Combining SetVideoSend and SetSource into one method.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 260 matching lines @@
     SetFrom(&video_noise_reduction, change.video_noise_reduction);
     SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
     SetFrom(&is_screencast, change.is_screencast);
   }
 
   bool operator==(const VideoOptions& o) const {
     return video_noise_reduction == o.video_noise_reduction &&
            screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps &&
            is_screencast == o.is_screencast;
   }
+  bool operator!=(const VideoOptions& o) const { return !(*this == o); }
 
   std::string ToString() const {
     std::ostringstream ost;
     ost << "VideoOptions {";
     ost << ToStringIfSet("noise reduction", video_noise_reduction);
     ost << ToStringIfSet("screencast min bitrate kbps",
                          screencast_min_bitrate_kbps);
     ost << ToStringIfSet("is_screencast ", is_screencast);
     ost << "}";
     return ost.str();
@@ skipping 688 matching lines @@
   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
   virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
   virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const = 0;
   virtual bool SetRtpParameters(uint32_t ssrc,
                                 const webrtc::RtpParameters& parameters) = 0;
   // Gets the currently set codecs/payload types to be used for outgoing media.
   virtual bool GetSendCodec(VideoCodec* send_codec) = 0;
   // Starts or stops transmission (and potentially capture) of local video.
   virtual bool SetSend(bool send) = 0;
   // Configure stream for sending.
+  // |ssrc| must be non-zero.
   virtual bool SetVideoSend(uint32_t ssrc,
                             bool enable,
-                            const VideoOptions* options) = 0;
+                            const VideoOptions* options,
+                            VideoCapturer* capturer) = 0;
   // Sets the sink object to be used for the specified stream.
   // If SSRC is 0, the renderer is used for the 'default' stream.
   virtual bool SetSink(uint32_t ssrc,
                        rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0;
-  // If |ssrc| is 0, replace the default capturer (engine capturer) with
-  // |capturer|. If |ssrc| is non zero create a new stream with |ssrc| as SSRC.
-  virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) = 0;
   // Gets quality stats for the channel.
   virtual bool GetStats(VideoMediaInfo* info) = 0;
 };
 
 enum DataMessageType {
   // Chrome-Internal use only.  See SctpDataMediaChannel for the actual PPID
   // values.
   DMT_NONE = 0,
   DMT_CONTROL = 1,
   DMT_BINARY = 2,
@@ skipping 106 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_