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fix(audio): eliminate Android output stutter via Oboe config + lock-free callback (#20)

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* fix(audio): eliminate Android output stutter via Oboe config + lock-free callback

Phase 1 — Oboe configuration:
- Change output stream from Usage::VoiceCommunication to Usage::Game with
  ContentType::Sonification to avoid forcing the Legacy (OpenSL ES) data
  path on most devices (Oboe issue #2075)
- Switch output format from i16 Mono to f32 Stereo, matching Qint's proven
  configuration and eliminating per-callback downmix conversion
- Set buffer size to 2x burst after stream open, reducing default buffer
  from 8-20x burst to 2x burst for lower latency
- Remove scratch Mutex<Vec<f32>>; callback writes directly to Oboe buffer

Phase 2 — Lock-free output callback:
- Add audio_event_queue.rs: lock-free SPSC bridge using crossbeam ArrayQueue
  with separate packet (lossy) and control (reliable) channels
- OutputCallback now owns AudioHandler directly (no Arc<Mutex<>> on Android)
- Inbound forwarder pushes packets via AudioEventProducer (no mutex)
- set_client_volume pushes control commands via event queue on Android
- iOS/desktop Arc<Mutex<AudioHandler>> path unchanged

* fix(audio): address PR #20 review findings

- Store AudioEventConsumer directly in OutputCallback to eliminate
  per-callback Arc clone on the real-time audio thread
- Add SAFETY comment for the unsafe from_raw_parts_mut transmute
- Bound set_client_volume spin-loop to 64 retries with warn log
- Remove redundant crossbeam-utils direct dependency
- Regenerate license inventory for new crossbeam deps (CI fix)

* fix(audio): use ASCII TODO punctuation
This commit is contained in:
Edison Jwa
2026-06-05 13:57:53 +09:00
committed by GitHub
parent fb2a8e0a80
commit 2902a8bcd5
8 changed files with 510 additions and 82 deletions
Download Patch File Download Diff File Expand all files Collapse all files
Cargo.lock
Generated
+21
View File
@@ -423,6 +423,7 @@ dependencies = [
"coreaudio-rs",
"cpal",
"criterion",
"crossbeam",
"dhat",
"dispatch2",
"futures-util",
@@ -803,6 +804,17 @@ version = "1.2.0"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "790eea4361631c5e7d22598ecd5723ff611904e3344ce8720784c93e3d83d40b"
[[package]]
name = "crossbeam"
version = "0.8.4"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "1137cd7e7fc0fb5d3c5a8678be38ec56e819125d8d7907411fe24ccb943faca8"
dependencies = [
"crossbeam-epoch",
"crossbeam-queue",
"crossbeam-utils",
]
[[package]]
name = "crossbeam-channel"
version = "0.5.15"
@@ -831,6 +843,15 @@ dependencies = [
"crossbeam-utils",
]
[[package]]
name = "crossbeam-queue"
version = "0.3.12"
source = "registry+https://github.com/rust-lang/crates.io-index"
checksum = "0f58bbc28f91df819d0aa2a2c00cd19754769c2fad90579b3592b1c9ba7a3115"
dependencies = [
"crossbeam-utils",
]
[[package]]
name = "crossbeam-utils"
version = "0.8.21"
crates/chanora_audio/Cargo.toml
+1
View File
@@ -29,6 +29,7 @@ audiopus = "0.3.0-rc.0"
tsclientlib = { git = "https://github.com/ReSpeak/tsclientlib.git", rev = "04aa2491", default-features = false, features = ["audio"] }
tokio = { version = "1", features = ["sync", "rt", "macros", "time"] }
rustfft = "6.2.0"
crossbeam = { version = "0.8", default-features = false, features = ["alloc", "crossbeam-queue"] }
[target.'cfg(all(not(target_os = "android"), not(target_os = "ios"), not(target_os = "macos")))'.dependencies]
# Desktop audio I/O for Windows capture/playback and Linux capture.
crates/chanora_audio/src/android_voice_unit.rs
+93 -53
View File
@@ -44,6 +44,7 @@ use std::sync::{Arc, Mutex};
use audiopus::coder::Encoder as OpusEncoder;
use tracing::{debug, info, warn};
use crate::audio_event_queue::{AudioCommand, AudioEventQueue};
use crate::mobile_voice_backend::{
clear_android_audio_diagnostics, latency_tier_for, next_input_preset_after,
next_sharing_mode_after, publish_android_audio_diagnostics, AchievedInputPreset,
@@ -63,7 +64,7 @@ use oboe::{
AudioInputCallback, AudioInputStreamSafe, AudioOutputCallback, AudioOutputStreamSafe,
AudioStream, AudioStreamAsync, AudioStreamBase, AudioStreamBuilder, AudioStreamSafe,
DataCallbackResult, Input as OboeInput, InputPreset, Mono, Output as OboeOutput,
PerformanceMode, SessionId, SharingMode, Usage,
PerformanceMode, SessionId, SharingMode, Stereo, Usage,
};
use crate::processor::AudioProcessor;
@@ -518,14 +519,14 @@ impl AudioInputCallback for InputCallback {
//
// Mirrors the iOS VPIO render callback. Pulls mixed 48 kHz stereo f32
// from `AudioHandler::fill_buffer`, applies output gain + mute, and
// writes mono i16 to the Oboe output buffer.
// writes stereo f32 directly to the Oboe output buffer.
struct OutputCallback {
handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
handler: AudioHandler<SessionAudioId>,
event_consumer: crate::audio_event_queue::AudioEventConsumer,
output_gain: Arc<AtomicU32>,
output_muted: Arc<AtomicBool>,
event_tx: BackendEventTx,
scratch: Arc<Mutex<Vec<f32>>>,
render_reference: Arc<RenderReferenceBuffer>,
audio_processing_stats: Arc<crate::SharedAudioProcessingStats>,
pending_render_ref: [f32; crate::frame::FRAME_10MS_SAMPLES],
@@ -533,51 +534,65 @@ struct OutputCallback {
}
impl AudioOutputCallback for OutputCallback {
type FrameType = (i16, Mono);
type FrameType = (f32, Stereo);
fn on_audio_ready(
&mut self,
_stream: &mut dyn AudioOutputStreamSafe,
frames: &mut [i16],
frames: &mut [(f32, f32)],
) -> DataCallbackResult {
let _ = catch_unwind(AssertUnwindSafe(|| {
let needed = frames.len() * 2; // stereo
let scratch = &mut self.scratch.lock().unwrap();
if scratch.len() < needed {
scratch.resize(needed, 0.0);
} else {
for s in &mut scratch[..needed] {
// SAFETY: `frames: &mut [(f32, f32)]` is an interleaved stereo
// buffer. `(f32, f32)` has the same size (8 bytes) and alignment
// (4 bytes) as `[f32; 2]`, so reinterpreting the slice as a flat
// `&mut [f32]` of length `frames.len() * 2` is sound. The Rust
// reference does not *guarantee* `#[repr(Rust)]` tuple layout,
// but (a) both fields are identical F32 primitives with no
// padding possible, and (b) the oboe crate uses
// `#[repr(transparent)]` on its frame type alias so the ABI
// contract is upheld at the FFI boundary. `frames` is not
// accessed again after `buf` is created, so no aliasing UB.
let buf: &mut [f32] = unsafe {
std::slice::from_raw_parts_mut(frames.as_mut_ptr() as *mut f32, frames.len() * 2)
};
for s in buf.iter_mut() {
*s = 0.0;
}
}
match self.handler.try_lock() {
Ok(mut h) => {
let _ = h.fill_buffer(&mut scratch[..needed]);
}
Err(std::sync::TryLockError::WouldBlock) => {}
Err(std::sync::TryLockError::Poisoned(e)) => {
warn!(
target: "chanora_audio",
"AudioHandler mutex poisoned: {}",
e
);
for cmd in self.event_consumer.drain_controls() {
match cmd {
AudioCommand::SetVolume(id, vol) => {
if let Some(q) = self.handler.get_mut_queues().get_mut(&id) {
q.volume = vol;
}
}
AudioCommand::RemoveClient(id) => {
self.handler.get_mut_queues().remove(&id);
}
}
}
for pkt in self.event_consumer.drain_packets(50) {
if let Err(e) = self.handler.handle_packet(pkt.client_id, pkt.data) {
debug!(target: "chanora_audio", error = %e, "decode failed");
}
}
let _ = self.handler.fill_buffer(buf);
let gain = f32::from_bits(self.output_gain.load(Ordering::Relaxed));
let muted = self.output_muted.load(Ordering::Relaxed);
let _ = crate::voice_render::downmix_stereo_f32_to_mono_i16(
&scratch[..needed],
frames,
gain,
muted,
);
if muted {
for s in buf.iter_mut() {
*s = 0.0;
}
} else if gain != 1.0 {
for s in buf.iter_mut() {
*s *= gain;
}
}
self.audio_processing_stats
.update_render(crate::frame::dbfs(&scratch[..needed]), frames.len() as u32);
.update_render(crate::frame::dbfs(buf), frames.len() as u32);
// Accumulate the full render callback into 10 ms mono chunks so
// AEC sees consistent reference timing even when output callbacks
// are shorter or longer than 10 ms.
