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Avoid waking up encoder thread when audio send stream is stopped.

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Remove the default enabled "WebRTC-Audio-FixTimestampStall" field trial which was rolled out 2 years ago without any issues.

Also change the include audio level indication member to be atomic since it is accessed on multiple threads.

Bug: webrtc:14804
Change-Id: I4c5145e1fb03351154162b4293a3bd870e4793cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290886
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39125}
This commit is contained in:
Jakob Ivarsson 2023-01-17 15:40:36 +01:00 committed by WebRTC LUCI CQ
parent 9a8aa209ab
commit 478f3b786e
1 changed files with 20 additions and 37 deletions
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57
audio/channel_send.cc
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@ -197,16 +197,13 @@ class ChannelSend : public ChannelSendInterface,
std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
std::unique_ptr<AudioCodingModule> audio_coding_;
uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
// uses
// This is just an offset, RTP module will add its own random offset.
uint32_t timestamp_ RTC_GUARDED_BY(audio_thread_race_checker_) = 0;
RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_);
bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
// VoeRTP_RTCP
// TODO(henrika): can today be accessed on the main thread and on the
// task queue; hence potential race.
bool _includeAudioLevelIndication;
bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false;
bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_) = false;
// RtcpBandwidthObserver
const std::unique_ptr<VoERtcpObserver> rtcp_observer_;
@ -219,7 +216,8 @@ class ChannelSend : public ChannelSendInterface,
SequenceChecker construction_thread_;
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_) = false;
std::atomic<bool> include_audio_level_indication_ = false;
std::atomic<bool> encoder_queue_is_active_ = false;
// E2EE Audio Frame Encryption
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
@ -233,8 +231,6 @@ class ChannelSend : public ChannelSendInterface,
rtc::scoped_refptr<ChannelSendFrameTransformerDelegate>
frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_);
const bool fixing_timestamp_stall_;
mutable Mutex rtcp_counter_mutex_;
RtcpPacketTypeCounter rtcp_packet_type_counter_
RTC_GUARDED_BY(rtcp_counter_mutex_);
@ -373,7 +369,7 @@ int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
uint32_t rtp_timestamp,
rtc::ArrayView<const uint8_t> payload,
int64_t absolute_capture_timestamp_ms) {
if (_includeAudioLevelIndication) {
if (include_audio_level_indication_.load()) {
// Store current audio level in the RTP sender.
// The level will be used in combination with voice-activity state
// (frameType) to add an RTP header extension
@ -463,11 +459,6 @@ ChannelSend::ChannelSend(
const FieldTrialsView& field_trials)
: ssrc_(ssrc),
event_log_(rtc_event_log),
_timeStamp(0), // This is just an offset, RTP module will add it's own
// random offset
input_mute_(false),
previous_frame_muted_(false),
_includeAudioLevelIndication(false),
rtcp_observer_(new VoERtcpObserver(this)),
feedback_observer_(feedback_observer),
rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
@ -475,8 +466,6 @@ ChannelSend::ChannelSend(
new RateLimiter(clock, kMaxRetransmissionWindowMs)),
frame_encryptor_(frame_encryptor),
crypto_options_(crypto_options),
fixing_timestamp_stall_(
field_trials.IsDisabled("WebRTC-Audio-FixTimestampStall")),
encoder_queue_(task_queue_factory->CreateTaskQueue(
"AudioEncoder",
TaskQueueFactory::Priority::NORMAL)) {
@ -540,10 +529,7 @@ void ChannelSend::StartSend() {
RTC_DCHECK_EQ(0, ret);
// It is now OK to start processing on the encoder task queue.
encoder_queue_.PostTask([this] {
RTC_DCHECK_RUN_ON(&encoder_queue_);
encoder_queue_is_active_ = true;
});
encoder_queue_is_active_.store(true);
}
void ChannelSend::StopSend() {
@ -552,11 +538,12 @@ void ChannelSend::StopSend() {
return;
}
sending_ = false;
encoder_queue_is_active_.store(false);
// Wait until all pending encode tasks are executed.
rtc::Event flush;
encoder_queue_.PostTask([this, &flush]() {
RTC_DCHECK_RUN_ON(&encoder_queue_);
encoder_queue_is_active_ = false;
flush.Set();
});
flush.Wait(rtc::Event::kForever);
@ -701,7 +688,7 @@ void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
_includeAudioLevelIndication = enable;
include_audio_level_indication_.store(enable);
if (enable) {
rtp_rtcp_->RegisterRtpHeaderExtension(AudioLevel::Uri(), id);
} else {
@ -815,17 +802,19 @@ void ChannelSend::ProcessAndEncodeAudio(
RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
RTC_DCHECK_LE(audio_frame->num_channels_, 8);
audio_frame->timestamp_ = timestamp_;
timestamp_ += audio_frame->samples_per_channel_;
if (!encoder_queue_is_active_.load()) {
return;
}
// Profile time between when the audio frame is added to the task queue and
// when the task is actually executed.
audio_frame->UpdateProfileTimeStamp();
encoder_queue_.PostTask(
[this, audio_frame = std::move(audio_frame)]() mutable {
RTC_DCHECK_RUN_ON(&encoder_queue_);
if (!encoder_queue_is_active_) {
if (fixing_timestamp_stall_) {
_timeStamp +=
static_cast<uint32_t>(audio_frame->samples_per_channel_);
}
if (!encoder_queue_is_active_.load()) {
return;
}
// Measure time between when the audio frame is added to the task queue
@ -838,7 +827,7 @@ void ChannelSend::ProcessAndEncodeAudio(
AudioFrameOperations::Mute(audio_frame.get(), previous_frame_muted_,
is_muted);
if (_includeAudioLevelIndication) {
if (include_audio_level_indication_.load()) {
size_t length =
audio_frame->samples_per_channel_ * audio_frame->num_channels_;
RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes);
@ -851,10 +840,6 @@ void ChannelSend::ProcessAndEncodeAudio(
}
previous_frame_muted_ = is_muted;
// Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
// The ACM resamples internally.
audio_frame->timestamp_ = _timeStamp;
// This call will trigger AudioPacketizationCallback::SendData if
// encoding is done and payload is ready for packetization and
// transmission. Otherwise, it will return without invoking the
@ -863,8 +848,6 @@ void ChannelSend::ProcessAndEncodeAudio(
RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
return;
}
_timeStamp += static_cast<uint32_t>(audio_frame->samples_per_channel_);
});
}