| 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()");
|
|
|