ABANDONED: Remove the default receive 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 165 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 voe_channel() const { return voe_channel_; }

[pthatcher1, 2015/10/02 02:33:31]

Can you rename this default_channel_id() or default_send_channel_id()?

[the sun, 2015/10/02 11:34:20]

Done.

   bool valid() const { return voe_channel_ != -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();
@@ skipping 42 matching lines @@
   void OnError(uint32 ssrc, int error);
 
   int GetReceiveChannelNum(uint32 ssrc) const;
   int GetSendChannelNum(uint32 ssrc) const;
 
  private:
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
   bool SetOptions(const AudioOptions& options);
+  bool SetRecvOptions(int channel_id);
   bool SetMaxSendBandwidth(int bps);
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetRecvRtpHeaderExtensions(
       const std::vector<RtpHeaderExtension>& extensions);
   bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer);
   bool MuteStream(uint32 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 ResetRecvCodecs(int channel);
   bool SetPlayout(int channel, bool playout);
-  static uint32 ParseSsrc(const void* data, size_t len, bool rtcp);
   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, 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);
   bool DeleteChannel(int channel);
-  bool InConferenceMode() const {
-    return options_.conference_mode.GetWithDefaultIfUnset(false);
-  }
   bool IsDefaultChannel(int channel_id) const {
     return channel_id == voe_channel();
   }
   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 ssrc);
   void RemoveAudioReceiveStream(uint32 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 voe_channel_;

[pthatcher1, 2015/10/02 02:33:31]

Can you rename this default_channel_id_ or default_send_channel_id_?

[the sun, 2015/10/02 11:34:20]

Done.

   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_;
 
+  uint32_t default_recv_ssrc_;
+  int default_recv_channel_id_;

[pthatcher1, 2015/10/02 02:33:31]

Why do we store the default_recv_channel_id_ when we can just call
GetRecvChannelNum at any time go get the same value?

[the sun, 2015/10/02 11:34:20]

You're right; I was using it as a flag anyway.

+
   // send_channels_ contains the channels which are being used for sending.
   // When the default channel (voe_channel) is used for sending, it is
   // contained in send_channels_, otherwise not.
   ChannelMap send_channels_;
   std::vector<RtpHeaderExtension> send_extensions_;
-  uint32 default_receive_ssrc_;
-  // Note the default channel (voe_channel()) can reside in both
-  // receive_channels_ and send_channels_ in non-conference mode and in that
-  // case it will only be there if a non-zero default_receive_ssrc_ is set.
-  ChannelMap receive_channels_;  // for multiple sources
+  ChannelMap receive_channels_;
   std::map<uint32, webrtc::AudioReceiveStream*> receive_streams_;
   std::map<uint32, 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_;
-
-  // 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_