]> git.sesse.net Git - vlc/blobdiff - modules/audio_filter/chorus_flanger.c
Use _WIN32 rather than WIN32 (same for WIN64)
[vlc] / modules / audio_filter / chorus_flanger.c
index 9581c381f5bc7b837a2dee8445f0e8d8190f13ea..4d2251181738cfbd05f47d1b596860b3030c25e0 100644 (file)
@@ -1,33 +1,27 @@
 /*****************************************************************************
- * 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
@@ -38,6 +32,7 @@
 #include <vlc_plugin.h>
 
 #include <vlc_aout.h>
+#include <vlc_filter.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 * );
+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 aout_filter_sys_t
+struct filter_sys_t
 {
     /* TODO: Cleanup and optimise */
     int i_cumulative;
@@ -57,15 +54,15 @@ struct aout_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;
 };
 
 /*****************************************************************************
@@ -75,23 +72,24 @@ struct aout_filter_sys_t
 
 vlc_module_begin ()
     set_description( N_("Sound Delay") )
-    set_shortname( N_("delay") )
+    set_shortname( N_("Delay") )
+    set_help( N_("Add a delay effect to the sound") )
     set_category( CAT_AUDIO )
     set_subcategory( SUBCAT_AUDIO_AFILTER )
     add_shortcut( "delay" )
-    add_float( "delay-time", 40, NULL, 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, NULL, N_("Sweep Depth"),
+    add_float( "sweep-depth", 6, 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"),
+    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 )
@@ -113,37 +111,24 @@ static inline float small_value()
  */
 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 ) );
+    filter_t *p_filter = (filter_t*)p_this;
+    filter_sys_t *p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) );
     if( !p_sys )
         return VLC_ENOMEM;
 
-    p_sys->i_channels       = aout_FormatNbChannels( &p_filter->input );
+    p_sys->i_channels       = aout_FormatNbChannels( &p_filter->fmt_in.audio );
     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" );
+    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)
     {
@@ -168,50 +153,51 @@ static int Open( vlc_object_t *p_this )
 
     /* 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 );
+                + p_sys->f_sweepDepth ) * p_filter->fmt_in.audio.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 );
+            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->input.i_rate < 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->input.i_rate ) ;
+            ( 7 * p_sys->f_sweepDepth * p_filter->fmt_in.audio.i_rate ) ;
     }
-    p_sys->i_sampleRate = p_filter->input.i_rate;
+    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
@@ -225,47 +211,38 @@ static inline void sanitize( float * f_value )
 
 /**
  * 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
+ * @return 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 )
+static block_t *DoWork( filter_t *p_filter, block_t *p_in_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;
+    struct filter_sys_t *p_sys = p_filter->p_sys;
+    int i_chan;
+    unsigned i_samples = p_in_buf->i_nb_samples; /* number of samples */
+    /* maximum number of samples to offset in buffer */
+    int i_maxOffset = floor( p_sys->f_sweepDepth * p_sys->i_sampleRate / 1000 );
+    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 ;
-
-    p_out_buf->i_nb_samples = p_in_buf->i_nb_samples;
-    p_out_buf->i_nb_bytes = p_in_buf->i_nb_bytes;
+    float *p_ptr, f_temp = 0;/* f_diff = 0, f_frac = 0;*/
 
     /* Process each sample */
-    for( i = 0; i < i_samples ; i++ )
+    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;
@@ -273,47 +250,50 @@ static void DoWork( aout_instance_t *p_aout, aout_filter_t *p_filter,
             }
         }
         /* 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;
+        int offset = floor( p_sys->f_offset );
+        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;
         }
 
     }
-    return;
+    return p_in_buf;
 }
 
 /**
@@ -322,9 +302,101 @@ static void DoWork( aout_instance_t *p_aout, aout_filter_t *p_filter,
  */
 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 );
+    filter_t *p_filter = ( filter_t* )p_this;
+    filter_sys_t *p_sys = p_filter->p_sys;
+
+    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;
+}