[Stats] Expose totalPacketSendDelay for audio as well.
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}
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WebRTC LUCI CQ
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@ -2758,6 +2758,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
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voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
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voice_media_info.senders[0].local_stats[0].ssrc = 1;
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voice_media_info.senders[0].packets_sent = 2;
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voice_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(1);
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voice_media_info.senders[0].retransmitted_packets_sent = 20;
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voice_media_info.senders[0].payload_bytes_sent = 3;
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voice_media_info.senders[0].header_and_padding_bytes_sent = 12;
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@ -2795,6 +2796,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
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expected_audio.transport_id = "TTransportName1";
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expected_audio.codec_id = "COTTransportName1_42";
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expected_audio.packets_sent = 2;
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expected_audio.total_packet_send_delay = 1;
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expected_audio.retransmitted_packets_sent = 20;
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expected_audio.bytes_sent = 3;
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expected_audio.header_bytes_sent = 12;
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@ -3192,6 +3194,7 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
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voice_media_info.senders[0].local_stats.push_back(cricket::SsrcSenderInfo());
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voice_media_info.senders[0].local_stats[0].ssrc = 1;
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voice_media_info.senders[0].packets_sent = 2;
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voice_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(0.5);
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voice_media_info.senders[0].retransmitted_packets_sent = 20;
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voice_media_info.senders[0].payload_bytes_sent = 3;
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voice_media_info.senders[0].header_and_padding_bytes_sent = 4;
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@ -3228,6 +3231,7 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
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expected_audio.transport_id = "TTransportName1";
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expected_audio.codec_id = "COTTransportName1_42";
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expected_audio.packets_sent = 2;
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expected_audio.total_packet_send_delay = 0.5;
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expected_audio.retransmitted_packets_sent = 20;
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expected_audio.bytes_sent = 3;
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expected_audio.header_bytes_sent = 4;
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