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Commit 57dfca01 authored by Sukrit Sangwan's avatar Sukrit Sangwan Committed by Jean-Baptiste Kempf
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Chorus: cleanup and quality fixes 

Signed-off-by: default avatarJean-Baptiste Kempf <jb@videolan.org>
parent f5b61062
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Showing with 142 additions and 52 deletions
modules/audio_filter/chorus_flanger.c
View file @ 57dfca01
/*****************************************************************************
* chorus_flanger.c
* chorus_flanger: Basic chorus/flanger/delay audio filter
*****************************************************************************
* Copyright (C) 2009 the VideoLAN team
* Copyright (C) 2009-12 the VideoLAN team
* $Id$
*
* 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
* it under the terms of the GNU General Public License as published by
......@@ -21,13 +22,6 @@
* 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
# include "config.h"
#endif
......@@ -47,6 +41,9 @@
static int Open ( vlc_object_t * );
static void Close ( vlc_object_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
{
......@@ -57,15 +54,15 @@ struct filter_sys_t
float f_wetLevel, f_dryLevel;
float f_sweepDepth, f_sweepRate;
float f_step,f_offset;
int i_step,i_offset;
float f_offset;
int i_step;
float f_temp;
float f_sinMultiplier;
/* This data is for the the circular queue which stores the samples. */
int i_bufferLength;
float * pf_delayLineStart, * pf_delayLineEnd;
float * pf_write;
float * p_delayLineStart, * p_delayLineEnd;
float * p_write;
};
/*****************************************************************************
......@@ -80,7 +77,7 @@ vlc_module_begin ()
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_AFILTER )
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 )
add_float( "sweep-depth", 6, N_("Sweep Depth"),
N_("Time in milliseconds of the maximum sweep depth. Thus, the sweep "
......@@ -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_dryLevel = var_CreateGetFloat( p_this, "dry-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)
{
......@@ -176,27 +179,25 @@ static int Open( vlc_object_t *p_this )
p_sys->f_sweepRate, p_filter->fmt_in.audio.i_rate );
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);
return VLC_EGENERIC;
}
p_sys->pf_delayLineStart = calloc( p_sys->i_bufferLength, sizeof( float ) );
if( !p_sys->pf_delayLineStart )
p_sys->p_delayLineStart = calloc( p_sys->i_bufferLength, sizeof( float ) );
if( !p_sys->p_delayLineStart )
{
free( p_sys );
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;
p_sys->p_delayLineEnd = p_sys->p_delayLineStart + p_sys->i_bufferLength;
p_sys->p_write = p_sys->p_delayLineStart;
if( p_sys->f_sweepDepth < small_value() ||
p_filter->fmt_in.audio.i_rate < small_value() ) {
......@@ -211,7 +212,6 @@ static int Open( vlc_object_t *p_this )
return VLC_SUCCESS;
}
/**
* sanitize: Helper function to eliminate small amplitudes
* @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 )
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 *p_ptr, f_temp = 0;/* f_diff = 0, f_frac = 0;*/
/* Process each sample */
for( unsigned i = 0; i < i_samples ; i++ )
{
/* Use a sine function as a oscillator wave. TODO */
/* f_offset = sinf( ( p_sys->i_cumulative ) * p_sys->f_sinMultiplier ) *
* (int)floor(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;
/* Sine function as a oscillator wave to calculate sweep */
p_sys->i_cumulative += p_sys->i_step;
p_sys->f_offset = sinf( (p_sys->i_cumulative) * p_sys->f_sinMultiplier )
* floorf(p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000);
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->f_offset = i_maxOffset;
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->f_offset = -i_maxOffset;
......@@ -269,43 +265,45 @@ 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;
p_ptr = p_sys->p_write + ( i_maxOffset - offset ) * p_sys->i_channels;
/* 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 */
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++ )
{
f_diff = *( pf_ptr + p_sys->i_channels + i_chan )
- *( pf_ptr + i_chan );
f_temp = ( *( pf_ptr + i_chan ) );//+ f_diff * f_frac);
/* if( p_ptr <= p_sys->p_delayLineStart + p_sys->i_channels )
f_diff = *(p_sys->p_delayLineEnd + i_chan) - p_ptr[i_chan];
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 */
sanitize(&f_temp);
p_out[i_chan] = p_sys->f_dryLevel * p_in[i_chan] +
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;
}
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++ )
*( p_sys->pf_delayLineEnd - p_sys->i_channels + i_chan )
= *( p_sys->pf_delayLineStart + i_chan );
*( p_sys->p_delayLineEnd - p_sys->i_channels + i_chan )
= *( p_sys->p_delayLineStart + i_chan );
p_in += p_sys->i_channels;
p_out += p_sys->i_channels;
p_sys->pf_write += p_sys->i_channels;
if( p_sys->pf_write == p_sys->pf_delayLineEnd - p_sys->i_channels )
p_sys->p_write += 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 )
filter_t *p_filter = ( filter_t* )p_this;
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 );
}
/******************************************************************************
* 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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