Commit 57dfca01 authored by Sukrit Sangwan's avatar Sukrit Sangwan Committed by Jean-Baptiste Kempf

Chorus: cleanup and quality fixes

Signed-off-by: default avatarJean-Baptiste Kempf <jb@videolan.org>
parent f5b61062
/***************************************************************************** /*****************************************************************************
* chorus_flanger.c * chorus_flanger: Basic chorus/flanger/delay audio filter
***************************************************************************** *****************************************************************************
* Copyright (C) 2009 the VideoLAN team * Copyright (C) 2009-12 the VideoLAN team
* $Id$ * $Id$
* *
* Author: Srikanth Raju < srikiraju at gmail dot com > * Authors: Srikanth Raju < srikiraju at gmail dot com >
* Sukrit Sangwan < sukritsangwan at gmail dot com >
* *
* This program is free software; you can redistribute it and/or modify * This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by * it under the terms of the GNU General Public License as published by
...@@ -21,13 +22,6 @@ ...@@ -21,13 +22,6 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
*****************************************************************************/ *****************************************************************************/
/**
* Basic chorus/flanger/delay audio filter
* This implements a variable delay filter for VLC. It has some issues with
* interpolation and sounding 'correct'.
*/
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
# include "config.h" # include "config.h"
#endif #endif
...@@ -47,6 +41,9 @@ ...@@ -47,6 +41,9 @@
static int Open ( vlc_object_t * ); static int Open ( vlc_object_t * );
static void Close ( vlc_object_t * ); static void Close ( vlc_object_t * );
static block_t *DoWork( filter_t *, block_t * ); static block_t *DoWork( filter_t *, block_t * );
static int paramCallback( vlc_object_t *, char const *, vlc_value_t ,
vlc_value_t , void * );
static int reallocate_buffer( filter_t *, filter_sys_t * );
struct filter_sys_t struct filter_sys_t
{ {
...@@ -57,15 +54,15 @@ struct filter_sys_t ...@@ -57,15 +54,15 @@ struct filter_sys_t
float f_wetLevel, f_dryLevel; float f_wetLevel, f_dryLevel;
float f_sweepDepth, f_sweepRate; float f_sweepDepth, f_sweepRate;
float f_step,f_offset; float f_offset;
int i_step,i_offset; int i_step;
float f_temp; float f_temp;
float f_sinMultiplier; float f_sinMultiplier;
/* This data is for the the circular queue which stores the samples. */ /* This data is for the the circular queue which stores the samples. */
int i_bufferLength; int i_bufferLength;
float * pf_delayLineStart, * pf_delayLineEnd; float * p_delayLineStart, * p_delayLineEnd;
float * pf_write; float * p_write;
}; };
/***************************************************************************** /*****************************************************************************
...@@ -80,7 +77,7 @@ vlc_module_begin () ...@@ -80,7 +77,7 @@ vlc_module_begin ()
set_category( CAT_AUDIO ) set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_AFILTER ) set_subcategory( SUBCAT_AUDIO_AFILTER )
add_shortcut( "delay" ) add_shortcut( "delay" )
add_float( "delay-time", 40, N_("Delay time"), add_float( "delay-time", 20, N_("Delay time"),
N_("Time in milliseconds of the average delay. Note average"), true ) N_("Time in milliseconds of the average delay. Note average"), true )
add_float( "sweep-depth", 6, N_("Sweep Depth"), add_float( "sweep-depth", 6, N_("Sweep Depth"),
N_("Time in milliseconds of the maximum sweep depth. Thus, the sweep " N_("Time in milliseconds of the maximum sweep depth. Thus, the sweep "
...@@ -144,6 +141,12 @@ static int Open( vlc_object_t *p_this ) ...@@ -144,6 +141,12 @@ static int Open( vlc_object_t *p_this )
p_sys->f_feedbackGain = var_CreateGetFloat( p_this, "feedback-gain" ); p_sys->f_feedbackGain = var_CreateGetFloat( p_this, "feedback-gain" );
p_sys->f_dryLevel = var_CreateGetFloat( p_this, "dry-mix" ); p_sys->f_dryLevel = var_CreateGetFloat( p_this, "dry-mix" );
p_sys->f_wetLevel = var_CreateGetFloat( p_this, "wet-mix" ); p_sys->f_wetLevel = var_CreateGetFloat( p_this, "wet-mix" );
var_AddCallback( p_this, "delay-time", paramCallback, p_sys );
var_AddCallback( p_this, "sweep-depth", paramCallback, p_sys );
var_AddCallback( p_this, "sweep-rate", paramCallback, p_sys );
var_AddCallback( p_this, "feedback-gain", paramCallback, p_sys );
var_AddCallback( p_this, "dry-mix", paramCallback, p_sys );
var_AddCallback( p_this, "wet-mix", paramCallback, p_sys );
if( p_sys->f_delayTime < 0.0) if( p_sys->f_delayTime < 0.0)
{ {
...@@ -176,27 +179,25 @@ static int Open( vlc_object_t *p_this ) ...@@ -176,27 +179,25 @@ static int Open( vlc_object_t *p_this )
p_sys->f_sweepRate, p_filter->fmt_in.audio.i_rate ); p_sys->f_sweepRate, p_filter->fmt_in.audio.i_rate );
if( p_sys->i_bufferLength <= 0 ) if( p_sys->i_bufferLength <= 0 )
{ {
msg_Err( p_filter, "Delay-time, Sampl rate or Channels was incorrect" ); msg_Err( p_filter, "Delay-time, Sample rate or Channels was incorrect" );
free(p_sys); free(p_sys);
return VLC_EGENERIC; return VLC_EGENERIC;
} }
p_sys->pf_delayLineStart = calloc( p_sys->i_bufferLength, sizeof( float ) ); p_sys->p_delayLineStart = calloc( p_sys->i_bufferLength, sizeof( float ) );
if( !p_sys->pf_delayLineStart ) if( !p_sys->p_delayLineStart )
{ {
free( p_sys ); free( p_sys );
return VLC_ENOMEM; return VLC_ENOMEM;
} }
p_sys->i_cumulative = 0; 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->i_step = p_sys->f_sweepRate > 0 ? 1 : 0;
p_sys->f_offset = 0; p_sys->f_offset = 0;
p_sys->i_offset = 0;
p_sys->f_temp = 0; p_sys->f_temp = 0;
p_sys->pf_delayLineEnd = p_sys->pf_delayLineStart + p_sys->i_bufferLength; p_sys->p_delayLineEnd = p_sys->p_delayLineStart + p_sys->i_bufferLength;
p_sys->pf_write = p_sys->pf_delayLineStart; p_sys->p_write = p_sys->p_delayLineStart;
if( p_sys->f_sweepDepth < small_value() || if( p_sys->f_sweepDepth < small_value() ||
p_filter->fmt_in.audio.i_rate < small_value() ) { p_filter->fmt_in.audio.i_rate < small_value() ) {
...@@ -211,7 +212,6 @@ static int Open( vlc_object_t *p_this ) ...@@ -211,7 +212,6 @@ static int Open( vlc_object_t *p_this )
return VLC_SUCCESS; return VLC_SUCCESS;
} }
/** /**
* sanitize: Helper function to eliminate small amplitudes * sanitize: Helper function to eliminate small amplitudes
* @param f_value pointer to value to clean * @param f_value pointer to value to clean
...@@ -239,28 +239,24 @@ static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf ) ...@@ -239,28 +239,24 @@ static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
float *p_out = (float*)p_in_buf->p_buffer; float *p_out = (float*)p_in_buf->p_buffer;
float *p_in = (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 *p_ptr, f_temp = 0;/* f_diff = 0, f_frac = 0;*/
/* Process each sample */ /* Process each sample */
for( unsigned i = 0; i < i_samples ; i++ ) for( unsigned i = 0; i < i_samples ; i++ )
{ {
/* Use a sine function as a oscillator wave. TODO */ /* Sine function as a oscillator wave to calculate sweep */
/* f_offset = sinf( ( p_sys->i_cumulative ) * p_sys->f_sinMultiplier ) * p_sys->i_cumulative += p_sys->i_step;
* (int)floor(p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000); p_sys->f_offset = sinf( (p_sys->i_cumulative) * p_sys->f_sinMultiplier )
