gstreamer/ext/audiofile/gstafsrc.c
Stefan Kost 36c2fc253b renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition
Original commit message from CVS:
* examples/indexing/indexmpeg.c: (main):
* ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio),
(gst_artsdsink_close_audio), (gst_artsdsink_change_state):
* ext/artsd/gstartsdsink.h:
* ext/audiofile/gstafparse.c: (gst_afparse_open_file),
(gst_afparse_close_file):
* ext/audiofile/gstafparse.h:
* ext/audiofile/gstafsink.c: (gst_afsink_open_file),
(gst_afsink_close_file), (gst_afsink_chain),
(gst_afsink_change_state):
* ext/audiofile/gstafsink.h:
* ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
(gst_afsrc_close_file), (gst_afsrc_change_state):
* ext/audiofile/gstafsrc.h:
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init):
* ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackbin.c: (gst_jack_bin_init),
(gst_jack_bin_change_state):
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init):
* ext/nas/nassink.c: (gst_nassink_open_audio),
(gst_nassink_close_audio), (gst_nassink_change_state):
* ext/nas/nassink.h:
* ext/polyp/polypsink.c: (gst_polypsink_init):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state):
* ext/sdl/sdlvideosink.h:
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
* ext/sndfile/gstsf.c: (gst_sf_set_property),
(gst_sf_change_state), (gst_sf_release_request_pad),
(gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop):
* ext/sndfile/gstsf.h:
* ext/swfdec/gstswfdec.c: (gst_swfdec_init):
* ext/tarkin/gsttarkindec.c: (gst_tarkindec_init):
* gst/apetag/apedemux.c: (gst_ape_demux_init):
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init):
* gst/festival/gstfestival.c: (gst_festival_change_state):
* gst/festival/gstfestival.h:
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
* gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init),
(gst_multifilesink_set_location), (gst_multifilesink_open_file),
(gst_multifilesink_close_file), (gst_multifilesink_next_file),
(gst_multifilesink_pad_query), (gst_multifilesink_handle_event),
(gst_multifilesink_chain), (gst_multifilesink_change_state):
* gst/multifilesink/gstmultifilesink.h:
* gst/videodrop/gstvideodrop.c: (gst_videodrop_init):
* sys/cdrom/gstcdplayer.c: (cdplayer_init):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init),
(dxr3audiosink_open), (dxr3audiosink_close),
(dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3),
(dxr3audiosink_change_state):
* sys/dxr3/dxr3audiosink.h:
* sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open),
(dxr3spusink_close), (dxr3spusink_chain),
(dxr3spusink_change_state):
* sys/dxr3/dxr3spusink.h:
* sys/dxr3/dxr3videosink.c: (dxr3videosink_init),
(dxr3videosink_open), (dxr3videosink_close),
(dxr3videosink_write_data), (dxr3videosink_change_state):
* sys/dxr3/dxr3videosink.h:
* sys/glsink/glimagesink.c: (gst_glimagesink_init):
* sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
(gst_qcamsrc_open), (gst_qcamsrc_close):
* sys/qcam/gstqcamsrc.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
* sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get),
(gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
(gst_vcdsrc_change_state), (gst_vcdsrc_recalculate):
* sys/vcd/vcdsrc.h:
renamed GST_FLAGS macros to GST_OBJECT_FLAGS
moved bitshift from macro to enum definition
2005-10-12 14:29:55 +00:00

403 lines
11 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.
*/
#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"
/* elementfactory information */
static GstElementDetails afsrc_details = {
"Audiofile Src",
"Source/Audio",
"Read audio files from disk using libaudiofile",
"Thomas <thomas@apestaart.org>",
};
/* 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 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_details (element_class, &afsrc_details);
}
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 (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:
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;
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;
}