--- /dev/null
+/* 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)
projects / platform / upstream / gst-plugins-base.git / commitdiff