This was never even ported to 0.10.
APEXSINK_DIR =
endif
-# if USE_AUDIOFILE
-# AUDIOFILE_DIR=audiofile
-# else
-AUDIOFILE_DIR=
-# endif
-
if USE_BS2B
BS2B_DIR=bs2b
else
+++ /dev/null
-
-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
+++ /dev/null
-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
+++ /dev/null
-/* 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)
+++ /dev/null
-/* 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;
-}
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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;
-}
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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;
-}
+++ /dev/null
-/* 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__ */
-ext/audiofile/gstafsink.c
-ext/audiofile/gstafsrc.c
ext/sndfile/gstsfsink.c
ext/sndfile/gstsfsrc.c
sys/dxr3/dxr3audiosink.c
projects / platform / upstream / gstreamer.git / commitdiff