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

webrtc/media/engine/webrtcvoiceengine.h

 /*
  *  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 227 matching lines @@
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendCodecs(int channel, const webrtc::RtpParameters& rtp_parameters);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
   bool MuteStream(uint32_t ssrc, bool mute);
 
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
-  void ChangePlayout(bool playout);
   int CreateVoEChannel();
   bool DeleteVoEChannel(int channel);
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
   bool SetMaxSendBitrate(int bps);
   bool SetChannelSendParameters(int channel,
                                 const webrtc::RtpParameters& parameters);
   bool SetMaxSendBitrate(int channel, int bps);
   bool HasSendCodec() const {
     return send_codec_spec_.codec_inst.pltype != -1;
   }
   bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
   void SetupRecording();
 
   rtc::ThreadChecker worker_thread_checker_;
 
   WebRtcVoiceEngine* const engine_ = nullptr;
   std::vector<AudioCodec> send_codecs_;
   std::vector<AudioCodec> recv_codecs_;
   int max_send_bitrate_bps_ = 0;
   AudioOptions options_;
   rtc::Optional<int> dtmf_payload_type_;
-  bool desired_playout_ = false;
   bool recv_transport_cc_enabled_ = false;
   bool recv_nack_enabled_ = false;
   bool playout_ = false;
   bool send_ = false;
   webrtc::Call* const call_ = nullptr;
 
   // SSRC of unsignalled receive stream, or -1 if there isn't one.
   int64_t default_recv_ssrc_ = -1;
   // Volume for unsignalled stream, which may be set before the stream exists.
   double default_recv_volume_ = 1.0;
@@ skipping 12 matching lines @@
   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   SendCodecSpec send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_