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Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent

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These two new histograms relate to the packet-loss concealment that
happens when audio packets are lost or late for decoding, and the
NetEq must resort to extrapolating audio from the previously
decoded data.

Bug: webrtc:9126
Change-Id: I99cc97e653169fb742da0092653ab99fd10e5d7b
Reviewed-on: https://webrtc-review.googlesource.com/67861
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22812}
This commit is contained in:
Henrik Lundin
2018-04-10 15:10:26 +02:00
committed by Commit Bot
parent f0482ea9dd
commit 3ef3bfc2aa
5 changed files with 140 additions and 1 deletions
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3
modules/audio_coding/neteq/neteq_impl.h
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@ -17,6 +17,7 @@
#include "api/optional.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/defines.h"
#include "modules/audio_coding/neteq/expand_uma_logger.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/packet.h" // Declare PacketList.
#include "modules/audio_coding/neteq/random_vector.h"
@ -440,6 +441,8 @@ class NetEqImpl : public webrtc::NetEq {
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
RTC_GUARDED_BY(crit_sect_);
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);