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fix some potential audio issues, clean up the code a bit, and suppress some warnings #737

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Merged
xianshijing-lk merged 2 commits into from
Oct 11, 2025
Merged

fix some potential audio issues, clean up the code a bit, and suppress some warnings #737

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bf8f8e0
fix some potential audio issues, clean up the code a bit, and surpres…
xianshijing-lk Oct 10, 2025
bfc10e2
use assign to initialize the buffer
xianshijing-lk Oct 10, 2025
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4 changes: 2 additions & 2 deletions livekit/src/room/utils/utf8_chunk.rs
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Original file line number Diff line number Diff line change
Expand Up @@ -59,11 +59,11 @@ impl<'a> Iterator for Utf8AwareChunks<'a> {
pub trait Utf8AwareChunkExt {
/// Splits the bytes into chunks of the specified size, ensuring that
/// UTF-8 character boundaries are respected.
fn utf8_aware_chunks(&self, chunk_size: usize) -> Utf8AwareChunks;
fn utf8_aware_chunks(&self, chunk_size: usize) -> Utf8AwareChunks<'_>;
}

impl Utf8AwareChunkExt for [u8] {
fn utf8_aware_chunks(&self, chunk_size: usize) -> Utf8AwareChunks {
fn utf8_aware_chunks(&self, chunk_size: usize) -> Utf8AwareChunks<'_> {
Utf8AwareChunks::new(self, chunk_size)
}
}
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3 changes: 2 additions & 1 deletion webrtc-sys/build.rs
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Expand Up @@ -216,7 +216,8 @@ fn main() {
.file("src/objc_video_factory.mm")
.file("src/objc_video_frame_buffer.mm")
.flag("-stdlib=libc++")
.flag("-std=c++20");
.flag("-std=c++20")
.flag("-Wno-nullability-completeness");
}
"ios" => {
println!("cargo:rustc-link-lib=framework=Foundation");
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3 changes: 2 additions & 1 deletion webrtc-sys/include/livekit/audio_track.h
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Expand Up @@ -17,6 +17,7 @@
#pragma once

#include <memory>
#include <vector>

#include "api/audio/audio_frame.h"
#include "api/audio_options.h"
Expand Down Expand Up @@ -137,7 +138,7 @@ class AudioTrackSource {
void (*on_complete_)(const SourceContext*) RTC_GUARDED_BY(mutex_);

int missed_frames_ RTC_GUARDED_BY(mutex_) = 0;
int16_t* silence_buffer_ = nullptr;
std::vector<int16_t> silence_buffer_;

int sample_rate_;
int num_channels_;
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10 changes: 6 additions & 4 deletions webrtc-sys/src/audio_track.cpp
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Expand Up @@ -150,7 +150,7 @@ AudioTrackSource::InternalSource::InternalSource(

int samples10ms = sample_rate / 100 * num_channels;

silence_buffer_ = new int16_t[samples10ms]();
silence_buffer_.assign(samples10ms, 0);
queue_size_samples_ = queue_size_ms / 10 * samples10ms;
notify_threshold_samples_ = queue_size_samples_; // TODO: this is currently
// using x2 the queue size
Expand All @@ -164,18 +164,21 @@ AudioTrackSource::InternalSource::InternalSource(
audio_queue_.get(),
[this, samples10ms]() {
webrtc::MutexLock lock(&mutex_);
constexpr int kBitsPerSample = sizeof(int16_t) * 8;

if (buffer_.size() >= samples10ms) {
// Reset |missed_frames_| to 0 so that it won't keep sending silence to webrtc due to audio callback timing drifts.
missed_frames_ = 0;
for (auto sink : sinks_)
sink->OnData(buffer_.data(), sizeof(int16_t) * 8, sample_rate_,
sink->OnData(buffer_.data(), kBitsPerSample, sample_rate_,
num_channels_, samples10ms / num_channels_);

buffer_.erase(buffer_.begin(), buffer_.begin() + samples10ms);
} else {
missed_frames_++;
if (missed_frames_ >= silence_frames_threshold) {
for (auto sink : sinks_)
sink->OnData(silence_buffer_, sizeof(int16_t) * 8, sample_rate_,
sink->OnData(silence_buffer_.data(), kBitsPerSample, sample_rate_,
num_channels_, samples10ms / num_channels_);
}
}
Expand All @@ -192,7 +195,6 @@ AudioTrackSource::InternalSource::InternalSource(
}

AudioTrackSource::InternalSource::~InternalSource() {
delete[] silence_buffer_;
}

bool AudioTrackSource::InternalSource::capture_frame(
Expand Down