gstreamer/ext/audiofile/gstafparse.c
Ronald S. Bultje 95011fd7e8 New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri...
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
2003-07-06 20:49:52 +00:00

532 lines
15 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstafparse.c:
*
* 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 <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#include "gstafparse.h"
/* elementfactory information */
static GstElementDetails afparse_details = {
"Audiofile Parse",
"Codec/Parser",
"LGPL",
"Audiofile parser for audio/raw",
VERSION,
"Steve Baker <stevebaker_org@yahoo.co.uk>",
"(C) 2002"
};
/* AFParse signals and args */
enum {
/* FILL ME */
SIGNAL_HANDOFF,
LAST_SIGNAL
};
enum {
ARG_0,
};
/* added a src factory function to force audio/raw MIME type */
GST_PAD_TEMPLATE_FACTORY (afparse_src_factory,
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"audiofile_src",
"audio/x-raw-int",
"endianness", GST_PROPS_INT (G_BYTE_ORDER),
"signed", GST_PROPS_LIST (GST_PROPS_BOOLEAN (TRUE), GST_PROPS_BOOLEAN (FALSE)),
"width", GST_PROPS_INT_RANGE (8, 16),
"depth", GST_PROPS_INT_RANGE (8, 16),
"rate", GST_PROPS_INT_RANGE (1, G_MAXINT),
"channels", GST_PROPS_INT_RANGE (1, 2)
)
)
GST_PAD_TEMPLATE_FACTORY (afparse_sink_factory,
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"afparse_sink_aiff",
"audio/x-aiff",
NULL
),
GST_CAPS_NEW (
"afparse_sink_wav",
"audio/x-wav",
NULL
),
GST_CAPS_NEW (
"afparse_sink_snd",
"audio/x-au",
NULL
)
)
static void gst_afparse_class_init(GstAFParseClass *klass);
static void gst_afparse_init (GstAFParse *afparse);
static gboolean gst_afparse_open_file(GstAFParse *afparse);
static void gst_afparse_close_file(GstAFParse *afparse);
static void gst_afparse_loop(GstElement *element);
static void gst_afparse_set_property(GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec);
static void gst_afparse_get_property(GObject *object, guint prop_id, GValue *value, GParamSpec *pspec);
static ssize_t gst_afparse_vf_read (AFvirtualfile *vfile, void *data, size_t nbytes);
static long gst_afparse_vf_length (AFvirtualfile *vfile);
static ssize_t gst_afparse_vf_write (AFvirtualfile *vfile, const void *data, size_t nbytes);
static void gst_afparse_vf_destroy(AFvirtualfile *vfile);
static long gst_afparse_vf_seek (AFvirtualfile *vfile, long offset, int is_relative);
static long gst_afparse_vf_tell (AFvirtualfile *vfile);
GType
gst_afparse_get_type (void)
{
static GType afparse_type = 0;
if (!afparse_type) {
static const GTypeInfo afparse_info = {
sizeof (GstAFParseClass), NULL,
NULL,
(GClassInitFunc) gst_afparse_class_init,
NULL,
NULL,
sizeof (GstAFParse),
0,
(GInstanceInitFunc) gst_afparse_init,
};
afparse_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAFParse", &afparse_info, 0);
}
return afparse_type;
}
static void
gst_afparse_class_init (GstAFParseClass *klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
gobject_class->set_property = gst_afparse_set_property;
gobject_class->get_property = gst_afparse_get_property;
}
static void
gst_afparse_init (GstAFParse *afparse)
{
afparse->srcpad = gst_pad_new_from_template (afparse_src_factory (), "src");
gst_element_add_pad (GST_ELEMENT (afparse), afparse->srcpad);
afparse->sinkpad = gst_pad_new_from_template (afparse_sink_factory (), "sink");
gst_element_add_pad (GST_ELEMENT (afparse), afparse->sinkpad);
gst_element_set_loop_function (GST_ELEMENT (afparse), gst_afparse_loop);
afparse->vfile = af_virtual_file_new();
afparse->vfile->closure = NULL;
afparse->vfile->read = gst_afparse_vf_read;
afparse->vfile->length = gst_afparse_vf_length;
afparse->vfile->write = gst_afparse_vf_write;
afparse->vfile->destroy = gst_afparse_vf_destroy;
afparse->vfile->seek = gst_afparse_vf_seek;
afparse->vfile->tell = gst_afparse_vf_tell;
afparse->frames_per_read = 1024;
afparse->curoffset = 0;
afparse->seq = 0;
afparse->file = NULL;
/* default values, should never be needed */
afparse->channels = 2;
afparse->width = 16;
afparse->rate = 44100;
afparse->type = AF_FILE_WAVE;
afparse->endianness_data = 1234;
afparse->endianness_wanted = 1234;
afparse->timestamp = 0LL;
}
static void
gst_afparse_loop(GstElement *element)
{
GstAFParse *afparse;
GstBuffer *buf;
GstBufferPool *bufpool;
gint numframes = 0, frames_to_bytes, frames_per_read, bytes_per_read;
guint8 *data;
gboolean bypass_afread = TRUE;
GstByteStream *bs;
int s_format, v_format, s_width, v_width;
afparse = GST_AFPARSE(element);
afparse->vfile->closure = bs = gst_bytestream_new (afparse->sinkpad);
/* just stop if we cannot open the file */
if (!gst_afparse_open_file (afparse)){
gst_bytestream_destroy ((GstByteStream *) afparse->vfile->closure);
gst_pad_push (afparse->srcpad, GST_BUFFER(gst_event_new (GST_EVENT_EOS)));
gst_element_set_eos (GST_ELEMENT (afparse));
return;
}
/* if audiofile changes the data in any way, we have to access
* the audio data via afReadFrames. Otherwise we can just access
* the data directly. */
afGetSampleFormat(afparse->file, AF_DEFAULT_TRACK, &s_format, &s_width);
afGetVirtualSampleFormat(afparse->file, AF_DEFAULT_TRACK, &v_format, &v_width);
if (afGetCompression != AF_COMPRESSION_NONE ||
afGetByteOrder(afparse->file, AF_DEFAULT_TRACK) != afGetVirtualByteOrder(afparse->file, AF_DEFAULT_TRACK) ||
s_format != v_format ||
s_width != v_width) {
bypass_afread = FALSE;
}
if (bypass_afread){
GST_DEBUG("will bypass afReadFrames\n");
}
frames_to_bytes = afparse->channels * afparse->width / 8;
frames_per_read = afparse->frames_per_read;
bytes_per_read = frames_per_read * frames_to_bytes;
bufpool = gst_buffer_pool_get_default (bytes_per_read, 8);
afSeekFrame(afparse->file, AF_DEFAULT_TRACK, 0);
if (bypass_afread){
GstEvent *event = NULL;
guint32 waiting;
guint32 got_bytes;
do {
got_bytes = gst_bytestream_read (bs, &buf, bytes_per_read);
if (got_bytes == 0) {
/* we need to check for an event. */
gst_bytestream_get_status (bs, &waiting, &event);
if (event && GST_EVENT_TYPE(event) == GST_EVENT_EOS) {
gst_pad_push (afparse->srcpad,
GST_BUFFER (gst_event_new (GST_EVENT_EOS)));
gst_element_set_eos (GST_ELEMENT (afparse));
break;
}
}
else {
GST_BUFFER_TIMESTAMP(buf) = afparse->timestamp;
gst_pad_push (afparse->srcpad, buf);
if (got_bytes != bytes_per_read){
/* this shouldn't happen very often */
/* FIXME calculate the timestamps based on the fewer bytes received */
}
else {
afparse->timestamp += frames_per_read * 1E9 / afparse->rate;
}
}
}
while (TRUE);
}
else {
do {
