1 /*****************************************************************************
3 *****************************************************************************
4 * Copyright (C) 2009 the VideoLAN team
7 * Author: Srikanth Raju < srikiraju at gmail dot com >
9 * This program is free software; you can redistribute it and/or modify
10 * it under the terms of the GNU General Public License as published by
11 * the Free Software Foundation; either version 2 of the License, or
12 * (at your option) any later version.
14 * This program is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
17 * GNU General Public License for more details.
19 * You should have received a copy of the GNU General Public License
20 * along with this program; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
22 *****************************************************************************/
25 * Basic chorus/flanger/delay audio filter
26 * This implements a variable delay filter for VLC. It has some issues with
27 * interpolation and sounding 'correct'.
37 #include <vlc_common.h>
38 #include <vlc_plugin.h>
41 #include <vlc_filter.h>
43 /*****************************************************************************
45 *****************************************************************************/
47 static int Open ( vlc_object_t * );
48 static void Close ( vlc_object_t * );
49 static block_t *DoWork( filter_t *, block_t * );
53 /* TODO: Cleanup and optimise */
55 int i_channels, i_sampleRate;
56 float f_delayTime, f_feedbackGain; /* delayTime is in milliseconds */
57 float f_wetLevel, f_dryLevel;
58 float f_sweepDepth, f_sweepRate;
60 float f_step,f_offset;
63 float f_sinMultiplier;
65 /* This data is for the the circular queue which stores the samples. */
67 float * pf_delayLineStart, * pf_delayLineEnd;
71 /*****************************************************************************
73 *****************************************************************************/
77 set_description( N_("Sound Delay") )
78 set_shortname( N_("Delay") )
79 set_help( N_("Add a delay effect to the sound") )
80 set_category( CAT_AUDIO )
81 set_subcategory( SUBCAT_AUDIO_AFILTER )
82 add_shortcut( "delay" )
83 add_float( "delay-time", 40, NULL, N_("Delay time"),
84 N_("Time in milliseconds of the average delay. Note average"), true )
85 add_float( "sweep-depth", 6, NULL, N_("Sweep Depth"),
86 N_("Time in milliseconds of the maximum sweep depth. Thus, the sweep "
87 "range will be delay-time +/- sweep-depth."), true )
88 add_float( "sweep-rate", 6, NULL, N_("Sweep Rate"),
89 N_("Rate of change of sweep depth in milliseconds shift per second "
91 add_float_with_range( "feedback-gain", 0.5, -0.9, 0.9, NULL,
92 N_("Feedback Gain"), N_("Gain on Feedback loop"), true )
93 add_float_with_range( "wet-mix", 0.4, -0.999, 0.999, NULL,
94 N_("Wet mix"), N_("Level of delayed signal"), true )
95 add_float_with_range( "dry-mix", 0.4, -0.999, 0.999, NULL,
96 N_("Dry Mix"), N_("Level of input signal"), true )
97 set_capability( "audio filter", 0 )
98 set_callbacks( Open, Close )
102 * small_value: Helper function
103 * return high pass cutoff
105 static inline float small_value()
107 /* allows for 2^-24, should be enough for 24-bit DACs at least */
108 return ( 1.0 / 16777216.0 );
112 * Open: initialize and create stuff
115 static int Open( vlc_object_t *p_this )
117 filter_t *p_filter = (filter_t*)p_this;
120 if ( !AOUT_FMTS_SIMILAR( &p_filter->fmt_in.audio, &p_filter->fmt_out.audio ) )
122 msg_Err( p_filter, "input and output formats are not similar" );
126 if( p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 ||
127 p_filter->fmt_out.audio.i_format != VLC_CODEC_FL32 )
129 p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
130 p_filter->fmt_out.audio.i_format = VLC_CODEC_FL32;
131 msg_Warn( p_filter, "bad input or output format" );
134 p_filter->pf_audio_filter = DoWork;
136 p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
140 p_sys->i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
141 p_sys->f_delayTime = var_CreateGetFloat( p_this, "delay-time" );
142 p_sys->f_sweepDepth = var_CreateGetFloat( p_this, "sweep-depth" );
143 p_sys->f_sweepRate = var_CreateGetFloat( p_this, "sweep-rate" );
144 p_sys->f_feedbackGain = var_CreateGetFloat( p_this, "feedback-gain" );
145 p_sys->f_dryLevel = var_CreateGetFloat( p_this, "dry-mix" );
146 p_sys->f_wetLevel = var_CreateGetFloat( p_this, "wet-mix" );
148 if( p_sys->f_delayTime < 0.0)
150 msg_Err( p_filter, "Delay Time is invalid" );
155 if( p_sys->f_sweepDepth > p_sys->f_delayTime || p_sys->f_sweepDepth < 0.0 )
157 msg_Err( p_filter, "Sweep Depth is invalid" );
162 if( p_sys->f_sweepRate < 0.0 )
