s| s.as_str())
}
/// Get the canonical federation room hash for a room.
/// Always uses the configured global room name, not the client-provided name.
pub fn global_room_hash(&self, room: &str) -> [u8; 8] {
if let Some(canonical) = self.resolve_global_room(room) {
room_hash(canonical)
} else {
room_hash(room)
}
}
/// Start federation — spawns connection loops + event dispatcher.
pub async fn run(self: Arc<Self>) {
if self.peers.is_empty() && self.global_rooms.is_empty() {
return;
}
info!(
peers = self.peers.len(),
global_rooms = self.global_rooms.len(),
"federation starting"
);
let mut handles = Vec::new();
// Per-peer outbound connection loops
for peer in &self.peers {
let this = self.clone();
let peer = peer.clone();
handles.push(tokio::spawn(async move {
run_peer_loop(this, peer).await;
}));
}
// Room event dispatcher
let room_events = {
let mgr = self.room_mgr.lock().await;
mgr.subscribe_events()
};
let this = self.clone();
handles.push(tokio::spawn(async move {
run_room_event_dispatcher(this, room_events).await;
}));
// Stale presence sweeper — purges remote participants from dead peers
let this = self.clone();
handles.push(tokio::spawn(async move {
run_stale_presence_sweeper(this).await;
}));
for h in handles {
let _ = h.await;
}
}
/// Handle an inbound federation connection from a recognized peer.
pub async fn handle_inbound(
self: &Arc<Self>,
transport: Arc<QuinnTransport>,
peer_config: PeerConfig,
) {
let peer_fp = normalize_fp(&peer_config.fingerprint);
let label = peer_config.label.unwrap_or_else(|| peer_config.url.clone());
info!(peer = %label, "inbound federation link active");
if let Err(e) = run_federation_link(self.clone(), transport, peer_fp, label.clone()).await {
warn!(peer = %label, "inbound federation link ended: {e}");
}
}
/// Get all remote participants for a room from all peer links.
/// Deduplicates by fingerprint (same participant may appear via multiple links).
pub async fn get_remote_participants(&self, room: &str) -> Vec<wzp_proto::packet::RoomParticipant> {
let canonical = self.resolve_global_room(room);
let links = self.peer_links.lock().await;
let mut result = Vec::new();
for link in links.values() {
// Check canonical name
if let Some(c) = canonical {
if let Some(remote) = link.remote_participants.get(c) {
result.extend(remote.iter().cloned());
}
// Also check raw room name, but only if different from canonical
if c != room {
if let Some(remote) = link.remote_participants.get(room) {
result.extend(remote.iter().cloned());
}
}
} else {
if let Some(remote) = link.remote_participants.get(room) {
result.extend(remote.iter().cloned());
}
}
}
// Deduplicate by fingerprint
let mut seen = HashSet::new();
result.retain(|p| seen.insert(p.fingerprint.clone()));
result
}
/// Forward locally-generated media to all connected peers.
/// For locally-originated media, we send to ALL peers (they decide whether to deliver).
/// For forwarded media (multi-hop), handle_datagram filters by active_rooms.
pub async fn forward_to_peers(&self, room_name: &str, room_hash: &[u8; 8], media_data: &Bytes) {
let links = self.peer_links.lock().await;
if links.is_empty() {
return;
}
for (_fp, link) in links.iter() {
let mut tagged = Vec::with_capacity(8 + media_data.len());
tagged.extend_from_slice(room_hash);
tagged.extend_from_slice(media_data);
match link.transport.send_raw_datagram(&tagged) {
Ok(()) => {
self.metrics.federation_packets_forwarded
.with_label_values(&[&link.label, "out"]).inc();
}
Err(e) => warn!(peer = %link.label, "federation send error: {e}"),
}
}
}
// ── Trust verification (kept from previous implementation) ──
pub fn find_peer_by_fingerprint(&self, fp: &str) -> Option<&PeerConfig> {
self.peers.iter().find(|p| normalize_fp(&p.fingerprint) == normalize_fp(fp))
}
pub fn find_peer_by_addr(&self, addr: SocketAddr) -> Option<&PeerConfig> {
let addr_ip = addr.ip();
self.peers.iter().find(|p| {
p.url.parse::<SocketAddr>()
.map(|sa| sa.ip() == addr_ip)
.unwrap_or(false)
})
}
pub fn find_trusted_by_fingerprint(&self, fp: &str) -> Option<&TrustedConfig> {
self.trusted.iter().find(|t| normalize_fp(&t.fingerprint) == normalize_fp(fp))
}
pub fn check_inbound_trust(&self, addr: SocketAddr, hello_fp: &str) -> Option<String> {
if let Some(peer) = self.find_peer_by_addr(addr) {
return Some(peer.label.clone().unwrap_or_else(|| peer.url.clone()));
}
if let Some(trusted) = self.find_trusted_by_fingerprint(hello_fp) {
return Some(trusted.label.clone().unwrap_or_else(|| hello_fp[..16].to_string()));
}
None
}
}
// ── Outbound media egress task ──
/// Drains the federation media channel and forwards to active peers.
pub async fn run_federation_media_egress(
fm: Arc<FederationManager>,
mut rx: tokio::sync::mpsc::Receiver<FederationMediaOut>,
) {
let mut count: u64 = 0;
while let Some(out) = rx.recv().await {
count += 1;
if count == 1 || count % 250 == 0 {
info!(room = %out.room_name, count, "federation egress: forwarding media");
}
fm.forward_to_peers(&out.room_name, &out.room_hash, &out.data).await;
}
info!(total = count, "federation egress task ended");
}
// ── Room event dispatcher ──
/// Watches RoomManager events and sends GlobalRoomActive/Inactive to peers.
async fn run_room_event_dispatcher(
fm: Arc<FederationManager>,
mut events: tokio::sync::broadcast::Receiver<RoomEvent>,
) {
loop {
match events.recv().await {
Ok(RoomEvent::LocalJoin { room }) => {
if fm.is_global_room(&room) {
let participants = {
let mgr = fm.room_mgr.lock().await;
mgr.local_participant_list(&room)
};
info!(room = %room, count = participants.len(), "global room now active, announcing to peers");
let msg = SignalMessage::GlobalRoomActive { room, participants };
let links = fm.peer_links.lock().await;
for link in links.values() {
let _ = link.transport.send_signal(&msg).await;
}
}
}
Ok(RoomEvent::LocalLeave { room }) => {
if fm.is_global_room(&room) {
info!(room = %room, "global room now inactive, announcing to peers");
let msg = SignalMessage::GlobalRoomInactive { room };
let links = fm.peer_links.lock().await;
for link in links.values() {
let _ = link.transport.send_signal(&msg).await;
}
}
}
Err(tokio::sync::broadcast::error::RecvError::Lagged(n)) => {
warn!(missed = n, "room event receiver lagged");
}
Err(tokio::sync::broadcast::error::RecvError::Closed) => break,
}
}
}
// ── Stale presence sweeper ──
/// Periodically checks for stale remote participants and purges them.
/// This handles the case where a peer link dies without sending GlobalRoomInactive
/// (e.g., QUIC timeout, network partition, crash).
async fn run_stale_presence_sweeper(fm: Arc<FederationManager>) {
let mut interval = tokio::time::interval(Duration::from_secs(5));
loop {
interval.tick().await;
let stale_threshold = Duration::from_secs(REMOTE_PARTICIPANT_STALE_SECS);
// Find peers with stale remote_participants whose link is also gone or idle
let stale_rooms: Vec<(String, String)> = {
let links = fm.peer_links.lock().await;
let mut stale = Vec::new();
for (fp, link) in links.iter() {
if link.last_seen.elapsed() > stale_threshold && !link.remote_participants.is_empty() {
for room in link.remote_participants.keys() {
stale.push((fp.clone(), room.clone()));
}
}
}
stale
};
if stale_rooms.is_empty() {
continue;
}
// Purge stale entries and collect affected rooms
let mut affected_rooms = HashSet::new();
{
let mut links = fm.peer_links.lock().await;
for (fp, room) in &stale_rooms {
if let Some(link) = links.get_mut(fp.as_str()) {
if link.last_seen.elapsed() > stale_threshold {
info!(peer = %link.label, room = %room, "purging stale remote participants (no data for {}s)", link.last_seen.elapsed().as_secs());
link.remote_participants.remove(room);
link.active_rooms.remove(room);
affected_rooms.insert(room.clone());
}
}
}
}
// Broadcast updated RoomUpdate for affected rooms
for room in &affected_rooms {
let mgr = fm.room_mgr.lock().await;
for local_room in mgr.active_rooms() {
if fm.resolve_global_room(&local_room) == fm.resolve_global_room(room) {
let mut all_participants = mgr.local_participant_list(&local_room);
let remote = fm.get_remote_participants(&local_room).await;
all_participants.extend(remote);
let mut seen = HashSet::new();
all_participants.retain(|p| seen.insert(p.fingerprint.clone()));
let update = SignalMessage::RoomUpdate {
count: all_participants.len() as u32,
participants: all_participants,
};
let senders = mgr.local_senders(&local_room);
drop(mgr);
room::broadcast_signal(&senders, &update).await;
info!(room = %room, "swept stale presence — broadcast updated RoomUpdate");
break;
}
}
}
}
}
// ── Peer connection management ──
/// Persistent connection loop for one peer — reconnects with backoff.
async fn run_peer_loop(fm: Arc<FederationManager>, peer: PeerConfig) {
let mut backoff = Duration::from_secs(5);
loop {
info!(peer_url = %peer.url, label = ?peer.label, "federation: connecting to peer...");
match connect_to_peer(&fm, &peer).await {
Ok(transport) => {
backoff = Duration::from_secs(5);
let peer_fp = normalize_fp(&peer.fingerprint);
let label = peer.label.clone().unwrap_or_else(|| peer.url.clone());
if let Err(e) = run_federation_link(fm.clone(), transport, peer_fp, label).await {
warn!(peer_url = %peer.url, "federation link ended: {e}");
}
}
Err(e) => {
warn!(peer_url = %peer.url, backoff_s = backoff.as_secs(), "federation connect failed: {e}");
}
}
tokio::time::sleep(backoff).await;
backoff = (backoff * 2).min(Duration::from_secs(300));
}
}
/// Connect to a peer relay and send hello.
async fn connect_to_peer(fm: &FederationManager, peer: &PeerConfig) -> Result<Arc<QuinnTransport>, anyhow::Error> {
let addr: SocketAddr = peer.url.parse()?;
let client_cfg = wzp_transport::client_config();
let conn = wzp_transport::connect(&fm.endpoint, addr, "_federation", client_cfg).await?;
let transport = Arc::new(QuinnTransport::new(conn));
// Send hello with our TLS fingerprint
let hello = SignalMessage::FederationHello {
tls_fingerprint: fm.local_tls_fp.clone(),
};
transport.send_signal(&hello).await
.map_err(|e| anyhow::anyhow!("federation hello send failed: {e}"))?;
info!(peer_url = %peer.url, label = ?peer.label, "federation: connected (hello sent)");
Ok(transport)
}
// ── Federation link (runs on a single QUIC connection) ──
/// Run the federation link: exchange global room state and forward media.
async fn run_federation_link(
fm: Arc<FederationManager>,
transport: Arc<QuinnTransport>,
peer_fp: String,
peer_label: String,
) -> Result<(), anyhow::Error> {
// Register peer link + metrics
fm.metrics.federation_peer_status.with_label_values(&[&peer_label]).set(1);
{
let mut links = fm.peer_links.lock().await;
links.insert(peer_fp.clone(), PeerLink {
transport: transport.clone(),
label: peer_label.clone(),
active_rooms: HashSet::new(),
remote_participants: HashMap::new(),
last_seen: Instant::now(),
});
}
// Announce our currently active global rooms to this new peer
// Collect all announcements first, then send (avoid holding locks across await)
let announcements = {
let mgr = fm.room_mgr.lock().await;
let active = mgr.active_rooms();
let mut msgs = Vec::new();
// Local rooms
for room_name in &active {
if fm.is_global_room(room_name) {
let participants = mgr.local_participant_list(room_name);
info!(peer = %peer_label, room = %room_name, participants = participants.len(), "announcing local global room to new peer");
msgs.push(SignalMessage::GlobalRoomActive { room: room_name.clone(), participants });
}
}
// Remote rooms from OTHER peers (for multi-hop propagation)
let links = fm.peer_links.lock().await;
for (fp, link) in links.iter() {
if fp != &peer_fp {
for (room, participants) in &link.remote_participants {
if fm.is_global_room(room) {
info!(peer = %peer_label, room = %room, via = %link.label, "propagating remote room to new peer");
msgs.push(SignalMessage::GlobalRoomActive {
room: room.clone(),
participants: participants.clone(),
});
}
}
}
}
msgs
};
for msg in &announcements {
