* modules/stream_out/transcode.c, modules/audio_filter/format.c: fixed a bunch of stupid mistakes.

modules/audio_filter/format.c

@@ -40,6 +40,7 @@ static int  Open ( vlc_object_t * );
 static block_t *Float32toS16( filter_t *, block_t * );
 static block_t *Float32toU16( filter_t *, block_t * );
 static block_t *S16toFloat32( filter_t *, block_t * );
+static block_t *S16Invert   ( filter_t *, block_t * );
 
 /*****************************************************************************
  * Module descriptor
@@ -62,16 +63,23 @@ static int Open( vlc_object_t *p_this )
     {
         p_filter->pf_audio_filter = Float32toS16;
     }
-    else if ( p_filter->fmt_in.i_codec == VLC_FOURCC('f','l','3','2') &&
-              p_filter->fmt_out.i_codec == AUDIO_FMT_U16_NE )
+    else if( p_filter->fmt_in.i_codec == VLC_FOURCC('f','l','3','2') &&
+             p_filter->fmt_out.i_codec == AUDIO_FMT_U16_NE )
     {
         p_filter->pf_audio_filter = Float32toU16;
     }
-    else if ( p_filter->fmt_in.i_codec == AUDIO_FMT_S16_NE &&
-              p_filter->fmt_out.i_codec == VLC_FOURCC('f','l','3','2') )
+    else if( p_filter->fmt_in.i_codec == AUDIO_FMT_S16_NE &&
+             p_filter->fmt_out.i_codec == VLC_FOURCC('f','l','3','2') )
     {
         p_filter->pf_audio_filter = S16toFloat32;
     }
+    else if( ( p_filter->fmt_in.i_codec == VLC_FOURCC('s','1','6','l') &&
+               p_filter->fmt_out.i_codec == VLC_FOURCC('s','1','6','b') ) ||
+             ( p_filter->fmt_in.i_codec == VLC_FOURCC('s','1','6','b') &&
+               p_filter->fmt_out.i_codec == VLC_FOURCC('s','1','6','l') ) )
+    {
+        p_filter->pf_audio_filter = S16Invert;
+    }
     else return VLC_EGENERIC;
     
     msg_Dbg( p_this, "%4.4s->%4.4s, bits per sample: %i",
@@ -91,7 +99,7 @@ static block_t *Float32toS16( filter_t *p_filter, block_t *p_block )
     float *p_in = (float *)p_block->p_buffer;
     int16_t *p_out = (int16_t *)p_in;
 
-    for( i = p_block->i_buffer/ p_filter->fmt_in.audio.i_bitspersample; i-- ; )
+    for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
     {
 #if 0
         /* Slow version. */
@@ -119,7 +127,7 @@ static block_t *Float32toU16( filter_t *p_filter, block_t *p_block )
     float *p_in = (float *)p_block->p_buffer;
     uint16_t *p_out = (uint16_t *)p_in;
 
-    for( i = p_block->i_buffer/ p_filter->fmt_in.audio.i_bitspersample; i-- ; )
+    for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
     {
         if ( *p_in >= 1.0 ) *p_out = 65535;
         else if ( *p_in < -1.0 ) *p_out = 0;
@@ -146,10 +154,10 @@ static block_t *S16toFloat32( filter_t *p_filter, block_t *p_block )
         return NULL;
     }
 
-    p_in = (int16_t *)(p_block->p_buffer + p_block->i_buffer) - 1;
-    p_out = (float *)(p_block_out->p_buffer + p_block_out->i_buffer) - 1;
+    p_in = (int16_t *)p_block->p_buffer;
+    p_out = (float *)p_block_out->p_buffer;
 
-    for( i = p_block->i_buffer/ p_filter->fmt_in.audio.i_bitspersample; i-- ; )
+    for( i = p_block->i_buffer*8/p_filter->fmt_in.audio.i_bitspersample; i--; )
     {
 #if 0
         /* Slow version */
@@ -163,7 +171,7 @@ static block_t *S16toFloat32( filter_t *p_filter, block_t *p_block )
         *p_out = u.f - 384.0;
 #endif
 
-        p_in--; p_out--;
+        p_in++; p_out++;
     }
 
     p_block_out->i_samples = p_block->i_samples;
@@ -172,5 +180,23 @@ static block_t *S16toFloat32( filter_t *p_filter, block_t *p_block )
     p_block_out->i_length = p_block->i_length;
     p_block_out->i_rate = p_block->i_rate;
 
+    p_block->pf_release( p_block );
     return p_block_out;
 }
+
+static block_t *S16Invert( filter_t *p_filter, block_t *p_block )
+{
+    int i;
+    uint8_t *p_in = (uint8_t *)p_block->p_buffer;
+    uint8_t tmp;
+
+    for( i = 0; i < p_block->i_buffer / 2; i++ )
+    {
+        tmp = p_in[0];
+        p_in[0] = p_in[1];
+        p_in[1] = tmp;
+        p_in += 2;
+    }
+
+    return p_block;
+}

modules/stream_out/transcode.c

@@ -818,17 +818,30 @@ int audio_BitsPerSample( vlc_fourcc_t i_format )
     {
     case VLC_FOURCC('u','8',' ',' '):
     case VLC_FOURCC('s','8',' ',' '):
-        return 1;
+        return 8;
 
     case VLC_FOURCC('u','1','6','l'):
     case VLC_FOURCC('s','1','6','l'):
     case VLC_FOURCC('u','1','6','b'):
     case VLC_FOURCC('s','1','6','b'):
-        return 2;
-
+        return 16;
+
+    case VLC_FOURCC('u','2','4','l'):
+    case VLC_FOURCC('s','2','4','l'):
+    case VLC_FOURCC('u','2','4','b'):
+    case VLC_FOURCC('s','2','4','b'):
+        return 24;
+
+    case VLC_FOURCC('u','3','2','l'):
+    case VLC_FOURCC('s','3','2','l'):
+    case VLC_FOURCC('u','3','2','b'):
+    case VLC_FOURCC('s','3','2','b'):
     case VLC_FOURCC('f','l','3','2'):
     case VLC_FOURCC('f','i','3','2'):
-        return 4;
+        return 32;
+
+    case VLC_FOURCC('f','l','6','4'):
+        return 64;
     }
 
     return 0;
@@ -892,7 +905,8 @@ static int transcode_audio_new( sout_stream_t *p_stream,
             id->p_encoder->fmt_out.audio.i_physical_channels;
     id->p_encoder->fmt_in.audio.i_channels =
         id->p_encoder->fmt_out.audio.i_channels;
-    id->p_encoder->fmt_in.audio.i_bitspersample = 16;
+    id->p_encoder->fmt_in.audio.i_bitspersample =
+        audio_BitsPerSample( id->p_encoder->fmt_in.i_codec );
 
     id->p_encoder->p_cfg = p_stream->p_sys->p_audio_cfg;
 
@@ -906,6 +920,8 @@ static int transcode_audio_new( sout_stream_t *p_stream,
         return VLC_EGENERIC;
     }
     id->p_encoder->fmt_in.audio.i_format = id->p_encoder->fmt_in.i_codec;
+    id->p_encoder->fmt_in.audio.i_bitspersample =
+        audio_BitsPerSample( id->p_encoder->fmt_in.i_codec );
 
     /* Check if we need a filter for chroma conversion or resizing */
     if( id->p_decoder->fmt_out.i_codec !=
@@ -934,6 +950,9 @@ static int transcode_audio_new( sout_stream_t *p_stream,
             return VLC_EGENERIC;
         }
 
+        id->pp_filter[0]->fmt_out.audio.i_bitspersample = 
+            audio_BitsPerSample( id->pp_filter[0]->fmt_out.i_codec );
+
         /* Try a 2 stage conversion */
         if( id->pp_filter[0]->fmt_out.i_codec !=
             id->p_encoder->fmt_in.i_codec )
@@ -1067,7 +1086,7 @@ static aout_buffer_t *audio_new_buffer( decoder_t *p_dec, int i_samples )
 
     if( p_dec->fmt_out.audio.i_bitspersample )
     {
-        i_size = i_samples * p_dec->fmt_out.audio.i_bitspersample *
+        i_size = i_samples * p_dec->fmt_out.audio.i_bitspersample / 8 *
             p_dec->fmt_out.audio.i_channels;
     }
     else if( p_dec->fmt_out.audio.i_bytes_per_frame &&