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