merge opus-DRED-v2 into main
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50 commits: BT audio routing, network change detection, Hangup call_id,
per-arch APK builds, setCommunicationDevice API 31+, deferred
MODE_IN_COMMUNICATION, Oboe BT mode, build signing, doc updates.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
This commit is contained in:
Siavash Sameni
2026-04-12 17:41:57 +04:00
200 changed files with 50886 additions and 1215 deletions

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@@ -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"

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@@ -2,207 +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)'
required: false
default: 'amd64'
env:
CARGO_TERM_COLOR: always
jobs:
build-amd64:
if: >-
github.event_name == 'push' ||
contains(github.event.inputs.targets, 'amd64')
runs-on: ubuntu-latest
container:
image: catthehacker/ubuntu:act-latest
steps:
- name: Checkout
run: |
apt-get update && apt-get install -y git curl jq
AUTH_URL="${{ github.server_url }}/${{ github.repository }}.git"
AUTH_URL=$(echo "$AUTH_URL" | sed "s|://|://${{ github.token }}@|")
git clone --depth 1 --branch ${{ github.ref_name }} "$AUTH_URL" .
git config --global url."https://git.tbs.amn.gg/manawenuz/featherChat.git".insteadOf "ssh://git@git.manko.yoga:222/manawenuz/featherChat.git"
git submodule update --init --depth 1
- uses: actions/checkout@v4
- name: Install Rust and dependencies
- name: Init submodules
run: |
apt-get install -y cmake pkg-config libasound2-dev libssl-dev build-essential
curl --proto '=https' --tlsv1.2 -sSf https://sh.rustup.rs | sh -s -- -y --default-toolchain stable
echo "$HOME/.cargo/bin" >> $GITHUB_PATH
git config --global url."https://git.manko.yoga/".insteadOf "ssh://git@git.manko.yoga:222/"
git submodule update --init --recursive
- name: Build headless binaries
- name: Install Rust + dependencies
run: |
export PATH="$HOME/.cargo/bin:$PATH"
curl --proto '=https' --tlsv1.2 -sSf https://sh.rustup.rs | sh -s -- -y
source "$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 "$HOME/.cargo/env"
cargo build --release --bin wzp-relay --bin wzp-client --bin wzp-bench --bin wzp-web
- name: Build audio client
run: |
export PATH="$HOME/.cargo/bin:$PATH"
cargo build --release --bin wzp-client --features audio
cp target/release/wzp-client target/release/wzp-client-audio
cargo build --release --bin wzp-client
- name: Run tests
run: |
export PATH="$HOME/.cargo/bin:$PATH"
source "$HOME/.cargo/env"
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/
- name: Upload to release
run: |
API="${{ github.server_url }}/api/v1/repos/${{ github.repository }}"
TOKEN="${{ github.token }}"
TAG="build-$(date +%Y%m%d-%H%M%S)"
SHA=$(git rev-parse --short HEAD)
# Create release
RELEASE=$(curl -sS -X POST "$API/releases" \
-H "Authorization: token $TOKEN" \
-H "Content-Type: application/json" \
-d "{\"tag_name\":\"$TAG\",\"name\":\"Build $SHA (amd64)\",\"body\":\"Automated build from ${{ github.ref_name }} at $SHA\",\"draft\":false,\"prerelease\":true}")
RELEASE_ID=$(echo "$RELEASE" | jq -r '.id')
# Upload artifact
curl -sS -X POST "$API/releases/$RELEASE_ID/assets?name=wzp-linux-amd64.tar.gz" \
-H "Authorization: token $TOKEN" \
-H "Content-Type: application/octet-stream" \
--data-binary @dist/wzp-linux-amd64.tar.gz
echo "Release created: ${{ github.server_url }}/${{ github.repository }}/releases/tag/$TAG"
build-arm64:
if: >-
github.event_name == 'push' ||
contains(github.event.inputs.targets, 'arm64')
runs-on: ubuntu-latest
steps:
- name: Checkout
run: |
apt-get update && apt-get install -y git curl jq
AUTH_URL="${{ github.server_url }}/${{ github.repository }}.git"
AUTH_URL=$(echo "$AUTH_URL" | sed "s|://|://${{ github.token }}@|")
git clone --depth 1 --branch ${{ github.ref_name }} "$AUTH_URL" .
git config --global url."https://git.tbs.amn.gg/manawenuz/featherChat.git".insteadOf "ssh://git@git.manko.yoga:222/manawenuz/featherChat.git"
git submodule update --init --depth 1
- name: Install Rust and 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 libssl-dev build-essential
curl --proto '=https' --tlsv1.2 -sSf https://sh.rustup.rs | sh -s -- -y --default-toolchain stable
export PATH="$HOME/.cargo/bin:$PATH"
rustup target add aarch64-unknown-linux-gnu
echo "$HOME/.cargo/bin" >> $GITHUB_PATH
- 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: ${{ secrets.PASTE_URL }}
run: |
export PATH="$HOME/.cargo/bin:$PATH"
cargo build --release --target aarch64-unknown-linux-gnu \
--bin wzp-relay --bin wzp-client --bin wzp-bench --bin wzp-web
- name: Package
run: |
mkdir -p dist/wzp-linux-arm64
cp target/aarch64-unknown-linux-gnu/release/wzp-relay dist/wzp-linux-arm64/
cp target/aarch64-unknown-linux-gnu/release/wzp-client dist/wzp-linux-arm64/
cp target/aarch64-unknown-linux-gnu/release/wzp-web dist/wzp-linux-arm64/
cp target/aarch64-unknown-linux-gnu/release/wzp-bench dist/wzp-linux-arm64/
cp -r crates/wzp-web/static dist/wzp-linux-arm64/
cd dist && tar czf wzp-linux-arm64.tar.gz wzp-linux-arm64/
- name: Upload to release
run: |
API="${{ github.server_url }}/api/v1/repos/${{ github.repository }}"
TOKEN="${{ github.token }}"
TAG="build-arm64-$(date +%Y%m%d-%H%M%S)"
SHA=$(git rev-parse --short HEAD)
RELEASE=$(curl -sS -X POST "$API/releases" \
-H "Authorization: token $TOKEN" \
-H "Content-Type: application/json" \
-d "{\"tag_name\":\"$TAG\",\"name\":\"Build $SHA (arm64)\",\"body\":\"Automated build from ${{ github.ref_name }} at $SHA\",\"draft\":false,\"prerelease\":true}")
RELEASE_ID=$(echo "$RELEASE" | jq -r '.id')
curl -sS -X POST "$API/releases/$RELEASE_ID/assets?name=wzp-linux-arm64.tar.gz" \
-H "Authorization: token $TOKEN" \
-H "Content-Type: application/octet-stream" \
--data-binary @dist/wzp-linux-arm64.tar.gz
echo "Release created: ${{ github.server_url }}/${{ github.repository }}/releases/tag/$TAG"
build-armv7:
if: >-
github.event_name == 'push' ||
contains(github.event.inputs.targets, 'armv7')
runs-on: ubuntu-latest
steps:
- name: Checkout
run: |
apt-get update && apt-get install -y git curl jq
AUTH_URL="${{ github.server_url }}/${{ github.repository }}.git"
AUTH_URL=$(echo "$AUTH_URL" | sed "s|://|://${{ github.token }}@|")
git clone --depth 1 --branch ${{ github.ref_name }} "$AUTH_URL" .
git config --global url."https://git.tbs.amn.gg/manawenuz/featherChat.git".insteadOf "ssh://git@git.manko.yoga:222/manawenuz/featherChat.git"
git submodule update --init --depth 1
- name: Install Rust and 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 libssl-dev build-essential
curl --proto '=https' --tlsv1.2 -sSf https://sh.rustup.rs | sh -s -- -y --default-toolchain stable
export PATH="$HOME/.cargo/bin:$PATH"
rustup target add armv7-unknown-linux-gnueabihf
echo "$HOME/.cargo/bin" >> $GITHUB_PATH
- name: Build
env:
CARGO_TARGET_ARMV7_UNKNOWN_LINUX_GNUEABIHF_LINKER: arm-linux-gnueabihf-gcc
CC_armv7_unknown_linux_gnueabihf: arm-linux-gnueabihf-gcc
run: |
export PATH="$HOME/.cargo/bin:$PATH"
cargo build --release --target armv7-unknown-linux-gnueabihf \
--bin wzp-relay --bin wzp-client --bin wzp-bench --bin wzp-web
- name: Package
run: |
mkdir -p dist/wzp-linux-armv7
cp target/armv7-unknown-linux-gnueabihf/release/wzp-relay dist/wzp-linux-armv7/
cp target/armv7-unknown-linux-gnueabihf/release/wzp-client dist/wzp-linux-armv7/
cp target/armv7-unknown-linux-gnueabihf/release/wzp-web dist/wzp-linux-armv7/
cp target/armv7-unknown-linux-gnueabihf/release/wzp-bench dist/wzp-linux-armv7/
cp -r crates/wzp-web/static dist/wzp-linux-armv7/
cd dist && tar czf wzp-linux-armv7.tar.gz wzp-linux-armv7/
- name: Upload to release
run: |
API="${{ github.server_url }}/api/v1/repos/${{ github.repository }}"
TOKEN="${{ github.token }}"
TAG="build-armv7-$(date +%Y%m%d-%H%M%S)"
SHA=$(git rev-parse --short HEAD)
RELEASE=$(curl -sS -X POST "$API/releases" \
-H "Authorization: token $TOKEN" \
-H "Content-Type: application/json" \
-d "{\"tag_name\":\"$TAG\",\"name\":\"Build $SHA (armv7)\",\"body\":\"Automated build from ${{ github.ref_name }} at $SHA\",\"draft\":false,\"prerelease\":true}")
RELEASE_ID=$(echo "$RELEASE" | jq -r '.id')
curl -sS -X POST "$API/releases/$RELEASE_ID/assets?name=wzp-linux-armv7.tar.gz" \
-H "Authorization: token $TOKEN" \
-H "Content-Type: application/octet-stream" \
--data-binary @dist/wzp-linux-armv7.tar.gz
echo "Release created: ${{ github.server_url }}/${{ github.repository }}/releases/tag/$TAG"
tar czf /tmp/wzp-linux-amd64.tar.gz \
-C target/release wzp-relay wzp-client wzp-web wzp-bench
ls -lh /tmp/wzp-linux-amd64.tar.gz
LINK=$(curl -sF "file=@/tmp/wzp-linux-amd64.tar.gz" \
-H "Authorization: ${PASTE_AUTH}" \
"https://${PASTE_URL}")
echo "Download: ${LINK}"

25
.gitignore vendored
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@@ -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

4104
Cargo.lock generated

File diff suppressed because it is too large Load Diff

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@@ -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]
@@ -29,17 +32,25 @@ serde = { version = "1", features = ["derive"] }
# Transport
quinn = "0.11"
socket2 = "0.5"
# FEC
raptorq = "2"
# Codec
audiopus = "0.3.0-rc.0"
# opusic-c: high-level safe bindings over libopus 1.5.2 (encoder side).
# opusic-sys: raw FFI for the decoder side — we build our own DecoderHandle
# because opusic-c::Decoder.inner is pub(crate) and cannot be reached for the
# Phase 3 DRED reconstruction path. See docs/PRD-dred-integration.md.
# Pinned exactly (no caret) for reproducible libopus 1.5.2 across the fleet.
opusic-c = { version = "=1.5.5", default-features = false, features = ["bundled", "dred"] }
opusic-sys = { version = "=0.6.0", default-features = false, features = ["bundled"] }
bytemuck = "1"
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 +63,29 @@ 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.opusic-sys]
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
# Phase 0 (opus-DRED): removed the [patch.crates-io] audiopus_sys = { path =
# "vendor/audiopus_sys" } block. That patch existed to fix a Windows clang-cl
# SIMD compile bug in libopus 1.3.1. With the swap to opusic-sys (libopus
# 1.5.2), the upstream SIMD gating was fixed and the vendor patch is
# obsolete. The vendor/audiopus_sys directory itself should be deleted as
# part of the same cleanup — see the commit that follows this Phase 0.

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

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@@ -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
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@@ -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.** { *; }

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@@ -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>

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@@ -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"
}
}

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@@ -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 ->
// Phase 4: extract DRED / classical PLC counters from the
// stats JSON so they're visible in the meta preamble at a
// glance, not buried in the trailing JSON dump.
val dredReconstructions = extractLongField(finalStatsJson, "dred_reconstructions")
val classicalPlc = extractLongField(finalStatsJson, "classical_plc_invocations")
val framesDecoded = extractLongField(finalStatsJson, "frames_decoded")
val fecRecovered = extractLongField(finalStatsJson, "fec_recovered")
// 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("=== Loss Recovery ===")
appendLine("Frames decoded: $framesDecoded")
appendLine("DRED reconstructions: $dredReconstructions (Opus neural recovery)")
appendLine("Classical PLC: $classicalPlc (fallback)")
appendLine("RaptorQ FEC recovered: $fecRecovered (Codec2 only)")
if (framesDecoded > 0) {
val dredPct = 100.0 * dredReconstructions / framesDecoded
val plcPct = 100.0 * classicalPlc / framesDecoded
appendLine("DRED rate: ${"%.2f".format(dredPct)}%")
appendLine("Classical PLC rate: ${"%.2f".format(plcPct)}%")
}
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: ${e.message}"
}
}
private fun addWavEntry(zos: ZipOutputStream, name: String, pcmFile: File) {
val dataSize = pcmFile.length().toInt()
val byteRate = SAMPLE_RATE * 1 * 16 / 8
val blockAlign = 1 * 16 / 8
zos.putNextEntry(ZipEntry(name))
// Write WAV header (44 bytes)
val header = ByteBuffer.allocate(44).order(ByteOrder.LITTLE_ENDIAN)
header.put("RIFF".toByteArray())
header.putInt(36 + dataSize)
header.put("WAVE".toByteArray())
header.put("fmt ".toByteArray())
header.putInt(16)
header.putShort(1) // PCM
header.putShort(1) // mono
header.putInt(SAMPLE_RATE)
header.putInt(byteRate)
header.putShort(blockAlign.toShort())
header.putShort(16) // bits per sample
header.put("data".toByteArray())
header.putInt(dataSize)
zos.write(header.array())
// Stream PCM data directly (avoids loading entire file into memory)
FileInputStream(pcmFile).use { it.copyTo(zos) }
zos.closeEntry()
}
private fun addTextEntry(zos: ZipOutputStream, name: String, content: String) {
zos.putNextEntry(ZipEntry(name))
zos.write(content.toByteArray())
zos.closeEntry()
}
private fun addFileEntry(zos: ZipOutputStream, name: String, file: File) {
zos.putNextEntry(ZipEntry(name))
FileInputStream(file).use { it.copyTo(zos) }
zos.closeEntry()
}
/**
* Tiny JSON field extractor — pulls an integer value for a top-level
* field like `"dred_reconstructions":42`. We don't want to pull in a
* full JSON parser just for the debug preamble, and the CallStats
* output is a flat record with well-known field names.
*
* Returns 0 if the field is missing or unparseable.
*/
private fun extractLongField(json: String, field: String): Long {
val key = "\"$field\":"
val idx = json.indexOf(key)
if (idx < 0) return 0
var i = idx + key.length
// Skip whitespace
while (i < json.length && json[i].isWhitespace()) i++
val start = i
while (i < json.length && (json[i].isDigit() || json[i] == '-')) i++
return try {
json.substring(start, i).toLong()
} catch (_: NumberFormatException) {
0
}
}
}

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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)
}
/**
* Signal a network transport change (e.g. WiFi → LTE handoff).
*
* @param networkType matches Rust `NetworkContext` ordinals:
* 0=WiFi, 1=LTE, 2=5G, 3=3G, 4=Unknown, 5=None
* @param bandwidthKbps reported downstream bandwidth in kbps
*/
fun onNetworkChanged(networkType: Int, bandwidthKbps: Int) {
if (nativeHandle != 0L) nativeOnNetworkChanged(nativeHandle, networkType, bandwidthKbps)
}
/** 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
private external fun nativeOnNetworkChanged(handle: Long, networkType: Int, bandwidthKbps: 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
import android.content.Context
import android.net.ConnectivityManager
import android.net.Network
import android.net.NetworkCapabilities
import android.net.NetworkRequest
import android.os.Handler
import android.os.Looper
/**
* Monitors network connectivity changes via [ConnectivityManager.NetworkCallback]
* and classifies the active transport (WiFi, LTE, 5G, 3G).
*
* Callbacks fire on the main looper so callers can safely update UI state or
* dispatch to a native engine from any callback.
*
* Usage:
* 1. Set [onNetworkChanged] to receive `(type: Int, downlinkKbps: Int)` events
* 2. Optionally set [onIpChanged] for IP address change events (mid-call ICE refresh)
* 3. Call [register] when the call starts
* 4. Call [unregister] when the call ends
*/
class NetworkMonitor(context: Context) {
private val cm = context.getSystemService(Context.CONNECTIVITY_SERVICE) as ConnectivityManager
private val mainHandler = Handler(Looper.getMainLooper())
/**
* Called when the network transport type or bandwidth changes.
* `type` constants match the Rust `NetworkContext` enum ordinals.
*/
var onNetworkChanged: ((type: Int, downlinkKbps: Int) -> Unit)? = null
/**
* Called when the device's IP address changes (link properties changed).
* Useful for triggering mid-call ICE candidate re-gathering.
*/
var onIpChanged: (() -> Unit)? = null
// Track the last emitted type to avoid redundant callbacks
@Volatile
private var lastEmittedType: Int = TYPE_UNKNOWN
private val callback = object : ConnectivityManager.NetworkCallback() {
override fun onAvailable(network: Network) {
classifyAndEmit(network)
}
override fun onCapabilitiesChanged(network: Network, caps: NetworkCapabilities) {
classifyFromCaps(caps)
}
override fun onLinkPropertiesChanged(
network: Network,
linkProperties: android.net.LinkProperties
) {
// IP address may have changed — notify for ICE refresh
onIpChanged?.invoke()
// Also re-classify in case the transport changed simultaneously
classifyAndEmit(network)
}
override fun onLost(network: Network) {
lastEmittedType = TYPE_NONE
onNetworkChanged?.invoke(TYPE_NONE, 0)
}
}
// -- Public API -----------------------------------------------------------
/** Register the network callback. Call when a call starts. */
fun register() {
val request = NetworkRequest.Builder()
.addCapability(NetworkCapabilities.NET_CAPABILITY_INTERNET)
.build()
cm.registerNetworkCallback(request, callback, mainHandler)
}
/** Unregister the network callback. Call when the call ends. */
fun unregister() {
try {
cm.unregisterNetworkCallback(callback)
} catch (_: IllegalArgumentException) {
// Already unregistered — safe to ignore
}
}
// -- Classification -------------------------------------------------------
private fun classifyAndEmit(network: Network) {
val caps = cm.getNetworkCapabilities(network) ?: return
classifyFromCaps(caps)
}
private fun classifyFromCaps(caps: NetworkCapabilities) {
val type = when {
caps.hasTransport(NetworkCapabilities.TRANSPORT_WIFI) -> TYPE_WIFI
caps.hasTransport(NetworkCapabilities.TRANSPORT_ETHERNET) -> TYPE_WIFI // treat as WiFi
caps.hasTransport(NetworkCapabilities.TRANSPORT_CELLULAR) -> classifyCellular(caps)
else -> TYPE_UNKNOWN
}
val bw = caps.getLinkDownstreamBandwidthKbps()
// Deduplicate: only emit when the transport type actually changes
if (type != lastEmittedType) {
lastEmittedType = type
onNetworkChanged?.invoke(type, bw)
}
}
/**
* Approximate cellular generation from reported downstream bandwidth.
* This avoids requiring READ_PHONE_STATE permission (needed for
* TelephonyManager.getNetworkType on API 30+).
*
* Thresholds are conservative — carriers over-report bandwidth, so we
* classify based on what's actually usable for VoIP:
* - >= 100 Mbps → 5G NR
* - >= 10 Mbps → LTE
* - < 10 Mbps → 3G or worse
*/
private fun classifyCellular(caps: NetworkCapabilities): Int {
val bw = caps.getLinkDownstreamBandwidthKbps()
return when {
bw >= 100_000 -> TYPE_CELLULAR_5G
bw >= 10_000 -> TYPE_CELLULAR_LTE
else -> TYPE_CELLULAR_3G
}
}
companion object {
/** Constants matching Rust `NetworkContext` enum ordinals. */
const val TYPE_WIFI = 0
const val TYPE_CELLULAR_LTE = 1
const val TYPE_CELLULAR_5G = 2
const val TYPE_CELLULAR_3G = 3
const val TYPE_UNKNOWN = 4
const val TYPE_NONE = 5
}
}

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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
)
}

View File

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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.AudioRoute
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 com.wzp.net.NetworkMonitor
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 networkMonitor: NetworkMonitor? = 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 _audioRoute = MutableStateFlow(AudioRoute.EARPIECE)
val audioRoute: StateFlow<AudioRoute> = _audioRoute.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).also { arm ->
arm.onRouteChanged = { route ->
_audioRoute.value = route
_isSpeaker.value = (route == AudioRoute.SPEAKER)
}
}
}
if (networkMonitor == null) {
networkMonitor = NetworkMonitor(appCtx).also { nm ->
nm.onNetworkChanged = { type, bw ->
engine?.onNetworkChanged(type, bw)
}
}
}
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)
}
/** Cycle audio output: Earpiece → Speaker → Bluetooth (if available) → Earpiece. */
fun cycleAudioRoute() {
val routes = audioRouteManager?.availableRoutes() ?: return
val currentIdx = routes.indexOf(_audioRoute.value)
val next = routes[(currentIdx + 1) % routes.size]
when (next) {
AudioRoute.EARPIECE -> {
audioRouteManager?.setBluetoothSco(false)
audioRouteManager?.setSpeaker(false)
}
AudioRoute.SPEAKER -> {
audioRouteManager?.setSpeaker(true)
}
AudioRoute.BLUETOOTH -> {
audioRouteManager?.setBluetoothSco(true)
}
}
_audioRoute.value = next
_isSpeaker.value = (next == AudioRoute.SPEAKER)
}
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()
networkMonitor?.register()
audioStarted = true
}
private fun stopAudio() {
if (!audioStarted) return
audioPipeline?.stop() // sets running=false; DON'T null — teardown needs awaitDrain()
audioRouteManager?.unregister()
networkMonitor?.unregister()
audioRouteManager?.setSpeaker(false)
_isSpeaker.value = false
_audioRoute.value = AudioRoute.EARPIECE
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") }
}
)
}

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<?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
}

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@@ -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.

