--- /dev/null
+/* GStreamer
+ * Copyright (C) <2011> Stefan Kost <ensonic@users.sf.net>
+ *
+ * gstbaseaudiovisualizer.h: base 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+/**
+ * SECTION:gstbaseaudiovisualizer
+ *
+ * A basclass 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
+
+/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
+ * with newer GLib versions (>= 2.31.0) */
+#define GLIB_DISABLE_DEPRECATION_WARNINGS
+
+#include <string.h>
+
+#include "gstbaseaudiovisualizer.h"
+
+GST_DEBUG_CATEGORY_STATIC (base_audio_visualizer_debug);
+#define GST_CAT_DEFAULT (base_audio_visualizer_debug)
+
+#define DEFAULT_SHADER GST_BASE_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_base_audio_visualizer_class_init (GstBaseAudioVisualizerClass *
+ klass);
+static void gst_base_audio_visualizer_init (GstBaseAudioVisualizer * scope,
+ GstBaseAudioVisualizerClass * g_class);
+static void gst_base_audio_visualizer_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_base_audio_visualizer_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+static void gst_base_audio_visualizer_dispose (GObject * object);
+
+static gboolean gst_base_audio_visualizer_src_negotiate (GstBaseAudioVisualizer
+ * scope);
+static gboolean gst_base_audio_visualizer_src_setcaps (GstBaseAudioVisualizer *
+ scope, GstCaps * caps);
+static gboolean gst_base_audio_visualizer_sink_setcaps (GstBaseAudioVisualizer *
+ scope, GstCaps * caps);
+
+static GstFlowReturn gst_base_audio_visualizer_chain (GstPad * pad,
+ GstObject * parent, GstBuffer * buffer);
+
+static gboolean gst_base_audio_visualizer_src_event (GstPad * pad,
+ GstObject * parent, GstEvent * event);
+static gboolean gst_base_audio_visualizer_sink_event (GstPad * pad,
+ GstObject * parent, GstEvent * event);
+
+static gboolean gst_base_audio_visualizer_src_query (GstPad * pad,
+ GstObject * parent, GstQuery * query);
+static gboolean gst_base_audio_visualizer_sink_query (GstPad * pad,
+ GstObject * parent, GstQuery * query);
+
+static GstStateChangeReturn gst_base_audio_visualizer_change_state (GstElement *
+ element, GstStateChange transition);
+
+/* shading functions */
+
+#define GST_TYPE_BASE_AUDIO_VISUALIZER_SHADER (gst_base_audio_visualizer_shader_get_type())
+static GType
+gst_base_audio_visualizer_shader_get_type (void)
+{
+ static GType shader_type = 0;
+ static const GEnumValue shaders[] = {
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_NONE, "None", "none"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE, "Fade", "fade"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP, "Fade and move up",
+ "fade-and-move-up"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN, "Fade and move down",
+ "fade-and-move-down"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT, "Fade and move left",
+ "fade-and-move-left"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT,
+ "Fade and move right",
+ "fade-and-move-right"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT,
+ "Fade and move horizontally out", "fade-and-move-horiz-out"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN,
+ "Fade and move horizontally in", "fade-and-move-horiz-in"},
+ {GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT,
+ "Fade and move vertically out", "fade-and-move-vert-out"},
+ {GST_BASE_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)) {
+ shader_type =
+ g_enum_register_static ("GstBaseAudioVisualizerShader", shaders);
+ }
+ return shader_type;
+}
+
+/* we're only supporting GST_VIDEO_FORMAT_xRGB right now) */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+
+#define SHADE1(_d, _s, _i, _r, _g, _b) \
+G_STMT_START { \
+ _d[_i] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
+ _i++; \
+ _d[_i] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
+ _i++; \
+ _d[_i] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
+ _i++; \
+ _d[_i++] = 0; \
+} G_STMT_END
+
+#define SHADE2(_d, _s, _j, _i, _r, _g, _b) \
+G_STMT_START { \
+ _d[_j++] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
+ _i++; \
+ _d[_j++] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
+ _i++; \
+ _d[_j++] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
+ _i++; \
+ _d[_j++] = 0; \
+ _i++; \
+} G_STMT_END
+
+#else
+
+#define SHADE1(_d, _s, _i, _r, _g, _b) \
+G_STMT_START { \
+ _d[_i++] = 0; \
+ _d[_i] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
+ _i++; \
