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Remove dead audiofile plugin

Browse source 
This was never even ported to 0.10.
This commit is contained in:
Tim-Philipp Müller 2016-01-31 19:09:57 +00:00
parent 914291808a
commit 0b4a3a4dbd
11 changed files with 0 additions and 1807 deletions
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6
ext/Makefile.am
View file

@ -16,12 +16,6 @@ else
APEXSINK_DIR = APEXSINK_DIR =
endif endif
# if USE_AUDIOFILE
# AUDIOFILE_DIR=audiofile
# else
AUDIOFILE_DIR=
# endif
if USE_BS2B if USE_BS2B
BS2B_DIR=bs2b BS2B_DIR=bs2b
else else

10
ext/audiofile/Makefile.am
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@ -1,10 +0,0 @@
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

39
ext/audiofile/README
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@ -1,39 +0,0 @@
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

47
ext/audiofile/gstaf.c
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@ -1,47 +0,0 @@
/* 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)

515
ext/audiofile/gstafparse.c
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@ -1,515 +0,0 @@
/* 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;
}

101
ext/audiofile/gstafparse.h
View file

@ -1,101 +0,0 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstafparse.h:
*
* 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.
*/
#ifndef __GST_AFPARSE_H__
#define __GST_AFPARSE_H__
#include <gst/gst.h>
#include <gst/bytestream/bytestream.h>
#include <audiofile.h> /* what else are we to do */
#include <af_vfs.h>
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#define GST_TYPE_AFPARSE \
(gst_afparse_get_type())
#define GST_AFPARSE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFPARSE,GstAFParse))
#define GST_AFPARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFPARSE,GstAFParseClass))
#define GST_IS_AFPARSE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFPARSE))
#define GST_IS_AFPARSE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFPARSE))
typedef struct _GstAFParse GstAFParse;
typedef struct _GstAFParseClass GstAFParseClass;
typedef enum {
GST_AFPARSE_OPEN = (GST_ELEMENT_FLAG_LAST << 0),
GST_AFPARSE_FLAG_LAST = (GST_ELEMENT_FLAG_LAST << 2),
} GstAFParseFlags;
struct _GstAFParse {
GstElement element;
GstPad *srcpad;
GstPad *sinkpad;
AFvirtualfile *vfile;
AFfilehandle file;
int format;
int channels;
int width;
unsigned int rate;
gboolean is_signed;
int type; /* type of output, compare to audiofile.h
* RAW, AIFF, AIFFC, NEXTSND, WAVE
*/
/* blocking */
gulong curoffset;
gulong bytes_per_read;
gint frames_per_read;
gulong seq;
gint64 timestamp;
/* FIXME : endianness is a little cryptic at this point */
int endianness_data; /* 4321 or 1234 */
int endianness_wanted; /* same thing, but what the output format wants */
int endianness_output; /* what the output endianness will be */
};
struct _GstAFParseClass {
GstElementClass parent_class;
/* signals */
void (*handoff) (GstElement *element,GstPad *pad);
};
gboolean gst_afparse_plugin_init (GstPlugin *plugin);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __GST_AFPARSE_H__ */

