uint8_t *in_ptr;
// Generate the affine transform scaling values
- float scale_width = ( float )iwidth / ( float )owidth;
- float scale_height = ( float )iheight / ( float )oheight;
+ int scale_width = ( iwidth << 16 ) / owidth;
+ int scale_height = ( iheight << 16 ) / oheight;
// Loop for the entirety of our output height.
for ( y = - out_y_range; y < out_y_range ; y ++ )
{
// Calculate the derived y value
- dy = scale_height * y;
+ dy = ( scale_height * y ) >> 16;
// Start at the beginning of the line
out_ptr = out_line;
for ( x = - out_x_range; x < out_x_range; x += 1 )
{
// Calculated the derived x
- dx = scale_width * x;
-
- // Check if x and y are in the valid input range.
- if ( abs( dx ) < in_x_range && abs( dy ) < in_y_range )
- {
- // We're in the input range for this row.
- in_ptr = in_line + ( dx >> 1 ) * 4 + 2 * ( x & 1 );
- *out_ptr ++ = *in_ptr ++;
- *out_ptr ++ = *in_ptr ++;
- }
- else
- {
- // We're not in the input range for this row.
- *out_ptr ++ = 16;
- *out_ptr ++ = 128;
- }
+ dx = ( scale_width * x ) >> 16;
+
+ // We're in the input range for this row.
+ in_ptr = in_line + ( dx << 1 );
+ *out_ptr ++ = *in_ptr ++;
+ in_ptr = in_line + ( ( dx >> 1 ) << 2 ) + ( ( x & 1 ) << 1 ) + 1;
+ *out_ptr ++ = *in_ptr;
}
// Move to next output line
int16_t *src, *dest;
//static int16_t *extra_src = NULL, *extra_dest = NULL;
static int extra_src_samples = 0, extra_dest_samples = 0;
- int frequency_src = 0, frequency_dest = 0;
- int channels_src = 0, channels_dest = 0;
- int samples_src = 0, samples_dest = 0;
+ int frequency_src = *channels, frequency_dest = *channels;
+ int channels_src = *channels, channels_dest = *channels;
+ int samples_src = *samples, samples_dest = *samples;
int i, j;
+ double d = 0, s = 0;
mlt_frame_get_audio( this, &p_dest, format, &frequency_dest, &channels_dest, &samples_dest );
- //fprintf( stderr, "frame dest samples %d channels %d position %f\n", samples_dest, channels_dest, mlt_properties_get_position( mlt_frame_properties( this ), "position" ) );
+ //fprintf( stderr, "frame dest samples %d channels %d position %lld\n", samples_dest, channels_dest, mlt_properties_get_position( mlt_frame_properties( this ), "position" ) );
mlt_frame_get_audio( that, &p_src, format, &frequency_src, &channels_src, &samples_src );
//fprintf( stderr, "frame src samples %d channels %d\n", samples_src, channels_src );
+ src = p_src;
+ dest = p_dest;
if ( channels_src > 6 )
channels_src = 0;
if ( channels_dest > 6 )
}
else
src = p_src;
-#else
- src = p_src;
- dest = p_dest;
#endif
// determine number of samples to process
{
for ( j = 0; j < *channels; j++ )
{
- double d = (double) dest[ i * channels_dest + j ];
- double s = (double) src[ i * channels_src + j ];
+ if ( j < channels_dest )
+ d = (double) dest[ i * channels_dest + j ];
+ if ( j < channels_src )
+ s = (double) src[ i * channels_src + j ];
dest[ i * channels_dest + j ] = s * weight + d * ( 1.0 - weight );
}
}