[vlc-devel] [PATCH] chorus_flanger.c : bit of cleanup

[Jean-Baptiste Kempf]

I think most of this code you remove was pending for future
improvements. Therefore, I am not sure this is a good idea to remove
them. See FIXME and TODO in the code.

On Wed, Mar 07, 2012 at 03:54:12PM -0800, Sukrit Sangwan wrote :
> 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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-- 
Best regards,

-- 
Jean-Baptiste Kempf
http://www.jbkempf.com/ - +33 672 704 734
Sent from my Electronic Device