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
|
/*
* Copyright 2015 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.
*/
// This file contains classes that implement RtpReceiverInterface.
// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
// transport (provided by webrtc::VoiceChannel/webrtc::VideoChannel)
#ifndef PC_RTP_RECEIVER_H_
#define PC_RTP_RECEIVER_H_
#include <stdint.h>
#include <optional>
#include <string>
#include <vector>
#include "api/dtls_transport_interface.h"
#include "api/media_stream_interface.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
#include "media/base/media_channel.h"
namespace webrtc {
// Internal class used by PeerConnection.
class RtpReceiverInternal : public RtpReceiverInterface {
public:
// Call on the signaling thread, to let the receiver know that the the
// embedded source object should enter a stopped/ended state and the track's
// state set to `kEnded`, a final state that cannot be reversed.
virtual void Stop() = 0;
// Sets the underlying MediaEngine channel associated with this RtpSender.
// A VoiceMediaChannel should be used for audio RtpSenders and
// a VideoMediaChannel should be used for video RtpSenders.
// NOTE:
// * SetMediaChannel(nullptr) must be called before the media channel is
// destroyed.
// * This method must be invoked on the worker thread.
virtual void SetMediaChannel(MediaReceiveChannelInterface* media_channel) = 0;
// Configures the RtpReceiver with the underlying media channel, with the
// given SSRC as the stream identifier.
virtual void SetupMediaChannel(uint32_t ssrc) = 0;
// Configures the RtpReceiver with the underlying media channel to receive an
// unsignaled receive stream.
virtual void SetupUnsignaledMediaChannel() = 0;
virtual void set_transport(
scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
// This SSRC is used as an identifier for the receiver between the API layer
// and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
virtual std::optional<uint32_t> ssrc() const = 0;
// Call this to notify the RtpReceiver when the first packet has been received
// on the corresponding channel.
virtual void NotifyFirstPacketReceived() = 0;
// Set the associated remote media streams for this receiver. The remote track
// will be removed from any streams that are no longer present and added to
// any new streams.
virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0;
// TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of
// set_stream_ids() as soon as downstream projects are no longer dependent on
// stream objects.
virtual void SetStreams(
const std::vector<scoped_refptr<MediaStreamInterface>>& streams) = 0;
// Returns an ID that changes if the attached track changes, but
// otherwise remains constant. Used to generate IDs for stats.
// The special value zero means that no track is attached.
virtual int AttachmentId() const = 0;
protected:
static int GenerateUniqueId();
static std::vector<scoped_refptr<MediaStreamInterface>> CreateStreamsFromIds(
std::vector<std::string> stream_ids);
};
} // namespace webrtc
#endif // PC_RTP_RECEIVER_H_
|