let _ = transport.send_signal(msg).await;
}
// Three concurrent tasks: signal recv + media recv + RTT monitor
let signal_transport = transport.clone();
let media_transport = transport.clone();
let rtt_transport = transport.clone();
let fm_signal = fm.clone();
let fm_media = fm.clone();
let fm_rtt = fm.clone();
let peer_fp_signal = peer_fp.clone();
let peer_fp_media = peer_fp.clone();
let label_signal = peer_label.clone();
let label_rtt = peer_label.clone();
let signal_task = async move {
loop {
match signal_transport.recv_signal().await {
Ok(Some(msg)) => {
handle_signal(&fm_signal, &peer_fp_signal, &label_signal, msg).await;
}
Ok(None) => break,
Err(e) => {
error!(peer = %label_signal, "federation signal error: {e}");
break;
}
}
}
};
let peer_label_media = peer_label.clone();
let media_task = async move {
let mut media_count: u64 = 0;
loop {
match media_transport.connection().read_datagram().await {
Ok(data) => {
media_count += 1;
if media_count == 1 || media_count % 250 == 0 {
info!(peer = %peer_label_media, media_count, len = data.len(), "federation: received datagram");
}
handle_datagram(&fm_media, &peer_fp_media, data).await;
}
Err(e) => {
info!(peer = %peer_label_media, "federation media task ended: {e}");
break;
}
}
}
};
// RTT monitor: periodically sample QUIC RTT for this peer
let rtt_task = async move {
loop {
tokio::time::sleep(Duration::from_secs(5)).await;
let rtt_ms = rtt_transport.connection().stats().path.rtt.as_millis() as f64;
}
};
tokio::select! {
_ = signal_task => {}
_ = media_task => {}
_ = rtt_task => {}
}
// Cleanup: remove peer link + metrics
fm.metrics.federation_peer_status.with_label_values(&[&peer_label]).set(0);
{
let mut links = fm.peer_links.lock().await;
links.remove(&peer_fp);
}
info!(peer = %peer_label, "federation link ended");
Ok(())
}
/// Handle an incoming federation signal.
async fn handle_signal(
fm: &Arc<FederationManager>,
peer_fp: &str,
peer_label: &str,
msg: SignalMessage,
) {
// Update last_seen for this peer
{
let mut links = fm.peer_links.lock().await;
if let Some(link) = links.get_mut(peer_fp) {
link.last_seen = Instant::now();
}
}
match msg {
SignalMessage::GlobalRoomActive { room, participants } => {
if fm.is_global_room(&room) {
info!(peer = %peer_label, room = %room, remote_participants = participants.len(), "peer has global room active");
let mut links = fm.peer_links.lock().await;
if let Some(link) = links.get_mut(peer_fp) {
link.active_rooms.insert(room.clone());
}
// Update active rooms metric
let total: usize = links.values().map(|l| l.active_rooms.len()).sum();
fm.metrics.federation_active_rooms.set(total as i64);
if let Some(link) = links.get_mut(peer_fp) {
// Tag remote participants with their relay label
let tagged: Vec<_> = participants.iter().map(|p| {
let mut tagged = p.clone();
if tagged.relay_label.is_none() {
tagged.relay_label = Some(link.label.clone());
}
tagged
}).collect();
link.remote_participants.insert(room.clone(), tagged);
}
// Propagate to other peers (with relay labels preserved)
let tagged_for_propagation = if let Some(link) = links.get(peer_fp) {
let label = link.label.clone();
participants.iter().map(|p| {
let mut t = p.clone();
if t.relay_label.is_none() {
t.relay_label = Some(label.clone());
}
t
}).collect::<Vec<_>>()
} else {
participants.clone()
};
for (fp, link) in links.iter() {
if fp != peer_fp {
let _ = link.transport.send_signal(&SignalMessage::GlobalRoomActive {
room: room.clone(),
participants: tagged_for_propagation.clone(),
}).await;
}
}
drop(links);
// Broadcast updated RoomUpdate to local clients in this room
// Find the local room name (may be hashed or raw)
let mgr = fm.room_mgr.lock().await;
for local_room in mgr.active_rooms() {
if fm.is_global_room(&local_room) && fm.resolve_global_room(&local_room) == fm.resolve_global_room(&room) {
// Build merged participant list: local + all remote (deduped)
let mut all_participants = mgr.local_participant_list(&local_room);
let links = fm.peer_links.lock().await;
for link in links.values() {
if let Some(canonical) = fm.resolve_global_room(&local_room) {
if let Some(remote) = link.remote_participants.get(canonical) {
all_participants.extend(remote.iter().cloned());
}
// Also check raw room name, but only if different from canonical
if canonical != local_room {
if let Some(remote) = link.remote_participants.get(&local_room) {
all_participants.extend(remote.iter().cloned());
}
}
}
}
// Deduplicate by fingerprint
let mut seen = HashSet::new();
all_participants.retain(|p| seen.insert(p.fingerprint.clone()));
let update = SignalMessage::RoomUpdate {
count: all_participants.len() as u32,
participants: all_participants,
};
let senders = mgr.local_senders(&local_room);
drop(links);
drop(mgr);
room::broadcast_signal(&senders, &update).await;
break;
}
}
}
}
SignalMessage::GlobalRoomInactive { room } => {
info!(peer = %peer_label, room = %room, "peer global room now inactive");
let mut links = fm.peer_links.lock().await;
if let Some(link) = links.get_mut(peer_fp) {
link.active_rooms.remove(&room);
// Clear remote participants for this peer+room
link.remote_participants.remove(&room);
// Also try canonical name
if let Some(canonical) = fm.resolve_global_room(&room) {
link.remote_participants.remove(canonical);
}
}
// Update active rooms metric
let total: usize = links.values().map(|l| l.active_rooms.len()).sum();
fm.metrics.federation_active_rooms.set(total as i64);
// Build remaining remote participants (from all peers except the one going inactive)
let remaining_remote: Vec<wzp_proto::packet::RoomParticipant> = {
let canonical = fm.resolve_global_room(&room);
let mut result = Vec::new();
for (fp, link) in links.iter() {
if fp == peer_fp { continue; }
if let Some(c) = canonical {
if let Some(remote) = link.remote_participants.get(c) {
result.extend(remote.iter().cloned());
}
}
}
let mut seen = HashSet::new();
result.retain(|p| seen.insert(p.fingerprint.clone()));
result
};
// Propagate to other peers: send updated GlobalRoomActive with revised list,
// or GlobalRoomInactive if no participants remain anywhere
let local_active = {
let mgr = fm.room_mgr.lock().await;
mgr.active_rooms().iter().any(|r| fm.resolve_global_room(r) == fm.resolve_global_room(&room))
};
let has_remaining = !remaining_remote.is_empty() || local_active;
// Collect peer transports to send to (avoid holding lock across await)
let peer_sends: Vec<_> = links.iter()
.filter(|(fp, _)| *fp != peer_fp)
.map(|(_, link)| link.transport.clone())
.collect();
drop(links);
if has_remaining {
// Send updated participant list to other peers
let mut updated_participants = remaining_remote.clone();
if local_active {
let mgr = fm.room_mgr.lock().await;
for local_room in mgr.active_rooms() {
if fm.resolve_global_room(&local_room) == fm.resolve_global_room(&room) {
updated_participants.extend(mgr.local_participant_list(&local_room));
break;
}
}
}
let msg = SignalMessage::GlobalRoomActive {
room: room.clone(),
participants: updated_participants,
};
for transport in &peer_sends {
let _ = transport.send_signal(&msg).await;
}
} else {
// No participants left anywhere — propagate inactive
let msg = SignalMessage::GlobalRoomInactive { room: room.clone() };
for transport in &peer_sends {
let _ = transport.send_signal(&msg).await;
}
}
// Broadcast updated RoomUpdate to local clients (remote participant removed)
let mgr = fm.room_mgr.lock().await;
for local_room in mgr.active_rooms() {
if fm.is_global_room(&local_room) && fm.resolve_global_room(&local_room) == fm.resolve_global_room(&room) {
let mut all_participants = mgr.local_participant_list(&local_room);
all_participants.extend(remaining_remote.iter().cloned());
// Deduplicate by fingerprint
let mut seen = HashSet::new();
all_participants.retain(|p| seen.insert(p.fingerprint.clone()));
let update = SignalMessage::RoomUpdate {
count: all_participants.len() as u32,
participants: all_participants,
};
let senders = mgr.local_senders(&local_room);
drop(mgr);
room::broadcast_signal(&senders, &update).await;
info!(room = %room, "broadcast updated presence (remote participant removed)");
break;
}
}
}
_ => {} // ignore other signals
}
}
/// Handle an incoming federation datagram (room-hash-tagged media).
async fn handle_datagram(
fm: &Arc<FederationManager>,
source_peer_fp: &str,
data: Bytes,
) {
if data.len() < 12 { return; } // 8-byte hash + min packet
let mut rh = [0u8; 8];
rh.copy_from_slice(&data[..8]);
let media_bytes = data.slice(8..);
let pkt = match wzp_proto::MediaPacket::from_bytes(media_bytes.clone()) {
Some(pkt) => pkt,
None => {
fm.event_log.emit(Event::new("federation_ingress_malformed").len(data.len()));
return;
}
};
// Event log: federation ingress
let peer_label = {
let links = fm.peer_links.lock().await;
links.get(source_peer_fp).map(|l| l.label.clone()).unwrap_or_default()
};
fm.event_log.emit(Event::new("federation_ingress").packet(&pkt).peer(&peer_label));
// Count inbound federation packet + update last_seen
fm.metrics.federation_packets_forwarded
.with_label_values(&[source_peer_fp, "in"]).inc();
{
let mut links = fm.peer_links.lock().await;
if let Some(link) = links.get_mut(source_peer_fp) {
link.last_seen = Instant::now();
}
}
// Dedup: drop packets we've already seen (multi-path duplicates).
// Key uses a hash of the actual payload bytes — unique per Opus frame,
// so different senders with the same seq/timestamp never collide.
let payload_hash = {
let mut h = 0u64;
for (i, &b) in media_bytes.iter().take(16).enumerate() {
h ^= (b as u64) << ((i % 8) * 8);
}
h
};
{
let mut dedup = fm.dedup.lock().await;
if dedup.is_dup(&rh, pkt.header.seq, payload_hash) {
fm.event_log.emit(Event::new("dedup_drop").seq(pkt.header.seq).peer(&peer_label));
return;
}
}
// Find room by hash — check local rooms AND global room config
let room_name = {
let mgr = fm.room_mgr.lock().await;
let active = mgr.active_rooms();
// First: check local rooms (has participants)
active.iter().find(|r| room_hash(r) == rh).cloned()
.or_else(|| active.iter().find(|r| fm.global_room_hash(r) == rh).cloned())
// Second: check global room config (hub relay may have no local participants)
.or_else(|| {
fm.global_rooms.iter().find(|name| room_hash(name) == rh).cloned()
})
};
let room_name = match room_name {
Some(r) => r,
None => {
fm.event_log.emit(Event::new("room_not_found").seq(pkt.header.seq).peer(&peer_label));
return;
}
};
// Rate limit per room
if FEDERATION_RATE_LIMIT_PPS > 0 {
let mut limiters = fm.rate_limiters.lock().await;
let limiter = limiters.entry(room_name.clone())
.or_insert_with(|| RateLimiter::new(FEDERATION_RATE_LIMIT_PPS));
if !limiter.allow() {
fm.event_log.emit(Event::new("rate_limit_drop").room(&room_name).seq(pkt.header.seq));
return;
}
}
// Deliver to all local participants — forward the raw bytes as-is.
// The original sender's MediaPacket is preserved exactly (no re-serialization).
let locals = {
let mgr = fm.room_mgr.lock().await;
mgr.local_senders(&room_name)
};
for sender in &locals {
match sender {
room::ParticipantSender::Quic(t) => {
if let Err(e) = t.send_raw_datagram(&media_bytes) {
fm.event_log.emit(Event::new("local_deliver_error").room(&room_name).seq(pkt.header.seq).reason(&e.to_string()));
warn!("federation local delivery error: {e}");
}
}
room::ParticipantSender::WebSocket(_) => { let _ = sender.send_raw(&pkt.payload).await; }
}
}
fm.event_log.emit(Event::new("local_deliver").room(&room_name).seq(pkt.header.seq).to_count(locals.len()));
// Multi-hop: forward to ALL other connected peers (not the source)
// Don't filter by active_rooms — the receiving peer decides whether to deliver
let links = fm.peer_links.lock().await;
for (fp, link) in links.iter() {
if fp != source_peer_fp {
let mut tagged = Vec::with_capacity(8 + media_bytes.len());
tagged.extend_from_slice(&rh);
tagged.extend_from_slice(&media_bytes);
let _ = link.transport.send_raw_datagram(&tagged);
}
}
}

