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Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay

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This metric was recently added to the standard (see https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay). This CL implements it for audio streams.

Bug: webrtc:14141
Change-Id: I79d918639cd12361ebbc28c2be41549e33fa7e2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262770
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37567}
This commit is contained in:
Ivo Creusen
2022-07-19 16:33:10 +02:00
committed by WebRTC LUCI CQ
parent e7696f771d
commit 1a84b565ac
21 changed files with 74 additions and 9 deletions
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4
pc/rtc_stats_collector_unittest.cc
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@ -2060,6 +2060,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
voice_media_info.receivers[0].jitter_ms = 4500;
voice_media_info.receivers[0].jitter_buffer_delay_seconds = 1.0;
voice_media_info.receivers[0].jitter_buffer_target_delay_seconds = 1.1;
voice_media_info.receivers[0].jitter_buffer_minimum_delay_seconds = 0.999;
voice_media_info.receivers[0].jitter_buffer_emitted_count = 2;
voice_media_info.receivers[0].total_samples_received = 3;
voice_media_info.receivers[0].concealed_samples = 4;
@ -2114,6 +2115,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
expected_audio.jitter = 4.5;
expected_audio.jitter_buffer_delay = 1.0;
expected_audio.jitter_buffer_target_delay = 1.1;
expected_audio.jitter_buffer_minimum_delay = 0.999;
expected_audio.jitter_buffer_emitted_count = 2;
expected_audio.total_samples_received = 3;
expected_audio.concealed_samples = 4;
@ -2180,6 +2182,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
video_media_info.receivers[0].jitter_ms = 1199;
video_media_info.receivers[0].jitter_buffer_delay_seconds = 3.456;
video_media_info.receivers[0].jitter_buffer_target_delay_seconds = 1.1;
video_media_info.receivers[0].jitter_buffer_minimum_delay_seconds = 0.999;
video_media_info.receivers[0].jitter_buffer_emitted_count = 13;
video_media_info.receivers[0].last_packet_received_timestamp_ms =
@ -2241,6 +2244,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
expected_video.jitter = 1.199;
expected_video.jitter_buffer_delay = 3.456;
expected_video.jitter_buffer_target_delay = 1.1;
expected_video.jitter_buffer_minimum_delay = 0.999;
expected_video.jitter_buffer_emitted_count = 13;
// `expected_video.last_packet_received_timestamp` should be undefined.
// `expected_video.content_type` should be undefined.