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 78 matching lines @@
   bool CheckNoRtp() { return rtp_packets_.empty(); }
   bool CheckNoRtcp() { return rtcp_packets_.empty(); }
   void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
   void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
   virtual bool AddSendStream(const StreamParams& sp) {
     if (std::find(send_streams_.begin(), send_streams_.end(), sp) !=
         send_streams_.end()) {
       return false;
     }
     send_streams_.push_back(sp);
+    rtp_parameters_[sp.first_ssrc()] = CreateRtpParametersWithOneEncoding();
     return true;
   }
   virtual bool RemoveSendStream(uint32_t ssrc) {
+    auto parameters_iterator = rtp_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_parameters_.end()) {
+      rtp_parameters_.erase(parameters_iterator);
+    }
     return RemoveStreamBySsrc(&send_streams_, ssrc);
   }
   virtual bool AddRecvStream(const StreamParams& sp) {
     if (std::find(receive_streams_.begin(), receive_streams_.end(), sp) !=
         receive_streams_.end()) {
       return false;
     }
     receive_streams_.push_back(sp);
     return true;
   }
   virtual bool RemoveRecvStream(uint32_t ssrc) {
     return RemoveStreamBySsrc(&receive_streams_, ssrc);
   }
+
+  virtual webrtc::RtpParameters GetRtpParameters(uint32_t ssrc) const {
+    auto parameters_iterator = rtp_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_parameters_.end()) {
+      return parameters_iterator->second;
+    }
+    return webrtc::RtpParameters();
+  }
+  virtual bool SetRtpParameters(uint32_t ssrc,
+                                const webrtc::RtpParameters& parameters) {
+    auto parameters_iterator = rtp_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_parameters_.end()) {
+      parameters_iterator->second = parameters;
+      return true;
+    }
+    // Replicate the behavior of the real media channel: return false
+    // when setting parameters for unknown SSRCs.
+    return false;
+  }
+
   bool IsStreamMuted(uint32_t ssrc) const {
     bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
     // If |ssrc = 0| check if the first send stream is muted.
     if (!ret && ssrc == 0 && !send_streams_.empty()) {
       return muted_streams_.find(send_streams_[0].first_ssrc()) !=
              muted_streams_.end();
     }
     return ret;
   }
   const std::vector<StreamParams>& send_streams() const {
@@ skipping 72 matching lines @@
  private:
   bool sending_;
   bool playout_;
   std::vector<RtpHeaderExtension> recv_extensions_;
   std::vector<RtpHeaderExtension> send_extensions_;
   std::list<std::string> rtp_packets_;
   std::list<std::string> rtcp_packets_;
   std::vector<StreamParams> send_streams_;
   std::vector<StreamParams> receive_streams_;
   std::set<uint32_t> muted_streams_;
+  std::map<uint32_t, webrtc::RtpParameters> rtp_parameters_;
   bool fail_set_send_codecs_;
   bool fail_set_recv_codecs_;
   uint32_t send_ssrc_;
   std::string rtcp_cname_;
   bool ready_to_send_;
 };
 
 class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
  public:
   struct DtmfInfo {
     DtmfInfo(uint32_t ssrc, int event_code, int duration)
         : 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) {
+      : engine_(engine), time_since_last_typing_(-1), max_bps_(-1) {
     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 bool SetPlayout(bool playout) {
     set_playout(playout);
     return true;
   }
   virtual void SetSend(bool send) { set_sending(send); }
   virtual bool SetAudioSend(uint32_t ssrc,
                             bool enable,
                             const AudioOptions* options,
                             AudioSource* source) {
     if (!SetLocalSource(ssrc, source)) {
@@ skipping 105 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 SetLocalSource(uint32_t ssrc, AudioSource* source) {
     auto it = local_sinks_.find(ssrc);
     if (source) {
       if (it != local_sinks_.end()) {
         ASSERT(it->second->source() == source);
@@ 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_sinks_;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
+  int max_bps_;
 };
 
 // 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);
 }
@@ skipping 17 matching lines @@
       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 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 408 matching lines @@
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
   DataChannelType last_channel_type_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_