Removed the legacy behavior of stopping playout when setting new receive codecs.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1687 matching lines @@
     } else {
       new_codecs.push_back(codec);
     }
   }
   if (new_codecs.empty()) {
     // There are no new codecs to configure. Already configured codecs are
     // never removed.
     return true;
   }
 
-  if (playout_) {
-    // Receive codecs can not be changed while playing. So we temporarily
-    // pause playout.
-    ChangePlayout(false);
-  }
-
   bool result = true;
   for (const AudioCodec& codec : new_codecs) {
     webrtc::CodecInst voe_codec = {0};
     if (WebRtcVoiceEngine::ToCodecInst(codec, &voe_codec)) {
       LOG(LS_INFO) << ToString(codec);
       voe_codec.pltype = codec.id;
       for (const auto& ch : recv_streams_) {
         if (engine()->voe()->codec()->SetRecPayloadType(
                 ch.second->channel(), voe_codec) == -1) {
           LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
                       ToString(voe_codec));
           result = false;
         }
       }
     } else {
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       result = false;
       break;
     }
   }
   if (result) {
     recv_codecs_ = codecs;
   }
 
-  if (desired_playout_ && !playout_) {
-    ChangePlayout(desired_playout_);
-  }
   return result;
 }
 
 // Utility function called from SetSendParameters() to extract current send
 // codec settings from the given list of codecs (originally from SDP). Both send
 // and receive streams may be reconfigured based on the new settings.
 bool WebRtcVoiceMediaChannel::SetSendCodecs(
     const std::vector<AudioCodec>& codecs) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   // TODO(solenberg): Validate input - that payload types don't overlap, are
@@ skipping 234 matching lines @@
   }
 
   if (engine()->voe()->codec()->SetSendCodec(channel, send_codec) == -1) {
     LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec));
     return false;
   }
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
-  desired_playout_ = playout;
-  return ChangePlayout(desired_playout_);
-}
-
-void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
-  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
+  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetPlayout");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   if (playout_ == playout) {
     return;
   }
 
   for (const auto& kv : recv_streams_) {
     kv.second->SetPlayout(playout);
   }
   playout_ = playout;
 }
@@ skipping 653 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE