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
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
|
/*
* Copyright (C) 2017-2022 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"
#if USE(GSTREAMER_WEBRTC)
#include "RealtimeIncomingSourceGStreamer.h"
#include "GStreamerCommon.h"
#include "NotImplemented.h"
#include <gst/app/gstappsink.h>
#include <wtf/text/MakeString.h>
#include <wtf/text/WTFString.h>
GST_DEBUG_CATEGORY(webkit_webrtc_incoming_media_debug);
#define GST_CAT_DEFAULT webkit_webrtc_incoming_media_debug
namespace WebCore {
RealtimeIncomingSourceGStreamer::RealtimeIncomingSourceGStreamer(const CaptureDevice& device)
: RealtimeMediaSource(device)
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_incoming_media_debug, "webkitwebrtcincoming", 0, "WebKit WebRTC incoming media");
});
}
bool RealtimeIncomingSourceGStreamer::setBin(const GRefPtr<GstElement>& bin)
{
ASSERT(!m_bin);
if (UNLIKELY(m_bin)) {
GST_ERROR_OBJECT(m_bin.get(), "Calling setBin twice on the same incoming source instance is not allowed");
return false;
}
m_bin = bin;
m_sink = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_bin.get()), "sink"));
g_object_set(m_sink.get(), "signal-handoffs", TRUE, nullptr);
auto handoffCallback = G_CALLBACK(+[](GstElement*, GstBuffer* buffer, GstPad* pad, gpointer userData) {
auto source = reinterpret_cast<RealtimeIncomingSourceGStreamer*>(userData);
auto caps = adoptGRef(gst_pad_get_current_caps(pad));
auto sample = adoptGRef(gst_sample_new(buffer, caps.get(), nullptr, nullptr));
// dispatchSample might trigger RealtimeMediaSource::notifySettingsDidChangeObservers()
// which expects to run in the main thread.
callOnMainThread([source, sample = WTFMove(sample)]() mutable {
source->dispatchSample(WTFMove(sample));
});
});
g_signal_connect(m_sink.get(), "preroll-handoff", handoffCallback, this);
g_signal_connect(m_sink.get(), "handoff", handoffCallback, this);
auto sinkPad = adoptGRef(gst_element_get_static_pad(m_sink.get(), "sink"));
gst_pad_add_probe(sinkPad.get(), static_cast<GstPadProbeType>(GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM), reinterpret_cast<GstPadProbeCallback>(+[](GstPad* pad, GstPadProbeInfo* info, gpointer userData) -> GstPadProbeReturn {
auto self = reinterpret_cast<RealtimeIncomingSourceGStreamer*>(userData);
if (info->type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) {
GRefPtr event = GST_PAD_PROBE_INFO_EVENT(info);
auto sink = adoptGRef(gst_pad_get_parent_element(pad));
self->handleDownstreamEvent(sink.get(), WTFMove(event));
return GST_PAD_PROBE_OK;
}
auto query = GST_PAD_PROBE_INFO_QUERY(info);
self->forEachClient([&](auto* appsrc) {
auto srcSrcPad = adoptGRef(gst_element_get_static_pad(appsrc, "src"));
gst_pad_peer_query(srcSrcPad.get(), query);
});
return GST_PAD_PROBE_OK;
}), this, nullptr);
return true;
}
const RealtimeMediaSourceCapabilities& RealtimeIncomingSourceGStreamer::capabilities()
{
return RealtimeMediaSourceCapabilities::emptyCapabilities();
}
bool RealtimeIncomingSourceGStreamer::hasClient(const GRefPtr<GstElement>& appsrc)
{
Locker lock { m_clientLock };
for (auto& client : m_clients.values()) {
if (client == appsrc)
return true;
}
return false;
}
int RealtimeIncomingSourceGStreamer::registerClient(GRefPtr<GstElement>&& appsrc)
{
Locker lock { m_clientLock };
static Atomic<int> counter = 1;
auto clientId = counter.exchangeAdd(1);
m_clients.add(clientId, WTFMove(appsrc));
return clientId;
}
void RealtimeIncomingSourceGStreamer::unregisterClient(int clientId)
{
Locker lock { m_clientLock };
GST_DEBUG_OBJECT(m_bin.get(), "Unregistering client %d", clientId);
m_clients.remove(clientId);
}
void RealtimeIncomingSourceGStreamer::forEachClient(Function<void(GstElement*)>&& applyFunction)
{
Locker lock { m_clientLock };
for (auto& client : m_clients.values())
applyFunction(client.get());
}
void RealtimeIncomingSourceGStreamer::handleUpstreamEvent(GRefPtr<GstEvent>&& event)
{
RELEASE_ASSERT(m_bin);
GST_DEBUG_OBJECT(m_bin.get(), "Handling %" GST_PTR_FORMAT, event.get());
auto pad = adoptGRef(gst_element_get_static_pad(m_sink.get(), "sink"));
gst_pad_push_event(pad.get(), event.leakRef());
}
bool RealtimeIncomingSourceGStreamer::handleUpstreamQuery(GstQuery* query)
{
RELEASE_ASSERT(m_bin);
GST_DEBUG_OBJECT(m_bin.get(), "Handling %" GST_PTR_FORMAT, query);
auto pad = adoptGRef(gst_element_get_static_pad(m_sink.get(), "sink"));
return gst_pad_peer_query(pad.get(), query);
}
void RealtimeIncomingSourceGStreamer::handleDownstreamEvent(GstElement* sink, GRefPtr<GstEvent>&& event)
{
switch (GST_EVENT_TYPE(event.get())) {
case GST_EVENT_STREAM_START:
case GST_EVENT_CAPS:
case GST_EVENT_SEGMENT:
case GST_EVENT_STREAM_COLLECTION:
return;
case GST_EVENT_LATENCY: {
GstClockTime minLatency, maxLatency;
if (gst_base_sink_query_latency(GST_BASE_SINK(sink), nullptr, nullptr, &minLatency, &maxLatency)) {
forEachClient([&](auto* appsrc) {
GST_DEBUG_OBJECT(sink, "Setting client latency to min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT, GST_TIME_ARGS(minLatency), GST_TIME_ARGS(maxLatency));
g_object_set(appsrc, "min-latency", minLatency, "max-latency", maxLatency, nullptr);
});
}
return;
}
default:
break;
}
forEachClient([&](auto* appsrc) {
auto pad = adoptGRef(gst_element_get_static_pad(appsrc, "src"));
GRefPtr eventCopy(event);
GST_DEBUG_OBJECT(sink, "Forwarding event %" GST_PTR_FORMAT " to client", eventCopy.get());
gst_pad_push_event(pad.get(), eventCopy.leakRef());
});
}
void RealtimeIncomingSourceGStreamer::tearDown()
{
notImplemented();
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
|