View File

@@ -15,25 +15,27 @@ use wzp_proto::{MediaTransport, QualityProfile, SignalMessage};
/// 5. Derive shared ChaCha20-Poly1305 session
/// 6. Send `CallAnswer` back
///
/// Returns the derived `CryptoSession` and the chosen `QualityProfile`.
/// Returns the derived `CryptoSession`, the chosen `QualityProfile`, the caller's fingerprint,
/// and the caller's alias (if provided in CallOffer).
pub async fn accept_handshake(
transport: &dyn MediaTransport,
seed: &[u8; 32],
) -> Result<(Box<dyn CryptoSession>, QualityProfile), anyhow::Error> {
) -> Result<(Box<dyn CryptoSession>, QualityProfile, String, Option<String>), anyhow::Error> {
// 1. Receive CallOffer
let offer = transport
.recv_signal()
.await?
.ok_or_else(|| anyhow::anyhow!("connection closed before receiving CallOffer"))?;
let (caller_identity_pub, caller_ephemeral_pub, caller_signature, supported_profiles) =
let (caller_identity_pub, caller_ephemeral_pub, caller_signature, supported_profiles, caller_alias) =
match offer {
SignalMessage::CallOffer {
identity_pub,
ephemeral_pub,
signature,
supported_profiles,
} => (identity_pub, ephemeral_pub, signature, supported_profiles),
alias,
} => (identity_pub, ephemeral_pub, signature, supported_profiles, alias),
other => {
return Err(anyhow::anyhow!(
"expected CallOffer, got {:?}",
@@ -76,25 +78,26 @@ pub async fn accept_handshake(
};
transport.send_signal(&answer).await?;
Ok((session, chosen_profile))
// Derive caller fingerprint: SHA-256(Ed25519 pub)[:16], formatted as xxxx:xxxx:...
// Must match the format used in signal registration and presence.
let caller_fp = {
use sha2::{Sha256, Digest};
let hash = Sha256::digest(&caller_identity_pub);
let fp = wzp_crypto::Fingerprint([
hash[0], hash[1], hash[2], hash[3], hash[4], hash[5], hash[6], hash[7],
hash[8], hash[9], hash[10], hash[11], hash[12], hash[13], hash[14], hash[15],
]);
fp.to_string()
};
Ok((session, chosen_profile, caller_fp, caller_alias))
}
/// Select the best quality profile from those the caller supports.
fn choose_profile(supported: &[QualityProfile]) -> QualityProfile {
// Prefer higher-quality profiles. Use GOOD as default if supported list is empty.
if supported.is_empty() {
return QualityProfile::GOOD;
}
// Pick the profile with the highest bitrate.
supported
.iter()
.max_by(|a, b| {
a.total_bitrate_kbps()
.partial_cmp(&b.total_bitrate_kbps())
.unwrap_or(std::cmp::Ordering::Equal)
})
.copied()
.unwrap_or(QualityProfile::GOOD)
// Cap at GOOD (24k) for now — studio tiers (32k/48k/64k) not yet tested
// for federation reliability (large packets may exceed path MTU).
QualityProfile::GOOD
}
#[cfg(test)]

View File

@@ -8,7 +8,11 @@
//! quality transitions.
pub mod auth;
pub mod call_registry;
pub mod config;
pub mod event_log;
pub mod federation;
pub mod signal_hub;
pub mod handshake;
pub mod metrics;
pub mod pipeline;