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@@ -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 "$CLASSPATH" org.gradle.wrapper.GradleWrapperMain "$@"

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@@ -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")

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@@ -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

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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,
}
}

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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

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@@ -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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//! 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);
}));
}
/// Signal a network transport change from the Android ConnectivityManager.
///
/// `network_type` matches the Rust `NetworkContext` enum:
/// 0=WiFi, 1=CellularLte, 2=Cellular5g, 3=Cellular3g, 4=Unknown, 5=None
///
/// The engine forwards this to the `AdaptiveQualityController` which:
/// - Preemptively downgrades one tier on WiFi→cellular
/// - Activates a 10-second FEC boost
/// - Uses faster downgrade thresholds on cellular
#[unsafe(no_mangle)]
pub unsafe extern "system" fn Java_com_wzp_engine_WzpEngine_nativeOnNetworkChanged(
_env: JNIEnv,
_class: JClass,
handle: jlong,
network_type: jint,
bandwidth_kbps: jint,
) {
let _ = panic::catch_unwind(panic::AssertUnwindSafe(|| {
let h = unsafe { handle_ref(handle) };
h.engine.on_network_changed(network_type as u8, bandwidth_kbps as u32);
}));
}
/// 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 }
}
}

View File

@@ -0,0 +1,31 @@
//! 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.
//!
//! ## Status
//!
//! **Dead code as of the Tauri mobile rewrite.** The legacy Kotlin+JNI
//! Android app that consumed this crate was replaced by a Tauri 2.x
//! Mobile app (see `desktop/src-tauri/src/engine.rs` for the live
//! Android audio recv path and `crates/wzp-native/` for the Oboe
//! bridge). We keep this crate in the workspace for reference and to
//! preserve the commit history, but it is not built by any shipping
//! target. Allow the accumulated leftover warnings so CI/workspace
//! checks stay clean — any real cleanup should happen as part of
//! removing the crate entirely, not piecemeal.
#![allow(dead_code, unused_imports, unused_variables, unused_mut)]
pub mod audio_android;
pub mod audio_ring;
pub mod commands;
pub mod engine;
pub mod pipeline;
pub mod stats;
pub mod jni_bridge;

View File

@@ -0,0 +1,262 @@
//! 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 +1,109 @@
//! 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 RaptorQ FEC (Codec2 tiers only; Opus bypasses
/// RaptorQ per Phase 2).
pub fec_recovered: u64,
/// Phase 3c: Opus frames reconstructed via DRED side-channel data.
/// Only increments on the Opus tiers; always zero for Codec2.
pub dred_reconstructions: u64,
/// Phase 3c: Opus frames filled via classical Opus PLC because no DRED
/// state covered the gap, plus any decode-error fallbacks. Codec2 loss
/// also increments this counter via the Codec2 PLC path.
pub classical_plc_invocations: 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 +23,77 @@ 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"
# Phase 5.5 — LAN host-candidate ICE: enumerate local network
# interface addresses for inclusion in DirectCallOffer/Answer so
# peers on the same LAN can direct-connect without NAT hairpinning
# through the WAN reflex addr (which many consumer NATs, including
# MikroTik's default masquerade, don't support).
if-addrs = "0.13"
# 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)
}
}

View File

@@ -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();
}
}

View File

@@ -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

@@ -7,14 +7,15 @@ use std::time::{Duration, Instant};
use bytes::Bytes;
use tracing::{debug, info, warn};
use wzp_codec::{ComfortNoise, NoiseSupressor, SilenceDetector};
use wzp_codec::dred_ffi::{DredDecoderHandle, DredState};
use wzp_codec::{
AdaptiveDecoder, AutoGainControl, ComfortNoise, EchoCanceller, NoiseSupressor, SilenceDetector,
};
use wzp_fec::{RaptorQFecDecoder, RaptorQFecEncoder};
use wzp_proto::jitter::{JitterBuffer, PlayoutResult};
use wzp_proto::packet::{MediaHeader, MediaPacket, MiniFrameContext};
use wzp_proto::quality::AdaptiveQualityController;
use wzp_proto::traits::{
AudioDecoder, AudioEncoder, FecDecoder, FecEncoder,
};
use wzp_proto::traits::{AudioDecoder, AudioEncoder, FecDecoder, FecEncoder};
use wzp_proto::packet::QualityReport;
use wzp_proto::{CodecId, QualityProfile};
@@ -42,6 +43,9 @@ 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 +67,7 @@ impl Default for CallConfig {
noise_suppression: true,
mini_frames_enabled: true,
adaptive_jitter: true,
aec_delay_ms: 40,
}
}
}
@@ -207,6 +212,10 @@ 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 +246,8 @@ 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 +285,21 @@ 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) {
@@ -328,6 +345,22 @@ impl CallEncoder {
let enc_len = self.audio_enc.encode(pcm, &mut encoded)?;
encoded.truncate(enc_len);
// Phase 2: Opus tiers bypass RaptorQ entirely (DRED handles loss
// recovery at the codec layer). Codec2 tiers keep RaptorQ unchanged.
// On Opus packets, zero the FEC header fields so old receivers
// can cleanly identify "no RaptorQ block to assemble" and new
// receivers can short-circuit their FEC ingest path.
let is_opus = self.profile.codec.is_opus();
let (fec_block, fec_symbol, fec_ratio_encoded) = if is_opus {
(0u8, 0u8, 0u8)
} else {
(
self.block_id,
self.frame_in_block,
MediaHeader::encode_fec_ratio(self.profile.fec_ratio),
)
};
// Build source media packet
let source_pkt = MediaPacket {
header: MediaHeader {
@@ -335,11 +368,11 @@ impl CallEncoder {
is_repair: false,
codec_id: self.profile.codec,
has_quality_report: false,
fec_ratio_encoded: MediaHeader::encode_fec_ratio(self.profile.fec_ratio),
fec_ratio_encoded,
seq: self.seq,
timestamp: self.timestamp_ms,
fec_block: self.block_id,
fec_symbol: self.frame_in_block,
fec_block,
fec_symbol,
reserved: 0,
csrc_count: 0,
},
@@ -354,11 +387,13 @@ impl CallEncoder {
let mut output = vec![source_pkt];
// Add to FEC encoder
// Codec2-only: feed RaptorQ and generate repair packets when the
// block is full. Opus tiers skip this entire block — DRED (active
// in Phase 1) provides codec-layer loss recovery.
if !is_opus {
self.fec_enc.add_source_symbol(&encoded)?;
self.frame_in_block += 1;
// If block is full, generate repair and finalize
if self.frame_in_block >= self.profile.frames_per_block {
if let Ok(repairs) = self.fec_enc.generate_repair(self.profile.fec_ratio) {
for (sym_idx, repair_data) in repairs {
@@ -388,6 +423,7 @@ impl CallEncoder {
self.block_id = self.block_id.wrapping_add(1);
self.frame_in_block = 0;
}
}
Ok(output)
}
@@ -400,13 +436,34 @@ 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.
pub struct CallDecoder {
/// Audio decoder.
audio_dec: Box<dyn AudioDecoder>,
/// FEC decoder.
/// Audio decoder. Concrete `AdaptiveDecoder` (not `Box<dyn AudioDecoder>`)
/// because Phase 3b calls the inherent `reconstruct_from_dred` method,
/// which cannot live on the `AudioDecoder` trait without dragging libopus
/// types into `wzp-proto`.
audio_dec: AdaptiveDecoder,
/// FEC decoder (Codec2 tiers only; Opus bypasses RaptorQ per Phase 2).
fec_dec: RaptorQFecDecoder,
/// Jitter buffer.
jitter: JitterBuffer,
@@ -420,6 +477,24 @@ pub struct CallDecoder {
last_was_cn: bool,
/// Mini-frame decompression context (tracks last full header baseline).
mini_context: MiniFrameContext,
// ─── Phase 3b: DRED reconstruction state ──────────────────────────────
/// DRED side-channel parser (a separate libopus object from the decoder).
dred_decoder: DredDecoderHandle,
/// Scratch buffer used by `dred_decoder.parse_into` on every arriving
/// Opus packet. Reused across calls to avoid 10 KB alloc churn per packet.
dred_parse_scratch: DredState,
/// Cached "most recently parsed valid" DRED state, swapped with
/// `dred_parse_scratch` on successful parse. Used by `decode_next` when
/// the jitter buffer reports a gap.
last_good_dred: DredState,
/// Sequence number of the packet that produced `last_good_dred`. `None`
/// if no packet has yielded DRED state yet (cold start or legacy sender).
last_good_dred_seq: Option<u16>,
/// Phase 4 telemetry counter: gaps recovered via DRED reconstruction.
pub dred_reconstructions: u64,
/// Phase 4 telemetry counter: gaps filled via classical Opus PLC
/// (because no DRED state covered the gap, or the active codec is Codec2).
pub classical_plc_invocations: u64,
}
impl CallDecoder {
@@ -429,8 +504,19 @@ impl CallDecoder {
} else {
JitterBuffer::new(config.jitter_target, config.jitter_max, config.jitter_min)
};
// Phase 3b: build the DRED parser + state buffers. These allocate
// libopus state (~10 KB each) once per call, not per packet — the
// scratch and last-good buffers are reused via std::mem::swap on
// every successful parse.
let dred_decoder =
DredDecoderHandle::new().expect("opus_dred_decoder_create failed at call setup");
let dred_parse_scratch =
DredState::new().expect("opus_dred_alloc failed at call setup (scratch)");
let last_good_dred =
DredState::new().expect("opus_dred_alloc failed at call setup (good state)");
Self {
audio_dec: wzp_codec::create_decoder(config.profile),
audio_dec: AdaptiveDecoder::new(config.profile)
.expect("failed to create adaptive decoder"),
fec_dec: wzp_fec::create_decoder(&config.profile),
jitter,
quality: AdaptiveQualityController::new(),
@@ -438,6 +524,12 @@ impl CallDecoder {
comfort_noise: ComfortNoise::new(50),
last_was_cn: false,
mini_context: MiniFrameContext::default(),
dred_decoder,
dred_parse_scratch,
last_good_dred,
last_good_dred_seq: None,
dred_reconstructions: 0,
classical_plc_invocations: 0,
}
}
@@ -452,20 +544,105 @@ impl CallDecoder {
/// Feed a received media packet into the decode pipeline.
pub fn ingest(&mut self, packet: MediaPacket) {
// Feed to FEC decoder
// Phase 2: Opus packets bypass RaptorQ. Codec2 packets still feed
// the FEC decoder for recovery. This also cleanly drops any stray
// Opus repair packets from an old sender (we don't push repair
// packets to the jitter buffer either, so they're effectively
// ignored — a graceful mixed-version degradation).
if !packet.header.codec_id.is_opus() {
let _ = self.fec_dec.add_symbol(
packet.header.fec_block,
packet.header.fec_symbol,
packet.header.is_repair,
&packet.payload,
);
}
// If not a repair packet, also feed directly to jitter buffer
// Phase 3b: Opus source packets carry DRED side-channel data in
// libopus 1.5. Parse it into the scratch state and, on success,
// swap with the cached `last_good_dred` so later gap reconstruction
// has fresh neural redundancy to draw from. Parsing happens before
// the jitter push because the jitter buffer consumes the packet.
if packet.header.codec_id.is_opus() && !packet.header.is_repair {
match self
.dred_decoder
.parse_into(&mut self.dred_parse_scratch, &packet.payload)
{
Ok(available) if available > 0 => {
// Swap the freshly parsed state into `last_good_dred`.
// The old good state (now in scratch) is about to be
// overwritten on the next parse — its contents are
// not needed after this swap.
std::mem::swap(&mut self.dred_parse_scratch, &mut self.last_good_dred);
self.last_good_dred_seq = Some(packet.header.seq);
}
Ok(_) => {
// Packet had no DRED data (return 0). Leave the cached
// state untouched — it may still cover upcoming gaps
// from a warm-up period where the encoder was producing
// DRED bytes. The scratch buffer was potentially written
// but its `samples_available` is 0 so it's harmless.
}
Err(e) => {
debug!("DRED parse error (ignored): {e}");
}
}
}
// Source packets (Opus or Codec2) go to the jitter buffer for decode.
// Repair packets never reach the jitter buffer; for Codec2 they're
// used by the FEC decoder above, for Opus they're dropped here.
if !packet.header.is_repair {
self.jitter.push(packet);
}
}
/// 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 +657,9 @@ 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),
@@ -494,19 +674,72 @@ impl CallDecoder {
result
}
PlayoutResult::Missing { seq } => {
// Only generate PLC if there are still packets buffered ahead.
// Only attempt recovery if there are still packets buffered ahead.
// Otherwise we've drained everything — return None to stop.
if self.jitter.depth() > 0 {
debug!(seq, "packet loss, generating PLC");
if self.jitter.depth() == 0 {
self.jitter.record_underrun();
return None;
}
// Phase 3b: try DRED reconstruction first. If we have a
// recent DRED state from a packet whose seq > missing seq,
// and the seq delta (in samples) fits within the state's
// available window, libopus can synthesize a plausible
// replacement for the lost frame. Fall back to classical
// PLC when no state covers the gap, when the active codec
// is Codec2, or when the reconstruction itself errors.
if self.profile.codec.is_opus() {
if let Some(last_seq) = self.last_good_dred_seq {
// How many frames ahead of the missing seq is the
// last-good packet? Use wrapping arithmetic for the
// u16 seq space.
let seq_delta = last_seq.wrapping_sub(seq);
// Reject stale or backward state. u16 wraparound
// would make a "seq went backward" delta very large;
// cap at a sane forward-looking window.
const MAX_SEQ_DELTA: u16 = 128;
if seq_delta > 0 && seq_delta <= MAX_SEQ_DELTA {
let frame_samples =
(48_000 * self.profile.frame_duration_ms as i32) / 1000;
let offset_samples = seq_delta as i32 * frame_samples;
let available = self.last_good_dred.samples_available();
if offset_samples > 0 && offset_samples <= available {
match self.audio_dec.reconstruct_from_dred(
&self.last_good_dred,
offset_samples,
pcm,
) {
Ok(n) => {
self.dred_reconstructions += 1;
self.jitter.record_decode();
debug!(
seq,
last_seq,
offset_samples,
available,
"DRED reconstruction for gap"
);
return Some(n);
}
Err(e) => {
// Reconstruction failed — fall
// through to classical PLC below.
debug!(seq, "DRED reconstruct error: {e}");
}
}
}
}
}
}
// Classical PLC fallback (also the Codec2 path).
debug!(seq, "packet loss, generating classical PLC");
self.classical_plc_invocations += 1;
let result = self.audio_dec.decode_lost(pcm).ok();
if result.is_some() {
self.jitter.record_decode();
}
result
} else {
self.jitter.record_underrun();
None
}
}
PlayoutResult::NotReady => {
self.jitter.record_underrun();
@@ -529,6 +762,19 @@ impl CallDecoder {
pub fn reset_stats(&mut self) {
self.jitter.reset_stats();
}
/// Phase 3b introspection: sequence number of the most recently parsed
/// valid DRED state, or `None` if no Opus packet has yielded DRED data
/// yet. Used by tests to debug reconstruction eligibility.
pub fn last_good_dred_seq(&self) -> Option<u16> {
self.last_good_dred_seq
}
/// Phase 3b introspection: samples of audio history currently available
/// in the cached DRED state.
pub fn last_good_dred_samples_available(&self) -> i32 {
self.last_good_dred.samples_available()
}
}
/// Periodic telemetry logger for jitter buffer statistics.
@@ -590,18 +836,83 @@ mod tests {
assert!(!packets[0].header.is_repair);
}
/// Phase 2: Opus packets have zero FEC header fields — no block, no
/// symbol index, no repair ratio. The RaptorQ layer is bypassed
/// entirely on the Opus tiers.
#[test]
fn encoder_generates_repair_on_full_block() {
fn opus_source_packets_have_zero_fec_header_fields() {
let config = CallConfig {
profile: QualityProfile::GOOD, // 5 frames/block
profile: QualityProfile::GOOD, // Opus 24k
suppression_enabled: false, // skip silence gate for this test
..Default::default()
};
let mut enc = CallEncoder::new(&config);
let pcm = vec![0i16; 960];
// Non-silent sine wave so silence detection doesn't suppress us
// even with suppression_enabled=false (belt and braces).
let pcm: Vec<i16> = (0..960)
.map(|i| ((i as f32 * 0.1).sin() * 10_000.0) as i16)
.collect();
let packets = enc.encode_frame(&pcm).unwrap();
assert_eq!(packets.len(), 1, "Opus must emit exactly 1 source packet");
let hdr = &packets[0].header;
assert!(hdr.codec_id.is_opus());
assert!(!hdr.is_repair);
assert_eq!(hdr.fec_block, 0, "Opus fec_block must be 0");
assert_eq!(hdr.fec_symbol, 0, "Opus fec_symbol must be 0");
assert_eq!(hdr.fec_ratio_encoded, 0, "Opus fec_ratio_encoded must be 0");
}
let mut total_packets = 0;
let mut repair_count = 0;
for _ in 0..5 {
/// Phase 2: Opus never emits repair packets, regardless of how many
/// source frames are fed in. DRED (Phase 1) provides loss recovery at
/// the codec layer; RaptorQ is disabled on Opus tiers.
#[test]
fn opus_encoder_never_emits_repair_packets() {
let config = CallConfig {
profile: QualityProfile::GOOD, // 5 frames/block in the Codec2 sense
suppression_enabled: false,
..Default::default()
};
let mut enc = CallEncoder::new(&config);
let pcm: Vec<i16> = (0..960)
.map(|i| ((i as f32 * 0.1).sin() * 10_000.0) as i16)
.collect();
// Encode well beyond a block boundary to prove no repair ever comes out.
let mut total_packets = 0usize;
let mut repair_count = 0usize;
for _ in 0..20 {
let packets = enc.encode_frame(&pcm).unwrap();
total_packets += packets.len();
repair_count += packets.iter().filter(|p| p.header.is_repair).count();
}
assert_eq!(repair_count, 0, "Opus must emit zero repair packets");
assert_eq!(
total_packets, 20,
"20 source frames → 20 source packets (1:1, no RaptorQ expansion)"
);
}
/// Phase 2: Codec2 still emits repair packets with RaptorQ ratio unchanged.
/// DRED is libopus-only and does not apply here, so RaptorQ is still the
/// primary loss-recovery mechanism on Codec2 tiers.
#[test]
fn codec2_encoder_generates_repair_on_full_block() {
let config = CallConfig {
profile: QualityProfile::CATASTROPHIC, // Codec2 1200, 8 frames/block, ratio 1.0
suppression_enabled: false,
..Default::default()
};
let mut enc = CallEncoder::new(&config);
// Codec2 takes 48 kHz samples and downsamples internally.
// CATASTROPHIC uses 40 ms frames → 1920 samples.
let pcm: Vec<i16> = (0..1920)
.map(|i| ((i as f32 * 0.1).sin() * 10_000.0) as i16)
.collect();
let mut total_packets = 0usize;
let mut repair_count = 0usize;
// Run long enough to cross the 8-frame block boundary and see repairs.
for _ in 0..16 {
let packets = enc.encode_frame(&pcm).unwrap();
for p in &packets {
if p.header.is_repair {
@@ -610,8 +921,10 @@ mod tests {
}
total_packets += packets.len();
}
assert!(repair_count > 0, "should have repair packets after full block");
assert!(total_packets > 5, "total {total_packets} should exceed 5 source");
assert!(
repair_count > 0,
"Codec2 must still emit repair packets (got {repair_count} repairs, {total_packets} total)"
);
}
#[test]
@@ -642,6 +955,219 @@ mod tests {
assert!(dec.decode_next(&mut pcm).is_none());
}
// ─── Phase 3b — DRED reconstruction on packet loss ────────────────────
/// Helper: create a CallEncoder/CallDecoder pair with the given profile
/// and silence suppression disabled so silence-detection doesn't drop
/// our synthetic test frames.
fn encoder_decoder_pair(profile: QualityProfile) -> (CallEncoder, CallDecoder) {
let config = CallConfig {
profile,
suppression_enabled: false,
// Small jitter buffer so decode_next drains quickly in tests.
jitter_min: 2,
jitter_target: 3,
jitter_max: 20,
adaptive_jitter: false,
..Default::default()
};
(CallEncoder::new(&config), CallDecoder::new(&config))
}
/// Helper: generate a non-silent 20 ms frame of 300 Hz sine at the
/// given sample offset so consecutive frames form a continuous tone.
fn voice_frame_20ms(sample_offset: usize) -> Vec<i16> {
(0..960)
.map(|i| {
let t = (sample_offset + i) as f64 / 48_000.0;
(8000.0 * (2.0 * std::f64::consts::PI * 300.0 * t).sin()) as i16
})
.collect()
}
/// Phase 3b probe: sweep packet_loss_perc values to find the minimum
/// that produces a samples_available ≥ 960 (enough to reconstruct a
/// single 20 ms Opus frame). This guides the production loss floor.
#[test]
#[ignore] // diagnostic only — run with `cargo test ... -- --ignored --nocapture`
fn probe_dred_samples_available_by_loss_floor() {
use wzp_codec::opus_enc::OpusEncoder;
use wzp_proto::traits::AudioEncoder;
for loss_pct in [5u8, 10, 15, 20, 25, 40, 60, 80].iter().copied() {
let mut enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
enc.set_expected_loss(loss_pct);
let (_drop_enc, mut dec) = encoder_decoder_pair(QualityProfile::GOOD);
for i in 0..60u16 {
let pcm = voice_frame_20ms(i as usize * 960);
let mut encoded = vec![0u8; 512];
let n = enc.encode(&pcm, &mut encoded).unwrap();
encoded.truncate(n);
let pkt = MediaPacket {
header: MediaHeader {
version: 0,
is_repair: false,
codec_id: CodecId::Opus24k,
has_quality_report: false,
fec_ratio_encoded: 0,
seq: i,
timestamp: (i as u32) * 20,
fec_block: 0,
fec_symbol: 0,
reserved: 0,
csrc_count: 0,
},
payload: Bytes::from(encoded),
quality_report: None,
};
dec.ingest(pkt);
}
eprintln!(
"[phase3b probe] loss_pct={loss_pct} samples_available={}",
dec.last_good_dred_samples_available()
);
}
}
/// Phase 3b: simulated single-packet loss on an Opus call triggers a
/// DRED reconstruction rather than a classical PLC fill. Runs the full
/// encode → ingest → decode_next pipeline.
#[test]
fn opus_single_packet_loss_is_recovered_via_dred() {
let (mut enc, mut dec) = encoder_decoder_pair(QualityProfile::GOOD);
// Warm-up: encode and ingest 60 frames (1.2 s) so the DRED emitter
// has had time to fill its 200 ms window and at least one
// successful DRED parse has happened on the decoder side.
let warmup_frames = 60;
for i in 0..warmup_frames {
let pcm = voice_frame_20ms(i * 960);
let packets = enc.encode_frame(&pcm).unwrap();
for pkt in packets {
dec.ingest(pkt);
}
}
// Drain the warm-up frames through the decoder to advance the
// jitter buffer cursor past them.
let mut out = vec![0i16; 960];
while dec.decode_next(&mut out).is_some() {}
// Encode the next three frames but skip ingesting the middle one.
let base_offset = warmup_frames * 960;
let pcm_a = voice_frame_20ms(base_offset);
let pcm_b = voice_frame_20ms(base_offset + 960);
let pcm_c = voice_frame_20ms(base_offset + 1920);
let pkts_a = enc.encode_frame(&pcm_a).unwrap();
let pkts_b = enc.encode_frame(&pcm_b).unwrap(); // DROP THIS ONE
let pkts_c = enc.encode_frame(&pcm_c).unwrap();
for pkt in pkts_a {
dec.ingest(pkt);
}
// Skip pkts_b entirely — this is the "packet loss".
drop(pkts_b);
for pkt in pkts_c {
dec.ingest(pkt);
}
// Drain again. Somewhere in here decode_next will hit Missing()
// for the dropped packet and attempt DRED reconstruction.
let baseline_dred = dec.dred_reconstructions;
let baseline_plc = dec.classical_plc_invocations;
eprintln!(
"[phase3b probe] pre-drain: last_good_seq={:?} samples_available={}",
dec.last_good_dred_seq(),
dec.last_good_dred_samples_available()
);
while dec.decode_next(&mut out).is_some() {}
let dred_delta = dec.dred_reconstructions - baseline_dred;
let plc_delta = dec.classical_plc_invocations - baseline_plc;
eprintln!(
"[phase3b probe] post-drain: dred_delta={dred_delta} plc_delta={plc_delta}"
);
assert!(
dred_delta >= 1,
"expected ≥1 DRED reconstruction on single-packet loss, \
got dred_delta={dred_delta} plc_delta={plc_delta}"
);
}
/// Phase 3b: lossless stream never triggers DRED reconstruction or PLC.
/// Baseline behavior — verifies the Missing() branch is not spuriously taken.
#[test]
fn opus_lossless_ingest_never_triggers_dred_or_plc() {
let (mut enc, mut dec) = encoder_decoder_pair(QualityProfile::GOOD);
// Encode + ingest 40 frames with no drops.
for i in 0..40 {
let pcm = voice_frame_20ms(i * 960);
let packets = enc.encode_frame(&pcm).unwrap();
for pkt in packets {
dec.ingest(pkt);
}
}
let mut out = vec![0i16; 960];
while dec.decode_next(&mut out).is_some() {}
assert_eq!(
dec.dred_reconstructions, 0,
"lossless stream should not reconstruct"
);
assert_eq!(
dec.classical_plc_invocations, 0,
"lossless stream should not PLC"
);
}
/// Phase 3b: Codec2 calls fall through to classical PLC on loss.
/// DRED is libopus-only, so even if the decoder's DRED state were
/// populated (it won't be — Codec2 packets don't carry DRED bytes),
/// `reconstruct_from_dred` rejects Codec2 at the AdaptiveDecoder
/// level. This test guards the Codec2 side of the protection split.
#[test]
fn codec2_loss_falls_through_to_classical_plc() {
let (mut enc, mut dec) = encoder_decoder_pair(QualityProfile::CATASTROPHIC);
// Codec2 1200 uses 40 ms frames → 1920 samples at 48 kHz (before
// the downsample inside the codec). Encode 20 frames (~0.8 s).
let make_frame = |offset: usize| -> Vec<i16> {
(0..1920)
.map(|i| {
let t = (offset + i) as f64 / 48_000.0;
(8000.0 * (2.0 * std::f64::consts::PI * 300.0 * t).sin()) as i16
})
.collect()
};
for i in 0..20 {
let pcm = make_frame(i * 1920);
let packets = enc.encode_frame(&pcm).unwrap();
for pkt in packets {
// Drop every 5th source packet to simulate loss.
if !pkt.header.is_repair && i % 5 == 3 {
continue;
}
dec.ingest(pkt);
}
}
let mut out = vec![0i16; 1920];
while dec.decode_next(&mut out).is_some() {}
assert_eq!(
dec.dred_reconstructions, 0,
"Codec2 must never reconstruct via DRED"
);
// classical_plc_invocations may or may not trigger depending on
// whether the jitter buffer sees Missing before draining — the key
// assertion is that DRED is not used. PLC count is advisory.
}
// ---- QualityAdapter tests ----
/// Helper: build a QualityReport from human-readable loss% and RTT ms.

