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
|
/*
* 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 "GStreamerRtpReceiverBackend.h"
#if ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
#include "GStreamerDtlsTransportBackend.h"
#include "GStreamerRtpReceiverTransformBackend.h"
#include "GStreamerWebRTCUtils.h"
#include "NotImplemented.h"
#include "RealtimeIncomingAudioSourceGStreamer.h"
#include "RealtimeIncomingVideoSourceGStreamer.h"
#include <wtf/TZoneMallocInlines.h>
#include <wtf/glib/GUniquePtr.h>
#include <wtf/text/StringToIntegerConversion.h>
GST_DEBUG_CATEGORY(webkit_webrtc_rtp_receiver_debug);
#define GST_CAT_DEFAULT webkit_webrtc_rtp_receiver_debug
namespace WebCore {
WTF_MAKE_TZONE_ALLOCATED_IMPL(GStreamerRtpReceiverBackend);
GStreamerRtpReceiverBackend::GStreamerRtpReceiverBackend(GRefPtr<GstWebRTCRTPTransceiver>&& rtcTransceiver)
: m_rtcTransceiver(WTFMove(rtcTransceiver))
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_rtp_receiver_debug, "webkitwebrtcrtpreceiver", 0, "WebKit WebRTC RTP Receiver");
});
g_object_get(m_rtcTransceiver.get(), "receiver", &m_rtcReceiver.outPtr(), nullptr);
}
RTCRtpParameters GStreamerRtpReceiverBackend::getParameters()
{
RTCRtpParameters parameters;
parameters.rtcp.reducedSize = true;
GRefPtr<GstCaps> caps;
g_object_get(m_rtcTransceiver.get(), "codec-preferences", &caps.outPtr(), nullptr);
if (!caps || gst_caps_is_any(caps.get()))
return parameters;
unsigned totalCodecs = gst_caps_get_size(caps.get());
for (unsigned i = 0; i < totalCodecs; i++) {
auto structure = gst_caps_get_structure(caps.get(), i);
RTCRtpCodecParameters codec;
if (auto pt = gstStructureGet<int>(structure, "payload"_s))
codec.payloadType = *pt;
auto media = gstStructureGetString(structure, "media"_s);
auto encodingName = gstStructureGetString(structure, "encoding-name"_s);
if (media && encodingName)
codec.mimeType = makeString(media, '/', encodingName.convertToASCIILowercase());
if (auto clockRate = gstStructureGet<uint64_t>(structure, "clock-rate"_s))
codec.clockRate = *clockRate;
if (auto channels = gstStructureGet<unsigned>(structure, "channels"_s))
codec.channels = *channels;
if (auto fmtpLine = gstStructureGetString(structure, "fmtp-line"_s))
codec.sdpFmtpLine = fmtpLine.toString();
parameters.codecs.append(WTFMove(codec));
gstStructureForeach(structure, [&](auto id, const auto value) -> bool {
auto name = gstIdToString(id);
if (!name.startsWith("extmap-"_s))
return true;
auto extensionId = parseInteger<unsigned short>(name.toStringWithoutCopying().substring(7));
if (!extensionId)
return true;
auto uri = String::fromLatin1(g_value_get_string(value));
parameters.headerExtensions.append({ uri, *extensionId });
return true;
});
}
return parameters;
}
Vector<RTCRtpContributingSource> GStreamerRtpReceiverBackend::getContributingSources() const
{
notImplemented();
return { };
}
Vector<RTCRtpSynchronizationSource> GStreamerRtpReceiverBackend::getSynchronizationSources() const
{
notImplemented();
return { };
}
Ref<RealtimeMediaSource> GStreamerRtpReceiverBackend::createSource(const String& trackKind, const String& trackId)
{
if (trackKind == "video"_s)
return RealtimeIncomingVideoSourceGStreamer::create(AtomString { trackId });
RELEASE_ASSERT(trackKind == "audio"_s);
return RealtimeIncomingAudioSourceGStreamer::create(AtomString { trackId });
}
Ref<RTCRtpTransformBackend> GStreamerRtpReceiverBackend::rtcRtpTransformBackend()
{
return GStreamerRtpReceiverTransformBackend::create(m_rtcReceiver);
}
std::unique_ptr<RTCDtlsTransportBackend> GStreamerRtpReceiverBackend::dtlsTransportBackend()
{
GRefPtr<GstWebRTCDTLSTransport> transport;
g_object_get(m_rtcReceiver.get(), "transport", &transport.outPtr(), nullptr);
if (!transport)
return nullptr;
return makeUnique<GStreamerDtlsTransportBackend>(WTFMove(transport));
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
|
