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

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
 #include <string.h>
 
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/include/tick_util.h"
 
 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),
       _dtmfEventFirstPacketSent(false),
       _dtmfPayloadType(-1),
       _dtmfTimestamp(0),
       _dtmfKey(0),
       _dtmfLengthSamples(0),
       _dtmfLevel(0),
       _dtmfTimeLastSent(0),
@@ skipping 111 matching lines @@
 
 int32_t RTPSenderAudio::SendAudio(FrameType frameType,
                                   int8_t payloadType,
                                   uint32_t captureTimeStamp,
                                   const uint8_t* payloadData,
                                   size_t dataSize,
                                   const RTPFragmentationHeader* fragmentation) {
   // 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;
   uint8_t audio_level_dbov;
   int8_t dtmf_payload_type;
   uint16_t packet_size_samples;
   {
     CriticalSectionScoped cs(_sendAudioCritsect.get());
     red_payload_type = _REDPayloadType;
     audio_level_dbov = _audioLevel_dBov;
     dtmf_payload_type = _dtmfPayloadType;
     packet_size_samples = _packetSizeSamples;
   }
 
   // Check if we have pending DTMFs to send
   if (!_dtmfEventIsOn && PendingDTMF()) {
     int64_t delaySinceLastDTMF =
         _clock->TimeInMilliseconds() - _dtmfTimeLastSent;
 
     if (delaySinceLastDTMF > 100) {
       // New tone to play
       _dtmfTimestamp = captureTimeStamp;
       if (NextDTMF(&key, &dtmfLengthMS, &_dtmfLevel) >= 0) {
         _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
   if (_dtmfEventIsOn) {
     if (frameType == kEmptyFrame) {
       // kEmptyFrame is used to drive the DTMF when in CN mode
       // it can be triggered more frequently than we want to send the
       // DTMF packets.
       if (packet_size_samples > (captureTimeStamp - _dtmfTimestampLastSent)) {
         // not time to send yet
@@ skipping 248 matching lines @@
                          dtmfTimeStamp, "seqnum", _rtpSender->SequenceNumber());
     retVal = _rtpSender->SendToNetwork(
         dtmfbuffer, 4, 12, TickTime::MillisecondTimestamp(),
         kAllowRetransmission, RtpPacketSender::kHighPriority);
     sendCount--;
   } while (sendCount > 0 && retVal == 0);
 
   return retVal;
 }
 }  // namespace webrtc