File: RealtimeOutgoingVideoSourceGStreamer.cpp

package info (click to toggle)
webkit2gtk 2.42.2-1~deb12u1
  • links: PTS, VCS
  • area: main
  • in suites: bookworm
  • size: 362,452 kB
  • sloc: cpp: 2,881,971; javascript: 282,447; ansic: 134,088; python: 43,789; ruby: 18,308; perl: 15,872; asm: 14,389; xml: 4,395; yacc: 2,350; sh: 2,074; java: 1,734; lex: 1,323; makefile: 288; pascal: 60
file content (385 lines) | stat: -rw-r--r-- 16,584 bytes parent folder | download | duplicates (2)
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
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
/*
 *  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 "RealtimeOutgoingVideoSourceGStreamer.h"

#if USE(GSTREAMER_WEBRTC)

#include "GStreamerCommon.h"
#include "GStreamerRegistryScanner.h"
#include "MediaStreamTrack.h"
#include "VideoEncoderPrivateGStreamer.h"

#include <wtf/glib/WTFGType.h>

GST_DEBUG_CATEGORY(webkit_webrtc_outgoing_video_debug);
#define GST_CAT_DEFAULT webkit_webrtc_outgoing_video_debug

namespace WebCore {

struct RealtimeOutgoingVideoSourceHolder {
    RefPtr<RealtimeOutgoingVideoSourceGStreamer> source;
};
WEBKIT_DEFINE_ASYNC_DATA_STRUCT(RealtimeOutgoingVideoSourceHolder)


RealtimeOutgoingVideoSourceGStreamer::RealtimeOutgoingVideoSourceGStreamer(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_video_debug, "webkitwebrtcoutgoingvideo", 0, "WebKit WebRTC outgoing video");
    });
    registerWebKitGStreamerElements();

    static Atomic<uint64_t> sourceCounter = 0;
    gst_element_set_name(m_bin.get(), makeString("outgoing-video-source-", sourceCounter.exchangeAdd(1)).ascii().data());

    m_stats.reset(gst_structure_new_empty("webrtc-outgoing-video-stats"));
    startUpdatingStats();

    m_videoConvert = makeGStreamerElement("videoconvert", nullptr);

    m_videoFlip = makeGStreamerElement("videoflip", nullptr);
    gst_util_set_object_arg(G_OBJECT(m_videoFlip.get()), "method", "automatic");

    // Variable framerate for canvas capture tracks requires further investigation, so disable it for now.
    if (!track.isCanvas()) {
        m_videoRate = makeGStreamerElement("videorate", nullptr);
        g_object_set(m_videoRate.get(), "skip-to-first", TRUE, nullptr);
        m_frameRateCapsFilter = makeGStreamerElement("capsfilter", nullptr);
        gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_videoRate.get(), m_frameRateCapsFilter.get(), nullptr);
    }

    m_encoder = gst_element_factory_make("webkitvideoencoder", nullptr);
    gst_bin_add_many(GST_BIN_CAST(m_bin.get()), m_videoFlip.get(), m_videoConvert.get(), m_encoder.get(), nullptr);
}

RTCRtpCapabilities RealtimeOutgoingVideoSourceGStreamer::rtpCapabilities() const
{
    auto& registryScanner = GStreamerRegistryScanner::singleton();
    return registryScanner.videoRtpCapabilities(GStreamerRegistryScanner::Configuration::Encoding);
}

void RealtimeOutgoingVideoSourceGStreamer::updateStats(GstBuffer*)
{
    uint64_t framesSent = 0;
    gst_structure_get_uint64(m_stats.get(), "frames-sent", &framesSent);
    framesSent++;

    if (m_encoder) {
        uint32_t bitrate;
        g_object_get(m_encoder.get(), "bitrate", &bitrate, nullptr);
        gst_structure_set(m_stats.get(), "bitrate", G_TYPE_DOUBLE, static_cast<double>(bitrate * 1024), nullptr);
    }

    gst_structure_set(m_stats.get(), "frames-sent", G_TYPE_UINT64, framesSent, "frames-encoded", G_TYPE_UINT64, framesSent, nullptr);
}

void RealtimeOutgoingVideoSourceGStreamer::teardown()
{
    stopUpdatingStats();
}

bool RealtimeOutgoingVideoSourceGStreamer::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;
    }

