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
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
|
/*
* 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 interfaces for RtpReceivers
// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
#ifndef API_RTP_RECEIVER_INTERFACE_H_
#define API_RTP_RECEIVER_INTERFACE_H_
#include <string>
#include <vector>
#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/frame_transformer_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/proxy.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/system/rtc_export.h"
namespace webrtc {
class RtpReceiverObserverInterface {
public:
// Note: Currently if there are multiple RtpReceivers of the same media type,
// they will all call OnFirstPacketReceived at once.
//
// In the future, it's likely that an RtpReceiver will only call
// OnFirstPacketReceived when a packet is received specifically for its
// SSRC/mid.
virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
protected:
virtual ~RtpReceiverObserverInterface() {}
};
class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
public:
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
// The dtlsTransport attribute exposes the DTLS transport on which the
// media is received. It may be null.
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport
// TODO(https://bugs.webrtc.org/907849) remove default implementation
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
// The list of streams that |track| is associated with. This is the same as
// the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
// https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
// TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
// TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of
// stream_ids() as soon as downstream projects are no longer dependent on
// stream objects.
virtual std::vector<std::string> stream_ids() const;
virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
// Audio or video receiver?
virtual cricket::MediaType media_type() const = 0;
// Not to be confused with "mid", this is a field we can temporarily use
// to uniquely identify a receiver until we implement Unified Plan SDP.
virtual std::string id() const = 0;
// The WebRTC specification only defines RTCRtpParameters in terms of senders,
// but this API also applies them to receivers, similar to ORTC:
// http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
virtual RtpParameters GetParameters() const = 0;
// TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium.
// Currently, doesn't support changing any parameters.
virtual bool SetParameters(const RtpParameters& parameters) { return false; }
// Does not take ownership of observer.
// Must call SetObserver(nullptr) before the observer is destroyed.
virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
// Sets the jitter buffer minimum delay until media playout. Actual observed
// delay may differ depending on the congestion control. |delay_seconds| is a
// positive value including 0.0 measured in seconds. |nullopt| means default
// value must be used.
virtual void SetJitterBufferMinimumDelay(
absl::optional<double> delay_seconds) = 0;
// TODO(zhihuang): Remove the default implementation once the subclasses
// implement this. Currently, the only relevant subclass is the
// content::FakeRtpReceiver in Chromium.
virtual std::vector<RtpSource> GetSources() const;
// Sets a user defined frame decryptor that will decrypt the entire frame
// before it is sent across the network. This will decrypt the entire frame
// using the user provided decryption mechanism regardless of whether SRTP is
// enabled or not.
virtual void SetFrameDecryptor(
rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
// Returns a pointer to the frame decryptor set previously by the
// user. This can be used to update the state of the object.
virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const;
// Sets a frame transformer between the depacketizer and the decoder to enable
// client code to transform received frames according to their own processing
// logic.
virtual void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
protected:
~RtpReceiverInterface() override = default;
};
// Define proxy for RtpReceiverInterface.
// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
// are called on is an implementation detail.
BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
streams)
PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
PROXY_CONSTMETHOD0(std::string, id)
PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>)
PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
PROXY_METHOD1(void,
SetFrameDecryptor,
rtc::scoped_refptr<FrameDecryptorInterface>)
PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>,
GetFrameDecryptor)
PROXY_METHOD1(void,
SetDepacketizerToDecoderFrameTransformer,
rtc::scoped_refptr<FrameTransformerInterface>)
END_PROXY_MAP()
} // namespace webrtc
#endif // API_RTP_RECEIVER_INTERFACE_H_
|
