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 class ReceiveStatistics; |
| 21 |
20 // This class is responsible for RTX decapsulation. The resulting media packets | 22 // This class is responsible for RTX decapsulation. The resulting media packets |
21 // are passed on to a sink representing the associated media stream. | 23 // are passed on to a sink representing the associated media stream. |
22 class RtxReceiveStream : public RtpPacketSinkInterface { | 24 class RtxReceiveStream : public RtpPacketSinkInterface { |
23 public: | 25 public: |
24 RtxReceiveStream(RtpPacketSinkInterface* media_sink, | 26 RtxReceiveStream(RtpPacketSinkInterface* media_sink, |
25 std::map<int, int> associated_payload_types, | 27 std::map<int, int> associated_payload_types, |
26 uint32_t media_ssrc); | 28 uint32_t media_ssrc, |
| 29 // TODO(nisse): Delete this argument, and |
| 30 // corresponding member variable, by moving the |
| 31 // responsibility for rtcp feedback to |
| 32 // RtpStreamReceiverController. |
| 33 ReceiveStatistics* rtp_receive_statistics = nullptr); |
27 ~RtxReceiveStream() override; | 34 ~RtxReceiveStream() override; |
28 // RtpPacketSinkInterface. | 35 // RtpPacketSinkInterface. |
29 void OnRtpPacket(const RtpPacketReceived& packet) override; | 36 void OnRtpPacket(const RtpPacketReceived& packet) override; |
30 | 37 |
31 private: | 38 private: |
32 RtpPacketSinkInterface* const media_sink_; | 39 RtpPacketSinkInterface* const media_sink_; |
33 // Map from rtx payload type -> media payload type. | 40 // Map from rtx payload type -> media payload type. |
34 const std::map<int, int> associated_payload_types_; | 41 const std::map<int, int> associated_payload_types_; |
35 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the | 42 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the |
36 // ssrc, and we should delete this. | 43 // ssrc, and we should delete this. |
37 const uint32_t media_ssrc_; | 44 const uint32_t media_ssrc_; |
| 45 ReceiveStatistics* const rtp_receive_statistics_; |
38 }; | 46 }; |
39 | 47 |
40 } // namespace webrtc | 48 } // namespace webrtc |
41 | 49 |
42 #endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ | 50 #endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ |
OLD | NEW |