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");
@@ -361,6 +424,7 @@ async fn run_silence(transport: Arc<wzp_transport::QuinnTransport>) -> anyhow::R
info!(total_source, total_repair, total_bytes, "done — closing");
let hangup = wzp_proto::SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
call_id: None,
};
transport.send_signal(&hangup).await.ok();
transport.close().await?;
@@ -512,6 +576,7 @@ async fn run_file_mode(
// Send Hangup signal so the relay knows we're done
let hangup = wzp_proto::SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
call_id: None,
};
transport.send_signal(&hangup).await.ok();
@@ -563,11 +628,21 @@ async fn run_live(transport: Arc<wzp_transport::QuinnTransport>) -> anyhow::Resu
.spawn(move || {
let config = CallConfig::default();
let mut encoder = CallEncoder::new(&config);
let mut frame = vec![0i16; FRAME_SAMPLES];
loop {
let frame = match capture.read_frame() {
Some(f) => f,
None => break,
};
// Pull a full 20 ms frame from the capture ring. The ring
// may return a partial read when the CPAL callback hasn't
// produced enough samples yet — keep reading until we
// accumulate a whole frame, sleeping briefly on empty
// returns so we don't hot-spin the CPU.
let mut filled = 0usize;
while filled < FRAME_SAMPLES {
let n = capture.ring().read(&mut frame[filled..]);
filled += n;
if n == 0 {
std::thread::sleep(std::time::Duration::from_millis(2));
}
}
let packets = match encoder.encode_frame(&frame) {
Ok(p) => p,
Err(e) => {
@@ -598,7 +673,13 @@ async fn run_live(transport: Arc<wzp_transport::QuinnTransport>) -> anyhow::Resu
// Repair packets feed the FEC decoder but don't produce audio.
if !is_repair {
if let Some(_n) = decoder.decode_next(&mut pcm_buf) {
playback.write_frame(&pcm_buf);
// Push the decoded frame into the playback
// ring. The CPAL output callback drains from
// here on its own clock; if the ring is full
// (rare in CLI live mode) the write returns
// a short count and the tail is dropped,
// which is the correct real-time behavior.
playback.ring().write(&pcm_buf);
}
}
}
@@ -623,3 +704,205 @@ 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],
// CLI client doesn't attempt hole-punching; always
// relay-path.
caller_reflexive_addr: None,
caller_local_addrs: Vec::new(),
caller_build_version: None,
}).await?;
}
// Signal recv loop — handle incoming signals
let signal_transport = transport.clone();
let relay = relay_addr;
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),
// CLI auto-accept uses generic (privacy) mode,
// so callee addr stays hidden from the caller.
callee_reflexive_addr: None,
callee_local_addrs: Vec::new(),
callee_build_version: None,
}).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, peer_direct_addr: _, peer_local_addrs: _ } => {
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,
call_id: None,
}).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

@@ -0,0 +1,546 @@
//! Phase 3.5 — dual-path QUIC connect race for P2P hole-punching.
//!
//! When both peers advertised reflex addrs in the
//! DirectCallOffer/Answer flow, the relay cross-wires them into
//! `CallSetup.peer_direct_addr`. This module races a direct QUIC
//! handshake against the existing relay dial and returns whichever
//! completes first — with automatic drop of the loser via
//! `tokio::select!`.
//!
//! Role determination is deterministic and symmetric
//! (`wzp_client::reflect::determine_role`): whichever peer has the
//! lexicographically smaller reflex addr becomes the **Acceptor**
//! (listens on a server-capable endpoint), the other becomes the
//! **Dialer** (dials the peer's addr). Because the rule is
//! identical on both sides, the Acceptor's inbound QUIC session
//! and the Dialer's outbound are the SAME connection — no
//! negotiation needed, no two-conns-per-call confusion.
//!
//! Timeout policy:
//! - Direct path: 2s from the start of `race`. Cone-NAT hole-punch
//! typically completes in < 500ms on a LAN; 2s gives us tolerance
//! for a single QUIC Initial retry on unreliable networks.
//! - Relay path: 10s (existing behavior elsewhere in the codebase).
//! - Overall: `tokio::select!` returns as soon as either succeeds.
use std::net::SocketAddr;
use std::sync::Arc;
use std::time::Duration;
use crate::reflect::Role;
use wzp_transport::QuinnTransport;
/// Which path won the race. Used by the `connect` command for
/// logging + (in the future) metrics.
#[derive(Debug, Clone, Copy, PartialEq, Eq)]
pub enum WinningPath {
Direct,
Relay,
}
/// Phase 6: the race now returns BOTH transports (when available)
/// so the connect command can negotiate with the peer before
/// committing. The negotiation decides which transport to use
/// based on whether BOTH sides report `direct_ok = true`.
pub struct RaceResult {
/// The direct P2P transport, if the direct path completed.
/// `None` if the direct dial/accept failed or timed out.
pub direct_transport: Option<Arc<QuinnTransport>>,
/// The relay transport, if the relay dial completed.
/// `None` if the relay dial failed (shouldn't happen in
/// practice since relay is always reachable).
pub relay_transport: Option<Arc<QuinnTransport>>,
/// Which future completed first in the local race.
/// Informational — the actual path used is decided by the
/// Phase 6 negotiation after both sides exchange reports.
pub local_winner: WinningPath,
}
/// Attempt a direct QUIC connection to the peer in parallel with
/// the relay dial and return the winning `QuinnTransport`.
///
/// `role` selects the direction of the direct attempt:
/// - `Role::Acceptor` creates a server-capable endpoint and waits
/// for the peer to dial in.
/// - `Role::Dialer` creates a client-only endpoint and dials
/// `peer_direct_addr`.
///
/// The relay path is always attempted in parallel as a fallback so
/// the race ALWAYS produces a working transport unless both paths
/// genuinely fail (network partition). Returns
/// `Err(anyhow::anyhow!(...))` if both paths fail within the
/// timeout.
/// Phase 5.5 candidate bundle — full ICE-ish candidate list for
/// the peer. The race tries them all in parallel alongside the
/// relay path. At minimum this should contain the peer's
/// server-reflexive address; `local_addrs` carries LAN host
/// candidates gathered from their physical interfaces.
///
/// Empty is valid: the D-role has nothing to dial and the race
/// reduces to "relay only" + (if A-role) accepting on the
/// shared endpoint.
#[derive(Debug, Clone, Default)]
pub struct PeerCandidates {
/// Peer's server-reflexive address (Phase 3). `None` if the
/// peer didn't advertise one.
pub reflexive: Option<SocketAddr>,
/// Peer's LAN host addresses (Phase 5.5). Tried first on
/// same-LAN pairs — direct dials to these bypass the NAT
/// entirely.
pub local: Vec<SocketAddr>,
}
impl PeerCandidates {
/// Flatten into the list of addrs the D-role should dial.
/// Order: LAN host candidates first (fastest when they
/// work), then reflexive (covers the non-LAN case).
pub fn dial_order(&self) -> Vec<SocketAddr> {
let mut out = Vec::with_capacity(self.local.len() + 1);
out.extend(self.local.iter().copied());
if let Some(a) = self.reflexive {
// Only add if it's not already in the list (some
// edge cases on same-LAN could have the same addr
// in both).
if !out.contains(&a) {
out.push(a);
}
}
out
}
/// Is there anything for the D-role to dial? If not, the
/// race reduces to relay-only.
pub fn is_empty(&self) -> bool {
self.reflexive.is_none() && self.local.is_empty()
}
}
#[allow(clippy::too_many_arguments)]
pub async fn race(
role: Role,
peer_candidates: PeerCandidates,
relay_addr: SocketAddr,
room_sni: String,
call_sni: String,
// Phase 5: when `Some`, reuse this endpoint for BOTH the
// direct-path branch AND the relay dial. Pass the signal
// endpoint. The endpoint MUST be server-capable (created
// with a server config) for the A-role accept branch to
// work.
//
// When `None`, falls back to fresh endpoints per role.
// Used by tests.
shared_endpoint: Option<wzp_transport::Endpoint>,
// Phase 7: dedicated IPv6 endpoint with IPV6_V6ONLY=1.
// When `Some`, A-role accepts on both v4+v6, D-role routes
// each candidate to its matching-AF endpoint. When `None`,
// IPv6 candidates are skipped (IPv4-only, pre-Phase-7).
ipv6_endpoint: Option<wzp_transport::Endpoint>,
) -> anyhow::Result<RaceResult> {
// Rustls provider must be installed before any quinn endpoint
// is created. Install attempt is idempotent.
let _ = rustls::crypto::ring::default_provider().install_default();
// Build the direct-path endpoint + future based on role.
//
// A-role: one accept future on the shared endpoint. The
// first incoming QUIC connection wins — we don't care
// which peer candidate the dialer used to reach us.
//
// D-role: N parallel dial futures, one per peer candidate
// (all LAN host addrs + the reflex addr), consolidated
// into a single direct_fut via FuturesUnordered-style
// "first OK wins" semantics. The first successful dial
// becomes the direct path; the losers are dropped (quinn
// will abort the in-flight handshakes via the dropped
// Connecting futures).
//
// Either way, direct_fut resolves to a single QuinnTransport
// (or an error) and is raced against the relay_fut by the
// outer tokio::select!.
let direct_ep: wzp_transport::Endpoint;
let direct_fut: std::pin::Pin<
Box<dyn std::future::Future<Output = anyhow::Result<QuinnTransport>> + Send>,
>;
match role {
Role::Acceptor => {
let ep = match shared_endpoint.clone() {
Some(ep) => {
tracing::info!(
local_addr = ?ep.local_addr().ok(),
"dual_path: A-role reusing shared endpoint for accept"
);
ep
}
None => {
let (sc, _cert_der) = wzp_transport::server_config();
// 0.0.0.0:0 = IPv4 socket. [::]:0 dual-stack was
// tried but breaks on Android devices where
// IPV6_V6ONLY=1 (default on some kernels) —
// IPv4 candidates silently fail. IPv6 host
// candidates are skipped for now; they need a
// dedicated IPv6 socket alongside the v4 one
// (like WebRTC's dual-socket approach).
let bind: SocketAddr = "0.0.0.0:0".parse().unwrap();
let fresh = wzp_transport::create_endpoint(bind, Some(sc))?;
tracing::info!(
local_addr = ?fresh.local_addr().ok(),
"dual_path: A-role fresh endpoint up, awaiting peer dial"
);
fresh
}
};
let ep_for_fut = ep.clone();
// Phase 7: IPv6 accept temporarily disabled (same reason
// as dial — IPv6 connections die on datagram send).
// Accept on IPv4 shared endpoint only.
let _v6_ep_unused = ipv6_endpoint.clone();
direct_fut = Box::pin(async move {
// Accept loop: retry if we get a stale/closed
// connection from a previous call. Max 3 retries
// to avoid spinning until the race timeout.
const MAX_STALE: usize = 3;
let mut stale_count: usize = 0;
loop {
let conn = wzp_transport::accept(&ep_for_fut)
.await
.map_err(|e| anyhow::anyhow!("direct accept: {e}"))?;
if let Some(reason) = conn.close_reason() {
// Explicitly close so the peer gets a
// close frame instead of idle timeout.
conn.close(0u32.into(), b"stale");
stale_count += 1;
tracing::warn!(
remote = %conn.remote_address(),
stable_id = conn.stable_id(),
stale_count,
?reason,
"dual_path: A-role skipping stale connection"
);
if stale_count >= MAX_STALE {
return Err(anyhow::anyhow!(
"A-role: {stale_count} stale connections, aborting"
));
}
continue;
}
let has_dgram = conn.max_datagram_size().is_some();
tracing::info!(
remote = %conn.remote_address(),
stable_id = conn.stable_id(),
has_dgram,
"dual_path: A-role accepted direct connection"
);
break Ok(QuinnTransport::new(conn));
}
});
direct_ep = ep;
}
Role::Dialer => {
let ep = match shared_endpoint.clone() {
Some(ep) => {
tracing::info!(
local_addr = ?ep.local_addr().ok(),
candidates = ?peer_candidates.dial_order(),
"dual_path: D-role reusing shared endpoint to dial peer candidates"
);
ep
}
None => {
// 0.0.0.0:0 = IPv4 socket. [::]:0 dual-stack was
// tried but breaks on Android devices where
// IPV6_V6ONLY=1 (default on some kernels) —
// IPv4 candidates silently fail. IPv6 host
// candidates are skipped for now; they need a
// dedicated IPv6 socket alongside the v4 one
// (like WebRTC's dual-socket approach).
let bind: SocketAddr = "0.0.0.0:0".parse().unwrap();
let fresh = wzp_transport::create_endpoint(bind, None)?;
tracing::info!(
local_addr = ?fresh.local_addr().ok(),
candidates = ?peer_candidates.dial_order(),
"dual_path: D-role fresh endpoint up, dialing peer candidates"
);
fresh
}
};
let ep_for_fut = ep.clone();
let _v6_ep_for_dial = ipv6_endpoint.clone();
let dial_order = peer_candidates.dial_order();
let sni = call_sni.clone();
direct_fut = Box::pin(async move {
if dial_order.is_empty() {
// No candidates — the race reduces to
// relay-only. Surface a stable error so the
// outer select falls through to relay_fut
// without a spurious "direct failed" warning.
// Use a pending future that never resolves so
// the select's "other side wins" branch is
// the natural outcome.
std::future::pending::<anyhow::Result<QuinnTransport>>().await
} else {
// Fan out N parallel dials via JoinSet. First
// `Ok` wins; `Err` from a single candidate is
// not fatal — we wait for the others. Only
// when ALL have failed do we return Err.
let mut set = tokio::task::JoinSet::new();
for (idx, candidate) in dial_order.iter().enumerate() {
// Phase 7: route each candidate to the
// endpoint matching its address family.
let candidate = *candidate;
// Phase 7: IPv6 dials temporarily disabled.
// IPv6 QUIC handshakes succeed but the
// connection dies immediately on datagram
// send ("connection lost"). Root cause is
// likely router-level IPv6 UDP filtering.
// Re-enable once IPv6 datagram delivery is
// verified on target networks.
if candidate.is_ipv6() {
tracing::debug!(
%candidate,
candidate_idx = idx,
"dual_path: skipping IPv6 candidate (disabled)"
);
continue;
}
let ep = ep_for_fut.clone();
let client_cfg = wzp_transport::client_config();
let sni = sni.clone();
set.spawn(async move {
let result = wzp_transport::connect(
&ep,
candidate,
&sni,
client_cfg,
)
.await;
(idx, candidate, result)
});
}
let mut last_err: Option<String> = None;
while let Some(join_res) = set.join_next().await {
let (idx, candidate, dial_res) = match join_res {
Ok(t) => t,
Err(e) => {
last_err = Some(format!("join {e}"));
continue;
}
};
match dial_res {
Ok(conn) => {
tracing::info!(
%candidate,
candidate_idx = idx,
remote = %conn.remote_address(),
stable_id = conn.stable_id(),
"dual_path: direct dial succeeded on candidate"
);
// Abort the remaining in-flight
// dials so they don't complete
// and leak QUIC sessions.
set.abort_all();
return Ok(QuinnTransport::new(conn));
}
Err(e) => {
tracing::debug!(
%candidate,
candidate_idx = idx,
error = %e,
"dual_path: direct dial failed, trying others"
);
last_err = Some(format!("candidate {candidate}: {e}"));
}
}
}
Err(anyhow::anyhow!(
"all {} direct candidates failed; last: {}",
dial_order.len(),
last_err.unwrap_or_else(|| "n/a".into())
))
}
});
direct_ep = ep;
}
}
// Relay path: classic dial to the relay's media room. Phase 5:
// reuse the shared endpoint here too so MikroTik-style NATs
// keep a stable external port across all flows from this
// client. Falls back to a fresh endpoint when not shared.
let relay_ep = match shared_endpoint.clone() {
Some(ep) => ep,
None => {
let relay_bind: SocketAddr = "[::]:0".parse().unwrap();
wzp_transport::create_endpoint(relay_bind, None)?
}
};
let relay_ep_for_fut = relay_ep.clone();
let relay_client_cfg = wzp_transport::client_config();
let relay_sni = room_sni.clone();
// Phase 5.5 direct-path head-start: hold the relay dial for
// 500ms before attempting it. On same-LAN cone-NAT pairs the
// direct dial finishes in ~30-100ms, so giving direct a 500ms
// head start means direct reliably wins when it's going to
// work at all. The worst case adds 500ms to the fall-back-
// to-relay scenario, which is imperceptible for users on
// setups where direct isn't available anyway.
//
// Prior behavior (immediate race) caused the relay to win
// ~105ms races on a MikroTik LAN because:
// - Acceptor role's direct_fut = accept() can only fire
// when the peer has completed its outbound LAN dial
// - Dialer role's parallel LAN dials need the peer's
// CallSetup processed + the race started on the other
// side before they can reach us
// - Meanwhile relay_fut is a plain dial that completes in
// whatever the client→relay RTT is (often <100ms)
//
// The 500ms head start is the minimum that empirically makes
// same-LAN direct reliably beat relay, without penalizing
// users who genuinely need the relay path.
const DIRECT_HEAD_START: Duration = Duration::from_millis(500);
let relay_fut = async move {
tokio::time::sleep(DIRECT_HEAD_START).await;
let conn =
wzp_transport::connect(&relay_ep_for_fut, relay_addr, &relay_sni, relay_client_cfg)
.await
.map_err(|e| anyhow::anyhow!("relay dial: {e}"))?;
Ok::<_, anyhow::Error>(QuinnTransport::new(conn))
};
// Phase 6: run both paths concurrently via tokio::spawn and
// collect BOTH results. The old tokio::select! approach dropped
// the loser, which meant the connect command couldn't negotiate
// with the peer — it had to commit to whichever path won locally.
//
// Now we spawn both as tasks, wait for the first to complete
// (that determines `local_winner`), then give the loser a short
// grace period to also complete. The connect command gets a
// RaceResult with both transports (when available) and uses the
// Phase 6 MediaPathReport exchange to decide which one to
// actually use for media.
tracing::info!(
?role,
candidates = ?peer_candidates.dial_order(),
%relay_addr,
"dual_path: racing direct vs relay"
);
let mut direct_task = tokio::spawn(
tokio::time::timeout(Duration::from_secs(2), direct_fut),
);
let mut relay_task = tokio::spawn(async move {
// Keep the 500ms head start so direct has a chance
tokio::time::sleep(Duration::from_millis(500)).await;
tokio::time::timeout(Duration::from_secs(5), relay_fut).await
});
// Wait for the first one to complete. This tells us the
// local_winner — but we DON'T commit to it yet. Phase 6
// negotiation decides the actual path.
let (mut direct_result, mut relay_result): (
Option<anyhow::Result<QuinnTransport>>,
Option<anyhow::Result<QuinnTransport>>,
) = (None, None);
let local_winner;
tokio::select! {
biased;
d = &mut direct_task => {
match d {
Ok(Ok(Ok(t))) => {
tracing::info!("dual_path: direct completed first");
direct_result = Some(Ok(t));
local_winner = WinningPath::Direct;
}
Ok(Ok(Err(e))) => {
tracing::warn!(error = %e, "dual_path: direct failed");
direct_result = Some(Err(anyhow::anyhow!("{e}")));
local_winner = WinningPath::Relay; // direct failed → relay is our only hope
}
Ok(Err(_)) => {
tracing::warn!("dual_path: direct timed out (2s)");
direct_result = Some(Err(anyhow::anyhow!("direct timeout")));
local_winner = WinningPath::Relay;
}
Err(e) => {
tracing::warn!(error = %e, "dual_path: direct task panicked");
direct_result = Some(Err(anyhow::anyhow!("direct task panic")));
local_winner = WinningPath::Relay;
}
}
}
r = &mut relay_task => {
match r {
Ok(Ok(Ok(t))) => {
tracing::info!("dual_path: relay completed first");
relay_result = Some(Ok(t));
local_winner = WinningPath::Relay;
}
Ok(Ok(Err(e))) => {
tracing::warn!(error = %e, "dual_path: relay failed");
relay_result = Some(Err(anyhow::anyhow!("{e}")));
local_winner = WinningPath::Direct;
}
Ok(Err(_)) => {
tracing::warn!("dual_path: relay timed out");
relay_result = Some(Err(anyhow::anyhow!("relay timeout")));
local_winner = WinningPath::Direct;
}
Err(e) => {
relay_result = Some(Err(anyhow::anyhow!("relay task panic: {e}")));
local_winner = WinningPath::Direct;
}
}
}
}
// Give the loser a short grace period (1s) to also complete.
// If it does, we have both transports for Phase 6 negotiation.
// If it doesn't, we still proceed with just the winner.
if direct_result.is_none() {
match tokio::time::timeout(Duration::from_secs(1), direct_task).await {
Ok(Ok(Ok(Ok(t)))) => { direct_result = Some(Ok(t)); }
Ok(Ok(Ok(Err(e)))) => { direct_result = Some(Err(anyhow::anyhow!("{e}"))); }
_ => { direct_result = Some(Err(anyhow::anyhow!("direct: no result in grace period"))); }
}
}
if relay_result.is_none() {
match tokio::time::timeout(Duration::from_secs(1), relay_task).await {
Ok(Ok(Ok(Ok(t)))) => { relay_result = Some(Ok(t)); }
Ok(Ok(Ok(Err(e)))) => { relay_result = Some(Err(anyhow::anyhow!("{e}"))); }
_ => { relay_result = Some(Err(anyhow::anyhow!("relay: no result in grace period"))); }
}
}
let direct_ok = direct_result.as_ref().map(|r| r.is_ok()).unwrap_or(false);
let relay_ok = relay_result.as_ref().map(|r| r.is_ok()).unwrap_or(false);
tracing::info!(
?local_winner,
direct_ok,
relay_ok,
"dual_path: race finished, both results collected for Phase 6 negotiation"
);
if !direct_ok && !relay_ok {
return Err(anyhow::anyhow!("both paths failed: no media transport available"));
}
let _ = (direct_ep, relay_ep, ipv6_endpoint);
Ok(RaceResult {
direct_transport: direct_result
.and_then(|r| r.ok())
.map(|t| Arc::new(t)),
relay_transport: relay_result
.and_then(|r| r.ok())
.map(|t| Arc::new(t)),
local_winner,
})
}

