diff options
| -rw-r--r-- | ext/Makefile.am | 6 | ||||
| -rw-r--r-- | ext/audiofile/Makefile.am | 10 | ||||
| -rw-r--r-- | ext/audiofile/README | 39 | ||||
| -rw-r--r-- | ext/audiofile/gstaf.c | 47 | ||||
| -rw-r--r-- | ext/audiofile/gstafparse.c | 515 | ||||
| -rw-r--r-- | ext/audiofile/gstafparse.h | 101 | ||||
| -rw-r--r-- | ext/audiofile/gstafsink.c | 490 | ||||
| -rw-r--r-- | ext/audiofile/gstafsink.h | 97 | ||||
| -rw-r--r-- | ext/audiofile/gstafsrc.c | 396 | ||||
| -rw-r--r-- | ext/audiofile/gstafsrc.h | 104 | ||||
| -rw-r--r-- | po/POTFILES.skip | 2 |
11 files changed, 0 insertions, 1807 deletions
diff --git a/ext/Makefile.am b/ext/Makefile.am index 15f2708a0..f2d9059f6 100644 --- a/ext/Makefile.am +++ b/ext/Makefile.am @@ -16,12 +16,6 @@ else APEXSINK_DIR = endif -# if USE_AUDIOFILE -# AUDIOFILE_DIR=audiofile -# else -AUDIOFILE_DIR= -# endif - if USE_BS2B BS2B_DIR=bs2b else diff --git a/ext/audiofile/Makefile.am b/ext/audiofile/Makefile.am deleted file mode 100644 index 27f7d3d37..000000000 --- a/ext/audiofile/Makefile.am +++ /dev/null @@ -1,10 +0,0 @@ - -plugin_LTLIBRARIES = libgstaudiofile.la - -libgstaudiofile_la_SOURCES = gstaf.c gstafsink.c gstafsrc.c gstafparse.c -libgstaudiofile_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_CFLAGS) $(AUDIOFILE_CFLAGS) -libgstaudiofile_la_LIBADD = $(AUDIOFILE_LIBS) -libgstaudiofile_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -libgstaudiofile_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS) - -noinst_HEADERS = gstafsink.h gstafsrc.h gstafparse.h diff --git a/ext/audiofile/README b/ext/audiofile/README deleted file mode 100644 index 4c52c0507..000000000 --- a/ext/audiofile/README +++ /dev/null @@ -1,39 +0,0 @@ -This plugin wraps the SGI Audiofile -(http://oss.sgi.com/projects/audiofile/) library into a src and sink -element. - -You can read from and write to the supported formats (WAVE, AIFF, AIFFC, -NEXTSND). - -What is supported : -* all the file formats -* integer sample data, both 2's complement and unsigned -* 8 or 16 bit width & depth (haven't tested others) -* sample rate -* some sort of endianness control - -What isn't supported yet : -* float data - -What you can do : -* src element only accepts location argument -* sink element accepts location, endianness and type - - location : file on the system to output - - endianness : at this time endianness is still a bit shady - you can either set 1234 or 4321; - setting it to 4321 will byteswap the buffer data - you might want to keep it at 1234 for now - - type : one of the file types - -Use gstreamer-inspect on afsink and afsrc to see all of the supported -options. - -Examples : - -* tools/gstreamer-launch afsrc location=/opt/media/wav/dark-480-16-m.wav ! afsink type=2 location=/opt/media/wav/dark-480-16-m.aiff - -Future plans : - -* add float support -* wrap up afsink and afsrc with pipe and fork to act like data convertors, - allowing arbitrary choice of sink and src element diff --git a/ext/audiofile/gstaf.c b/ext/audiofile/gstaf.c deleted file mode 100644 index 76d2a5440..000000000 --- a/ext/audiofile/gstaf.c +++ /dev/null @@ -1,47 +0,0 @@ -/* GStreamer - * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu> - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gstafsrc.h" -#include "gstafsink.h" -#include "gstafparse.h" - -gboolean gst_aftypes_plugin_init (GstPlugin * plugin); - -static gboolean -plugin_init (GstPlugin * plugin) -{ - if (!gst_library_load ("gstbytestream")) - return FALSE; - - gst_afsink_plugin_init (plugin); - gst_afsrc_plugin_init (plugin); - gst_afparse_plugin_init (plugin); - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - audiofile, - "Audiofile plugin", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, - GST_PACKAGE_ORIGIN) diff --git a/ext/audiofile/gstafparse.c b/ext/audiofile/gstafparse.c deleted file mode 100644 index 589a12238..000000000 --- a/ext/audiofile/gstafparse.c +++ /dev/null @@ -1,515 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * - * gstafparse.c: - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif -#include <gst/gst.h> -#include <gst/audio/audio.h> -#include <string.h> -#include "gstafparse.h" - -/* AFParse signals and args */ -enum -{ - /* FILL ME */ - SIGNAL_HANDOFF, - LAST_SIGNAL -}; - -enum -{ - ARG_0 -}; - -/* added a src factory function to force audio/raw MIME type */ -static GstStaticPadTemplate afparse_src_factory = -GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) { 8, 16 }, " - "depth = (int) { 8, 16 }, " "signed = (boolean) { true, false }") - ); - -static GstStaticPadTemplate afparse_sink_factory = - GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-aiff; " "audio/x-wav; " "audio/x-au") - ); - -static void gst_afparse_base_init (gpointer g_class); -static void gst_afparse_class_init (GstAFParseClass * klass); -static void gst_afparse_init (GstAFParse * afparse); - -static gboolean gst_afparse_open_file (GstAFParse * afparse); -static void gst_afparse_close_file (GstAFParse * afparse); - -static void gst_afparse_loop (GstElement * element); -static void gst_afparse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_afparse_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static ssize_t gst_afparse_vf_read (AFvirtualfile * vfile, void *data, - size_t nbytes); -static long gst_afparse_vf_length (AFvirtualfile * vfile); -static ssize_t gst_afparse_vf_write (AFvirtualfile * vfile, const void *data, - size_t nbytes); -static void gst_afparse_vf_destroy (AFvirtualfile * vfile); -static long gst_afparse_vf_seek (AFvirtualfile * vfile, long offset, - int is_relative); -static long gst_afparse_vf_tell (AFvirtualfile * vfile); - -GType -gst_afparse_get_type (void) -{ - static GType afparse_type = 0; - - if (!afparse_type) { - static const GTypeInfo afparse_info = { - sizeof (GstAFParseClass), - gst_afparse_base_init, - NULL, - (GClassInitFunc) gst_afparse_class_init, - NULL, - NULL, - sizeof (GstAFParse), - 0, - (GInstanceInitFunc) gst_afparse_init, - }; - - afparse_type = - g_type_register_static (GST_TYPE_ELEMENT, "GstAFParse", &afparse_info, - 0); - } - return afparse_type; -} - -static void -gst_afparse_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&afparse_src_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&afparse_sink_factory)); - - gst_element_class_set_static_metadata (element_class, "Audiofile demuxer", - "Codec/Demuxer/Audio", - "Audiofile parser for audio/raw", - "Steve Baker <stevebaker_org@yahoo.co.uk>"); -} - -static void -gst_afparse_class_init (GstAFParseClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - gobject_class->set_property = gst_afparse_set_property; - gobject_class->get_property = gst_afparse_get_property; - -} - -static void -gst_afparse_init (GstAFParse * afparse) -{ - afparse->srcpad = - gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT - (afparse), "src"), "src"); - gst_pad_use_explicit_caps (afparse->srcpad); - gst_element_add_pad (GST_ELEMENT (afparse), afparse->srcpad); - - afparse->sinkpad = - gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT - (afparse), "sink"), "sink"); - gst_element_add_pad (GST_ELEMENT (afparse), afparse->sinkpad); - - gst_element_set_loop_function (GST_ELEMENT (afparse), gst_afparse_loop); - - afparse->vfile = af_virtual_file_new (); - afparse->vfile->closure = NULL; - afparse->vfile->read = gst_afparse_vf_read; - afparse->vfile->length = gst_afparse_vf_length; - afparse->vfile->write = gst_afparse_vf_write; - afparse->vfile->destroy = gst_afparse_vf_destroy; - afparse->vfile->seek = gst_afparse_vf_seek; - afparse->vfile->tell = gst_afparse_vf_tell; - - afparse->frames_per_read = 1024; - afparse->curoffset = 0; - afparse->seq = 0; - - afparse->file = NULL; - /* default values, should never be needed */ - afparse->channels = 2; - afparse->width = 16; - afparse->rate = 44100; - afparse->type = AF_FILE_WAVE; - afparse->endianness_data = 1234; - afparse->endianness_wanted = 1234; - afparse->timestamp = 0LL; -} - -static void -gst_afparse_loop (GstElement * element) -{ - GstAFParse *afparse; - GstBuffer *buf; - gint numframes = 0, frames_to_bytes, frames_per_read, bytes_per_read; - guint8 *data; - gboolean bypass_afread = TRUE; - GstByteStream *bs; - int s_format, v_format, s_width, v_width; - - afparse = GST_AFPARSE (element); - - afparse->vfile->closure = bs = gst_bytestream_new (afparse->sinkpad); - - /* just stop if we cannot open the file */ - if (!gst_afparse_open_file (afparse)) { - gst_bytestream_destroy ((GstByteStream *) afparse->vfile->closure); - gst_pad_push (afparse->srcpad, GST_DATA (gst_event_new (GST_EVENT_EOS))); - gst_element_set_eos (GST_ELEMENT (afparse)); - return; - } - - /* if audiofile changes the data in any way, we have to access - * the audio data via afReadFrames. Otherwise we can just access - * the data directly. */ - afGetSampleFormat (afparse->file, AF_DEFAULT_TRACK, &s_format, &s_width); - afGetVirtualSampleFormat (afparse->file, AF_DEFAULT_TRACK, &v_format, - &v_width); - if (afGetCompression != AF_COMPRESSION_NONE - || afGetByteOrder (afparse->file, - AF_DEFAULT_TRACK) != afGetVirtualByteOrder (afparse->file, - AF_DEFAULT_TRACK) || s_format != v_format || s_width != v_width) { - bypass_afread = FALSE; - } - - if (bypass_afread) { - GST_DEBUG ("will bypass afReadFrames\n"); - } - - frames_to_bytes = afparse->channels * afparse->width / 8; - frames_per_read = afparse->frames_per_read; - bytes_per_read = frames_per_read * frames_to_bytes; - - afSeekFrame (afparse->file, AF_DEFAULT_TRACK, 0); - - if (bypass_afread) { - GstEvent *event = NULL; - guint32 waiting; - guint32 got_bytes; - - do { - - got_bytes = gst_bytestream_read (bs, &buf, bytes_per_read); - if (got_bytes == 0) { - /* we need to check for an event. */ - gst_bytestream_get_status (bs, &waiting, &event); - if (event && GST_EVENT_TYPE (event) == GST_EVENT_EOS) { - gst_pad_push (afparse->srcpad, - GST_DATA (gst_event_new (GST_EVENT_EOS))); - gst_element_set_eos (GST_ELEMENT (afparse)); - break; - } - } else { - GST_BUFFER_TIMESTAMP (buf) = afparse->timestamp; - gst_pad_push (afparse->srcpad, GST_DATA (buf)); - if (got_bytes != bytes_per_read) { - /* this shouldn't happen very often */ - /* FIXME calculate the timestamps based on the fewer bytes received */ - - } else { - afparse->timestamp += frames_per_read * 1E9 / afparse->rate; - } - } - } - while (TRUE); - - } else { - do { - buf = gst_buffer_new_and_alloc (bytes_per_read); - GST_BUFFER_TIMESTAMP (buf) = afparse->timestamp; - data = GST_BUFFER_DATA (buf); - numframes = - afReadFrames (afparse->file, AF_DEFAULT_TRACK, data, frames_per_read); - - /* events are handled in gst_afparse_vf_read so if there are no - * frames it must be EOS */ - if (numframes < 1) { - gst_buffer_unref (buf); - - gst_pad_push (afparse->srcpad, - GST_DATA (gst_event_new (GST_EVENT_EOS))); - gst_element_set_eos (GST_ELEMENT (afparse)); - break; - } - GST_BUFFER_SIZE (buf) = numframes * frames_to_bytes; - gst_pad_push (afparse->srcpad, GST_DATA (buf)); - afparse->timestamp += numframes * 1E9 / afparse->rate; - } - while (TRUE); - } - gst_afparse_close_file (afparse); - - gst_bytestream_destroy ((GstByteStream *) afparse->vfile->closure); - -} - - -static void -gst_afparse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstAFParse *afparse; - - afparse = GST_AFPARSE (object); - - switch (prop_id) { - default: - break; - } -} - -static void -gst_afparse_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAFParse *afparse; - - g_return_if_fail (GST_IS_AFPARSE (object)); - - afparse = GST_AFPARSE (object); - - switch (prop_id) { - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -gboolean -gst_afparse_plugin_init (GstPlugin * plugin) -{ - /* load audio support library */ - if (!gst_library_load ("gstaudio")) - return FALSE; - - if (!gst_element_register (plugin, "afparse", GST_RANK_NONE, - GST_TYPE_AFPARSE)) - return FALSE; - - return TRUE; -} - -/* this is where we open the audiofile */ -static gboolean -gst_afparse_open_file (GstAFParse * afparse) -{ - g_return_val_if_fail (!GST_OBJECT_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN), - FALSE); - - - /* open the file */ - GST_DEBUG ("opening vfile %p\n", afparse->vfile); - afparse->file = afOpenVirtualFile (afparse->vfile, "r", AF_NULL_FILESETUP); - if (afparse->file == AF_NULL_FILEHANDLE) { - /* this should never happen */ - g_warning ("ERROR: gstafparse: Could not open virtual file for reading\n"); - return FALSE; - } - - GST_DEBUG ("vfile opened\n"); - /* get the audiofile audio parameters */ - { - int sampleFormat, sampleWidth; - - afparse->channels = afGetChannels (afparse->file, AF_DEFAULT_TRACK); - afGetSampleFormat (afparse->file, AF_DEFAULT_TRACK, - &sampleFormat, &sampleWidth); - switch (sampleFormat) { - case AF_SAMPFMT_TWOSCOMP: - afparse->is_signed = TRUE; - break; - case AF_SAMPFMT_UNSIGNED: - afparse->is_signed = FALSE; - break; - case AF_SAMPFMT_FLOAT: - case AF_SAMPFMT_DOUBLE: - GST_DEBUG ("ERROR: float data not supported yet !\n"); - } - afparse->rate = (guint) afGetRate (afparse->file, AF_DEFAULT_TRACK); - afparse->width = sampleWidth; - GST_DEBUG ("input file: %d channels, %d width, %d rate, signed %s\n", - afparse->channels, afparse->width, afparse->rate, - afparse->is_signed ? "yes" : "no"); - } - - /* set caps on src */ - /*FIXME: add all the possible formats, especially float ! */ - gst_pad_set_explicit_caps (afparse->srcpad, - gst_caps_new_simple ("audio/x-raw-int", - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "signed", G_TYPE_BOOLEAN, afparse->is_signed, - "width", G_TYPE_INT, afparse->width, - "depth", G_TYPE_INT, afparse->width, - "rate", G_TYPE_INT, afparse->rate, - "channels", G_TYPE_INT, afparse->channels, NULL)); - - GST_OBJECT_FLAG_SET (afparse, GST_AFPARSE_OPEN); - - return TRUE; -} - -static void -gst_afparse_close_file (GstAFParse * afparse) -{ - g_return_if_fail (GST_OBJECT_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN)); - if (afCloseFile (afparse->file) != 0) { - g_warning ("afparse: oops, error closing !\n"); - } else { - GST_OBJECT_FLAG_UNSET (afparse, GST_AFPARSE_OPEN); - } -} - -static ssize_t -gst_afparse_vf_read (AFvirtualfile * vfile, void *data, size_t nbytes) -{ - GstByteStream *bs = (GstByteStream *) vfile->closure; - guint8 *bytes = NULL; - GstEvent *event = NULL; - guint32 waiting; - guint32 got_bytes; - - /*gchar *debug_str; */ - - got_bytes = gst_bytestream_peek_bytes (bs, &bytes, nbytes); - - while (got_bytes != nbytes) { - /* handle events */ - gst_bytestream_get_status (bs, &waiting, &event); - - /* FIXME this event handling isn't right yet */ - if (!event) { - /*g_print("no event found with %u bytes\n", got_bytes); */ - return 0; - } - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - return 0; - case GST_EVENT_FLUSH: - GST_DEBUG ("flush"); - break; - case GST_EVENT_DISCONTINUOUS: - GST_DEBUG ("seek done"); - got_bytes = gst_bytestream_peek_bytes (bs, &bytes, nbytes); - break; - default: - g_warning ("unknown event %d", GST_EVENT_TYPE (event)); - got_bytes = gst_bytestream_peek_bytes (bs, &bytes, nbytes); - } - } - - memcpy (data, bytes, got_bytes); - gst_bytestream_flush_fast (bs, got_bytes); - - /* debug_str = g_strndup((gchar*)bytes, got_bytes); - g_print("read %u bytes: %s\n", got_bytes, debug_str); - */ - return got_bytes; -} - -static long -gst_afparse_vf_seek (AFvirtualfile * vfile, long offset, int is_relative) -{ - GstByteStream *bs = (GstByteStream *) vfile->closure; - GstSeekType method; - guint64 current_offset = gst_bytestream_tell (bs); - - if (!is_relative) { - if ((guint64) offset == current_offset) { - /* this seems to happen before every read - bad audiofile */ - return offset; - } - - method = GST_SEEK_METHOD_SET; - } else { - if (offset == 0) - return current_offset; - method = GST_SEEK_METHOD_CUR; - } - - if (gst_bytestream_seek (bs, (gint64) offset, method)) { - GST_DEBUG ("doing seek to %d", (gint) offset); - return offset; - } - return 0; -} - -static long -gst_afparse_vf_length (AFvirtualfile * vfile) -{ - GstByteStream *bs = (GstByteStream *) vfile->closure; - guint64 length; - - length = gst_bytestream_length (bs); - GST_DEBUG ("doing length: %" G_GUINT64_FORMAT, length); - return length; -} - -static ssize_t -gst_afparse_vf_write (AFvirtualfile * vfile, const void *data, size_t nbytes) -{ - /* GstByteStream *bs = (GstByteStream*)vfile->closure; */ - g_warning ("shouldn't write to a readonly pad"); - return 0; -} - -static void -gst_afparse_vf_destroy (AFvirtualfile * vfile) -{ - /* GstByteStream *bs = (GstByteStream*)vfile->closure; */ - - GST_DEBUG ("doing destroy"); -} - -static long -gst_afparse_vf_tell (AFvirtualfile * vfile) -{ - GstByteStream *bs = (GstByteStream *) vfile->closure; - guint64 offset; - - offset = gst_bytestream_tell (bs); - GST_DEBUG ("doing tell: %" G_GUINT64_FORMAT, offset); - return offset; -} diff --git a/ext/audiofile/gstafparse.h b/ext/audiofile/gstafparse.h deleted file mode 100644 index 8bc04e0bd..000000000 --- a/ext/audiofile/gstafparse.h +++ /dev/null @@ -1,101 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * - * gstafparse.h: - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - - -#ifndef __GST_AFPARSE_H__ -#define __GST_AFPARSE_H__ - - -#include <gst/gst.h> -#include <gst/bytestream/bytestream.h> -#include <audiofile.h> /* what else are we to do */ -#include <af_vfs.h> - - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - - -#define