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";
