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Unified Diff: talk/app/webrtc/statscollector.cc

Issue 1469803004: Don't overwrite audio stats when they're not available. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Add DCHECK for audio level. Created 5 years, 1 month ago
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Index: talk/app/webrtc/statscollector.cc
diff --git a/talk/app/webrtc/statscollector.cc b/talk/app/webrtc/statscollector.cc
index 28686965e64666f7f225cb03c66c2946491f560e..893e5866b0312a8d064ca027541ae93228b8570c 100644
--- a/talk/app/webrtc/statscollector.cc
+++ b/talk/app/webrtc/statscollector.cc
@@ -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,
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