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vlc-1.1
Commit 0220726f authored by Jean-Paul Saman's avatar Jean-Paul Saman
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- Updated configure to build fixed32tos16 and fixed32tofloat32. 

  Desperately need a fixed32 mixer for testing the conversions.
- Uncommented unused functions in libmad.
parent ac4a6e1f
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configure
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configure.in
View file @ 0220726f
......@@ -444,7 +444,7 @@ PLUGINS="${PLUGINS} demux/mpeg/es demux/mpeg/audio demux/mpeg/mpeg_system demux/
PLUGINS="${PLUGINS} codec/mpeg_video/idct/idct codec/mpeg_video/idct/idctclassic codec/mpeg_video/motion/motion codec/mpeg_video/mpeg_video codec/spudec/spudec codec/spdif"
#PLUGINS="${PLUGINS} codec/a52old/imdct/imdct codec/a52old/downmix/downmix codec/mpeg_audio/mpeg_audio codec/a52old/a52old codec/lpcm/lpcm"
PLUGINS="${PLUGINS} video_filter/deinterlace/deinterlace video_filter/invert video_filter/wall video_filter/transform video_filter/distort video_filter/clone video_filter/crop"
PLUGINS="${PLUGINS} audio_filter/converter/float32tos16 audio_filter/converter/a52tospdif"
PLUGINS="${PLUGINS} audio_filter/converter/float32tos16 audio_filter/converter/a52tospdif audio_filter/converter/fixed32tos16 audio_filter/converter/fixed32tofloat32"
PLUGINS="${PLUGINS} audio_filter/resampler/trivial"
PLUGINS="${PLUGINS} audio_mixer/trivial audio_mixer/spdif"
PLUGINS="${PLUGINS} audio_output/file"
......
modules/codec/mad/libmad.c
View file @ 0220726f
......@@ -44,7 +44,7 @@ static void PrintFrameInfo(struct mad_header *Header);
enum mad_flow libmad_input(void *data, struct mad_stream *p_libmad_stream)
{
mad_adec_thread_t *p_mad_adec = (mad_adec_thread_t *) data;
size_t ReadSize, Remaining;
size_t ReadSize, Remaining;
unsigned char *ReadStart;
if ( p_mad_adec->p_fifo->b_die == 1 ) {
......@@ -164,110 +164,110 @@ enum mad_flow libmad_input(void *data, struct mad_stream *p_libmad_stream)
* }
*/
/*****************************************************************************
* support routines borrowed from mpg321 (file: mad.c), which is distributed
* under GPL license
*
* mpg321 was written by Joe Drew <drew@debian.org>, and based upon 'plaympeg'
* from the smpeg sources, which was written by various people from Loki Software
* (http://www.lokigames.com).
*
* It also incorporates some source from mad, written by Robert Leslie
*****************************************************************************/
/* The following two routines and data structure are from the ever-brilliant
Rob Leslie.
*/
struct audio_dither {
mad_fixed_t error[3];
mad_fixed_t random;
};
/*
* NAME: prng()
* DESCRIPTION: 32-bit pseudo-random number generator
*/
static inline unsigned long prng(unsigned long state)
{
return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
}
/*
* NAME: mpg321_s24_to_s16_pcm()
* DESCRIPTION: generic linear sample quantize and dither routine
*/
static inline signed int mpg321_s24_to_s16_pcm(unsigned int bits, mad_fixed_t sample,
struct audio_dither *dither)
{
unsigned int scalebits;
mad_fixed_t output, mask, random;
enum {
MIN = -MAD_F_ONE,
MAX = MAD_F_ONE - 1
};
/* noise shape */
sample += dither->error[0] - dither->error[1] + dither->error[2];
dither->error[2] = dither->error[1];
dither->error[1] = dither->error[0] / 2;
/* bias */
output = sample + (1L << (MAD_F_FRACBITS + 1 - bits - 1));
scalebits = MAD_F_FRACBITS + 1 - bits;
mask = (1L << scalebits) - 1;
/* dither */
random = prng(dither->random);
output += (random & mask) - (dither->random & mask);
dither->random = random;
/* clip */
if (output > MAX) {
output = MAX;
if (sample > MAX)
sample = MAX;
}
else if (output < MIN) {
output = MIN;
if (sample < MIN)
sample = MIN;
}
/* quantize */
output &= ~mask;
/* error feedback */
dither->error[0] = sample - output;
/* scale */
return output >> scalebits;
}
/*****************************************************************************
* s24_to_s16_pcm: Scale a 24 bit pcm sample to a 16 bit pcm sample.
