Imported Upstream version 0.11.93
diff --git a/gst/audiovisualizers/gstaudiovisualizer.c b/gst/audiovisualizers/gstaudiovisualizer.c
new file mode 100644
index 0000000..ac56a9f
--- /dev/null
+++ b/gst/audiovisualizers/gstaudiovisualizer.c
@@ -0,0 +1,1134 @@
+/* GStreamer
+ * Copyright (C) <2011> Stefan Kost <ensonic@users.sf.net>
+ *
+ * gstaudiovisualizer.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:gstaudiovisualizer
+ *
+ * A baseclass 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 "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 gboolean gst_audio_visualizer_sink_query (GstPad * pad,
+    GstObject * parent, GstQuery * query);
+
+static GstStateChangeReturn gst_audio_visualizer_change_state (GstElement *
+    element, GstStateChange transition);
+
+/* 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-BadGstAudioVisualizers", 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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 (GstAudioVisualizer * 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_audio_visualizer_change_shader (GstAudioVisualizer * scope)
+{
+  switch (scope->shader_type) {
+    case GST_AUDIO_VISUALIZER_SHADER_NONE:
+      scope->shader = NULL;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE:
+      scope->shader = shader_fade;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_UP:
+      scope->shader = shader_fade_and_move_up;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_DOWN:
+      scope->shader = shader_fade_and_move_down;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_LEFT:
+      scope->shader = shader_fade_and_move_left;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_RIGHT:
+      scope->shader = shader_fade_and_move_right;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_OUT:
+      scope->shader = shader_fade_and_move_horiz_out;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_HORIZ_IN:
+      scope->shader = shader_fade_and_move_horiz_in;
+      break;
+    case GST_AUDIO_VISUALIZER_SHADER_FADE_AND_MOVE_VERT_OUT:
+      scope->shader = shader_fade_and_move_vert_out;
+      break;
+    case GST_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_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-BadGstAudioVisualizers", &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;
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  GST_DEBUG_CATEGORY_INIT (audio_visualizer_debug, "baseaudiovisualizer",
+      0, "scope audio visualisation base 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);
+
+  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;
+
+  /* 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_audio_visualizer_chain));
+  gst_pad_set_event_function (scope->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_audio_visualizer_sink_event));
+  gst_pad_set_query_function (scope->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_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_audio_visualizer_src_event));
+  gst_pad_set_query_function (scope->srcpad,
+      GST_DEBUG_FUNCPTR (gst_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_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_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->shader_type = g_value_get_enum (value);
+      gst_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_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->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_audio_visualizer_dispose (GObject * object)
+{
+  GstAudioVisualizer *scope = GST_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_audio_visualizer_reset (GstAudioVisualizer * 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_audio_visualizer_sink_setcaps (GstAudioVisualizer * 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_audio_visualizer_src_setcaps (GstAudioVisualizer * scope, GstCaps * caps)
+{
+  GstVideoInfo info;
+  GstAudioVisualizerClass *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_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_audio_visualizer_src_negotiate (GstAudioVisualizer * 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);
+  target = gst_caps_fixate (target);
+
+  GST_DEBUG_OBJECT (scope, "final caps are %" GST_PTR_FORMAT, target);
+
+  gst_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_audio_visualizer_ensure_negotiated (GstAudioVisualizer * 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_audio_visualizer_src_negotiate (scope))
+      return GST_FLOW_NOT_NEGOTIATED;
+  }
+  return GST_FLOW_OK;
+}
+
+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;
+  gboolean (*render) (GstAudioVisualizer * scope, GstBuffer * audio,
+      GstBuffer * video);
+  gint bps, channels, rate;
+
+  scope = GST_AUDIO_VISUALIZER (parent);
+  klass = GST_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_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_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, &timestamp);
+
+      /* 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_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);
+      break;
+    }
+    case GST_EVENT_FLUSH_START:
+      res = gst_pad_push_event (scope->srcpad, event);
+      break;
+    case GST_EVENT_FLUSH_STOP:
+      gst_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_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->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_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_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:
+      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;
+}