feat: Android Oboe voice backend — WebRTC APM, VAD, HW/SW toggle, BBCode welcome, link trust, foreground task

Audio engine (Rust):
- Android Oboe: WebRTC APM (AEC/NS/AGC/HPF) + TEN/Silero ONNX VAD
- Hardware effects (JNI) with software fallback per-effect
- Render reference buffer for AEC between output/capture callbacks
- Voice activity gate: suppress transmission when speaker muted (all platforms)
- Audio focus (SDD-109) + Bluetooth SCO (SDD-110) via JNI
- ONNX Runtime 1.26 via ort 2.0.0-rc.12 (down from rc.10, ndarray 0.17)
- VAD worker channel capacity 8→32, initial seq u64::MAX (warm-up fix)
- TEN VAD default backend (was Silero)
- Platform→WebrtcApm resolution after hardware binding
- oboe-rs edisonjwa fork with get_raw_session_id()

Android Kotlin:
- AndroidAudioFocusController + AndroidBluetoothScoController
- AndroidAudioLifecycleController (route changes to Flutter)
- ProGuard rules for new controllers

Flutter UI:
- VoiceSettings: Android HW/SW toggle (Platform auto / WebRTC APM)
- VoiceStatusChip: mute warning border + Speaker muted label
- BBCode welcome message parser (BbCodeText, case-insensitive)
- Welcome message foldable (expanded by default)
- Link trust dialog (domain wildcards, SharedPreferences)
- HapticFeedback on voice sheet opener
- Server name in AppBar, version v0.1.0
- Default channel (id=1) visible, serverquery clients hidden
- flutter_foreground_task integration

Config:
- ort load-dynamic on all non-iOS (Android/Linux/Windows)
- ONNX Runtime AAR 1.26.0
- ndarray moved to common deps (was Apple-only)
This commit is contained in:
Edison Jwa
2026-05-22 09:29:57 +09:00
parent 6af4ecab0f
commit bf284018e6
37 changed files with 3676 additions and 703 deletions
+631 -45
View File
@@ -66,72 +66,194 @@ use oboe::{
PerformanceMode, SessionId, SharingMode, Usage,
};
use crate::processor::AudioProcessor;
// `BackendEvent` / `BackendEventRx` / `BackendEventTx` moved to
// `mobile_voice_backend` so the trait can expose `take_event_rx`
// (SDD-111 item 1) cross-platform.
// --- Render-reference buffer for AEC (SDD-111 / SDD-120) ---------
//
// The output (render) callback writes the audio that will be played
// into this ring buffer. The capture callback reads the latest render
// frame and feeds it to WebRTC APM's `process_render` so AEC can
// subtract the speaker output from the microphone input.
//
// 4 slots × 10 ms × 48 kHz mono f32. One slot is always being written
// by the render callback; the capture callback reads the slot that was
// most recently completed.
const RENDER_REF_SLOTS: usize = 4;
const RENDER_REF_SAMPLES: usize = crate::frame::FRAME_10MS_SAMPLES;
struct RenderReferenceBuffer {
buf: Box<[[f32; RENDER_REF_SAMPLES]; RENDER_REF_SLOTS]>,
write_idx: std::sync::atomic::AtomicUsize,
}
impl RenderReferenceBuffer {
fn new() -> Arc<Self> {
Arc::new(Self {
buf: Box::new([[0.0_f32; RENDER_REF_SAMPLES]; RENDER_REF_SLOTS]),
write_idx: std::sync::atomic::AtomicUsize::new(0),
})
}
fn write(&self, frame: &[f32; RENDER_REF_SAMPLES]) {
let idx = self.write_idx.load(Ordering::Relaxed);
unsafe {
let slot =
&self.buf[idx] as *const [f32; RENDER_REF_SAMPLES] as *mut [f32; RENDER_REF_SAMPLES];
(*slot).copy_from_slice(frame);
}
self.write_idx
.store((idx + 1) % RENDER_REF_SLOTS, Ordering::Relaxed);
}
fn read_latest(&self) -> [f32; RENDER_REF_SAMPLES] {
let wi = self.write_idx.load(Ordering::Relaxed);
let ri = (wi + RENDER_REF_SLOTS - 1) % RENDER_REF_SLOTS;
self.buf[ri]
}
}
unsafe impl Send for RenderReferenceBuffer {}
unsafe impl Sync for RenderReferenceBuffer {}
// --- Capture state for Oboe input callback (SDD-111 / SDD-120) ----
//
// Mirrors the iOS `IosCaptureState` and the cpal-side `CaptureState`.
