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
|
/*
* Copyright (C) 2019-2022 Igalia S.L. All rights reserved.
* Copyright (C) 2022 Metrological Group B.V.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "GStreamerRtpSenderBackend.h"
#if ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
#include "GStreamerDTMFSenderBackend.h"
#include "GStreamerDtlsTransportBackend.h"
#include "GStreamerPeerConnectionBackend.h"
#include "GStreamerRtpSenderTransformBackend.h"
#include "GStreamerWebRTCUtils.h"
#include "JSDOMPromiseDeferred.h"
#include "NotImplemented.h"
#include "RTCPeerConnection.h"
#include "RTCRtpSender.h"
#include "RealtimeOutgoingAudioSourceGStreamer.h"
#include "RealtimeOutgoingVideoSourceGStreamer.h"
#include "ScriptExecutionContext.h"
namespace WebCore {
template<typename Source>
static inline bool updateTrackSource(Source& source, MediaStreamTrack* track)
{
if (!track) {
source.stop();
return true;
}
return source.setTrack(track->privateTrack());
}
void GStreamerRtpSenderBackend::startSource()
{
switchOn(m_source, [](Ref<RealtimeOutgoingAudioSourceGStreamer>& source) {
source->start();
}, [](Ref<RealtimeOutgoingVideoSourceGStreamer>& source) {
source->start();
}, [](std::nullptr_t&) {
});
}
WARN_UNUSED_RETURN GRefPtr<GstElement> GStreamerRtpSenderBackend::stopSource()
{
switchOn(m_source, [](Ref<RealtimeOutgoingAudioSourceGStreamer>& source) {
source->stop();
return GRefPtr<GstElement>(source->bin());
}, [](Ref<RealtimeOutgoingVideoSourceGStreamer>& source) {
source->stop();
return GRefPtr<GstElement>(source->bin());
}, [](std::nullptr_t&) {
return GRefPtr<GstElement>(nullptr);
});
return nullptr;
}
bool GStreamerRtpSenderBackend::replaceTrack(RTCRtpSender& sender, MediaStreamTrack* track)
{
if (!track) {
auto stoppedSource = stopSource();
return true;
}
if (sender.track()) {
switchOn(m_source, [&](Ref<RealtimeOutgoingAudioSourceGStreamer>& source) {
ASSERT(track->source().type() == RealtimeMediaSource::Type::Audio);
source->stop();
source->setSource(track->privateTrack());
source->start();
}, [&](Ref<RealtimeOutgoingVideoSourceGStreamer>& source) {
ASSERT(track->source().type() == RealtimeMediaSource::Type::Video);
source->stop();
source->setSource(track->privateTrack());
source->start();
}, [](std::nullptr_t&) {
});
}
m_peerConnectionBackend->setSenderSourceFromTrack(*this, *track);
return true;
}
RTCRtpSendParameters GStreamerRtpSenderBackend::getParameters() const
{
return toRTCRtpSendParameters(m_initData.get());
}
void GStreamerRtpSenderBackend::setParameters(const RTCRtpSendParameters&, DOMPromiseDeferred<void>&& promise)
{
if (!m_rtcSender) {
promise.reject(NotSupportedError);
return;
}
notImplemented();
promise.resolve();
}
std::unique_ptr<RTCDTMFSenderBackend> GStreamerRtpSenderBackend::createDTMFBackend()
{
return makeUnique<GStreamerDTMFSenderBackend>();
}
Ref<RTCRtpTransformBackend> GStreamerRtpSenderBackend::rtcRtpTransformBackend()
{
return GStreamerRtpSenderTransformBackend::create(m_rtcSender);
}
void GStreamerRtpSenderBackend::setMediaStreamIds(const FixedVector<String>&)
{
notImplemented();
}
std::unique_ptr<RTCDtlsTransportBackend> GStreamerRtpSenderBackend::dtlsTransportBackend()
{
GRefPtr<GstWebRTCDTLSTransport> transport;
g_object_get(m_rtcSender.get(), "transport", &transport.outPtr(), nullptr);
return transport ? makeUnique<GStreamerDtlsTransportBackend>(transport) : nullptr;
}
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
|