+ _d[_i] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
+ _i++; \
+ _d[_i] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
+ _i++; \
+} G_STMT_END
+
+#define SHADE2(_d, _s, _j, _i, _r, _g, _b) \
+G_STMT_START { \
+ _d[_j++] = 0; \
+ _i++; \
+ _d[_j++] = (_s[_i] > _r) ? _s[_i] - _r : 0; \
+ _i++; \
+ _d[_j++] = (_s[_i] > _g) ? _s[_i] - _g : 0; \
+ _i++; \
+ _d[_j++] = (_s[_i] > _b) ? _s[_i] - _b : 0; \
+ _i++; \
+} G_STMT_END
+
+#endif
+
+static void
+shader_fade (GstBaseAudioVisualizer * scope, const guint8 * s, guint8 * d)
+{
+ guint i, bpf = scope->bpf;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ for (i = 0; i < bpf;) {
+ SHADE1 (d, s, i, r, g, b);
+ }
+}
+
+static void
+shader_fade_and_move_up (GstBaseAudioVisualizer * scope, const guint8 * s,
+ guint8 * d)
+{
+ guint i, j, bpf = scope->bpf;
+ guint bpl = 4 * scope->width;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ for (j = 0, i = bpl; i < bpf;) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+}
+
+static void
+shader_fade_and_move_down (GstBaseAudioVisualizer * scope, const guint8 * s,
+ guint8 * d)
+{
+ guint i, j, bpf = scope->bpf;
+ guint bpl = 4 * scope->width;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ for (j = bpl, i = 0; j < bpf;) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+}
+
+static void
+shader_fade_and_move_left (GstBaseAudioVisualizer * scope,
+ const guint8 * s, guint8 * d)
+{
+ guint i, j, k, bpf = scope->bpf;
+ guint w = scope->width;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ /* move to the left */
+ for (j = 0, i = 4; i < bpf;) {
+ for (k = 0; k < w - 1; k++) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ i += 4;
+ j += 4;
+ }
+}
+
+static void
+shader_fade_and_move_right (GstBaseAudioVisualizer * scope,
+ const guint8 * s, guint8 * d)
+{
+ guint i, j, k, bpf = scope->bpf;
+ guint w = scope->width;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ /* move to the left */
+ for (j = 4, i = 0; i < bpf;) {
+ for (k = 0; k < w - 1; k++) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ i += 4;
+ j += 4;
+ }
+}
+
+static void
+shader_fade_and_move_horiz_out (GstBaseAudioVisualizer * scope,
+ const guint8 * s, guint8 * d)
+{
+ guint i, j, bpf = scope->bpf / 2;
+ guint bpl = 4 * scope->width;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ /* move upper half up */
+ for (j = 0, i = bpl; i < bpf;) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ /* move lower half down */
+ for (j = bpf + bpl, i = bpf; j < bpf + bpf;) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+}
+
+static void
+shader_fade_and_move_horiz_in (GstBaseAudioVisualizer * scope,
+ const guint8 * s, guint8 * d)
+{
+ guint i, j, bpf = scope->bpf / 2;
+ guint bpl = 4 * scope->width;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ /* move upper half down */
+ for (i = 0, j = bpl; i < bpf;) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ /* move lower half up */
+ for (i = bpf + bpl, j = bpf; i < bpf + bpf;) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+}
+
+static void
+shader_fade_and_move_vert_out (GstBaseAudioVisualizer * scope,
+ const guint8 * s, guint8 * d)
+{
+ guint i, j, k, bpf = scope->bpf;
+ guint m = scope->width / 2;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ /* move left half to the left */
+ for (j = 0, i = 4; i < bpf;) {
+ for (k = 0; k < m; k++) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ j += 4 * m;
+ i += 4 * m;
+ }
+ /* move right half to the right */
+ for (j = 4 * (m + 1), i = 4 * m; j < bpf;) {
+ for (k = 0; k < m; k++) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ j += 4 * m;
+ i += 4 * m;
+ }
+}
+
+static void
+shader_fade_and_move_vert_in (GstBaseAudioVisualizer * scope,
+ const guint8 * s, guint8 * d)
+{
+ guint i, j, k, bpf = scope->bpf;
+ guint m = scope->width / 2;
+ guint r = (scope->shade_amount >> 16) & 0xff;
+ guint g = (scope->shade_amount >> 8) & 0xff;
+ guint b = (scope->shade_amount >> 0) & 0xff;
+
+ /* move left half to the right */
+ for (j = 4, i = 0; j < bpf;) {
+ for (k = 0; k < m; k++) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ j += 4 * m;
+ i += 4 * m;
+ }
+ /* move right half to the left */
+ for (j = 4 * m, i = 4 * (m + 1); i < bpf;) {
+ for (k = 0; k < m; k++) {
+ SHADE2 (d, s, j, i, r, g, b);
+ }
+ j += 4 * m;
+ i += 4 * m;
+ }
+}
+
+static void
+gst_base_audio_visualizer_change_shader (GstBaseAudioVisualizer * scope)
+{
+ switch (scope->shader_type) {