GST_TYPE_AFPARSE \ - (gst_afparse_get_type()) -#define GST_AFPARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFPARSE,GstAFParse)) -#define GST_AFPARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFPARSE,GstAFParseClass)) -#define GST_IS_AFPARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFPARSE)) -#define GST_IS_AFPARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFPARSE)) - -typedef struct _GstAFParse GstAFParse; -typedef struct _GstAFParseClass GstAFParseClass; - -typedef enum { - GST_AFPARSE_OPEN = (GST_ELEMENT_FLAG_LAST << 0), - - GST_AFPARSE_FLAG_LAST = (GST_ELEMENT_FLAG_LAST << 2), -} GstAFParseFlags; - -struct _GstAFParse { - GstElement element; - GstPad *srcpad; - GstPad *sinkpad; - - AFvirtualfile *vfile; - AFfilehandle file; - int format; - int channels; - int width; - unsigned int rate; - gboolean is_signed; - int type; /* type of output, compare to audiofile.h - * RAW, AIFF, AIFFC, NEXTSND, WAVE - */ - /* blocking */ - gulong curoffset; - gulong bytes_per_read; - gint frames_per_read; - - gulong seq; - gint64 timestamp; - /* FIXME : endianness is a little cryptic at this point */ - int endianness_data; /* 4321 or 1234 */ - int endianness_wanted; /* same thing, but what the output format wants */ - int endianness_output; /* what the output endianness will be */ -}; - -struct _GstAFParseClass { - GstElementClass parent_class; - - /* signals */ - void (*handoff) (GstElement *element,GstPad *pad); -}; - -gboolean gst_afparse_plugin_init (GstPlugin *plugin); - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - - -#endif /* __GST_AFPARSE_H__ */ diff --git a/ext/audiofile/gstafsink.c b/ext/audiofile/gstafsink.c deleted file mode 100644 index 6fb39842e..000000000 --- a/ext/audiofile/gstafsink.c +++ /dev/null @@ -1,490 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * - * gstafsink.c: - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gst/gst-i18n-plugin.h" - -#include <gst/gst.h> -#include <string.h> -#include <errno.h> - -#include "gstafsink.h" - -/* AFSink signals and args */ -enum -{ - /* FILL ME */ - SIGNAL_HANDOFF, - LAST_SIGNAL -}; - -enum -{ - ARG_0, - ARG_TYPE, - ARG_OUTPUT_ENDIANNESS, - ARG_LOCATION -}; - -/* added a sink factory function to force audio/raw MIME type */ -/* I think the caps can be broader, we need to change that somehow */ -static GstStaticPadTemplate afsink_sink_factory = -GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, 2 ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) { 8, 16 }, " - "depth = (int) { 8, 16 }, " "signed = (boolean) { true, false }") - ); - -/* we use an enum for the output type arg */ - -#define GST_TYPE_AFSINK_TYPES (gst_afsink_types_get_type()) -/* FIXME: fix the string ints to be string-converted from the audiofile.h types */ -static GType -gst_afsink_types_get_type (void) -{ - static GType afsink_types_type = 0; - static const GEnumValue afsink_types[] = { - {AF_FILE_RAWDATA, "0", "raw PCM"}, - {AF_FILE_AIFFC, "1", "AIFFC"}, - {AF_FILE_AIFF, "2", "AIFF"}, - {AF_FILE_NEXTSND, "3", "Next/SND"}, - {AF_FILE_WAVE, "4", "Wave"}, - {0, NULL, NULL}, - }; - - if (!afsink_types_type) { - afsink_types_type = - g_enum_register_static ("GstAudiosinkTypes", afsink_types); - } - return afsink_types_type; -} - -static void gst_afsink_base_init (gpointer g_class); -static void gst_afsink_class_init (GstAFSinkClass * klass); -static void gst_afsink_init (GstAFSink * afsink); - -static gboolean gst_afsink_open_file (GstAFSink * sink); -static void gst_afsink_close_file (GstAFSink * sink); - -static void gst_afsink_chain (GstPad * pad, GstData * _data); - -static void gst_afsink_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_afsink_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static gboolean gst_afsink_handle_event (GstPad * pad, GstEvent * event); - -static GstStateChangeReturn gst_afsink_change_state (GstElement * element, - GstStateChange transition); - -static GstElementClass *parent_class = NULL; -static guint gst_afsink_signals[LAST_SIGNAL] = { 0 }; - -GType -gst_afsink_get_type (void) -{ - static GType afsink_type = 0; - - if (!afsink_type) { - static const GTypeInfo afsink_info = { - sizeof (GstAFSinkClass), - gst_afsink_base_init, - NULL, - (GClassInitFunc) gst_afsink_class_init, - NULL, - NULL, - sizeof (GstAFSink), - 0, - (GInstanceInitFunc) gst_afsink_init, - }; - - afsink_type = - g_type_register_static (GST_TYPE_ELEMENT, "GstAFSink", &afsink_info, 0); - } - return afsink_type; -} - -static void -gst_afsink_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&afsink_sink_factory)); - gst_element_class_set_static_metadata (element_class, "Audiofile sink", - "Sink/Audio", - "Write audio streams to disk using libaudiofile", - "Thomas Vander Stichele <thomas@apestaart.org>"); -} - -static void -gst_afsink_class_init (GstAFSinkClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - parent_class = g_type_class_peek_parent (klass); - - gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass), - "location", ARG_LOCATION, G_PARAM_READWRITE, NULL); - - /* FIXME: add long property descriptions */ - g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TYPE, - g_param_spec_enum ("type", "type", "type", GST_TYPE_AFSINK_TYPES, 0, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - g_object_class_install_property (G_OBJECT_CLASS (klass), - ARG_OUTPUT_ENDIANNESS, g_param_spec_int ("endianness", "endianness", - "endianness", G_MININT, G_MAXINT, 0, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - gst_afsink_signals[SIGNAL_HANDOFF] = - g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, - G_STRUCT_OFFSET (GstAFSinkClass, handoff), NULL, NULL, - g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0); - - - gobject_class->set_property = gst_afsink_set_property; - gobject_class->get_property = gst_afsink_get_property; - - gstelement_class->change_state = gst_afsink_change_state; -} - -static void -gst_afsink_init (GstAFSink * afsink) -{ - /* GstPad *pad; this is now done in the struct */ - - afsink->sinkpad = - gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT - (afsink), "sink"), "sink"); - gst_element_add_pad (GST_ELEMENT (afsink), afsink->sinkpad); - - gst_pad_set_chain_function (afsink->sinkpad, gst_afsink_chain); - - afsink->filename = NULL; - afsink->file = NULL; - /* default values, should never be needed */ - afsink->channels = 2; - afsink->width = 16; - afsink->rate = 44100; - afsink->type = AF_FILE_WAVE; - afsink->endianness_data = 1234; - afsink->endianness_wanted = 1234; -} - -static void -gst_afsink_set_property (GObject * object, guint prop_id, const GValue * value, - GParamSpec * pspec) -{ - GstAFSink *sink; - - sink = GST_AFSINK (object); - - switch (prop_id) { - case ARG_LOCATION: - /* the element must be stopped or paused in order to do this */ - g_return_if_fail ((GST_STATE (sink) < GST_STATE_PLAYING) - || (GST_STATE (sink) == GST_STATE_PAUSED)); - g_free (sink->filename); - sink->filename = g_strdup (g_value_get_string (value)); - if ((GST_STATE (sink) == GST_STATE_PAUSED) - && (sink->filename != NULL)) { - gst_afsink_close_file (sink); - gst_afsink_open_file (sink); - } - - break; - case ARG_TYPE: - sink->type = g_value_get_enum (value); - break; - case ARG_OUTPUT_ENDIANNESS: - { - int end = g_value_get_int (value); - - if (end == 1234 || end == 4321) - sink->endianness_output = end; - } - break; - default: - break; - } -} - -static void -gst_afsink_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstAFSink *sink; - - g_return_if_fail (GST_IS_AFSINK (object)); - - sink = GST_AFSINK (object); - - switch (prop_id) { - case ARG_LOCATION: - g_value_set_string (value, sink->filename); - break; - case ARG_TYPE: - g_value_set_enum (value, sink->type); - break; - case ARG_OUTPUT_ENDIANNESS: - g_value_set_int (value, sink->endianness_output); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -gboolean -gst_afsink_plugin_init (GstPlugin * plugin) -{ - if (!gst_element_register (plugin, "afsink", GST_RANK_NONE, GST_TYPE_AFSINK)) - return FALSE; -#ifdef ENABLE_NLS - setlocale (LC_ALL, ""); - bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR); -#endif /* ENABLE_NLS */ - - return TRUE; -} - -/* this is where we open the audiofile */ -static gboolean -gst_afsink_open_file (GstAFSink * sink) -{ - AFfilesetup outfilesetup; - const GstCaps *caps; - GstStructure *structure; - int sample_format; /* audiofile's sample format, look in audiofile.h */ - int byte_order = 0; /* audiofile's byte order defines */ - - g_return_val_if_fail (!GST_OBJECT_FLAG_IS_SET (sink, GST_AFSINK_OPEN), FALSE); - - /* get the audio parameters */ - g_return_val_if_fail (GST_IS_PAD (sink->sinkpad), FALSE); - caps = GST_PAD_CAPS (sink->sinkpad); - - if (caps == NULL) { - g_critical ("gstafsink chain : Could not get caps of pad !\n"); - } else { - structure = gst_caps_get_structure (caps, 0); - gst_structure_get_int (structure, "channels", &sink->channels); - gst_structure_get_int (structure, "width", &sink->width); - gst_structure_get_int (structure, "rate", &sink->rate); - gst_structure_get_boolean (structure, "signed", &sink->is_signed); - gst_structure_get_int (structure, "endianness", &sink->endianness_data); - } - GST_DEBUG ("channels %d, width %d, rate %d, signed %s", - sink->channels, sink->width, sink->rate, sink->is_signed ? "yes" : "no"); - GST_DEBUG ("endianness: data %d, output %d", - sink->endianness_data, sink->endianness_output); - /* setup the output file */ - if (sink->is_signed) - sample_format = AF_SAMPFMT_TWOSCOMP; - else - sample_format = AF_SAMPFMT_UNSIGNED; - /* FIXME : this check didn't seem to work, so let the output endianness be set */ - /* - if (sink->endianness_data == sink->endianness_wanted) - byte_order = AF_BYTEORDER_LITTLEENDIAN; - else - byte_order = AF_BYTEORDER_BIGENDIAN; - */ - if (sink->endianness_output == 1234) - byte_order = AF_BYTEORDER_LITTLEENDIAN; - else - byte_order = AF_BYTEORDER_BIGENDIAN; - - outfilesetup = afNewFileSetup (); - afInitFileFormat (outfilesetup, sink->type); - afInitChannels (outfilesetup, AF_DEFAULT_TRACK, sink->channels); - afInitRate (outfilesetup, AF_DEFAULT_TRACK, sink->rate); - afInitSampleFormat (outfilesetup, AF_DEFAULT_TRACK, - sample_format, sink->width); - - /* open it */ - sink->file = afOpenFile (sink->filename, "w", outfilesetup); - if (sink->file == AF_NULL_FILEHANDLE) { - GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE, - (_("Could not open file \"%s\" for writing."), sink->filename), - ("system error: %s", strerror (errno))); - return FALSE; - } - - afFreeFileSetup (outfilesetup); -/* afSetVirtualByteOrder (sink->file, AF_DEFAULT_TRACK, byte_order); */ - - GST_OBJECT_FLAG_SET (sink, GST_AFSINK_OPEN); - - return TRUE; -} - -static void -gst_afsink_close_file (GstAFSink * sink) -{ -/* g_print ("DEBUG: closing sinkfile...