mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-03 05:05:52 +00:00
Code Issues 1 Projects Releases Wiki Activity

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

Browse source 
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.
This commit is contained in:
Wim Taymans 2004-06-17 13:45:50 +00:00
parent c163b09cc4
commit 7face56b42
4 changed files with 377 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

10
ChangeLog
View file

@ -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.

4
configure.ac
View file

@ -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

7
gst/audiorate/Makefile.am Normal file
View file

@ -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)

356
gst/audiorate/gstaudiorate.c Normal file
View file

@ -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)