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[Stats] Expose totalPacketSendDelay for audio as well.

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This information is now readily available. Let's expose it.

In practise we don't pace audio by default and the delay is ~0, however
we can tell that this metric is working as intended by setting
PacingController's pace_audio_ to true via the "WebRTC-Pacer-BlockAudio"
field trial. In this case chrome://webrtc-internals/ plots neats graphs
for audio send delay.

Bug: webrtc:10635
Change-Id: Iecfd93bb84ec61e5d54232769a9e7a500601b199
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280523
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38483}
This commit is contained in:
Henrik Boström
2022-10-26 16:53:03 +02:00
committed by WebRTC LUCI CQ
parent 86cfcc5eef
commit aebba7b468
10 changed files with 17 additions and 11 deletions
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2
api/stats/rtcstats_objects.h
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@ -549,8 +549,6 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final : public RTCRTPStreamStats {
RTCStatsMember<double> frames_per_second;
RTCStatsMember<uint32_t> frames_sent;
RTCStatsMember<uint32_t> huge_frames_sent;
// TODO(https://crbug.com/webrtc/10635): This is only implemented for video;
// implement it for audio as well.
RTCStatsMember<double> total_packet_send_delay;
// Enum type RTCQualityLimitationReason
RTCStatsMember<std::string> quality_limitation_reason;