mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 18:21:04 +00:00
visual: port to baseaudiovisualizer
Add a copy of the base class until it is stable. Right now the extra effects of the baseclass are not supported as the sublass overwrites the buffer instead of blending.
This commit is contained in:
parent
85565952ee
commit
5e6abc449b
5 changed files with 1421 additions and 756 deletions
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@ -1,12 +1,12 @@
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plugin_LTLIBRARIES = libgstlibvisual.la
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libgstlibvisual_la_SOURCES = plugin.c visual.c
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libgstlibvisual_la_SOURCES = plugin.c visual.c gstbaseaudiovisualizer.c
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libgstlibvisual_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBVISUAL_CFLAGS)
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libgstlibvisual_la_LIBADD = \
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$(top_builddir)/gst-libs/gst/audio/libgstaudio-$(GST_API_VERSION).la \
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$(top_builddir)/gst-libs/gst/video/libgstvideo-$(GST_API_VERSION).la \
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$(top_builddir)/gst-libs/gst/audio/libgstaudio-$(GST_API_VERSION).la \
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$(top_builddir)/gst-libs/gst/video/libgstvideo-$(GST_API_VERSION).la \
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$(GST_BASE_LIBS) $(LIBVISUAL_LIBS)
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libgstlibvisual_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstlibvisual_la_LIBTOOLFLAGS = --tag=disable-static
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noinst_HEADERS = visual.h
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noinst_HEADERS = visual.h gstbaseaudiovisualizer.h
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1134
ext/libvisual/gstbaseaudiovisualizer.c
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1134
ext/libvisual/gstbaseaudiovisualizer.c
Normal file
File diff suppressed because it is too large
Load diff
125
ext/libvisual/gstbaseaudiovisualizer.h
Normal file
125
ext/libvisual/gstbaseaudiovisualizer.h
Normal file
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@ -0,0 +1,125 @@
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/* GStreamer
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* Copyright (C) <2011> Stefan Kost <ensonic@users.sf.net>
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*
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* gstbaseaudiovisualizer.c: base class for audio visualisation elements
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
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*/
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#ifndef __GST_BASE_AUDIO_VISUALIZER_H__
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#define __GST_BASE_AUDIO_VISUALIZER_H__
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/video/video.h>
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#include <gst/audio/audio.h>
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#include <gst/base/gstadapter.h>
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G_BEGIN_DECLS
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#define GST_TYPE_BASE_AUDIO_VISUALIZER (gst_base_audio_visualizer_get_type())
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#define GST_BASE_AUDIO_VISUALIZER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_BASE_AUDIO_VISUALIZER,GstBaseAudioVisualizer))
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#define GST_BASE_AUDIO_VISUALIZER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_BASE_AUDIO_VISUALIZER,GstBaseAudioVisualizerClass))
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#define GST_IS_SYNAESTHESIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_BASE_AUDIO_VISUALIZER))
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#define GST_IS_SYNAESTHESIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_BASE_AUDIO_VISUALIZER))
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typedef struct _GstBaseAudioVisualizer GstBaseAudioVisualizer;
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typedef struct _GstBaseAudioVisualizerClass GstBaseAudioVisualizerClass;
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typedef void (*GstBaseAudioVisualizerShaderFunc)(GstBaseAudioVisualizer *scope, const guint8 *s, guint8 *d);
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/**
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* GstBaseAudioVisualizerShader:
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_NONE: no shading
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE: plain fading
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP: fade and move up
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN: fade and move down
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT: fade and move left
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT: fade and move right
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT: fade and move horizontally out
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN: fade and move horizontally in
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT: fade and move vertically out
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* @GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN: fade and move vertically in
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*
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* Different types of supported background shading functions.
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*/
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typedef enum {
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GST_BASE_AUDIO_VISUALIZER_SHADER_NONE,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT,
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GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN
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} GstBaseAudioVisualizerShader;
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struct _GstBaseAudioVisualizer
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{
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GstElement parent;
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/* pads */
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GstPad *srcpad, *sinkpad;
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GstBufferPool *pool;
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GstAdapter *adapter;
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GstBuffer *inbuf;
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guint8 *pixelbuf;
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GstBaseAudioVisualizerShader shader_type;
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GstBaseAudioVisualizerShaderFunc shader;
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guint32 shade_amount;
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guint spf; /* samples per video frame */
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guint req_spf; /* min samples per frame wanted by the subclass */
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/* video state */
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GstVideoInfo vinfo;
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GstVideoFormat video_format;
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gint fps_n, fps_d;
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gint width;
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gint height;
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guint64 frame_duration;
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guint bpf; /* bytes per frame */
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/* audio state */
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GstAudioInfo ainfo;
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/* configuration mutex */
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GMutex config_lock;
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/* QoS stuff *//* with LOCK */
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gdouble proportion;
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GstClockTime earliest_time;
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GstSegment segment;
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};
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struct _GstBaseAudioVisualizerClass
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{
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GstElementClass parent_class;
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/* virtual function, called whenever the format changes */
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gboolean (*setup) (GstBaseAudioVisualizer * scope);
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/* virtual function for rendering a frame */
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gboolean (*render) (GstBaseAudioVisualizer * scope, GstBuffer * audio, GstBuffer * video);
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};
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GType gst_base_audio_visualizer_get_type (void);
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G_END_DECLS
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#endif /* __GST_BASE_AUDIO_VISUALIZER_H__ */
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@ -30,11 +30,6 @@ GST_DEBUG_CATEGORY_EXTERN (libvisual_debug);
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/* amounf of samples before we can feed libvisual */
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#define VISUAL_SAMPLES 512
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#define DEFAULT_WIDTH 320
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#define DEFAULT_HEIGHT 240
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#define DEFAULT_FPS_N 25
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#define DEFAULT_FPS_D 1
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " GST_AUDIO_NE (S16) ", "
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"layout = (string) interleaved, " "channels = (int) { 1, 2 }, "
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"channel-mask = (bitmask) 0x3, "
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#if defined(VISUAL_API_VERSION) && VISUAL_API_VERSION >= 4000 && VISUAL_API_VERSION < 5000
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"rate = (int) { 8000, 11250, 22500, 32000, 44100, 48000, 96000 }"
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#else
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);
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static void gst_visual_init (GstVisual * visual);
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static void gst_visual_finalize (GObject * object);
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static GstStateChangeReturn gst_visual_change_state (GstElement * element,
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GstStateChange transition);
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static GstFlowReturn gst_visual_chain (GstPad * pad, GstObject * parent,
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GstBuffer * buffer);
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static gboolean gst_visual_sink_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_visual_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_visual_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static gboolean gst_visual_sink_setcaps (GstPad * pad, GstCaps * caps);
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static GstCaps *gst_visual_getcaps (GstPad * pad, GstCaps * filter);
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static gboolean gst_visual_setup (GstBaseAudioVisualizer * bscope);
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static gboolean gst_visual_render (GstBaseAudioVisualizer * bscope,
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GstBuffer * audio, GstBuffer * video);
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static GstElementClass *parent_class = NULL;
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(GInstanceInitFunc) gst_visual_init,
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};
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type = g_type_register_static (GST_TYPE_ELEMENT, "GstVisual", &info, 0);
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type =
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g_type_register_static (GST_TYPE_BASE_AUDIO_VISUALIZER, "GstVisual",
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&info, 0);
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}
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return type;
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}
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void
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gst_visual_class_init (gpointer g_class, gpointer class_data)
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{
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GstVisualClass *klass = GST_VISUAL_CLASS (g_class);
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GstElementClass *element = GST_ELEMENT_CLASS (g_class);
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GObjectClass *object = G_OBJECT_CLASS (g_class);
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GObjectClass *gobject_class = (GObjectClass *) g_class;
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GstElementClass *element_class = (GstElementClass *) g_class;
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GstBaseAudioVisualizerClass *scope_class =
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(GstBaseAudioVisualizerClass *) g_class;
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GstVisualClass *klass = (GstVisualClass *) g_class;
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klass->plugin = class_data;
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element->change_state = gst_visual_change_state;
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if (class_data == NULL) {
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parent_class = g_type_class_peek_parent (g_class);
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} else {
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klass->plugin->info->name, klass->plugin->info->version);
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/* FIXME: improve to only register what plugin supports? */
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gst_element_class_add_pad_template (element,
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_add_pad_template (element,
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_static_metadata (element,
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gst_element_class_set_static_metadata (element_class,
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longname, "Visualization",
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klass->plugin->info->about, "Benjamin Otte <otte@gnome.org>");
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g_free (longname);
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}
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object->finalize = gst_visual_finalize;
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gobject_class->finalize = gst_visual_finalize;
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scope_class->setup = GST_DEBUG_FUNCPTR (gst_visual_setup);
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scope_class->render = GST_DEBUG_FUNCPTR (gst_visual_render);
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}
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static void
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gst_visual_init (GstVisual * visual)
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{
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/* create the sink and src pads */
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visual->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_chain_function (visual->sinkpad, gst_visual_chain);
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gst_pad_set_event_function (visual->sinkpad, gst_visual_sink_event);
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gst_element_add_pad (GST_ELEMENT (visual), visual->sinkpad);
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visual->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_set_event_function (visual->srcpad, gst_visual_src_event);
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gst_pad_set_query_function (visual->srcpad, gst_visual_src_query);
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gst_element_add_pad (GST_ELEMENT (visual), visual->srcpad);
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visual->adapter = gst_adapter_new ();
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/* do nothing */
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}
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static void
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@ -178,716 +156,170 @@ gst_visual_finalize (GObject * object)
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{
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GstVisual *visual = GST_VISUAL (object);
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g_object_unref (visual->adapter);
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if (visual->pool)
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gst_object_unref (visual->pool);
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gst_visual_clear_actors (visual);
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GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
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}
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static void
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gst_visual_reset (GstVisual * visual)
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{
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gst_adapter_clear (visual->adapter);
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gst_segment_init (&visual->segment, GST_FORMAT_UNDEFINED);
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GST_OBJECT_LOCK (visual);
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visual->proportion = 1.0;
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visual->earliest_time = -1;
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GST_OBJECT_UNLOCK (visual);
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}
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static GstCaps *
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gst_visual_getcaps (GstPad * pad, GstCaps * filter)
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{
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GstCaps *ret;
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GstVisual *visual = GST_VISUAL (GST_PAD_PARENT (pad));
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int depths;
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if (!visual->actor) {
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ret = gst_pad_get_pad_template_caps (visual->srcpad);
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goto beach;
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}
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ret = gst_caps_new_empty ();
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depths = visual_actor_get_supported_depth (visual->actor);
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if (depths < 0) {
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/* FIXME: set an error */
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goto beach;
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}
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if (depths == VISUAL_VIDEO_DEPTH_GL) {
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/* We can't handle GL only plugins */
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goto beach;
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}
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GST_DEBUG_OBJECT (visual, "libvisual plugin supports depths %u (0x%04x)",
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depths, depths);
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/* if (depths & VISUAL_VIDEO_DEPTH_32BIT) Always supports 32bit output */
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#if G_BYTE_ORDER == G_BIG_ENDIAN
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gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("xRGB")));
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#else
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gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("BGRx")));
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#endif
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if (depths & VISUAL_VIDEO_DEPTH_24BIT) {
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#if G_BYTE_ORDER == G_BIG_ENDIAN
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gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("RGB")));
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#else
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gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("BGR")));
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#endif
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}
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if (depths & VISUAL_VIDEO_DEPTH_16BIT) {
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gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("RGB16")));
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}
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beach:
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if (filter) {
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GstCaps *intersection;
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intersection =
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gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (ret);
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ret = intersection;
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}
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GST_DEBUG_OBJECT (visual, "returning caps %" GST_PTR_FORMAT, ret);
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return ret;
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}
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static gboolean
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gst_visual_src_setcaps (GstVisual * visual, GstCaps * caps)
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gst_visual_setup (GstBaseAudioVisualizer * bscope)
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{
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gboolean res;
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GstStructure *structure;
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gint depth, pitch, rate;
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const gchar *fmt;
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GstVisual *visual = GST_VISUAL (bscope);
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gint pitch, depth;
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structure = gst_caps_get_structure (caps, 0);
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gst_visual_clear_actors (visual);
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GST_DEBUG_OBJECT (visual, "src pad got caps %" GST_PTR_FORMAT, caps);
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depth = bscope->vinfo.finfo->pixel_stride[0];
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if (bscope->vinfo.finfo->bits >= 8) {
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depth *= 8;
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}
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if (!gst_structure_get_int (structure, "width", &visual->width))
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goto error;
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if (!gst_structure_get_int (structure, "height", &visual->height))
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goto error;
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if (!(fmt = gst_structure_get_string (structure, "format")))
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goto error;
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if (!gst_structure_get_fraction (structure, "framerate", &visual->fps_n,
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&visual->fps_d))
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goto error;
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visual->actor =
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visual_actor_new (GST_VISUAL_GET_CLASS (visual)->plugin->info->plugname);
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visual->video = visual_video_new ();
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visual->audio = visual_audio_new ();
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/* can't have a play without actors */
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if (!visual->actor || !visual->video)
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goto no_actors;
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if (!strcmp (fmt, "BGR") || !strcmp (fmt, "RGB"))
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depth = 24;
|
||||
else if (!strcmp (fmt, "BGRx") || !strcmp (fmt, "xRGB"))
|
||||
depth = 32;
|
||||
else
|
||||
depth = 16;
|
||||
if (visual_actor_realize (visual->actor) != 0)
|
||||
goto no_realize;
|
||||
|
||||
visual_actor_set_video (visual->actor, visual->video);
|
||||
|
||||
visual_video_set_depth (visual->video,
|
||||
visual_video_depth_enum_from_value (depth));
|
||||
visual_video_set_dimension (visual->video, visual->width, visual->height);
|
||||
pitch = GST_ROUND_UP_4 (visual->width * visual->video->bpp);
|
||||
visual_video_set_dimension (visual->video, bscope->width, bscope->height);
|
||||
pitch = GST_ROUND_UP_4 (bscope->width * visual->video->bpp);
|
||||
visual_video_set_pitch (visual->video, pitch);
|
||||
visual_actor_video_negotiate (visual->actor, 0, FALSE, FALSE);
|
||||
|
||||
rate = GST_AUDIO_INFO_RATE (&visual->info);
|
||||
|
||||
/* precalc some values */
|
||||
visual->outsize = visual->video->height * pitch;
|
||||
visual->spf = gst_util_uint64_scale_int (rate, visual->fps_d, visual->fps_n);
|
||||
visual->duration =
|
||||
gst_util_uint64_scale_int (GST_SECOND, visual->fps_d, visual->fps_n);
|
||||
|
||||
res = gst_pad_set_caps (visual->srcpad, caps);
|
||||
|
||||
return res;
|
||||
|
||||
/* ERRORS */
|
||||
error:
|
||||
{
|
||||
GST_DEBUG_OBJECT (visual, "error parsing caps");
|
||||
return FALSE;
|
||||
}
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_visual_sink_setcaps (GstPad * pad, GstCaps * caps)
|
||||
{
|
||||
GstVisual *visual = GST_VISUAL (GST_PAD_PARENT (pad));
|
||||
GstAudioInfo info;
|
||||
gint rate;
|
||||
|
||||
if (!gst_audio_info_from_caps (&info, caps))
|
||||
goto invalid_caps;
|
||||
|
||||
visual->info = info;
|
||||
|
||||
rate = GST_AUDIO_INFO_RATE (&info);
|
||||
|
||||
/* this is how many samples we need to fill one frame at the requested
|
||||
* framerate. */
|
||||
if (visual->fps_n != 0) {
|
||||
visual->spf =
|
||||
gst_util_uint64_scale_int (rate, visual->fps_d, visual->fps_n);
|
||||
}
|
||||
GST_DEBUG_OBJECT (visual, "WxH: %dx%d, bpp: %d, pitch: %d, depth: %d",
|
||||
bscope->width, bscope->height, visual->video->bpp, pitch, depth);
|
||||
|
||||
return TRUE;
|
||||
|
||||
/* ERRORS */
|
||||
invalid_caps:
|
||||
{
|
||||
GST_ERROR_OBJECT (visual, "invalid caps received");
|
||||
return FALSE;
|
||||
}
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_vis_src_negotiate (GstVisual * visual)
|
||||
{
|
||||
GstCaps *othercaps, *target;
|
||||
GstStructure *structure;
|
||||
GstCaps *caps;
|
||||
GstQuery *query;
|
||||
GstBufferPool *pool = NULL;
|
||||
GstStructure *config;
|
||||
guint size, min, max;
|
||||
|
||||
caps = gst_pad_query_caps (visual->srcpad, NULL);
|
||||
|
||||
/* see what the peer can do */
|
||||
othercaps = gst_pad_peer_query_caps (visual->srcpad, caps);
|
||||
if (othercaps) {
|
||||
target = othercaps;
|
||||
gst_caps_unref (caps);
|
||||
|
||||
if (gst_caps_is_empty (target))
|
||||
goto no_format;
|
||||
|
||||
target = gst_caps_truncate (target);
|
||||
} else {
|
||||
/* need a copy, we'll be modifying it when fixating */
|
||||
target = gst_caps_ref (caps);
|
||||
}
|
||||
GST_DEBUG_OBJECT (visual, "before fixate caps %" GST_PTR_FORMAT, target);
|
||||
|
||||
target = gst_caps_make_writable (target);
|
||||
/* fixate in case something is not fixed. This does nothing if the value is
|
||||
* already fixed. For video we always try to fixate to something like
|
||||
* 320x240x25 by convention. */
|
||||
structure = gst_caps_get_structure (target, 0);
|
||||
gst_structure_fixate_field_nearest_int (structure, "width", DEFAULT_WIDTH);
|
||||
gst_structure_fixate_field_nearest_int (structure, "height", DEFAULT_HEIGHT);
|
||||
gst_structure_fixate_field_nearest_fraction (structure, "framerate",
|
||||
DEFAULT_FPS_N, DEFAULT_FPS_D);
|
||||
target = gst_caps_fixate (target);
|
||||
|
||||
GST_DEBUG_OBJECT (visual, "after fixate caps %" GST_PTR_FORMAT, target);
|
||||
|
||||
gst_visual_src_setcaps (visual, target);
|
||||
|
||||
/* try to get a bufferpool now */
|
||||
/* find a pool for the negotiated caps now */
|
||||
query = gst_query_new_allocation (target, TRUE);
|
||||
|
||||
if (!gst_pad_peer_query (visual->srcpad, query)) {
|
||||
/* not a problem, we deal with the defaults of the query */
|
||||
GST_DEBUG_OBJECT (visual, "allocation query failed");
|
||||
}
|
||||
|
||||
if (gst_query_get_n_allocation_pools (query) > 0) {
|
||||
gst_query_parse_nth_allocation_pool (query, 0, &pool, &size, &min, &max);
|
||||
|
||||
size = MAX (size, visual->outsize);
|
||||
} else {
|
||||
pool = NULL;
|
||||
size = visual->outsize;
|
||||
min = max = 0;
|
||||
}
|
||||
|
||||
if (pool == NULL) {
|
||||
/* no pool, just parameters, we can make our own */
|
||||
GST_DEBUG_OBJECT (visual, "no pool, making new pool");
|
||||
pool = gst_video_buffer_pool_new ();
|
||||
}
|
||||
|
||||
/* and configure */
|
||||
config = gst_buffer_pool_get_config (pool);
|
||||
gst_buffer_pool_config_set_params (config, target, size, min, max);
|
||||
gst_buffer_pool_set_config (pool, config);
|
||||
|
||||
if (visual->pool)
|
||||
gst_object_unref (visual->pool);
|
||||
visual->pool = pool;
|
||||
|
||||
/* and activate */
|
||||
gst_buffer_pool_set_active (pool, TRUE);
|
||||
|
||||
gst_caps_unref (target);
|
||||
|
||||
return TRUE;
|
||||
|
||||
/* ERRORS */
|
||||
no_format:
|
||||
{
|
||||
GST_ELEMENT_ERROR (visual, STREAM, FORMAT, (NULL),
|
||||
("could not negotiate output format"));
|
||||
gst_caps_unref (target);
|
||||
return FALSE;
|
||||
}
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_visual_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
||||
{
|
||||
GstVisual *visual;
|
||||
gboolean res;
|
||||
|
||||
visual = GST_VISUAL (parent);
|
||||
|
||||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_FLUSH_START:
|
||||
res = gst_pad_push_event (visual->srcpad, event);
|
||||
break;
|
||||
case GST_EVENT_FLUSH_STOP:
|
||||
/* reset QoS and adapter. */
|
||||
gst_visual_reset (visual);
|
||||
res = gst_pad_push_event (visual->srcpad, event);
|
||||
break;
|
||||
case GST_EVENT_CAPS:
|
||||
{
|
||||
GstCaps *caps;
|
||||
|
||||
gst_event_parse_caps (event, &caps);
|
||||
res = gst_visual_sink_setcaps (pad, caps);
|
||||
gst_event_unref (event);
|
||||
break;
|
||||
}
|
||||
case GST_EVENT_SEGMENT:
|
||||
{
|
||||
/* the newsegment values are used to clip the input samples
|
||||
* and to convert the incomming timestamps to running time so
|
||||
* we can do QoS */
|
||||
gst_event_copy_segment (event, &visual->segment);
|
||||
|
||||
/* and forward */
|
||||
res = gst_pad_push_event (visual->srcpad, event);
|
||||
break;
|
||||
}
|
||||
default:
|
||||
res = gst_pad_event_default (pad, parent, event);
|
||||
break;
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_visual_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
||||
{
|
||||
GstVisual *visual;
|
||||
gboolean res;
|
||||
|
||||
visual = GST_VISUAL (parent);
|
||||
|
||||
switch (GST_EVENT_TYPE (event)) {
|
||||
case GST_EVENT_QOS:
|
||||
{
|
||||
gdouble proportion;
|
||||
GstClockTimeDiff diff;
|
||||
GstClockTime timestamp;
|
||||
|
||||
gst_event_parse_qos (event, NULL, &proportion, &diff, ×tamp);
|
||||
|
||||
/* save stuff for the _chain function */
|
||||
GST_OBJECT_LOCK (visual);
|
||||
visual->proportion = proportion;
|
||||
if (diff >= 0)
|
||||
/* we're late, this is a good estimate for next displayable
|
||||
* frame (see part-qos.txt) */
|
||||
visual->earliest_time = timestamp + 2 * diff + visual->duration;
|
||||
else
|
||||
visual->earliest_time = timestamp + diff;
|
||||
|
||||
GST_OBJECT_UNLOCK (visual);
|
||||
|
||||
res = gst_pad_push_event (visual->sinkpad, event);
|
||||
break;
|
||||
}
|
||||
case GST_EVENT_RECONFIGURE:
|
||||
/* dont't forward */
|
||||
gst_event_unref (event);
|
||||
res = TRUE;
|
||||
break;
|
||||
default:
|
||||
res = gst_pad_event_default (pad, parent, event);
|
||||
break;
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_visual_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
|
||||
{
|
||||
gboolean res;
|
||||
GstVisual *visual;
|
||||
|
||||
visual = GST_VISUAL (parent);
|
||||
|
||||
switch (GST_QUERY_TYPE (query)) {
|
||||
case GST_QUERY_LATENCY:
|
||||
{
|
||||
/* We need to send the query upstream and add the returned latency to our
|
||||
* own */
|
||||
GstClockTime min_latency, max_latency;
|
||||
gboolean us_live;
|
||||
GstClockTime our_latency;
|
||||
guint max_samples;
|
||||
|
||||
if ((res = gst_pad_peer_query (visual->sinkpad, query))) {
|
||||
gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
|
||||
|
||||
GST_DEBUG_OBJECT (visual, "Peer latency: min %"
|
||||
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
||||
|
||||
/* the max samples we must buffer */
|
||||
max_samples = MAX (VISUAL_SAMPLES, visual->spf);
|
||||
our_latency =
|
||||
gst_util_uint64_scale_int (max_samples, GST_SECOND,
|
||||
GST_AUDIO_INFO_RATE (&visual->info));
|
||||
|
||||
GST_DEBUG_OBJECT (visual, "Our latency: %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (our_latency));
|
||||
|
||||
/* we add some latency but only if we need to buffer more than what
|
||||
* upstream gives us */
|
||||
min_latency += our_latency;
|
||||
if (max_latency != -1)
|
||||
max_latency += our_latency;
|
||||
|
||||
GST_DEBUG_OBJECT (visual, "Calculated total latency : min %"
|
||||
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
||||
|
||||
gst_query_set_latency (query, TRUE, min_latency, max_latency);
|
||||
}
|
||||
break;
|
||||
}
|
||||
case GST_QUERY_CAPS:
|
||||
{
|
||||
GstCaps *filter, *caps;
|
||||
|
||||
gst_query_parse_caps (query, &filter);
|
||||
caps = gst_visual_getcaps (pad, filter);
|
||||
gst_query_set_caps_result (query, caps);
|
||||
gst_caps_unref (caps);
|
||||
res = TRUE;
|
||||
}
|
||||
default:
|
||||
res = gst_pad_query_default (pad, parent, query);
|
||||
break;
|
||||
}
|
||||
|
||||
return res;
|
||||
}
|
||||
|
||||
/* Make sure we are negotiated */
|
||||
static GstFlowReturn
|
||||
ensure_negotiated (GstVisual * visual)
|
||||
{
|
||||
gboolean reconfigure;
|
||||
|
||||
reconfigure = gst_pad_check_reconfigure (visual->srcpad);
|
||||
|
||||
/* we don't know an output format yet, pick one */
|
||||
if (reconfigure || !gst_pad_has_current_caps (visual->srcpad)) {
|
||||
if (!gst_vis_src_negotiate (visual))
|
||||
return GST_FLOW_NOT_NEGOTIATED;
|
||||
}
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_visual_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
|
||||
{
|
||||
GstBuffer *outbuf = NULL;
|
||||
guint i;
|
||||
GstVisual *visual = GST_VISUAL (parent);
|
||||
GstFlowReturn ret = GST_FLOW_OK;
|
||||
guint avail;
|
||||
gint bpf, rate, channels;
|
||||
|
||||
GST_DEBUG_OBJECT (visual, "chain function called");
|
||||
|
||||
/* Make sure have an output format */
|
||||
ret = ensure_negotiated (visual);
|
||||
if (ret != GST_FLOW_OK) {
|
||||
gst_buffer_unref (buffer);
|
||||
goto beach;
|
||||
}
|
||||
|
||||
/* resync on DISCONT */
|
||||
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
|
||||
gst_adapter_clear (visual->adapter);
|
||||
}
|
||||
|
||||
rate = GST_AUDIO_INFO_RATE (&visual->info);
|
||||
bpf = GST_AUDIO_INFO_BPF (&visual->info);
|
||||
channels = GST_AUDIO_INFO_CHANNELS (&visual->info);
|
||||
|
||||
GST_DEBUG_OBJECT (visual,
|
||||
"Input buffer has %" G_GSIZE_FORMAT " samples, time=%" G_GUINT64_FORMAT,
|
||||
gst_buffer_get_size (buffer) / bpf, GST_BUFFER_TIMESTAMP (buffer));
|
||||
|
||||
gst_adapter_push (visual->adapter, buffer);
|
||||
|
||||
while (TRUE) {
|
||||
gboolean need_skip;
|
||||
const guint16 *data;
|
||||
guint64 dist, timestamp;
|
||||
GstMapInfo outmap;
|
||||
|
||||
GST_DEBUG_OBJECT (visual, "processing buffer");
|
||||
|
||||
avail = gst_adapter_available (visual->adapter);
|
||||
GST_DEBUG_OBJECT (visual, "avail now %u", avail);
|
||||
|
||||
/* we need at least VISUAL_SAMPLES samples */
|
||||
if (avail < VISUAL_SAMPLES * bpf)
|
||||
break;
|
||||
|
||||
/* we need at least enough samples to make one frame */
|
||||
if (avail < visual->spf * bpf)
|
||||
break;
|
||||
|
||||
/* get timestamp of the current adapter byte */
|
||||
timestamp = gst_adapter_prev_timestamp (visual->adapter, &dist);
|
||||
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
||||
/* convert bytes to time */
|
||||
dist /= bpf;
|
||||
timestamp += gst_util_uint64_scale_int (dist, GST_SECOND, rate);
|
||||
}
|
||||
|
||||
if (timestamp != -1) {
|
||||
gint64 qostime;
|
||||
|
||||
/* QoS is done on running time */
|
||||
qostime = gst_segment_to_running_time (&visual->segment, GST_FORMAT_TIME,
|
||||
timestamp);
|
||||
qostime += visual->duration;
|
||||
|
||||
GST_OBJECT_LOCK (visual);
|
||||
/* check for QoS, don't compute buffers that are known to be late */
|
||||
need_skip = visual->earliest_time != -1 &&
|
||||
qostime <= visual->earliest_time;
|
||||
GST_OBJECT_UNLOCK (visual);
|
||||
|
||||
if (need_skip) {
|
||||
GST_WARNING_OBJECT (visual,
|
||||
"QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (qostime), GST_TIME_ARGS (visual->earliest_time));
|
||||
goto skip;
|
||||
}
|
||||
}
|
||||
|
||||
/* Read VISUAL_SAMPLES samples per channel */
|
||||
data =
|
||||
(const guint16 *) gst_adapter_map (visual->adapter,
|
||||
VISUAL_SAMPLES * bpf);
|
||||
|
||||
#if defined(VISUAL_API_VERSION) && VISUAL_API_VERSION >= 4000 && VISUAL_API_VERSION < 5000
|
||||
{
|
||||
VisBuffer *lbuf, *rbuf;
|
||||
guint16 ldata[VISUAL_SAMPLES], rdata[VISUAL_SAMPLES];
|
||||
VisAudioSampleRateType vrate;
|
||||
|
||||
lbuf = visual_buffer_new_with_buffer (ldata, sizeof (ldata), NULL);
|
||||
rbuf = visual_buffer_new_with_buffer (rdata, sizeof (rdata), NULL);
|
||||
|
||||
if (channels == 2) {
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
ldata[i] = *data++;
|
||||
rdata[i] = *data++;
|
||||
}
|
||||
} else {
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
ldata[i] = *data;
|
||||
rdata[i] = *data++;
|
||||
}
|
||||
}
|
||||
|
||||
switch (rate) {
|
||||
case 8000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_8000;
|
||||
break;
|
||||
case 11250:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_11250;
|
||||
break;
|
||||
case 22500:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_22500;
|
||||
break;
|
||||
case 32000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_32000;
|
||||
break;
|
||||
case 44100:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_44100;
|
||||
break;
|
||||
case 48000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_48000;
|
||||
break;
|
||||
case 96000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_96000;
|
||||
break;
|
||||
default:
|
||||
visual_object_unref (VISUAL_OBJECT (lbuf));
|
||||
visual_object_unref (VISUAL_OBJECT (rbuf));
|
||||
GST_ERROR_OBJECT (visual, "unsupported rate %d", rate);
|
||||
ret = GST_FLOW_ERROR;
|
||||
goto beach;
|
||||
break;
|
||||
}
|
||||
|
||||
visual_audio_samplepool_input_channel (visual->audio->samplepool,
|
||||
lbuf,
|
||||
vrate, VISUAL_AUDIO_SAMPLE_FORMAT_S16,
|
||||
(char *) VISUAL_AUDIO_CHANNEL_LEFT);
|
||||
visual_audio_samplepool_input_channel (visual->audio->samplepool, rbuf,
|
||||
vrate, VISUAL_AUDIO_SAMPLE_FORMAT_S16,
|
||||
(char *) VISUAL_AUDIO_CHANNEL_RIGHT);
|
||||
|
||||
visual_object_unref (VISUAL_OBJECT (lbuf));
|
||||
visual_object_unref (VISUAL_OBJECT (rbuf));
|
||||
|
||||
}
|
||||
#else
|
||||
if (visual->channels == 2) {
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
visual->audio->plugpcm[0][i] = *data++;
|
||||
visual->audio->plugpcm[1][i] = *data++;
|
||||
}
|
||||
} else {
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
visual->audio->plugpcm[0][i] = *data;
|
||||
visual->audio->plugpcm[1][i] = *data++;
|
||||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
/* alloc a buffer if we don't have one yet, this happens
|
||||
* when we pushed a buffer in this while loop before */
|
||||
if (outbuf == NULL) {
|
||||
GST_DEBUG_OBJECT (visual, "allocating output buffer");
|
||||
ret = gst_buffer_pool_acquire_buffer (visual->pool, &outbuf, NULL);
|
||||
if (ret != GST_FLOW_OK) {
|
||||
gst_adapter_unmap (visual->adapter);
|
||||
goto beach;
|
||||
}
|
||||
}
|
||||
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
|
||||
visual_video_set_buffer (visual->video, outmap.data);
|
||||
visual_audio_analyze (visual->audio);
|
||||
visual_actor_run (visual->actor, visual->audio);
|
||||
visual_video_set_buffer (visual->video, NULL);
|
||||
gst_buffer_unmap (outbuf, &outmap);
|
||||
GST_DEBUG_OBJECT (visual, "rendered one frame");
|
||||
|
||||
gst_adapter_unmap (visual->adapter);
|
||||
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
||||
GST_BUFFER_DURATION (outbuf) = visual->duration;
|
||||
|
||||
ret = gst_pad_push (visual->srcpad, outbuf);
|
||||
outbuf = NULL;
|
||||
|
||||
skip:
|
||||
GST_DEBUG_OBJECT (visual, "finished frame, flushing %u samples from input",
|
||||
visual->spf);
|
||||
|
||||
/* Flush out the number of samples per frame */
|
||||
gst_adapter_flush (visual->adapter, visual->spf * bpf);
|
||||
|
||||
/* quit the loop if something was wrong */
|
||||
if (ret != GST_FLOW_OK)
|
||||
break;
|
||||
}
|
||||
|
||||
beach:
|
||||
|
||||
if (outbuf != NULL)
|
||||
gst_buffer_unref (outbuf);
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_visual_change_state (GstElement * element, GstStateChange transition)
|
||||
{
|
||||
GstVisual *visual = GST_VISUAL (element);
|
||||
GstStateChangeReturn ret;
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||
visual->actor =
|
||||
visual_actor_new (GST_VISUAL_GET_CLASS (visual)->plugin->info->
