port audioresample to basetransform

Original commit message from CVS:
port audioresample to basetransform
parent 41a43b86
2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
* configure.ac:
compile audioresample
* gst/audioresample/Makefile.am:
* gst/audioresample/buffer.c:
* gst/audioresample/functable.c:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
(resample_get_output_size_for_input):
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
port to use basetransform; doesn't work in all cases yet
2005-08-24 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/audioconvert/gstaudioconvert.c:
......
......@@ -369,8 +369,8 @@ dnl these are all the gst plug-ins, compilable without additional libs
GST_PLUGINS_ALL="\
adder \
audioconvert \
audioscale \
audiorate \
audioresample \
ffmpegcolorspace \
playback \
sine \
......@@ -892,6 +892,7 @@ gst/adder/Makefile
gst/audioconvert/Makefile
gst/audioscale/Makefile
gst/audiorate/Makefile
gst/audioresample/Makefile
gst/ffmpegcolorspace/Makefile
gst/playback/Makefile
gst/sine/Makefile
......
......@@ -15,7 +15,7 @@ resample_SOURCES = \
buffer.h
libgstaudioresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(LIBOIL_CFLAGS)
libgstaudioresample_la_LIBADD = $(LIBOIL_LIBS)
libgstaudioresample_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS)
libgstaudioresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS)
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
......@@ -3,10 +3,11 @@
#include "config.h"
#endif
#include <audioresample/buffer.h>
#include <glib.h>
#include <string.h>
#include <audioresample/debug.h>
#include "buffer.h"
#include "debug.h"
static void audioresample_buffer_free_mem (AudioresampleBuffer * buffer,
void *);
......
......@@ -26,8 +26,8 @@
#include <stdio.h>
#include <stdlib.h>
#include <audioresample/functable.h>
#include <audioresample/debug.h>
#include "functable.h"
#include "debug.h"
......
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* 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
......@@ -19,16 +19,17 @@
*/
/* 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
......@@ -40,7 +41,7 @@ GST_ELEMENT_DETAILS ("Audio scaler",
"Resample audio",
"David Schleef <ds@schleef.org>");
/* Audioresample signals and args */
/* GstAudioresample signals and args */
enum
{
/* FILL ME */
......@@ -79,63 +80,54 @@ enum
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_audioresample_base_init (gpointer g_class);
static void gst_audioresample_class_init (AudioresampleClass * klass);
static void gst_audioresample_init (Audioresample * audioresample);
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_chain (GstPad * pad, GstData * _data);
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);
static GstElementClass *parent_class = NULL;
/* 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 }; */
GType audioresample_get_type (void)
{
static GType audioresample_type = 0;
if (!audioresample_type)
{
static const GTypeInfo audioresample_info = {
sizeof (AudioresampleClass),
gst_audioresample_base_init,
NULL,
(GClassInitFunc) gst_audioresample_class_init,
NULL,
NULL,
sizeof (Audioresample), 0,
(GInstanceInitFunc) gst_audioresample_init,};
audioresample_type =
g_type_register_static (GST_TYPE_ELEMENT, "Audioresample",
&audioresample_info, 0);
}
return audioresample_type;
}
#define DEBUG_INIT(bla) \
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0, "audio resampling element");
static void gst_audioresample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
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));
static void gst_audioresample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
}
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));
static void gst_audioresample_class_init (AudioresampleClass * klass)
gst_element_class_set_details (gstelement_class,
&gst_audioresample_details);
}
static void gst_audioresample_class_init (GstAudioresampleClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audioresample_set_property;
gobject_class->get_property = gst_audioresample_get_property;
......@@ -145,240 +137,270 @@ static void gst_audioresample_class_init (AudioresampleClass * klass)
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0,
"audioresample element");
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_expand_caps (GstCaps * caps)
static void gst_audioresample_init (GstAudioresample * audioresample)
{
gint i;
ResampleState *r;
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
r = resample_new ();
audioresample->resample = r;
value = gst_structure_get_value (structure, "rate");
if (value == NULL) {
GST_ERROR ("caps structure doesn't have required rate field");
return;
}
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);
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0);
if (audioresample->resample) {
resample_free (audioresample->resample);
audioresample->resample = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstCaps *gst_audioresample_getcaps (GstPad * pad)
{
Audioresample *audioresample;
GstCaps *caps;
GstPad *otherpad;
/* vmethods */
gboolean
audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size) {
gint width, channels;
GstStructure *structure;
gboolean ret;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
g_return_val_if_fail (size, FALSE);
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
audioresample->srcpad;
caps = gst_pad_get_allowed_caps (otherpad);
/* 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);
gst_audioresample_expand_caps (caps);
*size = width * channels / 8;
return caps;
return TRUE;
}
static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps)
GstCaps *audioresample_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps)
{
Audioresample *audioresample;
GstPad *otherpad;
