s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine

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 90 matching lines @@
 
   // Video-specific config.
   struct Video {
     // Enable WebRTC CPU Overuse Detection. This flag comes from the
     // PeerConnection constraint 'googCpuOveruseDetection'.
     bool enable_cpu_overuse_detection = true;
 
     // Enable WebRTC suspension of video. No video frames will be sent
     // when the bitrate is below the configured minimum bitrate. This
     // flag comes from the PeerConnection constraint
-    // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel2 copies it
+    // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
     // to VideoSendStream::Config::suspend_below_min_bitrate.
     bool suspend_below_min_bitrate = false;
 
     // Set to true if the renderer has an algorithm of frame selection.
     // If the value is true, then WebRTC will hand over a frame as soon as
     // possible without delay, and rendering smoothness is completely the duty
     // of the renderer;
     // If the value is false, then WebRTC is responsible to delay frame release
     // in order to increase rendering smoothness.
     //
     // This flag comes from PeerConnection's RtcConfiguration, but is
     // currently only set by the command line flag
     // 'disable-rtc-smoothness-algorithm'.
-    // WebRtcVideoChannel2::AddRecvStream copies it to the created
+    // WebRtcVideoChannel::AddRecvStream copies it to the created
     // WebRtcVideoReceiveStream, where it is returned by the
     // SmoothsRenderedFrames method. This method is used by the
     // VideoReceiveStream, where the value is passed on to the
     // IncomingVideoStream constructor.
     bool disable_prerenderer_smoothing = false;
 
     // Enables periodic bandwidth probing in application-limited region.
     bool periodic_alr_bandwidth_probing = false;
   } video;
 
@@ skipping 200 matching lines @@
     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();
   }
 
   // Enable denoising? This flag comes from the getUserMedia
-  // constraint 'googNoiseReduction', and WebRtcVideoEngine2 passes it
+  // constraint 'googNoiseReduction', and WebRtcVideoEngine passes it
   // on to the codec options. Disabled by default.
   rtc::Optional<bool> video_noise_reduction;
   // Force screencast to use a minimum bitrate. This flag comes from
   // the PeerConnection constraint 'googScreencastMinBitrate'. It is
-  // copied to the encoder config by WebRtcVideoChannel2.
+  // copied to the encoder config by WebRtcVideoChannel.
   rtc::Optional<int> screencast_min_bitrate_kbps;
   // Set by screencast sources. Implies selection of encoding settings
   // suitable for screencast. Most likely not the right way to do
   // things, e.g., screencast of a text document and screencast of a
   // youtube video have different needs.
   rtc::Optional<bool> is_screencast;
 
  private:
   template <typename T>
   static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
@@ skipping 652 matching lines @@
       std::unique_ptr<webrtc::AudioSinkInterface> sink) = 0;
 };
 
 // TODO(deadbeef): Rename to VideoSenderParameters, since they're intended to
 // encapsulate all the parameters needed for a video RtpSender.
 struct VideoSendParameters : RtpSendParameters<VideoCodec> {
   // Use conference mode? This flag comes from the remote
   // description's SDP line 'a=x-google-flag:conference', copied over
   // by VideoChannel::SetRemoteContent_w, and ultimately used by
   // conference mode screencast logic in
-  // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
+  // WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig.
   // The special screencast behaviour is disabled by default.
   bool conference_mode = false;
 };
 
 // TODO(deadbeef): Rename to VideoReceiverParameters, since they're intended to
 // encapsulate all the parameters needed for a video RtpReceiver.
 struct VideoRecvParameters : RtpParameters<VideoCodec> {
 };
 
 class VideoMediaChannel : public MediaChannel {
@@ skipping 180 matching lines @@
                    const char*,
                    size_t> SignalDataReceived;
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_