Commit 535459a8 authored by Srikanth Raju's avatar Srikanth Raju Committed by Rémi Denis-Courmont

Chorus/Flanger audio filter Based on basic variable delay filter

Signed-off-by: Rémi Denis-Courmont's avatarRémi Denis-Courmont <remi@remlab.net>
parent 5b00f5cf
......@@ -4,3 +4,4 @@ SOURCES_normvol = normvol.c
SOURCES_audio_format = format.c
SOURCES_param_eq = param_eq.c
SOURCES_scaletempo = scaletempo.c
SOURCES_chorus_flanger = chorus_flanger.c
/*****************************************************************************
* chorus_flanger.c
*****************************************************************************
* Copyright (C) 2009 the VideoLAN team
* $Id$
*
* Author: Srikanth Raju < srikiraju 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
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* 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
#include <math.h>
#include <vlc_common.h>
#include <vlc_plugin.h>
#include <vlc_aout.h>
/*****************************************************************************
* Local prototypes
*****************************************************************************/
static int Open ( vlc_object_t * );
static void Close ( vlc_object_t * );
static void DoWork ( aout_instance_t * , aout_filter_t *,
aout_buffer_t * , aout_buffer_t * );
struct aout_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;
float f_sweepDepth, f_sweepRate;
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;
float * pf_delayLineStart, * pf_delayLineEnd;
float * pf_write;
};
/*****************************************************************************
* Module descriptor
*****************************************************************************/
vlc_module_begin ()
set_description( N_("Sound Delay") )
set_shortname( N_("delay") )
set_category( CAT_AUDIO )
set_subcategory( SUBCAT_AUDIO_AFILTER )
add_shortcut( "delay" )
add_float( "delay-time", 40, NULL, N_("Delay time"),
N_("Time in milliseconds of the average delay. Note average"), true )
add_float( "sweep-depth", 6, NULL, N_("Sweep Depth"),
N_("Time in milliseconds of the maximum sweep depth. Thus, the sweep "
"range will be delay-time +/- sweep-depth."), true )
add_float( "sweep-rate", 6, NULL, N_("Sweep Rate"),
N_("Rate of change of sweep depth in milliseconds shift per second "
"of play"), true )
add_float_with_range( "feedback-gain", 0.5, -0.9, 0.9, NULL,
N_("Feedback Gain"), N_("Gain on Feedback loop"), true )
add_float_with_range( "wet-mix", 0.4, -0.999, 0.999, NULL,
N_("Wet mix"), N_("Level of delayed signal"), true )
add_float_with_range( "dry-mix", 0.4, -0.999, 0.999, NULL,
N_("Dry Mix"), N_("Level of input signal"), true )
set_capability( "audio filter", 0 )
set_callbacks( Open, Close )
vlc_module_end ()
/**
* small_value: Helper function
* return high pass cutoff
*/
static inline float small_value()
{
/* allows for 2^-24, should be enough for 24-bit DACs at least */
return ( 1.0 / 16777216.0 );
}
/**
* Open: initialize and create stuff
* @param p_this
*/
static int Open( vlc_object_t *p_this )
{
aout_filter_t *p_filter = (aout_filter_t*)p_this;
aout_filter_sys_t *p_sys;
if ( !AOUT_FMTS_SIMILAR( &p_filter->input, &p_filter->output ) )
{
msg_Err( p_filter, "input and output formats are not similar" );
return VLC_EGENERIC;
}
if( p_filter->input.i_format != VLC_CODEC_FL32 ||
p_filter->output.i_format != VLC_CODEC_FL32 )
{
p_filter->input.i_format = VLC_CODEC_FL32;
p_filter->output.i_format = VLC_CODEC_FL32;
msg_Warn( p_filter, "bad input or output format" );
}
p_filter->pf_do_work = DoWork;
p_filter->b_in_place = true;
p_sys = p_filter->p_sys = malloc( sizeof( aout_filter_sys_t ) );
if( !p_sys )
return VLC_ENOMEM;
p_sys->i_channels = aout_FormatNbChannels( &p_filter->input );
p_sys->f_delayTime = var_CreateGetFloat( p_this, "delay-time" );
