Added support for logging the SSRC corresponding to AudioPlayout events.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 432 matching lines @@
   if (header.payload_type_frequency < 0)
     return false;
   return ReceivePacket(rtp_packet, rtp_packet_length, header, false);
 }
 
 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)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice,
                      VoEId(_instanceId,_channelId),
                      "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
         // In all likelihood, the audio in this frame is garbage. We return an
         // error so that the audio mixer module doesn't add it to the mix. As
         // a result, it won't be played out and the actions skipped here are
@@ skipping 249 matching lines @@
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
                  RtcEventLog* const event_log,
                  const Config& config)
   : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     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))),
     rtp_receive_statistics_(
         ReceiveStatistics::Create(Clock::GetRealTimeClock())),
     rtp_receiver_(
         RtpReceiver::CreateAudioReceiver(VoEModuleId(instanceId, channelId),
                                          Clock::GetRealTimeClock(),
                                          this,
                                          this,
@@ skipping 70 matching lines @@
     AudioCodingModule::Config acm_config;
     acm_config.id = VoEModuleId(instanceId, channelId);
     if (config.Get<NetEqCapacityConfig>().enabled) {
       // Clamping the buffer capacity at 20 packets. While going lower will
       // probably work, it makes little sense.
       acm_config.neteq_config.max_packets_in_buffer =
           std::max(20, config.Get<NetEqCapacityConfig>().capacity);
     }
     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();
     _inbandDtmfGenerator.Init();
     _outputAudioLevel.Clear();
 
     RtpRtcp::Configuration configuration;
     configuration.id = VoEModuleId(instanceId, channelId);
     configuration.audio = true;
     configuration.outgoing_transport = this;
@@ skipping 3329 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc