gstreamer/gst/audioresample/gstaudioresample.c
Thomas Vander Stichele 752a59192c port audioresample to basetransform
Original commit message from CVS:
port audioresample to basetransform
2005-08-24 14:08:58 +00:00

456 lines
14 KiB
C

/* GStreamer
* Copyright (C) 1999 Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 David A. Schleef <ds@schleef.org>
*
* 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.
*/
/* Element-Checklist-Version: 5 */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
/*#define DEBUG_ENABLED */
#include "gstaudioresample.h"
#include <gst/audio/audio.h>
#include <gst/base/gstbasetransform.h>
GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
#define GST_CAT_DEFAULT audioresample_debug
/* elementfactory information */
static GstElementDetails gst_audioresample_details =
GST_ELEMENT_DETAILS ("Audio scaler",
"Filter/Converter/Audio",
"Resample audio",
"David Schleef <ds@schleef.org>");
/* GstAudioresample signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_FILTERLEN
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS (\
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (boolean) true")
#if 0
/* disabled because it segfaults */
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) 32")
#endif
static GstStaticPadTemplate gst_audioresample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static GstStaticPadTemplate gst_audioresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_audioresample_base_init (gpointer g_class);
static void gst_audioresample_class_init (GstAudioresampleClass * klass);
static void gst_audioresample_init (GstAudioresample * audioresample);
static void gst_audioresample_dispose (GObject * object);
static void gst_audioresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audioresample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
/* vmethods */
gboolean audioresample_get_unit_size (GstBaseTransform * base,
GstCaps * caps, guint * size);
GstCaps *audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps);
gboolean audioresample_transform_size (GstBaseTransform * trans,
GstPadDirection direction, GstCaps * incaps, guint insize,
GstCaps * outcaps, guint * outsize);
gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps);
static GstFlowReturn audioresample_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
static void gst_audioresample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_sink_template));
gst_element_class_set_details (gstelement_class,
&gst_audioresample_details);
}
static void gst_audioresample_class_init (GstAudioresampleClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_audioresample_set_property;
gobject_class->get_property = gst_audioresample_get_property;
gobject_class->dispose = gst_audioresample_dispose;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
GST_DEBUG_FUNCPTR (audioresample_transform_size);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
GST_DEBUG_FUNCPTR (audioresample_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
GST_DEBUG_FUNCPTR (audioresample_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
GST_DEBUG_FUNCPTR (audioresample_transform);
}
static void gst_audioresample_init (GstAudioresample * audioresample)
{
ResampleState *r;
r = resample_new ();
audioresample->resample = r;
resample_set_filter_length (r, 64);
resample_set_format (r, RESAMPLE_FORMAT_S16);
}
static void gst_audioresample_dispose (GObject * object)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (object);
if (audioresample->resample) {
resample_free (audioresample->resample);
audioresample->resample = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
/* vmethods */
gboolean
audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size) {
gint width, channels;
GstStructure *structure;
gboolean ret;
g_return_val_if_fail (size, FALSE);
/* this works for both float and int */
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "channels", &channels);
g_return_val_if_fail (ret, FALSE);
*size = width * channels / 8;
return TRUE;
}
GstCaps *audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
GstCaps *temp, *res;
const GstCaps *templcaps;
GstStructure *structure;
temp = gst_caps_copy (caps);
structure = gst_caps_get_structure (temp, 0);
gst_structure_remove_field (structure, "rate");
templcaps = gst_pad_get_pad_template_caps (base->srcpad);
res = gst_caps_intersect (templcaps, temp);
gst_caps_unref (temp);
return res;
}
static gboolean
resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
{
GstStructure *structure;
gboolean ret;
gint myinrate, myoutrate;
int mychannels;
GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
structure = gst_caps_get_structure (incaps, 0);
/* FIXME: once it does float, set the correct format */
#if 0
if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
