Report stats from ChannelReceive::GetAudioFrameWithInfo at 1Hz. This is a change from the previous 100Hz frequency. Also changing the locks slightly in AcmReceiver so that grabbing the neteq lock right after we've let it go, isn't necessary inside of AcmReceiver::GetAudio and also to avoid grabbing the neteq lock while holding the AcmReceiver lock. Bug: webrtc:12868 Change-Id: If6ee35f3dca20eb5bdbc615123aa099ccecf57c5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221371 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34258}
diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h index 9781377..ea7079e 100644 --- a/api/neteq/neteq.h +++ b/api/neteq/neteq.h
@@ -214,11 +214,15 @@ // |data_| in |audio_frame| is not written, but should be interpreted as being // all zeros. For testing purposes, an override can be supplied in the // |action_override| argument, which will cause NetEq to take this action - // next, instead of the action it would normally choose. + // next, instead of the action it would normally choose. An optional output + // argument for fetching the current sample rate can be provided, which + // will return the same value as last_output_sample_rate_hz() but will avoid + // additional synchronization. // Returns kOK on success, or kFail in case of an error. virtual int GetAudio( AudioFrame* audio_frame, bool* muted, + int* current_sample_rate_hz = nullptr, absl::optional<Operation> action_override = absl::nullopt) = 0; // Replaces the current set of decoders with the given one.
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index a921031..0582171 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc
@@ -284,6 +284,15 @@ rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> frame_transformer_delegate_; + + // Counter that's used to control the frequency of reporting histograms + // from the `GetAudioFrameWithInfo` callback. + int audio_frame_interval_count_ RTC_GUARDED_BY(audio_thread_race_checker_) = + 0; + // Controls how many callbacks we let pass by before reporting callback stats. + // A value of 100 means 100 callbacks, each one of which represents 10ms worth + // of data, so the stats reporting frequency will be 1Hz (modulo failures). + constexpr static int kHistogramReportingInterval = 100; }; void ChannelReceive::OnReceivedPayloadData( @@ -332,7 +341,6 @@ rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) { RTC_DCHECK(frame_transformer); RTC_DCHECK(!frame_transformer_delegate_); - RTC_DCHECK(worker_thread_); RTC_DCHECK(worker_thread_->IsCurrent()); // Pass a callback to ChannelReceive::OnReceivedPayloadData, to be called by @@ -457,19 +465,22 @@ } audio_frame->packet_infos_ = RtpPacketInfos(packet_infos); - RTC_DCHECK(worker_thread_); - worker_thread_->PostTask(ToQueuedTask(worker_safety_, [this]() { - RTC_DCHECK_RUN_ON(&worker_thread_checker_); - RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs", - acm_receiver_.TargetDelayMs()); - const int jitter_buffer_delay = acm_receiver_.FilteredCurrentDelayMs(); - RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs", - jitter_buffer_delay + playout_delay_ms_); - RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverJitterBufferDelayMs", - jitter_buffer_delay); - RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDeviceDelayMs", - playout_delay_ms_); - })); + ++audio_frame_interval_count_; + if (audio_frame_interval_count_ >= kHistogramReportingInterval) { + audio_frame_interval_count_ = 0; + worker_thread_->PostTask(ToQueuedTask(worker_safety_, [this]() { + RTC_DCHECK_RUN_ON(&worker_thread_checker_); + RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs", + acm_receiver_.TargetDelayMs()); + const int jitter_buffer_delay = acm_receiver_.FilteredCurrentDelayMs(); + RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs", + jitter_buffer_delay + playout_delay_ms_); + RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverJitterBufferDelayMs", + jitter_buffer_delay); + RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDeviceDelayMs", + playout_delay_ms_); + })); + } return muted ? AudioMixer::Source::AudioFrameInfo::kMuted : AudioMixer::Source::AudioFrameInfo::kNormal;
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc index 0e615ca..3214ce6 100644 --- a/modules/audio_coding/acm2/acm_receiver.cc +++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -146,20 +146,22 @@ AudioFrame* audio_frame, bool* muted) { RTC_DCHECK(muted); - // Accessing members, take the lock. - MutexLock lock(&mutex_); - if (neteq_->GetAudio(audio_frame, muted) != NetEq::kOK) { + int current_sample_rate_hz = 0; + if (neteq_->GetAudio(audio_frame, muted, ¤t_sample_rate_hz) != + NetEq::kOK) { RTC_LOG(LERROR) << "AcmReceiver::GetAudio - NetEq Failed."; return -1; } - const int current_sample_rate_hz = neteq_->last_output_sample_rate_hz(); + RTC_DCHECK_NE(current_sample_rate_hz, 0); // Update if resampling is required. const bool need_resampling = (desired_freq_hz != -1) && (current_sample_rate_hz != desired_freq_hz); + // Accessing members, take the lock. + MutexLock lock(&mutex_); if (need_resampling && !resampled_last_output_frame_) { // Prime the resampler with the last frame. int16_t temp_output[AudioFrame::kMaxDataSizeSamples]; @@ -174,8 +176,8 @@ } } - // TODO(henrik.lundin) Glitches in the output may appear if the output rate - // from NetEq changes. See WebRTC issue 3923. + // TODO(bugs.webrtc.org/3923) Glitches in the output may appear if the output + // rate from NetEq changes. if (need_resampling) { // TODO(yujo): handle this more efficiently for muted frames. int samples_per_channel_int = resampler_.Resample10Msec(
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc index 6ac157f..e9ddbb9 100644 --- a/modules/audio_coding/neteq/neteq_impl.cc +++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -258,6 +258,7 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame, bool* muted, + int* current_sample_rate_hz, absl::optional<Operation> action_override) { TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio"); MutexLock lock(&mutex_); @@ -296,6 +297,11 @@ } } + if (current_sample_rate_hz) { + *current_sample_rate_hz = delayed_last_output_sample_rate_hz_.value_or( + last_output_sample_rate_hz_); + } + return kOK; }
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h index e130422..88da6dc 100644 --- a/modules/audio_coding/neteq/neteq_impl.h +++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -133,6 +133,7 @@ int GetAudio( AudioFrame* audio_frame, bool* muted, + int* current_sample_rate_hz = nullptr, absl::optional<Operation> action_override = absl::nullopt) override; void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc index 1369ead..8553307 100644 --- a/modules/audio_coding/neteq/neteq_unittest.cc +++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -1066,7 +1066,7 @@ expected_target_delay += neteq_->TargetDelayMs() * 2 * kSamples; // We have two packets in the buffer and kAccelerate operation will // extract 20 ms of data. - neteq_->GetAudio(&out_frame_, &muted, NetEq::Operation::kAccelerate); + neteq_->GetAudio(&out_frame_, &muted, nullptr, NetEq::Operation::kAccelerate); // Check jitter buffer delay. NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc index 0988d2c8..22f5ad6 100644 --- a/modules/audio_coding/neteq/tools/neteq_test.cc +++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -172,7 +172,7 @@ } AudioFrame out_frame; bool muted; - int error = neteq_->GetAudio(&out_frame, &muted, + int error = neteq_->GetAudio(&out_frame, &muted, nullptr, ActionToOperations(next_action_)); next_action_ = absl::nullopt; RTC_CHECK(!muted) << "The code does not handle enable_muted_state";