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
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
|
/*
* 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"
#include "RealtimeOutgoingAudioSourceGStreamer.h"
#if USE(GSTREAMER_WEBRTC)
#include "GStreamerCommon.h"
#include "GStreamerRegistryScanner.h"
#include "MediaStreamTrack.h"
#include <wtf/text/StringToIntegerConversion.h>
GST_DEBUG_CATEGORY(webkit_webrtc_outgoing_audio_debug);
#define GST_CAT_DEFAULT webkit_webrtc_outgoing_audio_debug
namespace WebCore {
RealtimeOutgoingAudioSourceGStreamer::RealtimeOutgoingAudioSourceGStreamer(const RefPtr<UniqueSSRCGenerator>& ssrcGenerator, const String& mediaStreamId, MediaStreamTrack& track)
: RealtimeOutgoingMediaSourceGStreamer(ssrcGenerator, mediaStreamId, track)
{
static std::once_flag debugRegisteredFlag;
std::call_once(debugRegisteredFlag, [] {
GST_DEBUG_CATEGORY_INIT(webkit_webrtc_outgoing_audio_debug, "webkitwebrtcoutgoingaudio", 0, "WebKit WebRTC outgoing audio");
});
static Atomic<uint64_t> sourceCounter = 0;
gst_element_set_name(m_bin.get(), makeString("outgoing-audio-source-", sourceCounter.exchangeAdd(1)).ascii().data());
m_audioconvert = makeGStreamerElement("audioconvert", nullptr);
m_audioresample = makeGStreamerElement("audioresample", nullptr);
gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_audioconvert.get(), m_audioresample.get(), nullptr);
}
RTCRtpCapabilities RealtimeOutgoingAudioSourceGStreamer::rtpCapabilities() const
{
auto& registryScanner = GStreamerRegistryScanner::singleton();
return registryScanner.audioRtpCapabilities(GStreamerRegistryScanner::Configuration::Encoding);
}
bool RealtimeOutgoingAudioSourceGStreamer::setPayloadType(const GRefPtr<GstCaps>& caps)
{
GST_DEBUG_OBJECT(m_bin.get(), "Setting payload caps: %" GST_PTR_FORMAT, caps.get());
auto* structure = gst_caps_get_structure(caps.get(), 0);
const char* encodingName = gst_structure_get_string(structure, "encoding-name");
if (!encodingName) {
GST_ERROR_OBJECT(m_bin.get(), "encoding-name not found");
return false;
}
auto encoding = makeString(encodingName).convertToASCIILowercase();
m_payloader = makeGStreamerElement(makeString("rtp"_s, encoding, "pay"_s).ascii().data(), nullptr);
if (UNLIKELY(!m_payloader)) {
GST_ERROR_OBJECT(m_bin.get(), "RTP payloader not found for encoding %s", encodingName);
return false;
}
if (encoding == "opus"_s) {
m_encoder = makeGStreamerElement("opusenc", nullptr);
if (!m_encoder)
return false;
// FIXME: Enable dtx too?
gst_util_set_object_arg(G_OBJECT(m_encoder.get()), "audio-type", "voice");
if (const char* useInbandFec = gst_structure_get_string(structure, "useinbandfec")) {
if (!strcmp(useInbandFec, "1"))
g_object_set(m_encoder.get(), "inband-fec", true, nullptr);
}
} else if (encoding == "g722"_s)
m_encoder = makeGStreamerElement("avenc_g722", nullptr);
else if (encoding == "pcma"_s)
m_encoder = makeGStreamerElement("alawenc", nullptr);
else if (encoding == "pcmu"_s)
m_encoder = makeGStreamerElement("mulawenc", nullptr);
else {
GST_ERROR_OBJECT(m_bin.get(), "Unsupported outgoing audio encoding: %s", encodingName);
return false;
}
if (!m_encoder) {
GST_ERROR_OBJECT(m_bin.get(), "Encoder not found for encoding %s", encodingName);
return false;
}
// FIXME: Re-enable auto-header-extension. Currently triggers caps negotiation error.