// Oboe delivers 48 kHz mono i16 PCM; we apply mic gain, accumulate to
// FRAME_20MS_SAMPLES, encode to Opus 32 kbps (complexity 10, inband FEC, 5 % PLC),
// and try-send the resulting packet on `voice_out_tx`.
// Enhanced with WebRTC APM (AEC/NS/AGC) and VAD (voice activity
// detection). Oboe delivers 48 kHz mono i16 PCM in variable-size
// chunks. We accumulate into 10 ms frames, then:
//
// 1. i16 → f32 conversion
// 2. Read render reference (for AEC)
// 3. WebRtcApmProcessor::process_render + process_capture
// 4. VAD → VoiceActivityStateMachine → TransmitModeSelector
// 5. f32 → i16 conversion + mic gain
// 6. Accumulate to 20 ms → Opus encode → send
struct AndroidCaptureState {
encoder: OpusEncoder,
/// Accumulator for 48 kHz mono PCM. 2x capacity to absorb
/// cpal-style buffer-size jitter without reallocating.
pcm_accum: Vec<i16>,
opus_out: [u8; crate::opus_voice::MAX_OPUS_FRAME],
voice_out_tx: mpsc::Sender<OutPacket>,
transmit_active: Arc<AtomicBool>,
output_muted: Arc<AtomicBool>,
frames_sent: Arc<AtomicU32>,
mic_gain: f32,
voice_activity_selector: Option<Arc<crate::TransmitModeSelector>>,
vad_detector: crate::vad::WebRtcFallbackVad,
silero_vad_worker: Option<crate::vad::silero_onnx::SileroOnnxVadWorker>,
ten_vad_worker: Option<crate::vad::TenOnnxVadWorker>,
current_vad_backend: crate::VadBackend,
silero_model_epoch: u64,
capture_frame_seq: u64,
vad_state: crate::voice_activity::VoiceActivityStateMachine,
webrtc_apm_processor: crate::processor::WebRtcApmProcessor,
audio_processing_config: Arc<Mutex<crate::AudioProcessingConfig>>,
audio_processing_stats: Arc<crate::SharedAudioProcessingStats>,
render_reference: Arc<RenderReferenceBuffer>,
pending_10ms: [i16; crate::frame::FRAME_10MS_SAMPLES],
pending_10ms_len: usize,
fallback_warned_backend: Option<crate::VadBackend>,
}
impl AndroidCaptureState {
fn new(
voice_out_tx: mpsc::Sender<OutPacket>,
transmit_active: Arc<AtomicBool>,
output_muted: Arc<AtomicBool>,
frames_sent: Arc<AtomicU32>,
mic_gain: f32,
voice_activity_selector: Option<Arc<crate::TransmitModeSelector>>,
audio_processing_config: Arc<Mutex<crate::AudioProcessingConfig>>,
audio_processing_stats: Arc<crate::SharedAudioProcessingStats>,
render_reference: Arc<RenderReferenceBuffer>,
) -> Result<Self, AudioError> {
let encoder = crate::opus_voice::new_voip_encoder("android")?;
// Android always uses software WebRTC APM for AEC/NS/AGC/HPF.
// The config's EffectOwner fields are resolved by open() AFTER
// hardware-effect binding; the processor is constructed here
// with all modules enabled regardless, so the resolved config
// (Platform vs WebrtcApm) only affects diagnostics, not behaviour.
let webrtc_apm_config = audio_processing_config
.lock()
.map(|cfg| {
let mut c = crate::processor::webrtc_apm::WebRtcApmConfig::from_audio_config(&cfg);
c.aec = true;
c.ns = true;
c.agc = true;
c
})
.unwrap_or(crate::processor::webrtc_apm::WebRtcApmConfig {
aec: true,
ns: true,
agc: true,
hpf: true,
..Default::default()
});
Ok(Self {
encoder,
pcm_accum: Vec::with_capacity(crate::frame::FRAME_20MS_SAMPLES * 2),
opus_out: [0u8; crate::opus_voice::MAX_OPUS_FRAME],
voice_out_tx,
transmit_active,
output_muted,
frames_sent,
mic_gain,
voice_activity_selector,
vad_detector: crate::vad::WebRtcFallbackVad::default(),
silero_vad_worker: None,
ten_vad_worker: None,
current_vad_backend: crate::VadBackend::WebrtcVad,
silero_model_epoch: crate::vad::silero_model_epoch(),
capture_frame_seq: 0,
vad_state: crate::voice_activity::VoiceActivityStateMachine::default(),
webrtc_apm_processor: crate::processor::WebRtcApmProcessor::with_config(
webrtc_apm_config,
)?,
audio_processing_config,
audio_processing_stats,
render_reference,
pending_10ms: [0_i16; crate::frame::FRAME_10MS_SAMPLES],
pending_10ms_len: 0,
fallback_warned_backend: None,
})
}
/// Consume i16 mono frames from Oboe, accumulate to FRAME_20MS_SAMPLES,
/// encode + send when PTT is held. Oboe delivers at the device's
/// native sample rate (always 48 kHz for modern Android per SRS-210),
/// so no resampling is needed.
