[vlc-devel] commit: Revert "bandlimited: factorize." (Pierre d'Herbemont )
git version control
git at videolan.org
Thu Dec 31 17:10:45 CET 2009
vlc | branch: master | Pierre d'Herbemont <pdherbemont at free.fr> | Thu Dec 31 17:10:24 2009 +0100| [b890a56baa23614689fc37c312e8242f58e8079f] | committer: Pierre d'Herbemont
Revert "bandlimited: factorize."
This reverts commit d6b5bc594818ae7448b180a3cb79b8fa55f923e6.
This wasn't intended from prime time. I guess.
> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=b890a56baa23614689fc37c312e8242f58e8079f
---
modules/audio_filter/resampler/bandlimited.c | 194 ++++++++++++++++----------
1 files changed, 122 insertions(+), 72 deletions(-)
diff --git a/modules/audio_filter/resampler/bandlimited.c b/modules/audio_filter/resampler/bandlimited.c
index 03d0c36..2f9267b 100644
--- a/modules/audio_filter/resampler/bandlimited.c
+++ b/modules/audio_filter/resampler/bandlimited.c
@@ -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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