*****************************************************************************/
static inline mad_fixed_t s24_to_s16_pcm(mad_fixed_t sample)
{
/* round */
sample += (1L << (MAD_F_FRACBITS - 16));
/* clip */
if (sample >= MAD_F_ONE)
sample = MAD_F_ONE - 1;
else if (sample < -MAD_F_ONE)
sample = -MAD_F_ONE;
/* quantize */
return sample >> (MAD_F_FRACBITS + 1 - 16);
}
///*****************************************************************************
// * support routines borrowed from mpg321 (file: mad.c), which is distributed
// * under GPL license
// *
// * mpg321 was written by Joe Drew <drew@debian.org>, and based upon 'plaympeg'
// * from the smpeg sources, which was written by various people from Loki Software
// * (http://www.lokigames.com).
// *
// * It also incorporates some source from mad, written by Robert Leslie
// *****************************************************************************/
//
///* The following two routines and data structure are from the ever-brilliant
// Rob Leslie.
//*/
//
//struct audio_dither {
// mad_fixed_t error[3];
// mad_fixed_t random;
//};
//
///*
//* NAME: prng()
//* DESCRIPTION: 32-bit pseudo-random number generator
//*/
//static inline unsigned long prng(unsigned long state)
//{
// return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
//}
//
///*
//* NAME: mpg321_s24_to_s16_pcm()
//* DESCRIPTION: generic linear sample quantize and dither routine
//*/
//static inline signed int mpg321_s24_to_s16_pcm(unsigned int bits, mad_fixed_t sample,
// struct audio_dither *dither)
//{
// unsigned int scalebits;
// mad_fixed_t output, mask, random;
//
// enum {
// MIN = -MAD_F_ONE,
// MAX = MAD_F_ONE - 1
// };
//
// /* noise shape */
// sample += dither->error[0] - dither->error[1] + dither->error[2];
//
// dither->error[2] = dither->error[1];
// dither->error[1] = dither->error[0] / 2;
//
// /* bias */
// output = sample + (1L << (MAD_F_FRACBITS + 1 - bits - 1));
//
// scalebits = MAD_F_FRACBITS + 1 - bits;
// mask = (1L << scalebits) - 1;
//
// /* dither */
// random = prng(dither->random);
// output += (random & mask) - (dither->random & mask);
//
// dither->random = random;
//
// /* clip */
// if (output > MAX) {
// output = MAX;
//
// if (sample > MAX)
// sample = MAX;
// }
// else if (output < MIN) {
// output = MIN;
//
// if (sample < MIN)
// sample = MIN;
// }
//
// /* quantize */
// output &= ~mask;
//
// /* error feedback */
// dither->error[0] = sample - output;
//
// /* scale */
// return output >> scalebits;
//}
//
///*****************************************************************************
// * s24_to_s16_pcm: Scale a 24 bit pcm sample to a 16 bit pcm sample.
// *****************************************************************************/
//static inline mad_fixed_t s24_to_s16_pcm(mad_fixed_t sample)
//{
// /* round */
// sample += (1L << (MAD_F_FRACBITS - 16));
//
// /* clip */
// if (sample >= MAD_F_ONE)
// sample = MAD_F_ONE - 1;
// else if (sample < -MAD_F_ONE)
// sample = -MAD_F_ONE;
//
// /* quantize */
// return sample >> (MAD_F_FRACBITS + 1 - 16);
//}
//
/*****************************************************************************
* libmad_ouput: this function is called just after the frame is decoded
*****************************************************************************/
......@@ -456,7 +456,7 @@ enum mad_flow libmad_output3(void *data, struct mad_header const *p_libmad_heade
}
p_mad_adec->last_date += (mtime_t)(p_libmad_pcm->length*2)
/ p_mad_adec->output_format.i_rate;
p_buffer->end_date = p_mad_adec->i_next_pts; // last_date;
p_buffer->end_date = p_mad_adec->last_date;
/* Interleave and keep buffers in mad_fixed_t format */
while (nsamples--)
......@@ -464,11 +464,12 @@ enum mad_flow libmad_output3(void *data, struct mad_header const *p_libmad_heade
/* left audio channel */
sample = *left_ch++;
#ifndef WORDS_BIGENDIAN
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
*(p_buffer->p_buffer)++ = (byte_t) (sample);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
#else
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
#endif
/* right audio channel */
if (p_libmad_pcm->channels == 2)
......@@ -476,11 +477,12 @@ enum mad_flow libmad_output3(void *data, struct mad_header const *p_libmad_heade
sample = *right_ch++;
} /* else reuse left audio channel */
#ifndef WORDS_BIGENDIAN
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
*(p_buffer->p_buffer)++ = (byte_t) (sample);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
#else
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
*(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 8);
// *(p_buffer->p_buffer)++ = (byte_t) (sample >> 0);
#endif
}
......
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