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fixed some GST_LIBS stuff added audiofile added gst-libs/audio building

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Original commit message from CVS:
fixed some GST_LIBS stuff
added audiofile
added gst-libs/audio building
This commit is contained in:
Thomas Vander Stichele 2001-12-21 11:46:15 +00:00
parent cc33fd24db
commit 3383f7c142
11 changed files with 1231 additions and 14 deletions
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4
Makefile.am
View file

@ -1,3 +1,3 @@
SUBDIRS=sys ext
SUBDIRS=gst sys ext gst-libs
DIST_SUBDIRS=sys ext
DIST_SUBDIRS=gst sys ext gst-libs

2
TODO
View file

@ -5,3 +5,5 @@
it better ;)
* check SDL optimisation flags
* check GST_* in configure.ac, there is too much in it

34
configure.ac
View file

@ -348,6 +348,16 @@ AC_SUBST(X_PRE_LIBS)
AC_SUBST(X_EXTRA_LIBS)
AC_SUBST(X_LIBS)
dnl ==========================================================================
dnl ============================= gst plugins ================================
dnl ==========================================================================
dnl *** sine ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SINE, true)
GST_CHECK_FEATURE(SINE, [sine plugin], sinesrc, [
HAVE_SINE="yes"
])
dnl ==========================================================================
dnl ============================= sys plugins ================================
dnl ==========================================================================
@ -428,11 +438,11 @@ GST_CHECK_FEATURE(ARTSC, [artsd plugins], artsdsink, [
dnl *** audiofile ***
dnl this check uses the GST_CHECK_CONFIGPROG macro
translit(dnm, m, l) AM_CONDITIONAL(USE_LIBAUDIOFILE, true)
GST_CHECK_FEATURE(LIBAUDIOFILE, [audiofile], afsink afsrc, [
translit(dnm, m, l) AC_SUBST(LIBAUDIOFILE_LIBS)
translit(dnm, m, l) AC_SUBST(LIBAUDIOFILE_CFLAGS)
GST_CHECK_CONFIGPROG(LIBAUDIOFILE, audiofile-config)
translit(dnm, m, l) AM_CONDITIONAL(USE_AUDIOFILE, true)
GST_CHECK_FEATURE(AUDIOFILE, [audiofile], afsink afsrc, [
translit(dnm, m, l) AC_SUBST(AUDIOFILE_LIBS)
translit(dnm, m, l) AC_SUBST(AUDIOFILE_CFLAGS)
GST_CHECK_CONFIGPROG(AUDIOFILE, audiofile-config)
])
dnl *** avifile ***
@ -1071,10 +1081,11 @@ AC_SUBST(LIBGST_LIBS)
AC_SUBST(LIBGST_CFLAGS)
dnl Vars for everyone else
GST_LIBS="\$(top_builddir)/gst/libgst.la $LIBGST_LIBS"
GST_CFLAGS="-I\$(top_srcdir) -I\$(top_srcdir)/include $LIBGST_CFLAGS"
AC_SUBST(GST_LIBS)
AC_SUBST(GST_CFLAGS)
dnl FIXME: is there a reason to add this top_builddir stuff ? don't think so
dnl GST_LIBS="\$(top_builddir)/gst/libgst.la $LIBGST_LIBS"
dnl GST_CFLAGS="-I\$(top_srcdir) -I\$(top_srcdir)/include $LIBGST_CFLAGS"
dnl AC_SUBST(GST_LIBS)
dnl AC_SUBST(GST_CFLAGS)
dnl #############################
dnl # Configure the subpackages #
@ -1127,6 +1138,8 @@ dnl stamp.h
dnl echo "$infomessages", infomessages="$infomessages"
AC_OUTPUT(
Makefile
gst/Makefile
gst/sine/Makefile
sys/Makefile
sys/oss/Makefile
sys/qcam/Makefile
@ -1135,11 +1148,14 @@ sys/vcd/Makefile
sys/vga/Makefile
sys/xvideo/Makefile
ext/Makefile
ext/audiofile/Makefile
ext/esd/Makefile
ext/lame/Makefile
ext/mad/Makefile
ext/sdl/Makefile
ext/vorbis/Makefile
gst-libs/Makefile
gst-libs/audio/Makefile
)
echo -e "configure: *** Plugins that will be built : $GST_PLUGINS_YES"