for chunk in scratch[..needed].chunks_exact(2) {
for chunk in buf.chunks_exact(2) {
self.pending_render_ref[self.pending_render_ref_len] = (chunk[0] + chunk[1]) * 0.5;
self.pending_render_ref_len += 1;
if self.pending_render_ref_len == crate::frame::FRAME_10MS_SAMPLES {
@@ -677,8 +692,6 @@ impl AndroidVoiceUnit {
.map_err(|e| BackendError::OpenFailed(format!("capture state init: {e}")))?,
));
let scratch = Arc::new(Mutex::new(Vec::with_capacity(8192)));
// --- Open input stream (SDD-112) ---------------------------
let input_builder = AudioStreamBuilder::default()
.set_direction::<OboeInput>()
@@ -766,8 +779,8 @@ impl AndroidVoiceUnit {
let output_builder = AudioStreamBuilder::default()
.set_direction::<OboeOutput>()
.set_sample_rate(cfg.sample_rate as i32)
.set_channel_count::<Mono>()
.set_format::<i16>()
.set_channel_count::<Stereo>()
.set_format::<f32>()
.set_performance_mode(if cfg.request_low_latency {
PerformanceMode::LowLatency
} else {
@@ -778,16 +791,21 @@ impl AndroidVoiceUnit {
} else {
SharingMode::Shared
})
.set_usage(Usage::VoiceCommunication)
.set_content_type(oboe::ContentType::Speech);
// Usage::Game avoids forcing the Legacy (OpenSL ES) data path
// that Usage::VoiceCommunication triggers on most devices.
// Android audio routing is already handled by
// AudioManager.MODE_IN_COMMUNICATION on the Flutter side.
.set_usage(Usage::Game)
.set_content_type(oboe::ContentType::Sonification);
let render_ref_for_output = render_ref_buf.clone();
let event_queue = params.event_producer.queue();
let output_cb = OutputCallback {
handler: params.handler.clone(),
handler: params.handler,
event_consumer: AudioEventQueue::consumer(&event_queue),
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,
audio_processing_stats: audio_processing_stats.clone(),
pending_render_ref: [0.0_f32; crate::frame::FRAME_10MS_SAMPLES],
@@ -806,20 +824,41 @@ impl AndroidVoiceUnit {
Self::open_output_fallback(
cfg,
&event_tx,
params.handler.clone(),
AudioHandler::new(),
AudioEventQueue::consumer(&event_queue),
params.output_gain.clone(),
params.output_muted.clone(),
audio_processing_stats.clone(),
scratch.clone(),
render_ref_buf,
)?
}
};
let output_frames_per_burst = output_stream.get_frames_per_burst();
if output_frames_per_burst > 0 {
let desired = output_frames_per_burst * 2;
match output_stream.set_buffer_size_in_frames(desired) {
Ok(actual) => {
debug!(
target: "chanora_audio",
desired,
actual,
"android: output buffer size tuned"
);
}
Err(e) => {
warn!(
target: "chanora_audio",
error = ?e,
"android: output buffer size tuning failed; using device default"
);
}
}
}
let output_perf = perf_from_oboe(output_stream.get_performance_mode());
let output_share = share_from_oboe(output_stream.get_sharing_mode());
let output_sample_rate = output_stream.get_sample_rate();
let output_frames_per_burst = output_stream.get_frames_per_burst();
// SDD-112 / SRS-210: structured "stream opened" event with
// achieved values. No PII; only platform-reported scalars.
