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Commit b890a56b authored by Pierre d'Herbemont's avatar Pierre d'Herbemont
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Revert "bandlimited: factorize." 

This reverts commit d6b5bc59.

This wasn't intended from prime time. I guess.
parent 3b0cbbed
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Showing with 122 additions and 72 deletions
modules/audio_filter/resampler/bandlimited.c
View file @ b890a56b
......@@ -93,74 +93,6 @@ vlc_module_begin ()
set_callbacks( OpenFilter, CloseFilter )
vlc_module_end ()
static void Resample_helper( filter_t *p_filter, float *p_in, float **pp_out,
int d_factor, int i_nb_channels,
block_t *p_out_buf, int *pi_out )
{
filter_sys_t *p_sys = p_filter->p_sys;
while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
if( d_factor >= 1 )
{
/* FilterFloatUP() is faster if we can use it */
/* Perform left-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, *pp_out,
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, *pp_out,
p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate,
1, i_nb_channels );
#if 0
/* Normalize for unity filter gain */
for( int i = 0; i < i_nb_channels; i++ )
{
*(p_out+i) *= d_old_scale_factor;
}
#endif
/* Sanity check */
if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
<= (unsigned int)*pi_out+1 )
{
*pp_out += i_nb_channels;
(*pi_out)++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
break;
}
}
else
{
/* Perform left-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, *pp_out,
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, *pp_out,
p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
1, i_nb_channels );
}
*pp_out += i_nb_channels;
(*pi_out)++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
}
/*****************************************************************************
* Resample: convert a buffer
*****************************************************************************/
......@@ -288,8 +220,67 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
continue;
}
Resample_helper( p_filter, p_in, &p_out, p_sys->d_old_factor,
i_nb_channels, p_out_buf, &i_out );
while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
if( p_sys->d_old_factor >= 1 )
{
/* FilterFloatUP() is faster if we can use it */
/* Perform left-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate,
1, i_nb_channels );
#if 0
/* Normalize for unity filter gain */
for( i = 0; i < i_nb_channels; i++ )
{
*(p_out+i) *= d_old_scale_factor;
}
#endif
/* Sanity check */
if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
<= (unsigned int)i_out+1 )
{
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
break;
}
}
else
{
/* Perform left-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
1, i_nb_channels );
}
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_in += i_nb_channels;
p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
......@@ -303,8 +294,67 @@ static block_t *Resample( filter_t * p_filter, block_t * p_in_buf )
}
for( ; i_in < i_in_nb - i_filter_wing; i_in++ )
{
Resample_helper( p_filter, p_in, &p_out, d_factor,
i_nb_channels, p_out_buf, &i_out );
while( p_sys->i_remainder < p_filter->fmt_out.audio.i_rate )
{
if( d_factor >= 1 )
{
/* FilterFloatUP() is faster if we can use it */
/* Perform left-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUP( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate,
1, i_nb_channels );
#if 0
/* Normalize for unity filter gain */
for( int i = 0; i < i_nb_channels; i++ )
{
*(p_out+i) *= d_old_scale_factor;
}
#endif
/* Sanity check */
if( p_out_buf->i_buffer/p_filter->fmt_in.audio.i_bytes_per_frame
<= (unsigned int)i_out+1 )
{
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
break;
}
}
else
{
/* Perform left-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in, p_out,
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
-1, i_nb_channels );
/* Perform right-wing inner product */
FilterFloatUD( SMALL_FILTER_FLOAT_IMP, SMALL_FILTER_FLOAT_IMPD,
SMALL_FILTER_NWING, p_in + i_nb_channels, p_out,
p_filter->fmt_out.audio.i_rate -
p_sys->i_remainder,
p_filter->fmt_out.audio.i_rate, p_filter->fmt_in.audio.i_rate,
1, i_nb_channels );
}
p_out += i_nb_channels;
i_out++;
p_sys->i_remainder += p_filter->fmt_in.audio.i_rate;
}
p_in += i_nb_channels;
p_sys->i_remainder -= p_filter->fmt_out.audio.i_rate;
......
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