*/ * floorf(p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000);
/* Triangle oscillator. Step using ints, because floats give rounding */
p_sys->i_offset+=p_sys->i_step;
p_sys->f_offset = p_sys->i_offset * p_sys->f_step;
if( abs( p_sys->i_step ) > 0 ) if( abs( p_sys->i_step ) > 0 )
{ {
if( p_sys->i_offset >= floor( p_sys->f_sweepDepth * if( p_sys->i_cumulative >= floor( p_sys->f_sweepDepth *
p_sys->i_sampleRate / p_sys->f_sweepRate )) p_sys->i_sampleRate / p_sys->f_sweepRate ))
{ {
p_sys->f_offset = i_maxOffset; p_sys->f_offset = i_maxOffset;
p_sys->i_step = -1 * ( p_sys->i_step ); p_sys->i_step = -1 * ( p_sys->i_step );
} }
if( p_sys->i_offset <= floor( -1 * p_sys->f_sweepDepth * if( p_sys->i_cumulative <= floor( -1 * p_sys->f_sweepDepth *
p_sys->i_sampleRate / p_sys->f_sweepRate ) ) p_sys->i_sampleRate / p_sys->f_sweepRate ) )
{ {
p_sys->f_offset = -i_maxOffset; p_sys->f_offset = -i_maxOffset;
...@@ -269,43 +265,45 @@ static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf ) ...@@ -269,43 +265,45 @@ static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
} }
/* Calculate position in delay */ /* Calculate position in delay */
int offset = floor( p_sys->f_offset ); int offset = floor( p_sys->f_offset );
pf_ptr = p_sys->pf_write + i_maxOffset * p_sys->i_channels + p_ptr = p_sys->p_write + ( i_maxOffset - offset ) * p_sys->i_channels;
offset * p_sys->i_channels;
/* Handle Overflow */ /* Handle Overflow */
if( pf_ptr < p_sys->pf_delayLineStart ) if( p_ptr < p_sys->p_delayLineStart )
{ {
pf_ptr += p_sys->i_bufferLength - p_sys->i_channels; p_ptr += p_sys->i_bufferLength - p_sys->i_channels;
} }
if( pf_ptr > p_sys->pf_delayLineEnd - 2*p_sys->i_channels ) if( p_ptr > p_sys->p_delayLineEnd - 2*p_sys->i_channels )
{ {
pf_ptr -= p_sys->i_bufferLength - p_sys->i_channels; p_ptr -= p_sys->i_bufferLength - p_sys->i_channels;
} }
/* For interpolation */ /* For interpolation */
f_frac = ( p_sys->f_offset - (int)p_sys->f_offset ); /* f_frac = ( p_sys->f_offset - (int)p_sys->f_offset );*/
for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ ) for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ )
{ {
f_diff = *( pf_ptr + p_sys->i_channels + i_chan ) /* if( p_ptr <= p_sys->p_delayLineStart + p_sys->i_channels )
- *( pf_ptr + i_chan ); f_diff = *(p_sys->p_delayLineEnd + i_chan) - p_ptr[i_chan];
f_temp = ( *( pf_ptr + i_chan ) );//+ f_diff * f_frac); else
f_diff = *( p_ptr - p_sys->i_channels + i_chan )
- p_ptr[i_chan];*/
f_temp = ( *( p_ptr + i_chan ) );//+ f_diff * f_frac;
/*Linear Interpolation. FIXME. This creates LOTS of noise */ /*Linear Interpolation. FIXME. This creates LOTS of noise */
sanitize(&f_temp); sanitize(&f_temp);
p_out[i_chan] = p_sys->f_dryLevel * p_in[i_chan] + p_out[i_chan] = p_sys->f_dryLevel * p_in[i_chan] +
p_sys->f_wetLevel * f_temp; p_sys->f_wetLevel * f_temp;
*( p_sys->pf_write + i_chan ) = p_in[i_chan] + *( p_sys->p_write + i_chan ) = p_in[i_chan] +
p_sys->f_feedbackGain * f_temp; p_sys->f_feedbackGain * f_temp;
} }
if( p_sys->pf_write == p_sys->pf_delayLineStart ) if( p_sys->p_write == p_sys->p_delayLineStart )
for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ ) for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ )
*( p_sys->pf_delayLineEnd - p_sys->i_channels + i_chan ) *( p_sys->p_delayLineEnd - p_sys->i_channels + i_chan )
= *( p_sys->pf_delayLineStart + i_chan ); = *( p_sys->p_delayLineStart + i_chan );
p_in += p_sys->i_channels; p_in += p_sys->i_channels;
p_out += p_sys->i_channels; p_out += p_sys->i_channels;
p_sys->pf_write += p_sys->i_channels; p_sys->p_write += p_sys->i_channels;
