--- /dev/null
+/* GStreamer
+ * Copyright (C) <2011> Stefan Kost <ensonic@users.sf.net>
+ * Copyright (C) <2015> Luis de Bethencourt <luis@debethencourt.com>
+ *
+ * gstaudiovisualizer.h: class for audio visualisation elements
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:gstaudiovisualizer
+ *
+ * A class for scopes (visualizers). It takes care of re-fitting the audio-rate
+ * to video-rate and handles renegotiation (downstream video size changes).
+ *
+ * It also provides several background shading effects. These effects are
+ * applied to a previous picture before the render() implementation can draw a
+ * new frame.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/video/video.h>
+#include <gst/video/gstvideometa.h>
+#include <gst/video/gstvideopool.h>
+
+#include "gstaudiovisualizer.h"
+
+GST_DEBUG_CATEGORY_STATIC (audio_visualizer_debug);
+#define GST_CAT_DEFAULT (audio_visualizer_debug)
+
+#define DEFAULT_SHADER GST_AUDIO_VISUALIZER_SHADER_FADE
+#define DEFAULT_SHADE_AMOUNT 0x000a0a0a
+
+enum
+{
+ PROP_0,
+ PROP_SHADER,
+ PROP_SHADE_AMOUNT
+};
+
+static GstBaseTransformClass *parent_class = NULL;
+
+static void gst_audio_visualizer_class_init (GstAudioVisualizerClass * klass);
+static void gst_audio_visualizer_init (GstAudioVisualizer * scope,
+ GstAudioVisualizerClass * g_class);
+static void gst_audio_visualizer_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_visualizer_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+static void gst_audio_visualizer_dispose (GObject * object);
+
+static gboolean gst_audio_visualizer_src_negotiate (GstAudioVisualizer * scope);
+static gboolean gst_audio_visualizer_src_setcaps (GstAudioVisualizer *
+ scope, GstCaps * caps);
+static gboolean gst_audio_visualizer_sink_setcaps (GstAudioVisualizer *
+ scope, GstCaps * caps);
+
+static GstFlowReturn gst_audio_visualizer_chain (GstPad * pad,
+ GstObject * parent, GstBuffer * buffer);
+
+static gboolean gst_audio_visualizer_src_event (GstPad * pad,
+ GstObject * parent, GstEvent * event);
+static gboolean gst_audio_visualizer_sink_event (GstPad * pad,
+ GstObject * parent, GstEvent * event);
+
+static gboolean gst_audio_visualizer_src_query (GstPad * pad,
+ GstObject * parent, GstQuery * query);
+
+static GstStateChangeReturn gst_audio_visualizer_change_state (GstElement *
+ element, GstStateChange transition);
+
+static gboolean gst_audio_visualizer_do_bufferpool (GstAudioVisualizer * scope,
+ GstCaps * outcaps);
+
+static gboolean
+default_decide_allocation (GstAudioVisualizer * scope, GstQuery * query);
+
+#define GST_AUDIO_VISUALIZER_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_AUDIO_VISUALIZER, GstAudioVisualizerPrivate))
+
+struct _GstAudioVisualizerPrivate
+{
+ gboolean negotiated;
+
+ GstBufferPool *pool;
+ gboolean pool_active;
+ GstAllocator *allocator;
+ GstAllocationParams params;
+ GstQuery *query;
+
+ /* pads */
+ GstPad *srcpad, *sinkpad;
+
+ GstAudioVisualizerShader shader_type;
+ GstAudioVisualizerShaderFunc shader;
+ guint32 shade_amount;
+
+ GstAdapter *adapter;
+
+ GstBuffer *inbuf;
+ GstBuffer *tempbuf;
+ GstVideoFrame tempframe;
+
+ guint spf; /* samples per video frame */
+ guint64 frame_duration;
+
+ /* QoS stuff *//* with LOCK */
+ gdouble proportion;
+ GstClockTime earliest_time;
+
+ guint dropped; /* frames dropped / not dropped */
+ guint processed;
+
+ /* configuration mutex */
+ GMutex config_lock;
+
+ GstSegment segment;
+};
+
+/* shading functions */
+
+#define GST_TYPE_AUDIO_VISUALIZER_SHADER (gst_audio_visualizer_shader_get_type())
+static GType
+gst_audio_visualizer_shader_get_type (void)
+{
+ static GType shader_type = 0;
+ static const GEnumValue shaders[] = {
+ {GST_AUDIO_VISUALIZER_SHADER_NONE, "None", "none"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE, "Fade", "fade"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP, "Fade and move up",
+ "fade-and-move-up"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN, "Fade and move down",
+ "fade-and-move-down"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT, "Fade and move left",
+ "fade-and-move-left"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT,
+ "Fade and move right",
+ "fade-and-move-right"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT,
+ "Fade and move horizontally out", "fade-and-move-horiz-out"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN,
+ "Fade and move horizontally in", "fade-and-move-horiz-in"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT,
+ "Fade and move vertically out", "fade-and-move-vert-out"},
+ {GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN,
+ "Fade and move vertically in", "fade-and-move-vert-in"},
+ {0, NULL, NULL},
+ };
+
+ if (G_UNLIKELY (shader_type == 0)) {
+ /* TODO: rename when exporting it as a library */
+ shader_type =
+ g_enum_register_static ("GstAudioVisualizerShader-ExtGoom2k1", shaders);
+ }
+ return shader_type;
+}
+
+/* we're only supporting GST_VIDEO_FORMAT_xRGB right now) */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+
+#define SHADE(_d, _s, _i, _r, _g, _b) \
+G_STMT_START { \
+ _d[_i * 4 + 0] = (_s[_i * 4 + 0] > _b) ? _s[_i * 4 + 0] - _b : 0; \
+ _d[_i * 4 + 1] = (_s[_i * 4 + 1] > _g) ? _s[_i * 4 + 1] - _g : 0; \
+ _d[_i * 4 + 2] = (_s[_i * 4 + 2] > _r) ? _s[_i * 4 + 2] - _r : 0; \
+ _d[_i * 4 + 3] = 0; \
+} G_STMT_END
+
+#else /* G_BYTE_ORDER == G_LITTLE_ENDIAN */
+
+#define SHADE(_d, _s, _i, _r, _g, _b) \
+G_STMT_START { \
+ _d[_i * 4 + 0] = 0; \
+ _d[_i * 4 + 1] = (_s[_i * 4 + 1] > _r) ? _s[_i * 4 + 1] - _r : 0; \
+ _d[_i * 4 + 2] = (_s[_i * 4 + 2] > _g) ? _s[_i * 4 + 2] - _g : 0; \
+ _d[_i * 4 + 3] = (_s[_i * 4 + 3] > _b) ? _s[_i * 4 + 3] - _b : 0; \
+} G_STMT_END
+
+#endif
+
+static void
+shader_fade (GstAudioVisualizer * scope, const GstVideoFrame * sframe,
+ GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *d;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ for (j = 0; j < height; j++) {
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ s += ss;
+ d += ds;
+ }
+}
+
+static void
+shader_fade_and_move_up (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *d;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ for (j = 1; j < height; j++) {
+ s += ss;
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ d += ds;
+ }
+}
+
+static void
+shader_fade_and_move_down (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *d;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ for (j = 1; j < height; j++) {
+ d += ds;
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ s += ss;
+ }
+}
+
+static void
+shader_fade_and_move_left (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *d;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ width -= 1;
+ s += 4;
+
+ /* move to the left */
+ for (j = 0; j < height; j++) {
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ d += ds;
+ s += ss;
+ }
+}
+
+static void
+shader_fade_and_move_right (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *d;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ width -= 1;
+ d += 4;
+
+ /* move to the right */
+ for (j = 0; j < height; j++) {
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ d += ds;
+ s += ss;
+ }
+}
+
+static void
+shader_fade_and_move_horiz_out (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *d;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ /* move upper half up */
+ for (j = 0; j < height / 2; j++) {
+ s += ss;
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ d += ds;
+ }
+ /* move lower half down */
+ for (j = 0; j < height / 2; j++) {
+ d += ds;
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ s += ss;
+ }
+}
+
+static void
+shader_fade_and_move_horiz_in (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *d;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ /* move upper half down */
+ for (j = 0; j < height / 2; j++) {
+ d += ds;
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ s += ss;
+ }
+ /* move lower half up */
+ for (j = 0; j < height / 2; j++) {
+ s += ss;
+ for (i = 0; i < width; i++) {
+ SHADE (d, s, i, r, g, b);
+ }
+ d += ds;
+ }
+}
+
+static void
+shader_fade_and_move_vert_out (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *s1, *d, *d1;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ for (j = 0; j < height; j++) {
+ /* move left half to the left */
+ s1 = s + 1;
+ for (i = 0; i < width / 2; i++) {
+ SHADE (d, s1, i, r, g, b);
+ }
+ /* move right half to the right */
+ d1 = d + 1;
+ for (; i < width - 1; i++) {
+ SHADE (d1, s, i, r, g, b);
+ }
+ s += ss;
+ d += ds;
+ }
+}
+
+static void
+shader_fade_and_move_vert_in (GstAudioVisualizer * scope,
+ const GstVideoFrame * sframe, GstVideoFrame * dframe)
+{
+ guint i, j;
+ guint r = (scope->priv->shade_amount >> 16) & 0xff;
+ guint g = (scope->priv->shade_amount >> 8) & 0xff;
+ guint b = (scope->priv->shade_amount >> 0) & 0xff;
+ guint8 *s, *s1, *d, *d1;
+ gint ss, ds, width, height;
+
+ s = GST_VIDEO_FRAME_PLANE_DATA (sframe, 0);
+ ss = GST_VIDEO_FRAME_PLANE_STRIDE (sframe, 0);
+ d = GST_VIDEO_FRAME_PLANE_DATA (dframe, 0);
+ ds = GST_VIDEO_FRAME_PLANE_STRIDE (dframe, 0);
+
+ width = GST_VIDEO_FRAME_WIDTH (sframe);
+ height = GST_VIDEO_FRAME_HEIGHT (sframe);
+
+ for (j = 0; j < height; j++) {
+ /* move left half to the right */
+ d1 = d + 1;
+ for (i = 0; i < width / 2; i++) {
+ SHADE (d1, s, i, r, g, b);
+ }
+ /* move right half to the left */
+ s1 = s + 1;
+ for (; i < width - 1; i++) {
+ SHADE (d, s1, i, r, g, b);
+ }
+ s += ss;
+ d += ds;
+ }
+}
+
+static void
+gst_audio_visualizer_change_shader (GstAudioVisualizer * scope)
+{
+ switch (scope->priv->shader_type) {
+ case GST_AUDIO_VISUALIZER_SHADER_NONE:
+ scope->priv->shader = NULL;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE:
+ scope->priv->shader = shader_fade;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP:
+ scope->priv->shader = shader_fade_and_move_up;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN:
+ scope->priv->shader = shader_fade_and_move_down;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT:
+ scope->priv->shader = shader_fade_and_move_left;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT:
+ scope->priv->shader = shader_fade_and_move_right;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT:
+ scope->priv->shader = shader_fade_and_move_horiz_out;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN:
+ scope->priv->shader = shader_fade_and_move_horiz_in;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT:
+ scope->priv->shader = shader_fade_and_move_vert_out;
+ break;
+ case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN:
+ scope->priv->shader = shader_fade_and_move_vert_in;
+ break;
+ default:
+ GST_ERROR ("invalid shader function");
+ scope->priv->shader = NULL;
+ break;
+ }
+}
+
+/* class */
+
+GType
+gst_audio_visualizer_get_type (void)
+{
+ static volatile gsize audio_visualizer_type = 0;
+
+ if (g_once_init_enter (&audio_visualizer_type)) {
+ static const GTypeInfo audio_visualizer_info = {
+ sizeof (GstAudioVisualizerClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_audio_visualizer_class_init,
+ NULL,
+ NULL,
+ sizeof (GstAudioVisualizer),
+ 0,
+ (GInstanceInitFunc) gst_audio_visualizer_init,
+ };
+ GType _type;
+
+ /* TODO: rename when exporting it as a library */
+ _type = g_type_register_static (GST_TYPE_ELEMENT,
+ "GstAudioVisualizer-ExtGoom2k1", &audio_visualizer_info,
+ G_TYPE_FLAG_ABSTRACT);
+ g_once_init_leave (&audio_visualizer_type, _type);
+ }
+ return (GType) audio_visualizer_type;
+}
+
+static void
+gst_audio_visualizer_class_init (GstAudioVisualizerClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *element_class = (GstElementClass *) klass;
+
+ g_type_class_add_private (klass, sizeof (GstAudioVisualizerPrivate));
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ GST_DEBUG_CATEGORY_INIT (audio_visualizer_debug,
+ "audiovisualizer-goom2k1", 0, "audio visualisation class");
+
+ gobject_class->set_property = gst_audio_visualizer_set_property;
+ gobject_class->get_property = gst_audio_visualizer_get_property;
+ gobject_class->dispose = gst_audio_visualizer_dispose;
+
+ element_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_audio_visualizer_change_state);
+
+ klass->decide_allocation = GST_DEBUG_FUNCPTR (default_decide_allocation);
+
+ g_object_class_install_property (gobject_class, PROP_SHADER,
+ g_param_spec_enum ("shader", "shader type",
+ "Shader function to apply on each frame",