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_NONE:
+ scope->shader = NULL;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE:
+ scope->shader = shader_fade;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP:
+ scope->shader = shader_fade_and_move_up;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN:
+ scope->shader = shader_fade_and_move_down;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT:
+ scope->shader = shader_fade_and_move_left;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT:
+ scope->shader = shader_fade_and_move_right;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT:
+ scope->shader = shader_fade_and_move_horiz_out;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN:
+ scope->shader = shader_fade_and_move_horiz_in;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT:
+ scope->shader = shader_fade_and_move_vert_out;
+ break;
+ case GST_BASE_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_IN:
+ scope->shader = shader_fade_and_move_vert_in;
+ break;
+ default:
+ GST_ERROR ("invalid shader function");
+ scope->shader = NULL;
+ break;
+ }
+}
+
+/* base class */
+
+GType
+gst_base_audio_visualizer_get_type (void)
+{
+ static volatile gsize base_audio_visualizer_type = 0;
+
+ if (g_once_init_enter (&base_audio_visualizer_type)) {
+ static const GTypeInfo base_audio_visualizer_info = {
+ sizeof (GstBaseAudioVisualizerClass),
+ NULL,
+ NULL,
+ (GClassInitFunc) gst_base_audio_visualizer_class_init,
+ NULL,
+ NULL,
+ sizeof (GstBaseAudioVisualizer),
+ 0,
+ (GInstanceInitFunc) gst_base_audio_visualizer_init,
+ };
+ GType _type;
+
+ _type = g_type_register_static (GST_TYPE_ELEMENT,
+ "GstBaseAudioVisualizer", &base_audio_visualizer_info,
+ G_TYPE_FLAG_ABSTRACT);
+ g_once_init_leave (&base_audio_visualizer_type, _type);
+ }
+ return (GType) base_audio_visualizer_type;
+}
+
+static void
+gst_base_audio_visualizer_class_init (GstBaseAudioVisualizerClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstElementClass *element_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ GST_DEBUG_CATEGORY_INIT (base_audio_visualizer_debug, "baseaudiovisualizer",
+ 0, "scope audio visualisation base class");
+
+ gobject_class->set_property = gst_base_audio_visualizer_set_property;
+ gobject_class->get_property = gst_base_audio_visualizer_get_property;
+ gobject_class->dispose = gst_base_audio_visualizer_dispose;
+
+ element_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_change_state);
+
+ 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_BASE_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_base_audio_visualizer_init (GstBaseAudioVisualizer * scope,
+ GstBaseAudioVisualizerClass * g_class)
+{
+ GstPadTemplate *pad_template;
+
+ /* 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->sinkpad = gst_pad_new_from_template (pad_template, "sink");
+ gst_pad_set_chain_function (scope->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_chain));
+ gst_pad_set_event_function (scope->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_sink_event));
+ gst_pad_set_query_function (scope->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_sink_query));
+ gst_element_add_pad (GST_ELEMENT (scope), scope->sinkpad);
+
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
+ g_return_if_fail (pad_template != NULL);
+ scope->srcpad = gst_pad_new_from_template (pad_template, "src");
+ gst_pad_set_event_function (scope->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_src_event));
+ gst_pad_set_query_function (scope->srcpad,
+ GST_DEBUG_FUNCPTR (gst_base_audio_visualizer_src_query));
+ gst_element_add_pad (GST_ELEMENT (scope), scope->srcpad);
+
+ scope->adapter = gst_adapter_new ();
+ scope->inbuf = gst_buffer_new ();
+
+ /* properties */
+ scope->shader_type = DEFAULT_SHADER;
+ gst_base_audio_visualizer_change_shader (scope);
+ scope->shade_amount = DEFAULT_SHADE_AMOUNT;
+
+ /* reset the initial video state */
+ scope->width = 320;
+ scope->height = 200;
+ scope->fps_n = 25; /* desired frame rate */
+ scope->fps_d = 1;
+ scope->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->config_lock);
+}
+
+static void
+gst_base_audio_visualizer_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstBaseAudioVisualizer *scope = GST_BASE_AUDIO_VISUALIZER (object);
+
+ switch (prop_id) {
+ case PROP_SHADER:
+ scope->shader_type = g_value_get_enum (value);
