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

webrtc/voice_engine/channel.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.
  */
 
@@ skipping 335 matching lines @@
     std::string transport_name =
         _externalTransport ? "external transport" : "WebRtc sockets";
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::SendRtcp() transmission using %s failed",
                  transport_name.c_str());
     return false;
   }
   return true;
 }
 
-void Channel::OnPlayTelephoneEvent(uint8_t event,
-                                   uint16_t lengthMs,
-                                   uint8_t volume) {
-  WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::OnPlayTelephoneEvent(event=%u, lengthMs=%u,"
-               " volume=%u)",
-               event, lengthMs, volume);
-}
-
 void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::OnIncomingSSRCChanged(SSRC=%d)", ssrc);
 
   // Update ssrc so that NTP for AV sync can be updated.
   _rtpRtcpModule->SetRemoteSSRC(ssrc);
 }
 
 void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
@@ skipping 349 matching lines @@
     : _instanceId(instanceId),
       _channelId(channelId),
       event_log_(event_log),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
-                                           this,
+                                           nullptr,
                                            this,
                                            this,
                                            rtp_payload_registry_.get())),
       telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
       _outputAudioLevel(),
       _externalTransport(false),
       _inputFilePlayerPtr(NULL),
       _outputFilePlayerPtr(NULL),
       _outputFileRecorderPtr(NULL),
       // Avoid conflict with other channels by adding 1024 - 1026,
@@ skipping 64 matching lines @@
   }
   acm_config.neteq_config.enable_fast_accelerate =
       config.Get<NetEqFastAccelerate>().enabled;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
   configuration.audio = true;
   configuration.outgoing_transport = this;
-  configuration.audio_messages = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
   configuration.bandwidth_callback = rtcp_observer_.get();
   if (pacing_enabled_) {
     configuration.paced_sender = rtp_packet_sender_proxy_.get();
     configuration.transport_sequence_number_allocator =
         seq_num_allocator_proxy_.get();
     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
   }
   configuration.event_log = event_log;
 
@@ skipping 1364 matching lines @@
 int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SendTelephoneEventOutband(...)");
   RTC_DCHECK_LE(0, event);
   RTC_DCHECK_GE(255, event);
   RTC_DCHECK_LE(0, duration_ms);
   RTC_DCHECK_GE(65535, duration_ms);
   if (!Sending()) {
     return -1;
   }
-
   if (_rtpRtcpModule->SendTelephoneEventOutband(
       event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
     _engineStatisticsPtr->SetLastError(
         VE_SEND_DTMF_FAILED, kTraceWarning,
         "SendTelephoneEventOutband() failed to send event");
     return -1;
   }
   return 0;
 }
 
@@ skipping 1325 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc