/*****************************************************************************
- * chorus_flanger.c
+ * chorus_flanger: Basic chorus/flanger/delay audio filter
*****************************************************************************
- * Copyright (C) 2009 the VideoLAN team
+ * Copyright (C) 2009-12 VLC authors and VideoLAN
* $Id$
*
- * 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
- * the Free Software Foundation; either version 2 of the License, or
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation; either version 2.1 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.
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU Lesser 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.
+ * You should have received a copy of the GNU Lesser 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
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
{
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;
};
/*****************************************************************************
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 "
add_float( "sweep-rate", 6, 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,
+ add_float_with_range( "feedback-gain", 0.5, -0.9, 0.9,
+ N_("Feedback gain"), N_("Gain on Feedback loop"), true )
+ add_float_with_range( "wet-mix", 0.4, -0.999, 0.999,
N_("Wet mix"), N_("Level of delayed signal"), true )
- add_float_with_range( "dry-mix", 0.4, -0.999, 0.999, NULL,
+ add_float_with_range( "dry-mix", 0.4, -0.999, 0.999,
N_("Dry Mix"), N_("Level of input signal"), true )
set_capability( "audio filter", 0 )
set_callbacks( Open, Close )
static int Open( vlc_object_t *p_this )
{
filter_t *p_filter = (filter_t*)p_this;
- filter_sys_t *p_sys;
-
- if ( !AOUT_FMTS_SIMILAR( &p_filter->fmt_in.audio, &p_filter->fmt_out.audio ) )
- {
- msg_Err( p_filter, "input and output formats are not similar" );
- return VLC_EGENERIC;
- }
-
- if( p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 ||
- p_filter->fmt_out.audio.i_format != VLC_CODEC_FL32 )
- {
- p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
- p_filter->fmt_out.audio.i_format = VLC_CODEC_FL32;
- msg_Warn( p_filter, "bad input or output format" );
- }
-
- p_filter->pf_audio_filter = DoWork;
-
- p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
+ filter_sys_t *p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
if( !p_sys )
return VLC_ENOMEM;
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)
{
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() ) {
}
p_sys->i_sampleRate = p_filter->fmt_in.audio.i_rate;
+ p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
+ p_filter->fmt_out.audio = p_filter->fmt_in.audio;
+ p_filter->pf_audio_filter = DoWork;
+
return VLC_SUCCESS;
}
-
/**
* sanitize: Helper function to eliminate small amplitudes
* @param f_value pointer to value to clean
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;
}
/* 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;
}
}
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;
+}
projects / vlc / blobdiff