@@ -1038,19 +1077,19 @@ impl AndroidVoiceUnit {
fn open_output_fallback(
cfg: &AndroidVoiceStreamConfig,
event_tx: &BackendEventTx,
handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
handler: AudioHandler<SessionAudioId>,
event_consumer: crate::audio_event_queue::AudioEventConsumer,
output_gain: Arc<AtomicU32>,
output_muted: Arc<AtomicBool>,
audio_processing_stats: Arc<crate::SharedAudioProcessingStats>,
scratch: Arc<Mutex<Vec<f32>>>,
render_reference: Arc<RenderReferenceBuffer>,
) -> Result<AudioStreamAsync<OboeOutput, OutputCallback>, BackendError> {
let cb = OutputCallback {
handler,
event_consumer,
output_gain,
output_muted,
event_tx: event_tx.clone(),
scratch,
render_reference,
audio_processing_stats,
pending_render_ref: [0.0_f32; crate::frame::FRAME_10MS_SAMPLES],
@@ -1059,12 +1098,13 @@ impl AndroidVoiceUnit {
let builder = AudioStreamBuilder::default()
.set_direction::<OboeOutput>()
.set_sample_rate(cfg.sample_rate as i32)
.set_channel_count::<Mono>()
.set_format::<i16>()
.set_channel_count::<Stereo>()
.set_format::<f32>()
.set_performance_mode(PerformanceMode::LowLatency)
.set_sharing_mode(SharingMode::Shared)
.set_usage(Usage::VoiceCommunication)
.set_content_type(oboe::ContentType::Speech)
// Same Usage::Game rationale as primary output builder above.
.set_usage(Usage::Game)
.set_content_type(oboe::ContentType::Sonification)
.set_callback(cb);
builder
.open_stream()
crates/chanora_audio/src/audio_event_queue.rs
+120
View File
@@ -0,0 +1,120 @@
use std::sync::atomic::{AtomicU64, Ordering};
use std::sync::Arc;
use chanora_protocol::InAudioBuf;
use crossbeam::queue::ArrayQueue;
use crate::engine::SessionAudioId;
const PACKET_QUEUE_CAPACITY: usize = 100;
const CONTROL_QUEUE_CAPACITY: usize = 32;
/// A raw inbound voice packet waiting to be inserted into AudioHandler.
pub struct AudioPacket {
/// Client whose TeamSpeak audio packet this belongs to.
pub client_id: SessionAudioId,
/// Raw inbound TeamSpeak audio payload accepted by AudioHandler::handle_packet.
pub data: InAudioBuf,
}
/// Control commands from the main thread to the audio callback.
pub enum AudioCommand {
/// Set a client's output volume.
SetVolume(SessionAudioId, f32),
/// Remove a client's decode queue.
// TODO: Wire to client disconnect path; handled in callback but no
// producer currently pushes this command.
RemoveClient(SessionAudioId),
}
/// Lock-free bridge between the inbound forwarder / main thread and the
/// audio callback. The callback owns the consumer halves.
pub struct AudioEventQueue {
/// Bounded lossy queue for raw voice packets. On overflow, the push
/// fails and the packet is dropped (counted via `packets_dropped`).
/// Capacity: 100 packets (~2 seconds at 50pps, far more than needed).
pub packet_queue: ArrayQueue<AudioPacket>,
/// Bounded reliable queue for control commands (volume, client removal).
/// On overflow, the caller retries. Capacity: 32 commands.
pub control_queue: ArrayQueue<AudioCommand>,
/// Atomic counter for dropped packets (for diagnostics).
pub packets_dropped: AtomicU64,
}
impl AudioEventQueue {
/// Create the Android audio event bridge with fixed queue capacities.
pub fn new() -> Arc<Self> {
Arc::new(Self {
packet_queue: ArrayQueue::new(PACKET_QUEUE_CAPACITY),
control_queue: ArrayQueue::new(CONTROL_QUEUE_CAPACITY),
packets_dropped: AtomicU64::new(0),
})
}
/// Create a producer handle sharing this queue.
pub fn producer(queue: &Arc<Self>) -> AudioEventProducer {
AudioEventProducer {
queue: Arc::clone(queue),
}
}
/// Create a consumer handle sharing this queue.
pub fn consumer(queue: &Arc<Self>) -> AudioEventConsumer {
AudioEventConsumer {
queue: Arc::clone(queue),
}
}
}
/// Producer side used by the inbound forwarder and engine control methods.