+ GST_TYPE_AUDIO_VISUALIZER_SHADER, DEFAULT_SHADER,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_SHADE_AMOUNT,
+ g_param_spec_uint ("shade-amount", "shade amount",
+ "Shading color to use (big-endian ARGB)", 0, G_MAXUINT32,
+ DEFAULT_SHADE_AMOUNT,
+ G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
+}
+
+static void
+gst_audio_visualizer_init (GstAudioVisualizer * scope,
+ GstAudioVisualizerClass * g_class)
+{
+ GstPadTemplate *pad_template;
+
+ scope->priv = GST_AUDIO_VISUALIZER_GET_PRIVATE (scope);
+
+ /* create the sink and src pads */
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
+ g_return_if_fail (pad_template != NULL);
+ scope->priv->sinkpad = gst_pad_new_from_template (pad_template, "sink");
+ gst_pad_set_chain_function (scope->priv->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_audio_visualizer_chain));
+ gst_pad_set_event_function (scope->priv->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_audio_visualizer_sink_event));
+ gst_element_add_pad (GST_ELEMENT (scope), scope->priv->sinkpad);
+
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
+ g_return_if_fail (pad_template != NULL);
+ scope->priv->srcpad = gst_pad_new_from_template (pad_template, "src");
+ gst_pad_set_event_function (scope->priv->srcpad,
+ GST_DEBUG_FUNCPTR (gst_audio_visualizer_src_event));
+ gst_pad_set_query_function (scope->priv->srcpad,
+ GST_DEBUG_FUNCPTR (gst_audio_visualizer_src_query));
+ gst_element_add_pad (GST_ELEMENT (scope), scope->priv->srcpad);
+
+ scope->priv->adapter = gst_adapter_new ();
+ scope->priv->inbuf = gst_buffer_new ();
+
+ /* properties */
+ scope->priv->shader_type = DEFAULT_SHADER;
+ gst_audio_visualizer_change_shader (scope);
+ scope->priv->shade_amount = DEFAULT_SHADE_AMOUNT;
+
+ /* reset the initial video state */
+ gst_video_info_init (&scope->vinfo);
+ scope->priv->frame_duration = GST_CLOCK_TIME_NONE;
+
+ /* reset the initial state */
+ gst_audio_info_init (&scope->ainfo);
+ gst_video_info_init (&scope->vinfo);
+
+ g_mutex_init (&scope->priv->config_lock);
+}
+
+static void
+gst_audio_visualizer_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAudioVisualizer *scope = GST_AUDIO_VISUALIZER (object);
+
+ switch (prop_id) {
+ case PROP_SHADER:
+ scope->priv->shader_type = g_value_get_enum (value);
+ gst_audio_visualizer_change_shader (scope);
+ break;
+ case PROP_SHADE_AMOUNT:
+ scope->priv->shade_amount = g_value_get_uint (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_visualizer_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAudioVisualizer *scope = GST_AUDIO_VISUALIZER (object);
+
+ switch (prop_id) {
+ case PROP_SHADER:
+ g_value_set_enum (value, scope->priv->shader_type);
+ break;
+ case PROP_SHADE_AMOUNT:
+ g_value_set_uint (value, scope->priv->shade_amount);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_visualizer_dispose (GObject * object)
+{
+ GstAudioVisualizer *scope = GST_AUDIO_VISUALIZER (object);
+
+ if (scope->priv->adapter) {
+ g_object_unref (scope->priv->adapter);
+ scope->priv->adapter = NULL;
+ }
+ if (scope->priv->inbuf) {
+ gst_buffer_unref (scope->priv->inbuf);
+ scope->priv->inbuf = NULL;
+ }
+ if (scope->priv->tempbuf) {
+ gst_video_frame_unmap (&scope->priv->tempframe);
+ gst_buffer_unref (scope->priv->tempbuf);
+ scope->priv->tempbuf = NULL;
+ }
+ if (scope->priv->config_lock.p) {
+ g_mutex_clear (&scope->priv->config_lock);
+ scope->priv->config_lock.p = NULL;
+ }
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_visualizer_reset (GstAudioVisualizer * scope)
+{
+ gst_adapter_clear (scope->priv->adapter);
+ gst_segment_init (&scope->priv->segment, GST_FORMAT_UNDEFINED);
+
+ GST_OBJECT_LOCK (scope);
+ scope->priv->proportion = 1.0;
+ scope->priv->earliest_time = -1;
+ scope->priv->dropped = 0;
+ scope->priv->processed = 0;
+ GST_OBJECT_UNLOCK (scope);
+}
+
+static gboolean
+gst_audio_visualizer_sink_setcaps (GstAudioVisualizer * scope, GstCaps * caps)
+{
+ GstAudioInfo info;
+
+ if (!gst_audio_info_from_caps (&info, caps))
+ goto wrong_caps;
+
+ scope->ainfo = info;
+
+ GST_DEBUG_OBJECT (scope, "audio: channels %d, rate %d",
+ GST_AUDIO_INFO_CHANNELS (&info), GST_AUDIO_INFO_RATE (&info));
+
+ if (!gst_audio_visualizer_src_negotiate (scope)) {
+ goto not_negotiated;
+ }
+
+ return TRUE;
+
+ /* Errors */
+wrong_caps:
+ {
+ GST_WARNING_OBJECT (scope, "could not parse caps");
+ return FALSE;
+ }
+not_negotiated:
+ {
+ GST_WARNING_OBJECT (scope, "failed to negotiate");
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_audio_visualizer_src_setcaps (GstAudioVisualizer * scope, GstCaps * caps)
+{
+ GstVideoInfo info;
+ GstAudioVisualizerClass *klass;
+ gboolean res;
+
+ if (!gst_video_info_from_caps (&info, caps))
+ goto wrong_caps;
+
+ klass = GST_AUDIO_VISUALIZER_CLASS (G_OBJECT_GET_CLASS (scope));
+
+ scope->vinfo = info;
+
+ scope->priv->frame_duration = gst_util_uint64_scale_int (GST_SECOND,
+ GST_VIDEO_INFO_FPS_D (&info), GST_VIDEO_INFO_FPS_N (&info));
+ scope->priv->spf =
+ gst_util_uint64_scale_int (GST_AUDIO_INFO_RATE (&scope->ainfo),
+ GST_VIDEO_INFO_FPS_D (&info), GST_VIDEO_INFO_FPS_N (&info));
+ scope->req_spf = scope->priv->spf;
+
+ if (scope->priv->tempbuf) {
+ gst_video_frame_unmap (&scope->priv->tempframe);
+ gst_buffer_unref (scope->priv->tempbuf);
+ }
+ scope->priv->tempbuf = gst_buffer_new_wrapped (g_malloc0 (scope->vinfo.size),
+ scope->vinfo.size);
+ gst_video_frame_map (&scope->priv->tempframe, &scope->vinfo,
+ scope->priv->tempbuf, GST_MAP_READWRITE);
+
+ if (klass->setup && !klass->setup (scope))
+ goto setup_failed;
+
+ GST_DEBUG_OBJECT (scope, "video: dimension %dx%d, framerate %d/%d",
+ GST_VIDEO_INFO_WIDTH (&info), GST_VIDEO_INFO_HEIGHT (&info),
+ GST_VIDEO_INFO_FPS_N (&info), GST_VIDEO_INFO_FPS_D (&info));
+ GST_DEBUG_OBJECT (scope, "blocks: spf %u, req_spf %u",
+ scope->priv->spf, scope->req_spf);
+
+ gst_pad_set_caps (scope->priv->srcpad, caps);
+
+ /* find a pool for the negotiated caps now */
+ res = gst_audio_visualizer_do_bufferpool (scope, caps);
+ gst_caps_unref (caps);
+
+ return res;
+
+ /* ERRORS */
+wrong_caps:
+ {
+ gst_caps_unref (caps);
+ GST_DEBUG_OBJECT (scope, "error parsing caps");
+ return FALSE;
+ }
+
+setup_failed:
+ {
+ GST_WARNING_OBJECT (scope, "failed to set up");
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_audio_visualizer_src_negotiate (GstAudioVisualizer * scope)
+{
+ GstCaps *othercaps, *target;
+ GstStructure *structure;
+ GstCaps *templ;
+ gboolean ret;
+
+ templ = gst_pad_get_pad_template_caps (scope->priv->srcpad);
+
+ GST_DEBUG_OBJECT (scope, "performing negotiation");
+
+ /* see what the peer can do */
+ othercaps = gst_pad_peer_query_caps (scope->priv->srcpad, NULL);
+ if (othercaps) {
+ target = gst_caps_intersect (othercaps, templ);
+ gst_caps_unref (othercaps);
+ gst_caps_unref (templ);
+
+ if (gst_caps_is_empty (target))
+ goto no_format;
+
+ target = gst_caps_truncate (target);
+ } else {
+ target = templ;
+ }
+
+ target = gst_caps_make_writable (target);
+ structure = gst_caps_get_structure (target, 0);
+ gst_structure_fixate_field_nearest_int (structure, "width", 320);
+ gst_structure_fixate_field_nearest_int (structure, "height", 200);
+ gst_structure_fixate_field_nearest_fraction (structure, "framerate", 25, 1);
+
+ target = gst_caps_fixate (target);
+
+ GST_DEBUG_OBJECT (scope, "final caps are %" GST_PTR_FORMAT, target);
+
+ ret = gst_audio_visualizer_src_setcaps (scope, target);
+
+ return ret;
+
+no_format:
+ {
+ gst_caps_unref (target);
+ return FALSE;
+ }
+}
+
+/* takes ownership of the pool, allocator and query */
+static gboolean
+gst_audio_visualizer_set_allocation (GstAudioVisualizer * scope,
+ GstBufferPool * pool, GstAllocator * allocator,
+ GstAllocationParams * params, GstQuery * query)
+{
+ GstAllocator *oldalloc;
+ GstBufferPool *oldpool;
+ GstQuery *oldquery;
+ GstAudioVisualizerPrivate *priv = scope->priv;
+
+ GST_OBJECT_LOCK (scope);
+ oldpool = priv->pool;
+ priv->pool = pool;
+ priv->pool_active = FALSE;
+
+ oldalloc = priv->allocator;
+ priv->allocator = allocator;
+
+ oldquery = priv->query;
+ priv->query = query;
+
+ if (params)
+ priv->params = *params;
+ else
+ gst_allocation_params_init (&priv->params);
+ GST_OBJECT_UNLOCK (scope);
+
+ if (oldpool) {
+ GST_DEBUG_OBJECT (scope, "deactivating old pool %p", oldpool);
+ gst_buffer_pool_set_active (oldpool, FALSE);
+ gst_object_unref (oldpool);
+ }
+ if (oldalloc) {
+ gst_object_unref (oldalloc);
+ }
+ if (oldquery) {
+ gst_query_unref (oldquery);
+ }
+ return TRUE;
+}
+
+static gboolean
+gst_audio_visualizer_do_bufferpool (GstAudioVisualizer * scope,
+ GstCaps * outcaps)
+{
+ GstQuery *query;
+ gboolean result = TRUE;
+ GstBufferPool *pool = NULL;
+ GstAudioVisualizerClass *klass;
+ GstAllocator *allocator;
+ GstAllocationParams params;
+
+ /* not passthrough, we need to allocate */
+ /* find a pool for the negotiated caps now */
+ GST_DEBUG_OBJECT (scope, "doing allocation query");
+ query = gst_query_new_allocation (outcaps, TRUE);
+
+ if (!gst_pad_peer_query (scope->priv->srcpad, query)) {
+ /* not a problem, we use the query defaults */
+ GST_DEBUG_OBJECT (scope, "allocation query failed");
+ }
+
+ klass = GST_AUDIO_VISUALIZER_GET_CLASS (scope);
+
+ GST_DEBUG_OBJECT (scope, "calling decide_allocation");
+ g_assert (klass->decide_allocation != NULL);
+ result = klass->decide_allocation (scope, query);
+
+ GST_DEBUG_OBJECT (scope, "ALLOCATION (%d) params: %" GST_PTR_FORMAT, result,
+ query);
+
+ if (!result)
+ goto no_decide_allocation;
+
+ /* we got configuration from our peer or the decide_allocation method,
+ * parse them */
+ if (gst_query_get_n_allocation_params (query) > 0) {
+ gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms);
+ } else {
+ allocator = NULL;
+ gst_allocation_params_init (¶ms);
+ }
+
+ if (gst_query_get_n_allocation_pools (query) > 0)
+ gst_query_parse_nth_allocation_pool (query, 0, &pool, NULL, NULL, NULL);
+
+ /* now store */
+ result =
+ gst_audio_visualizer_set_allocation (scope, pool, allocator, ¶ms,
+ query);
+
+ return result;
+
+ /* Errors */
+no_decide_allocation:
+ {
+ GST_WARNING_OBJECT (scope, "Subclass failed to decide allocation");
+ gst_query_unref (query);
+
+ return result;
+ }
+}
+
+static gboolean
+default_decide_allocation (GstAudioVisualizer * scope, GstQuery * query)
+{
+ GstCaps *outcaps;
+ GstBufferPool *pool;
+ guint size, min, max;
+ GstAllocator *allocator;
+ GstAllocationParams params;
+ GstStructure *config;
+ gboolean update_allocator;
+ gboolean update_pool;
+
+ gst_query_parse_allocation (query, &outcaps, NULL);
+
+ /* we got configuration from our peer or the decide_allocation method,
+ * parse them */
+ if (gst_query_get_n_allocation_params (query) > 0) {
+ /* try the allocator */
+ gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms);
+ update_allocator = TRUE;
+ } else {
+ allocator = NULL;
+ gst_allocation_params_init (¶ms);
+ update_allocator = FALSE;
+ }
+
+ if (gst_query_get_n_allocation_pools (query) > 0) {
+ gst_query_parse_nth_allocation_pool (query, 0, &pool, &size, &min, &max);
+ update_pool = TRUE;
+ } else {
+ pool = NULL;
+ size = GST_VIDEO_INFO_SIZE (&scope->vinfo);
+ min = max = 0;
+ update_pool = FALSE;
+ }
+
+ if (pool == NULL) {
+ /* we did not get a pool, make one ourselves then */
+ pool = gst_video_buffer_pool_new ();
+ }
+
+ config = gst_buffer_pool_get_config (pool);
+ gst_buffer_pool_config_set_params (config, outcaps, size, min, max);
+ gst_buffer_pool_config_set_allocator (config, allocator, ¶ms);
+ gst_buffer_pool_config_add_option (config, GST_BUFFER_POOL_OPTION_VIDEO_META);
+ gst_buffer_pool_set_config (pool, config);
+
+ if (update_allocator)
+ gst_query_set_nth_allocation_param (query, 0, allocator, ¶ms);
+ else
+ gst_query_add_allocation_param (query, allocator, ¶ms);
+
+ if (allocator)
+ gst_object_unref (allocator);
+
+ if (update_pool)
+ gst_query_set_nth_allocation_pool (query, 0, pool, size, min, max);
+ else
+ gst_query_add_allocation_pool (query, pool, size, min, max);
+
+ if (pool)
+ gst_object_unref (pool);
+
+ return TRUE;
+}
+
+static GstFlowReturn
+default_prepare_output_buffer (GstAudioVisualizer * scope, GstBuffer ** outbuf)
+{
+ GstAudioVisualizerPrivate *priv;
+
+ priv = scope->priv;
+
+ g_assert (priv->pool != NULL);
+
+ /* we can't reuse the input buffer */
+ if (!priv->pool_active) {
+ GST_DEBUG_OBJECT (scope, "setting pool %p active", priv->pool);
+ if (!gst_buffer_pool_set_active (priv->pool, TRUE))
