| Index: webrtc/voice_engine/channel.cc
|
| diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
|
| index 582bde5f26216f645340ef2102d87e485ff2fafb..7895e9b9e142a283c311730d953f03e06687c2db 100644
|
| --- a/webrtc/voice_engine/channel.cc
|
| +++ b/webrtc/voice_engine/channel.cc
|
| @@ -809,7 +809,6 @@ Channel::Channel(int32_t channelId,
|
| this,
|
| this,
|
| rtp_payload_registry_.get())),
|
| - telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
|
| _outputAudioLevel(),
|
| _externalTransport(false),
|
| // Avoid conflict with other channels by adding 1024 - 1026,
|
| @@ -979,7 +978,6 @@ int32_t Channel::Init() {
|
| // disabled by the user.
|
| // After StopListen (when no sockets exists), RTCP packets will no longer
|
| // be transmitted since the Transport object will then be invalid.
|
| - telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
|
| // RTCP is enabled by default.
|
| _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
|
| // --- Register all permanent callbacks
|
|
|