 Chromium Code Reviews
 Chromium Code Reviews Issue 3008773002:
  Use RtxReceiveStream.  (Closed)
    
  
    Issue 3008773002:
  Use RtxReceiveStream.  (Closed) 
  | OLD | NEW | 
|---|---|
| 1 /* | 1 /* | 
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 
| 3 * | 3 * | 
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license | 
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source | 
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found | 
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may | 
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. | 
| 9 */ | 9 */ | 
| 10 | 10 | 
| 11 #ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | 11 #ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | 
| 12 #define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | 12 #define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | 
| 13 | 13 | 
| 14 #include <map> | 14 #include <map> | 
| 15 | 15 | 
| 16 #include "webrtc/call/rtp_packet_sink_interface.h" | 16 #include "webrtc/call/rtp_packet_sink_interface.h" | 
| 17 | 17 | 
| 18 namespace webrtc { | 18 namespace webrtc { | 
| 19 | 19 | 
| 20 // TODO(nisse): Consider renaming, RtxReceiveStream looks similar to | |
| 
danilchap
2017/08/29 13:33:06
I doubt these kind of TODOs belong to code and eve
 
nisse-webrtc
2017/08/29 13:51:35
Rasmus wanted a TODO...
 | |
| 21 // VideoReceiveStream which is associated with Call. This class | |
| 22 // belongs to a different level, so RtxStreamReceiver might be a | |
| 23 // better name, more similar to RtpVideoStreamReceiver which belongs | |
| 24 // to the same level, and processes the media packets. | |
| 20 class RtxReceiveStream : public RtpPacketSinkInterface { | 25 class RtxReceiveStream : public RtpPacketSinkInterface { | 
| 21 public: | 26 public: | 
| 22 RtxReceiveStream(RtpPacketSinkInterface* media_sink, | 27 RtxReceiveStream(RtpPacketSinkInterface* media_sink, | 
| 23 std::map<int, int> rtx_payload_type_map, | 28 std::map<int, int> associated_payload_types, | 
| 
danilchap
2017/08/29 13:33:06
are these renames needed to use the RtxReceiveStre
 
nisse-webrtc
2017/08/29 13:51:35
Not strictly necessary, they're for consistency wi
 | |
| 24 uint32_t media_ssrc); | 29 uint32_t media_ssrc); | 
| 25 ~RtxReceiveStream() override; | 30 ~RtxReceiveStream() override; | 
| 26 // RtpPacketSinkInterface. | 31 // RtpPacketSinkInterface. | 
| 27 void OnRtpPacket(const RtpPacketReceived& packet) override; | 32 void OnRtpPacket(const RtpPacketReceived& packet) override; | 
| 28 | 33 | 
| 29 private: | 34 private: | 
| 30 RtpPacketSinkInterface* const media_sink_; | 35 RtpPacketSinkInterface* const media_sink_; | 
| 31 // Mapping rtx_payload_type_map_[rtx] = associated. | 36 // Map from rtx payload type -> media payload type. | 
| 32 const std::map<int, int> rtx_payload_type_map_; | 37 const std::map<int, int> associated_payload_types_; | 
| 33 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the | 38 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the | 
| 34 // ssrc, and we should delete this. | 39 // ssrc, and we should delete this. | 
| 35 const uint32_t media_ssrc_; | 40 const uint32_t media_ssrc_; | 
| 36 }; | 41 }; | 
| 37 | 42 | 
| 38 } // namespace webrtc | 43 } // namespace webrtc | 
| 39 | 44 | 
| 40 #endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | 45 #endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | 
| OLD | NEW |