490
ext/audiofile/gstafsink.c
View file

@ -1,490 +0,0 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstafsink.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-i18n-plugin.h"
#include <gst/gst.h>
#include <string.h>
#include <errno.h>
#include "gstafsink.h"
/* AFSink signals and args */
enum
{
/* FILL ME */
SIGNAL_HANDOFF,
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_TYPE,
ARG_OUTPUT_ENDIANNESS,
ARG_LOCATION
};
/* added a sink factory function to force audio/raw MIME type */
/* I think the caps can be broader, we need to change that somehow */
static GstStaticPadTemplate afsink_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 2 ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) { 8, 16 }, "
"depth = (int) { 8, 16 }, " "signed = (boolean) { true, false }")
);
/* we use an enum for the output type arg */
#define GST_TYPE_AFSINK_TYPES (gst_afsink_types_get_type())
/* FIXME: fix the string ints to be string-converted from the audiofile.h types */
static GType
gst_afsink_types_get_type (void)
{
static GType afsink_types_type = 0;
static const GEnumValue afsink_types[] = {
{AF_FILE_RAWDATA, "0", "raw PCM"},
{AF_FILE_AIFFC, "1", "AIFFC"},
{AF_FILE_AIFF, "2", "AIFF"},
{AF_FILE_NEXTSND, "3", "Next/SND"},
{AF_FILE_WAVE, "4", "Wave"},
{0, NULL, NULL},
};
if (!afsink_types_type) {
afsink_types_type =
g_enum_register_static ("GstAudiosinkTypes", afsink_types);
}
return afsink_types_type;
}
static void gst_afsink_base_init (gpointer g_class);
static void gst_afsink_class_init (GstAFSinkClass * klass);
static void gst_afsink_init (GstAFSink * afsink);
static gboolean gst_afsink_open_file (GstAFSink * sink);
static void gst_afsink_close_file (GstAFSink * sink);
static void gst_afsink_chain (GstPad * pad, GstData * _data);
static void gst_afsink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_afsink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_afsink_handle_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_afsink_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
static guint gst_afsink_signals[LAST_SIGNAL] = { 0 };
GType
gst_afsink_get_type (void)
{
static GType afsink_type = 0;
if (!afsink_type) {
static const GTypeInfo afsink_info = {
sizeof (GstAFSinkClass),
gst_afsink_base_init,
NULL,
(GClassInitFunc) gst_afsink_class_init,
NULL,
NULL,
sizeof (GstAFSink),
0,
(GInstanceInitFunc) gst_afsink_init,
};
afsink_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstAFSink", &afsink_info, 0);
}
return afsink_type;
}
static void
gst_afsink_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 (&afsink_sink_factory));
gst_element_class_set_static_metadata (element_class, "Audiofile sink",
"Sink/Audio",
"Write audio streams to disk using libaudiofile",
"Thomas Vander Stichele <thomas@apestaart.org>");
}
static void
gst_afsink_class_init (GstAFSinkClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass),
"location", ARG_LOCATION, G_PARAM_READWRITE, NULL);
/* FIXME: add long property descriptions */
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TYPE,
g_param_spec_enum ("type", "type", "type", GST_TYPE_AFSINK_TYPES, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
ARG_OUTPUT_ENDIANNESS, g_param_spec_int ("endianness", "endianness",
"endianness", G_MININT, G_MAXINT, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_afsink_signals[SIGNAL_HANDOFF] =
g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstAFSinkClass, handoff), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
gobject_class->set_property = gst_afsink_set_property;
gobject_class->get_property = gst_afsink_get_property;
gstelement_class->change_state = gst_afsink_change_state;
}
static void
gst_afsink_init (GstAFSink * afsink)
{
/* GstPad *pad; this is now done in the struct */
afsink->sinkpad =
gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