View File

@@ -13,9 +13,9 @@ use std::sync::Arc;
use std::time::Duration;
use tokio::sync::Mutex;
use tracing::{error, info};
use tracing::{error, info, warn};
use wzp_proto::MediaTransport;
use wzp_proto::{MediaTransport, SignalMessage};
use wzp_relay::config::RelayConfig;
use wzp_relay::metrics::RelayMetrics;
use wzp_relay::pipeline::{PipelineConfig, RelayPipeline};
@@ -23,12 +23,54 @@ use wzp_relay::presence::PresenceRegistry;
use wzp_relay::room::{self, RoomManager};
use wzp_relay::session_mgr::SessionManager;
fn parse_args() -> RelayConfig {
let mut config = RelayConfig::default();
/// Parsed CLI result — config + identity path.
struct CliResult {
config: RelayConfig,
identity_path: Option<String>,
config_file: Option<String>,
config_needs_create: bool,
}
fn parse_args() -> CliResult {
let args: Vec<String> = std::env::args().collect();
// First pass: extract --config and --identity
let mut config_file = None;
let mut identity_path = None;
let mut i = 1;
while i < args.len() {
match args[i].as_str() {
"--config" | "-c" => { i += 1; config_file = args.get(i).cloned(); }
"--identity" | "-i" => { i += 1; identity_path = args.get(i).cloned(); }
_ => {}
}
i += 1;
}
// Track if we need to create the config after identity is known
let config_needs_create = config_file.as_ref().map(|p| !std::path::Path::new(p).exists()).unwrap_or(false);
let mut config = if let Some(ref path) = config_file {
if config_needs_create {
// Will be re-created with personalized info after identity is loaded
RelayConfig::default()
} else {
wzp_relay::config::load_config(path)
.unwrap_or_else(|e| {
eprintln!("failed to load config from {path}: {e}");
std::process::exit(1);
})
}
} else {
RelayConfig::default()
};
// CLI flags override config file values
let mut i = 1;
while i < args.len() {
match args[i].as_str() {
"--config" | "-c" => { i += 1; } // already handled
"--identity" | "-i" => { i += 1; } // already handled
"--listen" => {
i += 1;
config.listen_addr = args.get(i).expect("--listen requires an address")
@@ -81,6 +123,28 @@ fn parse_args() -> RelayConfig {
args.get(i).expect("--static-dir requires a directory path").to_string(),
);
}
"--global-room" => {
i += 1;
config.global_rooms.push(wzp_relay::config::GlobalRoomConfig {
name: args.get(i).expect("--global-room requires a room name").to_string(),
});
}
"--debug-tap" => {
i += 1;
config.debug_tap = Some(
args.get(i).expect("--debug-tap requires a room name (or '*' for all)").to_string(),
);
}
"--event-log" => {
i += 1;
config.event_log = Some(
args.get(i).expect("--event-log requires a file path").to_string(),
);
}
"--version" | "-V" => {
println!("wzp-relay {}", env!("WZP_BUILD_HASH"));
std::process::exit(0);
}
"--mesh-status" => {
// Print mesh table from a fresh registry and exit.
// In practice this is useful after the relay has been running;
@@ -90,9 +154,11 @@ fn parse_args() -> RelayConfig {
std::process::exit(0);
}
"--help" | "-h" => {
eprintln!("Usage: wzp-relay [--listen <addr>] [--remote <addr>] [--auth-url <url>] [--metrics-port <port>] [--probe <addr>]... [--probe-mesh] [--mesh-status]");
eprintln!("Usage: wzp-relay [--config <path>] [--listen <addr>] [--remote <addr>] [--auth-url <url>] [--metrics-port <port>] [--probe <addr>]... [--probe-mesh] [--mesh-status]");
eprintln!();
eprintln!("Options:");
eprintln!(" -c, --config <path> Load config from TOML file (creates example if missing)");
eprintln!(" -i, --identity <path> Identity file path (creates if missing, uses OsRng)");
eprintln!(" --listen <addr> Listen address (default: 0.0.0.0:4433)");
eprintln!(" --remote <addr> Remote relay for forwarding (disables room mode)");
eprintln!(" --auth-url <url> featherChat auth endpoint (e.g., https://chat.example.com/v1/auth/validate)");
@@ -102,6 +168,8 @@ fn parse_args() -> RelayConfig {
eprintln!(" --probe-mesh Enable mesh mode (mark config flag, probes all --probe targets).");
eprintln!(" --mesh-status Print mesh health table and exit (diagnostic).");
eprintln!(" --trunking Enable trunk batching for outgoing media in room mode.");
eprintln!(" --global-room <name> Declare a room as global (bridged across federation). Repeatable.");
eprintln!(" --debug-tap <room> Log packet headers for a room ('*' for all rooms).");
eprintln!(" --ws-port <port> WebSocket listener port for browser clients (e.g., 8080).");
eprintln!(" --static-dir <dir> Directory to serve static files from (HTML/JS/WASM).");
eprintln!();
@@ -116,7 +184,7 @@ fn parse_args() -> RelayConfig {
}
i += 1;
}
config
CliResult { config, identity_path, config_file, config_needs_create }
}
struct RelayStats {
@@ -184,10 +252,29 @@ async fn run_downstream(
}
}
/// Detect a non-loopback IP address from local interfaces.
/// Prefers public IPs over private (10.x, 172.16-31.x, 192.168.x).
fn detect_public_ip() -> Option<String> {
use std::net::UdpSocket;
// Connect to a public address to find our outbound IP (doesn't actually send anything)
if let Ok(socket) = UdpSocket::bind("0.0.0.0:0") {
if socket.connect("8.8.8.8:80").is_ok() {
if let Ok(addr) = socket.local_addr() {
return Some(addr.ip().to_string());
}
}
}
None
}
/// Build-time git hash, set by build.rs or env.
const BUILD_GIT_HASH: &str = env!("WZP_BUILD_HASH");
#[tokio::main]
async fn main() -> anyhow::Result<()> {
let config = parse_args();
let CliResult { mut config, identity_path, config_file, config_needs_create } = parse_args();
tracing_subscriber::fmt().init();
info!(version = BUILD_GIT_HASH, "wzp-relay build");
rustls::crypto::ring::default_provider()
.install_default()
.expect("failed to install rustls crypto provider");
@@ -207,14 +294,115 @@ async fn main() -> anyhow::Result<()> {
tokio::spawn(wzp_relay::metrics::serve_metrics(port, m, p, rr));
}
// Generate ephemeral relay identity for crypto handshake
let relay_seed = wzp_crypto::Seed::generate();
// Load or generate relay identity
let relay_seed = {
let id_path = match identity_path {
Some(ref p) => std::path::PathBuf::from(p),
None => dirs::home_dir()
.unwrap_or_else(|| std::path::PathBuf::from("."))
.join(".wzp")
.join("relay-identity"),
};
if id_path.exists() {
if let Ok(hex) = std::fs::read_to_string(&id_path) {
if let Ok(s) = wzp_crypto::Seed::from_hex(hex.trim()) {
info!("loaded relay identity from {}", id_path.display());
s
} else {
warn!("corrupt identity file {}, generating new", id_path.display());
let s = wzp_crypto::Seed::generate();
let hex: String = s.0.iter().map(|b| format!("{b:02x}")).collect();
let _ = std::fs::write(&id_path, &hex);
s
}
} else {
let s = wzp_crypto::Seed::generate();
let hex: String = s.0.iter().map(|b| format!("{b:02x}")).collect();
let _ = std::fs::write(&id_path, &hex);
s
}
} else {
let s = wzp_crypto::Seed::generate();
if let Some(parent) = id_path.parent() {
let _ = std::fs::create_dir_all(parent);
}
let hex: String = s.0.iter().map(|b| format!("{b:02x}")).collect();
let _ = std::fs::write(&id_path, &hex);
info!("generated relay identity at {}", id_path.display());
s
}
};
let relay_fp = relay_seed.derive_identity().public_identity().fingerprint;
info!(addr = %config.listen_addr, fingerprint = %relay_fp, "WarzonePhone relay starting");
let (server_config, _cert) = wzp_transport::server_config();
let (server_config, cert_der) = wzp_transport::server_config_from_seed(&relay_seed.0);
let tls_fp = wzp_transport::tls_fingerprint(&cert_der);
info!(tls_fingerprint = %tls_fp, "TLS certificate (deterministic from relay identity)");
// Create personalized config file if it was missing
let public_ip = detect_public_ip();
if config_needs_create {
if let Some(ref path) = config_file {
let info = wzp_relay::config::RelayInfo {
listen_addr: config.listen_addr.to_string(),
tls_fingerprint: tls_fp.clone(),
public_ip: public_ip.clone(),
};
if let Err(e) = wzp_relay::config::load_or_create_config(path, Some(&info)) {
warn!("failed to create config: {e}");
}
}
}
// Print federation hint with our public IP + listen port + TLS fingerprint
let listen_port = config.listen_addr.port();
if let Some(ip) = &public_ip {
info!("federation: to peer with this relay, add to relay.toml:");
info!(" [[peers]]");
info!(" url = \"{ip}:{listen_port}\"");
info!(" fingerprint = \"{tls_fp}\"");
}
// Log configured peers and trusted relays
if !config.peers.is_empty() {
info!(count = config.peers.len(), "federation peers configured");
for p in &config.peers {
info!(url = %p.url, label = ?p.label, " peer");
}
}
if !config.trusted.is_empty() {
info!(count = config.trusted.len(), "trusted relays configured");
for t in &config.trusted {
info!(fingerprint = %t.fingerprint, label = ?t.label, " trusted");
}
}
let endpoint = wzp_transport::create_endpoint(config.listen_addr, Some(server_config))?;
// Compute the IP address we should advertise in CallSetup for direct
// calls. If the relay is bound to a specific IP, use it as-is; if bound
// to 0.0.0.0, use the trick of "connect" a UDP socket to an arbitrary
// external address and read its local_addr — the OS binds to whichever
// local interface IP would route packets to that destination, which is
// the primary outbound interface. This is the same IP clients on the
// LAN use to reach us.
let advertised_ip: std::net::IpAddr = {
let listen_ip = config.listen_addr.ip();
if !listen_ip.is_unspecified() {
listen_ip
} else {
// Probe via a dummy "connected" UDP socket. Never actually sends.
match std::net::UdpSocket::bind("0.0.0.0:0")
.and_then(|s| { s.connect("8.8.8.8:80").map(|_| s) })
.and_then(|s| s.local_addr())
{
Ok(a) if !a.ip().is_loopback() => a.ip(),
_ => std::net::IpAddr::from([127u8, 0, 0, 1]),
}
}
};
let advertised_addr_str = format!("{}:{}", advertised_ip, config.listen_addr.port());
info!(%advertised_addr_str, "relay advertised address for CallSetup");
// Forward mode
let remote_transport: Option<Arc<wzp_transport::QuinnTransport>> =
if let Some(remote_addr) = config.remote_relay {
@@ -230,9 +418,41 @@ async fn main() -> anyhow::Result<()> {
// Room manager (room mode only)
let room_mgr = Arc::new(Mutex::new(RoomManager::new()));
// Event log for protocol analysis
let event_log = wzp_relay::event_log::start_event_log(
config.event_log.as_ref().map(std::path::PathBuf::from)
);
// Federation manager
let global_room_set: std::collections::HashSet<String> = config.global_rooms.iter()
.map(|g| g.name.clone())
.collect();
let federation_mgr = if !config.peers.is_empty() || !config.trusted.is_empty() || !global_room_set.is_empty() {
let fm = Arc::new(wzp_relay::federation::FederationManager::new(
config.peers.clone(),
config.trusted.clone(),
global_room_set.clone(),
room_mgr.clone(),
endpoint.clone(),
tls_fp.clone(),
metrics.clone(),
event_log.clone(),
));
let fm_run = fm.clone();
tokio::spawn(async move { fm_run.run().await });
Some(fm)
} else {
None
};
// Session manager — enforces max concurrent sessions
let session_mgr = Arc::new(Mutex::new(SessionManager::new(config.max_sessions)));
// Signal hub + call registry for direct 1:1 calls
let signal_hub = Arc::new(Mutex::new(wzp_relay::signal_hub::SignalHub::new()));
let call_registry = Arc::new(Mutex::new(wzp_relay::call_registry::CallRegistry::new()));
// Spawn inter-relay health probes via ProbeMesh coordinator
if !config.probe_targets.is_empty() {
let mesh = wzp_relay::probe::ProbeMesh::new(
@@ -267,13 +487,32 @@ async fn main() -> anyhow::Result<()> {
} else {
info!("auth disabled — any client can connect (use --auth-url to enable)");
}
if !config.global_rooms.is_empty() {
info!(count = config.global_rooms.len(), "global rooms configured");
for g in &config.global_rooms {
info!(name = %g.name, " global room");
}
}
if let Some(ref tap) = config.debug_tap {
info!(filter = %tap, "debug tap enabled — logging packet headers");
}
info!("Listening for connections...");
loop {
let connection = match wzp_transport::accept(&endpoint).await {
Ok(conn) => conn,
Err(e) => { error!("accept: {e}"); continue; }
// Pull the next Incoming off the queue. Deliberately do NOT await
// the QUIC handshake here — move that into the per-connection
// spawned task below. Previously we used wzp_transport::accept
// which did both, which meant a single slow handshake would block
// the entire accept loop and prevent ALL subsequent connections
// from being processed. Surfaced as direct-call hangs where the
// callee's call-* connection never completes its QUIC handshake.
let incoming = match endpoint.accept().await {
Some(inc) => inc,
None => {
error!("endpoint.accept() returned None — endpoint closed");
break;
}
};
let remote_transport = remote_transport.clone();
@@ -283,10 +522,28 @@ async fn main() -> anyhow::Result<()> {
let relay_seed_bytes = relay_seed.0;
let metrics = metrics.clone();
let trunking_enabled = config.trunking_enabled;
let debug_tap = config.debug_tap.as_ref().map(|filter| room::DebugTap { room_filter: filter.clone() });
let presence = presence.clone();
let route_resolver = route_resolver.clone();
let federation_mgr = federation_mgr.clone();
let signal_hub = signal_hub.clone();
let call_registry = call_registry.clone();
let advertised_addr_str = advertised_addr_str.clone();
let incoming_addr = incoming.remote_address();
info!(%incoming_addr, "accept queue: new Incoming, spawning handshake task");
tokio::spawn(async move {
// Drive the QUIC handshake inside the spawned task so that
// slow or hung handshakes never block the outer accept loop.
let connection = match incoming.await {
