Added support for logging the SSRC corresponding to AudioPlayout events.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d602bb4fdfc47bd63d44c66e780d29129fb7c726..d64011e165edfe358ec8ad2d58a0b12e8f07fcdd 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -450,6 +450,11 @@ int32_t Channel::GetAudioFrame(int32_t id, AudioFrame* audioFrame)
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::GetAudioFrame(id=%d)", id);
 
+    if (event_log_) {
+      unsigned int ssrc;
+      GetLocalSSRC(ssrc);

[hlundin-webrtc, 2015/09/15 12:30:52]

Since GetLocalSSRC has a return value, I'd argue that you CHECK that it always
returns 0. (Currently, it does by definition, but who knows...)

[ivoc, 2015/09/15 14:55:16]

Done.

+      event_log_->LogAudioPlayout(ssrc);
+    }
     // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
     if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_,
                                        audioFrame) == -1)
@@ -719,6 +724,7 @@ Channel::Channel(int32_t channelId,
     volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
     _channelId(channelId),
+    event_log_(event_log),
     rtp_header_parser_(RtpHeaderParser::Create()),
     rtp_payload_registry_(
         new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
@@ -809,7 +815,6 @@ Channel::Channel(int32_t channelId,
     }
     acm_config.neteq_config.enable_fast_accelerate =
         config.Get<NetEqFastAccelerate>().enabled;
-    acm_config.event_log = event_log;
     audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
     _inbandDtmfQueue.ResetDtmf();