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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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5
modules/audio_coding/neteq/delay_manager.cc
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@ -101,6 +101,7 @@ void DelayManager::Update(int arrival_delay_ms, bool reordered) {
target_level_ms_ = std::max(
target_level_ms_, reorder_optimizer_->GetOptimalDelayMs().value_or(0));
}
unlimited_target_level_ms_ = target_level_ms_;
target_level_ms_ = std::max(target_level_ms_, effective_minimum_delay_ms_);
if (maximum_delay_ms_ > 0) {
target_level_ms_ = std::min(target_level_ms_, maximum_delay_ms_);
@ -134,6 +135,10 @@ int DelayManager::TargetDelayMs() const {
return target_level_ms_;
}
int DelayManager::UnlimitedTargetLevelMs() const {
return unlimited_target_level_ms_;
}
bool DelayManager::IsValidMinimumDelay(int delay_ms) const {
return 0 <= delay_ms && delay_ms <= MinimumDelayUpperBound();
}