+ gst_base_audio_visualizer_change_shader (scope);
+ break;
+ case PROP_SHADE_AMOUNT:
+ scope->shade_amount = g_value_get_uint (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_base_audio_visualizer_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstBaseAudioVisualizer *scope = GST_BASE_AUDIO_VISUALIZER (object);
+
+ switch (prop_id) {
+ case PROP_SHADER:
+ g_value_set_enum (value, scope->shader_type);
+ break;
+ case PROP_SHADE_AMOUNT:
+ g_value_set_uint (value, scope->shade_amount);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_base_audio_visualizer_dispose (GObject * object)
+{
+ GstBaseAudioVisualizer *scope = GST_BASE_AUDIO_VISUALIZER (object);
+
+ if (scope->adapter) {
+ g_object_unref (scope->adapter);
+ scope->adapter = NULL;
+ }
+ if (scope->inbuf) {
+ gst_buffer_unref (scope->inbuf);
+ scope->inbuf = NULL;
+ }
+ if (scope->pixelbuf) {
+ g_free (scope->pixelbuf);
+ scope->pixelbuf = NULL;
+ }
+ if (scope->config_lock.p) {
+ g_mutex_clear (&scope->config_lock);
+ scope->config_lock.p = NULL;
+ }
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_base_audio_visualizer_reset (GstBaseAudioVisualizer * scope)
+{
+ gst_adapter_clear (scope->adapter);
+ gst_segment_init (&scope->segment, GST_FORMAT_UNDEFINED);
+
+ GST_OBJECT_LOCK (scope);
+ scope->proportion = 1.0;
+ scope->earliest_time = -1;
+ GST_OBJECT_UNLOCK (scope);
+}
+
+static gboolean
+gst_base_audio_visualizer_sink_setcaps (GstBaseAudioVisualizer * scope,
+ GstCaps * caps)
+{
+ GstAudioInfo info;
+ gboolean res = TRUE;
+
+ 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));
+
+done:
+ return res;
+
+ /* Errors */
+wrong_caps:
+ {
+ GST_WARNING_OBJECT (scope, "could not parse caps");
+ res = FALSE;
+ goto done;
+ }
+}
+
+static gboolean
+gst_base_audio_visualizer_src_setcaps (GstBaseAudioVisualizer * scope,
+ GstCaps * caps)
+{
+ GstVideoInfo info;
+ GstBaseAudioVisualizerClass *klass;
+ GstStructure *structure;
+ gboolean res;
+
+ if (!gst_video_info_from_caps (&info, caps))
+ goto wrong_caps;
+
+ structure = gst_caps_get_structure (caps, 0);
+ if (!gst_structure_get_int (structure, "width", &scope->width) ||
+ !gst_structure_get_int (structure, "height", &scope->height) ||
+ !gst_structure_get_fraction (structure, "framerate", &scope->fps_n,
+ &scope->fps_d))
+ goto wrong_caps;
+
+ klass = GST_BASE_AUDIO_VISUALIZER_CLASS (G_OBJECT_GET_CLASS (scope));
+
+ scope->vinfo = info;
+ scope->video_format = info.finfo->format;
+
+ scope->frame_duration = gst_util_uint64_scale_int (GST_SECOND,
+ scope->fps_d, scope->fps_n);
+ scope->spf = gst_util_uint64_scale_int (GST_AUDIO_INFO_RATE (&scope->ainfo),
+ scope->fps_d, scope->fps_n);
+ scope->req_spf = scope->spf;
+
+ scope->bpf = scope->width * scope->height * 4;
+
+ if (scope->pixelbuf)
+ g_free (scope->pixelbuf);
+ scope->pixelbuf = g_malloc0 (scope->bpf);
+
+ if (klass->setup)
+ res = klass->setup (scope);
+
+ GST_DEBUG_OBJECT (scope, "video: dimension %dx%d, framerate %d/%d",
+ scope->width, scope->height, scope->fps_n, scope->fps_d);
+ GST_DEBUG_OBJECT (scope, "blocks: spf %u, req_spf %u",
+ scope->spf, scope->req_spf);
+
+ res = gst_pad_set_caps (scope->srcpad, caps);
+
+ return res;
+
+ /* ERRORS */
+wrong_caps:
+ {
+ GST_DEBUG_OBJECT (scope, "error parsing caps");
+ return FALSE;
+ }
+}
+
+static gboolean
+gst_base_audio_visualizer_src_negotiate (GstBaseAudioVisualizer * scope)
+{
+ GstCaps *othercaps, *target;
+ GstStructure *structure;
+ GstCaps *templ;
+ GstQuery *query;
+ GstBufferPool *pool;
+ GstStructure *config;
+ guint size, min, max;
+
+ templ = gst_pad_get_pad_template_caps (scope->srcpad);
+
+ GST_DEBUG_OBJECT (scope, "performing negotiation");
+
+ /* see what the peer can do */
+ othercaps = gst_pad_peer_query_caps (scope->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", scope->width);
+ gst_structure_fixate_field_nearest_int (structure, "height", scope->height);
+ gst_structure_fixate_field_nearest_fraction (structure, "framerate",
+ scope->fps_n, scope->fps_d);
+
+ GST_DEBUG_OBJECT (scope, "final caps are %" GST_PTR_FORMAT, target);
+
+ gst_base_audio_visualizer_src_setcaps (scope, 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 (scope->srcpad, query)) {
+ /* not a problem, we use the query defaults */
+ GST_DEBUG_OBJECT (scope, "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 = scope->bpf;
+ 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 (scope->pool) {
