[vlc-devel] [PATCH] chorus_flanger.c : bit of cleanup
Sukrit Sangwan
sukritsangwan at gmail.com
Thu Mar 8 00:54:12 CET 2012
diff --git a/modules/audio_filter/chorus_flanger.c
b/modules/audio_filter/chorus_flanger.c
index efeade6..f85451d 100644
--- a/modules/audio_filter/chorus_flanger.c
+++ b/modules/audio_filter/chorus_flanger.c
@@ -51,7 +51,6 @@ static block_t *DoWork( filter_t *, block_t * );
struct filter_sys_t
{
/* TODO: Cleanup and optimise */
- int i_cumulative;
int i_channels, i_sampleRate;
float f_delayTime, f_feedbackGain; /* delayTime is in milliseconds */
float f_wetLevel, f_dryLevel;
@@ -59,8 +58,6 @@ struct filter_sys_t
float f_step,f_offset;
int i_step,i_offset;
- float f_temp;
- float f_sinMultiplier;
/* This data is for the the circular queue which stores the samples. */
int i_bufferLength;
@@ -188,24 +185,14 @@ static int Open( vlc_object_t *p_this )
return VLC_ENOMEM;
}
- p_sys->i_cumulative = 0;
p_sys->f_step = p_sys->f_sweepRate / 1000.0;
p_sys->i_step = p_sys->f_sweepRate > 0 ? 1 : 0;
p_sys->f_offset = 0;
p_sys->i_offset = 0;
- p_sys->f_temp = 0;
p_sys->pf_delayLineEnd = p_sys->pf_delayLineStart +
p_sys->i_bufferLength;
p_sys->pf_write = p_sys->pf_delayLineStart;
- if( p_sys->f_sweepDepth < small_value() ||
- p_filter->fmt_in.audio.i_rate < small_value() ) {
- p_sys->f_sinMultiplier = 0.0;
- }
- else {
- p_sys->f_sinMultiplier = 11 * p_sys->f_sweepRate /
- ( 7 * p_sys->f_sweepDepth * p_filter->fmt_in.audio.i_rate ) ;
- }
p_sys->i_sampleRate = p_filter->fmt_in.audio.i_rate;
return VLC_SUCCESS;
@@ -239,7 +226,7 @@ static block_t *DoWork( filter_t *p_filter, block_t
*p_in_buf )
float *p_out = (float*)p_in_buf->p_buffer;
float *p_in = (float*)p_in_buf->p_buffer;
- float *pf_ptr, f_diff = 0, f_frac = 0, f_temp = 0 ;
+ float *pf_ptr, f_temp = 0 ;
/* Process each sample */
for( unsigned i = 0; i < i_samples ; i++ )
@@ -269,8 +256,7 @@ static block_t *DoWork( filter_t *p_filter, block_t
*p_in_buf )
}
/* Calculate position in delay */
int offset = floor( p_sys->f_offset );
- pf_ptr = p_sys->pf_write + i_maxOffset * p_sys->i_channels +
- offset * p_sys->i_channels;
+ pf_ptr = p_sys->pf_write + (i_maxOffset + offset) *
p_sys->i_channels;
/* Handle Overflow */
if( pf_ptr < p_sys->pf_delayLineStart )
@@ -282,12 +268,9 @@ static block_t *DoWork( filter_t *p_filter, block_t
*p_in_buf )
pf_ptr -= p_sys->i_bufferLength - p_sys->i_channels;
}
/* For interpolation */
- f_frac = ( p_sys->f_offset - (int)p_sys->f_offset );
for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ )
{
- f_diff = *( pf_ptr + p_sys->i_channels + i_chan )
- - *( pf_ptr + i_chan );
- f_temp = ( *( pf_ptr + i_chan ) );//+ f_diff * f_frac);
+ f_temp = *( pf_ptr + i_chan );
/*Linear Interpolation. FIXME. This creates LOTS of noise */
sanitize(&f_temp);
p_out[i_chan] = p_sys->f_dryLevel * p_in[i_chan] +
--
Regards
Sukrit Sangwan
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