Partial reads from the capture ring consumed samples that were then
discarded when the send loop retried from buf[0]. For 20ms codecs this
was invisible (single Oboe burst fills 960 samples in one read), but
40ms codecs (Opus6k, 1920 samples) needed 2 bursts — the first partial
read consumed 960 real samples and threw them away.
Result: Opus6k produced ~11 frames/s instead of 25 (~44% of expected).
Fix: expose wzp_native_audio_capture_available() and check it before
reading, matching the desktop capture_ring.available() pattern. Partial
reads no longer occur because we only read when enough samples exist.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Root cause: Oboe capture at 48kHz with InputPreset::VoiceCommunication
cannot open against a BT SCO device (only supports 8/16kHz). The stream
silently falls back to builtin mic, delivering zeros.
Fix: add bt_active flag to WzpOboeConfig. When set, capture skips
setSampleRate and setInputPreset, letting the system route to BT SCO
at its native rate. Oboe's SampleRateConversionQuality::Best resamples
to 48kHz for our ring buffers. Playout uses Usage::Media in BT mode.
New API: wzp_native_audio_start_bt() for BT mode, called from
set_bluetooth_sco(on=true). Normal audio_start() restores the
standard config when switching back to earpiece/speaker.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Build 4f2ad65 wired the Speaker button to AudioManager.setSpeakerphoneOn
but user testing found that flipping speakerphone on an active Oboe
VoiceCommunication stream silently tears down the AAudio streams on
Pixel-class devices — both capture and playout stop producing data.
Only ending the call and rejoining brings audio back (because the fresh
Oboe open runs with the new routing already applied).
Also the earpiece state showed up red in the UI because the button was
getting the `.muted` CSS class when speakerphoneOn=false. Earpiece is a
valid routing state, not a muted one.
Fix set_speakerphone Tauri command:
1. Flip AudioManager.setSpeakerphoneOn via JNI (as before).
2. If the Oboe backend is currently running, stop it, sleep 50 ms to
let AAudio finalise the transition, then start it again. The Rust
send/recv tokio tasks keep running across the gap — they just read
zero samples and write into the preserved ring buffers for a few
frames, which is acceptable. The AudioBackend singleton's ring
state is preserved across stop+start because it's in a 'static
OnceLock.
3. Debounce the UI click via speakerphoneBusy + spkBtn.disabled so
users can't queue up multiple toggles during the restart window.
Fix main.ts Speaker button:
- Remove the `.muted` classList toggle (added `.speaker-on` for CSS).
- Update label text to "🔊 Speaker" / "🔈 Earpiece" for clarity.
- On showCallScreen(), invoke is_speakerphone_on to sync the label
with the real AudioManager state, so it matches reality after a
rejoin (which was another symptom the user hit — the button label
desynced from the actual routing after ending and restarting a
call).
- Debounce click + disable button while the restart is in flight.
Drops #[allow(dead_code)] from wzp_native::audio_is_running now that it
is actually called from the set_speakerphone restart guard.
Replaces the Android-side CallEngine::start() stub with a real implementation
that mirrors the desktop start() body but routes all PCM through the
standalone wzp-native cdylib loaded at startup via libloading instead of
using CPAL.
- desktop/src-tauri/src/wzp_native.rs: new module with a static
OnceLock<libloading::Library> + cached raw fn pointers for every symbol
we need (version, hello, audio_start/stop, read_capture, write_playout,
is_running, capture/playout_latency_ms). init() resolves everything once
at startup; accessors return default values if init() never ran.
- desktop/src-tauri/src/lib.rs: drop the inline dlopen smoke test, add
`mod wzp_native;` behind target_os="android", and invoke
wzp_native::init() from the Tauri setup() callback so the library is
loaded + all symbols cached before any CallEngine can touch audio.
- desktop/src-tauri/src/engine.rs: the Android #[cfg] branch of
CallEngine::start() now does the full QUIC handshake + signal loop +
Opus send/recv tasks, calling wzp_native::audio_start() /
audio_read_capture() / audio_write_playout() instead of the desktop
CPAL rings. SyncWrapper now holds a placeholder Box<()> on Android
because the audio backend lives in a process-global singleton inside
libwzp_native.so rather than being owned per-engine.
Next step: build #39 on the remote docker builder and smoke-test on
Pixel 6 that the Connect button in the UI successfully brings up Oboe
and streams audio through the dlopen boundary.
