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1 /* | |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_API_CALL_FLEXFEC_RECEIVE_STREAM_H_ | |
12 #define WEBRTC_API_CALL_FLEXFEC_RECEIVE_STREAM_H_ | |
13 | |
14 #include <string> | |
15 #include <vector> | |
16 | |
17 #include "webrtc/api/call/transport.h" | |
18 #include "webrtc/config.h" | |
19 | |
20 namespace webrtc { | |
21 | |
22 // WORK IN PROGRESS! | |
23 // This class is under development and it is not yet intended for use outside | |
24 // of WebRTC. | |
25 // | |
26 // TODO(brandtr): Remove this comment when FlexFEC is ready for public use. | |
27 | |
28 class FlexfecReceiveStream { | |
29 public: | |
30 struct Stats { | |
31 std::string ToString(int64_t time_ms) const; | |
32 | |
33 // TODO(brandtr): Add appropriate stats here. | |
34 int flexfec_bitrate_bps; | |
35 }; | |
36 | |
37 struct Config { | |
38 std::string ToString() const; | |
39 | |
40 // Payload type for FlexFEC. | |
41 int payload_type = -1; | |
42 | |
43 // SSRC for FlexFEC stream to be received. | |
44 uint32_t remote_ssrc = 0; | |
45 | |
46 // Vector containing a single element, corresponding to the SSRC of the | |
47 // media stream being protected by this FlexFEC stream. The vector MUST have | |
48 // size 1. | |
49 // | |
50 // TODO(brandtr): Update comment above when we support multistream | |
51 // protection. | |
52 std::vector<uint32_t> protected_media_ssrcs; | |
53 | |
54 // SSRC for RTCP reports to be sent. | |
55 uint32_t local_ssrc = 0; | |
56 | |
57 // What RTCP mode to use in the reports. | |
58 RtcpMode rtcp_mode = RtcpMode::kCompound; | |
59 | |
60 // Transport for outgoing RTCP packets. | |
61 Transport* rtcp_send_transport = nullptr; | |
62 | |
63 // |transport_cc| is true whenever the send-side BWE RTCP feedback message | |
64 // has been negotiated. This is a prerequisite for enabling send-side BWE. | |
65 bool transport_cc = false; | |
66 | |
67 // RTP header extensions that have been negotiated for this track. | |
68 std::vector<RtpExtension> extensions; | |
69 }; | |
70 | |
71 // Starts stream activity. | |
72 // When a stream is active, it can receive and process packets. | |
73 virtual void Start() = 0; | |
74 // Stops stream activity. | |
75 // When a stream is stopped, it can't receive nor process packets. | |
76 virtual void Stop() = 0; | |
77 | |
78 virtual Stats GetStats() const = 0; | |
79 | |
80 protected: | |
81 virtual ~FlexfecReceiveStream() = default; | |
82 }; | |
83 | |
84 } // namespace webrtc | |
85 | |
86 #endif // WEBRTC_API_CALL_FLEXFEC_RECEIVE_STREAM_H_ | |
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