Ok(c) => c,
Err(e) => {
error!(%incoming_addr, "QUIC handshake failed: {e}");
return;
}
};
info!(%incoming_addr, "QUIC handshake complete");
let addr = connection.remote_address();
let room_name = connection
@@ -299,6 +556,23 @@ async fn main() -> anyhow::Result<()> {
let transport = Arc::new(wzp_transport::QuinnTransport::new(connection));
// Ping connections: client just measures QUIC connect RTT.
if room_name == "ping" {
info!(%addr, "ping connection (RTT probe)");
return;
}
// Version query: respond with build hash over a uni stream.
if room_name == "version" {
if let Ok(mut send) = transport.connection().open_uni().await {
let _ = send.write_all(BUILD_GIT_HASH.as_bytes()).await;
let _ = send.finish();
// Wait for client to read before closing
tokio::time::sleep(std::time::Duration::from_millis(100)).await;
}
return;
}
// Probe connections use SNI "_probe" to identify themselves.
// They skip auth + handshake and just do Ping->Pong + presence gossip.
if room_name == "_probe" {
@@ -385,6 +659,290 @@ async fn main() -> anyhow::Result<()> {
return;
}
// Federation connections use SNI "_federation"
if room_name == "_federation" {
if let Some(ref fm) = federation_mgr {
// Wait for FederationHello to identify the connecting relay
let hello_fp = match tokio::time::timeout(
std::time::Duration::from_secs(5),
transport.recv_signal(),
).await {
Ok(Ok(Some(wzp_proto::SignalMessage::FederationHello { tls_fingerprint }))) => tls_fingerprint,
_ => {
warn!(%addr, "federation: no hello received, closing");
return;
}
};
if let Some(label) = fm.check_inbound_trust(addr, &hello_fp) {
let peer_config = wzp_relay::config::PeerConfig {
url: addr.to_string(),
fingerprint: hello_fp,
label: Some(label.clone()),
};
let fm = fm.clone();
info!(%addr, label = %label, "inbound federation accepted (trusted)");
fm.handle_inbound(transport, peer_config).await;
} else {
warn!(%addr, fp = %hello_fp, "unknown relay wants to federate");
info!(" to accept, add to relay.toml:");
info!(" [[trusted]]");
info!(" fingerprint = \"{hello_fp}\"");
info!(" label = \"Relay at {addr}\"");
}
} else {
info!(%addr, "federation connection rejected (no federation configured)");
}
return;
}
// Direct calling: persistent signaling connection
if room_name == "_signal" {
info!(%addr, "signal connection");
// Optional auth
let auth_fp: Option<String> = if let Some(ref url) = auth_url {
match transport.recv_signal().await {
Ok(Some(SignalMessage::AuthToken { token })) => {
match wzp_relay::auth::validate_token(url, &token).await {
Ok(client) => Some(client.fingerprint),
Err(e) => {
error!(%addr, "signal auth failed: {e}");
return;
}
}
}
_ => { warn!(%addr, "signal: expected AuthToken"); return; }
}
} else {
None
};
// Wait for RegisterPresence
let (client_fp, client_alias) = match tokio::time::timeout(
std::time::Duration::from_secs(10),
transport.recv_signal(),
).await {
Ok(Ok(Some(SignalMessage::RegisterPresence { identity_pub, signature: _, alias }))) => {
// Compute fingerprint: SHA-256(Ed25519 pub key)[:16], same as Fingerprint type
let fp = {
use sha2::{Sha256, Digest};
let hash = Sha256::digest(&identity_pub);
let fingerprint = wzp_crypto::Fingerprint([
hash[0], hash[1], hash[2], hash[3], hash[4], hash[5], hash[6], hash[7],
hash[8], hash[9], hash[10], hash[11], hash[12], hash[13], hash[14], hash[15],
]);
fingerprint.to_string()
};
let fp = auth_fp.unwrap_or(fp);
(fp, alias)
}
_ => {
warn!(%addr, "signal: no RegisterPresence received");
return;
}
};
// Register in signal hub + presence
{
let mut hub = signal_hub.lock().await;
hub.register(client_fp.clone(), transport.clone(), client_alias.clone());
}
{
let mut reg = presence.lock().await;
reg.register_local(&client_fp, client_alias.clone(), None);
}
// Send ack
let _ = transport.send_signal(&SignalMessage::RegisterPresenceAck {
success: true,
error: None,
}).await;
info!(%addr, fingerprint = %client_fp, alias = ?client_alias, "signal client registered");
// Signal recv loop
loop {
match transport.recv_signal().await {
Ok(Some(msg)) => {
match msg {
SignalMessage::DirectCallOffer { ref target_fingerprint, ref call_id, ref caller_alias, .. } => {
let target_fp = target_fingerprint.clone();
let call_id = call_id.clone();
// Check if target is online
let online = {
let hub = signal_hub.lock().await;
hub.is_online(&target_fp)
};
if !online {
info!(%addr, target = %target_fp, "call target not online");
let _ = transport.send_signal(&SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
}).await;
continue;
}
// Create call in registry
{
let mut reg = call_registry.lock().await;
reg.create_call(call_id.clone(), client_fp.clone(), target_fp.clone());
}
// Forward offer to callee
info!(caller = %client_fp, callee = %target_fp, call_id = %call_id, "routing direct call offer");
let hub = signal_hub.lock().await;
if let Err(e) = hub.send_to(&target_fp, &msg).await {
warn!("failed to forward call offer: {e}");
}
// Send ringing to caller
drop(hub);
let _ = transport.send_signal(&SignalMessage::CallRinging {
call_id: call_id.clone(),
}).await;
}
SignalMessage::DirectCallAnswer { ref call_id, ref accept_mode, .. } => {
let call_id = call_id.clone();
let mode = *accept_mode;
let peer_fp = {
let reg = call_registry.lock().await;
reg.peer_fingerprint(&call_id, &client_fp).map(|s| s.to_string())
};
let Some(peer_fp) = peer_fp else {
warn!(call_id = %call_id, "answer for unknown call");
continue;
};
if mode == wzp_proto::CallAcceptMode::Reject {
info!(call_id = %call_id, "call rejected");
let mut reg = call_registry.lock().await;
reg.end_call(&call_id);
drop(reg);
let hub = signal_hub.lock().await;
let _ = hub.send_to(&peer_fp, &SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
}).await;
} else {
// Accept — create private room
let room = format!("call-{call_id}");
{
let mut reg = call_registry.lock().await;
reg.set_active(&call_id, mode, room.clone());
}
info!(call_id = %call_id, room = %room, mode = ?mode, "call accepted, creating room");
// Forward answer to caller
{
let hub = signal_hub.lock().await;
let _ = hub.send_to(&peer_fp, &msg).await;
}
// Send CallSetup to both parties.
//
// BUG FIX: the previous version of this used `addr.ip()`
// which is `connection.remote_address()` — the CLIENT'S
// IP, not the relay's. So CallSetup told both parties to
// dial the answerer's own IP, which meant the caller was
// sending QUIC Initials into the callee's client (no
// server listening there) and the callee was sending to
// itself. In both cases endpoint.connect() hung forever.
//
// Use the relay's precomputed advertised address instead.
let relay_addr_for_setup = advertised_addr_str.clone();
let setup = SignalMessage::CallSetup {
call_id: call_id.clone(),
room: room.clone(),
relay_addr: relay_addr_for_setup,
};
{
let hub = signal_hub.lock().await;
let _ = hub.send_to(&peer_fp, &setup).await;
let _ = hub.send_to(&client_fp, &setup).await;
}
}
}
SignalMessage::Hangup { .. } => {
// Forward hangup to all active calls for this user
let calls = {
let reg = call_registry.lock().await;
reg.calls_for_fingerprint(&client_fp)
.iter()
.map(|c| (c.call_id.clone(), if c.caller_fingerprint == client_fp {
c.callee_fingerprint.clone()
} else {
c.caller_fingerprint.clone()
}))
.collect::<Vec<_>>()
};
for (call_id, peer_fp) in &calls {
let hub = signal_hub.lock().await;
let _ = hub.send_to(peer_fp, &msg).await;
drop(hub);
let mut reg = call_registry.lock().await;
reg.end_call(call_id);
}
}
SignalMessage::Ping { timestamp_ms } => {
let _ = transport.send_signal(&SignalMessage::Pong { timestamp_ms }).await;
}
other => {
warn!(%addr, "signal: unexpected message: {:?}", std::mem::discriminant(&other));
}
}
}
Ok(None) => {
info!(%addr, "signal connection closed");
break;
}
Err(e) => {
warn!(%addr, "signal recv error: {e}");
break;
}
}
}
// Cleanup: unregister + end active calls
let active_calls = {
let reg = call_registry.lock().await;
reg.calls_for_fingerprint(&client_fp)
.iter()
.map(|c| (c.call_id.clone(), if c.caller_fingerprint == client_fp {
c.callee_fingerprint.clone()
} else {
c.caller_fingerprint.clone()
}))
.collect::<Vec<_>>()
};
for (call_id, peer_fp) in &active_calls {
let hub = signal_hub.lock().await;
let _ = hub.send_to(peer_fp, &SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
}).await;
drop(hub);
let mut reg = call_registry.lock().await;
reg.end_call(call_id);
}
{
let mut hub = signal_hub.lock().await;
hub.unregister(&client_fp);
}
{
let mut reg = presence.lock().await;
reg.unregister_local(&client_fp);
}
transport.close().await.ok();
return;
}
// Auth check: if --auth-url is set, expect first signal message to be a token
// Auth: if --auth-url is set, expect AuthToken as first signal
let authenticated_fp: Option<String> = if let Some(ref url) = auth_url {
@@ -431,7 +989,7 @@ async fn main() -> anyhow::Result<()> {
// Crypto handshake: verify client identity + negotiate quality profile
let handshake_start = std::time::Instant::now();
let (_crypto_session, _chosen_profile) = match wzp_relay::handshake::accept_handshake(
let (_crypto_session, _chosen_profile, caller_fp, caller_alias) = match wzp_relay::handshake::accept_handshake(
&*transport,
&relay_seed_bytes,
).await {
@@ -448,10 +1006,35 @@ async fn main() -> anyhow::Result<()> {
}
};
// Use the caller's identity fingerprint from the handshake
let participant_fp = authenticated_fp.clone().unwrap_or(caller_fp);
// ACL: call rooms (call-*) are restricted to the two authorized participants.
// Only the relay's call orchestrator creates these rooms — random clients can't join.
if room_name.starts_with("call-") {
let call_id = &room_name[5..]; // strip "call-" prefix
let authorized = {
let reg = call_registry.lock().await;
match reg.get(call_id) {
Some(call) => {
call.caller_fingerprint == participant_fp
|| call.callee_fingerprint == participant_fp
}
None => false, // unknown call — reject
}
};
if !authorized {
warn!(%addr, room = %room_name, fp = %participant_fp, "rejected: not authorized for this call room");
transport.close().await.ok();
return;
}
info!(%addr, room = %room_name, fp = %participant_fp, "authorized for call room");
}
// Register in presence registry
if let Some(ref fp) = authenticated_fp {
{
let mut reg = presence.lock().await;
reg.register_local(fp, None, Some(room_name.clone()));
reg.register_local(&participant_fp, None, Some(room_name.clone()));
}
info!(%addr, room = %room_name, "client joining");
@@ -500,16 +1083,55 @@ async fn main() -> anyhow::Result<()> {
metrics.active_sessions.inc();
// Call rooms: enforce 2-participant limit
if room_name.starts_with("call-") {
let mgr = room_mgr.lock().await;
if mgr.room_size(&room_name) >= 2 {
drop(mgr);
warn!(%addr, room = %room_name, "call room full (max 2 participants)");
metrics.active_sessions.dec();
let mut smgr = session_mgr.lock().await;
smgr.remove_session(session_id);
transport.close().await.ok();
return;
}
}
let participant_id = {
let mut mgr = room_mgr.lock().await;
match mgr.join(&room_name, addr, room::ParticipantSender::Quic(transport.clone()), authenticated_fp.as_deref()) {
Ok(id) => {
match mgr.join(
&room_name,
addr,
room::ParticipantSender::Quic(transport.clone()),
Some(&participant_fp),
caller_alias.as_deref(),
) {
Ok((id, update, senders)) => {
metrics.active_rooms.set(mgr.list().len() as i64);
drop(mgr); // release lock before async broadcast
// Merge federated participants into RoomUpdate if this is a global room
let merged_update = if let Some(ref fm) = federation_mgr {
if fm.is_global_room(&room_name) {
if let SignalMessage::RoomUpdate { count: _, participants: mut local_parts } = update {
let remote = fm.get_remote_participants(&room_name).await;
local_parts.extend(remote);
// Deduplicate by fingerprint
let mut seen = std::collections::HashSet::new();
local_parts.retain(|p| seen.insert(p.fingerprint.clone()));
SignalMessage::RoomUpdate {
count: local_parts.len() as u32,
participants: local_parts,
}
} else { update }
} else { update }
} else { update };
room::broadcast_signal(&senders, &merged_update).await;
id
}
Err(e) => {
error!(%addr, room = %room_name, "room join denied: {e}");
// Clean up the session we just created
metrics.active_sessions.dec();
let mut smgr = session_mgr.lock().await;
smgr.remove_session(session_id);
@@ -523,6 +1145,25 @@ async fn main() -> anyhow::Result<()> {
.iter()
.map(|b| format!("{b:02x}"))
.collect();
// Set up federation media channel if this is a global room
let (federation_tx, federation_room_hash) = if let Some(ref fm) = federation_mgr {
let is_global = fm.is_global_room(&room_name);
if is_global {
let canonical_hash = fm.global_room_hash(&room_name);
let (tx, rx) = tokio::sync::mpsc::channel(256);
let fm_clone = fm.clone();
tokio::spawn(async move {
wzp_relay::federation::run_federation_media_egress(fm_clone, rx).await;
});
info!(room = %room_name, canonical = ?fm.resolve_global_room(&room_name), "federation egress created (global room)");
(Some(tx), Some(canonical_hash))
} else {
(None, None)
}
} else {
(None, None)
};
room::run_participant(
room_mgr.clone(),
room_name,
@@ -531,6 +1172,9 @@ async fn main() -> anyhow::Result<()> {
metrics.clone(),
&session_id_str,
trunking_enabled,
debug_tap,
federation_tx,
federation_room_hash,
).await;
// Participant disconnected — clean up presence + per-session metrics
@@ -553,4 +1197,5 @@ async fn main() -> anyhow::Result<()> {
}
});
}
Ok(())
}

