gst/audiorate/: Added an audiorate converter that fills in gaps.

Original commit message from CVS:
* gst/audiorate/Makefile.am:
* gst/audiorate/gstaudiorate.c: (gst_audiorate_get_type),
(gst_audiorate_base_init), (gst_audiorate_class_init),
(gst_audiorate_link), (gst_audiorate_init), (gst_audiorate_chain),
(gst_audiorate_set_property), (gst_audiorate_get_property),
(gst_audiorate_change_state), (plugin_init):
Added an audiorate converter that fills in gaps.

4 files changed, 377 insertions, 0 deletions

diff --git a/ChangeLog b/ChangeLog
index 48e8b2e5f..4189da6b9 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,13 @@
+2004-06-17  Wim Taymans  <wim@fluendo.com>
+
+	* gst/audiorate/Makefile.am:
+	* gst/audiorate/gstaudiorate.c: (gst_audiorate_get_type),
+	(gst_audiorate_base_init), (gst_audiorate_class_init),
+	(gst_audiorate_link), (gst_audiorate_init), (gst_audiorate_chain),
+	(gst_audiorate_set_property), (gst_audiorate_get_property),
+	(gst_audiorate_change_state), (plugin_init):
+	Added an audiorate converter that fills in gaps.
+
 2004-06-17  Johan Dahlin  <johan@gnome.org>
 
 	* ext/tcp/*: Revert Zaheer changes.
diff --git a/configure.ac b/configure.ac
index b9f4c6560..05744b98c 100644
--- a/configure.ac
+++ b/configure.ac
@@ -340,6 +340,7 @@ GST_PLUGINS_ALL="\
 	asfdemux \
 	audioconvert \
 	audioscale \
+	audiorate \
 	auparse \
 	avi \
 	cdxaparse \
@@ -395,6 +396,7 @@ GST_PLUGINS_ALL="\
 	videoflip \
 	videofilter \
 	videomixer \
+	videorate \
 	videoscale \
 	videotestsrc \
 	volenv \
@@ -1781,6 +1783,7 @@ gst/adder/Makefile
 gst/alpha/Makefile
 gst/audioconvert/Makefile
 gst/audioscale/Makefile
+gst/audiorate/Makefile
 gst/auparse/Makefile
 gst/avi/Makefile
 gst/asfdemux/Makefile
@@ -1840,6 +1843,7 @@ gst/videofilter/Makefile
 gst/videoflip/Makefile
 gst/videomixer/Makefile
 gst/videoscale/Makefile
+gst/videorate/Makefile
 gst/videotestsrc/Makefile
 gst/volenv/Makefile
 gst/volume/Makefile
diff --git a/gst/audiorate/Makefile.am b/gst/audiorate/Makefile.am
new file mode 100644
index 000000000..fa0e736a3
--- /dev/null
+++ b/gst/audiorate/Makefile.am
@@ -0,0 +1,7 @@
+
+plugin_LTLIBRARIES = libgstaudiorate.la
+
+libgstaudiorate_la_SOURCES = gstaudiorate.c
+libgstaudiorate_la_CFLAGS = $(GST_CFLAGS)
+libgstaudiorate_la_LIBADD =
+libgstaudiorate_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
diff --git a/gst/audiorate/gstaudiorate.c b/gst/audiorate/gstaudiorate.c
new file mode 100644
index 000000000..1ff71b6d8
--- /dev/null
+++ b/gst/audiorate/gstaudiorate.c
@@ -0,0 +1,356 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#include <string.h>
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+
+#define GST_TYPE_AUDIORATE \
+  (gst_audiorate_get_type())
+#define GST_AUDIORATE(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORATE,GstAudiorate))
+#define GST_AUDIORATE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORATE,GstAudiorate))
+#define GST_IS_AUDIORATE(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORATE))
+#define GST_IS_AUDIORATE_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORATE))
+
+typedef struct _GstAudiorate GstAudiorate;
+typedef struct _GstAudiorateClass GstAudiorateClass;
+
+struct _GstAudiorate
+{
+  GstElement element;
+
+  GstPad *sinkpad, *srcpad;
+
+  /* audio state */
+  guint64 next_offset;
+
+  guint64 in, out, add, drop;
+};
+
+struct _GstAudiorateClass
+{
+  GstElementClass parent_class;
+};
+
+/* elementfactory information */
+static GstElementDetails audiorate_details =
+GST_ELEMENT_DETAILS ("Audio rate adjuster",
+    "Filter/Effect/Audio",
+    "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream",
+    "Wim Taymans <wim@fluendo.com>");
+
+/* GstAudiorate signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  ARG_0,
+  ARG_IN,
+  ARG_OUT,
+  ARG_ADD,
+  ARG_DROP,
+  /* FILL ME */
+};
+
+static GstStaticPadTemplate gst_audiorate_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
+    );
+
+static GstStaticPadTemplate gst_audiorate_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS)
+    );
+
+static void gst_audiorate_base_init (gpointer g_class);
+static void gst_audiorate_class_init (GstAudiorateClass * klass);
+static void gst_audiorate_init (GstAudiorate * audiorate);
+static void gst_audiorate_chain (GstPad * pad, GstData * _data);
+
+static void gst_audiorate_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audiorate_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static GstElementStateReturn gst_audiorate_change_state (GstElement * element);
+
+static GstElementClass *parent_class = NULL;
+
+/*static guint gst_audiorate_signals[LAST_SIGNAL] = { 0 }; */
+
+static GType
+gst_audiorate_get_type (void)
+{
+  static GType audiorate_type = 0;
+
+  if (!audiorate_type) {
+    static const GTypeInfo audiorate_info = {
+      sizeof (GstAudiorateClass),
+      gst_audiorate_base_init,
+      NULL,
+      (GClassInitFunc) gst_audiorate_class_init,
+      NULL,
+      NULL,
+      sizeof (GstAudiorate),
+      0,
+      (GInstanceInitFunc) gst_audiorate_init,
+    };
+
+    audiorate_type = g_type_register_static (GST_TYPE_ELEMENT,
+        "GstAudiorate", &audiorate_info, 0);
+  }
+
+  return audiorate_type;
+}
+
+static void
+gst_audiorate_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  gst_element_class_set_details (element_class, &audiorate_details);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_audiorate_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_audiorate_src_template));
+}
+static void
+gst_audiorate_class_init (GstAudiorateClass * klass)
+{
+  GObjectClass *object_class = G_OBJECT_CLASS (klass);
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  g_object_class_install_property (object_class, ARG_IN,
+      g_param_spec_uint64 ("in", "In",
+          "Number of input samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
+  g_object_class_install_property (object_class, ARG_OUT,
+      g_param_spec_uint64 ("out", "Out",
+          "Number of output samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
+  g_object_class_install_property (object_class, ARG_ADD,
+      g_param_spec_uint64 ("duplicate", "Duplicate",
+          "Number of added samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
+  g_object_class_install_property (object_class, ARG_DROP,
+      g_param_spec_uint64 ("drop", "Drop",
+          "Number of dropped samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE));
+
+  object_class->set_property = gst_audiorate_set_property;
+  object_class->get_property = gst_audiorate_get_property;
+
+  element_class->change_state = gst_audiorate_change_state;
+}
+
+static GstPadLinkReturn
+gst_audiorate_link (GstPad * pad, const GstCaps * caps)
+{
+  GstAudiorate *audiorate;
+  GstStructure *structure;
+  GstPad *otherpad;
+  GstPadLinkReturn res;
+
+  audiorate = GST_AUDIORATE (gst_pad_get_parent (pad));
+
+  otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
+      audiorate->srcpad;
+
+  res = gst_pad_try_set_caps (otherpad, caps);
+  if (GST_PAD_LINK_FAILED (res))
+    return res;
+
+  structure = gst_caps_get_structure (caps, 0);
+
+  return GST_PAD_LINK_OK;
+}
+
+static void
+gst_audiorate_init (GstAudiorate * audiorate)
+{
+  GST_FLAG_SET (audiorate, GST_ELEMENT_EVENT_AWARE);
+
+  GST_DEBUG ("gst_audiorate_init");
+  audiorate->sinkpad =
+      gst_pad_new_from_template (gst_static_pad_template_get
+      (&gst_audiorate_sink_template), "sink");
+  gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
+  gst_pad_set_chain_function (audiorate->sinkpad, gst_audiorate_chain);
+  gst_pad_set_link_function (audiorate->sinkpad, gst_audiorate_link);
+  gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
+
+  audiorate->srcpad =
+      gst_pad_new_from_template (gst_static_pad_template_get
+      (&gst_audiorate_src_template), "src");
+  gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
+  gst_pad_set_link_function (audiorate->srcpad, gst_audiorate_link);
+  gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
+
+  audiorate->in = 0;
+  audiorate->out = 0;
+  audiorate->drop = 0;
+  audiorate->add = 0;
+}
+
+static void
+gst_audiorate_chain (GstPad * pad, GstData * data)
+{
+  GstAudiorate *audiorate;
+  GstBuffer *buf;
+  GstClockTime in_time, in_duration;
+  guint64 in_offset, in_offset_end;
+  gint in_size;
+
+  audiorate = GST_AUDIORATE (gst_pad_get_parent (pad));
+
+  if (GST_IS_EVENT (data)) {
+    GstEvent *event = GST_EVENT (data);
+
+    gst_pad_event_default (pad, event);
+    return;
+  }
+
+  audiorate->in++;
+
+  buf = GST_BUFFER (data);
+  in_time = GST_BUFFER_TIMESTAMP (buf);
+  in_duration = GST_BUFFER_DURATION (buf);
+  in_size = GST_BUFFER_SIZE (buf);
+  in_offset = GST_BUFFER_OFFSET (buf);
+  in_offset_end = GST_BUFFER_OFFSET_END (buf);
+
+  /* do we need to insert samples */
+  if (in_offset > audiorate->next_offset) {
+    GstBuffer *fill;
+    gint bytes_per_sample, fillsize;
+    guint64 fillsamples;
+
+    /* FIXME: use caps to get this */
+    bytes_per_sample = in_size / (in_offset_end - in_offset);
+
+    fillsamples = in_offset - audiorate->next_offset;
+    fillsize = fillsamples * bytes_per_sample;
+
+    fill = gst_buffer_new_and_alloc (fillsize);
+    memset (GST_BUFFER_DATA (fill), 0, fillsize);
+
+    GST_BUFFER_DURATION (fill) = in_duration * fillsize / in_size;
+    GST_BUFFER_TIMESTAMP (fill) = in_time - GST_BUFFER_DURATION (fill);
+    GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
+    GST_BUFFER_OFFSET_END (fill) = in_offset;
+
+    gst_pad_push (audiorate->srcpad, GST_DATA (fill));
+    audiorate->out++;
+    audiorate->add++;
+  } else if (in_offset < audiorate->next_offset) {
+    g_warning ("overlapping samples, implement me");
+    audiorate->drop++;
+  }
+  gst_pad_push (audiorate->srcpad, GST_DATA (buf));
+  audiorate->out++;
+
+  audiorate->next_offset = in_offset_end;
+}
+
+static void
+gst_audiorate_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec)
+{
+  //GstAudiorate *audiorate = GST_AUDIORATE (object);
+
+  switch (prop_id) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audiorate_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec)
+{
+  GstAudiorate *audiorate = GST_AUDIORATE (object);
+
+  switch (prop_id) {
+    case ARG_IN:
+      g_value_set_uint64 (value, audiorate->in);
+      break;
+    case ARG_OUT:
+      g_value_set_uint64 (value, audiorate->out);
+      break;
+    case ARG_ADD:
+      g_value_set_uint64 (value, audiorate->add);
+      break;
+    case ARG_DROP:
+      g_value_set_uint64 (value, audiorate->drop);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstElementStateReturn
+gst_audiorate_change_state (GstElement * element)
+{
+  GstAudiorate *audiorate = GST_AUDIORATE (element);
+
+  switch (GST_STATE_TRANSITION (element)) {
+    case GST_STATE_PAUSED_TO_READY:
+      break;
+    case GST_STATE_READY_TO_PAUSED:
+      audiorate->next_offset = 0;
+      break;
+    default:
+      break;
+  }
+
+  if (parent_class->change_state)
+    return parent_class->change_state (element);
+
+  return GST_STATE_SUCCESS;
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
+      GST_TYPE_AUDIORATE);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "audiorate",
+    "Adjusts audio frames",
+    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE, GST_ORIGIN)