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 t
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(); |