(afsink), "sink"), "sink");
gst_element_add_pad (GST_ELEMENT (afsink), afsink->sinkpad);
gst_pad_set_chain_function (afsink->sinkpad, gst_afsink_chain);
afsink->filename = NULL;
afsink->file = NULL;
/* default values, should never be needed */
afsink->channels = 2;
afsink->width = 16;
afsink->rate = 44100;
afsink->type = AF_FILE_WAVE;
afsink->endianness_data = 1234;
afsink->endianness_wanted = 1234;
}
static void
gst_afsink_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstAFSink *sink;
sink = GST_AFSINK (object);
switch (prop_id) {
case ARG_LOCATION:
/* the element must be stopped or paused in order to do this */
g_return_if_fail ((GST_STATE (sink) < GST_STATE_PLAYING)
|| (GST_STATE (sink) == GST_STATE_PAUSED));
g_free (sink->filename);
sink->filename = g_strdup (g_value_get_string (value));
if ((GST_STATE (sink) == GST_STATE_PAUSED)
&& (sink->filename != NULL)) {
gst_afsink_close_file (sink);
gst_afsink_open_file (sink);
}
break;
case ARG_TYPE:
sink->type = g_value_get_enum (value);
break;
case ARG_OUTPUT_ENDIANNESS:
{
int end = g_value_get_int (value);
if (end == 1234 || end == 4321)
sink->endianness_output = end;
}
break;
default:
break;
}
}
static void
gst_afsink_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAFSink *sink;
g_return_if_fail (GST_IS_AFSINK (object));
sink = GST_AFSINK (object);
switch (prop_id) {
case ARG_LOCATION:
g_value_set_string (value, sink->filename);
break;
case ARG_TYPE:
g_value_set_enum (value, sink->type);
break;
case ARG_OUTPUT_ENDIANNESS:
g_value_set_int (value, sink->endianness_output);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_afsink_plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "afsink", GST_RANK_NONE, GST_TYPE_AFSINK))
return FALSE;
#ifdef ENABLE_NLS
setlocale (LC_ALL, "");
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
#endif /* ENABLE_NLS */
return TRUE;
}
/* this is where we open the audiofile */
static gboolean
gst_afsink_open_file (GstAFSink * sink)
{
AFfilesetup outfilesetup;
const GstCaps *caps;
GstStructure *structure;
int sample_format; /* audiofile's sample format, look in audiofile.h */
int byte_order = 0; /* audiofile's byte order defines */
g_return_val_if_fail (!GST_OBJECT_FLAG_IS_SET (sink, GST_AFSINK_OPEN), FALSE);
/* get the audio parameters */
g_return_val_if_fail (GST_IS_PAD (sink->sinkpad), FALSE);
caps = GST_PAD_CAPS (sink->sinkpad);
if (caps == NULL) {
g_critical ("gstafsink chain : Could not get caps of pad !\n");
} else {
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "channels", &sink->channels);
gst_structure_get_int (structure, "width", &sink->width);
gst_structure_get_int (structure, "rate", &sink->rate);
gst_structure_get_boolean (structure, "signed", &sink->is_signed);
gst_structure_get_int (structure, "endianness", &sink->endianness_data);
}
GST_DEBUG ("channels %d, width %d, rate %d, signed %s",
sink->channels, sink->width, sink->rate, sink->is_signed ? "yes" : "no");
GST_DEBUG ("endianness: data %d, output %d",
sink->endianness_data, sink->endianness_output);
/* setup the output file */
if (sink->is_signed)
sample_format = AF_SAMPFMT_TWOSCOMP;
else
sample_format = AF_SAMPFMT_UNSIGNED;
/* FIXME : this check didn't seem to work, so let the output endianness be set */
/*
if (sink->endianness_data == sink->endianness_wanted)
byte_order = AF_BYTEORDER_LITTLEENDIAN;
else
byte_order = AF_BYTEORDER_BIGENDIAN;
*/
if (sink->endianness_output == 1234)
byte_order = AF_BYTEORDER_LITTLEENDIAN;
else
byte_order = AF_BYTEORDER_BIGENDIAN;
outfilesetup = afNewFileSetup ();
afInitFileFormat (outfilesetup, sink->type);
afInitChannels (outfilesetup, AF_DEFAULT_TRACK, sink->channels);
afInitRate (outfilesetup, AF_DEFAULT_TRACK, sink->rate);
afInitSampleFormat (outfilesetup, AF_DEFAULT_TRACK,
sample_format, sink->width);
/* open it */
