Enable setting the maximum bitrate limit in RtpSender.

webrtc/media/base/fakemediaengine.h

Index: webrtc/media/base/fakemediaengine.h
diff --git a/webrtc/media/base/fakemediaengine.h b/webrtc/media/base/fakemediaengine.h
index afd262bb5e92263951cc85e7be3256e5d867801c..78530ed919f61b5c022e4f4223e1086e7b3da550 100644
--- a/webrtc/media/base/fakemediaengine.h
+++ b/webrtc/media/base/fakemediaengine.h
@@ -225,7 +225,9 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
   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);
   }
@@ -237,7 +239,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
     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) &&
@@ -249,6 +251,16 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
     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;
@@ -377,7 +389,10 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
     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);
@@ -410,6 +425,8 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
   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.
@@ -425,7 +442,9 @@ 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);
   }
 
@@ -448,6 +467,14 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
             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));
@@ -544,6 +571,7 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
   std::map<uint32_t, VideoCapturer*> capturers_;
   VideoOptions options_;
   int max_bps_;
+  webrtc::RTCRtpParameters rtp_parameters_;
 };
 
 class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {