[vlc-devel] [PATCH 2/4] Adds the compressor module to VLC
Ronald Wright
logiconcepts819 at gmail.com
Thu Jul 1 10:33:00 CEST 2010
---
modules/LIST | 1 +
modules/audio_filter/Modules.am | 2 +
modules/audio_filter/compressor.c | 975 +++++++++++++++++++++++++++++++++++++
3 files changed, 978 insertions(+), 0 deletions(-)
create mode 100644 modules/audio_filter/compressor.c
diff --git a/modules/LIST b/modules/LIST
index 820da0a..8f77c57 100644
--- a/modules/LIST
+++ b/modules/LIST
@@ -72,6 +72,7 @@ $Id$
* chorus_flanger: variable delay audio filter
* clone: Clone video filter
* colorthres: Theshold color based on similarity to reference color Video filter
+ * compressor: Dynamic range compressor
* converter_fixed: Fixed-point audio format conversions
* crop: Crop video filter
* croppadd: Crop/Padd image filter
diff --git a/modules/audio_filter/Modules.am b/modules/audio_filter/Modules.am
index eae73fc..7767cbe 100644
--- a/modules/audio_filter/Modules.am
+++ b/modules/audio_filter/Modules.am
@@ -1,5 +1,6 @@
SUBDIRS = channel_mixer converter resampler spatializer
SOURCES_equalizer = equalizer.c equalizer_presets.h
+SOURCES_compressor = compressor.c
SOURCES_normvol = normvol.c
SOURCES_audiobargraph_a = audiobargraph_a.c
SOURCES_param_eq = param_eq.c
@@ -9,6 +10,7 @@ SOURCES_chorus_flanger = chorus_flanger.c
libvlc_LTLIBRARIES += \
libaudiobargraph_a_plugin.la \
libchorus_flanger_plugin.la \
+ libcompressor_plugin.la \
libequalizer_plugin.la \
libnormvol_plugin.la \
libparam_eq_plugin.la \
diff --git a/modules/audio_filter/compressor.c b/modules/audio_filter/compressor.c
new file mode 100644
index 0000000..0c60745
--- /dev/null
+++ b/modules/audio_filter/compressor.c
@@ -0,0 +1,975 @@
+/*****************************************************************************
+ * compressor.c: dynamic range compressor, ported from plugins from LADSPA SWH
+ *****************************************************************************
+ * Copyright (C) 2010 Ronald Wright
+ * $Id$
+ *
+ * Author: Ronald Wright <logiconcepts819 at gmail.com>
+ * Original author: Steve Harris <steve at plugin.org.uk>
+ *
+ * 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.
+ *****************************************************************************/
+
+/*****************************************************************************
+ * Preamble
+ *****************************************************************************/
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <math.h>
+#include <stdint.h>
+
+#include <vlc_common.h>
+#include <vlc_plugin.h>
+
+#include <vlc_aout.h>
+#include <vlc_filter.h>
+
+/*****************************************************************************
+* Local prototypes.
+*****************************************************************************/
+
+#define A_TBL (256)
+
+#define DB_TABLE_SIZE (1024)
+#define DB_MIN (-60.0f)
+#define DB_MAX (24.0f)
+#define LIN_TABLE_SIZE (1024)
+#define LIN_MIN (0.0000000002f)
+#define LIN_MAX (9.0f)
+#define DB_DEFAULT_CUBE
+#define RMS_BUF_SIZE (960)
+#define LOOKAHEAD_SIZE ((RMS_BUF_SIZE)<<1)
+
+#define LN_2_2 (0.34657359f) /* ln(2)/2 */
+#define PEAK_BW (0.3f) /* Peak EQ bandwidth (octaves) */
+#define SHELF_SLOPE (1.5f) /* Shelf EQ slope (arb. units) */
+
+#define LIN_INTERP(f,a,b) ((a) + (f) * ( (b) - (a) ))
+#define LIMIT(v,l,u) (v < l ? l : ( v > u ? u : v ))
+
+typedef struct
+{
+ float pf_buf[RMS_BUF_SIZE];
+ unsigned int i_pos;
+ unsigned int i_count;
+ float f_sum;
+
+} rms_env;
+
+typedef struct
+{
+ struct
+ {
+ float pf_vals[AOUT_CHAN_MAX];
+ float f_lev_in;
+
+ } p_buf[LOOKAHEAD_SIZE];
+ unsigned int i_pos;
+ unsigned int i_count;
+
+} lookahead;
+
+typedef struct
+{
+ float f_a1;
+ float f_a2;
+ float f_b0;
+ float f_b1;
+ float f_b2;
+ float f_x1;
+ float f_x2;
+ float f_y1;
+ float f_y2;
+
+} biquad;
+
+struct filter_sys_t
+{
+ float f_amp;
+ float pf_as[A_TBL];
+ unsigned int i_count;
+ float f_env;
+ float f_env_peak;
+ float f_env_rms;
+ float f_gain;
+ float f_gain_out;
+ rms_env rms;
+ float f_sum;
+ lookahead la;
+
+ float pf_db_data[DB_TABLE_SIZE];
+ float pf_lin_data[LIN_TABLE_SIZE];
+
+ biquad p_filters[3 * AOUT_CHAN_MAX];
+
+ vlc_mutex_t lock;
+
+ float f_lo_pregain;
+ float f_mid_pregain;
+ float f_hi_pregain;
+ float f_rms_peak;
+ float f_attack;
+ float f_release;
+ float f_threshold;
+ float f_ratio;
+ float f_knee;
+ float f_makeup_gain;
+};
+
+typedef union
+{
+ float f;
+ int32_t i;
+
+} ls_pcast32;
+
+static int Open ( vlc_object_t * );
+static void Close ( vlc_object_t * );
+static block_t *DoWork ( filter_t *, block_t * );
+
+static void DbInit ( filter_sys_t * );
+static float Db2Lin ( float, filter_sys_t * );
+static float Lin2Db ( float, filter_sys_t * );
+#ifdef DB_DEFAULT_CUBE
+static float CubeInterp ( const float, const float, const float,
+ const float, const float );
+#endif
+static float FlushToZero ( float );
+static void RoundToZero ( float * );
+static float Max ( float, float );
+static float Clamp ( float, float, float );
+static int Round ( float );
+static float RmsEnvProcess ( rms_env *, const float );
+static void BufferProcess ( float *, int, float, float, lookahead * );
+static void EqSetParams ( biquad *, float, float, float, float );
+static void HsSetParams ( biquad *, float, float, float, float );
+static float BiquadRun ( biquad *, const float );
+
+static int LoPregainCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int MidPregainCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int HiPregainCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int RMSPeakCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int AttackCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int ReleaseCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int ThresholdCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int RatioCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int KneeCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+static int MakeupGainCallback ( vlc_object_t *, char const *, vlc_value_t,
+ vlc_value_t, void * );
+
+/*****************************************************************************
+ * Module descriptor
+ *****************************************************************************/
+
+#define LO_TEXT N_( "Lo pre-gain" )
+#define LO_LONGTEXT N_( "Set the low pre-gain in dB (-70 ... 6)." )
+
+#define MID_TEXT N_( "Mid pre-gain" )
+#define MID_LONGTEXT N_( "Set the mid pre-gain in dB (-70 ... 6)." )
+
+#define HI_TEXT N_( "Hi pre-gain" )
+#define HI_LONGTEXT N_( "Set the high pre-gain in dB (-70 ... 6)." )
+
+#define RMS_PEAK_TEXT N_( "RMS/peak" )
+#define RMS_PEAK_LONGTEXT N_( "Set the RMS/peak (0 ... 1)." )
+
+#define ATTACK_TEXT N_( "Attack time" )
+#define ATTACK_LONGTEXT N_( \
+ "Set the attack time in milliseconds (1.5 ... 400)." )
+
+#define RELEASE_TEXT N_( "Release time" )
+#define RELEASE_LONGTEXT N_( \
+ "Set the release time in milliseconds (2 ... 800)." )
+
+#define THRESHOLD_TEXT N_( "Threshold level" )
+#define THRESHOLD_LONGTEXT N_( "Set the threshold level in dB (-30 ... 0)." )
+
+#define RATIO_TEXT N_( "Ratio" )
+#define RATIO_LONGTEXT N_( "Set the ratio (n:1) (1 ... 20)." )
+
+#define KNEE_TEXT N_( "Knee radius" )
+#define KNEE_LONGTEXT N_( "Set the knee radius in dB (1 ... 10)." )
+
+#define MAKEUP_GAIN_TEXT N_( "Makeup gain" )
+#define MAKEUP_GAIN_LONGTEXT N_( "Set the makeup gain in dB (0 ... 24)." )
+
+vlc_module_begin()
+ set_shortname( _("Compressor") )
+ set_description( _("Dynamic range compressor") )
+ set_capability( "audio filter", 1 )
+ set_category( CAT_AUDIO )
+ set_subcategory( SUBCAT_AUDIO_AFILTER )
+
+ add_float( "compressor-lo-pregain", 0.0, NULL, LO_TEXT,
+ LO_LONGTEXT, false )
+ add_float( "compressor-mid-pregain", 0.0, NULL, MID_TEXT,
+ MID_LONGTEXT, false )
+ add_float( "compressor-hi-pregain", 0.0, NULL, HI_TEXT,
+ HI_LONGTEXT, false )
+ add_float( "compressor-rms-peak", 0.0, NULL, RMS_PEAK_TEXT,
+ RMS_PEAK_LONGTEXT, false )
+ add_float( "compressor-attack", 25.0, NULL, ATTACK_TEXT,
+ ATTACK_LONGTEXT, false )
+ add_float( "compressor-release", 100.0, NULL, RELEASE_TEXT,
+ RELEASE_LONGTEXT, false )
+ add_float( "compressor-threshold", -11.0, NULL, THRESHOLD_TEXT,
+ THRESHOLD_LONGTEXT, false )
+ add_float( "compressor-ratio", 8.0, NULL, RATIO_TEXT,
+ RATIO_LONGTEXT, false )
+ add_float( "compressor-knee", 2.5, NULL, KNEE_TEXT,
+ KNEE_LONGTEXT, false )
+ add_float( "compressor-makeup-gain", 7.0, NULL, MAKEUP_GAIN_TEXT,
+ MAKEUP_GAIN_LONGTEXT, false )
+ set_callbacks( Open, Close )
+ add_shortcut( "compressor" )
+vlc_module_end ()
+
+/*****************************************************************************
+ * Open: initialize interface
+ *****************************************************************************/
+
+static int Open( vlc_object_t *p_this )
+{
+ filter_t *p_filter = (filter_t*)p_this;
+ vlc_object_t *p_aout = p_filter->p_parent;
+ float f_sample_rate = p_filter->fmt_in.audio.i_rate;
+ struct filter_sys_t *p_sys;
+ float f_num;
+
+ 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" );
+ return VLC_EGENERIC;
+ }
+ if( !AOUT_FMTS_SIMILAR( &p_filter->fmt_in.audio,
+ &p_filter->fmt_out.audio ) )
+ {
+ p_filter->fmt_out.audio = p_filter->fmt_in.audio;
+ msg_Warn( p_filter, "input and output formats are not similar" );
+ return VLC_EGENERIC;
+ }
+
