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Fixes issue where WebRTC.Audio.RecordSampleRateOffsetInPercent can report 100%

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Bug: b/113648245
Change-Id: I5fe22b553177cf7f53095b691077b3efd7c6bb59
Reviewed-on: https://webrtc-review.googlesource.com/100241
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24733}
This commit is contained in:
henrika 2018-09-13 16:57:01 +02:00 committed by Commit Bot
parent c0af56b9fb
commit c5fe166dbc
1 changed files with 16 additions and 15 deletions
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31
modules/audio_device/audio_device_buffer.cc
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@ -400,7 +400,8 @@ void AudioDeviceBuffer::LogStats(LogState 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) {
if (++num_stat_reports_ > 2 &&
static_cast<size_t>(time_since_last) > kTimerIntervalInMilliseconds / 2) {
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;
@ -410,14 +411,14 @@ void AudioDeviceBuffer::LogStats(LogState state) {
((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: "
<< 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;
}
RTC_LOG(INFO) << "[REC : " << time_since_last << "msec, "
<< 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);
@ -428,14 +429,14 @@ void AudioDeviceBuffer::LogStats(LogState state) {
((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: "
<< 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;
}
RTC_LOG(INFO) << "[PLAY: " << time_since_last << "msec, "
<< 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;