gstreamer/ext/audiofile/gstafsrc.c
Christian Schaller 1d5f318e0d plugins part of license field patch
Original commit message from CVS:
plugins part of license field patch
2002-09-18 19:02:52 +00:00

413 lines
12 KiB
C

/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
*
* gstafsrc.c:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include "gstafsrc.h"
/* elementfactory information */
static GstElementDetails afsrc_details = {
"Audiofile Src",
"Source/Audio",
"LGPL",
"Read audio files from disk using libaudiofile",
VERSION,
"Thomas <thomas@apestaart.org>",
"(C) 2001"
};
/* AFSrc signals and args */
enum {
/* FILL ME */
SIGNAL_HANDOFF,
LAST_SIGNAL
};
enum {
ARG_0,
ARG_LOCATION
};
/* added a src factory function to force audio/raw MIME type */
/* I think the caps can be broader, we need to change that somehow */
GST_PAD_TEMPLATE_FACTORY (afsrc_src_factory,
"src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_CAPS_NEW (
"audiofile_src",
"audio/raw",
"format", GST_PROPS_STRING ("int"),
"law", GST_PROPS_INT (0),
"endianness", GST_PROPS_INT (G_BYTE_ORDER),
"signed", GST_PROPS_LIST (
GST_PROPS_BOOLEAN (TRUE),
GST_PROPS_BOOLEAN (FALSE)
),
"width", GST_PROPS_INT_RANGE (8, 16),
"depth", GST_PROPS_INT_RANGE (8, 16),
"rate", GST_PROPS_INT_RANGE (4000, 48000), /*FIXME*/
"channels", GST_PROPS_INT_RANGE (1, 2)
)
);
/* we use an enum for the output type arg */
#define GST_TYPE_AFSRC_TYPES (gst_afsrc_types_get_type())
/* FIXME: fix the string ints to be string-converted from the audiofile.h types */
/* defined but not used
static GType
gst_afsrc_types_get_type (void)
{
static GType afsrc_types_type = 0;
static GEnumValue afsrc_types[] = {
{AF_FILE_RAWDATA, "0", "raw PCM"},
{AF_FILE_AIFFC, "1", "AIFFC"},
{AF_FILE_AIFF, "2", "AIFF"},
{AF_FILE_NEXTSND, "3", "Next/SND"},
{AF_FILE_WAVE, "4", "Wave"},
{0, NULL, NULL},
};
if (!afsrc_types_type)
{
afsrc_types_type = g_enum_register_static ("GstAudiosrcTypes", afsrc_types);
}
return afsrc_types_type;
}
*/
static void gst_afsrc_class_init (GstAFSrcClass *klass);
static void gst_afsrc_init (GstAFSrc *afsrc);
static gboolean gst_afsrc_open_file (GstAFSrc *src);
static void gst_afsrc_close_file (GstAFSrc *src);
static GstBuffer* gst_afsrc_get (GstPad *pad);
static void gst_afsrc_set_property (GObject *object, guint prop_id,
const GValue *value, GParamSpec *pspec);
static void gst_afsrc_get_property (GObject *object, guint prop_id,
GValue *value, GParamSpec *pspec);
static GstElementStateReturn gst_afsrc_change_state (GstElement *element);
static GstElementClass *parent_class = NULL;
static guint gst_afsrc_signals[LAST_SIGNAL] = { 0 };
GType
gst_afsrc_get_type (void)
{
static GType afsrc_type = 0;
if (!afsrc_type) {
static const GTypeInfo afsrc_info = {
sizeof (GstAFSrcClass), NULL,
NULL,
(GClassInitFunc) gst_afsrc_class_init,
NULL,
NULL,
sizeof (GstAFSrc),
0,
(GInstanceInitFunc) gst_afsrc_init,
};
afsrc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAFSrc", &afsrc_info, 0);
}
return afsrc_type;
}
static void
gst_afsrc_class_init (GstAFSrcClass *klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass*)klass;
gstelement_class = (GstElementClass*)klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gst_element_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 (afsrc_src_factory (), "src");
gst_element_add_pad (GST_ELEMENT (afsrc), afsrc->srcpad);
gst_pad_set_get_function (afsrc->srcpad, gst_afsrc_get);
afsrc->bytes_per_read = 4096;
afsrc->curoffset = 0;
afsrc->seq = 0;
afsrc->filename = NULL;
afsrc->file = NULL;
/* default values, should never be needed */
afsrc->channels = 2;
afsrc->width = 16;
afsrc->rate = 44100;
afsrc->type = AF_FILE_WAVE;
afsrc->endianness_data = 1234;
afsrc->endianness_wanted = 1234;
afsrc->framestamp = 0;
}
static GstBuffer *
gst_afsrc_get (GstPad *pad)
{
GstAFSrc *src;
GstBuffer *buf;
glong readbytes, readframes;
glong frameCount;
g_return_val_if_fail (pad != NULL, NULL);
src = GST_AFSRC (gst_pad_get_parent (pad));
buf = gst_buffer_new ();
g_return_val_if_fail (buf, NULL);
GST_BUFFER_DATA (buf) = (gpointer) g_malloc (src->bytes_per_read);
/* calculate frameCount to read based on file info */
frameCount = src->bytes_per_read / (src->channels * src->width / 8);
/* g_print ("DEBUG: gstafsrc: going to read %ld frames\n", frameCount); */
readframes = afReadFrames (src->file, AF_DEFAULT_TRACK, GST_BUFFER_DATA (buf),
frameCount);