164 msg_Err( p_filter, "Sweep Rate is invalid" );
169 /* Max delay = delay + depth. Min = delay - depth */
170 p_sys->i_bufferLength = p_sys->i_channels * ( (int)( ( p_sys->f_delayTime
171 + p_sys->f_sweepDepth ) * p_filter->fmt_in.audio.i_rate/1000 ) + 1 );
173 msg_Dbg( p_filter , "Buffer length:%d, Channels:%d, Sweep Depth:%f, Delay "
174 "time:%f, Sweep Rate:%f, Sample Rate: %d", p_sys->i_bufferLength,
175 p_sys->i_channels, p_sys->f_sweepDepth, p_sys->f_delayTime,
176 p_sys->f_sweepRate, p_filter->fmt_in.audio.i_rate );
177 if( p_sys->i_bufferLength <= 0 )
179 msg_Err( p_filter, "Delay-time, Sampl rate or Channels was incorrect" );
184 p_sys->pf_delayLineStart = calloc( p_sys->i_bufferLength, sizeof( float ) );
185 if( !p_sys->pf_delayLineStart )
191 p_sys->i_cumulative = 0;
192 p_sys->f_step = p_sys->f_sweepRate / 1000.0;
193 p_sys->i_step = p_sys->f_sweepRate > 0 ? 1 : 0;
198 p_sys->pf_delayLineEnd = p_sys->pf_delayLineStart + p_sys->i_bufferLength;
199 p_sys->pf_write = p_sys->pf_delayLineStart;
201 if( p_sys->f_sweepDepth < small_value() ||
202 p_filter->fmt_in.audio.i_rate < small_value() ) {
203 p_sys->f_sinMultiplier = 0.0;
206 p_sys->f_sinMultiplier = 11 * p_sys->f_sweepRate /
207 ( 7 * p_sys->f_sweepDepth * p_filter->fmt_in.audio.i_rate ) ;
209 p_sys->i_sampleRate = p_filter->fmt_in.audio.i_rate;
216 * sanitize: Helper function to eliminate small amplitudes
217 * @param f_value pointer to value to clean
219 static inline void sanitize( float * f_value )
221 if ( fabs( *f_value ) < small_value() )
227 * DoWork : delays and finds the value of the current frame
228 * @param p_filter This filter object
229 * @param p_in_buf Input buffer
230 * @return Output buffer
232 static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
234 struct filter_sys_t *p_sys = p_filter->p_sys;
236 unsigned i_samples = p_in_buf->i_nb_samples; /* number of samples */
237 /* maximum number of samples to offset in buffer */
238 int i_maxOffset = floor( p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000 );
239 float *p_out = (float*)p_in_buf->p_buffer;
240 float *p_in = (float*)p_in_buf->p_buffer;
242 float *pf_ptr, f_diff = 0, f_frac = 0, f_temp = 0 ;
244 /* Process each sample */
245 for( unsigned i = 0; i < i_samples ; i++ )
247 /* Use a sine function as a oscillator wave. TODO */
248 /* f_offset = sinf( ( p_sys->i_cumulative ) * p_sys->f_sinMultiplier ) *
249 * (int)floor(p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000);
252 /* Triangle oscillator. Step using ints, because floats give rounding */
253 p_sys->i_offset+=p_sys->i_step;
254 p_sys->f_offset = p_sys->i_offset * p_sys->f_step;
255 if( abs( p_sys->i_step ) > 0 )
257 if( p_sys->i_offset >= floor( p_sys->f_sweepDepth *
258 p_sys->i_sampleRate / p_sys->f_sweepRate ))
260 p_sys->f_offset = i_maxOffset;
261 p_sys->i_step = -1 * ( p_sys->i_step );
263 if( p_sys->i_offset <= floor( -1 * p_sys->f_sweepDepth *
264 p_sys->i_sampleRate / p_sys->f_sweepRate ) )
266 p_sys->f_offset = -i_maxOffset;
267 p_sys->i_step = -1 * ( p_sys->i_step );
270 /* Calculate position in delay */
271 int offset = floor( p_sys->f_offset );
272 pf_ptr = p_sys->pf_write + i_maxOffset * p_sys->i_channels +
273 offset * p_sys->i_channels;
275 /* Handle Overflow */
276 if( pf_ptr < p_sys->pf_delayLineStart )
278 pf_ptr += p_sys->i_bufferLength - p_sys->i_channels;
280 if( pf_ptr > p_sys->pf_delayLineEnd - 2*p_sys->i_channels )
282 pf_ptr -= p_sys->i_bufferLength - p_sys->i_channels;
284 /* For interpolation */
285 f_frac = ( p_sys->f_offset - (int)p_sys->f_offset );
286 for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ )
288 f_diff = *( pf_ptr + p_sys->i_channels + i_chan )
289 - *( pf_ptr + i_chan );
290 f_temp = ( *( pf_ptr + i_chan ) );//+ f_diff * f_frac);
291 /*Linear Interpolation. FIXME. This creates LOTS of noise */
293 p_out[i_chan] = p_sys->f_dryLevel * p_in[i_chan] +
294 p_sys->f_wetLevel * f_temp;
295 *( p_sys->pf_write + i_chan ) = p_in[i_chan] +
296 p_sys->f_feedbackGain * f_temp;
298 if( p_sys->pf_write == p_sys->pf_delayLineStart )
299 for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ )
300 *( p_sys->pf_delayLineEnd - p_sys->i_channels + i_chan )
301 = *( p_sys->pf_delayLineStart + i_chan );
303 p_in += p_sys->i_channels;
304 p_out += p_sys->i_channels;
305 p_sys->pf_write += p_sys->i_channels;
306 if( p_sys->pf_write == p_sys->pf_delayLineEnd - p_sys->i_channels )
308 p_sys->pf_write = p_sys->pf_delayLineStart;
317 * @param p_this pointer to this filter object
319 static void Close( vlc_object_t *p_this )
321 filter_t *p_filter = ( filter_t* )p_this;
322 filter_sys_t *p_sys = p_filter->p_sys;
324 free( p_sys->pf_delayLineStart );