Delete telephone-event handling from RTPReceiverAudio.

Bug: webrtc:7135
Change-Id: Ic8b96f44ba25ff9265570dd43d3c76ed0177abfb
Reviewed-on: https://webrtc-review.googlesource.com/91125
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24172}

2 files changed, 0 insertions, 4 deletions

diff --git a/audio/channel.cc b/audio/channel.cc
index 297c11142e..971d68b831 100644
--- a/audio/channel.cc
+++ b/audio/channel.cc
@@ -529,7 +529,6 @@ Channel::Channel(ProcessThread* module_process_thread,
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
                                            this,
                                            rtp_payload_registry_.get())),
-      telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
       _outputAudioLevel(),
       _timeStamp(0),  // This is just an offset, RTP module will add it's own
                       // random offset
@@ -616,7 +615,6 @@ void 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);
 
diff --git a/audio/channel.h b/audio/channel.h
index 8d1a26b20e..9816e2928f 100644
--- a/audio/channel.h
+++ b/audio/channel.h
@@ -56,7 +56,6 @@ class RTPReceiverAudio;
 class RtpPacketReceived;
 class RtpRtcp;
 class RtpTransportControllerSendInterface;
-class TelephoneEventHandler;
 
 struct SenderInfo;
 
@@ -342,7 +341,6 @@ class Channel
   std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
-  TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
   AudioSinkInterface* audio_sink_ = nullptr;