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
|
/*
* Copyright (C) 2015 Igalia S.L. All rights reserved.
* Copyright (C) 2015 Metrological. All rights reserved.
*
* 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
*/
#ifndef MediaPlayerPrivateGStreamerOwr_h
#define MediaPlayerPrivateGStreamerOwr_h
#if ENABLE(MEDIA_STREAM) && USE(GSTREAMER) && USE(OPENWEBRTC)
#include "MediaPlayerPrivateGStreamerBase.h"
#include "RealtimeMediaSource.h"
typedef struct _OwrGstVideoRenderer OwrGstVideoRenderer;
typedef struct _OwrGstAudioRenderer OwrGstAudioRenderer;
namespace WebCore {
class MediaStreamPrivate;
class RealtimeMediaSourceOwr;
class MediaPlayerPrivateGStreamerOwr : public MediaPlayerPrivateGStreamerBase, private RealtimeMediaSource::Observer {
public:
explicit MediaPlayerPrivateGStreamerOwr(MediaPlayer*);
~MediaPlayerPrivateGStreamerOwr();
static void registerMediaEngine(MediaEngineRegistrar);
private:
GstElement* createVideoSink() override;
GstElement* audioSink() const override { return m_audioSink.get(); }
bool isLiveStream() const override { return true; }
String engineDescription() const override { return "OpenWebRTC"; }
void load(const String&) override;
#if ENABLE(MEDIA_SOURCE)
void load(const String&, MediaSourcePrivateClient*) override { }
#endif
void load(MediaStreamPrivate&) override;
void cancelLoad() override { }
void prepareToPlay() override { }
void play() override;
void pause() override;
bool hasVideo() const override;
bool hasAudio() const override;
float duration() const override { return 0; }
float currentTime() const override;
void seek(float) override { }
bool seeking() const override { return false; }
void setRate(float) override { }
void setPreservesPitch(bool) override { }
bool paused() const override { return m_paused; }
bool hasClosedCaptions() const override { return false; }
void setClosedCaptionsVisible(bool) override { };
float maxTimeSeekable() const override { return 0; }
std::unique_ptr<PlatformTimeRanges> buffered() const override { return std::make_unique<PlatformTimeRanges>(); }
bool didLoadingProgress() const override;
unsigned long long totalBytes() const override { return 0; }
bool canLoadPoster() const override { return false; }
void setPoster(const String&) override { }
// RealtimeMediaSource::Observer implementation.
void sourceStopped() override final;
void sourceMutedChanged() override final;
void sourceSettingsChanged() override final;
bool preventSourceFromStopping() override final;
static void getSupportedTypes(HashSet<String, ASCIICaseInsensitiveHash>&);
static MediaPlayer::SupportsType supportsType(const MediaEngineSupportParameters&);
static bool initializeGStreamerAndGStreamerDebugging();
void createGSTAudioSinkBin();
void loadingFailed(MediaPlayer::NetworkState error);
bool internalLoad();
void stop();
bool m_paused { true };
bool m_stopped { true };
RefPtr<RealtimeMediaSourceOwr> m_videoSource;
RefPtr<RealtimeMediaSourceOwr> m_audioSource;
GRefPtr<GstElement> m_audioSink;
RefPtr<MediaStreamPrivate> m_streamPrivate;
GRefPtr<OwrGstVideoRenderer> m_videoRenderer;
GRefPtr<OwrGstAudioRenderer> m_audioRenderer;
};
} // namespace WebCore
#endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER) && USE(OPENWEBRTC)
#endif // MediaPlayerPrivateGStreamerOwr_h
|