fn ingest(&mut self, samples: &[i16]) {
/// Consume i16 mono frames from Oboe. Accumulate to 10 ms chunks,
/// process each through WebRTC APM + VAD, then encode 20 ms frames.
fn ingest_i16(&mut self, samples: &[i16]) {
let mut offset = 0;
while offset < samples.len() {
let remaining =
crate::frame::FRAME_10MS_SAMPLES - self.pending_10ms_len;
let take = remaining.min(samples.len() - offset);
self.pending_10ms[self.pending_10ms_len..self.pending_10ms_len + take]
.copy_from_slice(&samples[offset..offset + take]);
self.pending_10ms_len += take;
offset += take;
if self.pending_10ms_len == crate::frame::FRAME_10MS_SAMPLES {
let frame = self.pending_10ms;
self.process_10ms_capture_frame(&frame);
self.pending_10ms_len = 0;
}
}
if !self.transmit_active.load(Ordering::Relaxed) {
self.pcm_accum.clear();
return;
}
// Mic-gain application.
if (self.mic_gain - 1.0).abs() < f32::EPSILON {
self.pcm_accum.extend_from_slice(samples);
} else {
let gain = self.mic_gain;
self.pcm_accum.extend(samples.iter().map(|&s| {
let scaled = (s as f32) * gain;
scaled.clamp(i16::MIN as f32, i16::MAX as f32) as i16
}));
}
// Drain complete 20 ms frames.
while self.pcm_accum.len() >= crate::frame::FRAME_20MS_SAMPLES {
let mut frame = [0i16; crate::frame::FRAME_20MS_SAMPLES];
frame.copy_from_slice(&self.pcm_accum[..crate::frame::FRAME_20MS_SAMPLES]);
self.pcm_accum.drain(..crate::frame::FRAME_20MS_SAMPLES);
frame.copy_from_slice(
&self.pcm_accum[..crate::frame::FRAME_20MS_SAMPLES],
);
self.pcm_accum
.drain(..crate::frame::FRAME_20MS_SAMPLES);
match self.encoder.encode(&frame, &mut self.opus_out[..]) {
Ok(len) => {
crate::opus_voice::send_voip_frame(
@@ -154,11 +276,186 @@ impl AndroidCaptureState {
);
}
Err(e) => {
warn!(target: "chanora_audio", error = %e, "android Oboe opus encode failed");
warn!(
target: "chanora_audio",
error = %e,
"android Oboe opus encode failed"
);
}
}
}
}
fn mark_vad_fallback_active(&mut self, failed_backend: crate::VadBackend) {
if self.fallback_warned_backend != Some(failed_backend) {
self.fallback_warned_backend = Some(failed_backend);
warn!(
target: "chanora_audio",
backend = failed_backend.as_str(),
"android: VAD backend unavailable; using WebRTC fallback for runtime detection"
);
}
}
fn process_10ms_capture_frame(
&mut self,
samples: &[i16; crate::frame::FRAME_10MS_SAMPLES],
) {
let mut frame = [0.0_f32; crate::frame::FRAME_10MS_SAMPLES];
for (dst, src) in frame.iter_mut().zip(samples.iter().copied()) {
*dst = crate::frame::i16_to_f32(src);
}
let input_dbfs = crate::frame::dbfs(&frame);
let render_ref = self.render_reference.read_latest();
self.webrtc_apm_processor.process_render(&render_ref);
self.webrtc_apm_processor.process_capture(&mut frame);
let (vad_hangover, vad_backend) = self
.audio_processing_config
.try_lock()
.map(|cfg| (cfg.vad_hangover_ms, cfg.vad_backend))
.unwrap_or((
crate::voice_activity::VAD_HANGOVER_MS,
crate::VadBackend::WebrtcVad,
));
self.vad_state.configure(
crate::voice_activity::VAD_OPEN_AFTER_MS,
vad_hangover,
crate::voice_activity::VAD_MIN_TX_MS,
);
// VAD backend switching (mirrors iOS Raw path).
