[vlc-commits] Ugly resampler: implement all linear formats

[Rémi Denis-Courmont]

vlc | branch: master | Rémi Denis-Courmont <remi at remlab.net> | Tue May 31 18:20:41 2011 +0300| [38feb73c11735e76ea233bdd0366e925c6d158dd] | committer: Rémi Denis-Courmont

Ugly resampler: implement all linear formats

> http://git.videolan.org/gitweb.cgi/vlc.git/?a=commit;h=38feb73c11735e76ea233bdd0366e925c6d158dd
---

 modules/audio_filter/resampler/ugly.c |   31 ++++++++++++++-----------------
 1 files changed, 14 insertions(+), 17 deletions(-)

diff --git a/modules/audio_filter/resampler/ugly.c b/modules/audio_filter/resampler/ugly.c
index a1a46cf..616d836 100644
--- a/modules/audio_filter/resampler/ugly.c
+++ b/modules/audio_filter/resampler/ugly.c
@@ -59,17 +59,14 @@ static int Create( vlc_object_t *p_this )
 {
     filter_t * p_filter = (filter_t *)p_this;
 
-    if ( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
-          || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
-          || p_filter->fmt_in.audio.i_physical_channels
-              != p_filter->fmt_out.audio.i_physical_channels
-          || p_filter->fmt_in.audio.i_original_channels
-              != p_filter->fmt_out.audio.i_original_channels
-          || (p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32
-               && p_filter->fmt_in.audio.i_format != VLC_CODEC_FI32) )
-    {
+    if( p_filter->fmt_in.audio.i_rate == p_filter->fmt_out.audio.i_rate
+     || p_filter->fmt_in.audio.i_format != p_filter->fmt_out.audio.i_format
+     || p_filter->fmt_in.audio.i_physical_channels
+                                 != p_filter->fmt_out.audio.i_physical_channels
+     || p_filter->fmt_in.audio.i_original_channels
+                                 != p_filter->fmt_out.audio.i_original_channels
+     || AOUT_FMT_NON_LINEAR( &p_filter->fmt_in.audio ) )
         return VLC_EGENERIC;
-    }
 
     p_filter->pf_audio_filter = DoWork;
     return VLC_SUCCESS;
@@ -87,12 +84,12 @@ static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
     block_t *p_out_buf = p_in_buf;
     unsigned int i_out_nb = p_in_buf->i_nb_samples
         * p_filter->fmt_out.audio.i_rate / p_filter->fmt_in.audio.i_rate;
-    const unsigned int i_sample_bytes =
-        aout_FormatNbChannels( &p_filter->fmt_in.audio ) * sizeof(int32_t);
+    const unsigned framesize = (p_filter->fmt_in.audio.i_bitspersample / 8)
+        * aout_FormatNbChannels( &p_filter->fmt_in.audio );
 
     if( p_filter->fmt_out.audio.i_rate > p_filter->fmt_in.audio.i_rate )
     {
-        p_out_buf = block_Alloc( i_out_nb * i_sample_bytes );
+        p_out_buf = block_Alloc( i_out_nb * framesize );
         if( !p_out_buf )
             goto out;
     }
@@ -102,7 +99,7 @@ static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
     unsigned int i_remainder = 0;
 
     p_out_buf->i_nb_samples = i_out_nb;
-    p_out_buf->i_buffer = i_out_nb * i_sample_bytes;
+    p_out_buf->i_buffer = i_out_nb * framesize;
     p_out_buf->i_pts = p_in_buf->i_pts;
     p_out_buf->i_length = p_out_buf->i_nb_samples *
         1000000 / p_filter->fmt_out.audio.i_rate;
@@ -110,14 +107,14 @@ static block_t *DoWork( filter_t * p_filter, block_t * p_in_buf )
     while( i_out_nb )
     {
         if( p_out != p_in )
-            memcpy( p_out, p_in, i_sample_bytes );
-        p_out += i_sample_bytes;
+            memcpy( p_out, p_in, framesize );
+        p_out += framesize;
         i_out_nb--;
 
         i_remainder += p_filter->fmt_in.audio.i_rate;
         while( i_remainder >= p_filter->fmt_out.audio.i_rate )
         {
-            p_in += i_sample_bytes;
+            p_in += framesize;
             i_remainder -= p_filter->fmt_out.audio.i_rate;
         }
     }