View File

@@ -16,6 +16,13 @@ pub struct RelayMetrics {
pub bytes_forwarded: IntCounter,
pub auth_attempts: IntCounterVec,
pub handshake_duration: Histogram,
// Federation metrics
pub federation_peer_status: IntGaugeVec,
pub federation_peer_rtt_ms: GaugeVec,
pub federation_packets_forwarded: IntCounterVec,
pub federation_packets_deduped: IntCounter,
pub federation_packets_rate_limited: IntCounter,
pub federation_active_rooms: IntGauge,
// Per-session metrics
pub session_buffer_depth: IntGaugeVec,
pub session_loss_pct: GaugeVec,
@@ -60,6 +67,28 @@ impl RelayMetrics {
)
.expect("metric");
let federation_peer_status = IntGaugeVec::new(
Opts::new("wzp_federation_peer_status", "Peer connection status (0=disconnected, 1=connected)"),
&["peer"],
).expect("metric");
let federation_peer_rtt_ms = GaugeVec::new(
Opts::new("wzp_federation_peer_rtt_ms", "QUIC RTT to federated peer in milliseconds"),
&["peer"],
).expect("metric");
let federation_packets_forwarded = IntCounterVec::new(
Opts::new("wzp_federation_packets_forwarded_total", "Packets forwarded to/from federated peers"),
&["peer", "direction"],
).expect("metric");
let federation_packets_deduped = IntCounter::with_opts(
Opts::new("wzp_federation_packets_deduped_total", "Duplicate federation packets dropped"),
).expect("metric");
let federation_packets_rate_limited = IntCounter::with_opts(
Opts::new("wzp_federation_packets_rate_limited_total", "Federation packets dropped by rate limiter"),
).expect("metric");
let federation_active_rooms = IntGauge::with_opts(
Opts::new("wzp_federation_active_rooms", "Number of federated rooms currently active"),
).expect("metric");
let session_buffer_depth = IntGaugeVec::new(
Opts::new(
"wzp_relay_session_jitter_buffer_depth",
@@ -107,6 +136,12 @@ impl RelayMetrics {
registry.register(Box::new(bytes_forwarded.clone())).expect("register");
registry.register(Box::new(auth_attempts.clone())).expect("register");
registry.register(Box::new(handshake_duration.clone())).expect("register");
registry.register(Box::new(federation_peer_status.clone())).expect("register");
registry.register(Box::new(federation_peer_rtt_ms.clone())).expect("register");
registry.register(Box::new(federation_packets_forwarded.clone())).expect("register");
registry.register(Box::new(federation_packets_deduped.clone())).expect("register");
registry.register(Box::new(federation_packets_rate_limited.clone())).expect("register");
registry.register(Box::new(federation_active_rooms.clone())).expect("register");
registry.register(Box::new(session_buffer_depth.clone())).expect("register");
registry.register(Box::new(session_loss_pct.clone())).expect("register");
registry.register(Box::new(session_rtt_ms.clone())).expect("register");
@@ -120,6 +155,12 @@ impl RelayMetrics {
bytes_forwarded,
auth_attempts,
handshake_duration,
federation_peer_status,
federation_peer_rtt_ms,
federation_packets_forwarded,
federation_packets_deduped,
federation_packets_rate_limited,
federation_active_rooms,
session_buffer_depth,
session_loss_pct,
session_rtt_ms,