    GRefPtr<GstCaps> encoderCaps;
    if (encoding == "vp8"_s) {
        if (gstObjectHasProperty(m_payloader.get(), "picture-id-mode"))
            gst_util_set_object_arg(G_OBJECT(m_payloader.get()), "picture-id-mode", "15-bit");

        encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-vp8"));
    } else if (encoding == "vp9"_s) {
        if (gstObjectHasProperty(m_payloader.get(), "picture-id-mode"))
            gst_util_set_object_arg(G_OBJECT(m_payloader.get()), "picture-id-mode", "15-bit");

        encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-vp9"));
        if (const char* vp9Profile = gst_structure_get_string(structure, "vp9-profile-id"))
            gst_caps_set_simple(encoderCaps.get(), "profile", G_TYPE_STRING, vp9Profile, nullptr);
    } else if (encoding == "h264"_s) {
        encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-h264"));
        // FIXME: https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/893
        // gst_util_set_object_arg(G_OBJECT(m_payloader.get()), "aggregate-mode", "zero-latency");
        // g_object_set(m_payloader.get(), "config-interval", -1, nullptr);

        const char* profile = gst_structure_get_string(structure, "profile");
        if (!profile)
            profile = "baseline";
        gst_caps_set_simple(encoderCaps.get(), "profile", G_TYPE_STRING, profile, nullptr);
    } else if (encoding == "h265"_s) {
        encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-h265"));
        // FIXME: profile tier level?
    } else if (encoding == "av1"_s) {
        encoderCaps = adoptGRef(gst_caps_new_empty_simple("video/x-av1"));
    } else {
        GST_ERROR_OBJECT(m_bin.get(), "Unsupported outgoing video 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 (!videoEncoderSetFormat(WEBKIT_VIDEO_ENCODER(m_encoder.get()), WTFMove(encoderCaps))) {
        GST_ERROR_OBJECT(m_bin.get(), "Unable to set encoder format");
        return false;
    }

    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(GST_BIN_CAST(m_bin.get()), m_payloader.get());

    auto encoderSinkPad = adoptGRef(gst_element_get_static_pad(m_encoder.get(), "sink"));
    if (!gst_pad_is_linked(encoderSinkPad.get())) {
        if (!gst_element_link_many(m_outgoingSource.get(), m_inputSelector.get(), m_videoFlip.get(), nullptr)) {
            GST_ERROR_OBJECT(m_bin.get(), "Unable to link outgoing source to videoflip");
            return false;
        }

        GstElement* tail = m_videoFlip.get();
        if (m_videoRate) {
            if (!gst_element_link_many(m_videoFlip.get(), m_videoRate.get(), m_frameRateCapsFilter.get(), nullptr)) {
                GST_ERROR_OBJECT(m_bin.get(), "Unable to link outgoing source to videorate");
                return false;
            }
            tail = m_frameRateCapsFilter.get();
        }
        if (!gst_element_link_many(tail, m_videoConvert.get(), m_preEncoderQueue.get(), m_encoder.get(), nullptr)) {
            GST_ERROR_OBJECT(m_bin.get(), "Unable to link outgoing source to encoder");
            return false;
        }
    }

    return gst_element_link_many(m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
}

void RealtimeOutgoingVideoSourceGStreamer::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_unlink_many(m_encoder.get(), m_payloader.get(), m_postEncoderQueue.get(), nullptr);
        gst_bin_remove(GST_BIN_CAST(m_bin.get()), m_payloader.get());
        m_payloader.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-video-new-codec-prefs");
    m_isStopped = false;
}