+ gst_buffer_pool_set_active (scope->pool, FALSE);
+ gst_object_unref (scope->pool);
+ }
+ scope->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;
+ }
+}
+
+/* make sure we are negotiated */
+static GstFlowReturn
+gst_base_audio_visualizer_ensure_negotiated (GstBaseAudioVisualizer * scope)
+{
+ gboolean reconfigure;
+
+ reconfigure = gst_pad_check_reconfigure (scope->srcpad);
+
+ /* we don't know an output format yet, pick one */
+ if (reconfigure || !gst_pad_has_current_caps (scope->srcpad)) {
+ if (!gst_base_audio_visualizer_src_negotiate (scope))
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+ return GST_FLOW_OK;
+}
+
+static GstFlowReturn
+gst_base_audio_visualizer_chain (GstPad * pad, GstObject * parent,
+ GstBuffer * buffer)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBaseAudioVisualizer *scope;
+ GstBaseAudioVisualizerClass *klass;
+ GstBuffer *inbuf;
+ guint64 dist, ts;
+ guint avail, sbpf;
+ gpointer adata;
+ gboolean (*render) (GstBaseAudioVisualizer * scope, GstBuffer * audio,
+ GstBuffer * video);
+ gint bps, channels, rate;
+
+ scope = GST_BASE_AUDIO_VISUALIZER (parent);
+ klass = GST_BASE_AUDIO_VISUALIZER_CLASS (G_OBJECT_GET_CLASS (scope));
+
+ render = klass->render;
+
+ GST_LOG_OBJECT (scope, "chainfunc called");
+
+ /* resync on DISCONT */
+ if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
+ gst_adapter_clear (scope->adapter);
+ }
+
+ /* Make sure have an output format */
+ ret = gst_base_audio_visualizer_ensure_negotiated (scope);
+ if (ret != GST_FLOW_OK) {
+ gst_buffer_unref (buffer);
+ goto beach;
+ }
+ 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->adapter, buffer);
+
+ g_mutex_lock (&scope->config_lock);
+
+ /* this is what we want */
+ sbpf = scope->req_spf * channels * sizeof (gint16);
+
+ inbuf = scope->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->adapter);
+ GST_LOG_OBJECT (scope, "avail: %u, bpf: %u", avail, sbpf);
+ while (avail >= sbpf) {
+ GstBuffer *outbuf;
+ GstMapInfo map;
+
+ /* get timestamp of the current adapter content */
+ ts = gst_adapter_prev_timestamp (scope->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);
+ }
+
+ if (GST_CLOCK_TIME_IS_VALID (ts)) {
+ gint64 qostime;
+ gboolean need_skip;
+
+ qostime =
+ gst_segment_to_running_time (&scope->segment, GST_FORMAT_TIME, ts) +
+ scope->frame_duration;
+
+ GST_OBJECT_LOCK (scope);
+ /* check for QoS, don't compute buffers that are known to be late */
+ need_skip = scope->earliest_time != -1 && qostime <= scope->earliest_time;
+ GST_OBJECT_UNLOCK (scope);
+
+ if (need_skip) {
+ GST_WARNING_OBJECT (scope,
+ "QoS: skip ts: %" GST_TIME_FORMAT ", earliest: %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (qostime), GST_TIME_ARGS (scope->earliest_time));
+ goto skip;
+ }
+ }
+
+ g_mutex_unlock (&scope->config_lock);
+ ret = gst_buffer_pool_acquire_buffer (scope->pool, &outbuf, NULL);
+ g_mutex_lock (&scope->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 */
+ gst_object_sync_values (GST_OBJECT (scope), ts);
+
+ GST_BUFFER_TIMESTAMP (outbuf) = ts;
+ GST_BUFFER_DURATION (outbuf) = scope->frame_duration;
+
+ gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
+ if (scope->shader) {
+ memcpy (map.data, scope->pixelbuf, scope->bpf);
+ } else {
+ memset (map.data, 0, scope->bpf);
+ }
+
+ /* this can fail as the data size we need could have changed */
+ if (!(adata = (gpointer) gst_adapter_map (scope->adapter, sbpf)))
+ break;
+
+ 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 (render) {
+ if (!render (scope, inbuf, outbuf)) {
+ ret = GST_FLOW_ERROR;
+ } else {
+ /* run various post processing (shading and geometri transformation */
+ if (scope->shader) {
+ scope->shader (scope, map.data, scope->pixelbuf);
+ }
+ }
+ }
+
+ gst_buffer_unmap (outbuf, &map);
+ gst_buffer_resize (outbuf, 0, scope->bpf);
+
+ g_mutex_unlock (&scope->config_lock);
+ ret = gst_pad_push (scope->srcpad, outbuf);
+ outbuf = NULL;
+ g_mutex_lock (&scope->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->adapter, sbpf);
+ gst_adapter_unmap (scope->adapter);
+ } else if (avail >= sbpf) {
+ /* just flush a bit and stop */
+ gst_adapter_flush (scope->adapter, (avail - sbpf));
+ gst_adapter_unmap (scope->adapter);
+ break;
+ }
+ avail = gst_adapter_available (scope->adapter);
+
+ if (ret != GST_FLOW_OK)
+ break;
+ }
+
+ g_mutex_unlock (&scope->config_lock);
+
+beach:
+ return ret;
+}
+
+static gboolean
