Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within …

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 35 matching lines @@
 #include "webrtc/call.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 
 namespace webrtc {
 class VideoEngine;
 }
 
 namespace cricket {
 
-// WebRtcSoundclipStream is an adapter object that allows a memory stream to be
-// passed into WebRtc, and support looping.
-class WebRtcSoundclipStream : public webrtc::InStream {
- public:
-  WebRtcSoundclipStream(const char* buf, size_t len)
-      : mem_(buf, len), loop_(true) {
-  }
-  void set_loop(bool loop) { loop_ = loop; }
-
-  int Read(void* buf, size_t len) override;
-  int Rewind() override;
-
- private:
-  rtc::MemoryStream mem_;
-  bool loop_;
-};
-
 // WebRtcMonitorStream is used to monitor a stream coming from WebRtc.
 // For now we just dump the data.
 class WebRtcMonitorStream : public webrtc::OutStream {
   bool Write(const void* buf, size_t len) override { return true; }
 };
 
 class AudioDeviceModule;
 class AudioRenderer;
 class VoETraceWrapper;
 class VoEWrapper;
@@ skipping 226 matching lines @@
   bool GetActiveStreams(AudioInfo::StreamList* actives) override;
   int GetOutputLevel() override;
   int GetTimeSinceLastTyping() override;
   void SetTypingDetectionParameters(int time_window,
                                     int cost_per_typing,
                                     int reporting_threshold,
                                     int penalty_decay,
                                     int type_event_delay) override;
   bool SetOutputScaling(uint32 ssrc, double left, double right) override;
 
-  bool SetRingbackTone(const char* buf, int len) override;
-  bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) override;
   bool CanInsertDtmf() override;
   bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) override;
 
   void OnPacketReceived(rtc::Buffer* packet,
                         const rtc::PacketTime& packet_time) override;
   void OnRtcpReceived(rtc::Buffer* packet,
                       const rtc::PacketTime& packet_time) override;
   void OnReadyToSend(bool ready) override {}
   bool GetStats(VoiceMediaInfo* info) override;
   // Gets last reported error from WebRtc voice engine.  This should be only
@@ skipping 83 matching lines @@
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
   bool SetChannelSendRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
 
   rtc::ThreadChecker thread_checker_;
 
   WebRtcVoiceEngine* const engine_;
   const int voe_channel_;
-  rtc::scoped_ptr<WebRtcSoundclipStream> ringback_tone_;
-  std::set<int> ringback_channels_;  // channels playing ringback
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
   rtc::scoped_ptr<webrtc::CodecInst> send_codec_;
   bool send_bitrate_setting_;
   int send_bitrate_bps_;
   AudioOptions options_;
   bool dtmf_allowed_;
   bool desired_playout_;
   bool nack_enabled_;
   bool playout_;
@@ skipping 21 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_