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author | Steve Baker <steve@stevebaker.org> | 2002-04-21 12:04:54 +0000 |
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committer | Steve Baker <steve@stevebaker.org> | 2002-04-21 12:04:54 +0000 |
commit | a175fe373c7fd7835a574009dc253afaa9062766 (patch) | |
tree | a898e8fdddef551692d32e3d2142cd21748712d5 /ext | |
parent | 6863dd92408eb5bd758e6c297324e6b2dc81042b (diff) | |
download | gst-plugins-bad-a175fe373c7fd7835a574009dc253afaa9062766.tar.gz gst-plugins-bad-a175fe373c7fd7835a574009dc253afaa9062766.tar.bz2 gst-plugins-bad-a175fe373c7fd7835a574009dc253afaa9062766.zip |
afparse will eventually be able to support pipelines like: gst-launch filesrc location=file.aiff ! afparse ! osssink
Original commit message from CVS:
afparse will eventually be able to support pipelines like:
gst-launch filesrc location=file.aiff ! afparse ! osssink
However at the moment it doesn't, because the public function afOpenVirtualFile isn't actually implemented yet. I implemented it with the audiofile CVS but it now segfaults the very first time one of the virtual file callbacks is called.
So, I'm committing this for posterity but it is not being built. Hopefully the audiofile lib will be released with working virtual file support soon.
Diffstat (limited to 'ext')
-rw-r--r-- | ext/audiofile/gstafparse.c | 525 | ||||
-rw-r--r-- | ext/audiofile/gstafparse.h | 103 |
2 files changed, 628 insertions, 0 deletions
diff --git a/ext/audiofile/gstafparse.c b/ext/audiofile/gstafparse.c new file mode 100644 index 00000000..b29a94c3 --- /dev/null +++ b/ext/audiofile/gstafparse.c @@ -0,0 +1,525 @@ +/* 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#include <gst/gst.h> +#include <gst/audio/audio.h> +#include "gstafparse.h" + + +static GstElementDetails afparse_details = { + "Audiofile Parse", + "Src", + "Audiofile parser for audio/raw", + VERSION, + "Steve Baker <stevebaker_org@yahoo.co.uk>", + "(C) 2002" +}; + + +/* 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 */ +GST_PAD_TEMPLATE_FACTORY (afparse_src_factory, + "src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "audiofile_src", + "audio/raw", + "format", GST_PROPS_STRING ("int"), + "law", GST_PROPS_INT (0), + "endianness", GST_PROPS_INT (G_BYTE_ORDER), + "signed", GST_PROPS_LIST (GST_PROPS_BOOLEAN (TRUE), GST_PROPS_BOOLEAN (FALSE)), + "width", GST_PROPS_INT_RANGE (8, 16), + "depth", GST_PROPS_INT_RANGE (8, 16), + "rate", GST_PROPS_INT_RANGE (1, G_MAXINT), + "channels", GST_PROPS_INT_RANGE (1, 2) + ) +) + +GST_PAD_TEMPLATE_FACTORY (afparse_sink_factory, + "sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_CAPS_NEW ( + "afparse_sink", + "audio/audiofile", + NULL + ) +) + +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); + +static GstCaps* gst_afparse_type_find(GstBuffer *buf, gpointer private); +static GstElementStateReturn gst_afparse_change_state (GstElement *element); + +static GstElementClass *parent_class = NULL; + +static GstTypeDefinition aftype_definitions[] = { + { "aftypes audio/audiofile", "audio/audiofile", ".wav .aiff .aif .aifc", gst_afparse_type_find }, + { NULL, NULL, NULL, NULL }, +}; + +GType +gst_afparse_get_type (void) +{ + static GType afparse_type = 0; + + if (!afparse_type) { + static const GTypeInfo afparse_info = { + sizeof (GstAFParseClass), NULL, + 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_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; + + gstelement_class->change_state = gst_afparse_change_state; +} + +static void +gst_afparse_init (GstAFParse *afparse) +{ + afparse->srcpad = gst_pad_new_from_template (afparse_src_factory (), "src"); + gst_element_add_pad (GST_ELEMENT (afparse), afparse->srcpad); + + afparse->sinkpad = gst_pad_new_from_template (afparse_sink_factory (), "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; + + /* default to low latency */ + afparse->frames_per_read = 64; + 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; + GstBufferPool *bufpool; + gint