|
||||
plugname);
|
||||
visual->video = visual_video_new ();
|
||||
visual->audio = visual_audio_new ();
|
||||
/* can't have a play without actors */
|
||||
if (!visual->actor || !visual->video)
|
||||
goto no_actors;
|
||||
|
||||
if (visual_actor_realize (visual->actor) != 0)
|
||||
goto no_realize;
|
||||
|
||||
visual_actor_set_video (visual->actor, visual->video);
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
gst_visual_reset (visual);
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
if (visual->pool) {
|
||||
gst_buffer_pool_set_active (visual->pool, FALSE);
|
||||
gst_object_unref (visual->pool);
|
||||
visual->pool = NULL;
|
||||
}
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||
gst_visual_clear_actors (visual);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return ret;
|
||||
|
||||
/* ERRORS */
|
||||
no_actors:
|
||||
{
|
||||
GST_ELEMENT_ERROR (visual, LIBRARY, INIT, (NULL),
|
||||
("could not create actors"));
|
||||
gst_visual_clear_actors (visual);
|
||||
return GST_STATE_CHANGE_FAILURE;
|
||||
return FALSE;
|
||||
}
|
||||
no_realize:
|
||||
{
|
||||
GST_ELEMENT_ERROR (visual, LIBRARY, INIT, (NULL),
|
||||
("could not realize actor"));
|
||||
gst_visual_clear_actors (visual);
|
||||
return GST_STATE_CHANGE_FAILURE;
|
||||
return FALSE;
|
||||
}
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_visual_render (GstBaseAudioVisualizer * bscope, GstBuffer * audio,
|
||||
GstBuffer * video)
|
||||
{
|
||||
GstVisual *visual = GST_VISUAL (bscope);
|
||||
GstMapInfo amap, vmap;
|
||||
const guint16 *adata;
|
||||
gint i, channels;
|
||||
gboolean res = TRUE;
|
||||
|
||||
gst_buffer_map (audio, &amap, GST_MAP_READ);
|
||||
gst_buffer_map (video, &vmap, GST_MAP_WRITE);
|
||||
|
||||
visual_video_set_buffer (visual->video, vmap.data);
|
||||
|
||||
channels = GST_AUDIO_INFO_CHANNELS (&bscope->ainfo);
|
||||
adata = (const guint16 *) amap.data;
|
||||
|
||||
#if defined(VISUAL_API_VERSION) && VISUAL_API_VERSION >= 4000 && VISUAL_API_VERSION < 5000
|
||||
{
|
||||
VisBuffer *lbuf, *rbuf;
|
||||
guint16 ldata[VISUAL_SAMPLES], rdata[VISUAL_SAMPLES];
|
||||
VisAudioSampleRateType vrate;
|
||||
|
||||
lbuf = visual_buffer_new_with_buffer (ldata, sizeof (ldata), NULL);
|
||||
rbuf = visual_buffer_new_with_buffer (rdata, sizeof (rdata), NULL);
|
||||
|
||||
if (channels == 2) {
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
ldata[i] = *adata++;
|
||||
rdata[i] = *adata++;
|
||||
}
|
||||
} else {
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
ldata[i] = *adata;
|
||||
rdata[i] = *adata++;
|
||||
}
|
||||
}
|
||||
|
||||
/* TODO(ensonic): move to setup */
|
||||
switch (bscope->ainfo.rate) {
|
||||
case 8000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_8000;
|
||||
break;
|
||||
case 11250:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_11250;
|
||||
break;
|
||||
case 22500:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_22500;
|
||||
break;
|
||||
case 32000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_32000;
|
||||
break;
|
||||
case 44100:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_44100;
|
||||
break;
|
||||
case 48000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_48000;
|
||||
break;
|
||||
case 96000:
|
||||
vrate = VISUAL_AUDIO_SAMPLE_RATE_96000;
|
||||
break;
|
||||
default:
|
||||
visual_object_unref (VISUAL_OBJECT (lbuf));
|
||||
visual_object_unref (VISUAL_OBJECT (rbuf));
|
||||
GST_ERROR_OBJECT (visual, "unsupported rate %d", bscope->ainfo.rate);
|
||||
res = FALSE;
|
||||
goto done;
|
||||
}
|
||||
|
||||
visual_audio_samplepool_input_channel (visual->audio->samplepool,
|
||||
lbuf,
|
||||
vrate, VISUAL_AUDIO_SAMPLE_FORMAT_S16,
|
||||
(char *) VISUAL_AUDIO_CHANNEL_LEFT);
|
||||
visual_audio_samplepool_input_channel (visual->audio->samplepool, rbuf,
|
||||
vrate, VISUAL_AUDIO_SAMPLE_FORMAT_S16,
|
||||
(char *) VISUAL_AUDIO_CHANNEL_RIGHT);
|
||||
|
||||
visual_object_unref (VISUAL_OBJECT (lbuf));
|
||||
visual_object_unref (VISUAL_OBJECT (rbuf));
|
||||
|
||||
}
|
||||
#else
|
||||
if (channels == 2) {
|
||||
guint16 *ldata = visual->audio->plugpcm[0];
|
||||
guint16 *rdata = visual->audio->plugpcm[1];
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
ldata[i] = *adata++;
|
||||
rdata[i] = *adata++;
|
||||
}
|
||||
} else {
|
||||
for (i = 0; i < VISUAL_SAMPLES; i++) {
|
||||
ldata[i] = *adata;
|
||||
rdata[i] = *adata++;
|
||||
}
|
||||
}
|
||||
#endif
|
||||
|
||||
visual_audio_analyze (visual->audio);
|
||||
visual_actor_run (visual->actor, visual->audio);
|
||||
visual_video_set_buffer (visual->video, NULL);
|
||||
|
||||
GST_DEBUG_OBJECT (visual, "rendered one frame");
|
||||
done:
|
||||
gst_buffer_unmap (video, &vmap);
|
||||
gst_buffer_unmap (audio, &amap);
|
||||
return res;
|
||||
}
|
||||
|
|
|
@ -28,6 +28,8 @@
|
|||
#include <gst/audio/audio.h>
|
||||
#include <libvisual/libvisual.h>
|
||||
|
||||
#include "gstbaseaudiovisualizer.h"
|
||||
|
||||
G_BEGIN_DECLS
|
||||
|
||||
#define GST_TYPE_VISUAL (gst_visual_get_type())
|
||||
|
@ -42,45 +44,17 @@ typedef struct _GstVisualClass GstVisualClass;
|
|||
|
||||
struct _GstVisual
|
||||
{
|
||||
GstElement element;
|
||||
|
||||
/* pads */
|
||||
GstPad *sinkpad;
|
||||
GstPad *srcpad;
|
||||
GstSegment segment;
|
||||
GstBaseAudioVisualizer element;
|
||||
|
||||
/* libvisual stuff */
|
||||
VisAudio *audio;
|
||||
VisVideo *video;
|
||||
VisActor *actor;
|
||||
|
||||
/* audio/video state */
|
||||
GstAudioInfo info;
|
||||
|
||||
/* framerate numerator & denominator */
|
||||
gint fps_n;
|
||||
gint fps_d;
|
||||
gint width;
|
||||
gint height;
|
||||
GstClockTime duration;
|
||||
guint outsize;
|
||||
GstBufferPool *pool;
|
||||
|
||||
/* samples per frame based on caps */
|
||||
guint spf;
|
||||
|
||||
/* state stuff */
|
||||
GstAdapter *adapter;
|
||||
guint count;
|
||||
|
||||
/* QoS stuff *//* with LOCK */
|
||||
gdouble proportion;
|
||||
GstClockTime earliest_time;
|
||||
};
|
||||
|
||||
struct _GstVisualClass
|
||||
{
|
||||
GstElementClass parent_class;
|
||||
GstBaseAudioVisualizerClass parent_class;
|
||||
|
||||
VisPluginRef *plugin;
|
||||
};
|
||||
|
|
Loading…
Reference in a new issue