p_sys->f_sweepDepth = var_CreateGetFloat( p_this, "sweep-depth" );
p_sys->f_sweepRate = var_CreateGetFloat( p_this, "sweep-rate" );
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" );
if( p_sys->f_delayTime < 0.0)
{
msg_Err( p_filter, "Delay Time is invalid" );
free(p_sys);
return VLC_EGENERIC;
}
if( p_sys->f_sweepDepth > p_sys->f_delayTime || p_sys->f_sweepDepth < 0.0 )
{
msg_Err( p_filter, "Sweep Depth is invalid" );
free( p_sys );
return VLC_EGENERIC;
}
if( p_sys->f_sweepRate < 0.0 )
{
msg_Err( p_filter, "Sweep Rate is invalid" );
free( p_sys );
return VLC_EGENERIC;
}
/* Max delay = delay + depth. Min = delay - depth */
p_sys->i_bufferLength = p_sys->i_channels * ( (int)( ( p_sys->f_delayTime
+ p_sys->f_sweepDepth ) * p_filter->input.i_rate/1000 ) + 1 );
msg_Dbg( p_filter , "Buffer length:%d, Channels:%d, Sweep Depth:%f, Delay "
"time:%f, Sweep Rate:%f, Sample Rate: %d", p_sys->i_bufferLength,
p_sys->i_channels, p_sys->f_sweepDepth, p_sys->f_delayTime,
p_sys->f_sweepRate, p_filter->input.i_rate );
if( p_sys->i_bufferLength <= 0 )
{
msg_Err( p_filter, "Delay-time, Sampl 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 )
{
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;
if( p_sys->f_sweepDepth < small_value() ||
p_filter->input.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->input.i_rate ) ;
}
p_sys->i_sampleRate = p_filter->input.i_rate;
return VLC_SUCCESS;
}
/**
* sanitize: Helper function to eliminate small amplitudes
* @param f_value pointer to value to clean
*/
static inline void sanitize( float * f_value )
{
if ( fabs( *f_value ) < small_value() )
*f_value = 0.0f;
}
/**
* DoWork : delays and finds the value of the current frame
* @param p_aout Audio output object
* @param p_filter This filter object
* @param p_in_buf Input buffer
* @param p_out_buf Output buffer
*/
static void DoWork( aout_instance_t *p_aout, aout_filter_t *p_filter,
aout_buffer_t *p_in_buf, aout_buffer_t *p_out_buf )
{
struct aout_filter_sys_t *p_sys = p_filter->p_sys;
int i, i_chan;
int i_samples = p_in_buf->i_nb_samples; /* Gives the number of samples */
int i_maxOffset = (int)floor( p_sys->f_sweepDepth * p_sys->i_sampleRate /
1000 ); /*maximum number of samples to offset in buffer */
float *p_out = (float*)p_out_buf->p_buffer;
float *p_in = (float*)p_in_buf->p_buffer;
float *pf_ptr, f_diff = 0, f_frac = 0, f_temp = 0 ;
p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
/* Process each sample */
for( 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;
if( abs( p_sys->i_step ) > 0 )
{
if( p_sys->i_offset >= 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 *
p_sys->i_sampleRate / p_sys->f_sweepRate ) )
{
p_sys->f_offset = -i_maxOffset;
p_sys->i_step = -1 * ( p_sys->i_step );
}
}
/* Calculate position in delay */
pf_ptr = p_sys->pf_write + i_maxOffset * p_sys->i_channels +
(int)( floor( p_sys->f_offset ) ) * p_sys->i_channels;
/* Handle Overflow */
if( pf_ptr < p_sys->pf_delayLineStart )
{
pf_ptr += p_sys->i_bufferLength - p_sys->i_channels;
}
if( pf_ptr > p_sys->pf_delayLineEnd - 2*p_sys->i_channels )
{
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);
/*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->f_feedbackGain * f_temp;
}
if( p_sys->pf_write == p_sys->pf_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_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->pf_write = p_sys->pf_delayLineStart;
}
}
return;
}
/**
* Close: Destructor
* @param p_this pointer to this filter object
*/
static void Close( vlc_object_t *p_this )
{
aout_filter_t *p_filter = ( aout_filter_t* )p_this;
aout_filter_sys_t *p_sys = p_filter->p_sys;
free( p_sys->pf_delayLineStart );
free( p_sys );
}
Markdown is supported
0%
or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment