Commit 0b4a3a4d authored by Tim-Philipp Müller's avatar Tim-Philipp Müller
Browse files

Remove dead audiofile plugin

This was never even ported to 0.10.
parent 91429180
......@@ -16,12 +16,6 @@ else
APEXSINK_DIR =
endif
# if USE_AUDIOFILE
# AUDIOFILE_DIR=audiofile
# else
AUDIOFILE_DIR=
# endif
if USE_BS2B
BS2B_DIR=bs2b
else
......
plugin_LTLIBRARIES = libgstaudiofile.la
libgstaudiofile_la_SOURCES = gstaf.c gstafsink.c gstafsrc.c gstafparse.c
libgstaudiofile_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_CFLAGS) $(AUDIOFILE_CFLAGS)
libgstaudiofile_la_LIBADD = $(AUDIOFILE_LIBS)
libgstaudiofile_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstaudiofile_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstafsink.h gstafsrc.h gstafparse.h
This plugin wraps the SGI Audiofile
(http://oss.sgi.com/projects/audiofile/) library into a src and sink
element.
You can read from and write to the supported formats (WAVE, AIFF, AIFFC,
NEXTSND).
What is supported :
* all the file formats
* integer sample data, both 2's complement and unsigned
* 8 or 16 bit width & depth (haven't tested others)
* sample rate
* some sort of endianness control
What isn't supported yet :
* float data
What you can do :
* src element only accepts location argument
* sink element accepts location, endianness and type
- location : file on the system to output
- endianness : at this time endianness is still a bit shady
you can either set 1234 or 4321;
setting it to 4321 will byteswap the buffer data
you might want to keep it at 1234 for now
- type : one of the file types
Use gstreamer-inspect on afsink and afsrc to see all of the supported
options.
Examples :
* tools/gstreamer-launch afsrc location=/opt/media/wav/dark-480-16-m.wav ! afsink type=2 location=/opt/media/wav/dark-480-16-m.aiff
Future plans :
* add float support
* wrap up afsink and afsrc with pipe and fork to act like data convertors,
allowing arbitrary choice of sink and src element
/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstafsrc.h"
#include "gstafsink.h"
#include "gstafparse.h"
gboolean gst_aftypes_plugin_init (GstPlugin * plugin);
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_library_load ("gstbytestream"))
return FALSE;
gst_afsink_plugin_init (plugin);
gst_afsrc_plugin_init (plugin);
gst_afparse_plugin_init (plugin);
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
audiofile,
"Audiofile plugin", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN)
/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#include "gstafparse.h"
/* 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 */
static GstStaticPadTemplate afparse_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 8, 16 }, "
"depth = (int) { 8, 16 }, " "signed = (boolean) { true, false }")
);
static GstStaticPadTemplate afparse_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-aiff; " "audio/x-wav; " "audio/x-au")
);
static void gst_afparse_base_init (gpointer g_class);
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),
gst_afparse_base_init,
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_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&afparse_src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&afparse_sink_factory));
gst_element_class_set_static_metadata (element_class, "Audiofile demuxer",
"Codec/Demuxer/Audio",
"Audiofile parser for audio/raw",
"Steve Baker <stevebaker_org@yahoo.co.uk>");
}
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 (gst_element_get_pad_template (GST_ELEMENT
(afparse), "src"), "src");
gst_pad_use_explicit_caps (afparse->srcpad);
gst_element_add_pad (GST_ELEMENT (afparse), afparse->srcpad);
afparse->sinkpad =
gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
(afparse), "sink"), "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;
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_DATA (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;
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_DATA (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, GST_DATA (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_and_alloc (bytes_per_read);
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_DATA (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, GST_DATA (buf));
afparse->timestamp += numframes * 1E9 / afparse->rate;
}
while (TRUE);
}
gst_afparse_close_file (afparse);
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;
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 (GstPlugin * plugin)
{
/* load audio support library */
if (!gst_library_load ("gstaudio"))
return FALSE;
if (!gst_element_register (plugin, "afparse", GST_RANK_NONE,
GST_TYPE_AFPARSE))
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_OBJECT_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_set_explicit_caps (afparse->srcpad,
gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, afparse->is_signed,
"width", G_TYPE_INT, afparse->width,
"depth", G_TYPE_INT, afparse->width,
"rate", G_TYPE_INT, afparse->rate,
"channels", G_TYPE_INT, afparse->channels, NULL));
GST_OBJECT_FLAG_SET (afparse, GST_AFPARSE_OPEN);
return TRUE;
}
static void
gst_afparse_close_file (GstAFParse * afparse)
{
g_return_if_fail (GST_OBJECT_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN));
if (afCloseFile (afparse->file) != 0) {
g_warning ("afparse: oops, error closing !\n");
} else {
GST_OBJECT_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;
}