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_category( CAT_AUDIO )
80 set_subcategory( SUBCAT_AUDIO_AFILTER )
81 add_shortcut( "delay" )
82 add_float( "delay-time", 40, NULL, N_("Delay time"),
83 N_("Time in milliseconds of the average delay. Note average"), true )
84 add_float( "sweep-depth", 6, NULL, N_("Sweep Depth"),
85 N_("Time in milliseconds of the maximum sweep depth. Thus, the sweep "
86 "range will be delay-time +/- sweep-depth."), true )
87 add_float( "sweep-rate", 6, NULL, N_("Sweep Rate"),
88 N_("Rate of change of sweep depth in milliseconds shift per second "
90 add_float_with_range( "feedback-gain", 0.5, -0.9, 0.9, NULL,
91 N_("Feedback Gain"), N_("Gain on Feedback loop"), true )
92 add_float_with_range( "wet-mix", 0.4, -0.999, 0.999, NULL,
93 N_("Wet mix"), N_("Level of delayed signal"), true )
94 add_float_with_range( "dry-mix", 0.4, -0.999, 0.999, NULL,
95 N_("Dry Mix"), N_("Level of input signal"), true )
96 set_capability( "audio filter2", 0 )
97 set_callbacks( Open, Close )
101 * small_value: Helper function
102 * return high pass cutoff
104 static inline float small_value()
106 /* allows for 2^-24, should be enough for 24-bit DACs at least */
107 return ( 1.0 / 16777216.0 );
111 * Open: initialize and create stuff
114 static int Open( vlc_object_t *p_this )
116 filter_t *p_filter = (filter_t*)p_this;
119 if ( !AOUT_FMTS_SIMILAR( &p_filter->fmt_in.audio, &p_filter->fmt_out.audio ) )
121 msg_Err( p_filter, "input and output formats are not similar" );
125 if( p_filter->fmt_in.audio.i_format != VLC_CODEC_FL32 ||
126 p_filter->fmt_out.audio.i_format != VLC_CODEC_FL32 )
128 p_filter->fmt_in.audio.i_format = VLC_CODEC_FL32;
129 p_filter->fmt_out.audio.i_format = VLC_CODEC_FL32;
130 msg_Warn( p_filter, "bad input or output format" );
133 p_filter->pf_audio_filter = DoWork;
135 p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
139 p_sys->i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
140 p_sys->f_delayTime = var_CreateGetFloat( p_this, "delay-time" );
141 p_sys->f_sweepDepth = var_CreateGetFloat( p_this, "sweep-depth" );
142 p_sys->f_sweepRate = var_CreateGetFloat( p_this, "sweep-rate" );
143 p_sys->f_feedbackGain = var_CreateGetFloat( p_this, "feedback-gain" );
144 p_sys->f_dryLevel = var_CreateGetFloat( p_this, "dry-mix" );
145 p_sys->f_wetLevel = var_CreateGetFloat( p_this, "wet-mix" );
147 if( p_sys->f_delayTime < 0.0)
149 msg_Err( p_filter, "Delay Time is invalid" );
154 if( p_sys->f_sweepDepth > p_sys->f_delayTime || p_sys->f_sweepDepth < 0.0 )
156 msg_Err( p_filter, "Sweep Depth is invalid" );
161 if( p_sys->f_sweepRate < 0.0 )
163 msg_Err( p_filter, "Sweep Rate is invalid" );
168 /* Max delay = delay + depth. Min = delay - depth */
169 p_sys->i_bufferLength = p_sys->i_channels * ( (int)( ( p_sys->f_delayTime
170 + p_sys->f_sweepDepth ) * p_filter->fmt_in.audio.i_rate/1000 ) + 1 );
172 msg_Dbg( p_filter , "Buffer length:%d, Channels:%d, Sweep Depth:%f, Delay "
173 "time:%f, Sweep Rate:%f, Sample Rate: %d", p_sys->i_bufferLength,
174 p_sys->i_channels, p_sys->f_sweepDepth, p_sys->f_delayTime,
175 p_sys->f_sweepRate, p_filter->fmt_in.audio.i_rate );
176 if( p_sys->i_bufferLength <= 0 )
178 msg_Err( p_filter, "Delay-time, Sampl rate or Channels was incorrect" );
183 p_sys->pf_delayLineStart = calloc( p_sys->i_bufferLength, sizeof( float ) );
184 if( !p_sys->pf_delayLineStart )
190 p_sys->i_cumulative = 0;
191 p_sys->f_step = p_sys->f_sweepRate / 1000.0;
192 p_sys->i_step = p_sys->f_sweepRate > 0 ? 1 : 0;
197 p_sys->pf_delayLineEnd = p_sys->pf_delayLineStart + p_sys->i_bufferLength;
198 p_sys->pf_write = p_sys->pf_delayLineStart;
200 if( p_sys->f_sweepDepth < small_value() ||
201 p_filter->fmt_in.audio.i_rate < small_value() ) {
202 p_sys->f_sinMultiplier = 0.0;
205 p_sys->f_sinMultiplier = 11 * p_sys->f_sweepRate /
206 ( 7 * p_sys->f_sweepDepth * p_filter->fmt_in.audio.i_rate ) ;
208 p_sys->i_sampleRate = p_filter->fmt_in.audio.i_rate;
215 * sanitize: Helper function to eliminate small amplitudes
216 * @param f_value pointer to value to clean
218 static inline void sanitize( float * f_value )
220 if ( fabs( *f_value ) < small_value() )
226 * DoWork : delays and finds the value of the current frame
227 * @param p_filter This filter object
228 * @param p_in_buf Input buffer
229 * @return Output buffer
231 static block_t *DoWork( filter_t *p_filter, block_t *p_in_buf )
233 struct filter_sys_t *p_sys = p_filter->p_sys;
235 unsigned i_samples = p_in_buf->i_nb_samples; /* number of samples */
236 /* maximum number of samples to offset in buffer */
237 int i_maxOffset = floor( p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000 );
238 float *p_out = (float*)p_in_buf->p_buffer;
239 float *p_in = (float*)p_in_buf->p_buffer;
241 float *pf_ptr, f_diff = 0, f_frac = 0, f_temp = 0 ;
243 /* Process each sample */
244 for( unsigned i = 0; i < i_samples ; i++ )
246 /* Use a sine function as a oscillator wave. TODO */
247 /* f_offset = sinf( ( p_sys->i_cumulative ) * p_sys->f_sinMultiplier ) *
248 * (int)floor(p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000);
251 /* Triangle oscillator. Step using ints, because floats give rounding */
252 p_sys->i_offset+=p_sys->i_step;
253 p_sys->f_offset = p_sys->i_offset * p_sys->f_step;
254 if( abs( p_sys->i_step ) > 0 )
256 if( p_sys->i_offset >= floor( p_sys->f_sweepDepth *
257 p_sys->i_sampleRate / p_sys->f_sweepRate ))
259 p_sys->f_offset = i_maxOffset;
260 p_sys->i_step = -1 * ( p_sys->i_step );
262 if( p_sys->i_offset <= floor( -1 * p_sys->f_sweepDepth *
263 p_sys->i_sampleRate / p_sys->f_sweepRate ) )
265 p_sys->f_offset = -i_maxOffset;
266 p_sys->i_step = -1 * ( p_sys->i_step );
269 /* Calculate position in delay */
270 int offset = floor( p_sys->f_offset );
271 pf_ptr = p_sys->pf_write + i_maxOffset * p_sys->i_channels +
272 offset * p_sys->i_channels;
274 /* Handle Overflow */
275 if( pf_ptr < p_sys->pf_delayLineStart )
277 pf_ptr += p_sys->i_bufferLength - p_sys->i_channels;
279 if( pf_ptr > p_sys->pf_delayLineEnd - 2*p_sys->i_channels )
281 pf_ptr -= p_sys->i_bufferLength - p_sys->i_channels;
283 /* For interpolation */
284 f_frac = ( p_sys->f_offset - (int)p_sys->f_offset );
285 for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ )
287 f_diff = *( pf_ptr + p_sys->i_channels + i_chan )
288 - *( pf_ptr + i_chan );
289 f_temp = ( *( pf_ptr + i_chan ) );//+ f_diff * f_frac);
290 /*Linear Interpolation. FIXME. This creates LOTS of noise */
292 p_out[i_chan] = p_sys->f_dryLevel * p_in[i_chan] +
293 p_sys->f_wetLevel * f_temp;
294 *( p_sys->pf_write + i_chan ) = p_in[i_chan] +
295 p_sys->f_feedbackGain * f_temp;
297 if( p_sys->pf_write == p_sys->pf_delayLineStart )
298 for( i_chan = 0; i_chan < p_sys->i_channels; i_chan++ )
299 *( p_sys->pf_delayLineEnd - p_sys->i_channels + i_chan )
300 = *( p_sys->pf_delayLineStart + i_chan );
302 p_in += p_sys->i_channels;
303 p_out += p_sys->i_channels;
304 p_sys->pf_write += p_sys->i_channels;
305 if( p_sys->pf_write == p_sys->pf_delayLineEnd - p_sys->i_channels )
307 p_sys->pf_write = p_sys->pf_delayLineStart;
316 * @param p_this pointer to this filter object
318 static void Close( vlc_object_t *p_this )
320 filter_t *p_filter = ( filter_t* )p_this;
321 filter_sys_t *p_sys = p_filter->p_sys;
323 free( p_sys->pf_delayLineStart );