Rename ViEReceiver and move ownership to VideoReceiveStream.

webrtc/video/rtp_stream_receiver.h

 /*
  *  Copyright (c) 2012 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.
  */
 
-#ifndef WEBRTC_VIDEO_VIE_RECEIVER_H_
-#define WEBRTC_VIDEO_VIE_RECEIVER_H_
+#ifndef WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
+#define WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_
 
 #include <list>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
@@ skipping 14 matching lines @@
 class RtcpRttStats;
 class RtpHeaderParser;
 class RTPPayloadRegistry;
 class RtpReceiver;
 class Transport;
 
 namespace vcm {
 class VideoReceiver;
 }  // namespace vcm
 
-class ViEReceiver : public RtpData {
+class RtpStreamReceiver : public RtpData, public RtpFeedback  {
  public:
-  ViEReceiver(vcm::VideoReceiver* video_receiver,
-              RemoteBitrateEstimator* remote_bitrate_estimator,
-              RtpFeedback* rtp_feedback,
-              Transport* transport,
-              RtcpRttStats* rtt_stats,
-              PacedSender* paced_sender,
-              PacketRouter* packet_router);
-  ~ViEReceiver();
+  RtpStreamReceiver(vcm::VideoReceiver* video_receiver,
+                    RemoteBitrateEstimator* remote_bitrate_estimator,
+                    Transport* transport,
+                    RtcpRttStats* rtt_stats,
+                    PacedSender* paced_sender,
+                    PacketRouter* packet_router);
+  ~RtpStreamReceiver();
 
   bool SetReceiveCodec(const VideoCodec& video_codec);
 
   void SetNackStatus(bool enable, int max_nack_reordering_threshold);
   void SetRtxPayloadType(int payload_type, int associated_payload_type);
   // If set to true, the RTX payload type mapping supplied in
   // |SetRtxPayloadType| will be used when restoring RTX packets. Without it,
   // RTX packets will always be restored to the last non-RTX packet payload type
   // received.
   void SetUseRtxPayloadMappingOnRestore(bool val);
@@ skipping 19 matching lines @@
                   size_t rtp_packet_length,
                   const PacketTime& packet_time);
   bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
 
   // Implements RtpData.
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 const size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override;
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
+  // Implements RtpFeedback.
+  int32_t OnInitializeDecoder(const int8_t payload_type,
+                              const char payload_name[RTP_PAYLOAD_NAME_SIZE],
+                              const int frequency,
+                              const size_t channels,
+                              const uint32_t rate) override;
+  void OnIncomingSSRCChanged(const uint32_t ssrc) override;
+  void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {}
+
   ReceiveStatistics* GetReceiveStatistics() const;
 
   template <class T>
   class RegisterableCallback : public T {
    public:
     RegisterableCallback() : callback_(nullptr) {}
 
     void Set(T* callback) {
       rtc::CritScope lock(&critsect_);
       callback_ = callback;
@@ skipping 56 matching lines @@
   bool receiving_ GUARDED_BY(receive_cs_);
   uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
   bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
   int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_VIDEO_VIE_RECEIVER_H_
+#endif  // WEBRTC_VIDEO_RTP_STREAM_RECEIVER_H_