Introduce struct MediaConfig, with construction-time settings.

webrtc/media/base/mediachannel.h

Index: webrtc/media/base/mediachannel.h
diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
index 30d4a45ff21debbf7caa4ba80b17873046193bf4..99cbac30568d22f37162033fed3d418fdfec6a79 100644
--- a/webrtc/media/base/mediachannel.h
+++ b/webrtc/media/base/mediachannel.h
@@ -45,6 +45,8 @@
 #include "webrtc/media/base/videosinkinterface.h"
 // TODO(juberti): re-evaluate this include
 #include "talk/session/media/audiomonitor.h"
+// TODO(nisse): Improper include? Move declaration of MediaConfig elsewhere?
+#include "talk/app/webrtc/mediacontroller.h"

[pthatcher1, 2016/02/05 17:11:30]

We could put MediaConfig here in mediachannel.h or in its own mediaconfig.h. 
Putting it here seems wrong.

[nisse-webrtc, 2016/02/08 09:02:32]

mediachannel.h sounds reasonable to me. And then also in the cricket namespace?

 
 namespace rtc {
 class Buffer;
@@ -125,7 +127,6 @@ struct AudioOptions {
     SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
     SetFrom(&recording_sample_rate, change.recording_sample_rate);
     SetFrom(&playout_sample_rate, change.playout_sample_rate);
-    SetFrom(&dscp, change.dscp);
     SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
   }
 
@@ -152,7 +153,6 @@ struct AudioOptions {
         tx_agc_limiter == o.tx_agc_limiter &&
         recording_sample_rate == o.recording_sample_rate &&
         playout_sample_rate == o.playout_sample_rate &&
-        dscp == o.dscp &&
         combined_audio_video_bwe == o.combined_audio_video_bwe;
   }
 
@@ -183,7 +183,6 @@ struct AudioOptions {
     ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
     ost << ToStringIfSet("recording_sample_rate", recording_sample_rate);
     ost << ToStringIfSet("playout_sample_rate", playout_sample_rate);
-    ost << ToStringIfSet("dscp", dscp);
     ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
     ost << "}";
     return ost.str();
@@ -220,8 +219,6 @@ struct AudioOptions {
   rtc::Optional<bool> tx_agc_limiter;
   rtc::Optional<uint32_t> recording_sample_rate;
   rtc::Optional<uint32_t> playout_sample_rate;
-  // Set DSCP value for packet sent from audio channel.
-  rtc::Optional<bool> dscp;
   // Enable combined audio+bandwidth BWE.
   rtc::Optional<bool> combined_audio_video_bwe;
 
@@ -241,32 +238,23 @@ struct AudioOptions {
 struct VideoOptions {
   void SetAll(const VideoOptions& change) {
     SetFrom(&video_noise_reduction, change.video_noise_reduction);
-    SetFrom(&cpu_overuse_detection, change.cpu_overuse_detection);
     SetFrom(&conference_mode, change.conference_mode);
-    SetFrom(&dscp, change.dscp);
     SetFrom(&suspend_below_min_bitrate, change.suspend_below_min_bitrate);
     SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
-    SetFrom(&disable_prerenderer_smoothing,
-            change.disable_prerenderer_smoothing);
   }
 
   bool operator==(const VideoOptions& o) const {
     return video_noise_reduction == o.video_noise_reduction &&
-           cpu_overuse_detection == o.cpu_overuse_detection &&
            conference_mode == o.conference_mode &&
-           dscp == o.dscp &&
            suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
-           screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps &&
-           disable_prerenderer_smoothing == o.disable_prerenderer_smoothing;
+           screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps;
   }
 