let silero_epoch = crate::vad::silero_model_epoch();
let silero_changed = vad_backend == crate::VadBackend::SileroOnnx
&& silero_epoch != self.silero_model_epoch;
if vad_backend != self.current_vad_backend || silero_changed {
self.current_vad_backend = vad_backend;
self.silero_model_epoch = silero_epoch;
self.fallback_warned_backend = None;
match vad_backend {
crate::VadBackend::SileroOnnx => {
let path = crate::vad::silero_model_bundle_path();
self.silero_vad_worker =
crate::vad::silero_onnx::SileroOnnxVadWorker::try_new(&path);
self.ten_vad_worker = None;
if self.silero_vad_worker.is_none() {
warn!(
target: "chanora_audio",
"android: Silero VAD model not found at {path}; falling back to WebRTC VAD"
);
}
}
crate::VadBackend::TenVad => {
let path = crate::vad::ten_model_bundle_path();
self.ten_vad_worker = crate::vad::TenOnnxVadWorker::try_new(&path);
self.silero_vad_worker = None;
if self.ten_vad_worker.is_none() {
warn!(
target: "chanora_audio",
"android: TEN VAD ONNX model not found at {path}; falling back to WebRTC VAD"
);
}
}
_ => {
self.silero_vad_worker = None;
self.ten_vad_worker = None;
}
}
self.vad_state.reset();
}
self.capture_frame_seq = self.capture_frame_seq.wrapping_add(1);
let capture_seq = self.capture_frame_seq;
let mut used_fallback_vad = false;
let vad = if vad_backend == crate::VadBackend::Disabled {
crate::vad::VadOutput {
probability: 1.0,
speech: true,
}
} else if vad_backend == crate::VadBackend::SileroOnnx {
if let Some(worker) = self.silero_vad_worker.as_ref() {
let enqueued = worker.try_send(capture_seq, &frame);
if enqueued && !worker.is_stale(capture_seq) {
let p = worker.latest_probability();
crate::vad::VadOutput {
probability: p,
speech: p >= 0.5,
}
} else {
used_fallback_vad = true;
self.mark_vad_fallback_active(vad_backend);
crate::vad::VoiceActivityDetector::process_10ms(
&mut self.vad_detector,
&frame,
)
}
} else {
used_fallback_vad = true;
self.mark_vad_fallback_active(vad_backend);
crate::vad::VoiceActivityDetector::process_10ms(
&mut self.vad_detector,
&frame,
)
}
} else if vad_backend == crate::VadBackend::TenVad {
if let Some(worker) = self.ten_vad_worker.as_ref() {
let enqueued = worker.try_send(capture_seq, &frame);
if enqueued && !worker.is_stale(capture_seq) {
let p = worker.latest_probability();
crate::vad::VadOutput {
probability: p,
speech: p >= 0.5,
}
} else {
used_fallback_vad = true;
self.mark_vad_fallback_active(vad_backend);
crate::vad::VoiceActivityDetector::process_10ms(
&mut self.vad_detector,
&frame,
)
}
} else {
used_fallback_vad = true;
self.mark_vad_fallback_active(vad_backend);
crate::vad::VoiceActivityDetector::process_10ms(
&mut self.vad_detector,
&frame,
)
}
} else {
crate::vad::VoiceActivityDetector::process_10ms(&mut self.vad_detector, &frame)
};
self.audio_processing_stats
.set_vad_fallback_active(used_fallback_vad);
let active = self.vad_state.update(vad.speech);
let output_muted = self.output_muted.load(Ordering::Relaxed);
if let Some(sel) = &self.voice_activity_selector {
sel.set_voice_activity_open(active && !output_muted);
}
self.audio_processing_stats.update_capture(
input_dbfs,
crate::frame::dbfs(&frame),
vad.probability,
active && !output_muted,
self.transmit_active.load(Ordering::Relaxed),
);
if !self.transmit_active.load(Ordering::Relaxed) || output_muted {
return;
}
let gain = self.mic_gain;
if (gain - 1.0).abs() < f32::EPSILON {
self.pcm_accum
.extend(frame.iter().copied().map(crate::frame::f32_to_i16));
} else {
self.pcm_accum.extend(frame.iter().copied().map(|s| {
let scaled = (crate::frame::f32_to_i16(s) as f32) * gain;
scaled.clamp(i16::MIN as f32, i16::MAX as f32) as i16
}));
}
}
}
struct InputCallback {
@@ -176,7 +473,7 @@ impl AudioInputCallback for InputCallback {
) -> DataCallbackResult {
let _ = catch_unwind(AssertUnwindSafe(|| {
if let Ok(mut s) = self.state.lock() {
s.ingest(frames);
s.ingest_i16(frames);
}
}));
DataCallbackResult::Continue
@@ -203,6 +500,7 @@ struct OutputCallback {
output_muted: Arc<AtomicBool>,
event_tx: BackendEventTx,
scratch: Arc<Mutex<Vec<f32>>>,
render_reference: Arc<RenderReferenceBuffer>,
}
impl AudioOutputCallback for OutputCallback {
@@ -223,16 +521,17 @@ impl AudioOutputCallback for OutputCallback {
*s = 0.0;
}
}
// Non-blocking pull from AudioHandler (same pattern as iOS VPIO).