+gst_base_audio_visualizer_src_event (GstPad * pad, GstObject * parent,
+ GstEvent * event)
+{
+ gboolean res;
+ GstBaseAudioVisualizer *scope;
+
+ scope = GST_BASE_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->proportion = proportion;
+ if (diff >= 0)
+ /* we're late, this is a good estimate for next displayable
+ * frame (see part-qos.txt) */
+ scope->earliest_time = timestamp + 2 * diff + scope->frame_duration;
+ else
+ scope->earliest_time = timestamp + diff;
+ GST_OBJECT_UNLOCK (scope);
+
+ res = gst_pad_push_event (scope->sinkpad, event);
+ break;
+ }
+ case GST_EVENT_RECONFIGURE:
+ /* dont't forward */
+ gst_event_unref (event);
+ res = TRUE;
+ break;
+ default:
+ res = gst_pad_push_event (scope->sinkpad, event);
+ break;
+ }
+
+ return res;
+}
+
+static gboolean
+gst_base_audio_visualizer_sink_event (GstPad * pad, GstObject * parent,
+ GstEvent * event)
+{
+ gboolean res;
+ GstBaseAudioVisualizer *scope;
+
+ scope = GST_BASE_AUDIO_VISUALIZER (parent);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_CAPS:
+ {
+ GstCaps *caps;
+
+ gst_event_parse_caps (event, &caps);
+ res = gst_base_audio_visualizer_sink_setcaps (scope, caps);
+ break;
+ }
+ case GST_EVENT_FLUSH_START:
+ res = gst_pad_push_event (scope->srcpad, event);
+ break;
+ case GST_EVENT_FLUSH_STOP:
+ gst_base_audio_visualizer_reset (scope);
+ res = gst_pad_push_event (scope->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->segment);
+
+ res = gst_pad_push_event (scope->srcpad, event);
+ break;
+ }
+ default:
+ res = gst_pad_push_event (scope->srcpad, event);
+ break;
+ }
+
+ return res;
+}
+
+static gboolean
+gst_base_audio_visualizer_src_query (GstPad * pad, GstObject * parent,
+ GstQuery * query)
+{
+ gboolean res = FALSE;
+ GstBaseAudioVisualizer *scope;
+
+ scope = GST_BASE_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->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->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 gboolean
+gst_base_audio_visualizer_sink_query (GstPad * pad, GstObject * parent,
+ GstQuery * query)
+{
+ gboolean res = FALSE;
+
+ switch (GST_QUERY_TYPE (query)) {
+ default:
+ res = gst_pad_query_default (pad, parent, query);
+ break;
+ }
+ return res;
+}
+
+static GstStateChangeReturn
+gst_base_audio_visualizer_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstStateChangeReturn ret;
+ GstBaseAudioVisualizer *scope;
+
+ scope = GST_BASE_AUDIO_VISUALIZER (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_base_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:
+ if (scope->pool) {
+ gst_buffer_pool_set_active (scope->pool, FALSE);
+ gst_object_replace ((GstObject **) & scope->pool, NULL);
+ }
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
/* amounf of samples before we can feed libvisual */
#define VISUAL_SAMPLES 512
-#define DEFAULT_WIDTH 320
-#define DEFAULT_HEIGHT 240
-#define DEFAULT_FPS_N 25
-#define DEFAULT_FPS_D 1
-
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, " "channels = (int) { 1, 2 }, "
+ "channel-mask = (bitmask) 0x3, "
#if defined(VISUAL_API_VERSION) && VISUAL_API_VERSION >= 4000 && VISUAL_API_VERSION < 5000
"rate = (int) { 8000, 11250, 22500, 32000, 44100, 48000, 96000 }"
#else
);
-
static void gst_visual_init (GstVisual * visual);
static void gst_visual_finalize (GObject * object);
-static GstStateChangeReturn gst_visual_change_state (GstElement * element,
- GstStateChange transition);
-static GstFlowReturn gst_visual_chain (GstPad * pad, GstObject * parent,
- GstBuffer * buffer);
-static gboolean gst_visual_sink_event (GstPad * pad, GstObject * parent,
- GstEvent * event);
-static gboolean gst_visual_src_event (GstPad * pad, GstObject * parent,
- GstEvent * event);
-
-static gboolean gst_visual_src_query (GstPad * pad, GstObject * parent,
- GstQuery * query);
-
-static gboolean gst_visual_sink_setcaps (GstPad * pad, GstCaps * caps);
-static GstCaps *gst_visual_getcaps (GstPad * pad, GstCaps * filter);
+static gboolean gst_visual_setup (GstBaseAudioVisualizer * bscope);
+static gboolean gst_visual_render (GstBaseAudioVisualizer * bscope,
+ GstBuffer * audio, GstBuffer * video);
static GstElementClass *parent_class = NULL;
(GInstanceInitFunc) gst_visual_init,
};
- type = g_type_register_static (GST_TYPE_ELEMENT, "GstVisual", &info, 0);
+ type =
+ g_type_register_static (GST_TYPE_BASE_AUDIO_VISUALIZER, "GstVisual",
+ &info, 0);
}
return type;
}
void
gst_visual_class_init (gpointer g_class, gpointer class_data)
