Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/engine/webrtcvideoengine2.h

 /*
  *  Copyright (c) 2014 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 127 matching lines @@
                       const std::vector<VideoCodec>& recv_codecs,
                       WebRtcVideoEncoderFactory* external_encoder_factory,
                       WebRtcVideoDecoderFactory* external_decoder_factory);
   ~WebRtcVideoChannel2() override;
 
   // VideoMediaChannel implementation
   rtc::DiffServCodePoint PreferredDscp() const override;
 
   bool SetSendParameters(const VideoSendParameters& params) override;
   bool SetRecvParameters(const VideoRecvParameters& params) override;
+  webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const override;
+  bool SetRtpParameters(uint32_t ssrc,
+                        const webrtc::RtpParameters& parameters) override;
   bool GetSendCodec(VideoCodec* send_codec) override;
   bool SetSend(bool send) override;
   bool SetVideoSend(uint32_t ssrc,
                     bool mute,
                     const VideoOptions* options) override;
   bool AddSendStream(const StreamParams& sp) override;
   bool RemoveSendStream(uint32_t ssrc) override;
   bool AddRecvStream(const StreamParams& sp) override;
   bool AddRecvStream(const StreamParams& sp, bool default_stream);
   bool RemoveRecvStream(uint32_t ssrc) override;
@@ skipping 80 matching lines @@
         bool enable_cpu_overuse_detection,
         int max_bitrate_bps,
         const rtc::Optional<VideoCodecSettings>& codec_settings,
         const std::vector<webrtc::RtpExtension>& rtp_extensions,
         const VideoSendParameters& send_params);
     virtual ~WebRtcVideoSendStream();
 
     void SetOptions(const VideoOptions& options);
     // TODO(pbos): Move logic from SetOptions into this method.
     void SetSendParameters(const ChangedSendParameters& send_params);
+    bool SetRtpParameters(const webrtc::RtpParameters& parameters);
 
     void OnFrame(const cricket::VideoFrame& frame) override;
     bool SetCapturer(VideoCapturer* capturer);
     void MuteStream(bool mute);
     bool DisconnectCapturer();
 
     void Start();
     void Stop();
 
+    webrtc::RtpParameters rtp_parameters() const { return rtp_parameters_; }
+
     // Implements webrtc::LoadObserver.
     void OnLoadUpdate(Load load) override;
 
     const std::vector<uint32_t>& GetSsrcs() const;
     VideoSenderInfo GetVideoSenderInfo();
     void FillBandwidthEstimationInfo(BandwidthEstimationInfo* bwe_info);
 
    private:
     // Parameters needed to reconstruct the underlying stream.
     // webrtc::VideoSendStream doesn't support setting a lot of options on the
@@ skipping 64 matching lines @@
     void DestroyVideoEncoder(AllocatedEncoder* encoder)
         EXCLUSIVE_LOCKS_REQUIRED(lock_);
     void SetCodec(const VideoCodecSettings& codec)
         EXCLUSIVE_LOCKS_REQUIRED(lock_);
     void RecreateWebRtcStream() EXCLUSIVE_LOCKS_REQUIRED(lock_);
     webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
         const Dimensions& dimensions,
         const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_);
     void SetDimensions(int width, int height)
         EXCLUSIVE_LOCKS_REQUIRED(lock_);
+    bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
 
     rtc::ThreadChecker thread_checker_;
     rtc::AsyncInvoker invoker_;
     rtc::Thread* worker_thread_;
     const std::vector<uint32_t> ssrcs_;
     const std::vector<SsrcGroup> ssrc_groups_;
     webrtc::Call* const call_;
     rtc::VideoSinkWants sink_wants_;
     // Counter used for deciding if the video resolution is currently
     // restricted by CPU usage. It is reset if |capturer_| is changed.
     int cpu_restricted_counter_;
     // Total number of times resolution as been requested to be changed due to
     // CPU adaptation.
     int number_of_cpu_adapt_changes_;
     VideoCapturer* capturer_;
     WebRtcVideoEncoderFactory* const external_encoder_factory_
         GUARDED_BY(lock_);
 
     rtc::CriticalSection lock_;
     webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
+    // Contains settings that are the same for all streams in the MediaChannel,
+    // such as codecs, header extensions, and the global bitrate limit for the
+    // entire channel.
     VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
+    // Contains settings that are unique for each stream, such as max_bitrate.
+    // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
+    // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
+    // one stream per MediaChannel.
+    webrtc::RtpParameters rtp_parameters_;
     bool pending_encoder_reconfiguration_ GUARDED_BY(lock_);
     VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_);
     AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_);
     Dimensions last_dimensions_ GUARDED_BY(lock_);
     webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) =
         webrtc::kVideoRotation_0;
 
     bool sending_ GUARDED_BY(lock_);
     bool muted_ GUARDED_BY(lock_);
 
@@ skipping 144 matching lines @@
   // TODO(deadbeef): Don't duplicate information between
   // send_params/recv_params, rtp_extensions, options, etc.
   VideoSendParameters send_params_;
   VideoOptions default_send_options_;
   VideoRecvParameters recv_params_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_