sink->file = afOpenFile (sink->filename, "w", outfilesetup);
if (sink->file == AF_NULL_FILEHANDLE) {
GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
(_("Could not open file \"%s\" for writing."), sink->filename),
("system error: %s", strerror (errno)));
return FALSE;
}
afFreeFileSetup (outfilesetup);
/* afSetVirtualByteOrder (sink->file, AF_DEFAULT_TRACK, byte_order); */
GST_OBJECT_FLAG_SET (sink, GST_AFSINK_OPEN);
return TRUE;
}
static void
gst_afsink_close_file (GstAFSink * sink)
{
/* g_print ("DEBUG: closing sinkfile...\n"); */
g_return_if_fail (GST_OBJECT_FLAG_IS_SET (sink, GST_AFSINK_OPEN));
/* g_print ("DEBUG: past flag test\n"); */
/* if (fclose (sink->file) != 0) */
if (afCloseFile (sink->file) != 0) {
GST_ELEMENT_ERROR (sink, RESOURCE, CLOSE,
(_("Error closing file \"%s\"."), sink->filename), GST_ERROR_SYSTEM);
} else {
GST_OBJECT_FLAG_UNSET (sink, GST_AFSINK_OPEN);
}
}
/**
* gst_afsink_chain:
* @pad: the pad this afsink is connected to
* @buf: the buffer that has to be absorbed
*
* take the buffer from the pad and write to file if it's open
*/
static void
gst_afsink_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf;
GstAFSink *afsink;
int ret = 0;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
if (GST_IS_EVENT (_data)) {
gst_afsink_handle_event (pad, GST_EVENT (_data));
return;
}
buf = GST_BUFFER (_data);
afsink = GST_AFSINK (gst_pad_get_parent (pad));
/* we use audiofile now
if (GST_OBJECT_FLAG_IS_SET (afsink, GST_AFSINK_OPEN))
{
bytes_written = fwrite (GST_BUFFER_DATA (buf), 1, GST_BUFFER_SIZE (buf), afsink->file);
if (bytes_written < GST_BUFFER_SIZE (buf))
{
printf ("afsink : Warning : %d bytes should be written, only %d bytes written\n",
GST_BUFFER_SIZE (buf), bytes_written);
}
}
*/
if (!GST_OBJECT_FLAG_IS_SET (afsink, GST_AFSINK_OPEN)) {
/* it's not open yet, open it */
if (!gst_afsink_open_file (afsink))
g_print ("WARNING: gstafsink: can't open file !\n");
/* return FALSE; Can't return value */
}
if (GST_OBJECT_FLAG_IS_SET (afsink, GST_AFSINK_OPEN)) {
int frameCount = 0;
frameCount =
GST_BUFFER_SIZE (buf) / ((afsink->width / 8) * afsink->channels);
/* g_print ("DEBUG: writing %d frames ", frameCount); */
ret = afWriteFrames (afsink->file, AF_DEFAULT_TRACK,
GST_BUFFER_DATA (buf), frameCount);
if (ret == AF_BAD_WRITE || ret == AF_BAD_LSEEK) {
printf ("afsink : Warning : afWriteFrames returned an error (%d)\n", ret);
}
}
gst_buffer_unref (buf);
g_signal_emit (G_OBJECT (afsink), gst_afsink_signals[SIGNAL_HANDOFF], 0);
}
static GstStateChangeReturn
gst_afsink_change_state (GstElement * element, GstStateChange transition)
{
g_return_val_if_fail (GST_IS_AFSINK (element), GST_STATE_CHANGE_FAILURE);
/* if going to NULL? then close the file */
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
/* printf ("DEBUG: afsink state change: null pending\n"); */
if (GST_OBJECT_FLAG_IS_SET (element, GST_AFSINK_OPEN)) {
/* g_print ("DEBUG: trying to close the sink file\n"); */
gst_afsink_close_file (GST_AFSINK (element));
}
}
/*
else
this has been moved to the chain function, since it's only then that
the caps are set and can be known
{
g_print ("DEBUG: it's not going to null\n");
if (!GST_OBJECT_FLAG_IS_SET (element, GST_AFSINK_OPEN))
{
g_print ("DEBUG: GST_AFSINK_OPEN not set\n");
if (!gst_afsink_open_file (GST_AFSINK (element)))
{
g_print ("DEBUG: element tries to open file\n");
return GST_STATE_CHANGE_FAILURE;
}
}
}
*/
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}
/* this function was copied from sinesrc */
static gboolean
gst_afsink_handle_event (GstPad * pad, GstEvent * event)
{
GstAFSink *afsink;
afsink = GST_AFSINK (gst_pad_get_parent (pad));
GST_DEBUG ("DEBUG: afsink: got event");
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
gst_afsink_close_file (afsink);
break;
default:
break;
}
gst_pad_event_default (pad, event);
return TRUE;
}

97
ext/audiofile/gstafsink.h
View file

@ -1,97 +0,0 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstafsink.h:
*
* 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.
*/
#ifndef __GST_AFSINK_H__
#define __GST_AFSINK_H__
#include <gst/gst.h>
#include <audiofile.h> /* what else are we to do */
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#define GST_TYPE_AFSINK \
(gst_afsink_get_type())
#define GST_AFSINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFSINK,GstAFSink))
#define GST_AFSINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFSINK,GstAFSinkClass))
#define GST_IS_AFSINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFSINK))
#define GST_IS_AFSINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFSINK))
typedef struct _GstAFSink GstAFSink;
typedef struct _GstAFSinkClass GstAFSinkClass;
typedef enum {
GST_AFSINK_OPEN = (GST_ELEMENT_FLAG_LAST << 0),
GST_AFSINK_FLAG_LAST = (GST_ELEMENT_FLAG_LAST << 2),
} GstAFSinkFlags;
struct _GstAFSink {
GstElement element;
GstPad *sinkpad;
gchar *filename;
/* FILE *file; */
/* AFfilesetup outfilesetup; */
AFfilehandle file;
int format;
int channels;
int width;
unsigned int rate;
gboolean is_signed;
int type; /* type of output, compare to audiofile.h
* RAW, AIFF, AIFFC, NEXTSND, WAVE
*/
/* FIXME : endianness is a little cryptic at this point */
int endianness_data; /* 4321 or 1234 */
int endianness_wanted; /* same thing, but what the output format wants */
int endianness_output; /* what the output endianness will be */
};
struct _GstAFSinkClass {
GstElementClass parent_class;
/* signals */
void (*handoff) (GstElement *element,GstPad *pad);
};
GType gst_afsink_get_type (void);
gboolean gst_afsink_plugin_init (GstPlugin *plugin);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __GST_AFSINK_H__ */

