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Don't overwrite audio stats when they're not available.

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Chromium implements AudioProcessorInterface::GetStats(), but other
clients may not. The existing stats were getting overwritten with
default AudioProcessorStats values in that case.

Now, we only overwrite the stats if the track has an
AudioProcessorInterface. Also, move signal level out of
SetAudioProcessingStats() to avoid the "don't set if it's -1" pattern.

Review URL: https://codereview.webrtc.org/1469803004

Cr-Commit-Position: refs/heads/master@{#10831}
This commit is contained in:
andrew 2015-11-27 17:27:35 -08:00 committed by Commit bot
parent 7e43138c08
commit 2fe1cb0f0a
1 changed files with 27 additions and 21 deletions
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48
talk/app/webrtc/statscollector.cc
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@ -115,17 +115,17 @@ void ExtractCommonReceiveProperties(const cricket::MediaReceiverInfo& info,
report->AddString(StatsReport::kStatsValueNameCodecName, info.codec_name);
}
void SetAudioProcessingStats(StatsReport* report, int signal_level,
bool typing_noise_detected, int echo_return_loss,
int echo_return_loss_enhancement, int echo_delay_median_ms,
float aec_quality_min, int echo_delay_std_ms) {
void SetAudioProcessingStats(StatsReport* report,
bool typing_noise_detected,
int echo_return_loss,
int echo_return_loss_enhancement,
int echo_delay_median_ms,
float aec_quality_min,
int echo_delay_std_ms) {
report->AddBoolean(StatsReport::kStatsValueNameTypingNoiseState,
typing_noise_detected);
report->AddFloat(StatsReport::kStatsValueNameEchoCancellationQualityMin,
aec_quality_min);
// Don't overwrite the previous signal level if it's not available now.
if (signal_level >= 0)
report->AddInt(StatsReport::kStatsValueNameAudioInputLevel, signal_level);
const IntForAdd ints[] = {
{ StatsReport::kStatsValueNameEchoReturnLoss, echo_return_loss },
{ StatsReport::kStatsValueNameEchoReturnLossEnhancement,
@ -182,11 +182,14 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
ExtractCommonSendProperties(info, report);
SetAudioProcessingStats(report, info.audio_level, info.typing_noise_detected,
info.echo_return_loss, info.echo_return_loss_enhancement,
info.echo_delay_median_ms, info.aec_quality_min, info.echo_delay_std_ms);
SetAudioProcessingStats(
report, info.typing_noise_detected, info.echo_return_loss,
info.echo_return_loss_enhancement, info.echo_delay_median_ms,
info.aec_quality_min, info.echo_delay_std_ms);
RTC_DCHECK_GE(info.audio_level, 0);
const IntForAdd ints[] = {
{ StatsReport::kStatsValueNameAudioInputLevel, info.audio_level},
{ StatsReport::kStatsValueNameJitterReceived, info.jitter_ms },
{ StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
{ StatsReport::kStatsValueNamePacketsSent, info.packets_sent },
@ -891,21 +894,24 @@ void StatsCollector::UpdateReportFromAudioTrack(AudioTrackInterface* track,
RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
RTC_DCHECK(track != NULL);
int signal_level = 0;
if (!track->GetSignalLevel(&signal_level))
signal_level = -1;
// Don't overwrite report values if they're not available.
int signal_level;
if (track->GetSignalLevel(&signal_level)) {
RTC_DCHECK_GE(signal_level, 0);
report->AddInt(StatsReport::kStatsValueNameAudioInputLevel, signal_level);
}
rtc::scoped_refptr<AudioProcessorInterface> audio_processor(
track->GetAudioProcessor());
auto audio_processor(track->GetAudioProcessor());
AudioProcessorInterface::AudioProcessorStats stats;
if (audio_processor.get())
if (audio_processor.get()) {
AudioProcessorInterface::AudioProcessorStats stats;
audio_processor->GetStats(&stats);
SetAudioProcessingStats(report, signal_level, stats.typing_noise_detected,
stats.echo_return_loss, stats.echo_return_loss_enhancement,
stats.echo_delay_median_ms, stats.aec_quality_min,
stats.echo_delay_std_ms);
SetAudioProcessingStats(
report, stats.typing_noise_detected, stats.echo_return_loss,
stats.echo_return_loss_enhancement, stats.echo_delay_median_ms,
stats.aec_quality_min, stats.echo_delay_std_ms);
}
}
bool StatsCollector::GetTrackIdBySsrc(uint32_t ssrc,