View File

@@ -10,7 +10,7 @@ use std::time::Duration;
use bytes::Bytes;
use tokio::sync::Mutex;
use tracing::{error, info, warn};
use tracing::{debug, error, info, trace, warn};
use wzp_proto::packet::TrunkFrame;
use wzp_proto::MediaTransport;
@@ -18,6 +18,38 @@ use wzp_proto::MediaTransport;
use crate::metrics::RelayMetrics;
use crate::trunk::TrunkBatcher;
/// Debug tap: logs packet metadata for matching rooms.
#[derive(Clone)]
pub struct DebugTap {
/// Room name filter ("*" = all rooms, or specific room name/hash).
pub room_filter: String,
}
impl DebugTap {
pub fn matches(&self, room_name: &str) -> bool {
self.room_filter == "*" || self.room_filter == room_name
}
pub fn log_packet(&self, room: &str, dir: &str, addr: &std::net::SocketAddr, pkt: &wzp_proto::MediaPacket, fan_out: usize) {
let h = &pkt.header;
info!(
target: "debug_tap",
room = %room,
dir = dir,
addr = %addr,
seq = h.seq,
codec = ?h.codec_id,
ts = h.timestamp,
fec_block = h.fec_block,
fec_sym = h.fec_symbol,
repair = h.is_repair,
len = pkt.payload.len(),
fan_out,
"TAP"
);
}
}
/// Unique participant ID within a room.
pub type ParticipantId = u64;
@@ -27,6 +59,22 @@ fn next_id() -> ParticipantId {
NEXT_PARTICIPANT_ID.fetch_add(1, Ordering::Relaxed)
}
/// Events emitted by RoomManager for federation to observe.
#[derive(Clone, Debug)]
pub enum RoomEvent {
/// First local participant joined this room.
LocalJoin { room: String },
/// Last local participant left this room.
LocalLeave { room: String },
}
/// Outbound federation media from a local participant.
pub struct FederationMediaOut {
pub room_name: String,
pub room_hash: [u8; 8],
pub data: Bytes,
}
/// How to send data to a participant — either via QUIC transport or WebSocket channel.
#[derive(Clone)]
pub enum ParticipantSender {
@@ -67,11 +115,24 @@ impl ParticipantSender {
}
}
/// Broadcast a signal message to a list of participant senders.
pub async fn broadcast_signal(senders: &[ParticipantSender], msg: &wzp_proto::SignalMessage) {
for sender in senders {
if let ParticipantSender::Quic(t) = sender {
if let Err(e) = t.send_signal(msg).await {
warn!("broadcast_signal error: {e}");
}
}
}
}
/// A participant in a room.
struct Participant {
id: ParticipantId,
_addr: std::net::SocketAddr,
sender: ParticipantSender,
fingerprint: Option<String>,
alias: Option<String>,
}
/// A room holding multiple participants.
@@ -86,10 +147,16 @@ impl Room {
}
}
fn add(&mut self, addr: std::net::SocketAddr, sender: ParticipantSender) -> ParticipantId {
fn add(
&mut self,
addr: std::net::SocketAddr,
sender: ParticipantSender,
fingerprint: Option<String>,
alias: Option<String>,
) -> ParticipantId {
let id = next_id();
info!(room_size = self.participants.len() + 1, participant = id, %addr, "joined room");
self.participants.push(Participant { id, _addr: addr, sender });
self.participants.push(Participant { id, _addr: addr, sender, fingerprint, alias });
id
}
@@ -106,6 +173,23 @@ impl Room {
.collect()
}
/// Build a RoomUpdate participant list.
fn participant_list(&self) -> Vec<wzp_proto::packet::RoomParticipant> {
self.participants
.iter()
.map(|p| wzp_proto::packet::RoomParticipant {
fingerprint: p.fingerprint.clone().unwrap_or_default(),
alias: p.alias.clone(),
relay_label: None, // local participant
})
.collect()
}
/// Get all senders (for broadcasting to everyone including the joiner).
fn all_senders(&self) -> Vec<ParticipantSender> {
self.participants.iter().map(|p| p.sender.clone()).collect()
}
fn is_empty(&self) -> bool {
self.participants.is_empty()
}
@@ -122,24 +206,35 @@ pub struct RoomManager {
/// When `None`, rooms are open (no auth mode). When `Some`, only listed
/// fingerprints can join the corresponding room.
acl: Option<HashMap<String, HashSet<String>>>,
/// Channel for room lifecycle events (federation subscribes).
event_tx: tokio::sync::broadcast::Sender<RoomEvent>,
}
impl RoomManager {
pub fn new() -> Self {
let (event_tx, _) = tokio::sync::broadcast::channel(64);
Self {
rooms: HashMap::new(),
acl: None,
event_tx,
}
}
/// Create a room manager with ACL enforcement enabled.
pub fn with_acl() -> Self {
let (event_tx, _) = tokio::sync::broadcast::channel(64);
Self {
rooms: HashMap::new(),
acl: Some(HashMap::new()),
event_tx,
}
}
/// Subscribe to room lifecycle events (for federation).
pub fn subscribe_events(&self) -> tokio::sync::broadcast::Receiver<RoomEvent> {
self.event_tx.subscribe()
}
/// Grant a fingerprint access to a room.
pub fn allow(&mut self, room_name: &str, fingerprint: &str) {
if let Some(ref mut acl) = self.acl {
@@ -165,20 +260,32 @@ impl RoomManager {
}
}
/// Join a room. Returns the participant ID or an error if unauthorized.
/// Join a room. Returns (participant_id, room_update_msg, all_senders) for broadcasting.
pub fn join(
&mut self,
room_name: &str,
addr: std::net::SocketAddr,
sender: ParticipantSender,
fingerprint: Option<&str>,
) -> Result<ParticipantId, String> {
alias: Option<&str>,
) -> Result<(ParticipantId, wzp_proto::SignalMessage, Vec<ParticipantSender>), String> {
if !self.is_authorized(room_name, fingerprint) {
warn!(room = room_name, fingerprint = ?fingerprint, "unauthorized room join attempt");
return Err("not authorized for this room".to_string());
}
let was_empty = !self.rooms.contains_key(room_name)
|| self.rooms.get(room_name).map_or(true, |r| r.is_empty());
let room = self.rooms.entry(room_name.to_string()).or_insert_with(Room::new);
Ok(room.add(addr, sender))
let id = room.add(addr, sender, fingerprint.map(|s| s.to_string()), alias.map(|s| s.to_string()));
if was_empty {
let _ = self.event_tx.send(RoomEvent::LocalJoin { room: room_name.to_string() });
}
let update = wzp_proto::SignalMessage::RoomUpdate {
count: room.len() as u32,
participants: room.participant_list(),
};
let senders = room.all_senders();
Ok((id, update, senders))
}
/// Join a room via WebSocket. Convenience wrapper around `join()`.
@@ -189,17 +296,49 @@ impl RoomManager {
sender: tokio::sync::mpsc::Sender<Bytes>,
fingerprint: Option<&str>,
) -> Result<ParticipantId, String> {
self.join(room_name, addr, ParticipantSender::WebSocket(sender), fingerprint)
let (id, _update, _senders) = self.join(room_name, addr, ParticipantSender::WebSocket(sender), fingerprint, None)?;
Ok(id)
}
/// Leave a room. Removes the room if empty.
pub fn leave(&mut self, room_name: &str, participant_id: ParticipantId) {
/// Get list of active room names.
pub fn active_rooms(&self) -> Vec<String> {
self.rooms.keys().cloned().collect()
}
/// Get participant list for a room (fingerprint + alias).
pub fn local_participant_list(&self, room_name: &str) -> Vec<wzp_proto::packet::RoomParticipant> {
self.rooms.get(room_name)
.map(|room| room.participant_list())
.unwrap_or_default()
}
/// Get all senders for participants in a room (for federation inbound media delivery).
pub fn local_senders(&self, room_name: &str) -> Vec<ParticipantSender> {
self.rooms.get(room_name)
.map(|room| room.participants.iter()
.map(|p| p.sender.clone())
.collect())
.unwrap_or_default()
}
/// Leave a room. Returns (room_update_msg, remaining_senders) for broadcasting, or None if room is now empty.
pub fn leave(&mut self, room_name: &str, participant_id: ParticipantId) -> Option<(wzp_proto::SignalMessage, Vec<ParticipantSender>)> {
if let Some(room) = self.rooms.get_mut(room_name) {
room.remove(participant_id);
if room.is_empty() {
self.rooms.remove(room_name);
let _ = self.event_tx.send(RoomEvent::LocalLeave { room: room_name.to_string() });
info!(room = room_name, "room closed (empty)");
return None;
}
let update = wzp_proto::SignalMessage::RoomUpdate {
count: room.len() as u32,
participants: room.participant_list(),
};
let senders = room.all_senders();
Some((update, senders))
} else {
None
}
}
@@ -298,6 +437,9 @@ pub async fn run_participant(
metrics: Arc<RelayMetrics>,
session_id: &str,
trunking_enabled: bool,
debug_tap: Option<DebugTap>,
federation_tx: Option<tokio::sync::mpsc::Sender<FederationMediaOut>>,
federation_room_hash: Option<[u8; 8]>,
) {
if trunking_enabled {
run_participant_trunked(
@@ -306,7 +448,7 @@ pub async fn run_participant(
.await;
} else {
run_participant_plain(
room_mgr, room_name, participant_id, transport, metrics, session_id,
room_mgr, room_name, participant_id, transport, metrics, session_id, debug_tap, federation_tx, federation_room_hash,
)
.await;
}
@@ -320,58 +462,145 @@ async fn run_participant_plain(
transport: Arc<wzp_transport::QuinnTransport>,
metrics: Arc<RelayMetrics>,
session_id: &str,
debug_tap: Option<DebugTap>,
federation_tx: Option<tokio::sync::mpsc::Sender<FederationMediaOut>>,
federation_room_hash: Option<[u8; 8]>,
) {
let addr = transport.connection().remote_address();
let mut packets_forwarded = 0u64;
let mut last_recv_instant = std::time::Instant::now();
let mut max_recv_gap_ms = 0u64;
let mut max_forward_ms = 0u64;
let mut send_errors = 0u64;
let mut last_log_instant = std::time::Instant::now();
info!(
room = %room_name,
participant = participant_id,
%addr,
session = session_id,
"forwarding loop started (plain)"
);
loop {
let recv_start = std::time::Instant::now();
let pkt = match transport.recv_media().await {
Ok(Some(pkt)) => pkt,
Ok(None) => {
info!(%addr, participant = participant_id, "disconnected");
info!(%addr, participant = participant_id, forwarded = packets_forwarded, "disconnected (stream ended)");
break;
}
Err(e) => {
let msg = e.to_string();
if msg.contains("timed out") || msg.contains("reset") || msg.contains("closed") {
info!(%addr, participant = participant_id, "connection closed: {e}");
info!(%addr, participant = participant_id, forwarded = packets_forwarded, "connection closed: {e}");
} else {
error!(%addr, participant = participant_id, "recv error: {e}");
error!(%addr, participant = participant_id, forwarded = packets_forwarded, "recv error: {e}");
}
break;
}
};
let recv_gap_ms = last_recv_instant.elapsed().as_millis() as u64;
last_recv_instant = std::time::Instant::now();
if recv_gap_ms > max_recv_gap_ms {
max_recv_gap_ms = recv_gap_ms;
}
// Log if recv gap is suspiciously large (>200ms = missed ~10 packets)
if recv_gap_ms > 200 {
warn!(
room = %room_name,
participant = participant_id,
recv_gap_ms,
seq = pkt.header.seq,
"large recv gap"
);
}
// Update per-session quality metrics if a quality report is present
if let Some(ref report) = pkt.quality_report {
metrics.update_session_quality(session_id, report);
}
// Get current list of other participants
let lock_start = std::time::Instant::now();
let others = {
let mgr = room_mgr.lock().await;
mgr.others(&room_name, participant_id)
};
let lock_ms = lock_start.elapsed().as_millis() as u64;
if lock_ms > 10 {
warn!(
room = %room_name,
participant = participant_id,
lock_ms,
"slow room_mgr lock"
);
}
// Debug tap: log packet metadata
if let Some(ref tap) = debug_tap {
if tap.matches(&room_name) {
tap.log_packet(&room_name, "in", &addr, &pkt, others.len());
}
}
// Forward to all others
let fwd_start = std::time::Instant::now();
let pkt_bytes = pkt.payload.len() as u64;
for other in &others {
match other {
ParticipantSender::Quic(t) => {
let _ = t.send_media(&pkt).await;
if let Err(e) = t.send_media(&pkt).await {
send_errors += 1;
if send_errors <= 5 || send_errors % 100 == 0 {
warn!(
room = %room_name,
participant = participant_id,
peer = %t.connection().remote_address(),
total_send_errors = send_errors,
"send_media error: {e}"
);
}
}
}
ParticipantSender::WebSocket(_) => {
// WS clients receive raw payload bytes
let _ = other.send_raw(&pkt.payload).await;
}
}
}
// Federation: forward to active peer relays via channel
if let Some(ref fed_tx) = federation_tx {
let data = pkt.to_bytes();
let _ = fed_tx.try_send(FederationMediaOut {
room_name: room_name.clone(),
room_hash: federation_room_hash.unwrap_or_else(|| crate::federation::room_hash(&room_name)),
data,
});
}
let fwd_ms = fwd_start.elapsed().as_millis() as u64;
if fwd_ms > max_forward_ms {
max_forward_ms = fwd_ms;
}
if fwd_ms > 50 {
warn!(
room = %room_name,
participant = participant_id,
fwd_ms,
fan_out = others.len(),
"slow forward"
);
}
let fan_out = others.len() as u64;
metrics.packets_forwarded.inc_by(fan_out);
metrics.bytes_forwarded.inc_by(pkt_bytes * fan_out);
packets_forwarded += 1;
if packets_forwarded % 500 == 0 {
// Periodic stats log every 5 seconds
if last_log_instant.elapsed() >= Duration::from_secs(5) {
let room_size = {
let mgr = room_mgr.lock().await;
mgr.room_size(&room_name)
@@ -381,14 +610,24 @@ async fn run_participant_plain(
participant = participant_id,
forwarded = packets_forwarded,
room_size,
fan_out,
max_recv_gap_ms,
max_forward_ms,
send_errors,
"participant stats"
);
max_recv_gap_ms = 0;
max_forward_ms = 0;
last_log_instant = std::time::Instant::now();
}
}
// Clean up
// Clean up — leave room and broadcast update to remaining participants
let mut mgr = room_mgr.lock().await;
mgr.leave(&room_name, participant_id);
if let Some((update, senders)) = mgr.leave(&room_name, participant_id) {
drop(mgr); // release lock before async broadcast
broadcast_signal(&senders, &update).await;
}
}
/// Trunked forwarding loop — batches outgoing packets per peer.
@@ -404,6 +643,19 @@ async fn run_participant_trunked(
let addr = transport.connection().remote_address();
let mut packets_forwarded = 0u64;
let mut last_recv_instant = std::time::Instant::now();
let mut max_recv_gap_ms = 0u64;
let mut max_forward_ms = 0u64;
let mut send_errors = 0u64;
let mut last_log_instant = std::time::Instant::now();
info!(
room = %room_name,
participant = participant_id,
%addr,
session = session_id,
"forwarding loop started (trunked)"
);
// Per-peer TrunkedForwarders, keyed by the raw pointer of the peer
// transport (stable for the Arc's lifetime). We use the remote address
@@ -425,24 +677,50 @@ async fn run_participant_trunked(
let pkt = match result {
Ok(Some(pkt)) => pkt,
Ok(None) => {
info!(%addr, participant = participant_id, "disconnected");
info!(%addr, participant = participant_id, forwarded = packets_forwarded, "disconnected (stream ended)");
break;
}
Err(e) => {
error!(%addr, participant = participant_id, "recv error: {e}");
error!(%addr, participant = participant_id, forwarded = packets_forwarded, "recv error: {e}");
break;
}
};
let recv_gap_ms = last_recv_instant.elapsed().as_millis() as u64;
last_recv_instant = std::time::Instant::now();
if recv_gap_ms > max_recv_gap_ms {
max_recv_gap_ms = recv_gap_ms;
}
if recv_gap_ms > 200 {
warn!(
room = %room_name,
participant = participant_id,
recv_gap_ms,
seq = pkt.header.seq,
"large recv gap (trunked)"
);
}
if let Some(ref report) = pkt.quality_report {
metrics.update_session_quality(session_id, report);
}
let lock_start = std::time::Instant::now();
let others = {
let mgr = room_mgr.lock().await;
mgr.others(&room_name, participant_id)
};
let lock_ms = lock_start.elapsed().as_millis() as u64;
if lock_ms > 10 {
warn!(
room = %room_name,
participant = participant_id,
lock_ms,
"slow room_mgr lock (trunked)"
);
}
let fwd_start = std::time::Instant::now();
let pkt_bytes = pkt.payload.len() as u64;
for other in &others {
match other {
@@ -452,21 +730,44 @@ async fn run_participant_trunked(
.entry(peer_addr)
.or_insert_with(|| TrunkedForwarder::new(t.clone(), sid_bytes));
if let Err(e) = fwd.send(&pkt).await {
let _ = e;
send_errors += 1;
if send_errors <= 5 || send_errors % 100 == 0 {
warn!(
room = %room_name,
participant = participant_id,
peer = %peer_addr,
total_send_errors = send_errors,
"trunked send error: {e}"
);
}
}
}
ParticipantSender::WebSocket(_) => {
// WS clients bypass trunking — send raw payload directly
let _ = other.send_raw(&pkt.payload).await;
}
}
}
let fwd_ms = fwd_start.elapsed().as_millis() as u64;
if fwd_ms > max_forward_ms {
max_forward_ms = fwd_ms;
}
if fwd_ms > 50 {
warn!(
room = %room_name,
participant = participant_id,
fwd_ms,
fan_out = others.len(),
"slow forward (trunked)"
);
}
let fan_out = others.len() as u64;
metrics.packets_forwarded.inc_by(fan_out);
metrics.bytes_forwarded.inc_by(pkt_bytes * fan_out);
packets_forwarded += 1;
if packets_forwarded % 500 == 0 {
// Periodic stats every 5 seconds
if last_log_instant.elapsed() >= Duration::from_secs(5) {
let room_size = {
let mgr = room_mgr.lock().await;
mgr.room_size(&room_name)
@@ -476,15 +777,30 @@ async fn run_participant_trunked(
participant = participant_id,
forwarded = packets_forwarded,
room_size,
fan_out,
max_recv_gap_ms,
max_forward_ms,
send_errors,
"participant stats (trunked)"
);
max_recv_gap_ms = 0;
max_forward_ms = 0;
last_log_instant = std::time::Instant::now();
}
}
_ = flush_interval.tick() => {
for fwd in forwarders.values_mut() {
if let Err(e) = fwd.flush().await {
let _ = e;
send_errors += 1;
if send_errors <= 5 || send_errors % 100 == 0 {
warn!(
room = %room_name,
participant = participant_id,
total_send_errors = send_errors,
"trunk flush error: {e}"
);
}
}
}
}
@@ -497,7 +813,10 @@ async fn run_participant_trunked(
}
let mut mgr = room_mgr.lock().await;
mgr.leave(&room_name, participant_id);
if let Some((update, senders)) = mgr.leave(&room_name, participant_id) {
drop(mgr);
broadcast_signal(&senders, &update).await;
}
}
/// Parse up to the first 2 bytes of a hex session-id string into `[u8; 2]`.