+ goto activate_failed;
+ priv->pool_active = TRUE;
+ }
+ GST_DEBUG_OBJECT (scope, "using pool alloc");
+
+ return gst_buffer_pool_acquire_buffer (priv->pool, outbuf, NULL);
+
+ /* ERRORS */
+activate_failed:
+ {
+ GST_ELEMENT_ERROR (scope, RESOURCE, SETTINGS,
+ ("failed to activate bufferpool"), ("failed to activate bufferpool"));
+ return GST_FLOW_ERROR;
+ }
+}
+
+static GstFlowReturn
+gst_audio_visualizer_chain (GstPad * pad, GstObject * parent,
+ GstBuffer * buffer)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstAudioVisualizer *scope;
+ GstAudioVisualizerClass *klass;
+ GstBuffer *inbuf;
+ guint64 dist, ts;
+ guint avail, sbpf;
+ gpointer adata;
+ gint bps, channels, rate;
+
+ scope = GST_AUDIO_VISUALIZER (parent);
+ klass = GST_AUDIO_VISUALIZER_CLASS (G_OBJECT_GET_CLASS (scope));
+
+ GST_LOG_OBJECT (scope, "chainfunc called");
+
+ /* resync on DISCONT */
+ if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+ gst_adapter_clear (scope->priv->adapter);
+ }
+
+ /* Make sure have an output format */
+ if (gst_pad_check_reconfigure (scope->priv->srcpad)) {
+ if (!gst_audio_visualizer_src_negotiate (scope)) {
+ gst_pad_mark_reconfigure (scope->priv->srcpad);
+ goto not_negotiated;
+ }
+ }
+
+ channels = GST_AUDIO_INFO_CHANNELS (&scope->ainfo);
+ rate = GST_AUDIO_INFO_RATE (&scope->ainfo);
+ bps = GST_AUDIO_INFO_BPS (&scope->ainfo);
+
+ if (bps == 0) {
+ ret = GST_FLOW_NOT_NEGOTIATED;
+ goto beach;
+ }
+
+ gst_adapter_push (scope->priv->adapter, buffer);
+
+ g_mutex_lock (&scope->priv->config_lock);
+
+ /* this is what we want */
+ sbpf = scope->req_spf * channels * sizeof (gint16);
+
+ inbuf = scope->priv->inbuf;
+ /* FIXME: the timestamp in the adapter would be different */
+ gst_buffer_copy_into (inbuf, buffer, GST_BUFFER_COPY_METADATA, 0, -1);
+
+ /* this is what we have */
+ avail = gst_adapter_available (scope->priv->adapter);
+ GST_LOG_OBJECT (scope, "avail: %u, bpf: %u", avail, sbpf);
+ while (avail >= sbpf) {
+ GstBuffer *outbuf;
+ GstVideoFrame outframe;
+
+ /* get timestamp of the current adapter content */
+ ts = gst_adapter_prev_pts (scope->priv->adapter, &dist);
+ if (GST_CLOCK_TIME_IS_VALID (ts)) {
+ /* convert bytes to time */
+ dist /= bps;
+ ts += gst_util_uint64_scale_int (dist, GST_SECOND, rate);
+ }
+
+ /* check for QoS, don't compute buffers that are known to be late */
+ if (GST_CLOCK_TIME_IS_VALID (ts)) {
+ GstClockTime earliest_time;
+ gdouble proportion;
+ gint64 qostime;
+
+ qostime =
+ gst_segment_to_running_time (&scope->priv->segment,
+ GST_FORMAT_TIME, ts) + scope->priv->frame_duration;
+
+ GST_OBJECT_LOCK (scope);
+ earliest_time = scope->priv->earliest_time;
+ proportion = scope->priv->proportion;
+ GST_OBJECT_UNLOCK (scope);
+
+ if (GST_CLOCK_TIME_IS_VALID (earliest_time) && qostime <= earliest_time) {
+ GstClockTime stream_time, jitter;
+ GstMessage *qos_msg;
+
+ GST_DEBUG_OBJECT (scope,
+ "QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (qostime), GST_TIME_ARGS (earliest_time));
+
+ ++scope->priv->dropped;
+ stream_time = gst_segment_to_stream_time (&scope->priv->segment,
+ GST_FORMAT_TIME, ts);
+ jitter = GST_CLOCK_DIFF (qostime, earliest_time);
+ qos_msg = gst_message_new_qos (GST_OBJECT (scope), FALSE, qostime,
+ stream_time, ts, GST_BUFFER_DURATION (buffer));
+ gst_message_set_qos_values (qos_msg, jitter, proportion, 1000000);
+ gst_message_set_qos_stats (qos_msg, GST_FORMAT_BUFFERS,
+ scope->priv->processed, scope->priv->dropped);
+ gst_element_post_message (GST_ELEMENT (scope), qos_msg);
+
+ goto skip;
+ }
+ }
+
+ ++scope->priv->processed;
+
+ g_mutex_unlock (&scope->priv->config_lock);
+ ret = default_prepare_output_buffer (scope, &outbuf);
+ g_mutex_lock (&scope->priv->config_lock);
+ /* recheck as the value could have changed */
+ sbpf = scope->req_spf * channels * sizeof (gint16);
+
+ /* no buffer allocated, we don't care why. */
+ if (ret != GST_FLOW_OK)
+ break;
+
+ /* sync controlled properties */
+ if (GST_CLOCK_TIME_IS_VALID (ts))
+ gst_object_sync_values (GST_OBJECT (scope), ts);
+
+ GST_BUFFER_TIMESTAMP (outbuf) = ts;
+ GST_BUFFER_DURATION (outbuf) = scope->priv->frame_duration;
+
+ /* this can fail as the data size we need could have changed */
+ if (!(adata = (gpointer) gst_adapter_map (scope->priv->adapter, sbpf)))
+ break;
+
+ gst_video_frame_map (&outframe, &scope->vinfo, outbuf, GST_MAP_READWRITE);
+
+ if (scope->priv->shader) {
+ gst_video_frame_copy (&outframe, &scope->priv->tempframe);
+ } else {
+ /* gst_video_frame_clear() or is output frame already cleared */
+ gint i;
+
+ for (i = 0; i < scope->vinfo.finfo->n_planes; i++) {
+ memset (outframe.data[i], 0, outframe.map[i].size);
+ }
+ }
+
+ gst_buffer_replace_all_memory (inbuf,
+ gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, adata, sbpf, 0,
+ sbpf, NULL, NULL));
+
+ /* call class->render() vmethod */
+ if (klass->render) {
+ if (!klass->render (scope, inbuf, &outframe)) {
+ ret = GST_FLOW_ERROR;
+ gst_video_frame_unmap (&outframe);
+ goto beach;
+ } else {
+ /* run various post processing (shading and geometric transformation) */
+ /* FIXME: SHADER assumes 32bpp */
+ if (scope->priv->shader &&
+ GST_VIDEO_INFO_COMP_PSTRIDE (&scope->vinfo, 0) == 4) {
+ scope->priv->shader (scope, &outframe, &scope->priv->tempframe);
+ }
+ }
+ }
+ gst_video_frame_unmap (&outframe);
+
+ g_mutex_unlock (&scope->priv->config_lock);
+ ret = gst_pad_push (scope->priv->srcpad, outbuf);
+ outbuf = NULL;
+ g_mutex_lock (&scope->priv->config_lock);
+
+ skip:
+ /* recheck as the value could have changed */
+ sbpf = scope->req_spf * channels * sizeof (gint16);
+ GST_LOG_OBJECT (scope, "avail: %u, bpf: %u", avail, sbpf);
+ /* we want to take less or more, depending on spf : req_spf */
+ if (avail - sbpf >= sbpf) {
+ gst_adapter_flush (scope->priv->adapter, sbpf);
+ gst_adapter_unmap (scope->priv->adapter);
+ } else if (avail >= sbpf) {
+ /* just flush a bit and stop */
+ gst_adapter_flush (scope->priv->adapter, (avail - sbpf));
+ gst_adapter_unmap (scope->priv->adapter);
+ break;
+ }
+ avail = gst_adapter_available (scope->priv->adapter);
+
+ if (ret != GST_FLOW_OK)
+ break;
+ }
+
+ g_mutex_unlock (&scope->priv->config_lock);
+
+beach:
+ return ret;
+
+ /* ERRORS */
+not_negotiated:
+ {
+ GST_DEBUG_OBJECT (scope, "Failed to renegotiate");
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+}
+
+static gboolean
+gst_audio_visualizer_src_event (GstPad * pad, GstObject * parent,
+ GstEvent * event)
+{
+ gboolean res;
+ GstAudioVisualizer *scope;
+
+ scope = GST_AUDIO_VISUALIZER (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 (scope);
+ scope->priv->proportion = proportion;
+ if (diff >= 0)
+ /* we're late, this is a good estimate for next displayable
+ * frame (see part-qos.txt) */
+ scope->priv->earliest_time = timestamp + 2 * diff +
+ scope->priv->frame_duration;
+ else
+ scope->priv->earliest_time = timestamp + diff;
+ GST_OBJECT_UNLOCK (scope);
+
+ res = gst_pad_push_event (scope->priv->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_audio_visualizer_sink_event (GstPad * pad, GstObject * parent,
+ GstEvent * event)
+{
+ gboolean res;
+ GstAudioVisualizer *scope;
+
+ scope = GST_AUDIO_VISUALIZER (parent);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_CAPS:
+ {
+ GstCaps *caps;
+
+ gst_event_parse_caps (event, &caps);
+ res = gst_audio_visualizer_sink_setcaps (scope, caps);
+ gst_event_unref (event);
+ break;
+ }
+ case GST_EVENT_FLUSH_STOP:
+ gst_audio_visualizer_reset (scope);
+ res = gst_pad_push_event (scope->priv->srcpad, 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, &scope->priv->segment);
+
+ res = gst_pad_push_event (scope->priv->srcpad, event);
+ break;
+ }
+ default:
+ res = gst_pad_event_default (pad, parent, event);
+ break;
+ }
+
+ return res;
+}
+
+static gboolean
+gst_audio_visualizer_src_query (GstPad * pad, GstObject * parent,
+ GstQuery * query)
+{
+ gboolean res = FALSE;
+ GstAudioVisualizer *scope;
+
+ scope = GST_AUDIO_VISUALIZER (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;
+ gint rate = GST_AUDIO_INFO_RATE (&scope->ainfo);
+
+ if (rate == 0)
+ break;
+
+ if ((res = gst_pad_peer_query (scope->priv->sinkpad, query))) {
+ gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
+
+ GST_DEBUG_OBJECT (scope, "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 buffer */
+ max_samples = MAX (scope->req_spf, scope->priv->spf);
+ our_latency = gst_util_uint64_scale_int (max_samples, GST_SECOND, rate);
+
+ GST_DEBUG_OBJECT (scope, "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 (scope, "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;
+ }
+ default:
+ res = gst_pad_query_default (pad, parent, query);
+ break;
+ }
+
+ return res;
+}
+
+static GstStateChangeReturn
+gst_audio_visualizer_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstStateChangeReturn ret;
+ GstAudioVisualizer *scope;
+
+ scope = GST_AUDIO_VISUALIZER (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_audio_visualizer_reset (scope);
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_audio_visualizer_set_allocation (scope, NULL, NULL, NULL, NULL);
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
/* gstgoom.c: implementation of goom drawing element
* Copyright (C) <2001> Richard Boulton <richard@tartarus.org>
* (C) <2006> Wim Taymans <wim at fluendo dot com>
+ * (C) <2015> Luis de Bethencourt <luis@debethencourt.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
#endif
#include <string.h>
-#include <gst/gst.h>
#include "gstgoom.h"
-#include <gst/video/video.h>
-#include <gst/audio/audio.h>
#include "goom_core.h"
GST_DEBUG_CATEGORY_STATIC (goom_debug);
#define DEFAULT_FPS_N 25
#define DEFAULT_FPS_D 1
-/* signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
-
-enum
-{
- PROP_0
- /* FILL ME */
-};
-
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
static void gst_goom_finalize (GObject * object);
-static GstStateChangeReturn gst_goom_change_state (GstElement * element,
- GstStateChange transition);
+static gboolean gst_goom_setup (GstAudioVisualizer * base);
+static gboolean gst_goom_render (GstAudioVisualizer * base, GstBuffer * audio,
+ GstVideoFrame * video);
-static GstFlowReturn gst_goom_chain (GstPad * pad, GstObject * parent,
- GstBuffer * buffer);
-static gboolean gst_goom_src_event (GstPad * pad, GstObject * parent,
- GstEvent * event);
-static gboolean gst_goom_sink_event (GstPad * pad, GstObject * parent,
- GstEvent * event);
-
-static gboolean gst_goom_src_query (GstPad * pad, GstObject * parent,
- GstQuery * query);
-
-static gboolean gst_goom_src_negotiate (GstGoom * goom);
-
-#define gst_goom_parent_class parent_class
typedef GstGoom GstGoom2k1;
typedef GstGoomClass GstGoom2k1Class;
-G_DEFINE_TYPE (GstGoom2k1, gst_goom, GST_TYPE_ELEMENT);
+
+G_DEFINE_TYPE (GstGoom2k1, gst_goom, GST_TYPE_AUDIO_VISUALIZER);
static void
gst_goom_class_init (GstGoomClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
+ GstAudioVisualizerClass *visualizer_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
-
- parent_class = g_type_class_peek_parent (klass);
+ visualizer_class = (GstAudioVisualizerClass *) klass;
gobject_class->finalize = gst_goom_finalize;
- gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_goom_change_state);
-
gst_element_class_set_static_metadata (gstelement_class,
"GOOM: what a GOOM! 2k1 edition", "Visualization",
"Takes frames of data and outputs video frames using the GOOM 2k1 filter",
gst_static_pad_template_get (&src_template));
GST_DEBUG_CATEGORY_INIT (goom_debug, "goom", 0, "goom visualisation element");
+
+ visualizer_class->setup = GST_DEBUG_FUNCPTR (gst_goom_setup);
+ visualizer_class->render = GST_DEBUG_FUNCPTR (gst_goom_render);
}
static void
gst_goom_init (GstGoom * goom)
{
- /* create the sink and src pads */
- goom->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
- gst_pad_set_chain_function (goom->sinkpad,
- GST_DEBUG_FUNCPTR (gst_goom_chain));
- gst_pad_set_event_function (goom->sinkpad,
- GST_DEBUG_FUNCPTR (gst_goom_sink_event));
- gst_element_add_pad (GST_ELEMENT (goom), goom->sinkpad);
-
- goom->srcpad = gst_pad_new_from_static_template (&src_template, "src");
- gst_pad_set_event_function (goom->srcpad,
- GST_DEBUG_FUNCPTR (gst_goom_src_event));
- gst_pad_set_query_function (goom->srcpad,
- GST_DEBUG_FUNCPTR (gst_goom_src_query));
- gst_element_add_pad (GST_ELEMENT (goom), goom->srcpad);
-
- goom->adapter = gst_adapter_new ();
-
goom->width = DEFAULT_WIDTH;
goom->height = DEFAULT_HEIGHT;
