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Expose new audio stats on the API

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Several new audio stats were recently standardized and implemented in
WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887. This CL
adds these to the GetStats API.

Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27839}
This commit is contained in:
Ivo Creusen
2019-04-30 09:45:21 +02:00
committed by Commit Bot
parent e847481dc8
commit 8d8ffdbcca
12 changed files with 88 additions and 3 deletions
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8
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -251,6 +251,8 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
acm_stat->silentConcealedSamples =
neteq_lifetime_stat.silent_concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
acm_stat->jitterBufferEmittedCount =
@ -262,6 +264,12 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
acm_stat->interruptionCount = neteq_lifetime_stat.interruption_count;
acm_stat->totalInterruptionDurationMs =
neteq_lifetime_stat.total_interruption_duration_ms;
acm_stat->insertedSamplesForDeceleration =
neteq_lifetime_stat.inserted_samples_for_deceleration;
acm_stat->removedSamplesForAcceleration =
neteq_lifetime_stat.removed_samples_for_acceleration;
acm_stat->fecPacketsReceived = neteq_lifetime_stat.fec_packets_received;
acm_stat->fecPacketsDiscarded = neteq_lifetime_stat.fec_packets_discarded;
NetEqOperationsAndState neteq_operations_and_state =
neteq_->GetOperationsAndState();

5
modules/audio_coding/include/audio_coding_module_typedefs.h
View File

@ -84,9 +84,14 @@ struct NetworkStatistics {
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
uint64_t totalSamplesReceived;
uint64_t concealedSamples;
uint64_t silentConcealedSamples;
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferEmittedCount;
uint64_t insertedSamplesForDeceleration;
uint64_t removedSamplesForAcceleration;
uint64_t fecPacketsReceived;
uint64_t fecPacketsDiscarded;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
uint16_t currentPacketLossRate;