Remove VoEAudioProcessing interface.

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 428 matching lines @@
         "Channel::SendData() failed to send data to RTP/RTCP module");
     return -1;
   }
 
   _lastLocalTimeStamp = timeStamp;
   _lastPayloadType = payloadType;
 
   return 0;
 }
 
-int32_t Channel::InFrameType(FrameType frame_type) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::InFrameType(frame_type=%d)", frame_type);
-
-  rtc::CritScope cs(&_callbackCritSect);
-  _sendFrameType = (frame_type == kAudioFrameSpeech);
-  return 0;
-}
-
 bool Channel::SendRtp(const uint8_t* data,
                       size_t len,
                       const PacketOptions& options) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
 
   rtc::CritScope cs(&_callbackCritSect);
 
   if (_transportPtr == NULL) {
     WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, _channelId),
@@ skipping 418 matching lines @@
       rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
       capture_start_rtp_time_stamp_(-1),
       capture_start_ntp_time_ms_(-1),
       _engineStatisticsPtr(NULL),
       _outputMixerPtr(NULL),
       _moduleProcessThreadPtr(NULL),
       _audioDeviceModulePtr(NULL),
       _voiceEngineObserverPtr(NULL),
       _callbackCritSectPtr(NULL),
       _transportPtr(NULL),
-      _sendFrameType(0),
       input_mute_(false),
       previous_frame_muted_(false),
       _outputGain(1.0f),
       _mixFileWithMicrophone(false),
       _lastLocalTimeStamp(0),
       _lastPayloadType(0),
       _includeAudioLevelIndication(false),
       transport_overhead_per_packet_(0),
       rtp_overhead_per_packet_(0),
       _outputSpeechType(AudioFrame::kNormalSpeech),
@@ skipping 112 matching lines @@
 
   // Ensure that RTCP is enabled by default for the created channel.
   // Note that, the module will keep generating RTCP until it is explicitly
   // 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
-  const bool fail = (audio_coding_->RegisterTransportCallback(this) == -1) ||
-                    (audio_coding_->RegisterVADCallback(this) == -1);
-
-  if (fail) {
+  if (audio_coding_->RegisterTransportCallback(this) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_INIT_CHANNEL, kTraceError,
         "Channel::Init() callbacks not registered");
     return -1;
   }
 
   // --- Register all supported codecs to the receiving side of the
   // RTP/RTCP module
 
   CodecInst codec;
@@ skipping 1244 matching lines @@
       _engineStatisticsPtr->SetLastError(
           VE_RTP_RTCP_MODULE_ERROR, kTraceError,
           "SetSendTelephoneEventPayloadType() failed to register send"
           "payload type");
       return -1;
     }
   }
   return 0;
 }
 
-int Channel::VoiceActivityIndicator(int& activity) {
-  activity = _sendFrameType;
-  return 0;
-}
-
 int Channel::SetLocalSSRC(unsigned int ssrc) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetLocalSSRC()");
   if (channel_state_.Get().sending) {
     _engineStatisticsPtr->SetLastError(VE_ALREADY_SENDING, kTraceError,
                                        "SetLocalSSRC() already sending");
     return -1;
   }
   _rtpRtcpModule->SetSSRC(ssrc);
   return 0;
@@ skipping 698 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