View File

@@ -96,6 +96,7 @@ pub fn signal_to_call_type(signal: &SignalMessage) -> CallSignalType {
SignalMessage::Hangup { .. } => CallSignalType::Hangup,
SignalMessage::Rekey { .. } => CallSignalType::Offer, // reuse
SignalMessage::QualityUpdate { .. } => CallSignalType::Offer, // reuse
SignalMessage::LossRecoveryUpdate { .. } => CallSignalType::Offer, // reuse (telemetry)
SignalMessage::Ping { .. } | SignalMessage::Pong { .. } => CallSignalType::Offer,
SignalMessage::AuthToken { .. } => CallSignalType::Offer,
SignalMessage::Hold => CallSignalType::Hold,
@@ -109,6 +110,27 @@ 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
// NAT reflection is a client↔relay control exchange that
// never crosses the featherChat bridge — if it ever reaches
// this mapper something is wrong, but we still have to give
// an answer. "Offer" is the generic catch-all.
SignalMessage::Reflect
| SignalMessage::ReflectResponse { .. } => CallSignalType::Offer, // control-plane
// Phase 4 cross-relay forwarding envelope — strictly a
// relay-to-relay message, never rides the featherChat
// bridge. Catch-all mapping for completeness.
SignalMessage::FederatedSignalForward { .. } => CallSignalType::Offer,
SignalMessage::MediaPathReport { .. } => CallSignalType::Offer, // control-plane
}
}
@@ -124,6 +146,7 @@ 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"));
@@ -141,11 +164,13 @@ mod tests {
ephemeral_pub: [0; 32],
signature: vec![],
supported_profiles: vec![],
alias: None,
};
assert!(matches!(signal_to_call_type(&offer), CallSignalType::Offer));
let hangup = SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
call_id: None,
};
assert!(matches!(signal_to_call_type(&hangup), CallSignalType::Hangup));

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

@@ -8,16 +8,77 @@
#[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;
pub mod echo_test;
pub mod featherchat;
pub mod handshake;
pub mod dual_path;
pub mod metrics;
pub mod reflect;
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

@@ -0,0 +1,679 @@
//! Multi-relay NAT reflection ("STUN for QUIC" — Phase 2).
//!
//! Phase 1 (`SignalMessage::Reflect` / `ReflectResponse`) lets a
//! client ask a single relay "what source address do you see for
//! me?". Phase 2 queries N relays in parallel and classifies the
//! results into a NAT type so the future P2P hole-punching path
//! can decide whether a direct QUIC handshake is viable:
//!
//! - All relays return the same `(ip, port)` → **Cone NAT**.
//! Endpoint-independent mapping, P2P hole-punching viable,
//! `consensus_addr` is the one address to advertise.
//! - Same ip, different ports → **Symmetric port-dependent NAT**.
//! The mapping changes per destination, so the advertised addr
//! wouldn't match what a peer actually sees; fall back to
//! relay-mediated path.
//! - Different ips → multi-homed / anycast / broken DNS, treat as
//! `Multiple` and do not attempt P2P.
//! - 0 or 1 successful probes → `Unknown`, not enough data.
//!
//! A probe is a throwaway QUIC signal connection: open endpoint,
//! connect, RegisterPresence (with a zero identity — the relay
//! accepts this exactly like the main signaling path does), send
//! Reflect, read ReflectResponse, close. Each probe gets its own
//! ephemeral quinn::Endpoint so the OS assigns a fresh source port
//! per relay — if we shared one endpoint across probes, a
//! symmetric NAT in front of the client would map every probe to
//! the same port and we couldn't detect it.
use std::net::SocketAddr;
use std::time::{Duration, Instant};
use serde::Serialize;
use wzp_proto::{MediaTransport, SignalMessage};
use wzp_transport::{client_config, create_endpoint, QuinnTransport};
/// Result of one probe against one relay. Always returned so the
/// UI can render per-relay status even when some fail.
#[derive(Debug, Clone, Serialize)]
pub struct NatProbeResult {
pub relay_name: String,
pub relay_addr: String,
/// `Some` on successful probe, `None` on failure.
pub observed_addr: Option<String>,
/// End-to-end wall-clock from connect start to ReflectResponse
/// received, in milliseconds. `Some` only on success.
pub latency_ms: Option<u32>,
/// Human-readable error on failure.
pub error: Option<String>,
}
/// Aggregated classification over N `NatProbeResult`s.
#[derive(Debug, Clone, Serialize)]
pub struct NatDetection {
pub probes: Vec<NatProbeResult>,
pub nat_type: NatType,
/// When `nat_type == Cone`, the one address all probes agreed
/// on. `None` for every other case.
pub consensus_addr: Option<String>,
}
/// NAT classification. See module doc for semantics.
#[derive(Debug, Clone, Copy, Serialize, PartialEq, Eq)]
pub enum NatType {
Cone,
SymmetricPort,
Multiple,
Unknown,
}
/// Probe a single relay with a QUIC connection.
///
/// # Endpoint reuse (Phase 5 — Nebula-style architecture)
///
/// If `existing_endpoint` is `Some`, the probe uses that socket
/// instead of creating a fresh one. This is the desired mode in
/// production: a port-preserving NAT (MikroTik masquerade, most
/// consumer routers) gives a **stable** external port for the
/// one socket, so the reflex addr observed by ANY relay is the
/// SAME addr and matches what a peer would see on a direct dial.
/// Pass the signal endpoint here.
///
/// If `None`, creates a fresh one-shot endpoint. Kept for:
/// - tests that spin up isolated probes
/// - the "I'm not registered yet" case where there's no signal
/// endpoint to reuse
///
/// NOTE on NAT-type detection: the pre-Phase-5 behavior of
/// forcing a fresh endpoint per probe was wrong — it made every
/// port-preserving NAT look symmetric because the classifier saw
/// a different external port for each fresh source port. With
/// one shared socket, the classifier reflects the REAL NAT
/// behavior.
pub async fn probe_reflect_addr(
relay: SocketAddr,
timeout_ms: u64,
existing_endpoint: Option<wzp_transport::Endpoint>,
) -> Result<(SocketAddr, u32), String> {
// Install rustls provider idempotently — a second install on the
// same thread is a no-op.
let _ = rustls::crypto::ring::default_provider().install_default();
let endpoint = match existing_endpoint {
Some(ep) => ep,
None => {
let bind: SocketAddr = "0.0.0.0:0".parse().unwrap();
create_endpoint(bind, None).map_err(|e| format!("endpoint: {e}"))?
}
};
let start = Instant::now();
let probe = async {
// Open the signal connection.
let conn =
wzp_transport::connect(&endpoint, relay, "_signal", client_config())
.await
.map_err(|e| format!("connect: {e}"))?;
let transport = QuinnTransport::new(conn);
// The relay signal handler waits for a RegisterPresence
// before entering its main dispatch loop (see
// wzp-relay/src/main.rs). So a transient probe has to
// register with a zero identity first — the relay accepts
// the empty-signature form exactly as the main signaling
// path does in desktop/src-tauri/src/lib.rs register_signal.
transport
.send_signal(&SignalMessage::RegisterPresence {
identity_pub: [0u8; 32],
signature: vec![],
alias: None,
})
.await
.map_err(|e| format!("send RegisterPresence: {e}"))?;
// Drain the RegisterPresenceAck so the response to our
// Reflect doesn't land on an unexpected stream order.
match transport.recv_signal().await {
Ok(Some(SignalMessage::RegisterPresenceAck { success: true, .. })) => {}
Ok(Some(other)) => {
return Err(format!(
"unexpected pre-reflect signal: {:?}",
std::mem::discriminant(&other)
));
}
Ok(None) => return Err("connection closed before RegisterPresenceAck".into()),
Err(e) => return Err(format!("recv RegisterPresenceAck: {e}")),
}
// Send Reflect and await response.
transport
.send_signal(&SignalMessage::Reflect)
.await
.map_err(|e| format!("send Reflect: {e}"))?;
match transport.recv_signal().await {
Ok(Some(SignalMessage::ReflectResponse { observed_addr })) => {
let parsed: SocketAddr = observed_addr
.parse()
.map_err(|e| format!("parse observed_addr {observed_addr:?}: {e}"))?;
let latency_ms = start.elapsed().as_millis() as u32;
// Clean close so the relay's per-connection cleanup
// runs promptly and we don't leak file descriptors.
let _ = transport.close().await;
Ok((parsed, latency_ms))
}
Ok(Some(other)) => Err(format!(
"expected ReflectResponse, got {:?}",
std::mem::discriminant(&other)
)),
Ok(None) => Err("connection closed before ReflectResponse".into()),
Err(e) => Err(format!("recv ReflectResponse: {e}")),
}
};
let out = tokio::time::timeout(Duration::from_millis(timeout_ms), probe)
.await
.map_err(|_| format!("probe timeout ({timeout_ms}ms)"))??;
// `endpoint` is a quinn::Endpoint clone — an Arc under the
// hood. Letting it drop at end-of-scope is correct whether it
// was fresh (last ref → socket closes) or shared (ref count
// decrements, socket stays alive for the signal loop).
Ok(out)
}
/// Detect the client's NAT type by probing N relays in parallel and
/// classifying the returned addresses. Never errors — failing
/// probes surface via `NatProbeResult.error`; aggregate is always
/// returned.
///
/// # Endpoint reuse (Phase 5)
///
/// If `shared_endpoint` is `Some`, every probe reuses it. This is
/// the PRODUCTION behavior: all probes source from the same UDP
/// port, so port-preserving NATs map them to the same external
/// port, and the classifier reflects the real NAT type. Pass the
/// signal endpoint.
///
/// If `None`, each probe creates its own fresh endpoint — useful
/// in tests that don't have a signal endpoint, but produces
/// spurious `SymmetricPort` classifications against NATs that
/// would otherwise look cone-like.
pub async fn detect_nat_type(
relays: Vec<(String, SocketAddr)>,
timeout_ms: u64,
shared_endpoint: Option<wzp_transport::Endpoint>,
) -> NatDetection {
// Parallel probes via tokio::task::JoinSet so the wall-clock is
// bounded by the slowest probe, not the sum. JoinSet keeps the
// dep surface at just tokio — we already depend on it.
let mut set = tokio::task::JoinSet::new();
for (name, addr) in relays {
let ep = shared_endpoint.clone();
set.spawn(async move {
let result = probe_reflect_addr(addr, timeout_ms, ep).await;
(name, addr, result)
});
}
let mut probes = Vec::new();
while let Some(join_result) = set.join_next().await {
let (name, addr, result) = match join_result {
Ok(tuple) => tuple,
// Task panicked — surface as a synthetic failed probe so
// the aggregate still returns a reasonable shape. This
// shouldn't happen but we don't want one bad probe to
// poison the whole detection.
Err(join_err) => {
probes.push(NatProbeResult {
relay_name: "<panicked>".into(),
relay_addr: "unknown".into(),
observed_addr: None,
latency_ms: None,
error: Some(format!("probe task panicked: {join_err}")),
});
continue;
}
};
probes.push(match result {
Ok((observed, latency_ms)) => NatProbeResult {
relay_name: name,
relay_addr: addr.to_string(),
observed_addr: Some(observed.to_string()),
latency_ms: Some(latency_ms),
error: None,
},
Err(e) => NatProbeResult {
relay_name: name,
relay_addr: addr.to_string(),
observed_addr: None,
latency_ms: None,
error: Some(e),
},
});
}
let (nat_type, consensus_addr) = classify_nat(&probes);
NatDetection {
probes,
nat_type,
consensus_addr,
}
}
/// Enumerate LAN-local host candidates this client is reachable
/// on, paired with the given port (typically the signal
/// endpoint's bound port so that incoming dials land on the same
/// socket the advertised reflex addr points to).
///
/// Gathers BOTH IPv4 and IPv6 candidates:
///
/// - **IPv4**: RFC1918 private ranges (10/8, 172.16/12, 192.168/16)
/// and CGNAT shared-transition (100.64/10). Public IPv4 is
/// skipped because the reflex-addr path already covers it.
/// Loopback and link-local (169.254/16) are skipped.
///
/// - **IPv6**: ALL global-unicast addresses (2000::/3 — the real
/// routable IPv6 space) AND unique-local (fc00::/7). These
/// are directly dialable from a peer on the same LAN, and on
/// true dual-stack LANs (which most consumer ISPs now provide,
/// including Starlink) IPv6 often gives a direct path even
/// when IPv4 can't hairpin. Loopback (::1), unspecified (::),
/// and link-local (fe80::/10) are skipped — link-local would
/// require a scope ID to be useful and is basically never
/// reachable across interface boundaries.
///
/// The port must come from the caller — typically
/// `signal_endpoint.local_addr()?.port()`, so that the peer's
/// dials to these addresses land on the same socket that's
/// already listening (Phase 5 shared-endpoint architecture).
///
/// Safe to call from any thread; no I/O, no async. The `if-addrs`
/// crate reads the kernel's interface table via a single
/// getifaddrs(3) syscall.
pub fn local_host_candidates(v4_port: u16, v6_port: Option<u16>) -> Vec<SocketAddr> {
let Ok(ifaces) = if_addrs::get_if_addrs() else {
return Vec::new();
};
let mut out = Vec::new();
for iface in ifaces {
if iface.is_loopback() {
continue;
}
match iface.ip() {
std::net::IpAddr::V4(v4) => {
if v4.is_link_local() {
continue;
}
// Keep RFC1918 private ranges and CGNAT — those
// are the LAN-dialable addrs we actually want.
// Skip public v4 because the reflex addr already
// covers that path.
if v4.is_private() {
out.push(SocketAddr::new(std::net::IpAddr::V4(v4), v4_port));
} else if v4.octets()[0] == 100 && (v4.octets()[1] & 0xc0) == 0x40 {
// 100.64/10 CGNAT — rare but valid if two
// phones are on the same CGNAT-hairpinned
// carrier LAN (some hotspot setups).
out.push(SocketAddr::new(std::net::IpAddr::V4(v4), v4_port));
}
}
std::net::IpAddr::V6(v6) => {
// Phase 7: IPv6 host candidates via dedicated
// IPv6 socket. When v6_port is None, no IPv6
// endpoint exists — skip silently.
let Some(port) = v6_port else { continue };
if v6.is_loopback() || v6.is_unspecified() {
continue;
}
// fe80::/10 link-local — needs scope ID, not
// routable across interfaces.
if (v6.segments()[0] & 0xffc0) == 0xfe80 {
continue;
}
// Accept global unicast (2000::/3) and
// unique-local (fc00::/7).
let first_seg = v6.segments()[0];
let is_global = (first_seg & 0xe000) == 0x2000;
let is_ula = (first_seg & 0xfe00) == 0xfc00;
if is_global || is_ula {
out.push(SocketAddr::new(std::net::IpAddr::V6(v6), port));
}
}
}
}
out
}
/// Role assignment for the Phase 3.5 dual-path QUIC race.
///
/// Both peers already know two strings at CallSetup time: their
/// own server-reflexive address (queried via Phase 1 Reflect) and
/// the peer's (carried in `CallSetup.peer_direct_addr`). To avoid
/// a negotiation round-trip, both sides compare the two strings
/// lexicographically and agree on a deterministic role:
///
/// - **Acceptor** — lexicographically smaller addr. Listens for
/// an incoming direct connection from the peer. Does NOT dial.
/// - **Dialer** — lexicographically larger addr. Dials the
/// peer's direct addr. Does NOT listen.
///
/// Both roles ALSO dial the relay in parallel as a fallback.
/// Whichever future (direct or relay) completes first is used as
/// the media transport. Because the role is deterministic and
/// symmetric, both peers end up holding the same underlying QUIC
/// session on the direct path — A's accepted conn and D's dialed
/// conn are literally the same connection.
#[derive(Debug, Clone, Copy, PartialEq, Eq)]
pub enum Role {
/// This peer listens for the direct incoming connection.
Acceptor,
/// This peer dials the peer's direct address.
Dialer,
}
/// Compute the deterministic role for this peer in the dual-path
/// race. Returns `None` when no direct attempt is possible —
/// either peer didn't advertise a reflex addr, or the two addrs
/// are identical (same host on loopback / mis-advertised).
///
/// The caller should treat `None` as "skip direct, relay-only".
pub fn determine_role(
own_reflex_addr: Option<&str>,
peer_reflex_addr: Option<&str>,
) -> Option<Role> {
let (own, peer) = match (own_reflex_addr, peer_reflex_addr) {
(Some(o), Some(p)) => (o, p),
_ => return None,
};
match own.cmp(peer) {
std::cmp::Ordering::Less => Some(Role::Acceptor),
std::cmp::Ordering::Greater => Some(Role::Dialer),
// Equal addrs should never happen in production (both
// peers behind the same NAT mapping + same port would be
// a degenerate case). Guard against it so we don't infinite-
// loop waiting for a connection to ourselves.
std::cmp::Ordering::Equal => None,
}
}
/// Returns `true` if the address is in an RFC1918 / link-local /
/// loopback range and therefore cannot possibly be a post-NAT
/// reflex address from the public internet's point of view.
///
/// A probe against a relay ON THE SAME LAN as the client will
/// naturally report the client's LAN IP back (because there's no
/// NAT between them) — that observation is real but says nothing
/// about the client's public-internet-facing NAT state. Mixing
/// LAN reflex addrs with public-internet reflex addrs in
/// `classify_nat` would always report `Multiple` (different IPs)
/// and falsely warn about symmetric NAT. Filter them out before
/// classifying.
fn is_private_or_loopback(addr: &SocketAddr) -> bool {
match addr.ip() {
std::net::IpAddr::V4(v4) => {
let o = v4.octets();
v4.is_loopback()
|| v4.is_private() // 10/8, 172.16/12, 192.168/16
|| v4.is_link_local() // 169.254/16
|| (o[0] == 100 && (o[1] & 0xc0) == 0x40) // 100.64/10 CGNAT shared
}
std::net::IpAddr::V6(v6) => {
v6.is_loopback() || v6.is_unspecified() || (v6.segments()[0] & 0xffc0) == 0xfe80 // fe80::/10 link-local
}
}
}
/// Pure-function NAT classifier — split out for unit testing
/// without touching the network.
///
/// Only considers probes whose reflex addr is a **public-internet**
/// address. LAN / private / loopback reflex addrs are dropped
/// because they reflect the same-network path rather than the
/// real NAT state. CGNAT (100.64/10) is also treated as private
/// because the post-CGNAT address would be what we actually want
/// to classify on — but CGNAT is unreachable from outside the
/// carrier, so a relay seeing the CGNAT addr is on the same
/// carrier network and again not useful for classification.
pub fn classify_nat(probes: &[NatProbeResult]) -> (NatType, Option<String>) {
// First: parse every successful probe's observed addr.
let parsed: Vec<SocketAddr> = probes
.iter()
.filter_map(|p| p.observed_addr.as_deref().and_then(|s| s.parse().ok()))
.collect();
// Then: drop LAN / private / loopback reflex addrs. Those are
// legitimate observations by same-network relays, but they
// don't contribute to NAT-type classification because the
// client's real public-facing NAT mapping is not involved on
// that path. A relay on the same LAN always sees the client's
// LAN IP, regardless of whether the NAT beyond it is cone or
// symmetric.
let successes: Vec<SocketAddr> = parsed
.into_iter()
.filter(|a| !is_private_or_loopback(a))
.collect();
if successes.len() < 2 {
return (NatType::Unknown, None);
}
let first = successes[0];
let same_ip = successes.iter().all(|a| a.ip() == first.ip());
if !same_ip {
return (NatType::Multiple, None);
}
let same_port = successes.iter().all(|a| a.port() == first.port());
if same_port {
(NatType::Cone, Some(first.to_string()))
} else {
(NatType::SymmetricPort, None)
}
}
// ── Unit tests for the pure classifier ───────────────────────────
#[cfg(test)]
mod tests {
use super::*;
fn mk(addr: Option<&str>) -> NatProbeResult {
NatProbeResult {
relay_name: "test".into(),
relay_addr: "0.0.0.0:0".into(),
observed_addr: addr.map(|s| s.to_string()),
latency_ms: addr.map(|_| 10),
error: None,
}
}
#[test]
fn classify_empty_is_unknown() {
let (nt, addr) = classify_nat(&[]);
assert_eq!(nt, NatType::Unknown);
assert!(addr.is_none());
}
#[test]
fn classify_single_success_is_unknown() {
let probes = vec![mk(Some("192.0.2.1:4433"))];
let (nt, addr) = classify_nat(&probes);
assert_eq!(nt, NatType::Unknown);
assert!(addr.is_none());
}
#[test]
fn classify_two_identical_is_cone() {
let probes = vec![
mk(Some("192.0.2.1:4433")),
mk(Some("192.0.2.1:4433")),
];
let (nt, addr) = classify_nat(&probes);
assert_eq!(nt, NatType::Cone);
assert_eq!(addr.as_deref(), Some("192.0.2.1:4433"));
}
#[test]
fn classify_same_ip_different_ports_is_symmetric() {
let probes = vec![
mk(Some("192.0.2.1:4433")),
mk(Some("192.0.2.1:51234")),
];
let (nt, addr) = classify_nat(&probes);
assert_eq!(nt, NatType::SymmetricPort);
assert!(addr.is_none());
}
#[test]
fn classify_different_ips_is_multiple() {
let probes = vec![
mk(Some("192.0.2.1:4433")),
mk(Some("198.51.100.9:4433")),
];
let (nt, addr) = classify_nat(&probes);
assert_eq!(nt, NatType::Multiple);
assert!(addr.is_none());
}
#[test]
fn classify_drops_private_ip_probes() {
// One LAN probe + one public probe should behave like a
// single public probe — i.e. Unknown (not enough data to
// classify). This is the common real-world case: the user
// has a LAN relay + an internet relay configured, the LAN
// relay sees the LAN IP, the internet relay sees the WAN
// IP, and the old classifier would flag "Multiple" and
// falsely warn about symmetric NAT.
let probes = vec![
mk(Some("192.168.1.100:4433")), // LAN — must be dropped
mk(Some("203.0.113.5:4433")), // public (TEST-NET-3)
];
let (nt, _) = classify_nat(&probes);
assert_eq!(nt, NatType::Unknown);
}
#[test]
fn classify_drops_loopback_probes() {
let probes = vec![
mk(Some("127.0.0.1:4433")), // loopback — must be dropped
mk(Some("203.0.113.5:4433")), // public
mk(Some("203.0.113.5:4433")), // public, same addr
];
let (nt, addr) = classify_nat(&probes);
// Two public probes with identical addrs → Cone.
assert_eq!(nt, NatType::Cone);
assert_eq!(addr.as_deref(), Some("203.0.113.5:4433"));
}
#[test]
fn classify_drops_cgnat_probes() {
// 100.64.0.0/10 is the CGNAT shared-transition range.
// Filter treats it like RFC1918 — a relay that sees the
// client with a 100.64/10 addr is on the same CGNAT
// network and can't contribute to public NAT classification.
let probes = vec![
mk(Some("100.64.0.42:4433")), // CGNAT — dropped
mk(Some("203.0.113.5:4433")), // public
mk(Some("203.0.113.5:12345")), // public, different port
];
let (nt, _) = classify_nat(&probes);
// Two public probes same IP different port → SymmetricPort.
assert_eq!(nt, NatType::SymmetricPort);
}
#[test]
fn classify_two_lan_probes_is_unknown_not_cone() {
// Even if both probes come back from LAN relays, we can't
// say anything useful about the public NAT state. Unknown,
// not Cone.
let probes = vec![
mk(Some("192.168.1.100:4433")),
mk(Some("192.168.1.100:4433")),
];
let (nt, addr) = classify_nat(&probes);
assert_eq!(nt, NatType::Unknown);
assert!(addr.is_none());
}
#[test]
fn classify_mix_of_success_and_failure() {
let probes = vec![
mk(Some("192.0.2.1:4433")),
mk(None), // failed probe
mk(Some("192.0.2.1:4433")),
];
let (nt, addr) = classify_nat(&probes);
// Two successes both agree → Cone, ignore the failure row.
assert_eq!(nt, NatType::Cone);
assert_eq!(addr.as_deref(), Some("192.0.2.1:4433"));
}
#[test]
fn determine_role_smaller_is_acceptor() {
// Lexicographic: "192.0.2.1:4433" < "198.51.100.9:4433"
assert_eq!(
determine_role(Some("192.0.2.1:4433"), Some("198.51.100.9:4433")),
Some(Role::Acceptor)
);
}
#[test]
fn determine_role_larger_is_dialer() {
assert_eq!(
determine_role(Some("198.51.100.9:4433"), Some("192.0.2.1:4433")),
Some(Role::Dialer)
);
}
#[test]
fn determine_role_port_difference_matters() {
// Same ip, different ports — string compare still works
// because "4433" < "54321".
assert_eq!(
determine_role(Some("127.0.0.1:4433"), Some("127.0.0.1:54321")),
Some(Role::Acceptor)
);
assert_eq!(
determine_role(Some("127.0.0.1:54321"), Some("127.0.0.1:4433")),
Some(Role::Dialer)
);
}
#[test]
fn determine_role_equal_addrs_is_none() {
assert_eq!(
determine_role(Some("192.0.2.1:4433"), Some("192.0.2.1:4433")),
None
);
}
#[test]
fn determine_role_missing_side_is_none() {
assert_eq!(determine_role(None, Some("192.0.2.1:4433")), None);
assert_eq!(determine_role(Some("192.0.2.1:4433"), None), None);
assert_eq!(determine_role(None, None), None);
}
#[test]
fn determine_role_is_symmetric_across_peers() {
// Both peers compute roles independently; they must end
// up with opposite assignments (one Acceptor, one Dialer)
// so that each side ends up talking to the other.
let a = "192.0.2.1:4433";
let b = "198.51.100.9:4433";
let alice_role = determine_role(Some(a), Some(b));
let bob_role = determine_role(Some(b), Some(a));
assert_eq!(alice_role, Some(Role::Acceptor));
assert_eq!(bob_role, Some(Role::Dialer));
}
#[test]
fn classify_one_success_one_failure_is_unknown() {
let probes = vec![mk(Some("192.0.2.1:4433")), mk(None)];
let (nt, addr) = classify_nat(&probes);
assert_eq!(nt, NatType::Unknown);
assert!(addr.is_none());
}
}

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@@ -0,0 +1,213 @@
//! Phase 3.5 integration tests for the dual-path QUIC race.
//!
//! The race takes a role (Acceptor or Dialer), a peer_direct_addr,
//! a relay_addr, and two SNI strings, then returns whichever QUIC
//! handshake completes first wrapped in a `QuinnTransport`. These
//! tests validate that:
//!
//! 1. On loopback with two real clients playing A + D roles, the
//! direct path wins (fewer hops than relay).
//! 2. When the direct peer is dead (nothing listening) but the
//! relay is up, the relay wins within the fallback window.
//! 3. When both paths are dead, the race errors cleanly rather
//! than hanging forever.
//!
//! The "relay" in these tests is a minimal mock that just accepts
//! an incoming QUIC connection and drops it — we don't need any
//! protocol handling, just a TCP-ish listen-and-accept.
use std::net::{Ipv4Addr, SocketAddr};
use std::time::Duration;
use wzp_client::dual_path::{race, PeerCandidates, WinningPath};
use wzp_client::reflect::Role;
use wzp_transport::{create_endpoint, server_config};
/// Spin up a "relay-ish" mock server on loopback that accepts
/// incoming QUIC connections and does nothing with them. Used to
/// give the relay branch of the race a real target to dial.
/// Returns the bound address + a join handle (kept alive to keep
/// the endpoint up).
async fn spawn_mock_relay() -> (SocketAddr, tokio::task::JoinHandle<()>) {
let _ = rustls::crypto::ring::default_provider().install_default();
let (sc, _cert_der) = server_config();
let bind: SocketAddr = (Ipv4Addr::LOCALHOST, 0).into();
let ep = create_endpoint(bind, Some(sc)).expect("relay endpoint");
let addr = ep.local_addr().expect("local_addr");
let handle = tokio::spawn(async move {
// Accept loop — hold the connection alive for a short
// while so the race result isn't killed by the peer
// closing before the winning transport is returned.
while let Some(incoming) = ep.accept().await {
if let Ok(_conn) = incoming.await {
tokio::time::sleep(Duration::from_secs(5)).await;
}
}
});
(addr, handle)
}
// -----------------------------------------------------------------------
// Test 1: direct path wins when both sides are up
// -----------------------------------------------------------------------
//
// Spawn a mock relay, then set up a two-client test where one
// client plays the Acceptor role and the other plays the Dialer
// role. The Dialer's `peer_direct_addr` is the Acceptor's listen
// address. Because the direct path is a single loopback hop and
// the relay dial also terminates on loopback, both complete
// essentially instantly — the `biased` tokio::select in race()
// should pick direct.
#[tokio::test(flavor = "multi_thread", worker_threads = 4)]
async fn dual_path_direct_wins_on_loopback() {
let _ = rustls::crypto::ring::default_provider().install_default();
let (relay_addr, _relay_handle) = spawn_mock_relay().await;
// Acceptor task: run race(Role::Acceptor, peer_addr_placeholder, ...).
// Since the acceptor doesn't dial, the peer_direct_addr arg is
// unused on the direct branch but we still pass a placeholder
// because the API takes one. Use a stub addr that would error
// if it were ever dialed — proving the Acceptor really doesn't
// reach it.
let unused_addr: SocketAddr = "127.0.0.1:2".parse().unwrap();
// We can't race both sides in the same task because each race
// call has its own direct endpoint that needs to talk to the
// OTHER side's endpoint. So spawn the Acceptor in a task and
// let it expose its listen addr via a oneshot back to the test,
// then run the Dialer in the test's main task.
//
// There's a chicken-and-egg issue: the Acceptor's listen addr
// is only known after race() creates its endpoint. To avoid
// reaching into race()'s internals, we instead play a slight
// trick: create the Acceptor's endpoint ourselves (outside
// race()) to learn its addr, spin up an accept loop on it
// ourselves, and pass THAT addr as the Dialer's peer addr.
// This tests the Dialer->Acceptor handshake end-to-end without
// running the full race() on both sides.
let (sc, _cert_der) = server_config();
let acceptor_bind: SocketAddr = (Ipv4Addr::LOCALHOST, 0).into();
let acceptor_ep = create_endpoint(acceptor_bind, Some(sc)).expect("acceptor ep");
let acceptor_listen_addr = acceptor_ep.local_addr().expect("acceptor addr");
// Drop the external acceptor after the test finishes, not
// before — spawn a dedicated accept task.
let acceptor_accept_task = tokio::spawn(async move {
// Accept one connection and hold it for a while so the
// Dialer side can complete its QUIC handshake.
if let Some(incoming) = acceptor_ep.accept().await {
if let Ok(_conn) = incoming.await {
tokio::time::sleep(Duration::from_secs(5)).await;
}
}
});
// Now run the Dialer in the race — peer_direct_addr = acceptor's
// listen addr. The relay is the mock from above. Direct path
// should win.
let result = race(
Role::Dialer,
PeerCandidates {
reflexive: Some(acceptor_listen_addr),
local: Vec::new(),
},
relay_addr,
"test-room".into(),
"call-test".into(),
None, // Phase 5: tests use fresh endpoints (no shared signal)
)
.await
.expect("race must succeed");
assert!(result.direct_transport.is_some(), "direct transport should be available");
assert_eq!(result.local_winner, WinningPath::Direct, "direct should win on loopback");
// Cancel the acceptor accept task so the test finishes.
acceptor_accept_task.abort();
// Suppress unused-var warning for the placeholder.
let _ = unused_addr;
}
// -----------------------------------------------------------------------
// Test 2: relay wins when the direct peer is dead
// -----------------------------------------------------------------------
//
// Dialer role, peer_direct_addr = a port nothing is listening on,
// relay is the working mock. Direct dial will sit waiting for a
// QUIC handshake that never comes; the 2s direct timeout kicks in
// and the relay path wins the fallback.
#[tokio::test(flavor = "multi_thread", worker_threads = 4)]
async fn dual_path_relay_wins_when_direct_is_dead() {
let _ = rustls::crypto::ring::default_provider().install_default();
let (relay_addr, _relay_handle) = spawn_mock_relay().await;
// A port that nothing is listening on — dead direct target.
// Port 1 on loopback is almost never bound and UDP packets to
// it will be dropped silently, so the QUIC handshake times out.
let dead_peer: SocketAddr = "127.0.0.1:1".parse().unwrap();
let result = race(
Role::Dialer,
PeerCandidates {
reflexive: Some(dead_peer),
local: Vec::new(),
},
relay_addr,
"test-room".into(),
"call-test".into(),
None, // Phase 5: tests use fresh endpoints (no shared signal)
)
.await
.expect("race must succeed via relay fallback");
assert!(result.relay_transport.is_some(), "relay transport should be available");
assert_eq!(
result.local_winner,
WinningPath::Relay,
"relay should win when direct dial has nowhere to land"
);
}
// -----------------------------------------------------------------------
// Test 3: race errors cleanly when both paths are dead
// -----------------------------------------------------------------------
//
// Dialer role, peer_direct_addr = dead, relay_addr = dead.
// Expected: race returns an Err within ~7s (2s direct timeout +
// 5s relay timeout fallback).
#[tokio::test(flavor = "multi_thread", worker_threads = 4)]
async fn dual_path_errors_cleanly_when_both_paths_dead() {
let _ = rustls::crypto::ring::default_provider().install_default();
let dead_peer: SocketAddr = "127.0.0.1:1".parse().unwrap();
let dead_relay: SocketAddr = "127.0.0.1:2".parse().unwrap();
let start = std::time::Instant::now();
let result = race(
Role::Dialer,
PeerCandidates {
reflexive: Some(dead_peer),
local: Vec::new(),
},
dead_relay,
"test-room".into(),
"call-test".into(),
None, // Phase 5: tests use fresh endpoints (no shared signal)
)
.await;
let elapsed = start.elapsed();
assert!(result.is_err(), "both-dead must return Err");
// Upper bound: direct 2s timeout + relay 5s fallback + small
// slack for scheduling. If this blows, something is looping.
assert!(
elapsed < Duration::from_secs(10),
"race took too long to give up: {:?}",
elapsed
);
}

View File

@@ -83,12 +83,12 @@ async fn full_handshake_both_sides_derive_same_session() {
// Run client and relay handshakes concurrently.
let (client_result, relay_result) = tokio::join!(
wzp_client::handshake::perform_handshake(client_transport_clone.as_ref(), &client_seed),
wzp_client::handshake::perform_handshake(client_transport_clone.as_ref(), &client_seed, None),
wzp_relay::handshake::accept_handshake(relay_transport_clone.as_ref(), &relay_seed),
);
let mut client_session = client_result.expect("client handshake should succeed");
let (mut relay_session, chosen_profile) =
let (mut relay_session, chosen_profile, _caller_fp, _caller_alias) =
relay_result.expect("relay handshake should succeed");
// Verify a profile was chosen.
@@ -151,6 +151,7 @@ async fn handshake_rejects_tampered_signature() {
ephemeral_pub,
signature: bad_signature,
supported_profiles: vec![wzp_proto::QualityProfile::GOOD],
alias: None,
};
client_transport_clone
.send_signal(&offer)

View File

@@ -10,8 +10,17 @@ description = "WarzonePhone audio codec layer — Opus + Codec2 encoding/decodin
wzp-proto = { workspace = true }
tracing = { workspace = true }
# Opus bindings
audiopus = { workspace = true }
# Opus bindings — libopus 1.5.2.
# opusic-c for the encoder (set_dred_duration lives here in Phase 1).
# opusic-sys for the decoder — we wrap the raw *mut OpusDecoder ourselves
# because opusic-c::Decoder.inner is pub(crate), blocking the unified
# decoder + DRED path we need in Phase 3.
opusic-c = { workspace = true }
opusic-sys = { workspace = true }
# Zero-cost slice reinterpretation for the i16 ↔ u16 boundary between
# our PCM buffers and opusic-c's encode API.
bytemuck = { workspace = true }
# Pure-Rust Codec2 implementation
codec2 = { workspace = true }

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)
@@ -195,6 +199,27 @@ impl AdaptiveDecoder {
fn codec2_frame_samples(&self) -> usize {
self.codec2.frame_samples()
}
/// Reconstruct a lost frame from a previously parsed DRED state.
///
/// Phase 3b entry point for gap reconstruction. Dispatches to the
/// inner Opus decoder when active. Returns an error if the active
/// codec is Codec2 — DRED is libopus-only and has no Codec2 equivalent,
/// so callers must fall back to classical PLC on Codec2 tiers.
pub fn reconstruct_from_dred(
&mut self,
state: &crate::dred_ffi::DredState,
offset_samples: i32,
output: &mut [i16],
) -> Result<usize, CodecError> {
if is_codec2(self.active) {
return Err(CodecError::DecodeFailed(
"DRED reconstruction is Opus-only; Codec2 must use classical PLC".into(),
));
}
self.opus
.reconstruct_from_dred(state, offset_samples, output)
}
}
// ─── Tests ───────────────────────────────────────────────────────────────────

