blob: 0464a70a28aaa9c4bce7bcd629b7c90d944b38a2 (
plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
|
/*
* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef CALL_RECEIVE_STREAM_H_
#define CALL_RECEIVE_STREAM_H_
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_types.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
namespace webrtc {
// Common base interface for MediaReceiveStream based classes and
// FlexfecReceiveStream.
class ReceiveStream {
public:
// Receive-stream specific RTP settings.
// TODO(tommi): This struct isn't needed at this level anymore. Move it closer
// to where it's used.
struct ReceiveStreamRtpConfig {
// Synchronization source (stream identifier) to be received.
// This member will not change mid-stream and can be assumed to be const
// post initialization.
uint32_t remote_ssrc = 0;
// Sender SSRC used for sending RTCP (such as receiver reports).
// This value may change mid-stream and must be done on the same thread
// that the value is read on (i.e. packet delivery).
uint32_t local_ssrc = 0;
// Enable feedback for send side bandwidth estimation.
// See
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
// for details.
// This value may change mid-stream and must be done on the same thread
// that the value is read on (i.e. packet delivery).
bool transport_cc = false;
// RTP header extensions used for the received stream.
// This value may change mid-stream and must be done on the same thread
// that the value is read on (i.e. packet delivery).
std::vector<RtpExtension> extensions;
};
// Set/change the rtp header extensions. Must be called on the packet
// delivery thread.
virtual void SetRtpExtensions(std::vector<RtpExtension> extensions) = 0;
virtual RtpHeaderExtensionMap GetRtpExtensionMap() const = 0;
// Returns a bool for whether feedback for send side bandwidth estimation is
// enabled. See
// https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
// for details.
// This value may change mid-stream and must be done on the same thread
// that the value is read on (i.e. packet delivery).
virtual bool transport_cc() const = 0;
protected:
virtual ~ReceiveStream() {}
};
// Either an audio or video receive stream.
class MediaReceiveStream : public ReceiveStream {
public:
// Starts stream activity.
// When a stream is active, it can receive, process and deliver packets.
virtual void Start() = 0;
// Stops stream activity. Must be called to match with a previous call to
// `Start()`. When a stream has been stopped, it won't receive, decode,
// process or deliver packets to downstream objects such as callback pointers
// set in the config struct.
virtual void Stop() = 0;
virtual void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface>
frame_transformer) = 0;
virtual void SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
virtual std::vector<RtpSource> GetSources() const = 0;
};
} // namespace webrtc
#endif // CALL_RECEIVE_STREAM_H_
|