   std::string ToString() const {
     std::ostringstream ost;
     ost << "VideoOptions {";
     ost << ToStringIfSet("noise reduction", video_noise_reduction);
-    ost << ToStringIfSet("cpu overuse detection", cpu_overuse_detection);
     ost << ToStringIfSet("conference mode", conference_mode);
-    ost << ToStringIfSet("dscp", dscp);
     ost << ToStringIfSet("suspend below min bitrate",
                          suspend_below_min_bitrate);
     ost << ToStringIfSet("screencast min bitrate kbps",
@@ -279,11 +267,6 @@ struct VideoOptions {
   // constraint 'googNoiseReduction', and WebRtcVideoEngine2 passes it
   // on to the codec options. Disabled by default.
   rtc::Optional<bool> video_noise_reduction;
-  // Enable WebRTC Cpu Overuse Detection. This flag comes from the
-  // PeerConnection constraint 'googCpuOveruseDetection' and is
-  // checked in WebRtcVideoChannel2::OnLoadUpdate, where it's passed
-  // to VideoCapturer::video_adapter()->OnCpuResolutionRequest.
-  rtc::Optional<bool> cpu_overuse_detection;
   // 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
@@ -291,12 +274,6 @@ struct VideoOptions {
   // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
   // The special screencast behaviour is disabled by default.
   rtc::Optional<bool> conference_mode;
-  // Set DSCP value for packet sent from video channel. This flag
-  // comes from the PeerConnection constraint 'googDscp' and,
-  // WebRtcVideoChannel2::SetOptions checks it before calling
-  // MediaChannel::SetDscp. If enabled, rtc::DSCP_AF41 is used. If
-  // disabled, which is the default, rtc::DSCP_DEFAULT is used.
-  rtc::Optional<bool> dscp;
   // 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
@@ -307,22 +284,6 @@ struct VideoOptions {
   // the PeerConnection constraint 'googScreencastMinBitrate'. It is
   // copied to the encoder config by WebRtcVideoChannel2.
   rtc::Optional<int> screencast_min_bitrate_kbps;
-  // 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
-  // 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.
-  rtc::Optional<bool> disable_prerenderer_smoothing;
 
  private:
   template <typename T>
@@ -385,15 +346,22 @@ class MediaChannel : public sigslot::has_slots<> {
     virtual ~NetworkInterface() {}
   };
 
-  MediaChannel() : network_interface_(NULL) {}
+  MediaChannel(const webrtc::MediaConfig& config)
+      : network_interface_(NULL) {
+    enable_dscp_ = config.enable_dscp;
+  }
+  MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
   virtual ~MediaChannel() {}
 
   // Sets the abstract interface class for sending RTP/RTCP data.
   virtual void SetInterface(NetworkInterface *iface) {
     rtc::CritScope cs(&network_interface_crit_);
     network_interface_ = iface;
+    SetDscp(enable_dscp_ ? MediaTypeDscpValue() : rtc::DSCP_DEFAULT);

[pthatcher1, 2016/02/05 17:11:30]

Can you call this something like preferred_dscp() instead of
MediaTypeDscpValue()?

[nisse-webrtc, 2016/02/08 09:02:32]

Sure. But for consistency, shouldn't the method name use CamelCase (ugly as it
is)? I.e., PreferredDscp()?

[pbos-webrtc, 2016/02/08 10:38:48]

https://google.github.io/styleguide/cppguide.html#Function_Names

Regular functions have mixed case; "cheap" functions may use lower case with
underscores.

I think this qualifies as a cheap method, so I think you can choose.

[nisse-webrtc, 2016/02/08 12:22:30]

Ok. I used CamelCase for now, but I can change it if anyone wants, I don't have
a strong opinion here.

+  }
+  virtual rtc::DiffServCodePoint MediaTypeDscpValue() const {
+    return rtc::DSCP_DEFAULT;
   }
-
   // Called when a RTP packet is received.
   virtual void OnPacketReceived(rtc::Buffer* packet,
                                 const rtc::PacketTime& packet_time) = 0;
@@ -441,7 +409,7 @@ class MediaChannel : public sigslot::has_slots<> {
     return network_interface_->SetOption(type, opt, option);
   }
 
- protected:
+ private:
   // This method sets DSCP |value| on both RTP and RTCP channels.
   int SetDscp(rtc::DiffServCodePoint value) {
     int ret;
@@ -468,6 +436,7 @@ class MediaChannel : public sigslot::has_slots<> {
                    : network_interface_->SendRtcp(packet, options);
   }
 
+  bool enable_dscp_;
   // |network_interface_| can be accessed from the worker_thread and
   // from any MediaEngine threads. This critical section is to protect accessing
   // of network_interface_ object.
@@ -921,6 +890,7 @@ class VoiceMediaChannel : public MediaChannel {
   };
 
   VoiceMediaChannel() {}
+  VoiceMediaChannel(const webrtc::MediaConfig& config) : MediaChannel(config) {}
   virtual ~VoiceMediaChannel() {}
   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
@@ -984,6 +954,7 @@ class VideoMediaChannel : public MediaChannel {
   };
 
   VideoMediaChannel() {}
+  VideoMediaChannel(const webrtc::MediaConfig& config) : MediaChannel(config) {}
   virtual ~VideoMediaChannel() {}
 
   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;