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Relands Adds WebRTC.Audio.Record/PlayoutSampleRateOffsetInPercent UMA stats to native WebRTC

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First version was reverted in https://webrtc-review.googlesource.com/c/src/+/97941.
The issue is now fixed.

TBR=ivoc

Bug: b/113648245
Change-Id: If631fdea95aa963952f15e48e9d2d678797dc225
Reviewed-on: https://webrtc-review.googlesource.com/97942
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24573}
This commit is contained in:
henrika
2018-09-05 11:36:22 +02:00
committed by Commit Bot
parent b29409064e
commit cfbd26df1e
3 changed files with 46 additions and 34 deletions
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48
modules/audio_device/audio_device_buffer.cc
View File

@ -181,50 +181,42 @@ void AudioDeviceBuffer::StopRecording() {
}
int32_t AudioDeviceBuffer::SetRecordingSampleRate(uint32_t fsHz) {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
RTC_LOG(INFO) << "SetRecordingSampleRate(" << fsHz << ")";
rec_sample_rate_ = fsHz;
return 0;
}
int32_t AudioDeviceBuffer::SetPlayoutSampleRate(uint32_t fsHz) {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
RTC_LOG(INFO) << "SetPlayoutSampleRate(" << fsHz << ")";
play_sample_rate_ = fsHz;
return 0;
}
int32_t AudioDeviceBuffer::RecordingSampleRate() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
uint32_t AudioDeviceBuffer::RecordingSampleRate() const {
return rec_sample_rate_;
}
int32_t AudioDeviceBuffer::PlayoutSampleRate() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
uint32_t AudioDeviceBuffer::PlayoutSampleRate() const {
return play_sample_rate_;
}
int32_t AudioDeviceBuffer::SetRecordingChannels(size_t channels) {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
rec_channels_ = channels;
return 0;
}
int32_t AudioDeviceBuffer::SetPlayoutChannels(size_t channels) {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
play_channels_ = channels;
return 0;
}
size_t AudioDeviceBuffer::RecordingChannels() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
return rec_channels_;
}
size_t AudioDeviceBuffer::PlayoutChannels() const {
RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
return play_channels_;
}
@ -419,28 +411,52 @@ void AudioDeviceBuffer::LogStats(LogState state) {
stats_.max_play_level = 0;
}
// Log the latest statistics but skip the first round just after state was
// set to LOG_START. Hence, first printed log will be after ~10 seconds.
if (++num_stat_reports_ > 1 && time_since_last > 0) {
// Cache current sample rate from atomic members.
const uint32_t rec_sample_rate = rec_sample_rate_;
const uint32_t play_sample_rate = play_sample_rate_;
// Log the latest statistics but skip the first two rounds just after state
// was set to LOG_START to ensure that we have at least one full stable
// 10-second interval for sample-rate estimation. Hence, first printed log
// will be after ~20 seconds.
if (++num_stat_reports_ > 2 && time_since_last > 0) {
uint32_t diff_samples = stats.rec_samples - last_stats_.rec_samples;
float rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
uint32_t abs_diff_rate_in_percent = 0;
if (rec_sample_rate > 0) {
abs_diff_rate_in_percent = static_cast<uint32_t>(
0.5f +
((100.0f * std::abs(rate - rec_sample_rate)) / rec_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.RecordSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
}
RTC_LOG(INFO) << "[REC : " << time_since_last << "msec, "
<< rec_sample_rate_ / 1000 << "kHz] callbacks: "
<< rec_sample_rate / 1000 << "kHz] callbacks: "
<< stats.rec_callbacks - last_stats_.rec_callbacks << ", "
<< "samples: " << diff_samples << ", "
<< "rate: " << static_cast<int>(rate + 0.5) << ", "
<< "rate diff: " << abs_diff_rate_in_percent << "%, "
<< "level: " << stats.max_rec_level;
diff_samples = stats.play_samples - last_stats_.play_samples;
rate = diff_samples / (static_cast<float>(time_since_last) / 1000.0);
abs_diff_rate_in_percent = 0;
if (play_sample_rate > 0) {
abs_diff_rate_in_percent = static_cast<uint32_t>(
0.5f +
((100.0f * std::abs(rate - play_sample_rate)) / play_sample_rate));
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Audio.PlayoutSampleRateOffsetInPercent",
abs_diff_rate_in_percent);
}
RTC_LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
<< play_sample_rate_ / 1000 << "kHz] callbacks: "
<< play_sample_rate / 1000 << "kHz] callbacks: "
<< stats.play_callbacks - last_stats_.play_callbacks << ", "
<< "samples: " << diff_samples << ", "
<< "rate: " << static_cast<int>(rate + 0.5) << ", "
<< "rate diff: " << abs_diff_rate_in_percent << "%, "
<< "level: " << stats.max_play_level;
last_stats_ = stats;
}
last_stats_ = stats;
int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";