void RealtimeOutgoingVideoSourceGStreamer::connectFallbackSource()
{
    GST_DEBUG_OBJECT(m_bin.get(), "Connecting fallback video source");
    if (!m_fallbackPad) {
        m_fallbackSource = makeGStreamerElement("videotestsrc", nullptr);
        if (!m_fallbackSource) {
            WTFLogAlways("Unable to connect fallback videotestsrc element, expect broken behavior. Please install gst-plugins-base.");
            return;
        }

        gst_util_set_object_arg(G_OBJECT(m_fallbackSource.get()), "pattern", "black");

        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 RealtimeOutgoingVideoSourceGStreamer::unlinkOutgoingSource()
{
    GST_DEBUG_OBJECT(m_bin.get(), "Unlinking outgoing video source");
    if (m_statsPadProbeId) {
        auto binSrcPad = adoptGRef(gst_element_get_static_pad(m_bin.get(), "src"));
        gst_pad_remove_probe(binSrcPad.get(), m_statsPadProbeId);
        m_statsPadProbeId = 0;
    }

    auto srcPad = adoptGRef(gst_element_get_static_pad(m_outgoingSource.get(), "video_src0"));
    auto peerPad = adoptGRef(gst_pad_get_peer(srcPad.get()));
    if (!peerPad) {
        GST_DEBUG_OBJECT(m_bin.get(), "Outgoing video source not linked");
        return;
    }

    gst_pad_unlink(srcPad.get(), peerPad.get());
    gst_element_release_request_pad(m_inputSelector.get(), peerPad.get());
}

void RealtimeOutgoingVideoSourceGStreamer::linkOutgoingSource()
{
    GST_DEBUG_OBJECT(m_bin.get(), "Linking outgoing video source");
    auto srcPad = adoptGRef(gst_element_get_static_pad(m_outgoingSource.get(), "video_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);

    flush();
}

void RealtimeOutgoingVideoSourceGStreamer::startUpdatingStats()
{
    GST_DEBUG_OBJECT(m_bin.get(), "Starting buffer monitoring for stats gathering");
    auto holder = createRealtimeOutgoingVideoSourceHolder();
    holder->source = this;
    auto pad = adoptGRef(gst_element_get_static_pad(m_bin.get(), "src"));
    m_statsPadProbeId = gst_pad_add_probe(pad.get(), GST_PAD_PROBE_TYPE_BUFFER, [](GstPad*, GstPadProbeInfo* info, gpointer userData) -> GstPadProbeReturn {
        auto* holder = static_cast<RealtimeOutgoingVideoSourceHolder*>(userData);
        auto* buffer = GST_PAD_PROBE_INFO_BUFFER(info);
        holder->source->updateStats(buffer);
        return GST_PAD_PROBE_OK;
    }, holder, reinterpret_cast<GDestroyNotify>(destroyRealtimeOutgoingVideoSourceHolder));
}

void RealtimeOutgoingVideoSourceGStreamer::stopUpdatingStats()
{
    if (!m_statsPadProbeId)
        return;

    GST_DEBUG_OBJECT(m_bin.get(), "Stopping buffer monitoring for stats gathering");
    auto binSrcPad = adoptGRef(gst_element_get_static_pad(m_bin.get(), "src"));
    gst_pad_remove_probe(binSrcPad.get(), m_statsPadProbeId);
    m_statsPadProbeId = 0;
}

void RealtimeOutgoingVideoSourceGStreamer::sourceEnabledChanged()
{
    RealtimeOutgoingMediaSourceGStreamer::sourceEnabledChanged();
    if (m_enabled)
        startUpdatingStats();
    else
        stopUpdatingStats();
}

void RealtimeOutgoingVideoSourceGStreamer::flush()
{
    GST_DEBUG_OBJECT(m_bin.get(), "Requesting key-frame");
    gst_element_send_event(m_outgoingSource.get(), gst_video_event_new_downstream_force_key_unit(GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE, FALSE, 1));
}

void RealtimeOutgoingVideoSourceGStreamer::setParameters(GUniquePtr<GstStructure>&& parameters)
{
    m_parameters = WTFMove(parameters);
    GST_DEBUG_OBJECT(m_bin.get(), "New encoding parameters: %" GST_PTR_FORMAT, m_parameters.get());

    auto* encodingsValue = gst_structure_get_value(m_parameters.get(), "encodings");
    RELEASE_ASSERT(GST_VALUE_HOLDS_LIST(encodingsValue));
    if (UNLIKELY(!gst_value_list_get_size(encodingsValue))) {
        GST_WARNING_OBJECT(m_bin.get(), "Encodings list is empty, cancelling configuration");
        return;
    }

    auto* firstEncoding = gst_value_list_get_value(encodingsValue, 0);
    RELEASE_ASSERT(GST_VALUE_HOLDS_STRUCTURE(firstEncoding));
    auto* structure = gst_value_get_structure(firstEncoding);

    if (gst_structure_has_field(structure, "max-framerate")) {
        if (!m_videoRate)
            GST_WARNING_OBJECT(m_bin.get(), "Unable to configure max-framerate");
        else {
            unsigned long maxFrameRate;
            gst_structure_get(structure, "max-framerate", G_TYPE_ULONG, &maxFrameRate, nullptr);

            // Some decoder(s), like FFMpeg don't handle 1 FPS framerate, so set a minimum more likely to be accepted.
            if (maxFrameRate < 2)
                maxFrameRate = 2;

            int numerator, denominator;
            gst_util_double_to_fraction(static_cast<double>(maxFrameRate), &numerator, &denominator);

            auto caps = adoptGRef(gst_caps_new_simple("video/x-raw", "framerate", GST_TYPE_FRACTION, numerator, denominator, nullptr));
            g_object_set(m_frameRateCapsFilter.get(), "caps", caps.get(), nullptr);
        }
    }

    if (UNLIKELY(!m_encoder) || !gst_structure_has_field(structure, "max-bitrate"))
        return;

    unsigned long maxBitrate;
    gst_structure_get(structure, "max-bitrate", G_TYPE_ULONG, &maxBitrate, nullptr);

    // maxBitrate is expessed in bits/s but the encoder property is in Kbit/s.
    g_object_set(m_encoder.get(), "bitrate", static_cast<unsigned>(maxBitrate / 1024), nullptr);
}

void RealtimeOutgoingVideoSourceGStreamer::fillEncodingParameters(const GUniquePtr<GstStructure>& encodingParameters)
{
    if (m_videoRate) {
        GRefPtr<GstCaps> caps;
        g_object_get(m_frameRateCapsFilter.get(), "caps", &caps.outPtr(), nullptr);
        double maxFrameRate = 30.0;
        if (!gst_caps_is_any(caps.get())) {
            if (auto* structure = gst_caps_get_structure(caps.get(), 0)) {
                int numerator, denominator;
                if (gst_structure_get_fraction(structure, "framerate", &numerator, &denominator))
                    gst_util_fraction_to_double(numerator, denominator, &maxFrameRate);
            }
        }

        gst_structure_set(encodingParameters.get(), "max-framerate", G_TYPE_DOUBLE, maxFrameRate, nullptr);
    }

    unsigned long maxBitrate = 2048 * 1024;
    if (m_encoder) {
        uint32_t bitrate;
        g_object_get(m_encoder.get(), "bitrate", &bitrate, nullptr);
        maxBitrate = bitrate * 1024;
    }

    gst_structure_set(encodingParameters.get(), "max-bitrate", G_TYPE_ULONG, maxBitrate, nullptr);
}

#undef GST_CAT_DEFAULT

} // namespace WebCore

#endif // USE(GSTREAMER_WEBRTC)