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-rw-r--r--Makefile.am4
-rw-r--r--TODO2
-rw-r--r--configure.ac34
-rw-r--r--ext/Makefile.am11
-rw-r--r--ext/audiofile/Makefile.am14
-rw-r--r--ext/audiofile/README39
-rw-r--r--ext/audiofile/gstafsink.c508
-rw-r--r--ext/audiofile/gstafsink.h100
-rw-r--r--ext/audiofile/gstafsrc.c425
-rw-r--r--ext/audiofile/gstafsrc.h106
-rw-r--r--gst-libs/audio/Makefile.am2
11 files changed, 1231 insertions, 14 deletions
diff --git a/Makefile.am b/Makefile.am
index 099534f7..b9e2a8d3 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -1,3 +1,3 @@
-SUBDIRS=sys ext
+SUBDIRS=gst sys ext gst-libs
-DIST_SUBDIRS=sys ext
+DIST_SUBDIRS=gst sys ext gst-libs
diff --git a/TODO b/TODO
index 2a2dfec5..e01502ca 100644
--- a/TODO
+++ b/TODO
@@ -5,3 +5,5 @@
it better ;)
* check SDL optimisation flags
+
+* check GST_* in configure.ac, there is too much in it
diff --git a/configure.ac b/configure.ac
index 70c56260..555735b3 100644
--- a/configure.ac
+++ b/configure.ac
@@ -349,6 +349,16 @@ 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 : $GST_PLUGINS_YES"
diff --git a/ext/Makefile.am b/ext/Makefile.am
index c0ad3282..3d1dd792 100644
--- a/ext/Makefile.am
+++ b/ext/Makefile.am
@@ -1,3 +1,9 @@
+if USE_AUDIOFILE
+AUDIOFILE_DIR=audiofile
+else
+AUDIOFILE_DIR=
+endif
+
if USE_ESD
ESD_DIR=esd
else
@@ -29,6 +35,7 @@ VORBIS_DIR=
endif
-SUBDIRS=$(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
diff --git a/ext/audiofile/Makefile.am b/ext/audiofile/Makefile.am
new file mode 100644
index 00000000..7e1ca5ec
--- /dev/null
+++ b/ext/audiofile/Makefile.am
@@ -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)
+
diff --git a/ext/audiofile/README b/ext/audiofile/README
new file mode 100644
index 00000000..4c52c050
--- /dev/null
+++ b/ext/audiofile/README
@@ -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
diff --git a/ext/audiofile/gstafsink.c b/ext/audiofile/gstafsink.c
new file mode 100644
index 00000000..b4c584c3
--- /dev/null
+++ b/ext/audiofile/gstafsink.c
@@ -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;
+
+}
+*/
+
diff --git a/ext/audiofile/gstafsink.h b/ext/audiofile/gstafsink.h
new file mode 100644
index 00000000..297aed36
--- /dev/null
+++ b/ext/audiofile/gstafsink.h
@@ -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__ */
diff --git a/ext/audiofile/gstafsrc.c b/ext/audiofile/gstafsrc.c
new file mode 100644
index 00000000..ceb548f6
--- /dev/null
+++ b/ext/audiofile/gstafsrc.c
@@ -0,0 +1,425 @@
+/* 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;
+}
diff --git a/ext/audiofile/gstafsrc.h b/ext/audiofile/gstafsrc.h
new file mode 100644
index 00000000..38116e4b
--- /dev/null
+++ b/ext/audiofile/gstafsrc.h
@@ -0,0 +1,106 @@
+/* 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__ */
diff --git a/gst-libs/audio/Makefile.am b/gst-libs/audio/Makefile.am
index f1307e4a..89d1d9e1 100644
--- a/gst-libs/audio/Makefile.am
+++ b/gst-libs/audio/Makefile.am
@@ -1,4 +1,4 @@
-libdir = $(libdir)/gst
+## libdir = $(libdir)/gst
lib_LTLIBRARIES = libgstaudio.la
generated by cgit v1.2.1 (git 2.18.0) at 2024-12-28 02:40:13 +0000