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

@@ -0,0 +1,585 @@
//! Raw opusic-sys FFI wrappers for libopus 1.5.2 decoder + DRED reconstruction.
//!
//! # Why this module exists
//!
//! We cannot use `opusic_c::Decoder` because its inner `*mut OpusDecoder`
//! pointer is `pub(crate)` — not reachable from outside the opusic-c crate.
//! Phase 3 of the DRED integration needs to hand that same pointer to
//! `opus_decoder_dred_decode`, and running two parallel decoders (one from
//! opusic-c for normal audio, another from opusic-sys for DRED) would cause
//! the DRED-only decoder's internal state to drift out of sync with the
//! audio stream because it would not see normal decode calls.
//!
//! The fix is to own the raw decoder ourselves and use the same handle for
//! both normal decode AND DRED reconstruction. This module is the single
//! owner of `*mut OpusDecoder`, `*mut OpusDREDDecoder`, and `*mut OpusDRED`
//! in the WZP workspace.
//!
//! # Phase 3a scope
//!
//! Phase 0 added `DecoderHandle` (normal decode). Phase 3a adds:
//! - [`DredDecoderHandle`] — wraps `*mut OpusDREDDecoder` for parsing DRED
//! side-channel data out of arriving Opus packets.
//! - [`DredState`] — wraps `*mut OpusDRED` (a fixed 10,592-byte buffer
//! allocated by libopus) that holds parsed DRED state between the parse
//! and reconstruct steps.
//! - [`DredDecoderHandle::parse_into`] — wraps `opus_dred_parse`.
//! - [`DecoderHandle::reconstruct_from_dred`] — wraps `opus_decoder_dred_decode`.
//!
//! The pattern is: on every arriving Opus packet, the receiver calls
//! `parse_into` with a reusable `DredState`, then stores (seq, state_clone)
//! in a ring. On detected loss, the receiver computes the offset from the
//! freshest reachable DRED state and calls `reconstruct_from_dred` to
//! synthesize the missing audio.
use std::ptr::NonNull;
use opusic_sys::{
OPUS_OK, OpusDRED, OpusDREDDecoder, OpusDecoder as RawOpusDecoder, opus_decode,
opus_decoder_create, opus_decoder_destroy, opus_decoder_dred_decode, opus_dred_alloc,
opus_dred_decoder_create, opus_dred_decoder_destroy, opus_dred_free, opus_dred_parse,
};
use wzp_proto::CodecError;
/// libopus operates at 48 kHz for all Opus variants we use.
const SAMPLE_RATE_HZ: i32 = 48_000;
/// Mono.
const CHANNELS: i32 = 1;
/// Safe owner of a `*mut OpusDecoder` allocated via `opus_decoder_create`.
///
/// Releases the decoder in `Drop`. All FFI access goes through `&mut self`
/// methods, so there is no aliasing or race. The raw pointer is exposed via
/// [`Self::as_raw_ptr`] at a crate-internal visibility for the future Phase 3
/// DRED reconstruction path — external crates cannot reach it.
pub struct DecoderHandle {
inner: NonNull<RawOpusDecoder>,
}
impl DecoderHandle {
/// Allocate a new Opus decoder at 48 kHz mono.
pub fn new() -> Result<Self, CodecError> {
let mut error: i32 = OPUS_OK;
// SAFETY: opus_decoder_create writes to `error` and returns either a
// valid heap pointer or null. We check both before constructing the
// NonNull wrapper.
let ptr = unsafe { opus_decoder_create(SAMPLE_RATE_HZ, CHANNELS, &mut error) };
if error != OPUS_OK {
// Even if ptr is non-null on error, libopus contracts guarantee
// it is unusable — do not attempt to free it.
return Err(CodecError::DecodeFailed(format!(
"opus_decoder_create failed: err={error}"
)));
}
let inner = NonNull::new(ptr).ok_or_else(|| {
CodecError::DecodeFailed("opus_decoder_create returned null".into())
})?;
Ok(Self { inner })
}
/// Decode an Opus packet into PCM samples.
///
/// `pcm` must have enough capacity for the frame (960 for 20 ms, 1920
/// for 40 ms at 48 kHz mono). Returns the number of decoded samples
/// per channel — for mono streams this equals the total sample count.
pub fn decode(&mut self, packet: &[u8], pcm: &mut [i16]) -> Result<usize, CodecError> {
if packet.is_empty() {
return Err(CodecError::DecodeFailed("empty packet".into()));
}
if pcm.is_empty() {
return Err(CodecError::DecodeFailed("empty output buffer".into()));
}
// SAFETY: self.inner is a valid *mut OpusDecoder owned by this struct.
// `data` / `pcm` are live Rust slices, so their pointers and lengths
// are valid for the duration of the call. libopus reads len bytes
// from data and writes up to frame_size samples (per channel) to pcm.
let n = unsafe {
opus_decode(
self.inner.as_ptr(),
packet.as_ptr(),
packet.len() as i32,
pcm.as_mut_ptr(),
pcm.len() as i32,
/* decode_fec = */ 0,
)
};
if n < 0 {
return Err(CodecError::DecodeFailed(format!(
"opus_decode failed: err={n}"
)));
}
Ok(n as usize)
}
/// Generate packet-loss concealment audio for a missing frame.
///
/// Implemented via `opus_decode` with a null data pointer, per the
/// libopus API contract. `pcm` should be sized for the expected frame.
pub fn decode_lost(&mut self, pcm: &mut [i16]) -> Result<usize, CodecError> {
if pcm.is_empty() {
return Err(CodecError::DecodeFailed("empty output buffer".into()));
}
// SAFETY: same invariants as decode(). libopus documents that passing
// a null data pointer with len=0 triggers PLC synthesis into pcm.
let n = unsafe {
opus_decode(
self.inner.as_ptr(),
std::ptr::null(),
0,
pcm.as_mut_ptr(),
pcm.len() as i32,
/* decode_fec = */ 0,
)
};
if n < 0 {
return Err(CodecError::DecodeFailed(format!(
"opus_decode PLC failed: err={n}"
)));
}
Ok(n as usize)
}
/// Reconstruct audio from a `DredState` into the `output` buffer.
///
/// `offset_samples` is the sample position (positive, measured backward
/// from the packet anchor that produced `state`) where reconstruction
/// begins. `output.len()` must match the number of samples to synthesize.
///
/// The libopus API: `opus_decoder_dred_decode(st, dred, dred_offset, pcm,
/// frame_size)` where `dred_offset` is "position of the redundancy to
/// decode, in samples before the beginning of the real audio data in the
/// packet." Valid values: `0 < offset_samples < state.samples_available()`.
///
/// Returns the number of samples actually written (should equal
/// `output.len()` on success).
pub fn reconstruct_from_dred(
&mut self,
state: &DredState,
offset_samples: i32,
output: &mut [i16],
) -> Result<usize, CodecError> {
if output.is_empty() {
return Err(CodecError::DecodeFailed(
"empty reconstruction output buffer".into(),
));
}
if offset_samples <= 0 {
return Err(CodecError::DecodeFailed(format!(
"DRED offset must be positive (got {offset_samples})"
)));
}
if offset_samples > state.samples_available() {
return Err(CodecError::DecodeFailed(format!(
"DRED offset {offset_samples} exceeds available samples {}",
state.samples_available()
)));
}
// SAFETY: self.inner is a valid *mut OpusDecoder, state.inner is a
// valid *const OpusDRED populated by a prior parse_into call, and
// output is a live mutable slice. libopus reads from dred and writes
// exactly frame_size samples (the output.len()) to pcm.
let n = unsafe {
opus_decoder_dred_decode(
self.inner.as_ptr(),
state.inner.as_ptr(),
offset_samples,
output.as_mut_ptr(),
output.len() as i32,
)
};
if n < 0 {
return Err(CodecError::DecodeFailed(format!(
"opus_decoder_dred_decode failed: err={n}"
)));
}
Ok(n as usize)
}
}
impl Drop for DecoderHandle {
fn drop(&mut self) {
// SAFETY: we own the pointer and no further access happens after
// this call because Drop consumes self.
unsafe { opus_decoder_destroy(self.inner.as_ptr()) };
}
}
// SAFETY: The underlying OpusDecoder is a plain heap allocation with no
// thread-local or lock-free state. It is safe to move between threads
// (Send), and all method access is gated by &mut self so Rust's borrow
// checker prevents simultaneous access from multiple threads (Sync).
unsafe impl Send for DecoderHandle {}
unsafe impl Sync for DecoderHandle {}
// ─── DRED decoder (parser) ──────────────────────────────────────────────────
/// Safe owner of a `*mut OpusDREDDecoder` allocated via
/// `opus_dred_decoder_create`.
///
/// The DRED decoder is a **separate** libopus object from the regular
/// `OpusDecoder`. It's used exclusively for parsing DRED side-channel data
/// out of arriving Opus packets via [`Self::parse_into`]. Actual audio
/// reconstruction from the parsed state uses the regular `DecoderHandle`
/// via [`DecoderHandle::reconstruct_from_dred`].
pub struct DredDecoderHandle {
inner: NonNull<OpusDREDDecoder>,
}
impl DredDecoderHandle {
/// Allocate a new DRED decoder.
pub fn new() -> Result<Self, CodecError> {
let mut error: i32 = OPUS_OK;
// SAFETY: opus_dred_decoder_create writes to `error` and returns
// either a valid heap pointer or null. Both are checked.
let ptr = unsafe { opus_dred_decoder_create(&mut error) };
if error != OPUS_OK {
return Err(CodecError::DecodeFailed(format!(
"opus_dred_decoder_create failed: err={error}"
)));
}
let inner = NonNull::new(ptr).ok_or_else(|| {
CodecError::DecodeFailed("opus_dred_decoder_create returned null".into())
})?;
Ok(Self { inner })
}
/// Parse DRED side-channel data from an Opus packet into `state`.
///
/// Returns the number of samples of audio history available for
/// reconstruction, or 0 if the packet carries no DRED data. Subsequent
/// `DecoderHandle::reconstruct_from_dred` calls using this `state` can
/// reconstruct any sample position in `(0, samples_available]`.
///
/// libopus API: `opus_dred_parse(dred_dec, dred, data, len,
/// max_dred_samples, sampling_rate, dred_end, defer_processing)`. We
/// pass `max_dred_samples = 48000` (1 s at 48 kHz, the DRED maximum),
/// `sampling_rate = 48000`, `defer_processing = 0` (process immediately).
/// The `dred_end` output is the silence gap at the tail of the DRED
/// window; we subtract it from the total offset to give callers the
/// truly usable sample count.
pub fn parse_into(
&mut self,
state: &mut DredState,
packet: &[u8],
) -> Result<i32, CodecError> {
if packet.is_empty() {
state.samples_available = 0;
return Ok(0);
}
let mut dred_end: i32 = 0;
// SAFETY: self.inner is a valid *mut OpusDREDDecoder; state.inner is
// a valid *mut OpusDRED allocated via opus_dred_alloc; packet is a
// live slice; dred_end is a stack int. libopus reads packet bytes
// and writes parsed DRED state into *state.inner.
let ret = unsafe {
opus_dred_parse(
self.inner.as_ptr(),
state.inner.as_ptr(),
packet.as_ptr(),
packet.len() as i32,
/* max_dred_samples = */ 48_000, // 1s max per libopus 1.5
/* sampling_rate = */ 48_000,
&mut dred_end,
/* defer_processing = */ 0,
)
};
if ret < 0 {
state.samples_available = 0;
return Err(CodecError::DecodeFailed(format!(
"opus_dred_parse failed: err={ret}"
)));
}
// ret is the positive offset of the first decodable DRED sample,
// or 0 if no DRED is present. dred_end is the silence gap at the
// tail. The usable sample range is (dred_end, ret], so the count
// of usable samples is ret - dred_end. We store `ret` as the max
// usable offset — callers should pass dred_offset values in the
// range (dred_end, ret] to reconstruct_from_dred. For simplicity
// we expose just samples_available = ret and let callers treat
// the full window as valid (the silence gap is small and libopus
// handles minor boundary cases gracefully).
state.samples_available = ret;
Ok(ret)
}
}
impl Drop for DredDecoderHandle {
fn drop(&mut self) {
// SAFETY: we own the pointer and no further access happens after
// this call because Drop consumes self.
unsafe { opus_dred_decoder_destroy(self.inner.as_ptr()) };
}
}
// SAFETY: same reasoning as DecoderHandle — heap allocation with no
// thread-local state, &mut self access discipline prevents races.
unsafe impl Send for DredDecoderHandle {}
unsafe impl Sync for DredDecoderHandle {}
// ─── DRED state buffer ──────────────────────────────────────────────────────
/// Safe owner of a `*mut OpusDRED` allocated via `opus_dred_alloc`.
///
/// Holds a fixed-size (10,592-byte per libopus 1.5) buffer that
/// `DredDecoderHandle::parse_into` populates from an Opus packet. The state
/// is reusable — the caller can call `parse_into` again on the same
/// `DredState` to overwrite it with a fresh packet's data.
///
/// `samples_available` tracks the last-parsed result so reconstruction
/// callers don't need to thread the return value separately. A fresh
/// state (before any `parse_into`) has `samples_available == 0`.
pub struct DredState {
inner: NonNull<OpusDRED>,
samples_available: i32,
}
impl DredState {
/// Allocate a new DRED state buffer.
pub fn new() -> Result<Self, CodecError> {
let mut error: i32 = OPUS_OK;
// SAFETY: opus_dred_alloc writes to `error` and returns either a
// valid heap pointer or null.
let ptr = unsafe { opus_dred_alloc(&mut error) };
if error != OPUS_OK {
return Err(CodecError::DecodeFailed(format!(
"opus_dred_alloc failed: err={error}"
)));
}
let inner = NonNull::new(ptr)
.ok_or_else(|| CodecError::DecodeFailed("opus_dred_alloc returned null".into()))?;
Ok(Self {
inner,
samples_available: 0,
})
}
/// How many samples of audio history this state currently covers.
///
/// Returns 0 if the state is fresh or the last parse found no DRED
/// data. Otherwise returns the positive offset set by the most recent
/// `DredDecoderHandle::parse_into` call — the maximum valid
/// `offset_samples` value for `DecoderHandle::reconstruct_from_dred`.
pub fn samples_available(&self) -> i32 {
self.samples_available
}
/// Reset the state to "fresh" without freeing the underlying buffer.
/// The next `parse_into` will overwrite the contents.
pub fn reset(&mut self) {
self.samples_available = 0;
}
}
impl Drop for DredState {
fn drop(&mut self) {
// SAFETY: we own the pointer and no further access happens after
// this call because Drop consumes self.
unsafe { opus_dred_free(self.inner.as_ptr()) };
}
}
// SAFETY: same reasoning as DecoderHandle.
unsafe impl Send for DredState {}
unsafe impl Sync for DredState {}
#[cfg(test)]
mod tests {
use super::*;
#[test]
fn decoder_handle_creates_and_drops() {
let handle = DecoderHandle::new().expect("decoder create");
// Dropping the handle must not panic or leak — validated by miri
// and the absence of sanitizer complaints in CI.
drop(handle);
}
#[test]
fn decode_lost_produces_full_frame_of_silence_on_cold_start() {
let mut handle = DecoderHandle::new().unwrap();
// 20 ms @ 48 kHz mono.
let mut pcm = vec![0i16; 960];
let n = handle.decode_lost(&mut pcm).unwrap();
assert_eq!(n, 960);
// On a fresh decoder, PLC output is silence (no past audio to extend).
assert!(pcm.iter().all(|&s| s == 0));
}
#[test]
fn decode_empty_packet_errors() {
let mut handle = DecoderHandle::new().unwrap();
let mut pcm = vec![0i16; 960];
let err = handle.decode(&[], &mut pcm);
assert!(err.is_err());
}
// ─── Phase 3a — DRED decoder + state ────────────────────────────────────
#[test]
fn dred_decoder_handle_creates_and_drops() {
let h = DredDecoderHandle::new().expect("dred decoder create");
drop(h);
}
#[test]
fn dred_state_creates_and_drops() {
let s = DredState::new().expect("dred state alloc");
assert_eq!(s.samples_available(), 0);
drop(s);
}
#[test]
fn dred_state_reset_zeroes_counter() {
let mut s = DredState::new().unwrap();
s.samples_available = 480; // pretend a parse populated it
assert_eq!(s.samples_available(), 480);
s.reset();
assert_eq!(s.samples_available(), 0);
}
/// Phase 3a end-to-end: encode a DRED-enabled stream, parse state out
/// of packets, and reconstruct audio at a past offset. Validates the
/// full parse → reconstruct pipeline against a real libopus 1.5.2
/// encoder so we catch FFI-layer bugs early.
#[test]
fn dred_parse_and_reconstruct_roundtrip() {
use crate::opus_enc::OpusEncoder;
use wzp_proto::{AudioEncoder, QualityProfile};
// Encoder with DRED at Opus 24k / 200 ms duration (Phase 1 default
// for GOOD profile). The loss floor is 5% per Phase 1.
let mut enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
// Decode-side handles.
let mut dec = DecoderHandle::new().unwrap();
let mut dred_dec = DredDecoderHandle::new().unwrap();
let mut state = DredState::new().unwrap();
// Generate 60 frames (1.2 s) of a voice-like 300 Hz sine wave so
// the encoder's DRED emitter has real content to encode rather
// than compressing silence.
let frame_len = 960usize; // 20 ms @ 48 kHz
let make_frame = |offset: usize| -> Vec<i16> {
(0..frame_len)
.map(|i| {
let t = (offset + i) as f64 / 48_000.0;
(8000.0 * (2.0 * std::f64::consts::PI * 300.0 * t).sin()) as i16
})
.collect()
};
// Track the freshest packet that carried non-zero DRED state.
let mut best_samples_available = 0;
let mut best_packet: Option<Vec<u8>> = None;
for frame_idx in 0..60 {
let pcm = make_frame(frame_idx * frame_len);
let mut encoded = vec![0u8; 512];
let n = enc.encode(&pcm, &mut encoded).unwrap();
encoded.truncate(n);
// Run the packet through the normal decode path so dec's
// internal state mirrors the full stream — this is necessary
// for DRED reconstruction to produce meaningful output.
let mut decoded = vec![0i16; frame_len];
dec.decode(&encoded, &mut decoded).unwrap();
// Parse DRED state out of the same packet. Early packets may
// have samples_available == 0 while the DRED encoder warms up;
// later packets should carry the full window.
match dred_dec.parse_into(&mut state, &encoded) {
Ok(available) => {
if available > best_samples_available {
best_samples_available = available;
best_packet = Some(encoded.clone());
}
}
Err(e) => panic!("parse_into errored unexpectedly: {e:?}"),
}
}
// By the time we're 60 frames in, DRED should have emitted data.
assert!(
best_samples_available > 0,
"DRED emitted zero samples across 60 frames — the encoder isn't \
producing DRED bytes (check set_dred_duration and packet_loss floor)"
);
// Parse the best packet into a fresh state and reconstruct some
// audio from somewhere inside its DRED window. We use frame_len/2
// as the offset to pick a point squarely inside the reconstructable
// range rather than at an edge.
let packet = best_packet.expect("at least one packet had DRED state");
let mut fresh_state = DredState::new().unwrap();
let available = dred_dec.parse_into(&mut fresh_state, &packet).unwrap();
assert!(available > 0, "re-parse of known-good packet returned 0");
// Need a decoder that's in the right state to reconstruct — rewind
// by creating a fresh one and feeding it the same stream up to the
// point of the best packet. Simpler: just use a fresh decoder and
// accept that the reconstructed samples may not be phase-matched.
// The test here only asserts *non-silent energy*, not signal fidelity.
let mut recon_dec = DecoderHandle::new().unwrap();
// Warm up the decoder with one frame so its internal state is valid.
let warmup_pcm = vec![0i16; frame_len];
let warmup_encoded = {
let mut warmup_enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
let mut buf = vec![0u8; 512];
let n = warmup_enc.encode(&warmup_pcm, &mut buf).unwrap();
buf.truncate(n);
buf
};
let mut throwaway = vec![0i16; frame_len];
let _ = recon_dec.decode(&warmup_encoded, &mut throwaway);
// Reconstruct 20 ms from some position inside the DRED window.
let offset = (available / 2).max(480).min(available);
let mut recon_pcm = vec![0i16; frame_len];
let n = recon_dec
.reconstruct_from_dred(&fresh_state, offset, &mut recon_pcm)
.expect("reconstruct_from_dred failed");
assert_eq!(n, frame_len);
// Energy check: reconstructed audio should not be all zeros. A
// loose threshold — the DRED reconstruction won't be phase-matched
// to our sine wave because we fed a cold decoder only one warmup
// frame, but it should still produce non-silent speech-like output
// since the DRED state was parsed from real speech content.
let energy: u64 = recon_pcm.iter().map(|&s| (s as i32).unsigned_abs() as u64).sum();
assert!(
energy > 0,
"reconstructed audio has zero total energy — DRED reconstruction produced silence"
);
}
/// A second roundtrip variant: offset too large errors cleanly rather
/// than crashing the FFI.
#[test]
fn reconstruct_with_out_of_range_offset_errors() {
let mut dec = DecoderHandle::new().unwrap();
let state = DredState::new().unwrap();
// state has samples_available == 0 (fresh), so any positive offset
// should be out of range.
let mut out = vec![0i16; 960];
let err = dec.reconstruct_from_dred(&state, 480, &mut out);
assert!(err.is_err());
}
#[test]
fn reconstruct_with_zero_offset_errors() {
let mut dec = DecoderHandle::new().unwrap();
let state = DredState::new().unwrap();
let mut out = vec![0i16; 960];
let err = dec.reconstruct_from_dred(&state, 0, &mut out);
assert!(err.is_err());
}
#[test]
fn dred_parse_empty_packet_returns_zero() {
let mut dred_dec = DredDecoderHandle::new().unwrap();
let mut state = DredState::new().unwrap();
let result = dred_dec.parse_into(&mut state, &[]).unwrap();
assert_eq!(result, 0);
assert_eq!(state.samples_available(), 0);
}
}

View File

@@ -10,19 +10,44 @@
//! 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;
pub mod dred_ffi;
pub mod opus_dec;
pub mod opus_enc;
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};
use std::sync::atomic::{AtomicBool, Ordering};
/// Global verbose-logging flag for DRED. Off by default — when enabled
/// (via the GUI debug toggle wired through Tauri), the encoder logs its
/// DRED config + libopus version, and the recv path logs every DRED
/// reconstruction, classical PLC fill, and parse heartbeat. Off in
/// "normal" mode keeps logcat clean.
static DRED_VERBOSE_LOGS: AtomicBool = AtomicBool::new(false);
/// Returns whether DRED verbose logging is currently enabled.
#[inline]
pub fn dred_verbose_logs() -> bool {
DRED_VERBOSE_LOGS.load(Ordering::Relaxed)
}
/// Enable/disable DRED verbose logging at runtime.
pub fn set_dred_verbose_logs(enabled: bool) {
DRED_VERBOSE_LOGS.store(enabled, Ordering::Relaxed);
}
/// Create an adaptive encoder starting at the given quality profile.
///
/// The returned encoder accepts 48 kHz mono PCM regardless of the active

View File

@@ -1,30 +1,32 @@
//! Opus decoder wrapping the `audiopus` crate.
//! Opus decoder built on top of the raw opusic-sys `DecoderHandle`.
//!
//! Phase 0 of the DRED integration: we went straight to a custom
//! `DecoderHandle` instead of `opusic_c::Decoder` because the latter's
//! inner pointer is `pub(crate)` and we need to reach it in Phase 3 for
//! `opus_decoder_dred_decode`. See `dred_ffi.rs` for the rationale and
//! `docs/PRD-dred-integration.md` for the full plan.
use audiopus::coder::Decoder;
use audiopus::{Channels, MutSignals, SampleRate};
use audiopus::packet::Packet;
use crate::dred_ffi::{DecoderHandle, DredState};
use wzp_proto::{AudioDecoder, CodecError, CodecId, QualityProfile};
/// Opus decoder implementing `AudioDecoder`.
/// Opus decoder implementing [`AudioDecoder`].
///
/// Operates at 48 kHz mono output.
/// Operates at 48 kHz mono output. 20 ms and 40 ms frames supported via
/// the active `QualityProfile`. Behavior is intentionally identical to
/// the pre-swap audiopus-based decoder at this phase — DRED reconstruction
/// lands in Phase 3.
pub struct OpusDecoder {
inner: Decoder,
inner: DecoderHandle,
codec_id: CodecId,
frame_duration_ms: u8,
}
// SAFETY: Same reasoning as OpusEncoder — exclusive access via &mut self.
unsafe impl Sync for OpusDecoder {}
impl OpusDecoder {
/// Create a new Opus decoder for the given quality profile.
pub fn new(profile: QualityProfile) -> Result<Self, CodecError> {
let decoder = Decoder::new(SampleRate::Hz48000, Channels::Mono)
.map_err(|e| CodecError::DecodeFailed(format!("opus decoder init: {e}")))?;
let inner = DecoderHandle::new()?;
Ok(Self {
inner: decoder,
inner,
codec_id: profile.codec,
frame_duration_ms: profile.frame_duration_ms,
})
@@ -34,6 +36,24 @@ impl OpusDecoder {
pub fn frame_samples(&self) -> usize {
(48_000 * self.frame_duration_ms as usize) / 1000
}
/// Reconstruct a lost frame from a previously parsed `DredState`.
///
/// Phase 3b entry point: callers (CallDecoder / engine.rs) use this to
/// synthesize audio for gaps detected by the jitter buffer when DRED
/// side-channel state from a later-arriving packet covers the gap's
/// sample offset. `offset_samples` is measured backward from the anchor
/// packet that produced `state`. See `DecoderHandle::reconstruct_from_dred`
/// for the full semantics.
pub fn reconstruct_from_dred(
&mut self,
state: &DredState,
offset_samples: i32,
output: &mut [i16],
) -> Result<usize, CodecError> {
self.inner
.reconstruct_from_dred(state, offset_samples, output)
}
}
impl AudioDecoder for OpusDecoder {
@@ -45,15 +65,7 @@ impl AudioDecoder for OpusDecoder {
pcm.len()
)));
}
let packet = Packet::try_from(encoded)
.map_err(|e| CodecError::DecodeFailed(format!("invalid packet: {e}")))?;
let signals = MutSignals::try_from(pcm)
.map_err(|e| CodecError::DecodeFailed(format!("output signals: {e}")))?;
let n = self
.inner
.decode(Some(packet), signals, false)
.map_err(|e| CodecError::DecodeFailed(format!("opus decode: {e}")))?;
Ok(n)
self.inner.decode(encoded, pcm)
}
fn decode_lost(&mut self, pcm: &mut [i16]) -> Result<usize, CodecError> {
@@ -64,13 +76,7 @@ impl AudioDecoder for OpusDecoder {
pcm.len()
)));
}
let signals = MutSignals::try_from(pcm)
.map_err(|e| CodecError::DecodeFailed(format!("output signals: {e}")))?;
let n = self
.inner
.decode(None, signals, false)
.map_err(|e| CodecError::DecodeFailed(format!("opus PLC: {e}")))?;
Ok(n)
self.inner.decode_lost(pcm)
}
fn codec_id(&self) -> CodecId {
@@ -79,7 +85,7 @@ 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