readbytes = readframes * (src->channels * src->width / 8);
if (readbytes == 0) {
gst_element_set_eos (GST_ELEMENT (src));
return GST_BUFFER (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 buf;
}
static void
gst_afsrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
{
GstAFSrc *src;
/* it's not null if we got it, but it might not be ours */
src = GST_AFSRC (object);
switch (prop_id) {
case ARG_LOCATION:
if (src->filename)
g_free (src->filename);
src->filename = g_strdup (g_value_get_string (value));
break;
default:
break;
}
}
static void
gst_afsrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
{
GstAFSrc *src;
/* it's not null if we got it, but it might not be ours */
g_return_if_fail (GST_IS_AFSRC (object));
src = GST_AFSRC (object);
switch (prop_id) {
case ARG_LOCATION:
g_value_set_string (value, src->filename);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_afsrc_plugin_init (GModule *module, GstPlugin *plugin)
{
GstElementFactory *factory;
factory = gst_element_factory_new ("afsrc", GST_TYPE_AFSRC,
&afsrc_details);
g_return_val_if_fail (factory != NULL, FALSE);
gst_element_factory_add_pad_template (factory, GST_PAD_TEMPLATE_GET (afsrc_src_factory));
gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
/* load audio support library */
if (!gst_library_load ("gstaudio"))
return FALSE;
return TRUE;
}
/* this is where we open the audiofile */
static gboolean
gst_afsrc_open_file (GstAFSrc *src)
{
g_return_val_if_fail (!GST_FLAG_IS_SET (src, GST_AFSRC_OPEN), FALSE);
/* open the file */
src->file = afOpenFile (src->filename, "r", AF_NULL_FILESETUP);
if (src->file == AF_NULL_FILEHANDLE)
{
g_print ("ERROR: gstafsrc: Could not open file %s for reading\n",
src->filename);
gst_element_error (GST_ELEMENT (src), g_strconcat ("opening file \"",
src->filename, "\"", NULL));
return FALSE;
}
/* get the audiofile audio parameters */
{
int sampleFormat, sampleWidth;
src->channels = afGetChannels (src->file, AF_DEFAULT_TRACK);
afGetSampleFormat (src->file, AF_DEFAULT_TRACK,
&sampleFormat, &sampleWidth);
switch (sampleFormat)
{
case AF_SAMPFMT_TWOSCOMP:
src->is_signed = TRUE;
break;
case AF_SAMPFMT_UNSIGNED:
src->is_signed = FALSE;
break;
case AF_SAMPFMT_FLOAT:
case AF_SAMPFMT_DOUBLE:
GST_DEBUG (GST_CAT_PLUGIN_INFO,
"ERROR: float data not supported yet !\n");
}
src->rate = (guint) afGetRate (src->file, AF_DEFAULT_TRACK);
src->width = sampleWidth;
GST_DEBUG (GST_CAT_PLUGIN_INFO,
"input file: %d channels, %d width, %d rate, signed %s\n",
src->channels, src->width, src->rate,
src->is_signed ? "yes" : "no");
}
/* set caps on src */
/*FIXME: add all the possible formats, especially float ! */
gst_pad_try_set_caps (src->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 (src->is_signed),
"width", GST_PROPS_INT (src->width),
"depth", GST_PROPS_INT (src->width),
"rate", GST_PROPS_INT (src->rate),
"channels", GST_PROPS_INT (src->channels)
)
);
GST_FLAG_SET (src, GST_AFSRC_OPEN);
return TRUE;
}
static void
gst_afsrc_close_file (GstAFSrc *src)
{
/* g_print ("DEBUG: closing srcfile...\n"); */
g_return_if_fail (GST_FLAG_IS_SET (src, GST_AFSRC_OPEN));
/* g_print ("DEBUG: past flag test\n"); */
/* if (fclose (src->file) != 0) */
if (afCloseFile (src->file) != 0)
{
g_print ("WARNING: afsrc: oops, error closing !\n");
perror ("close");
gst_element_error (GST_ELEMENT (src), g_strconcat("closing file \"", src->filename, "\"", NULL));
}
else {
GST_FLAG_UNSET (src, GST_AFSRC_OPEN);
}
}
static GstElementStateReturn
gst_afsrc_change_state (GstElement *element)
{
g_return_val_if_fail (GST_IS_AFSRC (element), GST_STATE_FAILURE);
/* if going to NULL then close the file */
if (GST_STATE_PENDING (element) == GST_STATE_NULL)
{
/* printf ("DEBUG: afsrc state change: null pending\n"); */
if (GST_FLAG_IS_SET (element, GST_AFSRC_OPEN))
{
/* g_print ("DEBUG: trying to close the src file\n"); */
gst_afsrc_close_file (GST_AFSRC (element));
}
}
else if (GST_STATE_PENDING (element) == GST_STATE_READY)
{
/* g_print ("DEBUG: afsrc: ready state pending. This shouldn't happen at the *end* of a stream\n"); */
if (!GST_FLAG_IS_SET (element, GST_AFSRC_OPEN))
{
/* g_print ("DEBUG: GST_AFSRC_OPEN not set\n"); */
if (!gst_afsrc_open_file (GST_AFSRC (element)))
{
/* g_print ("DEBUG: element tries to open file\n"); */
return GST_STATE_FAILURE;
}
}
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
return GST_STATE_SUCCESS;
}