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Adds WebRTC.Audio.EncodingTaskQueueLatencyMs

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Part II of https://webrtc-review.googlesource.com/c/src/+/1584

Bug: webrtc:8206
Change-Id: I71ff164a884c61404d1c542d943dd12a5ee2de6f
Reviewed-on: https://webrtc-review.googlesource.com/4180
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20012}
This commit is contained in:
henrika 2017-09-28 09:39:34 +02:00 committed by Commit Bot
parent 7e9c614648
commit 4580217b56
1 changed files with 10 additions and 0 deletions
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10
voice_engine/channel.cc
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@ -38,6 +38,7 @@
#include "rtc_base/thread_checker.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/trace.h"
#include "voice_engine/statistics.h"
#include "voice_engine/utility.h"
@ -1643,6 +1644,9 @@ void Channel::ProcessAndEncodeAudio(const AudioFrame& audio_input) {
// either into pool of frames or into the task itself.
audio_frame->CopyFrom(audio_input);
audio_frame->id_ = ChannelId();
// 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(std::unique_ptr<rtc::QueuedTask>(
new ProcessAndEncodeAudioTask(std::move(audio_frame), this)));
}
@ -1682,6 +1686,12 @@ void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
RTC_DCHECK_LE(audio_input->num_channels_, 2);
RTC_DCHECK_EQ(audio_input->id_, ChannelId());
// Measure time between when the audio frame is added to the task queue and
// when the task is actually executed. Goal is to keep track of unwanted
// extra latency added by the task queue.
RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs",
audio_input->ElapsedProfileTimeMs());
bool is_muted = InputMute();
AudioFrameOperations::Mute(audio_input, previous_frame_muted_, is_muted);