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

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 8992425b537b0e32d329a470d58fe50d2619171d..eeecc64acc20b131c960ada57e02f7573e1ce7eb 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -628,17 +628,16 @@ Channel::NeededFrequency(int32_t id)
     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,
@@ -713,86 +712,92 @@ Channel::RecordFileEnded(int32_t id)
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
-                 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())),
-    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
-    _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()),
-    capture_start_rtp_time_stamp_(-1),
-    capture_start_ntp_time_ms_(-1),
-    _engineStatisticsPtr(NULL),
-    _outputMixerPtr(NULL),
-    _transmitMixerPtr(NULL),
-    _moduleProcessThreadPtr(NULL),
-    _audioDeviceModulePtr(NULL),
-    _voiceEngineObserverPtr(NULL),
-    _callbackCritSectPtr(NULL),
-    _transportPtr(NULL),
-    _rxVadObserverPtr(NULL),
-    _oldVadDecision(-1),
-    _sendFrameType(0),
-    _externalMixing(false),
-    _mixFileWithMicrophone(false),
-    _mute(false),
-    _panLeft(1.0f),
-    _panRight(1.0f),
-    _outputGain(1.0f),
-    _playOutbandDtmfEvent(false),
-    _playInbandDtmfEvent(false),
-    _lastLocalTimeStamp(0),
-    _lastPayloadType(0),
-    _includeAudioLevelIndication(false),
-    _outputSpeechType(AudioFrame::kNormalSpeech),
-    _average_jitter_buffer_delay_us(0),
-    least_required_delay_ms_(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))
-{
+                 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())),
+      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
+      _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()),
+      capture_start_rtp_time_stamp_(-1),
+      capture_start_ntp_time_ms_(-1),
+      _engineStatisticsPtr(NULL),
+      _outputMixerPtr(NULL),
+      _transmitMixerPtr(NULL),
+      _moduleProcessThreadPtr(NULL),
+      _audioDeviceModulePtr(NULL),
+      _voiceEngineObserverPtr(NULL),
+      _callbackCritSectPtr(NULL),
+      _transportPtr(NULL),
+      _rxVadObserverPtr(NULL),
+      _oldVadDecision(-1),
+      _sendFrameType(0),
+      _externalMixing(false),
+      _mixFileWithMicrophone(false),
+      _mute(false),
+      _panLeft(1.0f),
+      _panRight(1.0f),
+      _outputGain(1.0f),
+      _playOutbandDtmfEvent(false),
+      _playInbandDtmfEvent(false),
+      _lastLocalTimeStamp(0),
+      _lastPayloadType(0),
+      _includeAudioLevelIndication(false),
+      _outputSpeechType(AudioFrame::kNormalSpeech),
+      _average_jitter_buffer_delay_us(0),
+      least_required_delay_ms_(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)) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::Channel() - ctor");
     AudioCodingModule::Config acm_config;
@@ -805,6 +810,7 @@ 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();