Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index a8345435f90e7d3c65ccb2e5bae2a8ad5bfa1c99..1bfacbe5532fde209b5f7b97b685ae1c5dd2c2ea 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -575,9 +575,8 @@ bool Channel::OnRecoveredPacket(const uint8_t* rtp_packet, |
MixerParticipant::AudioFrameInfo Channel::GetAudioFrameWithMuted( |
int32_t id, |
AudioFrame* audioFrame) { |
- unsigned int ssrc; |
- RTC_CHECK_EQ(GetLocalSSRC(ssrc), 0); |
- event_log_proxy_->LogAudioPlayout(ssrc); |
+ |
+ event_log_proxy_->LogAudioPlayout(0); |
the sun
2017/02/16 13:52:32
Hang on - RtcEventLog will have the same SSRC for
terelius
2017/02/16 13:58:05
I've never used the SSRC for audio playouts. ivoc@
nisse-webrtc
2017/02/16 14:01:59
Ooops. I misunderstood the GetLocalSSRC call compl
terelius
2017/02/16 14:04:42
If the ssrc is no longer considered useful, please
ivoc
2017/02/16 14:27:06
The SSRC was added in https://codereview.webrtc.or
nisse-webrtc
2017/02/16 14:29:22
But if it's a received stream, we should be using
ivoc
2017/02/16 14:35:38
We only used it to tell the different channels apa
nisse-webrtc
2017/02/17 09:59:47
Undid this change, and then changed GetLocalSsrc t
|
// Get 10ms raw PCM data from the ACM (mixer limits output frequency) |
bool muted; |
if (audio_coding_->PlayoutData10Ms(audioFrame->sample_rate_hz_, audioFrame, |