[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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