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

webrtc/media/engine/webrtcvoiceengine.h

Index: webrtc/media/engine/webrtcvoiceengine.h
diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h
index 27ae593cf15e66cc82fc935696075f80ca04883f..71bd874347d9103547adb854a49de98457f53db6 100644
--- a/webrtc/media/engine/webrtcvoiceengine.h
+++ b/webrtc/media/engine/webrtcvoiceengine.h
@@ -245,7 +245,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   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) {
@@ -269,7 +268,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
   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;