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Expose jitter buffer flushes metric in new getStats api.

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Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#

Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}
This commit is contained in:
Ruslan Burakov
2018-11-22 17:21:10 +01:00
committed by Commit Bot
parent b357e54dd5
commit 8af8896596
12 changed files with 34 additions and 12 deletions
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5
modules/audio_coding/acm2/acm_receiver.cc
View File

@ -345,6 +345,11 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
NetEqOperationsAndState neteq_operations_and_state =
neteq_->GetOperationsAndState();
acm_stat->packetBufferFlushes =
neteq_operations_and_state.packet_buffer_flushes;
}
int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,

2
modules/audio_coding/include/audio_coding_module_typedefs.h
View File

@ -117,6 +117,8 @@ struct NetworkStatistics {
int maxWaitingTimeMs;
// added samples in off mode due to packet loss
size_t addedSamples;
// count of the number of buffer flushes
uint64_t packetBufferFlushes;
};
} // namespace webrtc