New flag is_screencast in VideoOptions.

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 234 matching lines @@
   }
 };
 
 // Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
 // Used to be flags, but that makes it hard to selectively apply options.
 // We are moving all of the setting of options to structs like this,
 // but some things currently still use flags.
 struct VideoOptions {
   void SetAll(const VideoOptions& change) {
     SetFrom(&video_noise_reduction, change.video_noise_reduction);
     SetFrom(&suspend_below_min_bitrate, change.suspend_below_min_bitrate);

[perkj_webrtc, 2016/02/21 18:09:06]

not this cl but is suspend_below_min_bitrate and screencast_min_bitrate_kbps
contraints from getusermedia? They should really but constraints or some kind of
settings on either rtpvideosender or peerconnection...  

[nisse-webrtc, 2016/02/22 07:52:34]

By the comment I added below, they're PeerConnection constraints. Not entirely
clear if they are per stream or not. So we'll need to sort that out later.

     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 &&
            suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
-           screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps;
+           screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps &&
+           is_screencast == o.is_screencast;
   }
 
   std::string ToString() const {
     std::ostringstream ost;
     ost << "VideoOptions {";
     ost << ToStringIfSet("noise reduction", video_noise_reduction);
     ost << ToStringIfSet("suspend below min bitrate",
                          suspend_below_min_bitrate);
     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
   // on to the codec options. Disabled by default.
   rtc::Optional<bool> video_noise_reduction;
   // 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
   // to VideoSendStream::Config::suspend_below_min_bitrate.
   rtc::Optional<bool> suspend_below_min_bitrate;
   // Force screencast to use a minimum bitrate. This flag comes from
   // the PeerConnection constraint 'googScreencastMinBitrate'. It is
   // copied to the encoder config by WebRtcVideoChannel2.
   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) {
     if (o) {
       *s = o;
     }
   }
 };
 
@@ skipping 806 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_