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 |