11
ext/Makefile.am
View file

@ -1,3 +1,9 @@
if USE_AUDIOFILE
AUDIOFILE_DIR=audiofile
else
AUDIOFILE_DIR=
endif
if USE_ESD
ESD_DIR=esd
else
@ -29,6 +35,7 @@ VORBIS_DIR=
endif
SUBDIRS=$(ESD_DIR) $(LAME_DIR) $(MAD_DIR) $(SDL_DIR) $(VORBIS_DIR)
SUBDIRS=$(AUDIOFILE_DIR) $(ESD_DIR) $(LAME_DIR) $(MAD_DIR) \
$(SDL_DIR) $(VORBIS_DIR)
DIST_SUBDIRS=esd lame mad sdl vorbis
DIST_SUBDIRS=audiofile esd lame mad sdl vorbis

14
ext/audiofile/Makefile.am Normal file
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@ -0,0 +1,14 @@
plugindir = $(libdir)/gst
plugin_LTLIBRARIES = libafsink.la libafsrc.la
libafsink_la_SOURCES = gstafsink.c
libafsrc_la_SOURCES = gstafsrc.c
noinst_HEADERS = gstafsink.h gstafsrc.h
libafsink_la_LIBADD = $(AUDIOFILE_LIBS)
libafsrc_la_LIBADD = $(AUDIOFILE_LIBS)
libafsink_la_CFLAGS = $(GST_CFLAGS)
libafsrc_la_CFLAGS = $(GST_CFLAGS)

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

508
ext/audiofile/gstafsink.c Normal file
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@ -0,0 +1,508 @@
/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/gst.h>
#include "gstafsink.h"
static GstElementDetails afsink_details = {
"Audiofile Sink",
"Sink",
"Audiofile sink for audio/raw",
VERSION,
"Thomas <thomas@apestaart.org>",
"(C) 2001"
};
/* 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 */
GST_PADTEMPLATE_FACTORY (afsink_sink_factory,
"sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"audiofile_sink",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"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 (4000, 48000), //FIXME
"channels", GST_PROPS_INT_RANGE (1, 2)
)
);
/* 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 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_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,GstBuffer *buf);
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 GstElementStateReturn gst_afsink_change_state (GstElement *element);
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), NULL,
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_class_init (GstAFSinkClass *klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gst_element_install_std_props (
GST_ELEMENT_CLASS (klass),
"location", ARG_LOCATION, G_PARAM_READWRITE,
NULL);
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)); // CHECKME!
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)); // CHECKME
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_PADTEMPLATE_GET (afsink_sink_factory), "sink");
gst_element_add_pad (GST_ELEMENT (afsink), afsink->sinkpad);
gst_pad_set_chain_function (afsink->sinkpad, gst_afsink_chain);
gst_pad_set_event_function (afsink->sinkpad, gst_afsink_handle_event);
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;
/* it's not null if we got it, but it might not be ours */
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));
if (sink->filename)
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;
/* it's not null if we got it, but it might not be ours */
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;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
factory = gst_elementfactory_new ("afsink", GST_TYPE_AFSINK,
&afsink_details);
g_return_val_if_fail (factory != NULL, FALSE);
gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (afsink_sink_factory));
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"afsink",
plugin_init
};
/* this is where we open the audiofile */
static gboolean
gst_afsink_open_file (GstAFSink *sink)
{
AFfilesetup outfilesetup;
GstCaps *caps;
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_FLAG_IS_SET (sink, GST_AFSINK_OPEN), FALSE);
/* open the file */
/* we use audiofile now
sink->file = fopen (sink->filename, "w");
if (sink->file == NULL) {
perror ("open");
gst_element_error (GST_ELEMENT (sink), g_strconcat("opening file \"", sink->filename, "\"", NULL));
return FALSE;
}
*/
/* get the audio parameters */
caps = NULL;
g_return_val_if_fail (GST_IS_PAD (sink->sinkpad), FALSE);
caps = GST_PAD_CAPS (sink->sinkpad);
if (caps == NULL)
{
// FIXME : Please change this to a better warning method !
printf ("WARNING: gstafsink chain : Could not get caps of pad !\n");
}
else
{
sink->channels = gst_caps_get_int (caps, "channels");
sink->width = gst_caps_get_int (caps, "width");
sink->rate = gst_caps_get_int (caps, "rate");
sink->is_signed = gst_caps_get_int (caps, "signed");
sink->endianness_data = gst_caps_get_int (caps, "endianness");
}
GST_DEBUG (GST_CAT_PLUGIN_INFO, "channels %d, width %d, rate %d, signed %s\n",
sink->channels, sink->width, sink->rate,
sink->is_signed ? "yes" : "no");
GST_DEBUG (GST_CAT_PLUGIN_INFO, "endianness: data %d, output %d\n",
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)
{
perror ("open");
gst_element_error (GST_ELEMENT (sink), g_strconcat("opening file \"", sink->filename, "\"", NULL));
return FALSE;
}
afFreeFileSetup (outfilesetup);
// afSetVirtualByteOrder (sink->file, AF_DEFAULT_TRACK, byte_order);
GST_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_FLAG_IS_SET (sink, GST_AFSINK_OPEN));
// g_print ("DEBUG: past flag test\n");
// if (fclose (sink->file) != 0)
if (afCloseFile (sink->file) != 0)
{
g_print ("WARNING: afsink: oops, error closing !\n");
perror ("close");
gst_element_error (GST_ELEMENT (sink), g_strconcat("closing file \"", sink->filename, "\"", NULL));
}
else {
GST_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, GstBuffer *buf)
{
GstAFSink *afsink;
int ret = 0;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
afsink = GST_AFSINK (gst_pad_get_parent (pad));
/* we use audiofile now
if (GST_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_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_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 GstElementStateReturn
gst_afsink_change_state (GstElement *element)
{
g_return_val_if_fail (GST_IS_AFSINK (element), GST_STATE_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_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_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_FAILURE;
}
}
}
*/
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_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 (0, "DEBUG: afsink: got event\n");
gst_afsink_close_file (afsink);
GST_FLAG_SET (pad, GST_PAD_EOS);
return TRUE;
}
/*
gboolean
gst_afsink_factory_init (GstElementFactory *factory)
{
GstPadTemplate *sink_pt;
sink_pt = afsink_sink_factory();
gst_elementfactory_add_padtemplate (factory, sink_pt);
return TRUE;
}
*/

100
ext/audiofile/gstafsink.h Normal file
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@ -0,0 +1,100 @@
/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_AFSINK_H__
#define __GST_AFSINK_H__
#include <config.h>
#include <gst/gst.h>
#include <audiofile.h> /* what else are we to do */
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
//GstElementDetails gst_afsink_details;
#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(obj) \
(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,
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_factory_init (GstElementFactory *factory);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __GST_AFSINK_H__ */

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/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/gst.h>
#include <gst-libs/audio/gstaudio.h>
#include "gstafsrc.h"
static GstElementDetails afsrc_details = {
"Audiofile Src",
"Src",
"Audiofile src for audio/raw",
VERSION,
"Thomas <thomas@apestaart.org>",
"(C) 2001"
};
/* 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 */
GST_PADTEMPLATE_FACTORY (afsrc_src_factory,
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"audiofile_src",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"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 (4000, 48000), //FIXME
"channels", GST_PROPS_INT_RANGE (1, 2)
)
);
/* 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 */
static GType
gst_afsrc_types_get_type (void)
{
static GType afsrc_types_type = 0;
static 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_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 GstBuffer* 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 GstElementStateReturn gst_afsrc_change_state (GstElement *element);
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), NULL,
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_class_init (GstAFSrcClass *klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gst_element_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)
{
// GstPad *pad; this is now done in the struct
/* no need for a template, caps are set based on file, right ? */
afsrc->srcpad = gst_pad_new ("src", GST_PAD_SRC);
gst_element_add_pad (GST_ELEMENT (afsrc), 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 GstBuffer *
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_signal_eos (GST_ELEMENT (src));
return NULL;
}
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 buf;
}
static void
gst_afsrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
{
GstAFSrc *src;
/* it's not null if we got it, but it might not be ours */
src = GST_AFSRC (object);
switch (prop_id) {
case ARG_LOCATION:
if (src->filename)
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;
/* it's not null if we got it, but it might not be ours */
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;
}
}
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
factory = gst_elementfactory_new ("afsrc", GST_TYPE_AFSRC,
&afsrc_details);
g_return_val_if_fail (factory != NULL, FALSE);
gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (afsrc_src_factory));
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
/* load audio support library */
if (!gst_library_load ("gstaudio"))
{
gst_info ("mad: could not load support library: 'gstaudio'\n");
return FALSE;
}
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"afsrc",
plugin_init
};
/* this is where we open the audiofile */
static gboolean
gst_afsrc_open_file (GstAFSrc *src)
{
g_return_val_if_fail (!GST_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)
{
g_print ("ERROR: gstafsrc: Could not open file %s for reading\n",
src->filename);
gst_element_error (GST_ELEMENT (src), g_strconcat ("opening file \"",
src->filename, "\"", NULL));
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 (GST_CAT_PLUGIN_INFO,
"ERROR: float data not supported yet !\n");
}
src->rate = (guint) afGetRate (src->file, AF_DEFAULT_TRACK);
src->width = sampleWidth;
GST_DEBUG (GST_CAT_PLUGIN_INFO,
"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 */
//FIXME: add all the possible formats, especially float ! */
gst_pad_set_caps (src->srcpad, gst_caps_new (
"af_src",
"audio/raw",
gst_props_new (
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0), //FIXME
"endianness", GST_PROPS_INT (G_BYTE_ORDER), //FIXME
"signed", GST_PROPS_BOOLEAN (src->is_signed),
"width", GST_PROPS_INT (src->width),
"depth", GST_PROPS_INT (src->width),
"rate", GST_PROPS_INT (src->rate),
"channels", GST_PROPS_INT (src->channels),
NULL
)
));
GST_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_FLAG_IS_SET (src, GST_AFSRC_OPEN));
// g_print ("DEBUG: past flag test\n");
// if (fclose (src->file) != 0)
if (afCloseFile (src->file) != 0)
{
g_print ("WARNING: afsrc: oops, error closing !\n");
perror ("close");
gst_element_error (GST_ELEMENT (src), g_strconcat("closing file \"", src->filename, "\"", NULL));
}
else {
GST_FLAG_UNSET (src, GST_AFSRC_OPEN);
}
}
static GstElementStateReturn
gst_afsrc_change_state (GstElement *element)
{
g_return_val_if_fail (GST_IS_AFSRC (element), GST_STATE_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_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_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_FAILURE;
}
}
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}

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/* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __GST_AFSRC_H__
#define __GST_AFSRC_H__
#include <config.h>
#include <gst/gst.h>
#include <audiofile.h> /* what else are we to do */
#ifdef __cplusplus
extern "C" {
#endif /* __cplusplus */
//GstElementDetails gst_afsrc_details;
#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(obj) \
(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,
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_factory_init (GstElementFactory *factory);
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif /* __GST_AFSRC_H__ */

2
gst-libs/audio/Makefile.am
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@ -1,4 +1,4 @@
libdir = $(libdir)/gst
## libdir = $(libdir)/gst
lib_LTLIBRARIES = libgstaudio.la