Adding getParameters/setParameters APIs to RtpReceiver.

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 81 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();
+    rtp_send_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);
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_parameters_.end()) {
+      rtp_send_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);
+    rtp_receive_parameters_[sp.first_ssrc()] =
+        CreateRtpParametersWithOneEncoding();
     return true;
   }
   virtual bool RemoveRecvStream(uint32_t ssrc) {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_parameters_.end()) {
+      rtp_receive_parameters_.erase(parameters_iterator);
+    }
     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()) {
+  virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const {
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_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()) {
+  virtual bool SetRtpSendParameters(uint32_t ssrc,
+                                    const webrtc::RtpParameters& parameters) {
+    auto parameters_iterator = rtp_send_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_send_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;
+  }
+
+  virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_parameters_.end()) {
+      return parameters_iterator->second;
+    }
+    return webrtc::RtpParameters();
+  }
+  virtual bool SetRtpReceiveParameters(
+      uint32_t ssrc,
+      const webrtc::RtpParameters& parameters) {
+    auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
+    if (parameters_iterator != rtp_receive_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()) !=
@@ skipping 88 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_;
+  std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
+  std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
   bool fail_set_send_codecs_;
   bool fail_set_recv_codecs_;
   uint32_t send_ssrc_;
   std::string rtcp_cname_;
   bool ready_to_send_;
   rtc::NetworkRoute last_network_route_;
   int num_network_route_changes_ = 0;
 };
 
 class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
@@ skipping 654 matching lines @@
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
   DataChannelType last_channel_type_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_