sndfile: rewrite sndfile for 1.0

Add a sfdec for a start. Instead of a source plugin, this is a demuxer/decoder combination. This makes it work with auto-plugging.
This commit is contained in:
Stefan Sauer 2013-12-17 17:56:32 +01:00
parent 6348113777
commit 5f89bee749
6 changed files with 603 additions and 81 deletions

View file

@ -346,7 +346,7 @@ GST_PLUGINS_NONPORTED=" cdxaparse \
linsys vcd \
apexsink dc1394 \
gsettings \
musepack nas sdl sndfile timidity \
musepack nas sdl timidity \
directdraw acm wininet \
xvid lv2 teletextdec sndio osx_video quicktime"
AC_SUBST(GST_PLUGINS_NONPORTED)
@ -1952,7 +1952,7 @@ AG_GST_CHECK_FEATURE(SMOOTHSTREAMING, [Smooth Streaming plug-in], smoothstreamin
dnl *** sndfile ***
translit(dnm, m, l) AM_CONDITIONAL(USE_SNDFILE, true)
AG_GST_CHECK_FEATURE(SNDFILE, [sndfile plug-in], sfsrc sfsink, [
AG_GST_CHECK_FEATURE(SNDFILE, [sndfile plug-in], sfdec sfenc, [
PKG_CHECK_MODULES(SNDFILE, sndfile >= 1.0.16, HAVE_SNDFILE="yes", HAVE_SNDFILE="no")
AC_SUBST(SNDFILE_CFLAGS)
AC_SUBST(SNDFILE_LIBS)

View file

@ -1,10 +1,9 @@
plugin_LTLIBRARIES = libgstsndfile.la
libgstsndfile_la_SOURCES = gstsf.c gstsfsrc.c gstsfsink.c
libgstsndfile_la_SOURCES = gstsf.c gstsfdec.c
libgstsndfile_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(SNDFILE_CFLAGS)
libgstsndfile_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(SNDFILE_LIBS)
libgstsndfile_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(SNDFILE_LIBS)
libgstsndfile_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstsndfile_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = gstsf.h gstsfsrc.h gstsfsink.h
noinst_HEADERS = gstsf.h gstsfdec.h

View file

@ -27,79 +27,86 @@
#include "gstsf.h"
/* sf formats */
GType
gst_sf_major_types_get_type (void)
GstCaps *
gst_sf_create_audio_template_caps (void)
{
static GType sf_major_types_type = 0;
static GEnumValue *sf_major_types = NULL;
GstCaps *caps = gst_caps_new_empty ();
SF_FORMAT_INFO format_info;
const gchar *fmt;
gint k, count;
if (!sf_major_types_type) {
SF_FORMAT_INFO format_info;
int k, count;
sf_command (NULL, SFC_GET_FORMAT_MAJOR_COUNT, &count, sizeof (gint));
sf_command (NULL, SFC_GET_FORMAT_MAJOR_COUNT, &count, sizeof (int));
for (k = 0; k < count; k++) {
format_info.format = k;
sf_command (NULL, SFC_GET_FORMAT_MAJOR, &format_info, sizeof (format_info));
sf_major_types = g_new0 (GEnumValue, count + 1);
for (k = 0; k < count; k++) {
format_info.format = k;
sf_command (NULL, SFC_GET_FORMAT_MAJOR, &format_info,
sizeof (format_info));
sf_major_types[k].value = format_info.format;
sf_major_types[k].value_name = g_strdup (format_info.name);
sf_major_types[k].value_nick = g_strdup (format_info.extension);
/* Irritatingly enough, there exist major_types with the same extension. Let's
just hope that sndfile gives us the list in alphabetical order, as it
currently does. */
if (k > 0
&& strcmp (sf_major_types[k].value_nick,
sf_major_types[k - 1].value_nick) == 0) {
g_free ((gchar *) sf_major_types[k].value_nick);
sf_major_types[k].value_nick =
g_strconcat (sf_major_types[k - 1].value_nick, "-",
sf_major_types[k].value_name, NULL);
g_strcanon ((gchar *) sf_major_types[k].value_nick,
G_CSET_A_2_Z G_CSET_a_2_z G_CSET_DIGITS "-", '-');
}
switch (format_info.format) {
case SF_FORMAT_AIFF: /* Apple/SGI AIFF format */
fmt = "audio/x-aiff";
break;
case SF_FORMAT_AU: /* Sun/NeXT AU format */
fmt = "audio/x-au";
break;
case SF_FORMAT_FLAC: /* FLAC lossless file format */
fmt = "audio/x-flac";
break;
case SF_FORMAT_IRCAM: /* Berkeley/IRCAM/CARL */
fmt = "audio/x-ircam";
break;
case SF_FORMAT_NIST: /* Sphere NIST format. */
fmt = "audio/x-nist";
break;
case SF_FORMAT_OGG: /* Xiph OGG container */
fmt = "audio/ogg";
break;
case SF_FORMAT_PAF: /* Ensoniq PARIS file format. */
fmt = "audio/x-paris";
break;
case SF_FORMAT_RAW: /* RAW PCM data. */
fmt = "audio/x-raw";
break;
case SF_FORMAT_SDS: /* Midi Sample Dump Standard */
fmt = "audio/x-sds";
break;
case SF_FORMAT_SVX: /* Amiga IFF / SVX8 / SV16 format. */
fmt = "audio/x-svx";
break;
case SF_FORMAT_VOC: /* VOC files. */
fmt = "audio/x-voc";
break;
case SF_FORMAT_WAV: /* Microsoft WAV format */
case SF_FORMAT_WAVEX: /* MS WAVE with WAVEFORMATEX */
fmt = "audio/x-wav";
break;
case SF_FORMAT_W64: /* Sonic Foundry's 64 bit RIFF/WAV */
fmt = "audio/x-w64";
break;
case SF_FORMAT_XI: /* Fasttracker 2 Extended Instrument */
fmt = "audio/x-xi";
break;
case SF_FORMAT_RF64: /* RF64 WAV file */
case SF_FORMAT_MAT4: /* Matlab (tm) V4.2 / GNU Octave 2.0 */
case SF_FORMAT_MAT5: /* Matlab (tm) V5.0 / GNU Octave 2.1 */
case SF_FORMAT_PVF: /* Portable Voice Format */
case SF_FORMAT_HTK: /* HMM Tool Kit format */
case SF_FORMAT_AVR: /* Audio Visual Research */
case SF_FORMAT_SD2: /* Sound Designer 2 */
case SF_FORMAT_CAF: /* Core Audio File format */
case SF_FORMAT_WVE: /* Psion WVE format */
case SF_FORMAT_MPC2K: /* Akai MPC 2000 sampler */
default:
fmt = NULL;
GST_WARNING ("format 0x%x: '%s' is not mapped", format_info.format,
format_info.name);
}
sf_major_types_type =
g_enum_register_static ("GstSndfileMajorTypes", sf_major_types);
}
return sf_major_types_type;
}
GType
gst_sf_minor_types_get_type (void)
{
static GType sf_minor_types_type = 0;
static GEnumValue *sf_minor_types = NULL;
if (!sf_minor_types_type) {
SF_FORMAT_INFO format_info;
int k, count;
sf_command (NULL, SFC_GET_FORMAT_SUBTYPE_COUNT, &count, sizeof (int));
sf_minor_types = g_new0 (GEnumValue, count + 1);
for (k = 0; k < count; k++) {
format_info.format = k;
sf_command (NULL, SFC_GET_FORMAT_SUBTYPE, &format_info,
sizeof (format_info));
sf_minor_types[k].value = format_info.format;
sf_minor_types[k].value_name = g_strdup (format_info.name);
sf_minor_types[k].value_nick = g_ascii_strdown (format_info.name, -1);
g_strcanon ((gchar *) sf_minor_types[k].value_nick,
G_CSET_a_2_z G_CSET_DIGITS "-", '-');
if (fmt != NULL) {
gst_caps_append_structure (caps, gst_structure_new_empty (fmt));
}
sf_minor_types_type =
g_enum_register_static ("GstSndfileMinorTypes", sf_minor_types);
}
return sf_minor_types_type;
return gst_caps_simplify (caps);
}
static gboolean
@ -112,12 +119,8 @@ plugin_init (GstPlugin * plugin)
bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
#endif /* ENABLE_NLS */
if (!gst_element_register (plugin, "sfsink", GST_RANK_NONE,
gst_sf_sink_get_type ()))
return FALSE;
if (!gst_element_register (plugin, "sfsrc", GST_RANK_NONE,
gst_sf_src_get_type ()))
if (!gst_element_register (plugin, "sfdec", GST_RANK_MARGINAL,
gst_sf_dec_get_type ()))
return FALSE;
return TRUE;
@ -126,5 +129,5 @@ plugin_init (GstPlugin * plugin)
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
sndfile,
"use libsndfile to read and write audio from and to files",
"use libsndfile to read and write various audio formats",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

View file

@ -28,6 +28,7 @@
G_BEGIN_DECLS
GstCaps *gst_sf_create_audio_template_caps (void);
#define GST_TYPE_SF_MAJOR_TYPES (gst_sf_major_types_get_type())
#define GST_TYPE_SF_MINOR_TYPES (gst_sf_minor_types_get_type())
@ -35,9 +36,7 @@ G_BEGIN_DECLS
GType gst_sf_major_types_get_type (void);
GType gst_sf_minor_types_get_type (void);
GType gst_sf_sink_get_type (void);
GType gst_sf_src_get_type (void);
GType gst_sf_dec_get_type (void);
G_END_DECLS

448
ext/sndfile/gstsfdec.c Normal file
View file

@ -0,0 +1,448 @@
/* GStreamer libsndfile plugin
* Copyright (C) 2013 Stefan Sauer <ensonic@users.sf.net>
*
* 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 self 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 "gstsfdec.h"
#include <gst/audio/audio.h>
enum
{
PROP_0,
PROP_LOCATION
};
#define FORMATS \
"{ "GST_AUDIO_NE (F32)", "GST_AUDIO_NE (S32)", "GST_AUDIO_NE (S16)" }"
static GstStaticPadTemplate sf_dec_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMATS ", "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
GST_DEBUG_CATEGORY_STATIC (gst_sf_dec_debug);
#define GST_CAT_DEFAULT gst_sf_dec_debug
#define DEFAULT_BUFFER_FRAMES (256)
static GstStateChangeReturn gst_sf_dec_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_sf_dec_sink_activate (GstPad * pad, GstObject * parent);
static gboolean gst_sf_dec_sink_activate_mode (GstPad * sinkpad,
GstObject * parent, GstPadMode mode, gboolean active);
static void gst_sf_dec_loop (GstPad * pad);
static gboolean gst_sf_dec_start (GstSFDec * bsrc);
static gboolean gst_sf_dec_stop (GstSFDec * bsrc);
#define _do_init \
GST_DEBUG_CATEGORY_INIT (gst_sf_dec_debug, "sfdec", 0, "sfdec element");
#define gst_sf_dec_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstSFDec, gst_sf_dec, GST_TYPE_ELEMENT, _do_init);
/* sf virtual io */
static sf_count_t
gst_sf_vio_get_filelen (void *user_data)
{
GstElement *element = GST_ELEMENT (user_data);
gint64 dur;
if (gst_element_query_duration (element, GST_FORMAT_BYTES, &dur)) {
return (sf_count_t) dur;
}
GST_WARNING_OBJECT (element, "query_duration failed");
return -1;
}
static sf_count_t
gst_sf_vio_tell (void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
return self->pos;
}
static sf_count_t
gst_sf_vio_seek (sf_count_t offset, int whence, void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
switch (whence) {
case SEEK_CUR:
self->pos += offset;
break;
case SEEK_SET:
self->pos = offset;
break;
case SEEK_END:
self->pos = gst_sf_vio_get_filelen (user_data) - offset;
break;
}
return (sf_count_t) self->pos;
}
static sf_count_t
gst_sf_vio_read (void *ptr, sf_count_t count, void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
GstBuffer *buffer = gst_buffer_new_wrapped_full (0, ptr, count, 0, count,
ptr, NULL);
if (gst_pad_pull_range (self->sinkpad, self->pos, count, &buffer) ==
GST_FLOW_OK) {
GST_DEBUG_OBJECT (self, "read %d bytes @ pos %" G_GUINT64_FORMAT,
(gint) count, self->pos);
self->pos += count;
return count;
}
GST_WARNING_OBJECT (self, "read failed");
return 0;
}
static sf_count_t
gst_sf_vio_write (const void *ptr, sf_count_t count, void *user_data)
{
GstSFDec *self = GST_SF_DEC (user_data);
GstBuffer *buffer = gst_buffer_new_wrapped (g_memdup (ptr, count), count);
if (gst_pad_push (self->srcpad, buffer) == GST_FLOW_OK) {
return count;
}
GST_WARNING_OBJECT (self, "write failed");
return 0;
}
SF_VIRTUAL_IO gst_sf_vio = {
&gst_sf_vio_get_filelen,
&gst_sf_vio_seek,
&gst_sf_vio_read,
&gst_sf_vio_write,
&gst_sf_vio_tell,
};
static void
gst_sf_dec_class_init (GstSFDecClass * klass)
{
GstElementClass *gstelement_class;
gstelement_class = GST_ELEMENT_CLASS (klass);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_sf_dec_change_state);
gst_element_class_set_static_metadata (gstelement_class, "Sndfile decoder",
"Demuxer/Audio",
"Read audio streams using libsndfile",
"Stefan Sauer <ensonic@user.sf.net>");
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sf_dec_src_factory));
gst_element_class_add_pad_template (gstelement_class,
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
gst_sf_create_audio_template_caps ()));
}
static void
gst_sf_dec_init (GstSFDec * self)
{
self->sinkpad = gst_pad_new_from_template (gst_element_class_get_pad_template
(GST_ELEMENT_GET_CLASS (self), "sink"), "sink");
gst_pad_set_activate_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_sf_dec_sink_activate));
gst_pad_set_activatemode_function (self->sinkpad,
GST_DEBUG_FUNCPTR (gst_sf_dec_sink_activate_mode));
gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
self->srcpad = gst_pad_new_from_static_template (&sf_dec_src_factory, "src");
/* TODO(ensonic): event + query functions */
gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
}
static GstStateChangeReturn
gst_sf_dec_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstSFDec *self = GST_SF_DEC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_sf_dec_start (self);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_sf_dec_stop (self);
break;
default:
break;
}
return ret;
}
static gboolean
gst_sf_dec_start (GstSFDec * self)
{
self->file = NULL;
return TRUE;
}
static gboolean
gst_sf_dec_stop (GstSFDec * self)
{
int err = 0;
g_return_val_if_fail (self->file != NULL, FALSE);
GST_INFO_OBJECT (self, "Closing sndfile stream");
if ((err = sf_close (self->file)))
goto close_failed;
self->file = NULL;
self->offset = 0;
self->channels = 0;
self->rate = 0;
return TRUE;
close_failed:
{
GST_ELEMENT_ERROR (self, RESOURCE, CLOSE,
("Could not close sndfile stream."),
("soundfile error: %s", sf_error_number (err)));
return FALSE;
}
}
static gboolean
gst_sf_dec_sink_activate (GstPad * sinkpad, GstObject * parent)
{
GstQuery *query;
gboolean pull_mode;
query = gst_query_new_scheduling ();
if (!gst_pad_peer_query (sinkpad, query)) {
gst_query_unref (query);
goto activate_push;
}
pull_mode = gst_query_has_scheduling_mode_with_flags (query,
GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
gst_query_unref (query);
if (!pull_mode)
goto activate_push;
GST_DEBUG_OBJECT (sinkpad, "activating pull");
return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
activate_push:
{
GST_DEBUG_OBJECT (sinkpad, "activating push");
return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, TRUE);
}
}
static gboolean
gst_sf_dec_sink_activate_mode (GstPad * sinkpad, GstObject * parent,
GstPadMode mode, gboolean active)
{
gboolean res;
switch (mode) {
case GST_PAD_MODE_PUSH:
res = FALSE; /* no push support */
break;
case GST_PAD_MODE_PULL:
if (active) {
/* if we have a scheduler we can start the task */
res = gst_pad_start_task (sinkpad, (GstTaskFunction) gst_sf_dec_loop,
sinkpad, NULL);
} else {
res = gst_pad_stop_task (sinkpad);
}
break;
default:
res = FALSE;
break;
}
return res;
}
static gboolean
gst_sf_dec_open_file (GstSFDec * self)
{
SF_INFO info;
GstCaps *caps;
GstStructure *s;
GstSegment seg;
gint width;
const gchar *format;
gchar *stream_id;
GST_DEBUG_OBJECT (self, "opening the stream");
info.format = 0;
self->file = sf_open_virtual (&gst_sf_vio, SFM_READ, &info, self);
if (!self->file)
goto open_failed;
stream_id =
gst_pad_create_stream_id (self->srcpad, GST_ELEMENT_CAST (self), NULL);
gst_pad_push_event (self->srcpad, gst_event_new_stream_start (stream_id));
g_free (stream_id);
self->channels = info.channels;
self->rate = info.samplerate;
/* TODO(ensonic): do something with info.seekable? */
/* TODO(ensonic): calculate duration info.frames */
/* negotiate srcpad caps */
if ((caps = gst_pad_get_allowed_caps (self->srcpad)) == NULL) {
caps = gst_pad_get_pad_template_caps (self->srcpad);
}
caps = gst_caps_make_writable (caps);
GST_DEBUG_OBJECT (self, "allowed caps %" GST_PTR_FORMAT, caps);
s = gst_caps_get_structure (caps, 0);
gst_structure_set (s,
"channels", G_TYPE_INT, self->channels,
"rate", G_TYPE_INT, self->rate, NULL);
if (!gst_structure_fixate_field_string (s, "format", GST_AUDIO_NE (S16)))
GST_WARNING_OBJECT (self, "Failed to fixate format to S16NE");
caps = gst_caps_fixate (caps);
GST_DEBUG_OBJECT (self, "fixated caps %" GST_PTR_FORMAT, caps);
/* configure to output the negotiated format */
s = gst_caps_get_structure (caps, 0);
format = gst_structure_get_string (s, "format");
if (g_str_equal (format, GST_AUDIO_NE (S32))) {
self->reader = (GstSFReader) sf_readf_int;
width = 32;
} else if (g_str_equal (format, GST_AUDIO_NE (S16))) {
self->reader = (GstSFReader) sf_readf_short;
width = 16;
} else {
self->reader = (GstSFReader) sf_readf_float;
width = 32;
}
self->bytes_per_frame = width * self->channels / 8;
gst_pad_set_caps (self->srcpad, caps);
gst_caps_unref (caps);
gst_segment_init (&seg, GST_FORMAT_TIME);
/* TODO: calculate seg.stop = song_length; */
gst_pad_push_event (self->srcpad, gst_event_new_segment (&seg));
return TRUE;
open_failed:
{
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ,
(_("Could not open sndfile stream for reading.")),
("soundfile error: %s", sf_strerror (NULL)));
return FALSE;
}
}
static void
gst_sf_dec_loop (GstPad * pad)
{
GstSFDec *self = GST_SF_DEC (GST_PAD_PARENT (pad));
GstBuffer *buf;
GstMapInfo map;
GstFlowReturn flow;
sf_count_t bytes_read;
guint num_frames = 1024; /* arbitrary */
if (G_UNLIKELY (!self->file)) {
/* not started yet */
if (!gst_sf_dec_open_file (self))
goto pause;
}
buf = gst_buffer_new_and_alloc (self->bytes_per_frame * num_frames);
gst_buffer_map (buf, &map, GST_MAP_WRITE);
bytes_read = self->reader (self->file, map.data, num_frames);
GST_DEBUG_OBJECT (self, "read %d / %d bytes = %d frames of audio",
(gint) bytes_read, (gint) map.size, num_frames);
gst_buffer_unmap (buf, &map);
if (G_UNLIKELY (bytes_read < 0))
goto could_not_read;
if (G_UNLIKELY (bytes_read == 0))
goto eos;
num_frames = bytes_read / self->bytes_per_frame;
GST_BUFFER_OFFSET (buf) = self->offset;
GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (self->offset,
GST_SECOND, self->rate);
self->offset += num_frames;
GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (self->offset,
GST_SECOND, self->rate) - GST_BUFFER_TIMESTAMP (buf);
flow = gst_pad_push (self->srcpad, buf);
if (flow != GST_FLOW_OK) {
GST_LOG_OBJECT (self, "pad push flow: %s", gst_flow_get_name (flow));
goto pause;
}
return;
/* ERROR */
could_not_read:
{
GST_ELEMENT_ERROR (self, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
gst_buffer_unref (buf);
goto pause;
}
eos:
{
GST_DEBUG_OBJECT (self, "EOS");
gst_buffer_unref (buf);
gst_pad_push_event (self->srcpad, gst_event_new_eos ());
goto pause;
}
pause:
{
GST_INFO_OBJECT (self, "Pausing");
gst_pad_pause_task (self->sinkpad);
}
}

73
ext/sndfile/gstsfdec.h Normal file
View file

@ -0,0 +1,73 @@
/* GStreamer libsndfile plugin
* Copyright (C) 2013 Stefan Sauer <ensonic@users.sf.net>
*
* 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_SF_DEC_H__
#define __GST_SF_DEC_H__
#include "gstsf.h"
#include <gst/base/gstbasesrc.h>
G_BEGIN_DECLS
#define GST_TYPE_SF_DEC \
(gst_sf_dec_get_type())
#define GST_SF_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_SF_DEC,GstSFDec))
#define GST_SF_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_SF_DEC,GstSFDecClass))
#define GST_IS_SF_DEC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_SF_DEC))
#define GST_IS_SF_DEC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_SF_DEC))
typedef struct _GstSFDec GstSFDec;
typedef struct _GstSFDecClass GstSFDecClass;
typedef sf_count_t (*GstSFReader)(SNDFILE *f, void *data, sf_count_t nframes);
struct _GstSFDec {
GstElement parent;
GstPad *sinkpad;
GstPad *srcpad;
guint64 pos;
SNDFILE *file;
sf_count_t offset;
GstSFReader reader;
gint bytes_per_frame;
gint channels;
gint rate;
};
struct _GstSFDecClass {
GstElementClass parent_class;
};
G_END_DECLS
#endif /* __GST_SF_DEC_H__ */