{
- GstVisualClass *klass = GST_VISUAL_CLASS (g_class);
- GstElementClass *element = GST_ELEMENT_CLASS (g_class);
- GObjectClass *object = G_OBJECT_CLASS (g_class);
+ GObjectClass *gobject_class = (GObjectClass *) g_class;
+ GstElementClass *element_class = (GstElementClass *) g_class;
+ GstBaseAudioVisualizerClass *scope_class =
+ (GstBaseAudioVisualizerClass *) g_class;
+ GstVisualClass *klass = (GstVisualClass *) g_class;
klass->plugin = class_data;
- element->change_state = gst_visual_change_state;
-
if (class_data == NULL) {
parent_class = g_type_class_peek_parent (g_class);
} else {
klass->plugin->info->name, klass->plugin->info->version);
/* FIXME: improve to only register what plugin supports? */
- gst_element_class_add_pad_template (element,
+ gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
- gst_element_class_add_pad_template (element,
+ gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
- gst_element_class_set_static_metadata (element,
+ gst_element_class_set_static_metadata (element_class,
longname, "Visualization",
klass->plugin->info->about, "Benjamin Otte <otte@gnome.org>");
g_free (longname);
}
- object->finalize = gst_visual_finalize;
+ gobject_class->finalize = gst_visual_finalize;
+
+ scope_class->setup = GST_DEBUG_FUNCPTR (gst_visual_setup);
+ scope_class->render = GST_DEBUG_FUNCPTR (gst_visual_render);
}
static void
gst_visual_init (GstVisual * visual)
{
- /* create the sink and src pads */
- visual->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
- gst_pad_set_chain_function (visual->sinkpad, gst_visual_chain);
- gst_pad_set_event_function (visual->sinkpad, gst_visual_sink_event);
- gst_element_add_pad (GST_ELEMENT (visual), visual->sinkpad);
-
- visual->srcpad = gst_pad_new_from_static_template (&src_template, "src");
- gst_pad_set_event_function (visual->srcpad, gst_visual_src_event);
- gst_pad_set_query_function (visual->srcpad, gst_visual_src_query);
- gst_element_add_pad (GST_ELEMENT (visual), visual->srcpad);
-
- visual->adapter = gst_adapter_new ();
+ /* do nothing */
}
static void
{
GstVisual *visual = GST_VISUAL (object);
- g_object_unref (visual->adapter);
- if (visual->pool)
- gst_object_unref (visual->pool);
gst_visual_clear_actors (visual);
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
}
-static void
-gst_visual_reset (GstVisual * visual)
-{
- gst_adapter_clear (visual->adapter);
- gst_segment_init (&visual->segment, GST_FORMAT_UNDEFINED);
-
- GST_OBJECT_LOCK (visual);
- visual->proportion = 1.0;
- visual->earliest_time = -1;
- GST_OBJECT_UNLOCK (visual);
-}
-
-static GstCaps *
-gst_visual_getcaps (GstPad * pad, GstCaps * filter)
+static gboolean
+gst_visual_setup (GstBaseAudioVisualizer * bscope)
{
- GstCaps *ret;
- GstVisual *visual = GST_VISUAL (GST_PAD_PARENT (pad));
- int depths;
-
- if (!visual->actor) {
- ret = gst_pad_get_pad_template_caps (visual->srcpad);
- goto beach;
- }
-
- ret = gst_caps_new_empty ();
- depths = visual_actor_get_supported_depth (visual->actor);
- if (depths < 0) {
- /* FIXME: set an error */
- goto beach;
- }
- if (depths == VISUAL_VIDEO_DEPTH_GL) {
- /* We can't handle GL only plugins */
- goto beach;
- }
-
- GST_DEBUG_OBJECT (visual, "libvisual plugin supports depths %u (0x%04x)",
- depths, depths);
- /* if (depths & VISUAL_VIDEO_DEPTH_32BIT) Always supports 32bit output */
-#if G_BYTE_ORDER == G_BIG_ENDIAN
- gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("xRGB")));
-#else
- gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("BGRx")));
-#endif
+ GstVisual *visual = GST_VISUAL (bscope);
+ gint pitch, depth;
- if (depths & VISUAL_VIDEO_DEPTH_24BIT) {
-#if G_BYTE_ORDER == G_BIG_ENDIAN
- gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("RGB")));
-#else
- gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("BGR")));
-#endif
- }
- if (depths & VISUAL_VIDEO_DEPTH_16BIT) {
- gst_caps_append (ret, gst_caps_from_string (GST_VIDEO_CAPS_MAKE ("RGB16")));
- }
-
-beach:
-
- if (filter) {
- GstCaps *intersection;
+ gst_visual_clear_actors (visual);
- intersection =
- gst_caps_intersect_full (filter, ret, GST_CAPS_INTERSECT_FIRST);
- gst_caps_unref (ret);
- ret = intersection;
+ depth = bscope->vinfo.finfo->pixel_stride[0];
+ if (bscope->vinfo.finfo->bits >= 8) {
+ depth *= 8;
}
- GST_DEBUG_OBJECT (visual, "returning caps %" GST_PTR_FORMAT, ret);
+ 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;
