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

Original commit message from CVS:
fixed some GST_LIBS stuff
added audiofile
added gst-libs/audio building
parent cc33fd24
SUBDIRS=sys ext
SUBDIRS=gst sys ext gst-libs
DIST_SUBDIRS=sys ext
DIST_SUBDIRS=gst sys ext gst-libs
......@@ -5,3 +5,5 @@
it better ;)
* check SDL optimisation flags
* check GST_* in configure.ac, there is too much in it
......@@ -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 : $GST_PLUGINS_YES"
......
if USE_AUDIOFILE
AUDIOFILE_DIR=audiofile
else
AUDIOFILE_DIR=
endif
if USE_ESD
ESD_DIR=esd
else
......@@ -29,6 +35,7 @@ VORBIS_DIR=
endif
SUBDIRS=$(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
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)
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) 1999,2000 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;
}