\n"); */ - g_return_if_fail (GST_OBJECT_FLAG_IS_SET (sink, GST_AFSINK_OPEN)); -/* g_print ("DEBUG: past flag test\n"); */ -/* if (fclose (sink->file) != 0) */ - if (afCloseFile (sink->file) != 0) { - GST_ELEMENT_ERROR (sink, RESOURCE, CLOSE, - (_("Error closing file \"%s\"."), sink->filename), GST_ERROR_SYSTEM); - } else { - GST_OBJECT_FLAG_UNSET (sink, GST_AFSINK_OPEN); - } -} - -/** - * gst_afsink_chain: - * @pad: the pad this afsink is connected to - * @buf: the buffer that has to be absorbed - * - * take the buffer from the pad and write to file if it's open - */ -static void -gst_afsink_chain (GstPad * pad, GstData * _data) -{ - GstBuffer *buf; - GstAFSink *afsink; - int ret = 0; - - g_return_if_fail (pad != NULL); - g_return_if_fail (GST_IS_PAD (pad)); - - if (GST_IS_EVENT (_data)) { - gst_afsink_handle_event (pad, GST_EVENT (_data)); - return; - } - - buf = GST_BUFFER (_data); - afsink = GST_AFSINK (gst_pad_get_parent (pad)); -/* we use audiofile now - if (GST_OBJECT_FLAG_IS_SET (afsink, GST_AFSINK_OPEN)) - { - bytes_written = fwrite (GST_BUFFER_DATA (buf), 1, GST_BUFFER_SIZE (buf), afsink->file); - if (bytes_written < GST_BUFFER_SIZE (buf)) - { - printf ("afsink : Warning : %d bytes should be written, only %d bytes written\n", - GST_BUFFER_SIZE (buf), bytes_written); - } - } -*/ - - if (!GST_OBJECT_FLAG_IS_SET (afsink, GST_AFSINK_OPEN)) { - /* it's not open yet, open it */ - if (!gst_afsink_open_file (afsink)) - g_print ("WARNING: gstafsink: can't open file !\n"); -/* return FALSE; Can't return value */ - } - - if (GST_OBJECT_FLAG_IS_SET (afsink, GST_AFSINK_OPEN)) { - int frameCount = 0; - - frameCount = - GST_BUFFER_SIZE (buf) / ((afsink->width / 8) * afsink->channels); - /* g_print ("DEBUG: writing %d frames ", frameCount); */ - ret = afWriteFrames (afsink->file, AF_DEFAULT_TRACK, - GST_BUFFER_DATA (buf), frameCount); - if (ret == AF_BAD_WRITE || ret == AF_BAD_LSEEK) { - printf ("afsink : Warning : afWriteFrames returned an error (%d)\n", ret); - } - } - - gst_buffer_unref (buf); - - g_signal_emit (G_OBJECT (afsink), gst_afsink_signals[SIGNAL_HANDOFF], 0); -} - -static GstStateChangeReturn -gst_afsink_change_state (GstElement * element, GstStateChange transition) -{ - g_return_val_if_fail (GST_IS_AFSINK (element), GST_STATE_CHANGE_FAILURE); - - /* if going to NULL? then close the file */ - if (GST_STATE_PENDING (element) == GST_STATE_NULL) { -/* printf ("DEBUG: afsink state change: null pending\n"); */ - if (GST_OBJECT_FLAG_IS_SET (element, GST_AFSINK_OPEN)) { -/* g_print ("DEBUG: trying to close the sink file\n"); */ - gst_afsink_close_file (GST_AFSINK (element)); - } - } -/* - - else - this has been moved to the chain function, since it's only then that - the caps are set and can be known - { - g_print ("DEBUG: it's not going to null\n"); - if (!GST_OBJECT_FLAG_IS_SET (element, GST_AFSINK_OPEN)) - { - g_print ("DEBUG: GST_AFSINK_OPEN not set\n"); - if (!gst_afsink_open_file (GST_AFSINK (element))) - { - g_print ("DEBUG: element tries to open file\n"); - return GST_STATE_CHANGE_FAILURE; - } - } - } -*/ - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - return GST_STATE_CHANGE_SUCCESS; -} - -/* this function was copied from sinesrc */ - -static gboolean -gst_afsink_handle_event (GstPad * pad, GstEvent * event) -{ - GstAFSink *afsink; - - afsink = GST_AFSINK (gst_pad_get_parent (pad)); - GST_DEBUG ("DEBUG: afsink: got event"); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_EOS: - gst_afsink_close_file (afsink); - break; - default: - break; - } - - gst_pad_event_default (pad, event); - - return TRUE; -} diff --git a/ext/audiofile/gstafsink.h b/ext/audiofile/gstafsink.h deleted file mode 100644 index 06800f1fd..000000000 --- a/ext/audiofile/gstafsink.h +++ /dev/null @@ -1,97 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * - * gstafsink.h: - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - - -#ifndef __GST_AFSINK_H__ -#define __GST_AFSINK_H__ - - -#include <gst/gst.h> -#include <audiofile.h> /* what else are we to do */ - - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - - -#define GST_TYPE_AFSINK \ - (gst_afsink_get_type()) -#define GST_AFSINK(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFSINK,GstAFSink)) -#define GST_AFSINK_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFSINK,GstAFSinkClass)) -#define GST_IS_AFSINK(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFSINK)) -#define GST_IS_AFSINK_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFSINK)) - -typedef struct _GstAFSink GstAFSink; -typedef struct _GstAFSinkClass GstAFSinkClass; - -typedef enum { - GST_AFSINK_OPEN = (GST_ELEMENT_FLAG_LAST << 0), - - GST_AFSINK_FLAG_LAST = (GST_ELEMENT_FLAG_LAST << 2), -} GstAFSinkFlags; - -struct _GstAFSink { - GstElement element; - GstPad *sinkpad; - - gchar *filename; -/* FILE *file; */ - -/* AFfilesetup outfilesetup; */ - AFfilehandle file; - int format; - int channels; - int width; - unsigned int rate; - gboolean is_signed; - int type; /* type of output, compare to audiofile.h - * RAW, AIFF, AIFFC, NEXTSND, WAVE - */ - /* FIXME : endianness is a little cryptic at this point */ - int endianness_data; /* 4321 or 1234 */ - int endianness_wanted; /* same thing, but what the output format wants */ - int endianness_output; /* what the output endianness will be */ -}; - -struct _GstAFSinkClass { - GstElementClass parent_class; - - /* signals */ - void (*handoff) (GstElement *element,GstPad *pad); -}; - -GType gst_afsink_get_type (void); -gboolean gst_afsink_plugin_init (GstPlugin *plugin); - - - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - - -#endif /* __GST_AFSINK_H__ */ diff --git a/ext/audiofile/gstafsrc.c b/ext/audiofile/gstafsrc.c deleted file mode 100644 index ce4db3963..000000000 --- a/ext/audiofile/gstafsrc.c +++ /dev/null @@ -1,396 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * - * gstafsrc.c: - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gst/gst-i18n-plugin.h" -#include <gst/gst.h> -#include <gst/audio/audio.h> -#include <string.h> -#include <errno.h> - -#include "gstafsrc.h" - -/* AFSrc signals and args */ -enum -{ - /* FILL ME */ - SIGNAL_HANDOFF, - LAST_SIGNAL -}; - -enum -{ - ARG_0, - ARG_LOCATION -}; - -/* added a src factory function to force audio/raw MIME type */ -/* I think the caps can be broader, we need to change that somehow */ -static GstStaticPadTemplate afsrc_src_factory = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) { 8, 16 }, " - "depth = (int) { 8, 16 }, " "signed = (boolean) { true, false }") - ); - -/* we use an enum for the output type arg */ - -#define GST_TYPE_AFSRC_TYPES (gst_afsrc_types_get_type()) - -/* FIXME: fix the string ints to be string-converted from the audiofile.h types */ -/* defined but not used -static GType -gst_afsrc_types_get_type (void) -{ - static GType afsrc_types_type = 0; - static const GEnumValue afsrc_types[] = { - {AF_FILE_RAWDATA, "0", "raw PCM"}, - {AF_FILE_AIFFC, "1", "AIFFC"}, - {AF_FILE_AIFF, "2", "AIFF"}, - {AF_FILE_NEXTSND, "3", "Next/SND"}, - {AF_FILE_WAVE, "4", "Wave"}, - {0, NULL, NULL}, - }; - - if (!afsrc_types_type) - { - afsrc_types_type = g_enum_register_static ("GstAudiosrcTypes", afsrc_types); - } - return afsrc_types_type; -} -*/ -static void gst_afsrc_base_init (gpointer g_class); -static void gst_afsrc_class_init (GstAFSrcClass * klass); -static void gst_afsrc_init (GstAFSrc * afsrc); - -static gboolean gst_afsrc_open_file (GstAFSrc * src); -static void gst_afsrc_close_file (GstAFSrc * src); - -static GstData *gst_afsrc_get (GstPad * pad); - -static void gst_afsrc_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_afsrc_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static GstStateChangeReturn gst_afsrc_change_state (GstElement * element, - GstStateChange transition); - -static GstElementClass *parent_class = NULL; -static guint gst_afsrc_signals[LAST_SIGNAL] = { 0 }; - -GType -gst_afsrc_get_type (void) -{ - static GType afsrc_type = 0; - - if (!afsrc_type) { - static const GTypeInfo afsrc_info = { - sizeof (GstAFSrcClass), - gst_afsrc_base_init, - NULL, - (GClassInitFunc) gst_afsrc_class_init, - NULL, - NULL, - sizeof (GstAFSrc), - 0, - (GInstanceInitFunc) gst_afsrc_init, - }; - - afsrc_type = - g_type_register_static (GST_TYPE_ELEMENT, "GstAFSrc", &afsrc_info, 0); - } - return afsrc_type; -} - -static void -gst_afsrc_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&afsrc_src_factory)); - gst_element_class_set_static_metadata (element_class, "Audiofile source", - "Source/Audio", - "Read audio files from disk using libaudiofile", - "Thomas <thomas@apestaart.org>"); -} - -static void -gst_afsrc_class_init (GstAFSrcClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - parent_class = g_type_class_peek_parent (klass); - - gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass), - "location", ARG_LOCATION, G_PARAM_READWRITE, NULL); - - gst_afsrc_signals[SIGNAL_HANDOFF] = - g_signal_new ("handoff", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, - G_STRUCT_OFFSET (GstAFSrcClass, handoff), NULL, NULL, - g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0); - - - gobject_class->set_property = gst_afsrc_set_property; - gobject_class->get_property = gst_afsrc_get_property; - - gstelement_class->change_state = gst_afsrc_change_state; -} - -static void -gst_afsrc_init (GstAFSrc * afsrc) -{ - /* no need for a template, caps are set based on file, right ? */ - afsrc->srcpad = - gst_pad_new_from_template (gst_element_get_pad_template (GST_ELEMENT - (afsrc), "src"), "src"); - gst_element_add_pad (GST_ELEMENT (afsrc), afsrc->srcpad); - gst_pad_use_explicit_caps (afsrc->srcpad); - gst_pad_set_get_function (afsrc->srcpad, gst_afsrc_get); - - afsrc->bytes_per_read = 4096; - afsrc->curoffset = 0; - afsrc->seq = 0; - - afsrc->filename = NULL; - afsrc->file = NULL; - /* default values, should never be needed */ - afsrc->channels = 2; - afsrc->width = 16; - afsrc->rate = 44100; - afsrc->type = AF_FILE_WAVE; - afsrc->endianness_data = 1234; - afsrc->endianness_wanted = 1234; - afsrc->framestamp = 0; -} - -static GstData * -gst_afsrc_get (GstPad * pad) -{ - GstAFSrc *src; - GstBuffer *buf; - - glong readbytes, readframes; - glong frameCount; - - g_return_val_if_fail (pad != NULL, NULL); - src = GST_AFSRC (gst_pad_get_parent (pad)); - - buf = gst_buffer_new (); - g_return_val_if_fail (buf, NULL); - - GST_BUFFER_DATA (buf) = (gpointer) g_malloc (src->bytes_per_read); - - /* calculate frameCount to read based on file info */ - - frameCount = src->bytes_per_read / (src->channels * src->width / 8); -/* g_print ("DEBUG: gstafsrc: going to read %ld frames\n", frameCount); */ - readframes = afReadFrames (src->file, AF_DEFAULT_TRACK, GST_BUFFER_DATA (buf), - frameCount); - readbytes = readframes * (src->channels * src->width / 8); - if (readbytes == 0) { - gst_element_set_eos (GST_ELEMENT (src)); - return GST_DATA (gst_event_new (GST_EVENT_EOS)); - } - - GST_BUFFER_SIZE (buf) = readbytes; - GST_BUFFER_OFFSET (buf) = src->curoffset; - - src->curoffset += readbytes; - - src->framestamp += gst_audio_frame_length (src->srcpad, buf); - GST_BUFFER_TIMESTAMP (buf) = src->framestamp * 1E9 - / gst_audio_frame_rate (src->srcpad); - /* printf ("DEBUG: afsrc: timestamp set on output buffer: %f sec\n", - GST_BUFFER_TIMESTAMP (buf) / 1E9); */ - -/* g_print("DEBUG: gstafsrc: pushed buffer of %ld bytes\n", readbytes); */ - return GST_DATA (buf); -} - -static void -gst_afsrc_set_property (GObject * object, guint prop_id, const GValue * value, - GParamSpec * pspec) -{ - GstAFSrc *src; - - src = GST_AFSRC (object); - - switch (prop_id) { - case ARG_LOCATION: - g_free (src->filename); - src->filename = g_strdup (g_value_get_string (value)); - break; - default: - break; - } -} - -static void -gst_afsrc_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstAFSrc *src; - - g_return_if_fail (GST_IS_AFSRC (object)); - - src = GST_AFSRC (object); - - switch (prop_id) { - case ARG_LOCATION: - g_value_set_string (value, src->filename); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -gboolean -gst_afsrc_plugin_init (GstPlugin * plugin) -{ - /* load audio support library */ - if (!gst_library_load ("gstaudio")) - return FALSE; - - if (!gst_element_register (plugin, "afsrc", GST_RANK_NONE, GST_TYPE_AFSRC)) - return FALSE; - -#ifdef ENABLE_NLS - setlocale (LC_ALL, ""); - bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR); -#endif /* ENABLE_NLS */ - - return TRUE; -} - - -/* this is where we open the audiofile */ -static gboolean -gst_afsrc_open_file (GstAFSrc * src) -{ - g_return_val_if_fail (!GST_OBJECT_FLAG_IS_SET (src, GST_AFSRC_OPEN), FALSE); - - /* open the file */ - src->file = afOpenFile (src->filename, "r", AF_NULL_FILESETUP); - if (src->file == AF_NULL_FILEHANDLE) { - GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, - (_("Could not open file \"%s\" for reading."), src->filename), - ("system error: %s", strerror (errno))); - return FALSE; - } - - /* get the audiofile audio parameters */ - { - int sampleFormat, sampleWidth; - - src->channels = afGetChannels (src->file, AF_DEFAULT_TRACK); - afGetSampleFormat (src->file, AF_DEFAULT_TRACK, - &sampleFormat, &sampleWidth); - switch (sampleFormat) { - case AF_SAMPFMT_TWOSCOMP: - src->is_signed = TRUE; - break; - case AF_SAMPFMT_UNSIGNED: - src->is_signed = FALSE; - break; - case AF_SAMPFMT_FLOAT: - case AF_SAMPFMT_DOUBLE: - GST_DEBUG ("ERROR: float data not supported yet !\n"); - } - src->rate = (guint) afGetRate (src->file, AF_DEFAULT_TRACK); - src->width = sampleWidth; - GST_DEBUG ("input file: %d channels, %d width, %d rate, signed %s\n", - src->channels, src->width, src->rate, src->is_signed ? "yes" : "no"); - } - - /* set caps on src */ - gst_pad_set_explicit_caps (src->srcpad, - gst_caps_new_simple ("audio/x-raw-int", - "endianness", G_TYPE_INT, G_BYTE_ORDER, - "signed", G_TYPE_BOOLEAN, src->is_signed, - "width", G_TYPE_INT, src->width, - "depth", G_TYPE_INT, src->width, - "rate", G_TYPE_INT, src->rate, - "channels", G_TYPE_INT, src->channels, NULL)); - - GST_OBJECT_FLAG_SET (src, GST_AFSRC_OPEN); - - return TRUE; -} - -static void -gst_afsrc_close_file (GstAFSrc * src) -{ -/* g_print ("DEBUG: closing srcfile...\n"); */ - g_return_if_fail (GST_OBJECT_FLAG_IS_SET (src, GST_AFSRC_OPEN)); -/* g_print ("DEBUG: past flag test\n"); */ -/* if (fclose (src->file) != 0) */ - if (afCloseFile (src->file) != 0) { - GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, - (_("Error closing file \"%s\"."), src->filename), GST_ERROR_SYSTEM); - } else { - GST_OBJECT_FLAG_UNSET (src, GST_AFSRC_OPEN); - } -} - -static GstStateChangeReturn -gst_afsrc_change_state (GstElement * element, GstStateChange transition) -{ - g_return_val_if_fail (GST_IS_AFSRC (element), GST_STATE_CHANGE_FAILURE); - - /* if going to NULL then close the file */ - if (GST_STATE_PENDING (element) == GST_STATE_NULL) { -/* printf ("DEBUG: afsrc state change: null pending\n"); */ - if (GST_OBJECT_FLAG_IS_SET (element, GST_AFSRC_OPEN)) { -/* g_print ("DEBUG: trying to close the src file\n"); */ - gst_afsrc_close_file (GST_AFSRC (element)); - } - } else if (GST_STATE_PENDING (element) == GST_STATE_READY) { -/* g_print ("DEBUG: afsrc: ready state pending. This shouldn't happen at the *end* of a stream\n"); */ - if (!GST_OBJECT_FLAG_IS_SET (element, GST_AFSRC_OPEN)) { -/* g_print ("DEBUG: GST_AFSRC_OPEN not set\n"); */ - if (!gst_afsrc_open_file (GST_AFSRC (element))) { -/* g_print ("DEBUG: element tries to open file\n"); */ - return GST_STATE_CHANGE_FAILURE; - } - } - } - - if (GST_ELEMENT_CLASS (parent_class)->change_state) - return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - return GST_STATE_CHANGE_SUCCESS; -} diff --git a/ext/audiofile/gstafsrc.h b/ext/audiofile/gstafsrc.h deleted file mode 100644 index 161a4a69b..000000000 --- a/ext/audiofile/gstafsrc.h +++ /dev/null @@ -1,104 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * - * gstafsrc.h: - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, - * Boston, MA 02110-1301, USA. - */ - - -#ifndef __GST_AFSRC_H__ -#define __GST_AFSRC_H__ - - -#include <gst/gst.h> -#include <audiofile.h> /* what else are we to do */ - - -#ifdef __cplusplus -extern "C" { -#endif /* __cplusplus */ - - -#define GST_TYPE_AFSRC \ - (gst_afsrc_get_type()) -#define GST_AFSRC(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFSRC,GstAFSrc)) -#define GST_AFSRC_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFSRC,GstAFSrcClass)) -#define GST_IS_AFSRC(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFSRC)) -#define GST_IS_AFSRC_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFSRC)) - -typedef struct _GstAFSrc GstAFSrc; -typedef struct _GstAFSrcClass GstAFSrcClass; - -typedef enum { - GST_AFSRC_OPEN = (GST_ELEMENT_FLAG_LAST << 0), - - GST_AFSRC_FLAG_LAST = (GST_ELEMENT_FLAG_LAST << 2), -} GstAFSrcFlags; - -struct _GstAFSrc { - GstElement element; - GstPad *srcpad; - - gchar *filename; -/* FILE *file; */ - -/* AFfilesetup outfilesetup; */ - AFfilehandle file; - int format; - int channels; - int width; - unsigned int rate; - gboolean is_signed; - int type; /* type of output, compare to audiofile.h - * RAW, AIFF, AIFFC, NEXTSND, WAVE - */ - /* blocking */ - gulong curoffset; - gulong bytes_per_read; - - gulong seq; - guint64 framestamp; - /* FIXME : endianness is a little cryptic at this point */ - int endianness_data; /* 4321 or 1234 */ - int endianness_wanted; /* same thing, but what the output format wants */ - int endianness_output; /* what the output endianness will be */ -}; - -struct _GstAFSrcClass { - GstElementClass parent_class; - - /* signals */ - void (*handoff) (GstElement *element,GstPad *pad); -}; - -GType gst_afsrc_get_type (void); -gboolean gst_afsrc_plugin_init (GstPlugin *plugin); - - - - -#ifdef __cplusplus -} -#endif /* __cplusplus */ - - -#endif /* __GST_AFSRC_H__ */ diff --git a/po/POTFILES.skip b/po/POTFILES.skip index e50058b29..ba519eef8 100644 --- a/po/POTFILES.skip +++ b/po/POTFILES.skip @@ -1,5 +1,3 @@ -ext/audiofile/gstafsink.c -ext/audiofile/gstafsrc.c ext/sndfile/gstsfsink.c ext/sndfile/gstsfsrc.c sys/dxr3/dxr3audiosink.c |