buf = gst_buffer_new_from_pool (bufpool, 0, 0);
GST_BUFFER_TIMESTAMP(buf) = afparse->timestamp;
data = GST_BUFFER_DATA(buf);
numframes = afReadFrames (afparse->file, AF_DEFAULT_TRACK, data, frames_per_read);
/* events are handled in gst_afparse_vf_read so if there are no
* frames it must be EOS */
if (numframes < 1){
gst_buffer_unref(buf);
gst_pad_push (afparse->srcpad, GST_BUFFER(gst_event_new (GST_EVENT_EOS)));
gst_element_set_eos (GST_ELEMENT (afparse));
break;
}
GST_BUFFER_SIZE(buf) = numframes * frames_to_bytes;
gst_pad_push (afparse->srcpad, buf);
afparse->timestamp += numframes * 1E9 / afparse->rate;
}
while (TRUE);
}
gst_afparse_close_file (afparse);
gst_buffer_pool_unref(bufpool);
gst_bytestream_destroy ((GstByteStream*) afparse->vfile->closure);
}
static void
gst_afparse_set_property (GObject *object, guint prop_id,
const GValue *value, GParamSpec *pspec)
{
GstAFParse *afparse;
/* it's not null if we got it, but it might not be ours */
afparse = GST_AFPARSE (object);
switch (prop_id) {
default:
break;
}
}
static void
gst_afparse_get_property (GObject *object, guint prop_id,
GValue *value, GParamSpec *pspec)
{
GstAFParse *afparse;
g_return_if_fail (GST_IS_AFPARSE (object));
afparse = GST_AFPARSE (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_afparse_plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
factory = gst_element_factory_new ("afparse", GST_TYPE_AFPARSE,
&afparse_details);
g_return_val_if_fail (factory != NULL, FALSE);
gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (afparse_src_factory));
gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (afparse_sink_factory));
/* gst_element_factory_set_rank (factory, GST_ELEMENT_RANK_PRIMARY);*/
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
/* load audio support library */
if (!gst_library_load ("gstaudio"))
return FALSE;
if (!gst_library_load ("gstbytestream"))
return FALSE;
return TRUE;
}
/* this is where we open the audiofile */
static gboolean
gst_afparse_open_file (GstAFParse *afparse)
{
g_return_val_if_fail (!GST_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN), FALSE);
/* open the file */
GST_DEBUG("opening vfile %p\n", afparse->vfile);
afparse->file = afOpenVirtualFile (afparse->vfile, "r", AF_NULL_FILESETUP);
if (afparse->file == AF_NULL_FILEHANDLE)
{
/* this should never happen */
g_warning ("ERROR: gstafparse: Could not open virtual file for reading\n");
return FALSE;
}
GST_DEBUG("vfile opened\n");
/* get the audiofile audio parameters */
{
int sampleFormat, sampleWidth;
afparse->channels = afGetChannels (afparse->file, AF_DEFAULT_TRACK);
afGetSampleFormat (afparse->file, AF_DEFAULT_TRACK,
&sampleFormat, &sampleWidth);
switch (sampleFormat)
{
case AF_SAMPFMT_TWOSCOMP:
afparse->is_signed = TRUE;
break;
case AF_SAMPFMT_UNSIGNED:
afparse->is_signed = FALSE;
break;
case AF_SAMPFMT_FLOAT:
case AF_SAMPFMT_DOUBLE:
GST_DEBUG ("ERROR: float data not supported yet !\n");
}
afparse->rate = (guint) afGetRate (afparse->file, AF_DEFAULT_TRACK);
afparse->width = sampleWidth;
GST_DEBUG (
"input file: %d channels, %d width, %d rate, signed %s\n",
afparse->channels, afparse->width, afparse->rate,
afparse->is_signed ? "yes" : "no");
}
/* set caps on src */
/*FIXME: add all the possible formats, especially float ! */
