Change a way, how receive stream is determined in DefaultAudioQualityAnalyzer.

Bug: webrtc:10138
Change-Id: I8955c30f0a5d98abeca029323396e469a2fb243b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136683
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27927}
This commit is contained in:
Artem Titov 2019-05-13 14:26:31 +02:00 committed by Commit Bot
parent 157b7814b9
commit 0379d8cfea

View File

@ -45,15 +45,10 @@ void DefaultAudioQualityAnalyzer::OnStatsReports(
if (strcmp(media_type->static_string_val(), kStatsAudioMediaType) != 0) {
continue;
}
const webrtc::StatsReport::Value* bytes_received = stats_report->FindValue(
StatsReport::StatsValueName::kStatsValueNameBytesReceived);
if (bytes_received == nullptr || bytes_received->int64_val() == 0) {
// Discarding stats in the following situations:
// - When bytes_received is not present, because NetEq stats are only
// available in recv-side SSRC.
// - When bytes_received is present but its value is 0. This means
// that media is not yet flowing so there is no need to keep this
// stats report into account (since all its fields would be 0).
if (stats_report->FindValue(
webrtc::StatsReport::kStatsValueNameBytesSent)) {
// If kStatsValueNameBytesSent is present, it means it's a send stream,
// but we need audio metrics for receive stream, so skip it.
continue;
}