gstreamer/gst/audiorate/gstaudiorate.c
Stefan Kost 377e2be9f3 make more debug catagories static
Original commit message from CVS:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* gst/audiorate/gstaudiorate.c:
make more debug catagories static
* tests/check/Makefile.am:
* tests/check/elements/adder.c: (message_received),
(test_event_message_received), (GST_START_TEST),
(test_play_twice_message_received), (adder_suite):
added test case for using element twice, extra bonus points for anyone
who can make these test run reliably
2006-05-23 20:38:56 +00:00

441 lines
13 KiB
C

/* 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_CAT_DEFAULT audio_rate_debug
GST_DEBUG_CATEGORY_STATIC (audio_rate_debug);
#define GST_TYPE_AUDIO_RATE \
(gst_audio_rate_get_type())
#define GST_AUDIO_RATE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_RATE,GstAudioRate))
#define GST_AUDIO_RATE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_RATE,GstAudioRate))
#define GST_IS_AUDIO_RATE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_RATE))
#define GST_IS_AUDIO_RATE_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_RATE))
typedef struct _GstAudioRate GstAudioRate;
typedef struct _GstAudioRateClass GstAudioRateClass;
struct _GstAudioRate
{
GstElement element;
GstPad *sinkpad, *srcpad;
gint bytes_per_sample;
/* audio state */
guint64 offset;
guint64 next_offset;
guint64 in, out, add, drop;
gboolean silent;
};
struct _GstAudioRateClass
{
GstElementClass parent_class;
};
/* elementfactory information */
static const GstElementDetails audio_rate_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
};
#define DEFAULT_SILENT TRUE
enum
{
ARG_0,
ARG_IN,
ARG_OUT,
ARG_ADD,
ARG_DROP,
ARG_SILENT,
/* FILL ME */
};
static GstStaticPadTemplate gst_audio_rate_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
);
static GstStaticPadTemplate gst_audio_rate_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS ";"
GST_AUDIO_FLOAT_PAD_TEMPLATE_CAPS)
);
static void gst_audio_rate_base_init (gpointer g_class);
static void gst_audio_rate_class_init (GstAudioRateClass * klass);
static void gst_audio_rate_init (GstAudioRate * audiorate);
static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstBuffer * buf);
static void gst_audio_rate_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_rate_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_audio_rate_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
/*static guint gst_audio_rate_signals[LAST_SIGNAL] = { 0 }; */
static GType
gst_audio_rate_get_type (void)
{
static GType audio_rate_type = 0;
if (!audio_rate_type) {
static const GTypeInfo audio_rate_info = {
sizeof (GstAudioRateClass),
gst_audio_rate_base_init,
NULL,
(GClassInitFunc) gst_audio_rate_class_init,
NULL,
NULL,
sizeof (GstAudioRate),
0,
(GInstanceInitFunc) gst_audio_rate_init,
};
audio_rate_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstAudioRate", &audio_rate_info, 0);
}
return audio_rate_type;
}
static void
gst_audio_rate_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &audio_rate_details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_rate_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_audio_rate_src_template));
}
static void
gst_audio_rate_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);
object_class->set_property = gst_audio_rate_set_property;
object_class->get_property = gst_audio_rate_get_property;
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 ("add", "Add",
"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));
g_object_class_install_property (object_class, ARG_SILENT,
g_param_spec_boolean ("silent", "silent",
"Don't emit notify for dropped and duplicated frames",
DEFAULT_SILENT, G_PARAM_READWRITE));
element_class->change_state = gst_audio_rate_change_state;
}
static gboolean
gst_audio_rate_setcaps (GstPad * pad, GstCaps * caps)
{
GstAudioRate *audiorate;
GstStructure *structure;
GstPad *otherpad;
gboolean ret = FALSE;
gint channels, width;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
audiorate->srcpad;
if (!gst_pad_set_caps (otherpad, caps))
goto beach;
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "channels", &channels) ||
!gst_structure_get_int (structure, "width", &width)) {
goto beach;
}
audiorate->bytes_per_sample = channels * (width / 8);
if (audiorate->bytes_per_sample == 0)
audiorate->bytes_per_sample = 1;
ret = TRUE;
beach:
gst_object_unref (audiorate);
return ret;
}
static void
gst_audio_rate_init (GstAudioRate * audiorate)
{
audiorate->sinkpad =
gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink");
gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain);
gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audio_rate_setcaps);
gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
audiorate->srcpad =
gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src");
gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
gst_pad_set_setcaps_function (audiorate->srcpad, gst_audio_rate_setcaps);
gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
audiorate->bytes_per_sample = 1;
audiorate->in = 0;
audiorate->out = 0;
audiorate->drop = 0;
audiorate->add = 0;
audiorate->silent = DEFAULT_SILENT;
}
static GstFlowReturn
gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
{
GstAudioRate *audiorate;
GstClockTime in_time, in_duration;
guint64 in_offset, in_offset_end;
guint in_size;
GstFlowReturn ret = GST_FLOW_OK;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
audiorate->in++;
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);
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT
", in_size:%u, in_offset:%lld, in_offset_end:%lld" ", ->next_offset:%lld",
GST_TIME_ARGS (in_time), GST_TIME_ARGS (in_duration), in_size, in_offset,
in_offset_end, audiorate->next_offset);
if (in_offset == GST_CLOCK_TIME_NONE || in_offset_end == GST_CLOCK_TIME_NONE) {
GST_WARNING_OBJECT (audiorate, "audiorate got buffer without offsets");
in_offset = audiorate->offset;
in_offset_end = audiorate->offset + in_size / audiorate->bytes_per_sample;
GST_WARNING_OBJECT (audiorate, "in_offset:%lld, in_offset_end:%lld",
in_offset, in_offset_end);
}
/* do we need to insert samples */
if (in_offset > audiorate->next_offset) {
GstBuffer *fill;
gint fillsize;
guint64 fillsamples;
fillsamples = in_offset - audiorate->next_offset;
fillsize = fillsamples * audiorate->bytes_per_sample;
fill = gst_buffer_new_and_alloc (fillsize);
memset (GST_BUFFER_DATA (fill), 0, fillsize);
GST_LOG_OBJECT (audiorate, "inserting %lld samples", fillsamples);
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;
if ((ret = gst_pad_push (audiorate->srcpad, fill) != GST_FLOW_OK))
goto beach;
audiorate->out++;
audiorate->add += fillsamples;
if (!audiorate->silent)
g_object_notify (G_OBJECT (audiorate), "add");
} else if (in_offset < audiorate->next_offset) {
/* need to remove samples */
if (in_offset_end <= audiorate->next_offset) {
guint64 drop = in_size / audiorate->bytes_per_sample;
audiorate->drop += drop;
GST_LOG_OBJECT (audiorate, "dropping %lld samples", drop);
/* we can drop the buffer completely */
gst_buffer_unref (buf);
if (!audiorate->silent)
g_object_notify (G_OBJECT (audiorate), "drop");
goto beach;
} else {
guint64 truncsamples;
guint truncsize, leftsize;
GstBuffer *trunc;
/* truncate buffer */
truncsamples = audiorate->next_offset - in_offset;
truncsize = truncsamples * audiorate->bytes_per_sample;
leftsize = in_size - truncsize;
trunc = gst_buffer_create_sub (buf, truncsize, leftsize);
GST_BUFFER_DURATION (trunc) = in_duration * leftsize / in_size;
GST_BUFFER_TIMESTAMP (trunc) =
in_time + in_duration - GST_BUFFER_DURATION (trunc);
GST_BUFFER_OFFSET (trunc) = audiorate->next_offset;
GST_BUFFER_OFFSET_END (trunc) = in_offset_end;
GST_LOG_OBJECT (audiorate, "truncating %lld samples", truncsamples);
gst_buffer_unref (buf);
buf = trunc;
audiorate->drop += truncsamples;
}
}
ret = gst_pad_push (audiorate->srcpad, buf);
audiorate->out++;
audiorate->next_offset = in_offset_end;
beach:
audiorate->offset += in_size / audiorate->bytes_per_sample;
return ret;
}
static void
gst_audio_rate_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstAudioRate *audiorate = GST_AUDIO_RATE (object);
switch (prop_id) {
case ARG_SILENT:
audiorate->silent = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_rate_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstAudioRate *audiorate = GST_AUDIO_RATE (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;
case ARG_SILENT:
g_value_set_boolean (value, audiorate->silent);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_audio_rate_change_state (GstElement * element, GstStateChange transition)
{
GstAudioRate *audiorate = GST_AUDIO_RATE (element);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
audiorate->offset = 0;
audiorate->next_offset = 0;
break;
default:
break;
}
if (parent_class->change_state)
return parent_class->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0,
"AudioRate stream fixer");
return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
GST_TYPE_AUDIO_RATE);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audiorate",
"Adjusts audio frames",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)