commit 8d8ffdbccacb6664cfa5f8d781fb7c2a362a6c10
author Ivo Creusen <ivoc@webrtc.org> Tue Apr 30 07:45:21 2019
committer Commit Bot <commit-bot@chromium.org> Fri May 03 10:10:15 2019
tree e1d9e484a8aefa1f16aab2cfbfaf9ed43ff83ce7
parent e847481dc8d57dd9aa287f7e51e42c3bcdb37e53

Expose new audio stats on the API

Several new audio stats were recently standardized and implemented in
WebRTC in https://webrtc-review.googlesource.com/c/src/+/133887. This CL
adds these to the GetStats API.

Bug: webrtc:10442, webrtc:10443, webrtc:10444
Change-Id: I0e898ac14777e82b1a9099b5e0a5584eb9cb5934
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134213
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27839}

call/audio_receive_stream.h

diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 9091afd..35c6ef7 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -38,6 +38,8 @@
     uint32_t remote_ssrc = 0;
     int64_t bytes_rcvd = 0;
     uint32_t packets_rcvd = 0;
+    uint64_t fec_packets_received = 0;
+    uint64_t fec_packets_discarded = 0;
     uint32_t packets_lost = 0;
     float fraction_lost = 0.0f;
     std::string codec_name;
@@ -54,9 +56,12 @@
     uint64_t total_samples_received = 0;
     double total_output_duration = 0.0;
     uint64_t concealed_samples = 0;
+    uint64_t silent_concealed_samples = 0;
     uint64_t concealment_events = 0;
     double jitter_buffer_delay_seconds = 0.0;
     uint64_t jitter_buffer_emitted_count = 0;
+    uint64_t inserted_samples_for_deceleration = 0;
+    uint64_t removed_samples_for_acceleration = 0;
     // Stats below DO NOT correspond directly to anything in the WebRTC stats
     float expand_rate = 0.0f;
     float speech_expand_rate = 0.0f;