Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 2aa4961cdc77d98490c607ed23945617228bf760..804294ac5408547e9e347bd5a743348e9f26dc67 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -21,12 +21,9 @@ namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
-RTPSenderAudio::RTPSenderAudio(Clock* clock,
-                               RTPSender* rtpSender,
-                               RtpAudioFeedback* audio_feedback)
+RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtpSender)
     : _clock(clock),
       _rtpSender(rtpSender),
-      _audioFeedback(audio_feedback),
       _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
       _packetSizeSamples(160),
       _dtmfEventIsOn(false),
@@ -158,7 +155,6 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
   // TODO(pwestin) Breakup function in smaller functions.
   size_t payloadSize = dataSize;
   size_t maxPayloadLength = _rtpSender->MaxPayloadLength();
-  bool dtmfToneStarted = false;
   uint16_t dtmfLengthMS = 0;
   uint8_t key = 0;
   int red_payload_type;
@@ -185,15 +181,10 @@ int32_t RTPSenderAudio::SendAudio(FrameType frameType,
         _dtmfEventFirstPacketSent = false;
         _dtmfKey = key;
         _dtmfLengthSamples = (kDtmfFrequencyHz / 1000) * dtmfLengthMS;
-        dtmfToneStarted = true;
         _dtmfEventIsOn = true;
       }
     }
   }
-  if (dtmfToneStarted) {
-    if (_audioFeedback)
-      _audioFeedback->OnPlayTelephoneEvent(key, dtmfLengthMS, _dtmfLevel);
-  }
 
   // A source MAY send events and coded audio packets for the same time
   // but we don't support it