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Side by Side Diff: webrtc/call/rtx_receive_stream.h

Issue 3006063002: Reland of Use RtxReceiveStream. (Closed)
Patch Set: Merge remote-tracking branch 'origin/master' into use-RtxStreamReceiver-reland Created 3 years, 3 months ago
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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
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): Should we make this argument
danilchap 2017/09/06 10:29:52 no, would rather see TODO to remove it once receiv
nisse-webrtc 2017/09/06 11:25:10 Done.
30 // mandatory? Ideally, generation of rtcp feedback
31 // should be a responsibility of the
32 // RtpStreamReceiverController, and then we
33 // wouldn't need it here.
34 ReceiveStatistics* rtp_receive_statistics = nullptr);
27 ~RtxReceiveStream() override; 35 ~RtxReceiveStream() override;
28 // RtpPacketSinkInterface. 36 // RtpPacketSinkInterface.
29 void OnRtpPacket(const RtpPacketReceived& packet) override; 37 void OnRtpPacket(const RtpPacketReceived& packet) override;
30 38
31 private: 39 private:
32 RtpPacketSinkInterface* const media_sink_; 40 RtpPacketSinkInterface* const media_sink_;
33 // Map from rtx payload type -> media payload type. 41 // Map from rtx payload type -> media payload type.
34 const std::map<int, int> associated_payload_types_; 42 const std::map<int, int> associated_payload_types_;
35 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the 43 // TODO(nisse): Ultimately, the media receive stream shouldn't care about the
36 // ssrc, and we should delete this. 44 // ssrc, and we should delete this.
37 const uint32_t media_ssrc_; 45 const uint32_t media_ssrc_;
46 ReceiveStatistics* const rtp_receive_statistics_;
38 }; 47 };
39 48
40 } // namespace webrtc 49 } // namespace webrtc
41 50
42 #endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_ 51 #endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
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