396
ext/audiofile/gstafsrc.c
View file

@ -1,396 +0,0 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstafsrc.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-i18n-plugin.h"
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>
#include <errno.h>
#include "gstafsrc.h"
/* AFSrc signals and args */
enum
{
/* FILL ME */
SIGNAL_HANDOFF,
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_LOCATION
};
/* added a src factory function to force audio/raw MIME type */
/* I think the caps can be broader, we need to change that somehow */
static GstStaticPadTemplate afsrc_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 }")
);
/* we use an enum for the output type arg */
#define GST_TYPE_AFSRC_TYPES (gst_afsrc_types_get_type())
/* FIXME: fix the string ints to be string-converted from the audiofile.h types */
/* defined but not used
static GType
gst_afsrc_types_get_type (void)
{
static GType afsrc_types_type = 0;
static const GEnumValue afsrc_types[] = {
{AF_FILE_RAWDATA, "0", "raw PCM"},
{AF_FILE_AIFFC, "1", "AIFFC"},
{AF_FILE_AIFF, "2", "AIFF"},
{AF_FILE_NEXTSND, "3", "Next/SND"},
{AF_FILE_WAVE, "4", "Wave"},
{0, NULL, NULL},
};
if (!afsrc_types_type)
{
afsrc_types_type = g_enum_register_static ("GstAudiosrcTypes", afsrc_types);
}
return afsrc_types_type;
}
*/
static void gst_afsrc_base_init (gpointer g_class);
static void gst_afsrc_class_init (GstAFSrcClass * klass);
static void gst_afsrc_init (GstAFSrc * afsrc);
static gboolean gst_afsrc_open_file (GstAFSrc * src);
static void gst_afsrc_close_file (GstAFSrc * src);
static GstData *gst_afsrc_get (GstPad * pad);
static void gst_afsrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_afsrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_afsrc_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
static guint gst_afsrc_signals[LAST_SIGNAL] = { 0 };
GType
gst_afsrc_get_type (void)
{
static GType afsrc_type = 0;
if (!afsrc_type) {
static const GTypeInfo afsrc_info = {
sizeof (GstAFSrcClass),
gst_afsrc_base_init,
NULL,
(GClassInitFunc) gst_afsrc_class_init,
NULL,
NULL,
sizeof (GstAFSrc),
0,
(GInstanceInitFunc) gst_afsrc_init,
};
afsrc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstAFSrc", &afsrc_info, 0);
}
return afsrc_type;
}
static void
gst_afsrc_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 (&afsrc_src_factory));
gst_element_class_set_static_metadata (element_class, "Audiofile source",
"Source/Audio",
"Read audio files from disk using libaudiofile",
"Thomas <thomas@apestaart.org>");
}
static void
gst_afsrc_class_init (GstAFSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass),
"location", ARG_LOCATION, G_PARAM_READWRITE, NULL);
gst_afsrc_signals[SIGNAL_HANDOFF] =
g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstAFSrcClass, handoff), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
gobject_class->set_property = gst_afsrc_set_property;
gobject_class->get_property = gst_afsrc_get_property;
gstelement_class->change_state = gst_afsrc_change_state;
}
static void
gst_afsrc_init (GstAFSrc * afsrc)
{
/* no need for a template, caps are set based on file, right ? */
afsrc->srcpad =
gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT
(afsrc), "src"), "src");
gst_element_add_pad (GST_ELEMENT (afsrc), afsrc->srcpad);
gst_pad_use_explicit_caps (afsrc->srcpad);
gst_pad_set_get_function (afsrc->srcpad, gst_afsrc_get);
afsrc->bytes_per_read = 4096;
afsrc->curoffset = 0;
afsrc->seq = 0;
afsrc->filename = NULL;
afsrc->file = NULL;
/* default values, should never be needed */
afsrc->channels = 2;
afsrc->width = 16;
afsrc->rate = 44100;
afsrc->type = AF_FILE_WAVE;
afsrc->endianness_data = 1234;
afsrc->endianness_wanted = 1234;
afsrc->framestamp = 0;
}
static GstData *
gst_afsrc_get (GstPad * pad)
{
GstAFSrc *src;
GstBuffer *buf;
glong readbytes, readframes;
glong frameCount;
g_return_val_if_fail (pad != NULL, NULL);
src = GST_AFSRC (gst_pad_get_parent (pad));
buf = gst_buffer_new ();
g_return_val_if_fail (buf, NULL);
GST_BUFFER_DATA (buf) = (gpointer) g_malloc (src->bytes_per_read);
/* calculate frameCount to read based on file info */
frameCount = src->bytes_per_read / (src->channels * src->width / 8);
/* g_print ("DEBUG: gstafsrc: going to read %ld frames\n", frameCount); */
readframes = afReadFrames (src->file, AF_DEFAULT_TRACK, GST_BUFFER_DATA (buf),
frameCount);
readbytes = readframes * (src->channels * src->width / 8);