gst_pad_try_set_caps (afparse->srcpad,
GST_CAPS_NEW (
"af_src",
"audio/x-raw-int",
"endianness", GST_PROPS_INT (G_BYTE_ORDER), /*FIXME */
"signed", GST_PROPS_BOOLEAN (afparse->is_signed),
"width", GST_PROPS_INT (afparse->width),
"depth", GST_PROPS_INT (afparse->width),
"rate", GST_PROPS_INT (afparse->rate),
"channels", GST_PROPS_INT (afparse->channels)
)
);
GST_FLAG_SET (afparse, GST_AFPARSE_OPEN);
return TRUE;
}
static void
gst_afparse_close_file (GstAFParse *afparse)
{
g_return_if_fail (GST_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN));
if (afCloseFile (afparse->file) != 0)
{
g_warning ("afparse: oops, error closing !\n");
}
else {
GST_FLAG_UNSET (afparse, GST_AFPARSE_OPEN);
}
}
static ssize_t
gst_afparse_vf_read (AFvirtualfile *vfile, void *data, size_t nbytes)
{
GstByteStream *bs = (GstByteStream*)vfile->closure;
guint8 *bytes = NULL;
GstEvent *event = NULL;
guint32 waiting;
guint32 got_bytes;
/*gchar *debug_str;*/
got_bytes = gst_bytestream_peek_bytes(bs, &bytes, nbytes);
while (got_bytes != nbytes){
/* handle events */
gst_bytestream_get_status (bs, &waiting, &event);
/* FIXME this event handling isn't right yet */
if (!event){
/*g_print("no event found with %u bytes\n", got_bytes);*/
return 0;
}
switch (GST_EVENT_TYPE(event)) {
case GST_EVENT_EOS:
return 0;
case GST_EVENT_FLUSH:
GST_DEBUG("flush");
break;
case GST_EVENT_DISCONTINUOUS:
GST_DEBUG("seek done");
got_bytes = gst_bytestream_peek_bytes(bs, &bytes, nbytes);
break;
default:
g_warning("unknown event %d", GST_EVENT_TYPE(event));
got_bytes = gst_bytestream_peek_bytes(bs, &bytes, nbytes);
}
}
memcpy(data, bytes, got_bytes);
gst_bytestream_flush_fast(bs, got_bytes);
/* debug_str = g_strndup((gchar*)bytes, got_bytes);
g_print("read %u bytes: %s\n", got_bytes, debug_str);
*/
return got_bytes;
}
static long
gst_afparse_vf_seek (AFvirtualfile *vfile, long offset, int is_relative)
{
GstByteStream *bs = (GstByteStream*)vfile->closure;
GstSeekType method;
guint64 current_offset = gst_bytestream_tell(bs);
if (!is_relative){
if ((guint64)offset == current_offset) {
/* this seems to happen before every read - bad audiofile */
return offset;
}
method = GST_SEEK_METHOD_SET;
}
else {
if (offset == 0) return current_offset;
method = GST_SEEK_METHOD_CUR;
}
if (gst_bytestream_seek(bs, (gint64)offset, method)){
GST_DEBUG("doing seek to %d", (gint)offset);
return offset;
}
return 0;
}
static long
gst_afparse_vf_length (AFvirtualfile *vfile)
{
GstByteStream *bs = (GstByteStream*)vfile->closure;
guint64 length;
length = gst_bytestream_length(bs);
GST_DEBUG("doing length: %" G_GUINT64_FORMAT, length);
return length;
}
static ssize_t
gst_afparse_vf_write (AFvirtualfile *vfile, const void *data, size_t nbytes)
{
/* GstByteStream *bs = (GstByteStream*)vfile->closure;*/
g_warning("shouldn't write to a readonly pad");
return 0;
}
static void
gst_afparse_vf_destroy(AFvirtualfile *vfile)
{
/* GstByteStream *bs = (GstByteStream*)vfile->closure;*/
GST_DEBUG("doing destroy");
}
static long
gst_afparse_vf_tell (AFvirtualfile *vfile)
{
GstByteStream *bs = (GstByteStream*)vfile->closure;
guint64 offset;
offset = gst_bytestream_tell(bs);
GST_DEBUG("doing tell: %" G_GUINT64_FORMAT, offset);
return offset;
}