Index: webrtc/voice_engine/channel.cc |
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc |
index a9fe6fdfa52f8902259f4f146491f3355fc88ce2..1d73db6d396a29ab1bc3ca12beee0c8ce1106ae0 100644 |
--- a/webrtc/voice_engine/channel.cc |
+++ b/webrtc/voice_engine/channel.cc |
@@ -446,15 +446,6 @@ int32_t Channel::SendData(FrameType frameType, |
return 0; |
} |
-int32_t Channel::InFrameType(FrameType frame_type) { |
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
- "Channel::InFrameType(frame_type=%d)", frame_type); |
- |
- rtc::CritScope cs(&_callbackCritSect); |
- _sendFrameType = (frame_type == kAudioFrameSpeech); |
- return 0; |
-} |
- |
bool Channel::SendRtp(const uint8_t* data, |
size_t len, |
const PacketOptions& options) { |
@@ -893,7 +884,6 @@ Channel::Channel(int32_t channelId, |
_voiceEngineObserverPtr(NULL), |
_callbackCritSectPtr(NULL), |
_transportPtr(NULL), |
- _sendFrameType(0), |
input_mute_(false), |
previous_frame_muted_(false), |
_outputGain(1.0f), |
@@ -1026,10 +1016,7 @@ int32_t Channel::Init() { |
// RTCP is enabled by default. |
_rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound); |
// --- Register all permanent callbacks |
- const bool fail = (audio_coding_->RegisterTransportCallback(this) == -1) || |
- (audio_coding_->RegisterVADCallback(this) == -1); |
- |
- if (fail) { |
+ if (audio_coding_->RegisterTransportCallback(this) == -1) { |
_engineStatisticsPtr->SetLastError( |
VE_CANNOT_INIT_CHANNEL, kTraceError, |
"Channel::Init() callbacks not registered"); |
@@ -2294,11 +2281,6 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type, |
return 0; |
} |
-int Channel::VoiceActivityIndicator(int& activity) { |
- activity = _sendFrameType; |
- return 0; |
-} |
- |
int Channel::SetLocalSSRC(unsigned int ssrc) { |
WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId), |
"Channel::SetLocalSSRC()"); |