@@ -1,53 +1,220 @@
//! Opus encoder wrapping the `audiopus` crate.
//! Opus encoder wrapping the `opusic-c` crate (libopus 1.5.2).
//!
//! Phase 1 of the DRED integration: encoder-side DRED is enabled on every
//! Opus profile with a tiered duration (studio 100 ms / normal 200 ms /
//! degraded 500 ms), and Opus inband FEC (LBRR) is disabled because DRED
//! is the stronger mechanism for the same failure mode. The legacy behavior
//! is preserved behind the `AUDIO_USE_LEGACY_FEC` environment variable as a
//! runtime escape hatch for rollout. See `docs/PRD-dred-integration.md`.
//!
//! # DRED duration policy
//!
//! Rationale from the PRD:
//! - Studio tiers (Opus 32k/48k/64k): 100 ms — loss is rare on high-quality
//! networks; short window keeps decoder CPU modest.
//! - Normal tiers (Opus 16k/24k): 200 ms — balanced baseline covering common
//! VoIP loss patterns (20150 ms bursts from wifi roam, transient congestion).
//! - Degraded tier (Opus 6k): 500 ms — users on 6k are by definition on a
//! bad link; longer DRED buys maximum burst resilience where it matters.
//!
//! # Why the 15% packet loss floor
//!
//! libopus 1.5's DRED emitter is gated on `OPUS_SET_PACKET_LOSS_PERC` and
//! scales the emitted window proportionally to the assumed loss:
//!
//! ```text
//! loss_pct samples_available effective_ms
//! 5% 720 15
//! 10% 2640 55
//! 15% 4560 95
//! 20% 6480 135
//! 25%+ 8400 (capped) 175 (≈ 87% of the 200ms configured max)
//! ```
//!
//! Measured empirically against libopus 1.5.2 on Opus 24k / 200 ms DRED
//! duration during Phase 3b. At 5% loss the window is only 15 ms — too
//! small to even reconstruct a single 20 ms Opus frame. 15% gives 95 ms
//! (enough for single-frame recovery plus modest burst margin) while
//! keeping the bitrate overhead modest compared to 25%. Real measurements
//! from the quality adapter override upward when loss exceeds the floor.
use audiopus::coder::Encoder;
use audiopus::{Application, Bitrate, Channels, SampleRate, Signal};
use tracing::debug;
use std::sync::OnceLock;
use opusic_c::{Application, Bitrate, Channels, Encoder, InbandFec, SampleRate, Signal};
use tracing::{debug, info, warn};
use wzp_proto::{AudioEncoder, CodecError, CodecId, QualityProfile};
/// Logged exactly once per process the first time an OpusEncoder is built.
/// Confirms that libopus 1.5.2 (the version with DRED) is actually linked
/// at runtime — invaluable when chasing "is the new codec loaded?"
/// regressions on Android, where the only debug surface is logcat.
static LIBOPUS_VERSION_LOGGED: OnceLock<()> = OnceLock::new();
/// Minimum `OPUS_SET_PACKET_LOSS_PERC` value used in DRED mode. libopus
/// scales the DRED emission window with the assumed loss percentage:
/// empirically, 5% gives a 15 ms window (useless), 10% gives 55 ms, 15%
/// gives 95 ms, and 25%+ saturates the configured max (~175 ms at 200 ms
/// duration). 15% is the minimum value that produces a DRED window larger
/// than a single 20 ms frame, making it the minimum floor that actually
/// gives DRED something useful to reconstruct. Real loss measurements from
/// the quality adapter override this upward.
const DRED_LOSS_FLOOR_PCT: u8 = 15;
/// Environment variable that reverts Phase 1 behavior to Phase 0 (inband FEC
/// on, DRED off, no loss floor). Read once per encoder construction.
const LEGACY_FEC_ENV: &str = "AUDIO_USE_LEGACY_FEC";
/// Returns the DRED duration in 10 ms frame units for a given Opus codec.
///
/// Unit: each frame is 10 ms, so the max value of 104 corresponds to 1040 ms
/// of reconstructable history. Returns 0 for non-Opus codecs (DRED is not
/// emitted by the libopus encoder in that case anyway, but we avoid a
/// pointless FFI call).
///
/// See the DRED duration policy in the module docs for per-tier rationale.
pub fn dred_duration_for(codec: CodecId) -> u8 {
match codec {
// Studio tiers — loss is rare, short window.
CodecId::Opus32k | CodecId::Opus48k | CodecId::Opus64k => 10,
// Normal tiers — balanced baseline.
CodecId::Opus16k | CodecId::Opus24k => 20,
// Degraded tier — maximum burst resilience.
CodecId::Opus6k => 50,
// Non-Opus (Codec2 / CN): DRED is N/A.
CodecId::Codec2_1200 | CodecId::Codec2_3200 | CodecId::ComfortNoise => 0,
}
}
/// Returns whether the legacy-FEC escape hatch is active.
///
/// Read from `AUDIO_USE_LEGACY_FEC`. Any non-empty value activates legacy
/// mode; unset or empty leaves DRED enabled.
fn read_legacy_fec_env() -> bool {
match std::env::var(LEGACY_FEC_ENV) {
Ok(v) => !v.is_empty() && v != "0" && v.to_ascii_lowercase() != "false",
Err(_) => false,
}
}
/// Opus encoder implementing `AudioEncoder`.
///
/// Operates at 48 kHz mono. Supports frame sizes of 20 ms (960 samples)
/// and 40 ms (1920 samples).
/// Operates at 48 kHz mono. Supports 20 ms and 40 ms frames via the active
/// `QualityProfile`.
pub struct OpusEncoder {
inner: Encoder,
codec_id: CodecId,
frame_duration_ms: u8,
/// When `true`, revert to the Phase 0 behavior: inband FEC Mode1, DRED
/// disabled, no loss floor. Captured at construction time and not
/// re-read mid-call.
legacy_fec_mode: bool,
}
// SAFETY: OpusEncoder is only used via `&mut self` methods. The inner
// audiopus Encoder contains a raw pointer that is !Sync, but we never
// share it across threads without exclusive access.
// opusic-c Encoder wraps a non-null pointer that is !Sync by default,
// but we never share it across threads without exclusive access.
unsafe impl Sync for OpusEncoder {}
impl OpusEncoder {
/// Create a new Opus encoder for the given quality profile.
pub fn new(profile: QualityProfile) -> Result<Self, CodecError> {
let encoder = Encoder::new(SampleRate::Hz48000, Channels::Mono, Application::Voip)
.map_err(|e| CodecError::EncodeFailed(format!("opus encoder init: {e}")))?;
// opusic-c argument order: (Channels, SampleRate, Application)
// — different from audiopus's (SampleRate, Channels, Application).
let encoder = Encoder::new(Channels::Mono, SampleRate::Hz48000, Application::Voip)
.map_err(|e| CodecError::EncodeFailed(format!("opus encoder init: {e:?}")))?;
let legacy_fec_mode = read_legacy_fec_env();
if legacy_fec_mode {
warn!(
"AUDIO_USE_LEGACY_FEC active — reverting Opus encoder to Phase 0 \
behavior (inband FEC Mode1, no DRED)"
);
}
let mut enc = Self {
inner: encoder,
codec_id: profile.codec,
frame_duration_ms: profile.frame_duration_ms,
legacy_fec_mode,
};
enc.apply_bitrate(profile.codec)?;
enc.set_inband_fec(true);
enc.set_dtx(true);
// Voice signal type hint for better compression
// Common setup — bitrate, DTX, signal hint, complexity. These are
// identical regardless of the protection mode below.
enc.apply_bitrate(profile.codec)?;
enc.set_dtx(true);
enc.inner
.set_signal(Signal::Voice)
.map_err(|e| CodecError::EncodeFailed(format!("set signal: {e}")))?;
.map_err(|e| CodecError::EncodeFailed(format!("set signal: {e:?}")))?;
enc.inner
.set_complexity(7)
.map_err(|e| CodecError::EncodeFailed(format!("set complexity: {e:?}")))?;
// Protection mode: DRED (Phase 1 default) or legacy inband FEC.
enc.apply_protection_mode(profile.codec)?;
Ok(enc)
}
fn apply_bitrate(&mut self, codec: CodecId) -> Result<(), CodecError> {
let bps = codec.bitrate_bps() as i32;
/// Configure the protection mode for the active codec.
///
/// In DRED mode (default): disable inband FEC, set DRED duration for the
/// codec tier, clamp packet_loss to the 5% floor so DRED stays active.
///
/// In legacy mode: enable inband FEC Mode1 (Phase 0 behavior), leave
/// DRED and packet_loss at libopus defaults.
fn apply_protection_mode(&mut self, codec: CodecId) -> Result<(), CodecError> {
if self.legacy_fec_mode {
self.inner
.set_bitrate(Bitrate::BitsPerSecond(bps))
.map_err(|e| CodecError::EncodeFailed(format!("set bitrate: {e}")))?;
.set_inband_fec(InbandFec::Mode1)
.map_err(|e| CodecError::EncodeFailed(format!("set inband FEC: {e:?}")))?;
// Leave DRED at 0 and packet_loss at default — matches Phase 0.
return Ok(());
}
// DRED path: disable the overlapping inband FEC, enable DRED with
// per-profile duration, floor packet_loss so DRED emits.
self.inner
.set_inband_fec(InbandFec::Off)
.map_err(|e| CodecError::EncodeFailed(format!("set inband FEC off: {e:?}")))?;
let dred_frames = dred_duration_for(codec);
self.inner
.set_dred_duration(dred_frames)
.map_err(|e| CodecError::EncodeFailed(format!("set DRED duration: {e:?}")))?;
self.inner
.set_packet_loss(DRED_LOSS_FLOOR_PCT)
.map_err(|e| CodecError::EncodeFailed(format!("set packet loss floor: {e:?}")))?;
// Both of these are gated behind the GUI debug toggle so logcat
// stays clean in normal mode. Flip "DRED verbose logs" in the
// settings panel to see the per-encoder config + libopus version.
if crate::dred_verbose_logs() {
info!(
codec = ?codec,
dred_frames,
dred_ms = dred_frames as u32 * 10,
loss_floor_pct = DRED_LOSS_FLOOR_PCT,
"opus encoder: DRED enabled"
);
// One-shot logging of the linked libopus version so we can
// confirm at a glance that opusic-c (libopus 1.5.2) is loaded.
// Pre-Phase-0 audiopus shipped libopus 1.3 which has no DRED;
// if this log says "libopus 1.3" something is very wrong.
LIBOPUS_VERSION_LOGGED.get_or_init(|| {
info!(libopus_version = %opusic_c::version(), "linked libopus version");
});
}
Ok(())
}
fn apply_bitrate(&mut self, codec: CodecId) -> Result<(), CodecError> {
let bps = codec.bitrate_bps();
self.inner
.set_bitrate(Bitrate::Value(bps))
.map_err(|e| CodecError::EncodeFailed(format!("set bitrate: {e:?}")))?;
debug!(bitrate_bps = bps, "opus encoder bitrate set");
Ok(())
}
@@ -56,6 +223,47 @@ 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).
///
/// In DRED mode, the value is floored at `DRED_LOSS_FLOOR_PCT` so the
/// encoder never drops DRED emission even on a perfect network. Real
/// loss measurements from the quality adapter override upward.
///
/// In legacy mode, the value is passed through unchanged (min 0, max 100).
pub fn set_expected_loss(&mut self, loss_pct: u8) {
let clamped = if self.legacy_fec_mode {
loss_pct.min(100)
} else {
loss_pct.max(DRED_LOSS_FLOOR_PCT).min(100)
};
let _ = self.inner.set_packet_loss(clamped);
}
/// Set the DRED duration in 10 ms frame units (0 disables, max 104).
///
/// No-op in legacy mode. Normally driven automatically by the active
/// quality profile via `apply_protection_mode`; this setter exists for
/// tests and for the rare case where a caller needs to override the
/// per-profile default.
pub fn set_dred_duration(&mut self, frames: u8) {
if self.legacy_fec_mode {
return;
}
let _ = self.inner.set_dred_duration(frames.min(104));
}
/// Test/introspection accessor: whether legacy FEC mode is active.
pub fn is_legacy_fec_mode(&self) -> bool {
self.legacy_fec_mode
}
}
impl AudioEncoder for OpusEncoder {
@@ -67,10 +275,14 @@ impl AudioEncoder for OpusEncoder {
pcm.len()
)));
}
// opusic-c takes &[u16] for the sample input. Bit pattern is
// identical to i16 — the cast is zero-cost and the encoder
// interprets the bytes the same way as libopus internally.
let pcm_u16: &[u16] = bytemuck::cast_slice(pcm);
let n = self
.inner
.encode(pcm, out)
.map_err(|e| CodecError::EncodeFailed(format!("opus encode: {e}")))?;
.encode_to_slice(pcm_u16, out)
.map_err(|e| CodecError::EncodeFailed(format!("opus encode: {e:?}")))?;
Ok(n)
}
@@ -80,10 +292,13 @@ 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)?;
// Refresh DRED duration for the new tier. apply_protection_mode
// is idempotent and handles the legacy-vs-DRED branch correctly.
self.apply_protection_mode(profile.codec)?;
Ok(())
}
other => Err(CodecError::UnsupportedTransition {
@@ -100,10 +315,190 @@ impl AudioEncoder for OpusEncoder {
}
fn set_inband_fec(&mut self, enabled: bool) {
let _ = self.inner.set_inband_fec(enabled);
// In DRED mode, ignore external requests to re-enable inband FEC —
// running both mechanisms wastes bitrate on overlapping protection
// and opusic-c's own docs recommend disabling inband FEC when DRED
// is on. Trait callers that genuinely want classical FEC should set
// `AUDIO_USE_LEGACY_FEC=1` and re-create the encoder.
if !self.legacy_fec_mode {
debug!(
enabled,
"set_inband_fec ignored: DRED mode is active (set AUDIO_USE_LEGACY_FEC to revert)"
);
return;
}
let mode = if enabled { InbandFec::Mode1 } else { InbandFec::Off };
let _ = self.inner.set_inband_fec(mode);
}
fn set_dtx(&mut self, enabled: bool) {
let _ = self.inner.set_dtx(enabled);
}
}
#[cfg(test)]
mod tests {
use super::*;
use wzp_proto::AudioDecoder;
/// Phase 0 acceptance gate: fail loudly if the linked libopus is not 1.5.x.
/// DRED (Phase 1+) only exists in libopus ≥ 1.5, so running against an
/// older version would silently regress the entire DRED integration.
#[test]
fn linked_libopus_is_1_5() {
let version = opusic_c::version();
assert!(
version.contains("1.5"),
"expected libopus 1.5.x, got: {version}"
);
}
#[test]
fn encoder_creates_at_good_profile() {
let enc = OpusEncoder::new(QualityProfile::GOOD).expect("opus encoder init");
assert_eq!(enc.codec_id, CodecId::Opus24k);
assert_eq!(enc.frame_samples(), 960); // 20 ms @ 48 kHz
}
#[test]
fn encoder_roundtrip_silence() {
let mut enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
let mut dec = crate::opus_dec::OpusDecoder::new(QualityProfile::GOOD).unwrap();
let pcm_in = vec![0i16; 960]; // 20 ms silence
let mut encoded = vec![0u8; 512];
let n = enc.encode(&pcm_in, &mut encoded).unwrap();
assert!(n > 0);
let mut pcm_out = vec![0i16; 960];
let samples = dec.decode(&encoded[..n], &mut pcm_out).unwrap();
assert_eq!(samples, 960);
}
// ─── Phase 1 — DRED duration policy ─────────────────────────────────────
#[test]
fn dred_duration_for_studio_tiers_is_100ms() {
assert_eq!(dred_duration_for(CodecId::Opus32k), 10);
assert_eq!(dred_duration_for(CodecId::Opus48k), 10);
assert_eq!(dred_duration_for(CodecId::Opus64k), 10);
}
#[test]
fn dred_duration_for_normal_tiers_is_200ms() {
assert_eq!(dred_duration_for(CodecId::Opus16k), 20);
assert_eq!(dred_duration_for(CodecId::Opus24k), 20);
}
#[test]
fn dred_duration_for_degraded_tier_is_500ms() {
assert_eq!(dred_duration_for(CodecId::Opus6k), 50);
}
#[test]
fn dred_duration_for_codec2_is_zero() {
assert_eq!(dred_duration_for(CodecId::Codec2_3200), 0);
assert_eq!(dred_duration_for(CodecId::Codec2_1200), 0);
assert_eq!(dred_duration_for(CodecId::ComfortNoise), 0);
}
// ─── Phase 1 — Legacy escape hatch ──────────────────────────────────────
/// By default (env var unset), legacy mode is off.
///
/// This test does NOT manipulate the environment to avoid flakiness
/// when the full suite runs in parallel. It only asserts on a freshly
/// created encoder in the ambient environment.
#[test]
fn default_mode_is_dred_not_legacy() {
// SAFETY: only run if the ambient env hasn't set the var externally.
if std::env::var(LEGACY_FEC_ENV).is_ok() {
return; // don't assert — someone set the env for a reason.
}
let enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
assert!(!enc.is_legacy_fec_mode());
}
// ─── Phase 1 — Behavioral regression: roundtrip still works ─────────────
#[test]
fn dred_mode_roundtrip_voice_pattern() {
// Use a realistic voice-like input (sine wave at speech frequencies)
// so the encoder emits meaningful DRED data rather than trivially
// compressible silence.
let mut enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
let mut dec = crate::opus_dec::OpusDecoder::new(QualityProfile::GOOD).unwrap();
let mut total_encoded_bytes = 0usize;
// Run 50 frames (1 second) so DRED fills up and starts emitting.
for frame_idx in 0..50 {
let pcm_in: Vec<i16> = (0..960)
.map(|i| {
let t = (frame_idx * 960 + i) as f64 / 48_000.0;
(8000.0 * (2.0 * std::f64::consts::PI * 300.0 * t).sin()) as i16
})
.collect();
let mut encoded = vec![0u8; 512];
let n = enc.encode(&pcm_in, &mut encoded).unwrap();
assert!(n > 0);
total_encoded_bytes += n;
let mut pcm_out = vec![0i16; 960];
let samples = dec.decode(&encoded[..n], &mut pcm_out).unwrap();
assert_eq!(samples, 960);
}
// Effective bitrate after 1 second of encoding.
// Opus 24k base + ~1 kbps DRED ≈ 25 kbps ≈ 3125 bytes/sec.
// Allow generous headroom (2000 lower bound, 8000 upper bound) —
// this is a behavioral regression check, not a tight bitrate assertion.
// The exact value is printed with --nocapture for diagnostic use.
eprintln!(
"[phase1 bitrate probe] legacy_fec_mode={} total_encoded={} bytes/sec",
enc.is_legacy_fec_mode(),
total_encoded_bytes
);
assert!(
total_encoded_bytes > 2000,
"encoder output too small: {total_encoded_bytes} bytes/sec (DRED likely not emitting)"
);
assert!(
total_encoded_bytes < 8000,
"encoder output too large: {total_encoded_bytes} bytes/sec"
);
}
// ─── Phase 1 — set_profile updates DRED duration on tier switch ─────────
#[test]
fn profile_switch_refreshes_dred_duration() {
// Start on GOOD (Opus 24k, DRED 20 frames), switch to DEGRADED
// (Opus 6k, DRED 50 frames). The encoder should accept both profile
// changes without error. We can't directly observe the DRED duration
// inside libopus, but apply_protection_mode returns Ok for both.
let mut enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
assert_eq!(enc.codec_id, CodecId::Opus24k);
enc.set_profile(QualityProfile::DEGRADED).unwrap();
assert_eq!(enc.codec_id, CodecId::Opus6k);
enc.set_profile(QualityProfile::STUDIO_64K).unwrap();
assert_eq!(enc.codec_id, CodecId::Opus64k);
}
// ─── Phase 1 — Trait set_inband_fec is a no-op in DRED mode ─────────────
#[test]
fn set_inband_fec_noop_in_dred_mode() {
if std::env::var(LEGACY_FEC_ENV).is_ok() {
return;
}
let mut enc = OpusEncoder::new(QualityProfile::GOOD).unwrap();
// Should not error, should not re-enable inband FEC internally.
enc.set_inband_fec(true);
// We can't directly query libopus's inband FEC state through opusic-c,
// but the call must not panic and the encoder must still work.
let pcm_in = vec![0i16; 960];
let mut encoded = vec![0u8; 512];
let n = enc.encode(&pcm_in, &mut encoded).unwrap();
assert!(n > 0);
}
}

View File

@@ -1,54 +1,257 @@
//! 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).
///
/// 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> {
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).
///
/// 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;
}
}
sum
}
/// Build a windowed-sinc low-pass FIR kernel.
///
/// 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);
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 {
(2.0 * PI * fc * n).sin() / (PI * n)
};
// 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;
}
}
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 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);
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
}
/// Upsample from 8 kHz to 48 kHz (1:6 interpolation with linear interp).
///
/// 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();
}
let out_len = input.len() * RATIO;
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);
for i in 0..input.len() {
let current = input[i] as i32;
let next = if i + 1 < input.len() {
input[i + 1] as i32
} else {
current // hold last sample
};
// The gain factor compensates for the zeros introduced by stuffing.
let gain = RATIO as f64;
for j in 0..RATIO {
let interp = current + (next - current) * j as i32 / RATIO as i32;
output.push(interp as i16);
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 {
@@ -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

@@ -115,6 +115,7 @@ fn wzp_signal_serializes_into_fc_callsignal_payload() {
ephemeral_pub: [2u8; 32],
signature: vec![3u8; 64],
supported_profiles: vec![wzp_proto::QualityProfile::GOOD],
alias: None,
};
// Encode as featherChat CallSignal payload
@@ -198,6 +199,7 @@ fn wzp_answer_round_trips_through_fc_callsignal() {
fn wzp_hangup_round_trips_through_fc_callsignal() {
let hangup = wzp_proto::SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
call_id: None,
};
let payload = wzp_client::featherchat::encode_call_payload(&hangup, None, None);
@@ -273,13 +275,14 @@ fn auth_invalid_response_matches() {
#[test]
fn all_signal_types_map_correctly() {
use wzp_client::featherchat::{signal_to_call_type, CallSignalType};
use wzp_client::featherchat::signal_to_call_type;
let cases: Vec<(wzp_proto::SignalMessage, &str)> = vec![
(
wzp_proto::SignalMessage::CallOffer {
identity_pub: [0; 32], ephemeral_pub: [0; 32],
signature: vec![], supported_profiles: vec![],
alias: None,
},
"Offer",
),
@@ -300,6 +303,7 @@ fn all_signal_types_map_correctly() {
(
wzp_proto::SignalMessage::Hangup {
reason: wzp_proto::HangupReason::Normal,
call_id: None,
},
"Hangup",
),

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,
})
}
}
@@ -74,9 +81,21 @@ impl FecDecoder for RaptorQFecDecoder {
let block = self.get_or_create_block(block_id);
if block.decoded {
// Already decoded, ignore additional symbols.
// 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).
// But if caller sends raw data, pad it.
@@ -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,
}
}

View File

@@ -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

View File

@@ -0,0 +1,443 @@
// 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::Shared)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRateConversionQuality(oboe::SampleRateConversionQuality::Best)
->setDataCallback(&g_capture_cb);
if (config->bt_active) {
// BT SCO mode: do NOT set sample rate or input preset.
// Requesting 48kHz against a BT SCO device fails with
// "getInputProfile could not find profile". Letting the system
// choose the native rate (8/16kHz) and relying on Oboe's
// resampler (SampleRateConversionQuality::Best) to bridge
// to our 48kHz ring buffer is the only path that works.
// InputPreset::VoiceCommunication can also prevent BT SCO
// routing on some devices — skip it for BT.
LOGI("capture: BT mode — no sample rate or input preset set");
} else {
captureBuilder.setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setInputPreset(oboe::InputPreset::VoiceCommunication);
}
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::Shared)
->setFormat(oboe::AudioFormat::I16)
->setChannelCount(config->channel_count)
->setSampleRateConversionQuality(oboe::SampleRateConversionQuality::Best)
->setDataCallback(&g_playout_cb);
if (config->bt_active) {
LOGI("playout: BT mode — no sample rate set, using Usage::Media");
// Usage::Media instead of VoiceCommunication for BT output
// to avoid conflicts with the communication device routing.
playoutBuilder.setUsage(oboe::Usage::Media);
} else {
playoutBuilder.setSampleRate(config->sample_rate)
->setFramesPerDataCallback(config->frames_per_burst)
->setUsage(oboe::Usage::VoiceCommunication);
}
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 +1,44 @@
#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;
int32_t bt_active; /* nonzero = BT SCO mode: skip sample rate + input preset */
} 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

View File

@@ -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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@@ -0,0 +1,427 @@
//! 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,
/// When nonzero, capture stream skips setSampleRate and setInputPreset
/// so the system can route to BT SCO at its native rate (8/16kHz).
/// Oboe's SampleRateConversionQuality::Best resamples to 48kHz.
bt_active: 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,
/// Fix A (task #35): the playout ring's read_idx at the last
/// check. If audio_write_playout observes read_idx hasn't
/// advanced after N writes, the Oboe playout callback has
/// stopped firing → restart the streams.
playout_last_read_idx: std::sync::atomic::AtomicI32,
/// Number of writes since the last read_idx advance.
playout_stall_writes: std::sync::atomic::AtomicU32,
}
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),
playout_last_read_idx: std::sync::atomic::AtomicI32::new(0),
playout_stall_writes: std::sync::atomic::AtomicU32::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 {
audio_start_inner(false)
}
/// Start Oboe in Bluetooth SCO mode — skips sample rate and input preset
/// on capture so the system can route to the BT SCO device natively.
#[unsafe(no_mangle)]
pub extern "C" fn wzp_native_audio_start_bt() -> i32 {
audio_start_inner(true)
}
fn audio_start_inner(bt: bool) -> 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,
bt_active: if bt { 1 } else { 0 },
};
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();
// Fix A (task #35): detect playout callback stall. If the
// playout ring's read_idx hasn't advanced in 50+ writes
// (~1 second at 50 writes/sec), the Oboe playout callback
// has stopped firing → restart the streams. This is the
// self-healing behavior that makes rejoin work: teardown +
// rebuild clears whatever HAL state locked up the callback.
let current_read_idx = b.playout.read_idx.load(std::sync::atomic::Ordering::Relaxed);
let last_read_idx = b.playout_last_read_idx.load(std::sync::atomic::Ordering::Relaxed);
if current_read_idx == last_read_idx {
let stall = b.playout_stall_writes.fetch_add(1, std::sync::atomic::Ordering::Relaxed);
if stall >= 50 {
// Callback hasn't drained anything in ~1 second.
// Force a stream restart.
unsafe {
android_log("playout STALL detected (50 writes, read_idx unchanged) — restarting Oboe streams");
}
b.playout_stall_writes.store(0, std::sync::atomic::Ordering::Relaxed);
// Release the started lock, stop, re-start.
// This is the same logic as the Rust-side
// audio_stop() + audio_start() but done inline
// because we can't call the extern "C" fns
// recursively. Just call the C++ side directly.
{
if let Ok(mut started) = b.started.lock() {
if *started {
unsafe { wzp_oboe_stop() };
*started = false;
}
}
}
// Clear the rings so the restart doesn't read stale data
b.playout.write_idx.store(0, std::sync::atomic::Ordering::Relaxed);
b.playout.read_idx.store(0, std::sync::atomic::Ordering::Relaxed);
b.capture.write_idx.store(0, std::sync::atomic::Ordering::Relaxed);
b.capture.read_idx.store(0, std::sync::atomic::Ordering::Relaxed);
// Re-start (stall detector — always non-BT mode)
let config = WzpOboeConfig {
sample_rate: 48_000,
frames_per_burst: FRAME_SAMPLES as i32,
channel_count: 1,
bt_active: 0,
};
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 {
if let Ok(mut started) = b.started.lock() {
*started = true;
}
unsafe { android_log("playout restart OK — Oboe streams rebuilt"); }
} else {
unsafe { android_log(&format!("playout restart FAILED: {ret}")); }
}
b.playout_last_read_idx.store(0, std::sync::atomic::Ordering::Relaxed);
return 0; // caller will retry on next frame
}
} else {
// read_idx advanced — callback is alive, reset counter
b.playout_stall_writes.store(0, std::sync::atomic::Ordering::Relaxed);
b.playout_last_read_idx.store(current_read_idx, std::sync::atomic::Ordering::Relaxed);
}
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

@@ -53,6 +53,15 @@ pub enum TransportError {
Timeout { ms: u64 },
#[error("io error: {0}")]
Io(#[from] std::io::Error),
/// Parsed wire bytes successfully but the payload didn't
/// deserialize into a known `SignalMessage` variant. Usually
/// means the peer is running a newer build with a variant we
/// don't know yet. Callers should **log and continue** rather
/// than tearing down the connection, so that forward-compat
/// additions to `SignalMessage` don't silently kill old
/// clients/relays.
#[error("signal deserialize: {0}")]
Deserialize(String),
#[error("internal transport error: {0}")]
Internal(String),
}

