Create AudioSendStreams 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 154 matching lines @@
   bool initialized_;
   AudioOptions options_;
 
   // Cache received extended_filter_aec, delay_agnostic_aec and experimental_ns
   // values, and apply them in case they are missing in the audio options. We
   // need to do this because SetExtraOptions() will revert to defaults for
   // options which are not provided.
   Settable<bool> extended_filter_aec_;
   Settable<bool> delay_agnostic_aec_;
   Settable<bool> experimental_ns_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcVoiceEngine);
 };
 
 // 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);
@@ skipping 73 matching lines @@
 
   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 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);
   bool SetSendCodec(int channel, const webrtc::CodecInst& send_codec);
   bool ChangePlayout(bool playout);
   bool ChangeSend(SendFlags send);
   bool ChangeSend(int channel, SendFlags send);
   bool ConfigureRecvChannel(int channel);
   int CreateVoEChannel();
@@ skipping 37 matching lines @@
   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_;
+  class WebRtcAudioSendStream;
+  std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
   std::vector<RtpHeaderExtension> send_extensions_;
-  ChannelMap receive_channels_;
+
+  class WebRtcAudioReceiveStream;
+  std::map<uint32_t, WebRtcAudioReceiveStream*> 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_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_