numframes, frames_to_bytes, frames_per_read, bytes_per_read; + guint8 *data; + + afparse = GST_AFPARSE(element); + + frames_to_bytes = afparse->channels * afparse->width / 8; + frames_per_read = afparse->frames_per_read; + bytes_per_read = frames_per_read * frames_to_bytes; + + bufpool = gst_buffer_pool_get_default (bytes_per_read, 8); + + do { + buf = gst_buffer_new_from_pool (bufpool, 0, 0); + 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_BUFFER(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, buf); + afparse->timestamp += numframes * 1E9 / afparse->rate; + } + while (TRUE); + + gst_bytestream_destroy((GstByteStream*)afparse->vfile->closure); + gst_buffer_pool_unref(bufpool); + +} + +static GstBuffer * +gst_afparse_get (GstPad *pad) +{ + GstAFParse *afparse; + GstBuffer *buf; + + glong readbytes, readframes; + glong frameCount; + + g_return_val_if_fail (pad != NULL, NULL); + afparse = GST_AFPARSE (gst_pad_get_parent (pad)); + + buf = gst_buffer_new (); + g_return_val_if_fail (buf, NULL); + + GST_BUFFER_DATA (buf) = (gpointer) g_malloc (afparse->bytes_per_read); + + /* calculate frameCount to read based on file info */ + + frameCount = afparse->bytes_per_read / (afparse->channels * afparse->width / 8); +/* g_print ("DEBUG: gstafparse: going to read %ld frames\n", frameCount); */ + readframes = afReadFrames (afparse->file, AF_DEFAULT_TRACK, GST_BUFFER_DATA (buf), frameCount); + readbytes = readframes * (afparse->channels * afparse->width / 8); + if (readbytes == 0) { + gst_element_set_eos (GST_ELEMENT (afparse)); + return GST_BUFFER (gst_event_new (GST_EVENT_EOS)); + } + + GST_BUFFER_SIZE (buf) = readbytes; + GST_BUFFER_OFFSET (buf) = afparse->curoffset; + + afparse->curoffset += readbytes; + + afparse->timestamp += gst_audio_frame_length (afparse->srcpad, buf); + GST_BUFFER_TIMESTAMP (buf) = afparse->timestamp * 1E9 / afparse->rate; + + return buf; +} + +static void +gst_afparse_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec) +{ + GstAFParse *afparse; + + /* it's not null if we got it, but it might not be ours */ + 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; + + /* it's not null if we got it, but it might not be ours */ + 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; + } +} + +static gboolean +plugin_init (GModule *module, GstPlugin *plugin) +{ + GstElementFactory *factory; + gint i=0; + + factory = gst_element_factory_new ("afparse", GST_TYPE_AFPARSE, + &afparse_details); + g_return_val_if_fail (factory != NULL, FALSE); + + gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (afparse_src_factory)); + gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (afparse_sink_factory)); + + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory)); + + while (aftype_definitions[i].name) { + GstTypeFactory *type; + + type = gst_type_factory_new (&aftype_definitions[i]); + gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (type)); + i++; + } + + /* load audio support library */ + if (!gst_library_load ("gstaudio")) + { + gst_info ("gstafparse/sink: could not load support library: 'gstaudio'\n"); + return FALSE; + } + if (!gst_library_load ("gstbytestream")) + { + gst_info ("gstafparse/sink: could not load support library: 'gstbytestream'\n"); + return FALSE; + } + return TRUE; +} + + +GstPluginDesc plugin_desc = { + GST_VERSION_MAJOR, + GST_VERSION_MINOR, + "afparse", + plugin_init +}; + +/* this is where we open the audiofile */ +static gboolean +gst_afparse_open_file (GstAFParse *afparse) +{ + g_return_val_if_fail (!GST_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN), FALSE); + + afparse->vfile->closure = gst_bytestream_new(afparse->srcpad); + + /* open the file */ + g_print("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; + } + + g_print("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 (GST_CAT_PLUGIN_INFO, "ERROR: float data not supported yet !