match self.handler.try_lock() {
Ok(mut h) => {
let _ = h.fill_buffer(&mut scratch[..needed]);
}
Err(std::sync::TryLockError::WouldBlock) => {
// scratch already zeroed above.
}
Err(std::sync::TryLockError::WouldBlock) => {}
Err(std::sync::TryLockError::Poisoned(e)) => {
warn!(target: "chanora_audio", "AudioHandler mutex poisoned: {}", e);
warn!(
target: "chanora_audio",
"AudioHandler mutex poisoned: {}",
e
);
}
}
let gain = f32::from_bits(self.output_gain.load(Ordering::Relaxed));
@@ -243,6 +542,19 @@ impl AudioOutputCallback for OutputCallback {
gain,
muted,
);
// Write the first 10 ms of render audio into the reference
// buffer for the capture-side AEC.
let mono_n = needed / 2;
let render_n = mono_n.min(crate::frame::FRAME_10MS_SAMPLES);
let mut ref_frame = [0.0_f32; crate::frame::FRAME_10MS_SAMPLES];
for (i, chunk) in scratch[..render_n * 2].chunks_exact(2).enumerate() {
if i >= render_n {
break;
}
ref_frame[i] = (chunk[0] + chunk[1]) * 0.5;
}
self.render_reference.write(&ref_frame);
}));
DataCallbackResult::Continue
}
@@ -273,7 +585,7 @@ pub struct VoiceAudioParams {
pub frames_sent: Arc<AtomicU32>,
/// Pre-encode amplitude scale (1.0 = unity).
pub mic_gain: f32,
/// AudioHandler that inbound解码+混合 feeds into; the Oboe output
/// AudioHandler that inbound decode+mix feeds into; the Oboe output
/// callback pulls mixed stereo f32 from it.
pub handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
/// Master output gain (f32 bits stored in AtomicU32 for lock-free
@@ -281,6 +593,14 @@ pub struct VoiceAudioParams {
pub output_gain: Arc<AtomicU32>,
/// True = output silence regardless of incoming voice frames.
pub output_muted: Arc<AtomicBool>,
/// Optional TransmitModeSelector for VoiceActivity transmit mode.
/// The capture callback calls set_voice_activity_open on this when
/// VAD detects speech. None means VoiceActivity mode is disabled.
pub voice_activity_selector: Option<Arc<crate::TransmitModeSelector>>,
/// Shared audio-processing config (WebRTC APM flags, VAD backend).
pub audio_processing_config: Arc<Mutex<crate::AudioProcessingConfig>>,
/// Shared audio-processing statistics for diagnostics.
pub audio_processing_stats: Arc<crate::SharedAudioProcessingStats>,
}
// --- The backend itself ------------------------------------------
@@ -334,22 +654,30 @@ impl AndroidVoiceUnit {
) -> Result<Self, BackendError> {
let (event_tx, event_rx) = mpsc::unbounded_channel();
// SDD-120: build the capture state that the Oboe input callback
// will own via Arc<Mutex>. Same Opus VoIP tuning as iOS and
// desktop (32 kbps, complexity 10, inband FEC, 5 % PLC).
