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
|
/*
* Copyright 2025 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.
*/
// Option structures for MediaSession APIs.
#ifndef PC_MEDIA_OPTIONS_H_
#define PC_MEDIA_OPTIONS_H_
#include <string>
#include <vector>
#include "api/crypto/crypto_options.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "media/base/codec.h"
#include "media/base/rid_description.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_description_factory.h"
#include "pc/simulcast_description.h"
namespace webrtc {
// Default RTCP CNAME for unit tests.
const char kDefaultRtcpCname[] = "DefaultRtcpCname";
// Options for an RtpSender contained with an media description/"m=" section.
// Note: Spec-compliant Simulcast and legacy simulcast are mutually exclusive.
struct SenderOptions {
std::string track_id;
std::vector<std::string> stream_ids;
// Use RIDs and Simulcast Layers to indicate spec-compliant Simulcast.
std::vector<RidDescription> rids;
SimulcastLayerList simulcast_layers;
// Use `num_sim_layers` to indicate legacy simulcast.
int num_sim_layers;
};
// Options for an individual media description/"m=" section.
struct MediaDescriptionOptions {
MediaDescriptionOptions(MediaType type,
const std::string& mid,
RtpTransceiverDirection direction,
bool stopped)
: type(type), mid(mid), direction(direction), stopped(stopped) {}
// TODO(deadbeef): When we don't support Plan B, there will only be one
// sender per media description and this can be simplified.
void AddAudioSender(const std::string& track_id,
const std::vector<std::string>& stream_ids);
void AddVideoSender(const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<RidDescription>& rids,
const SimulcastLayerList& simulcast_layers,
int num_sim_layers);
MediaType type;
std::string mid;
RtpTransceiverDirection direction;
bool stopped;
TransportOptions transport_options;
// Note: There's no equivalent "RtpReceiverOptions" because only send
// stream information goes in the local descriptions.
std::vector<SenderOptions> sender_options;
std::vector<RtpCodecCapability> codec_preferences;
std::vector<RtpHeaderExtensionCapability> header_extensions;
// Codecs to include in a generated offer or answer.
// If this is used, session-level codec lists MUST be ignored.
std::vector<Codec> codecs_to_include;
private:
// Doesn't DCHECK on `type`.
void AddSenderInternal(const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<RidDescription>& rids,
const SimulcastLayerList& simulcast_layers,
int num_sim_layers);
};
// Provides a mechanism for describing how m= sections should be generated.
// The m= section with index X will use media_description_options[X]. There
// must be an option for each existing section if creating an answer, or a
// subsequent offer.
struct MediaSessionOptions {
MediaSessionOptions() {}
bool has_audio() const { return HasMediaDescription(MediaType::AUDIO); }
bool has_video() const { return HasMediaDescription(MediaType::VIDEO); }
bool has_data() const { return HasMediaDescription(MediaType::DATA); }
bool HasMediaDescription(MediaType type) const;
bool vad_enabled = true; // When disabled, removes all CN codecs from SDP.
bool rtcp_mux_enabled = true;
bool bundle_enabled = false;
bool offer_extmap_allow_mixed = false;
bool raw_packetization_for_video = false;
std::string rtcp_cname = kDefaultRtcpCname;
CryptoOptions crypto_options;
// List of media description options in the same order that the media
// descriptions will be generated.
std::vector<MediaDescriptionOptions> media_description_options;
std::vector<IceParameters> pooled_ice_credentials;
// Use the draft-ietf-mmusic-sctp-sdp-03 obsolete syntax for SCTP
// datachannels.
// Default is true for backwards compatibility with clients that use
// this internal interface.
bool use_obsolete_sctp_sdp = true;
};
} // namespace webrtc
// Re-export symbols from the webrtc namespace for backwards compatibility.
// TODO(bugs.webrtc.org/4222596): Remove once all references are updated.
#ifdef WEBRTC_ALLOW_DEPRECATED_NAMESPACES
namespace cricket {
using ::webrtc::kDefaultRtcpCname;
using ::webrtc::MediaDescriptionOptions;
using ::webrtc::MediaSessionOptions;
using ::webrtc::SenderOptions;
} // namespace cricket
#endif // WEBRTC_ALLOW_DEPRECATED_NAMESPACES
#endif // PC_MEDIA_OPTIONS_H_
|