goom->fps_n = DEFAULT_FPS_N; /* desired frame rate */
goom_close (&(goom->goomdata));
- g_object_unref (goom->adapter);
-
- G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static void
-gst_goom_reset (GstGoom * goom)
-{
- gst_adapter_clear (goom->adapter);
- gst_segment_init (&goom->segment, GST_FORMAT_UNDEFINED);
-
- GST_OBJECT_LOCK (goom);
- goom->proportion = 1.0;
- goom->earliest_time = GST_CLOCK_TIME_NONE;
- GST_OBJECT_UNLOCK (goom);
-
- goom->dropped = 0;
- goom->processed = 0;
-}
-
-static gboolean
-gst_goom_sink_setcaps (GstGoom * goom, GstCaps * caps)
-{
- GstStructure *structure;
- gboolean res;
-
- structure = gst_caps_get_structure (caps, 0);
-
- res = gst_structure_get_int (structure, "channels", &goom->channels);
- res &= gst_structure_get_int (structure, "rate", &goom->rate);
- if (!res)
- return FALSE;
-
- goom->bps = goom->channels * sizeof (gint16);
-
- return gst_goom_src_negotiate (goom);
+ G_OBJECT_CLASS (gst_goom_parent_class)->finalize (object);
}
static gboolean
-gst_goom_src_setcaps (GstGoom * goom, GstCaps * caps)
+gst_goom_setup (GstAudioVisualizer * base)
{
- GstStructure *structure;
-
- structure = gst_caps_get_structure (caps, 0);
+ GstGoom *goom = GST_GOOM (base);
- if (!gst_structure_get_int (structure, "width", &goom->width) ||
- !gst_structure_get_int (structure, "height", &goom->height) ||
- !gst_structure_get_fraction (structure, "framerate", &goom->fps_n,
- &goom->fps_d))
- goto error;
+ goom->width = GST_VIDEO_INFO_WIDTH (&base->vinfo);
+ goom->height = GST_VIDEO_INFO_HEIGHT (&base->vinfo);
goom_set_resolution (&(goom->goomdata), goom->width, goom->height);
- /* size of the output buffer in bytes, depth is always 4 bytes */
- goom->outsize = goom->width * goom->height * 4;
- goom->duration =
- gst_util_uint64_scale_int (GST_SECOND, goom->fps_d, goom->fps_n);
- goom->spf = gst_util_uint64_scale_int (goom->rate, goom->fps_d, goom->fps_n);
- goom->bpf = goom->spf * goom->bps;
-
- GST_DEBUG_OBJECT (goom, "dimension %dx%d, framerate %d/%d, spf %d",
- goom->width, goom->height, goom->fps_n, goom->fps_d, goom->spf);
-
- return gst_pad_set_caps (goom->srcpad, caps);
-
- /* ERRORS */
-error:
- {
- GST_DEBUG_OBJECT (goom, "error parsing caps");
- return FALSE;
- }
-}
-
-static gboolean
-gst_goom_src_negotiate (GstGoom * goom)
-{
- GstCaps *othercaps, *target;
- GstStructure *structure;
- GstCaps *templ;
- GstQuery *query;
- GstBufferPool *pool = NULL;
- GstStructure *config;
- guint size, min, max;
-
- templ = gst_pad_get_pad_template_caps (goom->srcpad);
-
- GST_DEBUG_OBJECT (goom, "performing negotiation");
-
- /* see what the peer can do */
- othercaps = gst_pad_peer_query_caps (goom->srcpad, NULL);
- if (othercaps) {
- target = gst_caps_intersect (othercaps, templ);
- gst_caps_unref (templ);
- gst_caps_unref (othercaps);
-
- if (gst_caps_is_empty (target))
- goto no_format;
-
- target = gst_caps_truncate (target);
- } else {
- target = gst_caps_copy (templ);
- }
-
- 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);
-
- gst_goom_src_setcaps (goom, target);
-
- /* find a pool for the negotiated caps now */
- query = gst_query_new_allocation (target, TRUE);
-
- if (!gst_pad_peer_query (goom->srcpad, query)) {
- /* no problem, we use the query defaults */
- GST_DEBUG_OBJECT (goom, "ALLOCATION query failed");
- }
-
- if (gst_query_get_n_allocation_pools (query) > 0) {
- /* we got configuration from our peer, parse them */
- gst_query_parse_nth_allocation_pool (query, 0, &pool, &size, &min, &max);
- } else {
- pool = NULL;
- size = goom->outsize;
- min = max = 0;
- }
-
- if (pool == NULL) {
- /* we did not get a pool, make one ourselves then */
- pool = gst_buffer_pool_new ();
- }
-
- 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 (goom->pool) {
- gst_buffer_pool_set_active (goom->pool, FALSE);
- gst_object_unref (goom->pool);
- }
- goom->pool = pool;
-
- /* and activate */
- gst_buffer_pool_set_active (pool, TRUE);
-
- gst_caps_unref (target);
-
return TRUE;
-
-no_format:
- {
- gst_caps_unref (target);
- return FALSE;
- }
}
static gboolean
-gst_goom_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
-{
- gboolean res;
- GstGoom *goom;
-
- goom = GST_GOOM (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 (goom);
- goom->proportion = proportion;
- if (diff >= 0)
- /* we're late, this is a good estimate for next displayable
- * frame (see part-qos.txt) */
- goom->earliest_time = timestamp + 2 * diff + goom->duration;
- else
- goom->earliest_time = timestamp + diff;
- GST_OBJECT_UNLOCK (goom);
-
- res = gst_pad_event_default (pad, parent, event);
- break;
+gst_goom_render (GstAudioVisualizer * base, GstBuffer * audio,
+ GstVideoFrame * video)
+{
+ GstGoom *goom = GST_GOOM (base);
+ GstMapInfo amap;
+ gint16 *adata;
+ gint i;
+
+ /* get next GOOM_SAMPLES, we have at least this amount of samples */
+ gst_buffer_map (audio, &amap, GST_MAP_READ);
+ adata = (gint16 *) amap.data;
+
+ if (goom->channels == 2) {
+ for (i = 0; i < GOOM_SAMPLES; i++) {
+ goom->datain[0][i] = *adata++;
+ goom->datain[1][i] = *adata++;
}
- default:
- res = gst_pad_event_default (pad, parent, event);
- break;
- }
-
- return res;
-}
-
-static gboolean
-gst_goom_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
-{
- gboolean res;
- GstGoom *goom;
-
- goom = GST_GOOM (parent);
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_CAPS:
- {
- GstCaps *caps;
-
- gst_event_parse_caps (event, &caps);
- res = gst_goom_sink_setcaps (goom, caps);
- gst_event_unref (event);
- break;
- }
- case GST_EVENT_FLUSH_STOP:
- gst_goom_reset (goom);
- res = gst_pad_event_default (pad, parent, 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, &goom->segment);
-
- res = gst_pad_event_default (pad, parent, event);
- break;
- }
- default:
- res = gst_pad_event_default (pad, parent, event);
- break;
- }
-
- return res;
-}
-
-static gboolean
-gst_goom_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
-{
- gboolean res;
- GstGoom *goom;
-
- goom = GST_GOOM (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 (goom->sinkpad, query))) {
- gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
-
- GST_DEBUG_OBJECT (goom, "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 buffer */
