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Delete unused NetEq stats currentPacketLossRate, currentDiscardRate and addedSamples

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Bug: webrtc:11622
Change-Id: I097bb7284d952ada41f4f38dd7adf3536bd040ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183620
Reviewed-by: Minyue Li <minyue@google.com>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32148}
This commit is contained in:
Niels Möller
2020-09-21 10:05:32 +02:00
committed by Commit Bot
parent 4c87d83d03
commit 4461f059d1
13 changed files with 15 additions and 74 deletions
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8
modules/audio_coding/include/audio_coding_module_typedefs.h
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@ -97,12 +97,6 @@ struct NetworkStatistics {
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;
// Late loss rate; fraction between 0 and 1, scaled to Q14.
union {
RTC_DEPRECATED uint16_t currentDiscardRate;
};
// fraction (of original stream) of synthesized audio inserted through
// expansion (in Q14)
uint16_t currentExpandRate;
@ -129,8 +123,6 @@ struct NetworkStatistics {
int minWaitingTimeMs;
// max packet waiting time in the jitter buffer (ms)
int maxWaitingTimeMs;
// added samples in off mode due to packet loss
size_t addedSamples;
// count of the number of buffer flushes
uint64_t packetBufferFlushes;
// number of samples expanded due to delayed packets