int rate;
GstCaps *copy;
GstCaps *temp, *res;
const GstCaps *templcaps;
GstStructure *structure;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
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);
if (pad == audioresample->srcpad) {
otherpad = audioresample->sinkpad;
rate = audioresample->i_rate;
} else
{
otherpad = audioresample->srcpad;
rate = audioresample->o_rate;
}
if (!GST_PAD_IS_NEGOTIATING (otherpad))
return NULL;
if (gst_caps_get_size (caps) > 1)
return NULL;
copy = gst_caps_copy (caps);
structure = gst_caps_get_structure (copy, 0);
if (rate) {
if (gst_caps_structure_fixate_field_nearest_int (structure, "rate", rate)) {
return copy;
}
}
gst_caps_free (copy);
return NULL;
return res;
}
static GstPadLinkReturn gst_audioresample_link (GstPad * pad,
const GstCaps * caps)
static gboolean
resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
{
Audioresample *audioresample;
GstStructure *structure;
int rate;
int channels;
gboolean ret;
GstPad *otherpad;
gint myinrate, myoutrate;
int mychannels;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
audioresample->srcpad;
structure = gst_caps_get_structure (incaps, 0);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &rate);
ret &= gst_structure_get_int (structure, "channels", &channels);
if (!ret)
{
return GST_PAD_LINK_REFUSED;
/* 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
if (gst_pad_is_negotiated (otherpad))
{
GstCaps *othercaps = gst_caps_copy (caps);
int otherrate;
GstPadLinkReturn linkret;
ret = gst_structure_get_int (structure, "rate", &myinrate);
ret &= gst_structure_get_int (structure, "channels", &mychannels);
g_return_val_if_fail (ret, FALSE);
if (pad == audioresample->srcpad) {
otherrate = audioresample->i_rate;
} else {
otherrate = audioresample->o_rate;
}
gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL);
linkret = gst_pad_try_set_caps (otherpad, othercaps);
if (GST_PAD_LINK_FAILED (linkret)) {
return GST_PAD_LINK_REFUSED;
}
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;
audioresample->channels = channels;
resample_set_n_channels (audioresample->resample, audioresample->channels);
if (pad == audioresample->srcpad) {
audioresample->o_rate = rate;
resample_set_output_rate (audioresample->resample, audioresample->o_rate);
GST_DEBUG ("set o_rate to %d", rate);
} else {
audioresample->i_rate = rate;
resample_set_input_rate (audioresample->resample, audioresample->i_rate);
GST_DEBUG ("set i_rate to %d", rate);
}
resample_set_n_channels (state, mychannels);
resample_set_input_rate (state, myinrate);
resample_set_output_rate (state, myoutrate);
return GST_PAD_LINK_OK;
return TRUE;
}
static void gst_audioresample_init (Audioresample * audioresample)
gboolean audioresample_transform_size (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
guint * othersize)
{
ResampleState *r;
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;
}
audioresample->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioresample_sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad);
gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain);
gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->sinkpad,
gst_audioresample_getcaps);
gst_pad_set_fixate_function (audioresample->sinkpad,
gst_audioresample_fixate);
audioresample->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioresample_src_template), "src");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad);
gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->srcpad,
gst_audioresample_getcaps);
gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate);
/* 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);
}
r = resample_new ();
audioresample->resample = r;
/* 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);
}
resample_set_filter_length (r, 64);
resample_set_format (r, RESAMPLE_FORMAT_S16);
if (!use_internal) {
resample_free (state);
}
return ret;
}
static void gst_audioresample_dispose (GObject * object)
gboolean audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
Audioresample *audioresample = GST_AUDIORESAMPLE (object);
gboolean ret;
gint inrate, outrate;
int channels;
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
if (audioresample->resample) {
resample_free (audioresample->resample);
}
GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
GST_PTR_FORMAT, incaps, outcaps);
G_OBJECT_CLASS (parent_class)->dispose (object);
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 void gst_audioresample_chain (GstPad * pad, GstData * _data)
static GstFlowReturn
audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstBuffer *buf = GST_BUFFER (_data);
Audioresample *audioresample;
/* FIXME: this-> */
GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
ResampleState *r;
guchar *data;
gulong size;
int outsize;
GstBuffer *outbuf;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
if (!GST_IS_BUFFER (_data)) {
gst_pad_push (audioresample->srcpad, _data);
return;
}
/* 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 (buf);
size = GST_BUFFER_SIZE (buf);
data = GST_BUFFER_DATA (inbuf);
size = GST_BUFFER_SIZE (inbuf);
GST_DEBUG ("got buffer of %ld bytes", size);
GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
resample_add_input_data (r, data, size, (ResampleCallback) gst_data_unref,
buf);
resample_add_input_data (r, data, size, NULL, NULL);
outsize = resample_get_output_size (r);
/* FIXME this is audioresample being dumb. dunno why */
if (outsize == 0) {
GST_ERROR ("overriding outbuf size");
outsize = size;
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);
}
outbuf = gst_buffer_new_and_alloc (outsize);
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
gst_pad_push (audioresample->srcpad, GST_DATA (outbuf));
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