Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call.

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 610 matching lines @@
             if(_outputFilePlayerPtr->Frequency()>highestNeeded)
             {
                 highestNeeded=_outputFilePlayerPtr->Frequency();
             }
         }
     }
 
     return(highestNeeded);
 }
 
-int32_t
-Channel::CreateChannel(Channel*& channel,
-                       int32_t channelId,
-                       uint32_t instanceId,
-                       const Config& config)
-{
+int32_t Channel::CreateChannel(Channel*& channel,
+                               int32_t channelId,
+                               uint32_t instanceId,
+                               RtcEventLog* const event_log,
+                               const Config& config) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
                  "Channel::CreateChannel(channelId=%d, instanceId=%d)",
         channelId, instanceId);
 
-    channel = new Channel(channelId, instanceId, config);
+    channel = new Channel(channelId, instanceId, event_log, config);
     if (channel == NULL)
     {
         WEBRTC_TRACE(kTraceMemory, kTraceVoice,
                      VoEId(instanceId,channelId),
                      "Channel::CreateChannel() unable to allocate memory for"
                      " channel");
         return -1;
     }
     return 0;
 }
@@ skipping 54 matching lines @@
 
     _outputFileRecording = false;
     WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
                  VoEId(_instanceId,_channelId),
                  "Channel::RecordFileEnded() => output file recorder module is"
                  " shutdown");
 }
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
-                 const Config& config) :
-    _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
+                 RtcEventLog* const event_log,
+                 const Config& config)
+  : _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
     volume_settings_critsect_(*CriticalSectionWrapper::CreateCriticalSection()),
     _instanceId(instanceId),
     _channelId(channelId),
     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, this, rtp_payload_registry_.get())),
+    rtp_receive_statistics_(
+        ReceiveStatistics::Create(Clock::GetRealTimeClock())),
+    rtp_receiver_(
+        RtpReceiver::CreateAudioReceiver(VoEModuleId(instanceId, channelId),
+                                         Clock::GetRealTimeClock(),
+                                         this,
+                                         this,
+                                         this,
+                                         rtp_payload_registry_.get())),
     telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
     _outputAudioLevel(),
     _externalTransport(false),
     _inputFilePlayerPtr(NULL),
     _outputFilePlayerPtr(NULL),
     _outputFileRecorderPtr(NULL),
     // Avoid conflict with other channels by adding 1024 - 1026,
     // won't use as much as 1024 channels.
     _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
     _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
     _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
     _outputFileRecording(false),
     _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
     _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
     _outputExternalMedia(false),
     _inputExternalMediaCallbackPtr(NULL),
     _outputExternalMediaCallbackPtr(NULL),
-    _timeStamp(0), // This is just an offset, RTP module will add it's own random offset
+    _timeStamp(0),  // This is just an offset, RTP module will add it's own
+                    // random offset
     _sendTelephoneEventPayloadType(106),
     ntp_estimator_(Clock::GetRealTimeClock()),
     jitter_buffer_playout_timestamp_(0),
     playout_timestamp_rtp_(0),
     playout_timestamp_rtcp_(0),
     playout_delay_ms_(0),
     _numberOfDiscardedPackets(0),
     send_sequence_number_(0),
     ts_stats_lock_(CriticalSectionWrapper::CreateCriticalSection()),
     rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
@@ skipping 26 matching lines @@
     _average_jitter_buffer_delay_us(0),
     _previousTimestamp(0),
     _recPacketDelayMs(20),
     _RxVadDetection(false),
     _rxAgcIsEnabled(false),
     _rxNsIsEnabled(false),
     restored_packet_in_use_(false),
     rtcp_observer_(new VoERtcpObserver(this)),
     network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
     assoc_send_channel_lock_(CriticalSectionWrapper::CreateCriticalSection()),
-    associate_send_channel_(ChannelOwner(nullptr))
-{
+    associate_send_channel_(ChannelOwner(nullptr)) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::Channel() - ctor");
     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