r->format = GST_RESAMPLE_FLOAT;
} else {
r->format = GST_RESAMPLE_S16;
}
#endif
ret = gst_structure_get_int (structure, "rate", &myinrate);
ret &= gst_structure_get_int (structure, "channels", &mychannels);
g_return_val_if_fail (ret, FALSE);
structure = gst_caps_get_structure (outcaps, 0);
ret = gst_structure_get_int (structure, "rate", &myoutrate);
g_return_val_if_fail (ret, FALSE);
if (channels)
*channels = mychannels;
if (inrate)
*inrate = myinrate;
if (outrate)
*outrate = myoutrate;
resample_set_n_channels (state, mychannels);
resample_set_input_rate (state, myinrate);
resample_set_output_rate (state, myoutrate);
return TRUE;
}
gboolean audioresample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
ResampleState *state;
GstCaps *srccaps, *sinkcaps;
gboolean use_internal = FALSE; /* whether we use the internal state */
gboolean ret = TRUE;
/* FIXME: make sure incaps/outcaps get renamed to caps/othercaps, since
* interpretation depends on the direction */
if (direction == GST_PAD_SINK) {
sinkcaps = caps;
srccaps = othercaps;
} else {
sinkcaps = othercaps;
srccaps = caps;
}
/* if the caps are the ones that _set_caps got called with; we can use
* our own state; otherwise we'll have to create a state */
if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&
gst_caps_is_equal (srccaps, audioresample->srccaps)) {
use_internal = TRUE;
state = audioresample->resample;
} else {
state = resample_new ();
resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
}
/* we can use our own state to answer the question */
if (direction == GST_PAD_SINK) {
/* asked to convert size of an incoming buffer */
*othersize = resample_get_output_size_for_input (state, size);
} else {
/* take a best guess, this is called cheating */
*othersize = floor (size * state->i_rate / state->o_rate);
}
if (!use_internal) {
resample_free (state);
}
return ret;
}
gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
gboolean ret;
gint inrate, outrate;
int channels;
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,
&channels, &inrate, &outrate);
g_return_val_if_fail (ret, FALSE);
audioresample->channels = channels;
GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);
audioresample->i_rate = inrate;
GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);
audioresample->o_rate = outrate;
GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);
/* save caps so we can short-circuit in the size_transform if the caps
* are the same */
/* FIXME: clean them up in state change ? */
gst_caps_ref (incaps);
gst_caps_replace (&audioresample->sinkcaps, incaps);
gst_caps_ref (outcaps);
gst_caps_replace (&audioresample->srccaps, outcaps);
return TRUE;
}
static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
/* FIXME: this-> */
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
ResampleState *r;
guchar *data;
gulong size;
int outsize;
/* FIXME: move to _inplace */
#if 0
if (audioresample->passthru) {
gst_pad_push (audioresample->srcpad, GST_DATA (buf));
return;
}
#endif
r = audioresample->resample;
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
resample_add_input_data (r, data, size, NULL, NULL);
outsize = resample_get_output_size (r);
if (outsize != GST_BUFFER_SIZE (outbuf)) {
GST_WARNING_OBJECT (audioresample,
"overriding audioresample's outsize %d with outbuffer's size %d",
outsize, GST_BUFFER_SIZE (outbuf));
outsize = GST_BUFFER_SIZE (outbuf);
}
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
if (outsize != GST_BUFFER_SIZE (outbuf)) {
GST_WARNING_OBJECT (audioresample,
"audioresample, you bastard ! you only gave me %d bytes, not %d",
outsize, GST_BUFFER_SIZE (outbuf));
/* if the size we get is smaller than the buffer, it's still fine; we
* just waste a bit of space on the end */
if (outsize < GST_BUFFER_SIZE (outbuf)) {
GST_BUFFER_SIZE (outbuf) = outsize;
return GST_FLOW_OK;
} else {
/* this is an error that needs fixing in the resample library; we told
* it we wanted only GST_BUFFER_SIZE (outbuf), and it gave us more ! */
return GST_FLOW_ERROR;
}
}
return GST_FLOW_OK;
}
static void
gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
audioresample->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d\n",
audioresample->filter_length);
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
break;
default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audioresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
g_value_set_int (value, audioresample->filter_length);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean plugin_init (GstPlugin * plugin)
{
resample_init ();
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
GST_TYPE_AUDIORESAMPLE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audioresample",
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN);