/*****************************************************************************
* format.c : PCM format converter
*****************************************************************************
- * Copyright (C) 2002-2005 VideoLAN
+ * Copyright (C) 2002-2005 the VideoLAN team
* $Id$
*
* Authors: Christophe Massiot <massiot@via.ecp.fr>
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111, USA.
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/
/*****************************************************************************
* Preamble
*****************************************************************************/
-#include <stdlib.h> /* malloc(), free() */
-#include <string.h>
#include <vlc/vlc.h>
-#include <vlc/decoder.h>
+#include <vlc_aout.h>
+#include <vlc_block.h>
#include "vlc_filter.h"
#ifdef WORDS_BIGENDIAN
{ VLC_FOURCC('s','8',' ',' '), AOUT_FMT_U16_NE, S8toU16 },
{ VLC_FOURCC('s','8',' ',' '), AOUT_FMT_U16_IE, S8toU16Invert },
{ VLC_FOURCC('s','8',' ',' '), VLC_FOURCC('u','8',' ',' '), S8toU8 },
-
+
/* From u8 */
{ VLC_FOURCC('u','8',' ',' '), VLC_FOURCC('f','l','3','2'), U8toFloat32 },
{ VLC_FOURCC('u','8',' ',' '), AOUT_FMT_S16_NE, U8toS16 },
* Module descriptor
*****************************************************************************/
vlc_module_begin();
- set_description( _("audio filter for PCM format conversion") );
+ set_description( _("Audio filter for PCM format conversion") );
+ set_category( CAT_AUDIO );
+ set_subcategory( SUBCAT_AUDIO_MISC );
set_capability( "audio filter2", 1 );
set_callbacks( Open, NULL );
vlc_module_end();
if( ConvertTable[i].pf_convert == NULL )
return VLC_EGENERIC;
-
p_filter->pf_audio_filter = ConvertTable[i].pf_convert;
+ p_filter->fmt_out.audio = p_filter->fmt_in.audio;
+ p_filter->fmt_out.audio.i_format = p_filter->fmt_out.i_codec;
msg_Dbg( p_this, "%4.4s->%4.4s, bits per sample: %i",
(char *)&p_filter->fmt_in.i_codec,
uint8_t *p_out = (uint8_t *)p_in;
int32_t out;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 4; i--; )
{
if ( *p_in >= 1.0 ) out = 8388607;
else if ( *p_in < -1.0 ) out = -8388608;
else out = *p_in * 8388608.0;
#ifdef WORDS_BIGENDIAN
- *((int16_t *)p_out) = out >> 8;
- p_out[2] = out & 0xFF;
+ *((int16_t *)p_out) = out >> 8;
+ p_out[2] = out & 0xFF;
#else
- *((int16_t *)(p_out+1)) = out >> 8;
- p_out[0] = out & 0xFF;
+ *((int16_t *)(p_out+1)) = out >> 8;
+ p_out[0] = out & 0xFF;
#endif
p_in++; p_out += 3;
float *p_in = (float *)p_block->p_buffer;
int16_t *p_out = (int16_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 4; i--; )
{
#if 0
/* Slow version. */
float *p_in = (float *)p_block->p_buffer;
uint16_t *p_out = (uint16_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 4; i--; )
{
if ( *p_in >= 1.0 ) *p_out = 65535;
else if ( *p_in < -1.0 ) *p_out = 0;
int i;
p_block_out =
- p_filter->pf_audio_buffer_new( p_filter, p_block->i_buffer*4/3 );
+ p_filter->pf_audio_buffer_new( p_filter, p_block->i_buffer * 4 / 3 );
if( !p_block_out )
{
msg_Warn( p_filter, "can't get output buffer" );
p_in = p_block->p_buffer;
p_out = (float *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 3; i--; )
{
+ /* FIXME: unaligned reads */
#ifdef WORDS_BIGENDIAN
*p_out = ((float)( (((int32_t)*(int16_t *)(p_in)) << 8) + p_in[2]))
#else
uint8_t *p_in = (uint8_t *)p_block->p_buffer;
uint8_t *p_out = (uint8_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 3; i--; )
{
#ifdef WORDS_BIGENDIAN
*p_out++ = *p_in++;
p_in = (int16_t *)p_block->p_buffer;
p_out = (float *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
{
#if 0
/* Slow version */
static block_t *U16toFloat32( filter_t *p_filter, block_t *p_block )
{
block_t *p_block_out;
- int16_t *p_in;
+ uint16_t *p_in;
float *p_out;
int i;
return NULL;
}
- p_in = (int16_t *)p_block->p_buffer;
+ p_in = (uint16_t *)p_block->p_buffer;
p_out = (float *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
{
*p_out++ = (float)(*p_in++ - 32768) / 32768.0;
}
p_in = (uint8_t *)p_block->p_buffer;
p_out = (uint8_t *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
{
#ifdef WORDS_BIGENDIAN
*p_out++ = *p_in++;
int16_t *p_in = (int16_t *)p_block->p_buffer;
int8_t *p_out = (int8_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
*p_out++ = (*p_in++) >> 8;
p_block->i_buffer /= 2;
int16_t *p_in = (int16_t *)p_block->p_buffer;
uint8_t *p_out = (uint8_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
*p_out++ = ((*p_in++) + 32768) >> 8;
p_block->i_buffer /= 2;
int16_t *p_in = (int16_t *)p_block->p_buffer;
uint16_t *p_out = (uint16_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
*p_out++ = (*p_in++) + 32768;
return p_block;
uint16_t *p_in = (uint16_t *)p_block->p_buffer;
int8_t *p_out = (int8_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
*p_out++ = ((int)(*p_in++) - 32768) >> 8;
p_block->i_buffer /= 2;
uint16_t *p_in = (uint16_t *)p_block->p_buffer;
uint8_t *p_out = (uint8_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
*p_out++ = (*p_in++) >> 8;
p_block->i_buffer /= 2;
static block_t *U16toS16( filter_t *p_filter, block_t *p_block )
{
int i;
- int16_t *p_in = (int16_t *)p_block->p_buffer;
- uint16_t *p_out = (uint16_t *)p_in;
+ uint16_t *p_in = (uint16_t *)p_block->p_buffer;
+ int16_t *p_out = (int16_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer / 2; i--; )
*p_out++ = (int)(*p_in++) - 32768;
return p_block;
int8_t *p_in = (int8_t *)p_block->p_buffer;
uint8_t *p_out = (uint8_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer; i--; )
*p_out++ = ((*p_in++) + 128);
return p_block;
uint8_t *p_in = (uint8_t *)p_block->p_buffer;
int8_t *p_out = (int8_t *)p_in;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer; i--; )
*p_out++ = ((*p_in++) - 128);
return p_block;
p_in = (int8_t *)p_block->p_buffer;
p_out = (uint16_t *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer; i--; )
*p_out++ = ((*p_in++) + 128) << 8;
p_block_out->i_samples = p_block->i_samples;
p_in = (uint8_t *)p_block->p_buffer;
p_out = (int16_t *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer; i--; )
*p_out++ = ((*p_in++) - 128) << 8;
p_block_out->i_samples = p_block->i_samples;
p_in = (int8_t *)p_block->p_buffer;
p_out = (int16_t *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer; i--; )
*p_out++ = (*p_in++) << 8;
p_block_out->i_samples = p_block->i_samples;
p_in = (uint8_t *)p_block->p_buffer;
p_out = (uint16_t *)p_block_out->p_buffer;
- for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
+ for( i = p_block->i_buffer; i--; )
*p_out++ = (*p_in++) << 8;
p_block_out->i_samples = p_block->i_samples;