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Fix headless ADM playout drift below 100 Hz #244

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Fix headless ADM playout drift below 100 Hz #244

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7 changes: 5 additions & 2 deletions webrtc-jni/src/main/cpp/include/api/HeadlessAudioDeviceModule.h
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Original file line number Diff line number Diff line change
Expand Up @@ -166,8 +166,11 @@ namespace jni

size_t playoutFramesIn10MS_;
size_t recordingFramesIn10MS_;
int64_t lastCallPlayoutMillis_;
int64_t lastCallRecordMillis_;
// Absolute wall-clock deadline (ms) of the next 10 ms tick. Advanced by a
// fixed +10 each tick so scheduling/wake-up latency is corrected against the
// grid rather than accumulating into the frame period.
int64_t nextPlayoutMillis_;
int64_t nextRecordMillis_;

mutable webrtc::Mutex mutex_;
std::unique_ptr<webrtc::AudioDeviceBuffer> audio_device_buffer_ RTC_GUARDED_BY(mutex_);
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53 changes: 40 additions & 13 deletions webrtc-jni/src/main/cpp/src/api/HeadlessAudioDeviceModule.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -25,8 +25,8 @@ namespace jni
channels_(channels ? channels : 1),
playoutFramesIn10MS_(0),
recordingFramesIn10MS_(0),
lastCallPlayoutMillis_(0),
lastCallRecordMillis_(0),
nextPlayoutMillis_(0),
nextRecordMillis_(0),
audio_callback_(nullptr)
{
audio_device_buffer_ = std::make_unique<webrtc::AudioDeviceBuffer>(&env.task_queue_factory());
Expand Down Expand Up @@ -553,22 +553,35 @@ namespace jni
int64_t currentTime = webrtc::TimeMillis();
mutex_.Lock();

if (lastCallPlayoutMillis_ == 0 || currentTime - lastCallPlayoutMillis_ >= 10) {
// Seed the grid on the first tick.
if (nextPlayoutMillis_ == 0) {
nextPlayoutMillis_ = currentTime;
}

if (currentTime >= nextPlayoutMillis_) {
mutex_.Unlock();
audio_device_buffer_->RequestPlayoutData(playoutFramesIn10MS_);
mutex_.Lock();

audio_device_buffer_->GetPlayoutData(play_buffer_.data());

lastCallPlayoutMillis_ = currentTime;
// Advance the grid by a fixed 10 ms rather than re-anchoring to currentTime,
// so wake-up latency is corrected on the next tick instead of accumulating
// into the frame period (which otherwise pulls the effective rate below 100 Hz).
nextPlayoutMillis_ += 10;

// If we fell far behind (e.g. the thread was descheduled), resync to now
// instead of bursting frames to catch up.
if (nextPlayoutMillis_ < currentTime - 100) {
nextPlayoutMillis_ = currentTime;
}
}

int64_t sleepMillis = nextPlayoutMillis_ - webrtc::TimeMillis();
mutex_.Unlock();

int64_t deltaTimeMillis = webrtc::TimeMillis() - currentTime;

if (deltaTimeMillis < 10) {
webrtc::Thread::SleepMs(10 - deltaTimeMillis);
if (sleepMillis > 0) {
webrtc::Thread::SleepMs(sleepMillis);
}

return true;
Expand All @@ -588,7 +601,12 @@ namespace jni
int64_t currentTime = webrtc::TimeMillis();
mutex_.Lock();

if (lastCallRecordMillis_ == 0 || currentTime - lastCallRecordMillis_ >= 10) {
// Seed the grid on the first tick.
if (nextRecordMillis_ == 0) {
nextRecordMillis_ = currentTime;
}

if (currentTime >= nextRecordMillis_) {
size_t nSamplesOut = 0;
const size_t nBytesPerSample = sizeof(int16_t);
const size_t nChannels = channels_;
Expand Down Expand Up @@ -617,19 +635,28 @@ namespace jni
audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(), recordingFramesIn10MS_);
audio_device_buffer_->SetVQEData(/*play_delay_ms*/ 0, /*rec_delay_ms*/ 0);

lastCallRecordMillis_ = currentTime;
// Advance the grid by a fixed 10 ms rather than re-anchoring to currentTime,
// so wake-up latency is corrected on the next tick instead of accumulating
// into the frame period (which otherwise pulls the effective rate below 100 Hz).
nextRecordMillis_ += 10;

// If we fell far behind (e.g. the thread was descheduled), resync to now
// instead of bursting frames to catch up.
if (nextRecordMillis_ < currentTime - 100) {
nextRecordMillis_ = currentTime;
}

mutex_.Unlock();
audio_device_buffer_->DeliverRecordedData();
mutex_.Lock();
}
}

int64_t sleepMillis = nextRecordMillis_ - webrtc::TimeMillis();
mutex_.Unlock();

int64_t deltaTimeMillis = webrtc::TimeMillis() - currentTime;
if (deltaTimeMillis < 10) {
webrtc::Thread::SleepMs(10 - deltaTimeMillis);
if (sleepMillis > 0) {
webrtc::Thread::SleepMs(sleepMillis);
}

return true;
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