Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/base/fakemediaengine.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 207 matching lines @@
         : ssrc(ssrc),
           event_code(event_code),
           duration(duration) {}
     uint32_t ssrc;
     int event_code;
     int duration;
   };
   explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
                                  const AudioOptions& options)
       : engine_(engine),
-        time_since_last_typing_(-1) {
+        time_since_last_typing_(-1),
+        max_bps_(-1),
+        rtp_parameters_(webrtc::RTCRtpParameters::CreateDefault()) {

[pthatcher1, 2016/03/12 01:21:03]

It seems like building a one-encoding parameters should be defined here, not in
RtpParameters.  Actually, why do we need one encoding the parameters here?

[Taylor Brandstetter, 2016/03/12 01:57:06]

To answer the question: If you call GetParameters there should be something
returned. And I think it would be much better defined in mediaengine.h, not
fakemediaengine.h.

     output_scalings_[0] = 1.0;  // For default channel.
     SetOptions(options);
   }
   ~FakeVoiceMediaChannel();
   const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
   const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
   const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
   const std::vector<DtmfInfo>& dtmf_info_queue() const {
     return dtmf_info_queue_;
   }
   const AudioOptions& options() const { return options_; }
-
+  int max_bps() const { return max_bps_; }
   virtual bool SetSendParameters(const AudioSendParameters& params) {
     return (SetSendCodecs(params.codecs) &&
             SetSendRtpHeaderExtensions(params.extensions) &&
             SetMaxSendBandwidth(params.max_bandwidth_bps) &&
             SetOptions(params.options));
   }
 
   virtual bool SetRecvParameters(const AudioRecvParameters& params) {
     return (SetRecvCodecs(params.codecs) &&
             SetRecvRtpHeaderExtensions(params.extensions));
   }
+
+  virtual webrtc::RTCRtpParameters GetRtpParameters(uint32_t ssrc) {
+    return rtp_parameters_;
+  }
+  virtual bool SetRtpParameters(uint32_t ssrc,
+                                const webrtc::RTCRtpParameters& parameters) {
+    rtp_parameters_ = parameters;

[Taylor Brandstetter, 2016/03/12 01:57:06]

I know this is just a stub class, but for consistency we should probably make
rtp_parameters_ a map (on ssrc), in case the need arises later to call
Get/SetRtpParameters on multiple SSRCs. Just need to add/remove entries inside
Add/RemoveSendStream. Also, can move it to RtpHelper.

[skvlad, 2016/03/15 21:18:18]

Done.

+    return true;
+  }
+
   virtual bool SetPlayout(bool playout) {
     set_playout(playout);
     return true;
   }
   virtual bool SetSend(SendFlags flag) {
     return set_sending(flag != SEND_NOTHING);
   }
   virtual bool SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const AudioOptions* options,
@@ skipping 108 matching lines @@
     return true;
   }
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
     if (fail_set_send_codecs()) {
       // Fake the failure in SetSendCodecs.
       return false;
     }
     send_codecs_ = codecs;
     return true;
   }
-  bool SetMaxSendBandwidth(int bps) { return true; }
+  bool SetMaxSendBandwidth(int bps) {
+    max_bps_ = bps;
+    return true;
+  }
   bool SetOptions(const AudioOptions& options) {
     // Does a "merge" of current options and set options.
     options_.SetAll(options);
     return true;
   }
   bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer) {
     auto it = local_renderers_.find(ssrc);
     if (renderer) {
       if (it != local_renderers_.end()) {
         ASSERT(it->second->renderer() == renderer);
@@ skipping 12 matching lines @@
 
   FakeVoiceEngine* engine_;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
   std::map<uint32_t, double> output_scalings_;
   std::vector<DtmfInfo> dtmf_info_queue_;
   int time_since_last_typing_;
   AudioOptions options_;
   std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
+  int max_bps_;
+  webrtc::RTCRtpParameters rtp_parameters_;
 };
 
 // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
 inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
                             uint32_t ssrc,
                             int event_code,
                             int duration) {
   return (info.duration == duration && info.event_code == event_code &&
           info.ssrc == ssrc);
 }
 
 class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
  public:
   explicit FakeVideoMediaChannel(FakeVideoEngine* engine,
                                  const VideoOptions& options)
-      : engine_(engine), max_bps_(-1) {
+      : engine_(engine),
+        max_bps_(-1),
+        rtp_parameters_(webrtc::RTCRtpParameters::CreateDefault()) {
     SetOptions(options);
   }
 
   ~FakeVideoMediaChannel();
 
   const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
   const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
   const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
   bool rendering() const { return playout(); }
   const VideoOptions& options() const { return options_; }
   const std::map<uint32_t, rtc::VideoSinkInterface<VideoFrame>*>& sinks()
       const {
     return sinks_;
   }
   int max_bps() const { return max_bps_; }
   virtual bool SetSendParameters(const VideoSendParameters& params) {
     return (SetSendCodecs(params.codecs) &&
             SetSendRtpHeaderExtensions(params.extensions) &&
             SetMaxSendBandwidth(params.max_bandwidth_bps) &&
             SetOptions(params.options));
   }
 
+  virtual webrtc::RTCRtpParameters GetRtpParameters(uint32_t ssrc) {
+    return rtp_parameters_;
+  }
+  virtual bool SetRtpParameters(uint32_t ssrc,
+                                const webrtc::RTCRtpParameters& parameters) {
+    rtp_parameters_ = parameters;
+    return true;
+  }
   virtual bool SetRecvParameters(const VideoRecvParameters& params) {
     return (SetRecvCodecs(params.codecs) &&
             SetRecvRtpHeaderExtensions(params.extensions));
   }
   virtual bool AddSendStream(const StreamParams& sp) {
     return RtpHelper<VideoMediaChannel>::AddSendStream(sp);
   }
   virtual bool RemoveSendStream(uint32_t ssrc) {
     return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
   }
@@ skipping 76 matching lines @@
     return true;
   }
 
   FakeVideoEngine* engine_;
   std::vector<VideoCodec> recv_codecs_;
   std::vector<VideoCodec> send_codecs_;
   std::map<uint32_t, rtc::VideoSinkInterface<VideoFrame>*> sinks_;
   std::map<uint32_t, VideoCapturer*> capturers_;
   VideoOptions options_;
   int max_bps_;
+  webrtc::RTCRtpParameters rtp_parameters_;
 };
 
 class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
  public:
   explicit FakeDataMediaChannel(void* unused, const DataOptions& options)
       : send_blocked_(false), max_bps_(-1) {}
   ~FakeDataMediaChannel() {}
   const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
   const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
   const std::vector<DataCodec>& codecs() const { return send_codecs(); }
@@ skipping 312 matching lines @@
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
   DataChannelType last_channel_type_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_