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Adding packetsDiscarded to RTCReceivedRtpStreamStats.

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Bug: webrtc:12532, webrtc:7065, webrtc:8199
Change-Id: I3ba62ec65e5660e98787f629aec3ee7a0889207a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225261
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34468}
This commit is contained in:
Minyue Li
2021-07-07 15:53:38 +02:00
committed by WebRTC LUCI CQ
parent 6a9dec9392
commit 28a2c63526
13 changed files with 85 additions and 21 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -307,6 +307,8 @@ void AcmReceiver::GetNetworkStatistics(
neteq_->GetOperationsAndState();
acm_stat->packetBufferFlushes =
neteq_operations_and_state.packet_buffer_flushes;
acm_stat->packetsDiscarded =
neteq_operations_and_state.discarded_primary_packets;
}
int AcmReceiver::EnableNack(size_t max_nack_list_size) {

9
modules/audio_coding/include/audio_coding_module_typedefs.h
View File

@ -81,19 +81,22 @@ struct NetworkStatistics {
// adding extra delay due to "peaky jitter"
bool jitterPeaksFound;
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
uint64_t totalSamplesReceived;
uint64_t concealedSamples;
uint64_t silentConcealedSamples;
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferEmittedCount;
// Non standard stats propagated to spec complaint GetStats API.
uint64_t jitterBufferTargetDelayMs;
uint64_t insertedSamplesForDeceleration;
uint64_t removedSamplesForAcceleration;
uint64_t fecPacketsReceived;
uint64_t fecPacketsDiscarded;
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
// https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
uint64_t packetsDiscarded;
// Non standard stats propagated to spec complaint GetStats API.
uint64_t jitterBufferTargetDelayMs;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// fraction (of original stream) of synthesized audio inserted through
// expansion (in Q14)