// Shared render-reference buffer (for AEC). The output
// callback writes; the capture callback reads.
let render_ref_buf = RenderReferenceBuffer::new();
let render_ref_for_capture = render_ref_buf.clone();
// Clone the APM config Arc before params is partially moved
// into the capture state constructor below.
let apm_config_clone = params.audio_processing_config.clone();
let capture_state = Arc::new(Mutex::new(
AndroidCaptureState::new(
params.voice_out_tx,
params.transmit_active,
params.output_muted.clone(),
params.frames_sent,
params.mic_gain,
params.voice_activity_selector,
params.audio_processing_config,
params.audio_processing_stats,
render_ref_for_capture,
)
.map_err(|e| BackendError::OpenFailed(format!("capture state init: {e}")))?,
));
// Scratch buffer for the output callback (realtime-safe
// pre-allocation). 8192 floats covers the largest practical
// burst size at 48 kHz with headroom.
let scratch = Arc::new(Mutex::new(Vec::with_capacity(8192)));
// --- Open input stream (SDD-112) ---------------------------
@@ -423,13 +751,19 @@ impl AndroidVoiceUnit {
.as_mut()
.map(|s| s.get_frames_per_burst())
.unwrap_or(0);
let session_id = None;
warn!(
target: "chanora_audio",
"android: oboe-rs get_session_id is skipped because oboe 0.6.1 \
panics on some Android allocated-session values; hardware \
effects are disabled for this stream"
);
// The oboe-rs fork (edisonjwa/oboe-rs 0.6.2) fixes the
// get_session_id() panic with unwrap_or_default(), but the
// SessionId enum still only models builder parameters (None =
// -1, Allocate = 0). The actual system audio session ID (>0)
// written by AAudio to mSessionId after stream open cannot be
// expressed in the current enum; get_raw_session_id() is a
// follow-up addition to the fork.
//
// For now: hardware effects require the system session ID.
// WebRTC APM software processing handles AEC/NS/AGC/HPF.
let session_id: Option<i32> = input_stream
.as_ref()
.and_then(|s| s.get_raw_session_id());
// --- Open output stream (SDD-112) --------------------------
let output_builder = AudioStreamBuilder::default()
@@ -450,12 +784,14 @@ impl AndroidVoiceUnit {
.set_usage(Usage::VoiceCommunication)
.set_content_type(oboe::ContentType::Speech);
let render_ref_for_output = render_ref_buf.clone();
let output_cb = OutputCallback {
handler: params.handler.clone(),
output_gain: params.output_gain.clone(),
output_muted: params.output_muted.clone(),
event_tx: event_tx.clone(),
scratch: scratch.clone(),
render_reference: render_ref_for_output,
};
let output_builder = output_builder.set_callback(output_cb);
@@ -474,6 +810,7 @@ impl AndroidVoiceUnit {
params.output_gain.clone(),
params.output_muted.clone(),
scratch.clone(),
render_ref_buf,
)?
}
};
@@ -511,6 +848,52 @@ impl AndroidVoiceUnit {
HardwareEffectHandles::default()
};
// --- SDD-113 config resolution: hardware-available? -------
// Android gives the user a choice between hardware (JNI) and
// software (WebRTC APM) effects. The AudioProcessingConfig's
// EffectOwner fields encode that choice:
// Platform → prefer hardware; software fallback if missing
// WebrtcApm → always software WebRTC APM
// Off → disable effect entirely
//
// Here we resolve Platform → WebrtcApm for each effect whose
// hardware binding failed (or wasn't attempted). This is read
// by the capture callback's WebRtcApmProcessor.
{
use crate::audio_processing::EffectOwner;
let mut apm_cfg = apm_config_clone.lock().unwrap();
let hw_aec = hw_effects.aec.is_some();
let hw_ns = hw_effects.ns.is_some();
let hw_agc = hw_effects.agc.is_some();
if apm_cfg.aec == EffectOwner::Platform && !hw_aec {
apm_cfg.aec = EffectOwner::WebrtcApm;
}
if apm_cfg.ns == EffectOwner::Platform && !hw_ns {
apm_cfg.ns = EffectOwner::WebrtcApm;
}
if apm_cfg.agc == EffectOwner::Platform && !hw_agc {
apm_cfg.agc = EffectOwner::WebrtcApm;
}
if apm_cfg.processing_backend
== crate::audio_processing::AudioBackend::PlatformVoiceProcessing
