File: GStreamerWebRTCProvider.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 (141 lines) | stat: -rw-r--r-- 5,252 bytes parent folder | download | duplicates (5)
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
/*
 * Copyright (C) 2022 Metrological Group B.V.
 * Author: Philippe Normand <philn@igalia.com>
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library 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
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public License
 * aint with this library; see the file COPYING.LIB.  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 "GStreamerWebRTCProvider.h"

#include "ContentType.h"
#include "GStreamerRegistryScanner.h"
#include "MediaCapabilitiesDecodingInfo.h"
#include "MediaCapabilitiesEncodingInfo.h"
#include "MediaDecodingConfiguration.h"
#include "MediaEncodingConfiguration.h"
#include "NotImplemented.h"

namespace WebCore {

void WebRTCProvider::setH264HardwareEncoderAllowed(bool)
{
    // TODO: Hook this into GStreamerRegistryScanner.
    notImplemented();
}

UniqueRef<WebRTCProvider> WebRTCProvider::create()
{
    return makeUniqueRef<GStreamerWebRTCProvider>();
}

bool WebRTCProvider::webRTCAvailable()
{
    return true;
}

void WebRTCProvider::setActive(bool)
{
    notImplemented();
}

std::optional<RTCRtpCapabilities> GStreamerWebRTCProvider::receiverCapabilities(const String& kind)
{
    if (kind == "audio"_s)
        return audioDecodingCapabilities();
    if (kind == "video"_s)
        return videoDecodingCapabilities();

    return { };
}

std::optional<RTCRtpCapabilities> GStreamerWebRTCProvider::senderCapabilities(const String& kind)
{
    if (kind == "audio"_s)
        return audioEncodingCapabilities();
    if (kind == "video"_s)
        return videoEncodingCapabilities();
    return { };
}

void GStreamerWebRTCProvider::initializeAudioEncodingCapabilities()
{
    m_audioEncodingCapabilities = GStreamerRegistryScanner::singleton().audioRtpCapabilities(GStreamerRegistryScanner::Configuration::Encoding);
}

void GStreamerWebRTCProvider::initializeVideoEncodingCapabilities()
{
    ensureGStreamerInitialized();
    registerWebKitGStreamerVideoEncoder();
    m_videoEncodingCapabilities = GStreamerRegistryScanner::singleton().videoRtpCapabilities(GStreamerRegistryScanner::Configuration::Encoding);
    m_videoEncodingCapabilities->codecs.removeAllMatching([isSupportingVP9Profile0 = isSupportingVP9Profile0(), isSupportingVP9Profile2 = isSupportingVP9Profile2(), isSupportingH265 = isSupportingH265()](const auto& codec) {
        if (!isSupportingVP9Profile0 && codec.sdpFmtpLine == "profile-id=0"_s)
            return true;
        if (!isSupportingVP9Profile2 && codec.sdpFmtpLine == "profile-id=2"_s)
            return true;
        if (!isSupportingH265 && codec.mimeType == "video/H265"_s)
            return true;

        return false;
    });
}

void GStreamerWebRTCProvider::initializeAudioDecodingCapabilities()
{
    m_audioDecodingCapabilities = GStreamerRegistryScanner::singleton().audioRtpCapabilities(GStreamerRegistryScanner::Configuration::Decoding);
}

void GStreamerWebRTCProvider::initializeVideoDecodingCapabilities()
{
    m_videoDecodingCapabilities = GStreamerRegistryScanner::singleton().videoRtpCapabilities(GStreamerRegistryScanner::Configuration::Decoding);
    m_videoDecodingCapabilities->codecs.removeAllMatching([isSupportingVP9Profile0 = isSupportingVP9Profile0(), isSupportingVP9Profile2 = isSupportingVP9Profile2(), isSupportingH265 = isSupportingH265()](const auto& codec) {
        if (!isSupportingVP9Profile0 && codec.sdpFmtpLine == "profile-id=0"_s)
            return true;
        if (!isSupportingVP9Profile2 && codec.sdpFmtpLine == "profile-id=2"_s)
            return true;
        if (!isSupportingH265 && codec.mimeType == "video/H265"_s)
            return true;

        return false;
    });
}

std::optional<MediaCapabilitiesDecodingInfo> GStreamerWebRTCProvider::videoDecodingCapabilitiesOverride(const VideoConfiguration& configuration)
{
    MediaCapabilitiesDecodingInfo info;
    ContentType contentType { configuration.contentType };
    auto containerType = contentType.containerType();
    if (equalLettersIgnoringASCIICase(containerType, "video/vp8"_s)) {
        info.powerEfficient = false;
        info.smooth = isVPSoftwareDecoderSmooth(configuration);
    } else if (equalLettersIgnoringASCIICase(containerType, "video/vp9"_s)) {
        auto decodingInfo = computeVPParameters(configuration);
        info.powerEfficient = decodingInfo ? decodingInfo->powerEfficient : true;
        info.smooth = decodingInfo ? decodingInfo->smooth : isVPSoftwareDecoderSmooth(configuration);
    } else {
        // FIXME: Provide more granular H.264 decoder information.
        info.powerEfficient = true;
        info.smooth = true;
    }
    info.supported = true;
    return { info };
}

} // namespace WebCore

#endif // USE(GSTREAMER_WEBRTC)