- max_samples = MAX (GOOM_SAMPLES, goom->spf);
- our_latency =
- gst_util_uint64_scale_int (max_samples, GST_SECOND, goom->rate);
-
- GST_DEBUG_OBJECT (goom, "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 (goom, "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;
- }
- default:
- res = gst_pad_query_default (pad, parent, query);
- break;
- }
-
- return res;
-}
-
-/* make sure we are negotiated */
-static GstFlowReturn
-ensure_negotiated (GstGoom * goom)
-{
- if (gst_pad_check_reconfigure (goom->srcpad)) {
- if (!gst_goom_src_negotiate (goom))
- return GST_FLOW_NOT_NEGOTIATED;
- }
- return GST_FLOW_OK;
-}
-
-static GstFlowReturn
-gst_goom_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
-{
- GstGoom *goom;
- GstFlowReturn ret;
- GstBuffer *outbuf = NULL;
-
- goom = GST_GOOM (parent);
-
- /* If we don't have an output format yet, preallocate a buffer to try and
- * set one */
- if (goom->bps == 0) {
- gst_buffer_unref (buffer);
- ret = GST_FLOW_NOT_NEGOTIATED;
- goto beach;
- }
-
- /* Make sure have an output format */
- ret = ensure_negotiated (goom);
- if (ret != GST_FLOW_OK) {
- gst_buffer_unref (buffer);
- goto beach;
- }
-
- /* don't try to combine samples from discont buffer */
- if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
- gst_adapter_clear (goom->adapter);
- }
-
- GST_DEBUG_OBJECT (goom,
- "Input buffer has %" G_GSIZE_FORMAT " samples, time=%" G_GUINT64_FORMAT,
- gst_buffer_get_size (buffer) / goom->bps, GST_BUFFER_TIMESTAMP (buffer));
-
- /* Collect samples until we have enough for an output frame */
- gst_adapter_push (goom->adapter, buffer);
-
- ret = GST_FLOW_OK;
-
- while (TRUE) {
- const guint16 *data;
- guchar *out_frame;
- gint i;
- guint avail, to_flush;
- guint64 dist, timestamp;
-
- avail = gst_adapter_available (goom->adapter);
- GST_DEBUG_OBJECT (goom, "avail now %u", avail);
-
- /* we need GOOM_SAMPLES to get a meaningful result from goom. */
- if (avail < (GOOM_SAMPLES * goom->bps))
- break;
-
- /* we also need enough samples to produce one frame at least */
- if (avail < goom->bpf)
- break;
-
- GST_DEBUG_OBJECT (goom, "processing buffer");
-
- /* get timestamp of the current adapter byte */
- timestamp = gst_adapter_prev_pts (goom->adapter, &dist);
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- /* convert bytes to time */
- dist /= goom->bps;
- timestamp += gst_util_uint64_scale_int (dist, GST_SECOND, goom->rate);
- }
-
- /* check for QoS, don't compute buffers that are known to be late */
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- GstClockTime earliest_time;
- gdouble proportion;
- gint64 qostime;
-
- qostime = gst_segment_to_running_time (&goom->segment, GST_FORMAT_TIME,
- timestamp);
- qostime += goom->duration;
-
- GST_OBJECT_LOCK (goom);
- earliest_time = goom->earliest_time;
- proportion = goom->proportion;
- GST_OBJECT_UNLOCK (goom);
-
- if (GST_CLOCK_TIME_IS_VALID (earliest_time) && qostime <= earliest_time) {
- GstClockTime stream_time, jitter;
- GstMessage *qos_msg;
-
- GST_DEBUG_OBJECT (goom,
- "QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
- GST_TIME_ARGS (qostime), GST_TIME_ARGS (earliest_time));
-
- ++goom->dropped;
- stream_time = gst_segment_to_stream_time (&goom->segment,
- GST_FORMAT_TIME, timestamp);
- jitter = GST_CLOCK_DIFF (qostime, earliest_time);
- qos_msg = gst_message_new_qos (GST_OBJECT (goom), FALSE, qostime,
- stream_time, timestamp, GST_BUFFER_DURATION (buffer));
- gst_message_set_qos_values (qos_msg, jitter, proportion, 1000000);
- gst_message_set_qos_stats (qos_msg, GST_FORMAT_BUFFERS,
- goom->processed, goom->dropped);
- gst_element_post_message (GST_ELEMENT (goom), qos_msg);
- goto skip;
- }
- }
-
- ++goom->processed;
-
- /* get next GOOM_SAMPLES, we have at least this amount of samples */
- data =
- (const guint16 *) gst_adapter_map (goom->adapter,
- GOOM_SAMPLES * goom->bps);
-
- if (goom->channels == 2) {
- for (i = 0; i < GOOM_SAMPLES; i++) {
- goom->datain[0][i] = *data++;
- goom->datain[1][i] = *data++;
- }
- } else {
- for (i = 0; i < GOOM_SAMPLES; i++) {
- goom->datain[0][i] = *data;
- goom->datain[1][i] = *data++;
- }
- }
-
- /* 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 (goom, "allocating output buffer");
- ret = gst_buffer_pool_acquire_buffer (goom->pool, &outbuf, NULL);
- if (ret != GST_FLOW_OK) {
- gst_adapter_unmap (goom->adapter);
- goto beach;
- }
+ } else {
+ for (i = 0; i < GOOM_SAMPLES; i++) {
+ goom->datain[0][i] = *adata;
+ goom->datain[1][i] = *adata++;
}
-
- GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
- GST_BUFFER_DURATION (outbuf) = goom->duration;
-
- out_frame = (guchar *) goom_update (&(goom->goomdata), goom->datain);
- gst_buffer_fill (outbuf, 0, out_frame, goom->outsize);
-
- gst_adapter_unmap (goom->adapter);
-
- GST_DEBUG ("Pushing frame with time=%" GST_TIME_FORMAT ", duration=%"
- GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (goom->duration));
-
- ret = gst_pad_push (goom->srcpad, outbuf);
- outbuf = NULL;
-
- skip:
- /* Now flush the samples we needed for this frame, which might be more than
- * the samples we used (GOOM_SAMPLES). */
- to_flush = goom->bpf;
-
- GST_DEBUG_OBJECT (goom, "finished frame, flushing %u bytes from input",
- to_flush);
- gst_adapter_flush (goom->adapter, to_flush);
-
- if (ret != GST_FLOW_OK)
- break;
}
- if (outbuf != NULL)
- gst_buffer_unref (outbuf);
+ video->data[0] = goom_update (&(goom->goomdata), goom->datain);
+ gst_buffer_unmap (audio, &amap);
-beach:
-
- return ret;
-}
-
-static GstStateChangeReturn
-gst_goom_change_state (GstElement * element, GstStateChange transition)
-{
- GstGoom *goom = GST_GOOM (element);
- GstStateChangeReturn ret;
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- gst_goom_reset (goom);
- break;
- default:
- break;
- }
-
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- switch (transition) {
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- if (goom->pool) {
- gst_buffer_pool_set_active (goom->pool, FALSE);
- gst_object_replace ((GstObject **) & goom->pool, NULL);
- }
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
- }
-
- return ret;
+ return TRUE;
}
static gboolean
projects / platform / upstream / gst-plugins-good.git / commitdiff