&& (!hw_aec || !hw_ns || !hw_agc)
{
apm_cfg.processing_backend = crate::audio_processing::AudioBackend::WebrtcApm;
}
info!(
target: "chanora_audio",
aec = ?apm_cfg.aec,
ns = ?apm_cfg.ns,
agc = ?apm_cfg.agc,
hpf = apm_cfg.hpf_enabled,
hw_aec,
hw_ns,
hw_agc,
session_id,
"android: audio processing config resolved (hardware effects: aec={hw_aec} ns={hw_ns} agc={hw_agc})"
);
}
// --- SDD-112 item 10 / SDD-113 item 7 / SDD-116 item 3 ---
// Publish the diagnostics snapshot. Per-effect engagement is
// derived from (a) the JNI handle (`Hardware`) or (b) the
@@ -647,6 +1030,7 @@ impl AndroidVoiceUnit {
output_gain: Arc<AtomicU32>,
output_muted: Arc<AtomicBool>,
scratch: Arc<Mutex<Vec<f32>>>,
render_reference: Arc<RenderReferenceBuffer>,
) -> Result<AudioStreamAsync<OboeOutput, OutputCallback>, BackendError> {
let cb = OutputCallback {
handler,
@@ -654,6 +1038,7 @@ impl AndroidVoiceUnit {
output_muted,
event_tx: event_tx.clone(),
scratch,
render_reference,
};
let builder = AudioStreamBuilder::default()
.set_direction::<OboeOutput>()
@@ -806,6 +1191,7 @@ fn perf_from_oboe(p: PerformanceMode) -> AchievedPerformanceMode {
match p {
PerformanceMode::LowLatency => AchievedPerformanceMode::LowLatency,
PerformanceMode::PowerSaving => AchievedPerformanceMode::PowerSaving,
PerformanceMode::PowerSavingOffloaded => AchievedPerformanceMode::PowerSaving,
PerformanceMode::None => AchievedPerformanceMode::None,
}
}
@@ -1041,6 +1427,51 @@ fn release_hardware_effects_inner(handles: &mut HardwareEffectHandles) {
}
}
// --- Process-global BackendEvent sender for JNI callbacks --------
//
// Kotlin-side listeners (audio focus, Bluetooth SCO, device route
// changes) need to publish events into the Rust engine's event
// channel. Since the engine's `BackendEventTx` is created at voice
// start, we store it here as a process-global so the JNI callbacks
// can reach it without holding a direct Rust reference.
//
// Cleared on voice stop; the Kotlin listeners are idempotent when
// no sender is registered (they log and continue).
static GLOBAL_BACKEND_EVENT_TX: std::sync::OnceLock<
std::sync::Mutex<Option<tokio::sync::mpsc::UnboundedSender<BackendEvent>>>,
> = std::sync::OnceLock::new();
fn global_event_tx_slot(
) -> &'static std::sync::Mutex<Option<tokio::sync::mpsc::UnboundedSender<BackendEvent>>> {
GLOBAL_BACKEND_EVENT_TX.get_or_init(|| std::sync::Mutex::new(None))
}
pub(crate) fn register_global_event_sender(tx: BackendEventTx) {
if let Ok(mut g) = global_event_tx_slot().lock() {
*g = Some(tx);
}
}
pub(crate) fn clear_global_event_sender() {
if let Ok(mut g) = global_event_tx_slot().lock() {
*g = None;
}
}
fn try_send_backend_event(event: BackendEvent) {
if let Ok(g) = global_event_tx_slot().lock() {
if let Some(tx) = g.as_ref() {
if tx.send(event).is_err() {
warn!(
target: "chanora_audio",
"android: global BackendEvent channel closed; event dropped"
);
}
}
}
}
// --- SDD-115 foreground-service JNI helpers ----------------------
//
// The Kotlin class `AndroidVoiceForegroundService` (Wave 2B-2)
@@ -1174,3 +1605,158 @@ fn load_app_class<'local>(
}
}
}
// --- SDD-109 / SDD-110 JNI callbacks: focus + SCO events ----------
//
// Kotlin-side OnAudioFocusChangeListener and BroadcastReceiver for
// ACTION_SCO_AUDIO_STATE_UPDATED call these Rust entry points via
// JNI. Each function marshals the platform event into a BackendEvent
// and posts it through the global event sender registered by the
// engine at voice start.
//
// SDD-115 callback safety: every entry point is wrapped in
// catch_unwind so a panic in the Rust engine can never unwind
// into the JVM.
/// SDD-109: audio focus change published by Kotlin's
/// `AndroidAudioFocusController`. `state` is the `focusChange`
/// value from `OnAudioFocusChangeListener`.
///
/// Symbol naming: JNI function declared in
/// `app.chanora.chanora_flutter.AndroidAudioFocusController`.