View File

@@ -0,0 +1,105 @@
//! Persistent signaling connection manager.
//!
//! Tracks clients connected via `_signal` SNI. Routes call signals
//! (DirectCallOffer, DirectCallAnswer, Hangup) between registered users.
use std::collections::HashMap;
use std::sync::Arc;
use std::time::Instant;
use tracing::{info, warn};
use wzp_proto::{MediaTransport, SignalMessage};
use wzp_transport::QuinnTransport;
/// A client connected via `_signal` for direct calling.
pub struct SignalClient {
pub fingerprint: String,
pub alias: Option<String>,
pub transport: Arc<QuinnTransport>,
pub connected_at: Instant,
}
/// Manages persistent signaling connections.
pub struct SignalHub {
clients: HashMap<String, SignalClient>,
}
impl SignalHub {
pub fn new() -> Self {
Self {
clients: HashMap::new(),
}
}
/// Register a new signaling client.
pub fn register(&mut self, fp: String, transport: Arc<QuinnTransport>, alias: Option<String>) {
info!(fingerprint = %fp, alias = ?alias, "signal client registered");
self.clients.insert(fp.clone(), SignalClient {
fingerprint: fp,
alias,
transport,
connected_at: Instant::now(),
});
}
/// Unregister a signaling client. Returns the client if found.
pub fn unregister(&mut self, fp: &str) -> Option<SignalClient> {
let client = self.clients.remove(fp);
if client.is_some() {
info!(fingerprint = %fp, "signal client unregistered");
}
client
}
/// Look up a client by fingerprint.
pub fn get(&self, fp: &str) -> Option<&SignalClient> {
self.clients.get(fp)
}
/// Check if a fingerprint is online.
pub fn is_online(&self, fp: &str) -> bool {
self.clients.contains_key(fp)
}
/// Send a signal message to a client by fingerprint.
pub async fn send_to(&self, fp: &str, msg: &SignalMessage) -> Result<(), String> {
match self.clients.get(fp) {
Some(client) => {
client.transport.send_signal(msg).await
.map_err(|e| format!("send to {fp}: {e}"))
}
None => Err(format!("{fp} not online")),
}
}
/// Number of connected signaling clients.
pub fn online_count(&self) -> usize {
self.clients.len()
}
/// List all online fingerprints.
pub fn online_fingerprints(&self) -> Vec<&str> {
self.clients.keys().map(|s| s.as_str()).collect()
}
/// Get alias for a fingerprint.
pub fn alias(&self, fp: &str) -> Option<&str> {
self.clients.get(fp).and_then(|c| c.alias.as_deref())
}
}
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn register_unregister() {
let mut hub = SignalHub::new();
assert_eq!(hub.online_count(), 0);
assert!(!hub.is_online("alice"));
// Can't easily construct QuinnTransport in a unit test,
// so we just test the HashMap logic conceptually.
// Integration tests cover the full flow.
}
}

View File

@@ -16,6 +16,9 @@ async-trait = { workspace = true }
serde_json = "1"
rustls = { version = "0.23", default-features = false, features = ["ring", "std"] }
rcgen = "0.13"
ed25519-dalek = { workspace = true }
hkdf = { workspace = true }
sha2 = { workspace = true }
[dev-dependencies]
tokio = { workspace = true, features = ["rt-multi-thread", "macros"] }

View File

@@ -6,20 +6,74 @@ use std::time::Duration;
use quinn::crypto::rustls::QuicClientConfig;
use quinn::crypto::rustls::QuicServerConfig;
/// Create a server configuration with a self-signed certificate (for testing).
/// Create a server configuration with a self-signed certificate (random keypair).
///
/// Tunes QUIC transport parameters for lossy VoIP:
/// - 30s idle timeout
/// - 5s keep-alive interval
/// - DATAGRAM extension enabled
/// - Conservative flow control for bandwidth-constrained links
/// The certificate changes on every call. Use `server_config_from_seed` for
/// a deterministic certificate that survives relay restarts.
pub fn server_config() -> (quinn::ServerConfig, Vec<u8>) {
let cert_key = rcgen::generate_simple_self_signed(vec!["localhost".to_string()])
.expect("failed to generate self-signed cert");
let cert_der = rustls::pki_types::CertificateDer::from(cert_key.cert);
let key_der =
rustls::pki_types::PrivateKeyDer::try_from(cert_key.key_pair.serialize_der()).unwrap();
build_server_config(cert_der, key_der)
}
/// Create a server configuration with a deterministic self-signed certificate
/// derived from a 32-byte seed. Same seed = same cert = same TLS fingerprint.
pub fn server_config_from_seed(seed: &[u8; 32]) -> (quinn::ServerConfig, Vec<u8>) {
use ed25519_dalek::pkcs8::EncodePrivateKey;
use ed25519_dalek::SigningKey;
use hkdf::Hkdf;
use sha2::Sha256;
// Derive Ed25519 key bytes from seed via HKDF
let hk = Hkdf::<Sha256>::new(None, seed);
let mut ed_bytes = [0u8; 32];
hk.expand(b"wzp-tls-ed25519", &mut ed_bytes)
.expect("HKDF expand failed");
// Create Ed25519 signing key and export as PKCS8 DER
let signing_key = SigningKey::from_bytes(&ed_bytes);
let pkcs8_doc = signing_key.to_pkcs8_der()
.expect("failed to encode Ed25519 key as PKCS8");
let key_der_for_rcgen = rustls::pki_types::PrivateKeyDer::try_from(pkcs8_doc.as_bytes().to_vec())
.expect("failed to wrap PKCS8 DER");
// Create rcgen KeyPair from DER
let key_pair = rcgen::KeyPair::from_der_and_sign_algo(
&key_der_for_rcgen,
&rcgen::PKCS_ED25519,
)
.expect("failed to create KeyPair from seed-derived Ed25519 key");
// Build self-signed cert with this deterministic keypair
let params = rcgen::CertificateParams::new(vec!["localhost".to_string()])
.expect("failed to create CertificateParams");
let cert = params.self_signed(&key_pair).expect("failed to self-sign cert");
let cert_der = rustls::pki_types::CertificateDer::from(cert.der().to_vec());
let key_der = rustls::pki_types::PrivateKeyDer::try_from(key_pair.serialize_der())
.expect("failed to serialize key DER");
build_server_config(cert_der, key_der)
}
/// Compute a hex-formatted SHA-256 fingerprint of a DER-encoded certificate.
///
/// Format: `xx:xx:xx:xx:...` (32 bytes = 64 hex chars with colons).
pub fn tls_fingerprint(cert_der: &[u8]) -> String {
use sha2::{Sha256, Digest};
let hash = Sha256::digest(cert_der);
hash.iter()
.map(|b| format!("{b:02x}"))
.collect::<Vec<_>>()
.join(":")
}
fn build_server_config(
cert_der: rustls::pki_types::CertificateDer<'static>,
key_der: rustls::pki_types::PrivateKeyDer<'static>,
) -> (quinn::ServerConfig, Vec<u8>) {
let mut server_crypto = rustls::ServerConfig::builder()
.with_no_client_auth()
.with_single_cert(vec![cert_der.clone()], key_der)

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@@ -22,8 +22,13 @@ pub mod path_monitor;
pub mod quic;
pub mod reliable;
pub use config::{client_config, server_config};
pub use config::{client_config, server_config, server_config_from_seed, tls_fingerprint};
pub use connection::{accept, connect, create_endpoint};
pub use path_monitor::PathMonitor;
pub use quic::QuinnTransport;
pub use wzp_proto::{MediaTransport, PathQuality, TransportError};
// Re-export the quinn Endpoint type so downstream crates (wzp-desktop) can
// thread a shared endpoint between signaling and media connections without
// needing to depend on quinn directly.
pub use quinn::Endpoint;

View File

@@ -136,6 +136,11 @@ impl PathMonitor {
}
}
/// Get raw packet counts for debugging.
pub fn counts(&self) -> (u64, u64) {
(self.total_sent, self.total_received)
}
/// Estimate bandwidth in kbps from bytes received over time.
fn estimate_bandwidth_kbps(&self) -> u32 {
if let (Some(first), Some(last)) = (self.first_recv_time_ms, self.last_recv_time_ms) {
@@ -149,6 +154,27 @@ impl PathMonitor {
}
0
}
/// Detect whether a network handoff likely occurred.
///
/// Returns `true` if the most recent RTT jitter measurement exceeds 3x
/// the EWMA-smoothed jitter average, which is characteristic of a cellular
/// network handoff (tower switch, WiFi-to-cellular transition, etc.).
pub fn detect_handoff(&self) -> bool {
// We need at least two RTT observations to have a meaningful jitter value,
// and the EWMA must be non-zero to avoid division/multiplication by zero.
if self.jitter_ewma <= 0.0 {
return false;
}
if let (Some(last_rtt), Some(_)) = (self.last_rtt_ms, Some(self.rtt_ewma)) {
// Compute the most recent instantaneous jitter (RTT deviation from EWMA)
let instant_jitter = (last_rtt - self.rtt_ewma).abs();
instant_jitter > self.jitter_ewma * 3.0
} else {
false
}
}
}
impl Default for PathMonitor {

View File

@@ -33,6 +33,29 @@ impl QuinnTransport {
&self.connection
}
/// Send raw bytes as a QUIC datagram (no MediaPacket framing).
pub fn send_raw_datagram(&self, data: &[u8]) -> Result<(), TransportError> {
self.connection
.send_datagram(bytes::Bytes::copy_from_slice(data))
.map_err(|e| TransportError::Internal(format!("datagram: {e}")))
}
/// Close the QUIC connection immediately (synchronous, no async needed).
/// The relay will detect the close and remove this participant from the room.
pub fn close_now(&self) {
self.connection.close(quinn::VarInt::from_u32(0), b"hangup");
}
/// Feed an external RTT observation (e.g. from QUIC path stats) into the path monitor.
pub fn feed_rtt(&self, rtt_ms: u32) {
self.path_monitor.lock().unwrap().observe_rtt(rtt_ms);
}
/// Get raw packet counts from path monitor (sent, received).
pub fn monitor_counts(&self) -> (u64, u64) {
self.path_monitor.lock().unwrap().counts()
}
/// Get the maximum datagram payload size, if datagrams are supported.
pub fn max_datagram_size(&self) -> Option<usize> {
datagram::max_datagram_payload(&self.connection)
@@ -120,7 +143,7 @@ impl MediaTransport for QuinnTransport {
}
};
match datagram::deserialize_media(data) {
match datagram::deserialize_media(data.clone()) {
Some(packet) => {
// Record receive observation
{
@@ -133,8 +156,10 @@ impl MediaTransport for QuinnTransport {
Ok(Some(packet))
}
None => {
tracing::warn!("received malformed media datagram");
Ok(None)
tracing::warn!(len = data.len(), "skipping malformed media datagram, continuing");
// Don't return Ok(None) — that signals connection closed.
// Recurse to read the next datagram instead.
Box::pin(self.recv_media()).await
}
}
}

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