gstreamer/ext/vorbis/gstvorbisdeclib.h
Wim Taymans 33196cdd2c audio: change audio format syntax a little
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
2011-09-06 12:06:39 +02:00

184 lines
4.9 KiB
C

/* GStreamer
* Copyright (C) 2010 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
* Copyright (C) 2010 Nokia Corporation. All rights reserved.
* Contact: Stefan Kost <stefan.kost@nokia.com>
*
* Tremor modifications <2006>:
* Chris Lord, OpenedHand Ltd. <chris@openedhand.com>, http://www.o-hand.com/
*
* 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_VORBIS_DEC_LIB_H__
#define __GST_VORBIS_DEC_LIB_H__
#include <gst/gst.h>
#include <gst/audio/audio.h>
#ifndef TREMOR
#include <vorbis/codec.h>
typedef float vorbis_sample_t;
typedef ogg_packet ogg_packet_wrapper;
#define GST_VORBIS_DEC_DESCRIPTION "decode raw vorbis streams to float audio"
#define GST_VORBIS_AUDIO_FORMAT GST_AUDIO_FORMAT_F32
#define GST_VORBIS_AUDIO_FORMAT_STR GST_AUDIO_NE (F32)
#define GST_VORBIS_DEC_SRC_CAPS \
GST_STATIC_CAPS ("audio/x-raw, " \
"format = (string)" GST_VORBIS_AUDIO_FORMAT_STR ", " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 256 ]")
#define GST_VORBIS_DEC_DEFAULT_SAMPLE_WIDTH (32)
#define GST_VORBIS_DEC_GLIB_TYPE_NAME GstVorbisDec
static inline guint8 *
gst_ogg_packet_data (ogg_packet * p)
{
return (guint8 *) p->packet;
}
static inline gint
gst_ogg_packet_size (ogg_packet * p)
{
return p->bytes;
}
static inline void
gst_ogg_packet_wrapper_map (ogg_packet * packet, GstBuffer * buffer)
{
gsize size;
packet->packet = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
packet->bytes = size;
}
static inline void
gst_ogg_packet_wrapper_unmap (ogg_packet * packet, GstBuffer * buffer)
{
gst_buffer_unmap (buffer, packet->packet, packet->bytes);
}
static inline ogg_packet *
gst_ogg_packet_from_wrapper (ogg_packet_wrapper * packet)
{
return packet;
}
#else
#ifdef USE_TREMOLO
#include <Tremolo/ivorbiscodec.h>
#include <Tremolo/codec_internal.h>
typedef ogg_int16_t vorbis_sample_t;
#else
#include <tremor/ivorbiscodec.h>
typedef ogg_int32_t vorbis_sample_t;
#endif
typedef struct _ogg_packet_wrapper ogg_packet_wrapper;
struct _ogg_packet_wrapper {
ogg_packet packet;
ogg_reference ref;
ogg_buffer buf;
};
#define GST_VORBIS_DEC_DESCRIPTION "decode raw vorbis streams to integer audio"
#define GST_VORBIS_AUDIO_FORMAT GST_AUDIO_FORMAT_S16
#define GST_VORBIS_AUDIO_FORMAT_STR GST_AUDIO_NE (S16)
#define GST_VORBIS_DEC_SRC_CAPS \
GST_STATIC_CAPS ("audio/x-raw, " \
"format = (string) " GST_VORBIS_AUDIO_FORMAT_STR ", " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, 6 ]")
/* we need a different type name here */
#define GST_VORBIS_DEC_GLIB_TYPE_NAME GstIVorbisDec
/* and still have it compile */
typedef struct _GstVorbisDec GstIVorbisDec;
typedef struct _GstVorbisDecClass GstIVorbisDecClass;
/* compensate minor variation */
#define vorbis_synthesis(a, b) vorbis_synthesis (a, b, 1)
static inline guint8 *
gst_ogg_packet_data (ogg_packet * p)
{
return (guint8 *) p->packet->buffer->data;
}
static inline gint
gst_ogg_packet_size (ogg_packet * p)
{
return p->packet->buffer->size;
}
static inline void
gst_ogg_packet_wrapper_map (ogg_packet_wrapper * packet,
GstBuffer * buffer)
{
ogg_reference *ref = &packet->ref;
ogg_buffer *buf = &packet->buf;
gsize size;
buf->data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
buf->size = size;
buf->refcount = 1;
buf->ptr.owner = NULL;
buf->ptr.next = NULL;
ref->buffer = buf;
ref->begin = 0;
ref->length = buf->size;
ref->next = NULL;
packet->packet.packet = ref;
packet->packet.bytes = ref->length;
}
static inline void
gst_ogg_packet_wrapper_unmap (ogg_packet_wrapper * packet,
GstBuffer * buffer)
{
ogg_reference *ref = &packet->ref;
ogg_buffer *buf = &packet->buf;
gst_buffer_unmap (buffer, buf->data, buf->size);
}
static inline ogg_packet *
gst_ogg_packet_from_wrapper (ogg_packet_wrapper * packet)
{
return &(packet->packet);
}
#endif
typedef void (*CopySampleFunc)(vorbis_sample_t *out, vorbis_sample_t **in,
guint samples, gint channels);
CopySampleFunc get_copy_sample_func (gint channels);
#endif /* __GST_VORBIS_DEC_LIB_H__ */