- return ret;
-}
+ if (visual_actor_realize (visual->actor) != 0)
+ goto no_realize;
-static gboolean
-gst_visual_src_setcaps (GstVisual * visual, GstCaps * caps)
-{
- gboolean res;
- GstStructure *structure;
- gint depth, pitch, rate;
- const gchar *fmt;
-
- structure = gst_caps_get_structure (caps, 0);
-
- GST_DEBUG_OBJECT (visual, "src pad got caps %" GST_PTR_FORMAT, caps);
-
- if (!gst_structure_get_int (structure, "width", &visual->width))
- goto error;
- if (!gst_structure_get_int (structure, "height", &visual->height))
- goto error;
- if (!(fmt = gst_structure_get_string (structure, "format")))
- goto error;
- if (!gst_structure_get_fraction (structure, "framerate", &visual->fps_n,
- &visual->fps_d))
- goto error;
-
- if (!strcmp (fmt, "BGR") || !strcmp (fmt, "RGB"))
- depth = 24;
- else if (!strcmp (fmt, "BGRx") || !strcmp (fmt, "xRGB"))
- depth = 32;
- else
- depth = 16;
+ 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:
+no_actors:
{
- GST_ERROR_OBJECT (visual, "invalid caps received");
+ GST_ELEMENT_ERROR (visual, LIBRARY, INIT, (NULL),
+ ("could not create actors"));
+ gst_visual_clear_actors (visual);
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:
+no_realize:
{
- GST_ELEMENT_ERROR (visual, STREAM, FORMAT, (NULL),
- ("could not negotiate output format"));
- gst_caps_unref (target);
+ GST_ELEMENT_ERROR (visual, LIBRARY, INIT, (NULL),
+ ("could not realize actor"));
+ gst_visual_clear_actors (visual);
return FALSE;
}
}
static gboolean
-gst_visual_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
+gst_visual_render (GstBaseAudioVisualizer * bscope, GstBuffer * audio,
+ GstBuffer * video)
{
- 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;
- }
+ GstVisual *visual = GST_VISUAL (bscope);
+ GstMapInfo amap, vmap;
+ const guint16 *adata;
+ gint i, channels;
+ gboolean res = TRUE;
- 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;
+ gst_buffer_map (audio, &amap, GST_MAP_READ);
+ gst_buffer_map (video, &vmap, GST_MAP_WRITE);
- 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;
- }
- }
+ visual_video_set_buffer (visual->video, vmap.data);
- /* Read VISUAL_SAMPLES samples per channel */
- data =
- (const guint16 *) gst_adapter_map (visual->adapter,
- VISUAL_SAMPLES * bpf);
+ 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] = *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);
+ {
+ VisBuffer *lbuf, *rbuf;
+ guint16 ldata[VISUAL_SAMPLES], rdata[VISUAL_SAMPLES];
+ VisAudioSampleRateType vrate;
- visual_object_unref (VISUAL_OBJECT (lbuf));
- visual_object_unref (VISUAL_OBJECT (rbuf));
+ lbuf = visual_buffer_new_with_buffer (ldata, sizeof (ldata), NULL);
+ rbuf = visual_buffer_new_with_buffer (rdata, sizeof (rdata), NULL);
- }
-#else
- if (visual->channels == 2) {
+ if (channels == 2) {
for (i = 0; i < VISUAL_SAMPLES; i++) {
- visual->audio->plugpcm[0][i] = *data++;
- visual->audio->plugpcm[1][i] = *data++;
+ ldata[i] = *adata++;
+ rdata[i] = *adata++;
}
} else {
for (i = 0; i < VISUAL_SAMPLES; i++) {
- visual->audio->plugpcm[0][i] = *data;
- visual->audio->plugpcm[1][i] = *data++;
+ ldata[i] = *adata;
+ rdata[i] = *adata++;
}
}
-#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;
- }
+ /* 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;
}
- 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;
+ 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);
- skip:
- GST_DEBUG_OBJECT (visual, "finished frame, flushing %u samples from input",
- visual->spf);
+ visual_object_unref (VISUAL_OBJECT (lbuf));
+ visual_object_unref (VISUAL_OBJECT (rbuf));
- /* 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;
+#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
- return ret;
+ visual_audio_analyze (visual->audio);
+ visual_actor_run (visual->actor, visual->audio);
+ visual_video_set_buffer (visual->video, NULL);
- /* ERRORS */
-no_actors:
- {
- GST_ELEMENT_ERROR (visual, LIBRARY, INIT, (NULL),
- ("could not create actors"));
- gst_visual_clear_actors (visual);
- return GST_STATE_CHANGE_FAILURE;
- }
-no_realize:
- {
- GST_ELEMENT_ERROR (visual, LIBRARY, INIT, (NULL),
- ("could not realize actor"));
- gst_visual_clear_actors (visual);
- return GST_STATE_CHANGE_FAILURE;
- }
+ GST_DEBUG_OBJECT (visual, "rendered one frame");
+done:
+ gst_buffer_unmap (video, &vmap);
+ gst_buffer_unmap (audio, &amap);
+ return res;
}
projects / platform / upstream / gst-plugins-base.git / commitdiff