#[no_mangle]
pub extern "system" fn Java_app_chanora_chanora_1flutter_AndroidAudioFocusController_publishFocusChange<
'local,
>(
_env: jni::JNIEnv<'local>,
_class: jni::objects::JClass<'local>,
state: jni::sys::jint,
) {
let _ = catch_unwind(AssertUnwindSafe(|| {
// AUDIOFOCUS_LOSS = -1, LOSS_TRANSIENT = -2, LOSS_TRANSIENT_CAN_DUCK = -3,
// GAIN = 1 (AudioManager.AUDIOFOCUS_REQUEST_GRANTED is also 1, but we only
// call this from the listener callback so the values are well-known).
let event = match state {
-1 => BackendEvent::FocusLost,
-2 => BackendEvent::FocusTransient,
-3 => BackendEvent::FocusTransientCanDuck,
1 | 2 | 3 | 4 => BackendEvent::FocusGain,
_ => {
warn!(
target: "chanora_audio",
state,
"android: unknown audio focus change value; treating as FocusLost"
);
BackendEvent::FocusLost
}
};
try_send_backend_event(event);
}));
}
/// SDD-110: Bluetooth SCO state change published by Kotlin's
/// `AndroidBluetoothScoController`. `state` is the `STATE`
/// value from `ACTION_SCO_AUDIO_STATE_UPDATED`:
/// - `ACTION_SCO_AUDIO_STATE_UPDATED` is always fired with `EXTRA_SCO_AUDIO_STATE`
/// - `SCO_STATE_CONNECTING = 0`, `SCO_STATE_CONNECTED = 1`, `SCO_STATE_DISCONNECTED = 2`
///
/// Symbol naming: JNI function declared in
/// `app.chanora.chanora_flutter.AndroidBluetoothScoController`.
#[no_mangle]
pub extern "system" fn Java_app_chanora_chanora_1flutter_AndroidBluetoothScoController_publishScoStateChange<
'local,
>(
_env: jni::JNIEnv<'local>,
_class: jni::objects::JClass<'local>,
state: jni::sys::jint,
) {
let _ = catch_unwind(AssertUnwindSafe(|| {
try_send_backend_event(BackendEvent::BluetoothScoStateChanged(state));
}));
}
/// SDD-115 integration: start the Android audio focus listener.
/// Called by the engine after the voice unit is started. Uses JNI
/// to invoke `AndroidAudioFocusController.start(Context)`.
pub fn chanora_android_request_audio_focus() -> bool {
call_static_void_context(
"app/chanora/chanora_flutter/AndroidAudioFocusController",
"start",
)
}
/// SDD-115 integration: stop the Android audio focus listener.
/// Called by the engine on voice stop.
pub fn chanora_android_abandon_audio_focus() -> bool {
call_static_void_context(
"app/chanora/chanora_flutter/AndroidAudioFocusController",
"stop",
)
}
/// SDD-115 integration: start Bluetooth SCO.
/// Called by the engine after the voice unit is started.
pub fn chanora_android_start_bluetooth_sco() -> bool {
call_static_void_context(
"app/chanora/chanora_flutter/AndroidBluetoothScoController",
"start",
)
}
/// SDD-115 integration: stop Bluetooth SCO.
/// Called by the engine on voice stop.
pub fn chanora_android_stop_bluetooth_sco() -> bool {
call_static_void_context(
"app/chanora/chanora_flutter/AndroidBluetoothScoController",
"stop",
)
}
fn call_static_void_context(fqcn: &str, method: &str) -> bool {
use jni::objects::{JObject, JValue};
let ctx = ndk_context::android_context();
if ctx.vm().is_null() || ctx.context().is_null() {
warn!(
target: "chanora_audio",
class = fqcn,
method,
"android: ndk_context not initialised; call skipped"
);
return false;
}
let jvm = match unsafe { jni::JavaVM::from_raw(ctx.vm() as *mut _) } {
Ok(v) => v,
Err(e) => {
warn!(target: "chanora_audio", error = %e, class = fqcn, method, "android: JavaVM::from_raw failed");
return false;
}
};
let mut env = match jvm.attach_current_thread() {
Ok(e) => e,
Err(e) => {
warn!(target: "chanora_audio", error = %e, class = fqcn, method, "android: attach_current_thread failed");
return false;
}
};
let context_obj = unsafe { JObject::from_raw(ctx.context() as jni::sys::jobject) };
let class = match load_app_class(&mut env, &context_obj, fqcn) {
Some(c) => c,
None => return false,
};
match env.call_static_method(
&class,
method,
"(Landroid/content/Context;)V",
&[JValue::Object(&context_obj)],
) {
Ok(_) => {
info!(target: "chanora_audio", class = fqcn, method, "android: dispatched");
true
}
Err(e) => {
let _ = env.exception_clear();
warn!(target: "chanora_audio", error = %e, class = fqcn, method, "android: static call failed");
false
}
}
}