if( p_sys->pf_write == p_sys->pf_delayLineEnd - p_sys->i_channels ) if( p_sys->p_write == p_sys->p_delayLineEnd - p_sys->i_channels )
{ {
p_sys->pf_write = p_sys->pf_delayLineStart; p_sys->p_write = p_sys->p_delayLineStart;
} }
} }
...@@ -321,6 +319,98 @@ static void Close( vlc_object_t *p_this ) ...@@ -321,6 +319,98 @@ static void Close( vlc_object_t *p_this )
filter_t *p_filter = ( filter_t* )p_this; filter_t *p_filter = ( filter_t* )p_this;
filter_sys_t *p_sys = p_filter->p_sys; filter_sys_t *p_sys = p_filter->p_sys;
free( p_sys->pf_delayLineStart ); var_DelCallback( p_this, "delay-time", paramCallback, p_sys );
var_DelCallback( p_this, "sweep-depth", paramCallback, p_sys );
var_DelCallback( p_this, "sweep-rate", paramCallback, p_sys );
var_DelCallback( p_this, "feedback-gain", paramCallback, p_sys );
var_DelCallback( p_this, "wet-mix", paramCallback, p_sys );
var_DelCallback( p_this, "dry-mix", paramCallback, p_sys );
var_Destroy( p_this, "delay-time" );
var_Destroy( p_this, "sweep-depth" );
var_Destroy( p_this, "sweep-rate" );
var_Destroy( p_this, "feedback-gain" );
var_Destroy( p_this, "wet-mix" );
var_Destroy( p_this, "dry-mix" );
free( p_sys->p_delayLineStart );
free( p_sys ); free( p_sys );
} }
/******************************************************************************
* Callback to update parameters on the fly
******************************************************************************/
static int paramCallback( vlc_object_t *p_this, char const *psz_var,
vlc_value_t oldval, vlc_value_t newval, void *p_data )
{
filter_t *p_filter = (filter_t *)p_this;
filter_sys_t *p_sys = (filter_sys_t *) p_data;
if( !strncmp( psz_var, "delay-time", 10 ) )
{
/* if invalid value pretend everything is OK without updating value */
if( newval.f_float < 0 )
return VLC_SUCCESS;
p_sys->f_delayTime = newval.f_float;
if( !reallocate_buffer( p_filter, p_sys ) )
{
p_sys->f_delayTime = oldval.f_float;
p_sys->i_bufferLength = p_sys->i_channels * ( (int)
( ( p_sys->f_delayTime + p_sys->f_sweepDepth ) *
p_filter->fmt_in.audio.i_rate/1000 ) + 1 );
}
}
else if( !strncmp( psz_var, "sweep-depth", 11 ) )
{
if( newval.f_float < 0 || newval.f_float > p_sys->f_delayTime)
return VLC_SUCCESS;
p_sys->f_sweepDepth = newval.f_float;
if( !reallocate_buffer( p_filter, p_sys ) )
{
p_sys->f_sweepDepth = oldval.f_float;
p_sys->i_bufferLength = p_sys->i_channels * ( (int)
( ( p_sys->f_delayTime + p_sys->f_sweepDepth ) *
p_filter->fmt_in.audio.i_rate/1000 ) + 1 );
}
}
else if( !strncmp( psz_var, "sweep-rate", 10 ) )
{
if( newval.f_float > p_sys->f_sweepDepth )
return VLC_SUCCESS;
p_sys->f_sweepRate = newval.f_float;
/* Calculate new f_sinMultiplier */
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 ) ;
}
}
else if( !strncmp( psz_var, "feedback-gain", 13 ) )
p_sys->f_feedbackGain = newval.f_float;
else if( !strncmp( psz_var, "wet-mix", 7 ) )
p_sys->f_wetLevel = newval.f_float;
else if( !strncmp( psz_var, "dry-mix", 7 ) )
p_sys->f_dryLevel = newval.f_float;
return VLC_SUCCESS;
}
static int reallocate_buffer( filter_t *p_filter, filter_sys_t *p_sys )
{
p_sys->i_bufferLength = p_sys->i_channels * ( (int)( ( p_sys->f_delayTime
+ p_sys->f_sweepDepth ) * p_filter->fmt_in.audio.i_rate/1000 ) + 1 );
float *temp = realloc( p_sys->p_delayLineStart, p_sys->i_bufferLength );
if( unlikely( !temp ) )
{
msg_Err( p_filter, "Couldnt reallocate buffer for new delay." );
return 0;
}
free( p_sys->p_delayLineStart );
p_sys->p_delayLineStart = temp;
p_sys->p_delayLineEnd = p_sys->p_delayLineStart + p_sys->i_bufferLength;
free( temp );
return 1;
}
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