#[derive(Clone)]
pub struct AudioEventProducer {
queue: Arc<AudioEventQueue>,
}
impl AudioEventProducer {
/// Push a raw voice packet, incrementing the drop counter if full.
pub fn push_packet(&self, packet: AudioPacket) -> Result<(), AudioPacket> {
self.queue.packet_queue.push(packet).map_err(|packet| {
self.queue.packets_dropped.fetch_add(1, Ordering::Relaxed);
packet
})
}
/// Push a control command, returning it unchanged if the queue is full.
pub fn push_control(&self, cmd: AudioCommand) -> Result<(), AudioCommand> {
self.queue.control_queue.push(cmd)
}
/// Shared queue backing this producer.
pub fn queue(&self) -> Arc<AudioEventQueue> {
Arc::clone(&self.queue)
}
}
/// Consumer side used by the Android output callback.
pub struct AudioEventConsumer {
queue: Arc<AudioEventQueue>,
}
impl AudioEventConsumer {
/// Pop up to `cap` queued packets.
pub fn drain_packets(&self, cap: usize) -> impl Iterator<Item = AudioPacket> + '_ {
let mut drained = 0;
std::iter::from_fn(move || {
if drained >= cap {
return None;
}
let packet = self.queue.packet_queue.pop();
if packet.is_some() {
drained += 1;
}
packet
})
}
/// Pop all currently queued controls.
pub fn drain_controls(&self) -> impl Iterator<Item = AudioCommand> + '_ {
std::iter::from_fn(move || self.queue.control_queue.pop())
}
}
crates/chanora_audio/src/engine.rs
+43 -15
View File
@@ -52,6 +52,8 @@ use tsclientlib::audio::AudioHandler;
use chanora_protocol::{InboundVoice, OutPacket};
#[cfg(target_os = "android")]
use crate::audio_event_queue::{AudioCommand, AudioEventQueue, AudioPacket};
use crate::AudioError;
#[cfg(all(
@@ -305,8 +307,11 @@ pub struct AudioEngine {
output_muted: Arc<AtomicBool>,
audio_processing_config: Arc<Mutex<crate::AudioProcessingConfig>>,
audio_processing_stats: Arc<crate::SharedAudioProcessingStats>,
#[cfg(not(target_os = "android"))]
audio_handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
#[cfg(target_os = "android")]
audio_event_producer: crate::audio_event_queue::AudioEventProducer,
#[cfg(target_os = "android")]
voice_out_tx: mpsc::Sender<OutPacket>,
#[cfg(target_os = "android")]
voice_activity_selector: Option<Arc<crate::TransmitModeSelector>>,
@@ -482,7 +487,7 @@ impl AudioEngine {
voice_out_tx: mpsc::Sender<OutPacket>,
transmit_gate: crate::ptt::AudioTransmitGate,
frames_sent: Arc<AtomicU32>,
audio_handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
event_producer: crate::audio_event_queue::AudioEventProducer,
output_gain: Arc<AtomicU32>,
output_muted: Arc<AtomicBool>,
voice_activity_selector: Option<Arc<crate::TransmitModeSelector>>,
@@ -552,7 +557,8 @@ impl AudioEngine {
transmit_active: transmit_gate.flag_arc(),
frames_sent: frames_sent.clone(),
mic_gain,
handler: audio_handler.clone(),
handler: AudioHandler::new(),
event_producer: event_producer.clone(),
output_gain: output_gain.clone(),
output_muted: output_muted.clone(),
voice_activity_selector: voice_activity_selector.clone(),
@@ -572,7 +578,7 @@ impl AudioEngine {
voice_out_tx.clone(),
transmit_gate.clone(),
frames_sent.clone(),
audio_handler.clone(),
event_producer.clone(),
output_gain.clone(),
output_muted.clone(),
voice_activity_selector.clone(),
@@ -1007,8 +1013,8 @@ impl AudioEngine {
let audio_processing_config = Arc::new(Mutex::new(crate::AudioProcessingConfig::default()));
let audio_processing_stats = Arc::new(crate::SharedAudioProcessingStats::default());
let audio_handler: Arc<Mutex<AudioHandler<SessionAudioId>>> =
Arc::new(Mutex::new(AudioHandler::new()));
let event_queue = AudioEventQueue::new();
let event_producer = AudioEventQueue::producer(&event_queue);
if !cfg.mobile_voice_preset {
return Err(AudioError::Backend(
@@ -1069,7 +1075,8 @@ impl AudioEngine {
transmit_active: transmit_flag_for_capture,