// Align MTU with libwebrtc implementation, also helping to reduce packet fragmentation.
g_object_set(m_payloader.get(), "auto-header-extension", FALSE, "mtu", 1200, nullptr);
if (const char* minPTime = gst_structure_get_string(structure, "minptime")) {
auto time = String::fromLatin1(minPTime);
if (auto value = parseIntegerAllowingTrailingJunk<int64_t>(time))
g_object_set(m_payloader.get(), "min-ptime", *value * GST_MSECOND, nullptr);
}
int payloadType;
if (gst_structure_get_int(structure, "payload", &payloadType))
g_object_set(m_payloader.get(), "pt", payloadType, nullptr);
g_object_set(m_capsFilter.get(), "caps", caps.get(), nullptr);
gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_payloader.get(), m_encoder.get(), nullptr);
auto preEncoderSinkPad = adoptGRef(gst_element_get_static_pad(m_preEncoderQueue.get(), "sink"));
if (!gst_pad_is_linked(preEncoderSinkPad.get())) {
if (!gst_element_link_many(m_outgoingSource.get(), m_inputSelector.get(), m_audioconvert.get(), m_audioresample.get(), m_preEncoderQueue.get(), nullptr)) {
GST_ERROR_OBJECT(m_bin.get(), "Unable to link outgoing source to pre-encoder queue");
return false;
}
}
return gst_element_link_many(m_preEncoderQueue.get(), m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
}
void RealtimeOutgoingAudioSourceGStreamer::codecPreferencesChanged(const GRefPtr<GstCaps>& codecPreferences)
{
gst_element_set_locked_state(m_bin.get(), TRUE);
if (m_payloader) {
gst_element_set_state(m_payloader.get(), GST_STATE_NULL);
gst_element_set_state(m_encoder.get(), GST_STATE_NULL);
gst_element_unlink_many(m_preEncoderQueue.get(), m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
gst_bin_remove_many(GST_BIN_CAST(m_bin.get()), m_payloader.get(), m_encoder.get(), nullptr);
m_payloader.clear();
m_encoder.clear();
}
if (!setPayloadType(codecPreferences)) {
gst_element_set_locked_state(m_bin.get(), FALSE);
GST_ERROR_OBJECT(m_bin.get(), "Unable to link encoder to webrtcbin");
return;
}
gst_element_set_locked_state(m_bin.get(), FALSE);
gst_bin_sync_children_states(GST_BIN_CAST(m_bin.get()));
gst_element_sync_state_with_parent(m_bin.get());
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN_CAST(m_bin.get()), GST_DEBUG_GRAPH_SHOW_ALL, "outgoing-audio-new-codec-prefs");
m_isStopped = false;
}
void RealtimeOutgoingAudioSourceGStreamer::connectFallbackSource()
{
if (!m_fallbackPad) {
m_fallbackSource = makeGStreamerElement("audiotestsrc", nullptr);
if (!m_fallbackSource) {
WTFLogAlways("Unable to connect fallback audiotestsrc element, expect broken behavior. Please install gst-plugins-base.");
return;
}
gst_util_set_object_arg(G_OBJECT(m_fallbackSource.get()), "wave", "silence");
gst_bin_add(GST_BIN_CAST(m_bin.get()), m_fallbackSource.get());
m_fallbackPad = adoptGRef(gst_element_request_pad_simple(m_inputSelector.get(), "sink_%u"));
auto srcPad = adoptGRef(gst_element_get_static_pad(m_fallbackSource.get(), "src"));
gst_pad_link(srcPad.get(), m_fallbackPad.get());
gst_element_sync_state_with_parent(m_fallbackSource.get());
}
g_object_set(m_inputSelector.get(), "active-pad", m_fallbackPad.get(), nullptr);
}
void RealtimeOutgoingAudioSourceGStreamer::unlinkOutgoingSource()
{
auto srcPad = adoptGRef(gst_element_get_static_pad(m_outgoingSource.get(), "audio_src0"));
auto peerPad = adoptGRef(gst_pad_get_peer(srcPad.get()));
if (!peerPad)
return;
gst_pad_unlink(srcPad.get(), peerPad.get());
gst_element_release_request_pad(m_inputSelector.get(), peerPad.get());
}
void RealtimeOutgoingAudioSourceGStreamer::linkOutgoingSource()
{
auto srcPad = adoptGRef(gst_element_get_static_pad(m_outgoingSource.get(), "audio_src0"));
auto sinkPad = adoptGRef(gst_element_request_pad_simple(m_inputSelector.get(), "sink_%u"));
gst_pad_link(srcPad.get(), sinkPad.get());
g_object_set(m_inputSelector.get(), "active-pad", sinkPad.get(), nullptr);
}
void RealtimeOutgoingAudioSourceGStreamer::setParameters(GUniquePtr<GstStructure>&& parameters)
{
m_parameters = WTFMove(parameters);
}
#undef GST_CAT_DEFAULT
} // namespace WebCore
#endif // USE(GSTREAMER_WEBRTC)
|