if (readbytes == 0) {
gst_element_set_eos (GST_ELEMENT (src));
return GST_DATA (gst_event_new (GST_EVENT_EOS));
}
GST_BUFFER_SIZE (buf) = readbytes;
GST_BUFFER_OFFSET (buf) = src->curoffset;
src->curoffset += readbytes;
src->framestamp += gst_audio_frame_length (src->srcpad, buf);
GST_BUFFER_TIMESTAMP (buf) = src->framestamp * 1E9
/ gst_audio_frame_rate (src->srcpad);
/* printf ("DEBUG: afsrc: timestamp set on output buffer: %f sec\n",
GST_BUFFER_TIMESTAMP (buf) / 1E9); */
/* g_print("DEBUG: gstafsrc: pushed buffer of %ld bytes\n", readbytes); */
return GST_DATA (buf);
}
static void
gst_afsrc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstAFSrc *src;
src = GST_AFSRC (object);
switch (prop_id) {
case ARG_LOCATION:
g_free (src->filename);
src->filename = g_strdup (g_value_get_string (value));
break;
default:
break;
}
}
static void
gst_afsrc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAFSrc *src;
g_return_if_fail (GST_IS_AFSRC (object));
src = GST_AFSRC (object);
switch (prop_id) {
case ARG_LOCATION:
g_value_set_string (value, src->filename);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_afsrc_plugin_init (GstPlugin * plugin)
{
/* load audio support library */
if (!gst_library_load ("gstaudio"))
return FALSE;
if (!gst_element_register (plugin, "afsrc", GST_RANK_NONE, GST_TYPE_AFSRC))
return FALSE;
#ifdef ENABLE_NLS
setlocale (LC_ALL, "");
bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
#endif /* ENABLE_NLS */
return TRUE;
}
/* this is where we open the audiofile */
static gboolean
gst_afsrc_open_file (GstAFSrc * src)
{
g_return_val_if_fail (!GST_OBJECT_FLAG_IS_SET (src, GST_AFSRC_OPEN), FALSE);
/* open the file */
src->file = afOpenFile (src->filename, "r", AF_NULL_FILESETUP);
if (src->file == AF_NULL_FILEHANDLE) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
(_("Could not open file \"%s\" for reading."), src->filename),
("system error: %s", strerror (errno)));
return FALSE;
}
/* get the audiofile audio parameters */
{
int sampleFormat, sampleWidth;
src->channels = afGetChannels (src->file, AF_DEFAULT_TRACK);
afGetSampleFormat (src->file, AF_DEFAULT_TRACK,
&sampleFormat, &sampleWidth);
switch (sampleFormat) {
case AF_SAMPFMT_TWOSCOMP:
src->is_signed = TRUE;
break;
case AF_SAMPFMT_UNSIGNED:
src->is_signed = FALSE;
break;
case AF_SAMPFMT_FLOAT:
case AF_SAMPFMT_DOUBLE:
GST_DEBUG ("ERROR: float data not supported yet !\n");
}
src->rate = (guint) afGetRate (src->file, AF_DEFAULT_TRACK);
src->width = sampleWidth;
GST_DEBUG ("input file: %d channels, %d width, %d rate, signed %s\n",
src->channels, src->width, src->rate, src->is_signed ? "yes" : "no");
}
/* set caps on src */
gst_pad_set_explicit_caps (src->srcpad,
gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, src->is_signed,
"width", G_TYPE_INT, src->width,
"depth", G_TYPE_INT, src->width,
"rate", G_TYPE_INT, src->rate,
"channels", G_TYPE_INT, src->channels, NULL));
GST_OBJECT_FLAG_SET (src, GST_AFSRC_OPEN);
return TRUE;
}
static void
gst_afsrc_close_file (GstAFSrc * src)
{
/* g_print ("DEBUG: closing srcfile...\n"); */
g_return_if_fail (GST_OBJECT_FLAG_IS_SET (src, GST_AFSRC_OPEN));
/* g_print ("DEBUG: past flag test\n"); */
/* if (fclose (src->file) != 0) */
if (afCloseFile (src->file) != 0) {
GST_ELEMENT_ERROR (src, RESOURCE, CLOSE,
(_("Error closing file \"%s\"."), src->filename), GST_ERROR_SYSTEM);
} else {
GST_OBJECT_FLAG_UNSET (src, GST_AFSRC_OPEN);
}
}
static GstStateChangeReturn
gst_afsrc_change_state (GstElement * element, GstStateChange transition)
{
g_return_val_if_fail (GST_IS_AFSRC (element), GST_STATE_CHANGE_FAILURE);
/* if going to NULL then close the file */
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
/* printf ("DEBUG: afsrc state change: null pending\n"); */
if (GST_OBJECT_FLAG_IS_SET (element, GST_AFSRC_OPEN)) {
/* g_print ("DEBUG: trying to close the src file\n"); */
gst_afsrc_close_file (GST_AFSRC (element));
}
} else if (GST_STATE_PENDING (element) == GST_STATE_READY) {
/* g_print ("DEBUG: afsrc: ready state pending. This shouldn't happen at the *end* of a stream\n"); */
if (!GST_OBJECT_FLAG_IS_SET (element, GST_AFSRC_OPEN)) {
/* g_print ("DEBUG: GST_AFSRC_OPEN not set\n"); */
if (!gst_afsrc_open_file (GST_AFSRC (element))) {
/* g_print ("DEBUG: element tries to open file\n"); */
return GST_STATE_CHANGE_FAILURE;
}
}
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}

104
ext/audiofile/gstafsrc.h
View file

@ -1,104 +0,0 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstafsrc.h:
*
* 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.
*/
#ifndef __GST_AFSRC_H__
#define __GST_AFSRC_H__
#include <gst/gst.h>
#include <audiofile.h> /* what else are we to do */
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
#define GST_TYPE_AFSRC \
(gst_afsrc_get_type())
#define GST_AFSRC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFSRC,GstAFSrc))
#define GST_AFSRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFSRC,GstAFSrcClass))
#define GST_IS_AFSRC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFSRC))
#define GST_IS_AFSRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFSRC))
typedef struct _GstAFSrc GstAFSrc;
typedef struct _GstAFSrcClass GstAFSrcClass;
typedef enum {
GST_AFSRC_OPEN = (GST_ELEMENT_FLAG_LAST << 0),
GST_AFSRC_FLAG_LAST = (GST_ELEMENT_FLAG_LAST << 2),
} GstAFSrcFlags;
struct _GstAFSrc {
GstElement element;
GstPad *srcpad;
gchar *filename;
/* FILE *file; */
/* AFfilesetup outfilesetup; */
AFfilehandle file;
int format;
int channels;
int width;
unsigned int rate;
gboolean is_signed;
int type; /* type of output, compare to audiofile.h
* RAW, AIFF, AIFFC, NEXTSND, WAVE
*/
/* blocking */
gulong curoffset;
gulong bytes_per_read;
gulong seq;
guint64 framestamp;
/* FIXME : endianness is a little cryptic at this point */
int endianness_data; /* 4321 or 1234 */
int endianness_wanted; /* same thing, but what the output format wants */
int endianness_output; /* what the output endianness will be */
};
struct _GstAFSrcClass {
GstElementClass parent_class;
/* signals */
void (*handoff) (GstElement *element,GstPad *pad);
};
GType gst_afsrc_get_type (void);
gboolean gst_afsrc_plugin_init (GstPlugin *plugin);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __GST_AFSRC_H__ */

2
po/POTFILES.skip
View file

@ -1,5 +1,3 @@
ext/audiofile/gstafsink.c
ext/audiofile/gstafsrc.c
ext/sndfile/gstsfsink.c ext/sndfile/gstsfsink.c
ext/sndfile/gstsfsrc.c ext/sndfile/gstsfsrc.c
sys/dxr3/dxr3audiosink.c sys/dxr3/dxr3audiosink.c