| /macosx-10.9.5/WebCore-7537.78.1/platform/graphics/gstreamer/ |
| H A D | GRefPtrGStreamer.h | 26 typedef struct _GstElement GstElement; typedef in typeref:struct:_GstElement
37 template<> GRefPtr<GstElement> adoptGRef(GstElement* ptr);
38 template<> GstElement* refGPtr<GstElement>(GstElement* ptr);
39 template<> void derefGPtr<GstElement>(GstElement* ptr);
|
| H A D | GStreamerGWorld.h | 33 typedef struct _GstElement GstElement; typedef in typeref:struct:_GstElement
48 static PassRefPtr<GStreamerGWorld> createGWorld(GstElement*);
51 GstElement* pipeline() const { return m_pipeline; }
64 GStreamerGWorld(GstElement*);
65 GstElement* m_pipeline;
|
| H A D | MediaPlayerPrivateGStreamerBase.h | 39 typedef struct _GstElement GstElement; typedef in typeref:struct:_GstElement
114 GstElement* createVideoSink(GstElement* pipeline);
116 virtual GstElement* audioSink() const { return 0; }
120 GRefPtr<GstElement> m_webkitVideoSink;
121 GRefPtr<GstElement> m_videoSinkBin;
122 GstElement* m_fpsSink;
|
| H A D | GStreamerGWorld.cpp | 73 PassRefPtr<GStreamerGWorld> GStreamerGWorld::createGWorld(GstElement* pipeline)
78 GStreamerGWorld::GStreamerGWorld(GstElement* pipeline)
114 GstElement* platformVideoSink = gst_element_factory_make("autovideosink", "platformVideoSink");
115 GstElement* colorspace = gst_element_factory_make(gVideoConvertName, "colorspace");
116 GstElement* queue = gst_element_factory_make("queue", "queue");
117 GstElement* videoScale = gst_element_factory_make("videoscale", "videoScale");
120 GRefPtr<GstElement> videoSink;
121 GstElement* sinkPtr = 0;
126 GRefPtr<GstElement> tee = adoptGRef(gst_bin_get_by_name(GST_BIN(videoSink.get()), "videoTee"));
199 GstElement* sinkPt
[all...] |
| H A D | MediaPlayerPrivateGStreamer.h | 38 typedef struct _GstElement GstElement; typedef in typeref:struct:_GstElement
94 GstElement* audioSink() const;
132 GRefPtr<GstElement> m_playBin;
133 GRefPtr<GstElement> m_source;
164 GRefPtr<GstElement> m_webkitAudioSink;
169 GRefPtr<GstElement> m_autoAudioSink;
|
| H A D | GRefPtrGStreamer.cpp | 29 template <> GRefPtr<GstElement> adoptGRef(GstElement* ptr)
32 return GRefPtr<GstElement>(ptr, GRefPtrAdopt);
35 template <> GstElement* refGPtr<GstElement>(GstElement* ptr)
43 template <> void derefGPtr<GstElement>(GstElement* ptr)
|
| H A D | VideoSinkGStreamer.h | 65 GstElement* webkitVideoSinkNew(WebCore::GStreamerGWorld*);
67 GstElement* webkitVideoSinkNew();
|
| H A D | GStreamerVersioning.h | 50 void notifyGstTagsOnPad(GstElement*, GstPad*, GstTagList*);
|
| H A D | FullscreenVideoControllerGStreamer.cpp | 84 GstElement* pipeline = m_gstreamerGWorld->pipeline();
93 GstElement* pipeline = m_gstreamerGWorld->pipeline();
|
| H A D | MediaPlayerPrivateGStreamerBase.cpp | 508 GstElement* MediaPlayerPrivateGStreamerBase::createVideoSink(GstElement* pipeline)
532 GstElement* videoTee = gst_element_factory_make("tee", "videoTee");
533 GstElement* queue = gst_element_factory_make("queue", 0);
546 GstElement* actualVideoSink = 0;
|
| H A D | MediaPlayerPrivateGStreamer.cpp | 1025 GstElement* sinkPtr = 0;
1032 GstElement* MediaPlayerPrivateGStreamer::audioSink() const
1039 GstElement* srcPtr = 0;
1581 GstElement* scale = gst_element_factory_make("scaletempo", 0);
1587 GstElement* convert = gst_element_factory_make("audioconvert", 0);
1588 GstElement* resample = gst_element_factory_make("audioresample", 0);
1589 GstElement* sink = gst_element_factory_make("autoaudiosink", 0);
1595 GstElement* audioSink = gst_bin_new("audio-sink");
1629 GstElement* videoElement = createVideoSink(m_playBin.get());
|
| H A D | VideoSinkGStreamer.cpp | 435 GstElement* webkitVideoSinkNew(WebCore::GStreamerGWorld* gstGWorld)
437 GstElement* element = GST_ELEMENT(g_object_new(WEBKIT_TYPE_VIDEO_SINK, 0));
442 GstElement* webkitVideoSinkNew()
|
| H A D | GStreamerVersioning.cpp | 226 void notifyGstTagsOnPad(GstElement* element, GstPad* pad, GstTagList* tags)
|
| H A D | WebKitWebSourceGStreamer.cpp | 140 static GstStateChangeReturn webKitWebSrcChangeState(GstElement*, GstStateChange);
521 static GstStateChangeReturn webKitWebSrcChangeState(GstElement* element, GstStateChange transition)
609 GRefPtr<GstElement> src = adoptGRef(gst_pad_get_parent_element(pad));
|
| /macosx-10.9.5/WebCore-7537.78.1/platform/audio/gstreamer/ |
| H A D | AudioDestinationGStreamer.h | 26 typedef struct _GstElement GstElement; typedef in typeref:struct:_GstElement
55 GstElement* m_pipeline;
|
| H A D | AudioDestinationGStreamer.cpp | 74 static void onGStreamerWavparsePadAddedCallback(GstElement*, GstPad* pad, AudioDestinationGStreamer* destination)
92 GstElement* webkitAudioSrc = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC,
98 GstElement* wavParser = gst_element_factory_make("wavparse", 0);
132 GRefPtr<GstElement> audioSink = gst_element_factory_make("autoaudiosink", 0);
151 GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
|
| H A D | AudioFileReaderGStreamer.cpp | 92 GstElement* m_pipeline;
94 GRefPtr<GstElement> m_decodebin;
95 GRefPtr<GstElement> m_deInterleave;
139 static void onGStreamerDeinterleavePadAddedCallback(GstElement*, GstPad* pad, AudioFileReader* reader)
144 static void onGStreamerDeinterleaveReadyCallback(GstElement*, AudioFileReader* reader)
149 static void onGStreamerDecodebinPadAddedCallback(GstElement*, GstPad* pad, AudioFileReader* reader)
340 GstElement* queue = gst_element_factory_make("queue", 0);
341 GstElement* sink = gst_element_factory_make("appsink", 0);
380 GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
381 GstElement* audioResampl
[all...] |
| H A D | WebKitWebAudioSourceGStreamer.cpp | 64 GRefPtr<GstElement> interleave;
65 GRefPtr<GstElement> wavEncoder;
97 static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement*, GstStateChange);
247 GstElement* queue = gst_element_factory_make("queue", 0);
248 GstElement* capsfilter = gst_element_factory_make("capsfilter", 0);
249 GstElement* audioconvert = gst_element_factory_make("audioconvert", 0);
397 static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, GstStateChange transition)
|