Remove the default send channel in WVoE.

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 97 matching lines @@
   // Starts AEC dump using existing file.
   bool StartAecDump(rtc::PlatformFile file);
 
   // Starts recording an RtcEventLog using an existing file until 10 minutes
   // pass or the StopRtcEventLog function is called.
   bool StartRtcEventLog(rtc::PlatformFile file);
 
   // Stops recording the RtcEventLog.
   void StopRtcEventLog();
 
-  // Create a VoiceEngine Channel.
-  int CreateMediaVoiceChannel();
-
  private:
   void Construct();
   void ConstructCodecs();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
   // Every option that is "set" will be applied. Every option not "set" will be
   // ignored. This allows us to selectively turn on and off different options
   // easily at any time.
   bool ApplyOptions(const AudioOptions& options);
 
   // webrtc::TraceCallback:
   void Print(webrtc::TraceLevel level, const char* trace, int length) override;
 
   // webrtc::VoiceEngineObserver:
   void CallbackOnError(int channel_id, int errCode) override;
 
   // Given the device type, name, and id, find device id. Return true and
   // set the output parameter rtc_id if successful.
   bool FindWebRtcAudioDeviceId(
       bool is_input, const std::string& dev_name, int dev_id, int* rtc_id);
 
   void StartAecDump(const std::string& filename);
   void StopAecDump();
-  int CreateVoiceChannel(VoEWrapper* voe);
+  int CreateVoEChannel();
 
   static const int kDefaultLogSeverity = rtc::LS_WARNING;
 
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_ptr<VoEWrapper> voe_wrapper_;
   rtc::scoped_ptr<VoETraceWrapper> tracing_;
   // The external audio device manager
   webrtc::AudioDeviceModule* adm_;
   int log_filter_;
   std::string log_options_;
@@ skipping 23 matching lines @@
 // WebRtcVoiceMediaChannel is an implementation of VoiceMediaChannel that uses
 // WebRtc Voice Engine.
 class WebRtcVoiceMediaChannel : public VoiceMediaChannel,
                                 public webrtc::Transport {
  public:
   WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                           const AudioOptions& options,
                           webrtc::Call* call);
   ~WebRtcVoiceMediaChannel() override;
 
-  int default_send_channel_id() const { return default_send_channel_id_; }
-  bool valid() const { return default_send_channel_id_ != -1; }
   const AudioOptions& options() const { return options_; }
 
   bool SetSendParameters(const AudioSendParameters& params) override;
   bool SetRecvParameters(const AudioRecvParameters& params) override;
   bool SetPlayout(bool playout) override;
   bool PausePlayout();
   bool ResumePlayout();
   bool SetSend(SendFlags send) override;
   bool PauseSend();
   bool ResumeSend();
@@ skipping 58 matching lines @@
       const std::vector<RtpHeaderExtension>& extensions);
   bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer);
   bool MuteStream(uint32_t ssrc, bool mute);
 
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
   bool GetRedSendCodec(const AudioCodec& red_codec,
                        const std::vector<AudioCodec>& all_codecs,
                        webrtc::CodecInst* send_codec);
-  bool EnableRtcp(int channel);
   bool SetPlayout(int channel, bool playout);
   static Error WebRtcErrorToChannelError(int err_code);
 
   class WebRtcVoiceChannelRenderer;
   // Map of ssrc to WebRtcVoiceChannelRenderer object.  A new object of
   // WebRtcVoiceChannelRenderer will be created for every new stream and
   // will be destroyed when the stream goes away.
   typedef std::map<uint32_t, WebRtcVoiceChannelRenderer*> ChannelMap;
   typedef int (webrtc::VoERTP_RTCP::* ExtensionSetterFunction)(int, bool,
       unsigned char);
 
   void SetNack(int channel, bool nack_enabled);
-  void SetNack(const ChannelMap& channels, bool nack_enabled);
-  bool SetSendCodec(const webrtc::CodecInst& send_codec);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool ChangePlayout(bool playout);
   bool ChangeSend(SendFlags send);
   bool ChangeSend(int channel, SendFlags send);
-  void ConfigureSendChannel(int channel);
   bool ConfigureRecvChannel(int channel);
+  int CreateVoEChannel();
   bool DeleteChannel(int channel);
-  bool IsDefaultChannel(int channel_id) const {
-    return channel_id == default_send_channel_id_;
-  }
   bool IsDefaultRecvStream(uint32_t ssrc) {
     return default_recv_ssrc_ == static_cast<int64_t>(ssrc);
   }
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBitrateInternal(int bps);
 
   bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
                           const RtpHeaderExtension* extension);
   void RecreateAudioReceiveStreams();
   void AddAudioReceiveStream(uint32_t ssrc);
   void RemoveAudioReceiveStream(uint32_t ssrc);
   bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
 
   bool SetChannelRecvRtpHeaderExtensions(
     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 default_send_channel_id_;
   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_;
   bool typing_noise_detected_;
   SendFlags desired_send_;
   SendFlags send_;
   webrtc::Call* const call_;
 
   // 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;
+  // SSRC to use for RTCP receiver reports; default to 1 in case of no signaled
+  // send streams. See: https://code.google.com/p/webrtc/issues/detail?id=4740
+  uint32_t receiver_reports_ssrc_ = 1;
 
   // send_channels_ contains the channels which are being used for sending.
   // When the default channel (default_send_channel_id) is used for sending, it
   // is contained in send_channels_, otherwise not.
   ChannelMap send_channels_;
   std::vector<RtpHeaderExtension> send_extensions_;
   ChannelMap receive_channels_;
   std::map<uint32_t, webrtc::AudioReceiveStream*> receive_streams_;
   std::map<uint32_t, StreamParams> receive_stream_params_;
   // receive_channels_ can be read from WebRtc callback thread.  Access from
   // the WebRtc thread must be synchronized with edits on the worker thread.
   // Reads on the worker thread are ok.
   std::vector<RtpHeaderExtension> receive_extensions_;
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_