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
|
/*
* Copyright (C) 2021-2022 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 "GStreamerSctpTransportBackend.h"
#if ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
#include "GStreamerDtlsTransportBackend.h"
#include "GStreamerWebRTCUtils.h"
#include <wtf/TZoneMallocInlines.h>
GST_DEBUG_CATEGORY(webkit_webrtc_sctp_transport_debug);
#define GST_CAT_DEFAULT webkit_webrtc_sctp_transport_debug
namespace WebCore {
WTF_MAKE_TZONE_ALLOCATED_IMPL(RTCSctpTransportState);
static inline RTCSctpTransportState toRTCSctpTransportState(GstWebRTCSCTPTransportState state)
{
switch (state) {
case GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW:
case GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING:
return RTCSctpTransportState::Connecting;
case GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED:
return RTCSctpTransportState::Connected;
case GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED:
return RTCSctpTransportState::Closed;
}
RELEASE_ASSERT_NOT_REACHED();
}
GStreamerSctpTransportBackend::GStreamerSctpTransportBackend(GRefPtr<GstWebRTCSCTPTransport>&& transport)
: m_backend(WTFMove(transport))
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_sctp_transport_debug, "webkitwebrtcsctp", 0, "WebKit WebRTC SCTP transport");
});
ASSERT(m_backend);
}
GStreamerSctpTransportBackend::~GStreamerSctpTransportBackend()
{
unregisterClient();
}
UniqueRef<RTCDtlsTransportBackend> GStreamerSctpTransportBackend::dtlsTransportBackend()
{
GRefPtr<GstWebRTCDTLSTransport> transport;
g_object_get(m_backend.get(), "transport", &transport.outPtr(), nullptr);
return makeUniqueRef<GStreamerDtlsTransportBackend>(WTFMove(transport));
}
void GStreamerSctpTransportBackend::registerClient(RTCSctpTransportBackendClient& client)
{
ASSERT(!m_client);
m_client = client;
g_signal_connect_swapped(m_backend.get(), "notify::state", G_CALLBACK(+[](GStreamerSctpTransportBackend* backend) {
backend->stateChanged();
}), this);
}
void GStreamerSctpTransportBackend::unregisterClient()
{
g_signal_handlers_disconnect_by_data(m_backend.get(), this);
m_client.clear();
}
void GStreamerSctpTransportBackend::stateChanged()
{
if (!m_client)
return;
GstWebRTCSCTPTransportState transportState;
guint16 maxChannels;
uint64_t maxMessageSize;
g_object_get(m_backend.get(), "state", &transportState, "max-message-size", &maxMessageSize, "max-channels", &maxChannels, nullptr);
GST_DEBUG("Notifying SCTP transport state, max-message-size: %" G_GUINT64_FORMAT " max-channels: %" G_GUINT16_FORMAT, maxMessageSize, maxChannels);
callOnMainThread([client = m_client, transportState, maxChannels, maxMessageSize] {
if (!client)
return;
client->onStateChanged(toRTCSctpTransportState(transportState), maxMessageSize, maxChannels);
});
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // ENABLE(WEB_RTC) && USE(GSTREAMER_WEBRTC)
|