frames_sent: frames_sent.clone(),
mic_gain: cfg.mic_gain,
handler: audio_handler.clone(),
handler: AudioHandler::new(),
event_producer: event_producer.clone(),
output_gain: output_gain.clone(),
output_muted: output_muted.clone(),
voice_activity_selector: cfg.voice_activity_selector.clone(),
@@ -1103,7 +1110,7 @@ impl AudioEngine {
voice_out_tx.clone(),
transmit_gate.clone(),
frames_sent.clone(),
audio_handler.clone(),
event_producer.clone(),
output_gain.clone(),
output_muted.clone(),
cfg.voice_activity_selector.clone(),
@@ -1151,7 +1158,7 @@ impl AudioEngine {
let capture_active = true;
let (shutdown_tx, mut shutdown_rx) = tokio::sync::oneshot::channel();
let handler_for_task = audio_handler.clone();
let event_producer_for_task = event_producer.clone();
let frames_received_for_task = frames_received.clone();
tokio::spawn(async move {
loop {
@@ -1164,10 +1171,8 @@ impl AudioEngine {
match item {
Some(v) => {
let id = SessionAudioId(v.from_client);
let mut h = handler_for_task.lock().unwrap();
if let Err(e) = h.handle_packet(id, v.packet) {
debug!(target: "chanora_audio", error = %e, "decode failed");
} else {
let packet = AudioPacket { client_id: id, data: v.packet };
if event_producer_for_task.push_packet(packet).is_ok() {
frames_received_for_task.fetch_add(1, Ordering::Relaxed);
}
}
@@ -1186,7 +1191,7 @@ impl AudioEngine {
output_muted,
audio_processing_config,
audio_processing_stats,
audio_handler,
audio_event_producer: event_producer,
voice_out_tx,
voice_activity_selector: cfg.voice_activity_selector.clone(),
mic_gain: cfg.mic_gain,
@@ -1467,7 +1472,8 @@ impl AudioEngine {
transmit_active: self.transmit_gate.flag_arc(),
frames_sent: self.frames_sent.clone(),
mic_gain: self.mic_gain,
handler: self.audio_handler.clone(),
handler: AudioHandler::new(),
event_producer: self.audio_event_producer.clone(),
output_gain: self.output_gain.clone(),
output_muted: self.output_muted.clone(),
voice_activity_selector: self.voice_activity_selector.clone(),
@@ -1488,7 +1494,7 @@ impl AudioEngine {
self.voice_out_tx.clone(),
self.transmit_gate.clone(),
self.frames_sent.clone(),
self.audio_handler.clone(),
self.audio_event_producer.clone(),
self.output_gain.clone(),
self.output_muted.clone(),
self.voice_activity_selector.clone(),
@@ -1655,6 +1661,27 @@ impl AudioEngine {
/// `0.0..4.0`.
pub fn set_client_volume(&self, client_id: u64, volume: f32) {
let clamped = volume.clamp(0.0, 4.0);
#[cfg(target_os = "android")]
{
let mut cmd = AudioCommand::SetVolume(SessionAudioId(client_id), clamped);
for _ in 0..64 {
match self.audio_event_producer.push_control(cmd) {
Ok(()) => return,
Err(returned) => {
cmd = returned;
std::thread::yield_now();
}
}
}
tracing::warn!(
target: "chanora_audio",
client_id,
volume = clamped,
"set_client_volume: control queue full after 64 retries — volume not applied"
);
}
#[cfg(not(target_os = "android"))]
{
match self.audio_handler.lock() {
Ok(mut h) => {
if let Some(q) = h.get_mut_queues().get_mut(&SessionAudioId(client_id)) {
@@ -1673,6 +1700,7 @@ impl AudioEngine {
}
}
}
}
impl Drop for AudioEngine {
fn drop(&mut self) {
crates/chanora_audio/src/lib.rs
+2
View File
@@ -28,6 +28,8 @@
#![warn(missing_docs)]
#[cfg(target_os = "android")]
mod audio_event_queue;
pub mod audio_processing;
pub mod debug_wav;
mod engine;
crates/chanora_audio/src/mobile_voice_backend.rs
+8 -1
View File
@@ -67,7 +67,7 @@ pub type BackendEventTx = mpsc::UnboundedSender<BackendEvent>;
pub type AudioSessionId = i32;
/// Engine-owned state shared with mobile voice audio callbacks.
#[derive(Clone)]
#[cfg_attr(not(target_os = "android"), derive(Clone))]
pub(crate) struct VoiceAudioParams {
/// Opus-encoded voice packets sent on this channel toward the
/// protocol layer.
