Line data Source code
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_CALL_FLEXFEC_RECEIVE_STREAM_H_
12 : #define WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
13 :
14 : #include <stdint.h>
15 :
16 : #include <string>
17 : #include <vector>
18 :
19 : #include "webrtc/api/call/transport.h"
20 : #include "webrtc/config.h"
21 :
22 : namespace webrtc {
23 :
24 0 : class FlexfecReceiveStream {
25 : public:
26 : struct Stats {
27 : std::string ToString(int64_t time_ms) const;
28 :
29 : // TODO(brandtr): Add appropriate stats here.
30 : int flexfec_bitrate_bps;
31 : };
32 :
33 0 : struct Config {
34 : explicit Config(Transport* rtcp_send_transport)
35 : : rtcp_send_transport(rtcp_send_transport) {
36 : RTC_DCHECK(rtcp_send_transport);
37 : }
38 :
39 : std::string ToString() const;
40 :
41 : // Returns true if all RTP information is available in order to
42 : // enable receiving FlexFEC.
43 : bool IsCompleteAndEnabled() const;
44 :
45 : // Payload type for FlexFEC.
46 : int payload_type = -1;
47 :
48 : // SSRC for FlexFEC stream to be received.
49 : uint32_t remote_ssrc = 0;
50 :
51 : // Vector containing a single element, corresponding to the SSRC of the
52 : // media stream being protected by this FlexFEC stream. The vector MUST have
53 : // size 1.
54 : //
55 : // TODO(brandtr): Update comment above when we support multistream
56 : // protection.
57 : std::vector<uint32_t> protected_media_ssrcs;
58 :
59 : // SSRC for RTCP reports to be sent.
60 : uint32_t local_ssrc = 0;
61 :
62 : // What RTCP mode to use in the reports.
63 : RtcpMode rtcp_mode = RtcpMode::kCompound;
64 :
65 : // Transport for outgoing RTCP packets.
66 : Transport* rtcp_send_transport = nullptr;
67 :
68 : // |transport_cc| is true whenever the send-side BWE RTCP feedback message
69 : // has been negotiated. This is a prerequisite for enabling send-side BWE.
70 : bool transport_cc = false;
71 :
72 : // RTP header extensions that have been negotiated for this track.
73 : std::vector<RtpExtension> rtp_header_extensions;
74 : };
75 :
76 : // Starts stream activity.
77 : // When a stream is active, it can receive and process packets.
78 : virtual void Start() = 0;
79 : // Stops stream activity.
80 : // When a stream is stopped, it can't receive nor process packets.
81 : virtual void Stop() = 0;
82 :
83 : virtual Stats GetStats() const = 0;
84 :
85 : protected:
86 0 : virtual ~FlexfecReceiveStream() = default;
87 : };
88 :
89 : } // namespace webrtc
90 :
91 : #endif // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
|