Remove voe::Channel::StopReceive() and associated logic.

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 1204 matching lines @@
   if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
         "StartSend() RTP/RTCP failed to stop sending");
   }
   _rtpRtcpModule->SetSendingMediaStatus(false);
 
   return 0;
 }
 
-int32_t Channel::StartReceiving() {
+void Channel::ResetDiscardedPacketCount() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::StartReceiving()");
-  if (channel_state_.Get().receiving) {
-    return 0;
-  }
-  channel_state_.SetReceiving(true);
+               "Channel::ResetDiscardedPacketCount()");
   _numberOfDiscardedPackets = 0;
-  return 0;
-}
-
-int32_t Channel::StopReceiving() {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::StopReceiving()");
-  if (!channel_state_.Get().receiving) {
-    return 0;
-  }
-
-  channel_state_.SetReceiving(false);
-  return 0;
 }
 
 int32_t Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RegisterVoiceEngineObserver()");
   rtc::CritScope cs(&_callbackCritSect);
 
   if (_voiceEngineObserverPtr) {
     _engineStatisticsPtr->SetLastError(
         VE_INVALID_OPERATION, kTraceError,
@@ skipping 108 matching lines @@
 int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRecPayloadType()");
 
   if (channel_state_.Get().playing) {
     _engineStatisticsPtr->SetLastError(
         VE_ALREADY_PLAYING, kTraceError,
         "SetRecPayloadType() unable to set PT while playing");
     return -1;
   }
-  if (channel_state_.Get().receiving) {
-    _engineStatisticsPtr->SetLastError(
-        VE_ALREADY_LISTENING, kTraceError,
-        "SetRecPayloadType() unable to set PT while listening");
-    return -1;
-  }
 
   if (codec.pltype == -1) {
     // De-register the selected codec (RTP/RTCP module and ACM)
 
     int8_t pltype(-1);
     CodecInst rxCodec = codec;
 
     // Get payload type for the given codec
     rtp_payload_registry_->ReceivePayloadType(
         rxCodec.plname, rxCodec.plfreq, rxCodec.channels,
@@ skipping 1876 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