\n"); + } + afparse->rate = (guint) afGetRate (afparse->file, AF_DEFAULT_TRACK); + afparse->width = sampleWidth; + GST_DEBUG (GST_CAT_PLUGIN_INFO, + "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_try_set_caps (afparse->srcpad, + GST_CAPS_NEW ( + "af_src", + "audio/raw", + "format", GST_PROPS_STRING ("int"), + "law", GST_PROPS_INT (0), /*FIXME */ + "endianness", GST_PROPS_INT (G_BYTE_ORDER), /*FIXME */ + "signed", GST_PROPS_BOOLEAN (afparse->is_signed), + "width", GST_PROPS_INT (afparse->width), + "depth", GST_PROPS_INT (afparse->width), + "rate", GST_PROPS_INT (afparse->rate), + "channels", GST_PROPS_INT (afparse->channels) + ) + ); + + GST_FLAG_SET (afparse, GST_AFPARSE_OPEN); + + return TRUE; +} + +static void +gst_afparse_close_file (GstAFParse *afparse) +{ + g_return_if_fail (GST_FLAG_IS_SET (afparse, GST_AFPARSE_OPEN)); + if (afCloseFile (afparse->file) != 0) + { + g_warning ("afparse: oops, error closing !\n"); + } + else { + GST_FLAG_UNSET (afparse, GST_AFPARSE_OPEN); + } +} + +static GstElementStateReturn +gst_afparse_change_state (GstElement *element) +{ + g_return_val_if_fail (GST_IS_AFPARSE (element), GST_STATE_FAILURE); + + /* if going to NULL then close the file */ + if (GST_STATE_PENDING (element) == GST_STATE_NULL) + { +/* printf ("DEBUG: afparse state change: null pending\n"); */ + if (GST_FLAG_IS_SET (element, GST_AFPARSE_OPEN)) + { +/* g_print ("DEBUG: trying to close the src file\n"); */ + gst_afparse_close_file (GST_AFPARSE (element)); + } + } + else if (GST_STATE_PENDING (element) == GST_STATE_READY) + { +/* g_print ("DEBUG: afparse: ready state pending. This shouldn't happen at the *end* of a stream\n"); */ + if (!GST_FLAG_IS_SET (element, GST_AFPARSE_OPEN)) + { +/* g_print ("DEBUG: GST_AFPARSE_OPEN not set\n"); */ + if (!gst_afparse_open_file (GST_AFPARSE (element))) + { +/* g_print ("DEBUG: element tries to open file\n"); */ + return GST_STATE_FAILURE; + } + } + } + + if (GST_ELEMENT_CLASS (parent_class)->change_state) + return GST_ELEMENT_CLASS (parent_class)->change_state (element); + + return GST_STATE_SUCCESS; +} + +static ssize_t +gst_afparse_vf_read (AFvirtualfile *vfile, void *data, size_t nbytes) +{ + GstByteStream *bs = (GstByteStream*)vfile->closure; + guint8 *bytes; + g_print("doing read\n"); + bytes = gst_bytestream_peek_bytes(bs, nbytes); + if (!bytes){ + /* handle events */ + + return 0; + } + + gst_bytestream_flush_fast(bs, nbytes); + data = bytes; + return nbytes; +} + +static long +gst_afparse_vf_seek (AFvirtualfile *vfile, long offset, int is_relative) +{ + GstByteStream *bs = (GstByteStream*)vfile->closure; + GstSeekType type; + + g_print("doing seek\n"); + type = is_relative ? GST_SEEK_BYTEOFFSET_CUR : GST_SEEK_BYTEOFFSET_SET; + if (gst_bytestream_seek(bs, type, (gint64)offset)){ + return offset; + } + return 0; +} + +static long +gst_afparse_vf_length (AFvirtualfile *vfile) +{ + /*GstByteStream *bs = (GstByteStream*)vfile->closure;*/ + /* FIXME there is currently no practical way to do this. + * wait for the events rewrite to drop */ + g_warning("cannot get length at the moment"); + return G_MAXLONG; +} + +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;*/ + + g_print("doing destroy\n"); +} + +static long +gst_afparse_vf_tell (AFvirtualfile *vfile) +{ + /*GstByteStream *bs = (GstByteStream*)vfile->closure;*/ + g_print("doing tell\n"); + /* return gst_bytestream_tell(bs);*/ + return 0; +} + +static GstCaps* +gst_afparse_type_find(GstBuffer *buf, gpointer private) +{ + + return NULL; +} + diff --git a/ext/audiofile/gstafparse.h b/ext/audiofile/gstafparse.h new file mode 100644 index 00000000..72a40bb6 --- /dev/null +++ b/ext/audiofile/gstafparse.h @@ -0,0 +1,103 @@ +/* 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + + +#ifndef __GST_AFPARSE_H__ +#define __GST_AFPARSE_H__ + + +#include <config.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 */ + + +/*GstElementDetails gst_afparse_details;*/ + + +#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(obj) \ + (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, + + 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); +}; + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + + +#endif /* __GST_AFPARSE_H__ */ |