commit 3ef3bfc2aafa707985c9e9dcd4cfb6ccbc525628
author Henrik Lundin <henrik.lundin@webrtc.org> Tue Apr 10 15:10:26 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Tue Apr 10 21:32:55 2018 +0000
tree f0ae34d12d636be7cf1515576d4331761e246839
parent f0482ea9dd5772e98c18550b443f516d92c372ff

Add new histograms WebRTC.Audio.(Speech)ExpandRatePercent

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}

modules/audio_coding/neteq/neteq_impl.h

diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index bdeb020..3b7070f 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -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 @@
   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);