[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}
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

View File

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