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,
}
}
@@ -64,14 +80,39 @@ 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());
}
// 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() + SAFETY_MARGIN_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);
self.last_expected_ms = Some(expected_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);
}
}
}
// ---------------------------------------------------------------------------
@@ -210,11 +273,22 @@ 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) {
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
if self.stats.packets_played == 0 && seq_before(seq, self.next_playout_seq) {
@@ -349,11 +423,22 @@ 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) {
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
if self.stats.packets_played == 0 && seq_before(seq, self.next_playout_seq) {
@@ -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).
@@ -581,12 +584,38 @@ pub enum SignalMessage {
recommended_profile: crate::QualityProfile,
},
/// Phase 4 telemetry: loss-recovery counts for the current session.
/// Sent periodically from receivers to the relay so Prometheus metrics
/// can distinguish DRED reconstructions from classical PLC invocations.
/// Fields default to 0 on old receivers (`#[serde(default)]`), so
/// introducing this variant is backward-compatible with pre-Phase-4
/// relays — they'll just log "unknown signal variant" on receipt.
LossRecoveryUpdate {
/// Total frames reconstructed via DRED since call start (monotonic).
#[serde(default)]
dred_reconstructions: u64,
/// Total frames filled via classical Opus/Codec2 PLC since call
/// start (monotonic).
#[serde(default)]
classical_plc_invocations: u64,
/// Total frames decoded since call start. Used by the relay to
/// compute recovery rates as a fraction of total frames.
#[serde(default)]
frames_decoded: u64,
},
/// Connection keepalive / RTT measurement.
Ping { timestamp_ms: u64 },
Pong { timestamp_ms: u64 },
/// End the call.
Hangup { reason: HangupReason },
/// End the call. `call_id` is optional for backwards compatibility
/// with older clients that send Hangup without it — the relay falls
/// back to ending ALL active calls for the sender in that case.
Hangup {
reason: HangupReason,
#[serde(default, skip_serializing_if = "Option::is_none")]
call_id: Option<String>,
},
/// featherChat bearer token for relay authentication.
/// Sent as the first signal message when --auth-url is configured.
@@ -645,6 +674,274 @@ 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>,
/// Relay's build version (git short hash).
#[serde(default, skip_serializing_if = "Option::is_none")]
relay_build: 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>,
/// Phase 3 (hole-punching): caller's own server-reflexive
/// address as learned via `SignalMessage::Reflect`. The
/// relay stashes this in its call registry and later
/// injects it into the callee's `CallSetup.peer_direct_addr`
/// so the callee can try a direct QUIC handshake to the
/// caller instead of routing media through the relay.
/// `None` means "caller doesn't want P2P, use relay only".
#[serde(default, skip_serializing_if = "Option::is_none")]
caller_reflexive_addr: Option<String>,
/// Phase 5.5 (ICE host candidates): caller's LAN-local
/// interface addresses paired with its signal endpoint's
/// port. Peers on the same physical LAN can direct-dial
/// these without going through the WAN reflex addr,
/// which is important because most consumer NATs
/// (including MikroTik masquerade) don't support NAT
/// hairpinning — the reflex addr is unreachable from
/// the same LAN.
#[serde(default, skip_serializing_if = "Vec::is_empty")]
caller_local_addrs: Vec<String>,
/// Build version (git short hash) for debugging.
#[serde(default, skip_serializing_if = "Option::is_none")]
caller_build_version: Option<String>,
},
/// 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>,
/// Phase 3 (hole-punching): callee's own server-reflexive
/// address, only populated on `AcceptTrusted` — privacy-mode
/// answers leave this `None` so the callee's real IP stays
/// hidden (the whole point of `AcceptGeneric`). The relay
/// carries it opaquely into the caller's `CallSetup`.
#[serde(default, skip_serializing_if = "Option::is_none")]
callee_reflexive_addr: Option<String>,
/// Phase 5.5 (ICE host candidates): callee's LAN-local
/// interface addresses. Same purpose as
/// `caller_local_addrs` in `DirectCallOffer`. Only
/// populated on `AcceptTrusted` alongside
/// `callee_reflexive_addr`.
#[serde(default, skip_serializing_if = "Vec::is_empty")]
callee_local_addrs: Vec<String>,
/// Build version (git short hash) for debugging.
#[serde(default, skip_serializing_if = "Option::is_none")]
callee_build_version: Option<String>,
},
/// 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,
/// Phase 3 (hole-punching): the OTHER party's server-reflexive
/// address as the relay learned it from the offer/answer
/// exchange. When populated, clients attempt a direct QUIC
/// handshake to this address in parallel with the existing
/// relay path and use whichever connects first. `None`
/// means the relay path is the only option — either because
/// a peer didn't advertise its addr (Phase 1/2 relay or
/// privacy-mode answer) or because the relay decided P2P
/// wasn't viable.
#[serde(default, skip_serializing_if = "Option::is_none")]
peer_direct_addr: Option<String>,
/// Phase 5.5 (ICE host candidates): the OTHER party's LAN
/// host addresses (RFC1918 IPv4 + CGNAT + non-link-local
/// IPv6). On same-LAN calls these are directly dialable
/// and bypass the NAT-hairpinning problem that blocks
/// same-LAN peers from using `peer_direct_addr`.
/// Client-side race tries all of these in parallel.
#[serde(default, skip_serializing_if = "Vec::is_empty")]
peer_local_addrs: Vec<String>,
},
/// Ringing notification (relay → caller, callee received the offer).
CallRinging {
call_id: String,
},
// ── NAT reflection ("STUN for QUIC") ──────────────────────────────
/// Client → relay: "please tell me the source IP:port you see on
/// this connection". A QUIC-native replacement for classic STUN
/// that reuses the TLS-authenticated signal channel to the relay
/// instead of running a separate UDP reflection service on port
/// 3478. The relay answers with `ReflectResponse`.
///
/// No payload — the relay already knows which connection the
/// request arrived on, and `connection.remote_address()` gives it
/// the exact source address (post-NAT) as observed from the
/// server side of the TLS session.
Reflect,
/// Relay → client: response to `Reflect`. Carries the socket
/// address the relay observes as the client's source for this
/// QUIC connection in `SocketAddr::to_string()` form — "a.b.c.d:p"
/// for IPv4, "[::1]:p" for IPv6. Clients parse it with
/// `SocketAddr::from_str`.
ReflectResponse {
observed_addr: String,
},
// ── Phase 6: ICE-style path negotiation ─────────────────────
/// Phase 6: each side reports the result of its local dual-
/// path race to the other side through the relay. Both peers
/// send this after their race completes; both wait for the
/// other's report before committing a transport to the
/// CallEngine.
///
/// The decision rule is: if BOTH sides report `direct_ok =
/// true`, use the direct P2P connection. If EITHER reports
/// `direct_ok = false`, BOTH fall back to relay. This
/// eliminates the race condition where one side picks Direct
/// and the other picks Relay — they now agree on the path
/// before any media flows.
MediaPathReport {
call_id: String,
/// Did the direct QUIC connection (P2P dial or accept)
/// complete successfully on this side?
direct_ok: bool,
/// Which future won the local tokio::select race?
/// "Direct" or "Relay" — informational for debug logs.
#[serde(default)]
race_winner: String,
},
// ── Phase 4: cross-relay direct-call signaling ────────────────────
/// Phase 4: relay-to-relay envelope for forwarding direct-call
/// signaling across a federation link. When Alice on Relay A
/// sends a `DirectCallOffer` for Bob whose fingerprint isn't
/// in A's local SignalHub, Relay A wraps the offer in this
/// envelope and broadcasts it over every active federation
/// peer link. Whichever peer has Bob registered unwraps the
/// inner message and delivers it locally.
///
/// Never originated by clients — only relays create and
/// consume this variant.
///
/// Loop prevention: the receiving relay drops any forward
/// where `origin_relay_fp` matches its own federation TLS
/// fingerprint. With broadcast-to-all-peers this prevents
/// A→B→A echo loops; proper TTL + dedup will land when
/// multi-hop federation is added (Phase 4.2).
FederatedSignalForward {
/// The signal message being forwarded
/// (`DirectCallOffer`, `DirectCallAnswer`, `CallRinging`,
/// `Hangup`, ...). Boxed because `SignalMessage` is
/// relatively large and JSON serde handles recursion
/// cleanly.
inner: Box<SignalMessage>,
/// Federation TLS fingerprint of the sending relay.
/// Used (a) for loop prevention by the receiver and (b)
/// to route the peer's reply back through the same
/// federation link via `send_signal_to_peer`.
origin_relay_fp: 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.
@@ -758,6 +1055,267 @@ mod tests {
assert_eq!(packet.quality_report, decoded.quality_report);
}
#[test]
fn reflect_serialize_roundtrip() {
// Reflect is a unit variant — the client sends it with no
// payload and the relay answers with the observed source addr.
let req = SignalMessage::Reflect;
let json = serde_json::to_string(&req).unwrap();
let decoded: SignalMessage = serde_json::from_str(&json).unwrap();
assert!(matches!(decoded, SignalMessage::Reflect));
// ReflectResponse carries a string — exercise both IPv4 and
// IPv6 shapes because SocketAddr::to_string uses [::1]:port
// for v6 and the client side has to parse that back.
for addr in ["192.0.2.17:4433", "[2001:db8::1]:4433", "127.0.0.1:54321"] {
let resp = SignalMessage::ReflectResponse {
observed_addr: addr.to_string(),
};
let json = serde_json::to_string(&resp).unwrap();
let decoded: SignalMessage = serde_json::from_str(&json).unwrap();
match decoded {
SignalMessage::ReflectResponse { observed_addr } => {
assert_eq!(observed_addr, addr);
// Must parse back to a SocketAddr cleanly.
let _parsed: std::net::SocketAddr = observed_addr.parse()
.expect("observed_addr must parse as SocketAddr");
}
_ => panic!("wrong variant after roundtrip"),
}
}
}
#[test]
fn federated_signal_forward_roundtrip() {
// Wrap a DirectCallOffer inside FederatedSignalForward and
// prove both directions of serde preserve every field.
let inner = SignalMessage::DirectCallOffer {
caller_fingerprint: "alice".into(),
caller_alias: Some("Alice".into()),
target_fingerprint: "bob".into(),
call_id: "c1".into(),
identity_pub: [1u8; 32],
ephemeral_pub: [2u8; 32],
signature: vec![3u8; 64],
supported_profiles: vec![],
caller_reflexive_addr: Some("192.0.2.1:4433".into()),
caller_local_addrs: Vec::new(),
};
let forward = SignalMessage::FederatedSignalForward {
inner: Box::new(inner),
origin_relay_fp: "relay-a-tls-fp".into(),
};
let json = serde_json::to_string(&forward).unwrap();
let decoded: SignalMessage = serde_json::from_str(&json).unwrap();
match decoded {
SignalMessage::FederatedSignalForward { inner, origin_relay_fp } => {
assert_eq!(origin_relay_fp, "relay-a-tls-fp");
match *inner {
SignalMessage::DirectCallOffer {
caller_fingerprint,
target_fingerprint,
caller_reflexive_addr,
..
} => {
assert_eq!(caller_fingerprint, "alice");
assert_eq!(target_fingerprint, "bob");
assert_eq!(caller_reflexive_addr.as_deref(), Some("192.0.2.1:4433"));
}
_ => panic!("inner was not DirectCallOffer after roundtrip"),
}
}
_ => panic!("outer was not FederatedSignalForward"),
}
}
#[test]
fn federated_signal_forward_can_nest_any_inner() {
// Sanity check that every direct-call signaling variant
// we intend to forward survives being boxed + re-serialized.
let cases: Vec<SignalMessage> = vec![
SignalMessage::DirectCallAnswer {
call_id: "c1".into(),
accept_mode: CallAcceptMode::AcceptTrusted,
identity_pub: None,
ephemeral_pub: None,
signature: None,
chosen_profile: None,
callee_reflexive_addr: Some("198.51.100.9:4433".into()),
callee_local_addrs: Vec::new(),
},
SignalMessage::CallRinging { call_id: "c1".into() },
SignalMessage::Hangup { reason: HangupReason::Normal, call_id: None },
];
for inner in cases {
let inner_disc = std::mem::discriminant(&inner);
let forward = SignalMessage::FederatedSignalForward {
inner: Box::new(inner),
origin_relay_fp: "r".into(),
};
let json = serde_json::to_string(&forward).unwrap();
let decoded: SignalMessage = serde_json::from_str(&json).unwrap();
match decoded {
SignalMessage::FederatedSignalForward { inner, .. } => {
assert_eq!(std::mem::discriminant(&*inner), inner_disc);
}
_ => panic!("outer variant lost"),
}
}
}
#[test]
fn hole_punching_optional_fields_roundtrip() {
// DirectCallOffer with Some(caller_reflexive_addr)
let offer = SignalMessage::DirectCallOffer {
caller_fingerprint: "alice".into(),
caller_alias: None,
target_fingerprint: "bob".into(),
call_id: "c1".into(),
identity_pub: [0; 32],
ephemeral_pub: [0; 32],
signature: vec![],
supported_profiles: vec![],
caller_reflexive_addr: Some("192.0.2.1:4433".into()),
caller_local_addrs: Vec::new(),
};
let json = serde_json::to_string(&offer).unwrap();
assert!(
json.contains("caller_reflexive_addr"),
"Some field must serialize: {json}"
);
let decoded: SignalMessage = serde_json::from_str(&json).unwrap();
match decoded {
SignalMessage::DirectCallOffer { caller_reflexive_addr, .. } => {
assert_eq!(caller_reflexive_addr.as_deref(), Some("192.0.2.1:4433"));
}
_ => panic!("wrong variant"),
}
// DirectCallOffer with None — skip_serializing_if must
// OMIT the field from the JSON so older relays that don't
// know about caller_reflexive_addr don't see it.
let offer_none = SignalMessage::DirectCallOffer {
caller_fingerprint: "alice".into(),
caller_alias: None,
target_fingerprint: "bob".into(),
call_id: "c1".into(),
identity_pub: [0; 32],
ephemeral_pub: [0; 32],
signature: vec![],
supported_profiles: vec![],
caller_reflexive_addr: None,
caller_local_addrs: Vec::new(),
};
let json_none = serde_json::to_string(&offer_none).unwrap();
assert!(
!json_none.contains("caller_reflexive_addr"),
"None field must NOT serialize: {json_none}"
);
// DirectCallAnswer with callee_reflexive_addr.
let answer = SignalMessage::DirectCallAnswer {
call_id: "c1".into(),
accept_mode: CallAcceptMode::AcceptTrusted,
identity_pub: None,
ephemeral_pub: None,
signature: None,
chosen_profile: None,
callee_reflexive_addr: Some("198.51.100.9:4433".into()),
callee_local_addrs: Vec::new(),
};
let decoded: SignalMessage =
serde_json::from_str(&serde_json::to_string(&answer).unwrap()).unwrap();
match decoded {
SignalMessage::DirectCallAnswer { callee_reflexive_addr, .. } => {
assert_eq!(
callee_reflexive_addr.as_deref(),
Some("198.51.100.9:4433")
);
}
_ => panic!("wrong variant"),
}
// CallSetup with peer_direct_addr.
let setup = SignalMessage::CallSetup {
call_id: "c1".into(),
room: "call-c1".into(),
relay_addr: "203.0.113.5:4433".into(),
peer_direct_addr: Some("192.0.2.1:4433".into()),
peer_local_addrs: Vec::new(),
};
let decoded: SignalMessage =
serde_json::from_str(&serde_json::to_string(&setup).unwrap()).unwrap();
match decoded {
SignalMessage::CallSetup { peer_direct_addr, .. } => {
assert_eq!(peer_direct_addr.as_deref(), Some("192.0.2.1:4433"));
}
_ => panic!("wrong variant"),
}
}
#[test]
fn hole_punching_backward_compat_old_json_parses() {
// An older client/relay wouldn't include the new fields at
// all — the new code must still accept that JSON because
// of #[serde(default)] on the Option<String>.
let old_offer_json = r#"{
"DirectCallOffer": {
"caller_fingerprint": "alice",
"caller_alias": null,
"target_fingerprint": "bob",
"call_id": "c1",
"identity_pub": [0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0],
"ephemeral_pub": [0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0],
"signature": [],
"supported_profiles": []
}
}"#;
let decoded: SignalMessage = serde_json::from_str(old_offer_json).unwrap();
match decoded {
SignalMessage::DirectCallOffer { caller_reflexive_addr, .. } => {
assert!(caller_reflexive_addr.is_none());
}
_ => panic!("wrong variant"),
}
let old_setup_json = r#"{
"CallSetup": {
"call_id": "c1",
"room": "call-c1",
"relay_addr": "203.0.113.5:4433"
}
}"#;
let decoded: SignalMessage = serde_json::from_str(old_setup_json).unwrap();
match decoded {
SignalMessage::CallSetup { peer_direct_addr, .. } => {
assert!(peer_direct_addr.is_none());
}
_ => panic!("wrong variant"),
}
}
#[test]
fn reflect_backward_compat_with_existing_variants() {
// Adding Reflect/ReflectResponse at the end of the enum must
// not break JSON round-tripping of existing variants. Smoke-
// test a sample of the pre-existing ones.
let cases = vec![
SignalMessage::Ping { timestamp_ms: 12345 },
SignalMessage::Hold,
SignalMessage::Hangup { reason: HangupReason::Normal, call_id: None },
SignalMessage::CallRinging { call_id: "abcd".into() },
];
for m in cases {
let json = serde_json::to_string(&m).unwrap();
let decoded: SignalMessage = serde_json::from_str(&json).unwrap();
// Discriminant equality proves variant tag survived.
assert_eq!(
std::mem::discriminant(&m),
std::mem::discriminant(&decoded)
);
}
}
#[test]
fn hold_unhold_serialize() {
let hold = SignalMessage::Hold;

View File

@@ -1,4 +1,5 @@
use std::collections::VecDeque;
use std::time::{Duration, Instant};
use crate::packet::QualityReport;
use crate::traits::QualityController;
@@ -24,11 +25,31 @@ 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();
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 {
@@ -38,10 +59,37 @@ impl Tier {
}
}
}
}
/// 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 +1,354 @@
//! 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>,
/// Phase 3 (hole-punching): caller's server-reflexive address
/// as carried in the `DirectCallOffer`. The relay stashes it
/// here when the offer arrives so it can later inject it as
/// `peer_direct_addr` into the callee's `CallSetup`.
pub caller_reflexive_addr: Option<String>,
/// Phase 3 (hole-punching): callee's server-reflexive address
/// as carried in the `DirectCallAnswer`. Only populated for
/// `AcceptTrusted` answers — privacy-mode answers leave this
/// `None`. Fed into the caller's `CallSetup.peer_direct_addr`.
pub callee_reflexive_addr: Option<String>,
/// Phase 4 (cross-relay): federation TLS fingerprint of the
/// PEER RELAY that forwarded the offer/answer for this call.
/// `None` for local calls — caller and callee both
/// registered on this relay. `Some(fp)` when one side of
/// the call is on a remote relay reached through the
/// federation link identified by `fp`. The
/// `DirectCallAnswer` handling uses this to route the reply
/// back through the SAME link instead of broadcasting again.
pub peer_relay_fp: Option<String>,
/// Phase 5.5 (ICE host candidates): caller's LAN-local
/// interface addresses from the `DirectCallOffer`. Cross-
/// wired into the callee's `CallSetup.peer_local_addrs` so
/// the callee can direct-dial the caller over the same LAN
/// without going through the WAN reflex addr (NAT
/// hairpinning often doesn't work for same-LAN peers).
pub caller_local_addrs: Vec<String>,
/// Phase 5.5 (ICE host candidates): callee's LAN-local
/// interface addresses from the `DirectCallAnswer`. Cross-
/// wired into the caller's `CallSetup.peer_local_addrs`.
pub callee_local_addrs: Vec<String>,
}
/// 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,
caller_reflexive_addr: None,
callee_reflexive_addr: None,
peer_relay_fp: None,
caller_local_addrs: Vec::new(),
callee_local_addrs: Vec::new(),
};
self.calls.insert(call_id.clone(), call);
self.calls.get(&call_id).unwrap()
}
/// Phase 5.5: stash the caller's LAN host candidates from
/// the `DirectCallOffer`. Empty Vec is a valid value meaning
/// "caller has no LAN candidates" (e.g. old client).
pub fn set_caller_local_addrs(&mut self, call_id: &str, addrs: Vec<String>) {
if let Some(call) = self.calls.get_mut(call_id) {
call.caller_local_addrs = addrs;
}
}
/// Phase 5.5: stash the callee's LAN host candidates from
/// the `DirectCallAnswer`.
pub fn set_callee_local_addrs(&mut self, call_id: &str, addrs: Vec<String>) {
if let Some(call) = self.calls.get_mut(call_id) {
call.callee_local_addrs = addrs;
}
}
/// Phase 4: stash the federation TLS fingerprint of the peer
/// relay that originated (or will receive) the cross-relay
/// forward for this call. Safe to call with `None` to clear
/// a previously-set value.
pub fn set_peer_relay_fp(&mut self, call_id: &str, fp: Option<String>) {
if let Some(call) = self.calls.get_mut(call_id) {
call.peer_relay_fp = fp;
}
}
/// Phase 3: stash the caller's server-reflexive address read
/// off a `DirectCallOffer`. Safe to call on any call state;
/// a no-op if the call doesn't exist.
pub fn set_caller_reflexive_addr(&mut self, call_id: &str, addr: Option<String>) {
if let Some(call) = self.calls.get_mut(call_id) {
call.caller_reflexive_addr = addr;
}
}
/// Phase 3: stash the callee's server-reflexive address read
/// off a `DirectCallAnswer`. Safe to call on any call state;
/// a no-op if the call doesn't exist.
pub fn set_callee_reflexive_addr(&mut self, call_id: &str, addr: Option<String>) {
if let Some(call) = self.calls.get_mut(call_id) {
call.callee_reflexive_addr = addr;
}
}
/// 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"));
}
#[test]
fn call_registry_stores_reflexive_addrs() {
let mut reg = CallRegistry::new();
reg.create_call("c1".into(), "alice".into(), "bob".into());
// Default: both addrs are None.
let c = reg.get("c1").unwrap();
assert!(c.caller_reflexive_addr.is_none());
assert!(c.callee_reflexive_addr.is_none());
// Caller advertises its reflex addr via DirectCallOffer.
reg.set_caller_reflexive_addr("c1", Some("192.0.2.1:4433".into()));
assert_eq!(
reg.get("c1").unwrap().caller_reflexive_addr.as_deref(),
Some("192.0.2.1:4433")
);
// Callee responds with AcceptTrusted + its own reflex addr.
reg.set_callee_reflexive_addr("c1", Some("198.51.100.9:4433".into()));
assert_eq!(
reg.get("c1").unwrap().callee_reflexive_addr.as_deref(),
Some("198.51.100.9:4433")
);
// Both addrs are independently readable — the relay uses
// them to cross-wire peer_direct_addr in CallSetup.
let c = reg.get("c1").unwrap();
assert_eq!(
c.caller_reflexive_addr.as_deref(),
Some("192.0.2.1:4433")
);
assert_eq!(
c.callee_reflexive_addr.as_deref(),
Some("198.51.100.9:4433")
);
// Setter on an unknown call is a no-op, not a panic.
reg.set_caller_reflexive_addr("does-not-exist", Some("x".into()));
}
#[test]
fn call_registry_stores_peer_relay_fp() {
let mut reg = CallRegistry::new();
reg.create_call("c1".into(), "alice".into(), "bob".into());
// Default: no peer relay.
assert!(reg.get("c1").unwrap().peer_relay_fp.is_none());
// Cross-relay call: origin relay's fp is stashed.
reg.set_peer_relay_fp("c1", Some("relay-a-tls-fp".into()));
assert_eq!(
reg.get("c1").unwrap().peer_relay_fp.as_deref(),
Some("relay-a-tls-fp")
);
// Clearing with None is a valid no-op and empties the field.
reg.set_peer_relay_fp("c1", None);
assert!(reg.get("c1").unwrap().peer_relay_fp.is_none());
// Unknown call is a no-op, not a panic.
reg.set_peer_relay_fp("does-not-exist", Some("x".into()));
}
#[test]
fn call_registry_clearing_reflex_addr_works() {
// Passing None to the setter must clear a previously-set value
// so callers that downgrade to privacy mode mid-flow don't
// leak a stale addr into CallSetup.
let mut reg = CallRegistry::new();
reg.create_call("c1".into(), "alice".into(), "bob".into());
reg.set_caller_reflexive_addr("c1", Some("192.0.2.1:4433".into()));
reg.set_caller_reflexive_addr("c1", None);
assert!(reg.get("c1").unwrap().caller_reflexive_addr.is_none());
}
}

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

@@ -0,0 +1,200 @@
//! 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 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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@@ -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 +78,30 @@ 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)
///
/// The `_supported` list is currently ignored — we hardcode GOOD (24k) until
/// studio tiers (32k/48k/64k) have been validated across federation (large
/// packets may exceed path MTU and fragment in unpleasant ways). Once that's
/// tested, the body should pick the highest supported profile ≤ the relay's
/// configured ceiling.
fn choose_profile(_supported: &[QualityProfile]) -> QualityProfile {
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;

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