@@ -78,8 +78,15 @@ pub(crate) struct VoiceAudioParams {
pub frames_sent: Arc<AtomicU32>,
/// Pre-encode amplitude scale (1.0 = unity).
pub mic_gain: f32,
/// AudioHandler owned by the Android output callback.
#[cfg(target_os = "android")]
pub handler: AudioHandler<SessionAudioId>,
/// Producer used by Android engine tasks to feed the output callback.
#[cfg(target_os = "android")]
pub event_producer: crate::audio_event_queue::AudioEventProducer,
/// AudioHandler that inbound decode+mix feeds into; the output
/// callback pulls mixed stereo f32 from it.
#[cfg(not(target_os = "android"))]
pub handler: Arc<Mutex<AudioHandler<SessionAudioId>>>,
/// Master output gain (f32 bits stored in AtomicU32 for lock-free
/// cross-thread read from the realtime audio callback).
docs/security/license-inventory.md
+210 -1
View File
@@ -16,7 +16,7 @@ those terms.
| License | Crate count |
|---------|-------------|
| `Apache License 2.0` | 331 |
| `Apache License 2.0` | 333 |
| `MIT License` | 74 |
| `Unicode License v3` | 19 |
| `BSD 3-Clause &quot;New&quot; or &quot;Revised&quot; License` | 14 |
@@ -117,6 +117,7 @@ those terms.
| pin-utils | 0.1.0 | `Apache License 2.0` | <https://github.com/rust-lang-nursery/pin-utils> |
| ecdsa | 0.16.9 | `Apache License 2.0` | <https://github.com/RustCrypto/signatures/tree/master/ecdsa> |
| rfc6979 | 0.4.0 | `Apache License 2.0` | <https://github.com/RustCrypto/signatures/tree/master/rfc6979> |
| crossbeam | 0.8.4 | `Apache License 2.0` | <https://github.com/crossbeam-rs/crossbeam> |
| ppv-lite86 | 0.2.21 | `Apache License 2.0` | <https://github.com/cryptocorrosion/cryptocorrosion> |
| rustls-pki-types | 1.14.1 | `Apache License 2.0` | <https://github.com/rustls/pki-types> |
| keyring | 3.6.3 | `Apache License 2.0` | <https://github.com/hwchen/keyring-rs.git> |
@@ -144,6 +145,7 @@ those terms.
| critical-section | 1.2.0 | `Apache License 2.0` | <https://github.com/rust-embedded/critical-section> |
| crossbeam-channel | 0.5.15 | `Apache License 2.0` | <https://github.com/crossbeam-rs/crossbeam> |
| crossbeam-epoch | 0.9.18 | `Apache License 2.0` | <https://github.com/crossbeam-rs/crossbeam> |
| crossbeam-queue | 0.3.12 | `Apache License 2.0` | <https://github.com/crossbeam-rs/crossbeam> |
| crossbeam-utils | 0.8.21 | `Apache License 2.0` | <https://github.com/crossbeam-rs/crossbeam> |
| dbus-secret-service | 4.1.0 | `Apache License 2.0` | <https://github.com/brotskydotcom/dbus-secret-service.git> |
| displaydoc | 0.2.6 | `Apache License 2.0` | <https://github.com/yaahc/displaydoc> |
@@ -4993,6 +4995,213 @@ TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
END OF TERMS AND CONDITIONS
APPENDIX: How to apply the Apache License to your work.
To apply the Apache License to your work, attach the following
boilerplate notice, with the fields enclosed by brackets &quot;[]&quot;
replaced with your own identifying information. (Don&#x27;t include
the brackets!) The text should be enclosed in the appropriate
comment syntax for the file format. We also recommend that a
file or class name and description of purpose be included on the
same &quot;printed page&quot; as the copyright notice for easier
identification within third-party archives.
Copyright 2019 The Crossbeam Project Developers
Licensed under the Apache License, Version 2.0 (the &quot;License&quot;);
you may not use this file except in compliance with the License.
You may obtain a copy of the License at
http://www.apache.org/licenses/LICENSE-2.0
Unless required by applicable law or agreed to in writing, software
distributed under the License is distributed on an &quot;AS IS&quot; BASIS,
WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
See the License for the specific language governing permissions and
limitations under the License.
```
### Apache License 2.0
```
Apache License
Version 2.0, January 2004
http://www.apache.org/licenses/
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APPENDIX: How to apply the Apache License to your work.
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