+ /* Initialize the filter parameter structure */
+ p_sys = p_filter->p_sys = calloc( 1, sizeof(*p_sys) );
+ if( !p_sys )
+ {
+ return VLC_ENOMEM;
+ }
+
+ /* Initialize the attack lookup table */
+ p_sys->pf_as[0] = 1.0f;
+ for( int i = 1; i < A_TBL; i++ )
+ {
+ p_sys->pf_as[i] = expf( -1.0f / ( f_sample_rate * i / A_TBL ) );
+ }
+
+ /* Calculate the RMS and lookahead sizes from the sample rate */
+ f_num = 0.01f * f_sample_rate;
+ p_sys->rms.i_count = Round( Clamp( 0.5f * f_num, 1.0f, RMS_BUF_SIZE ) );
+ p_sys->la.i_count = Round( Clamp( f_num, 1.0f, LOOKAHEAD_SIZE ) );
+
+ /* Initialize decibel lookup tables */
+ DbInit( p_sys );
+
+ /* Restore the last saved settings */
+ p_sys->f_lo_pregain =
+ var_CreateGetFloat( p_aout, "compressor-lo-pregain" );
+ p_sys->f_mid_pregain =
+ var_CreateGetFloat( p_aout, "compressor-mid-pregain" );
+ p_sys->f_hi_pregain =
+ var_CreateGetFloat( p_aout, "compressor-hi-pregain" );
+ p_sys->f_rms_peak = var_CreateGetFloat( p_aout, "compressor-rms-peak" );
+ p_sys->f_attack = var_CreateGetFloat( p_aout, "compressor-attack" );
+ p_sys->f_release = var_CreateGetFloat( p_aout, "compressor-release" );
+ p_sys->f_threshold = var_CreateGetFloat( p_aout, "compressor-threshold" );
+ p_sys->f_ratio = var_CreateGetFloat( p_aout, "compressor-ratio" );
+ p_sys->f_knee = var_CreateGetFloat( p_aout, "compressor-knee" );
+ p_sys->f_makeup_gain =
+ var_CreateGetFloat( p_aout, "compressor-makeup-gain" );
+
+ /* Initialize the mutex */
+ vlc_mutex_init( &p_sys->lock );
+
+ /* Add our own callbacks */
+ var_AddCallback( p_aout, "compressor-lo-pregain", LoPregainCallback,
+ p_sys );
+ var_AddCallback( p_aout, "compressor-mid-pregain", MidPregainCallback,
+ p_sys );
+ var_AddCallback( p_aout, "compressor-hi-pregain", HiPregainCallback,
+ p_sys );
+ var_AddCallback( p_aout, "compressor-rms-peak", RMSPeakCallback, p_sys );
+ var_AddCallback( p_aout, "compressor-attack", AttackCallback, p_sys );
+ var_AddCallback( p_aout, "compressor-release", ReleaseCallback, p_sys );
+ var_AddCallback( p_aout, "compressor-threshold", ThresholdCallback, p_sys );
+ var_AddCallback( p_aout, "compressor-ratio", RatioCallback, p_sys );
+ var_AddCallback( p_aout, "compressor-knee", KneeCallback, p_sys );
+ var_AddCallback( p_aout, "compressor-makeup-gain", MakeupGainCallback,
+ p_sys );
+
+ /* Set the filter function */
+ p_filter->pf_audio_filter = DoWork;
+
+ /* At this stage, we are ready! */
+ msg_Dbg( p_filter, "compressor successfully initialized" );
+ return VLC_SUCCESS;
+}
+
+/*****************************************************************************
+ * Close: destroy interface
+ *****************************************************************************/
+
+static void Close( vlc_object_t *p_this )
+{
+ filter_t *p_filter = (filter_t*)p_this;
+ vlc_object_t *p_aout = p_filter->p_parent;
+ struct filter_sys_t *p_sys = p_filter->p_sys;
+
+ /* Remove our callbacks */
+ var_DelCallback( p_aout, "compressor-rms-peak", RMSPeakCallback, p_sys );
+ var_DelCallback( p_aout, "compressor-attack", AttackCallback, p_sys );
+ var_DelCallback( p_aout, "compressor-release", ReleaseCallback, p_sys );
+ var_DelCallback( p_aout, "compressor-threshold", ThresholdCallback, p_sys );
+ var_DelCallback( p_aout, "compressor-ratio", RatioCallback, p_sys );
+ var_DelCallback( p_aout, "compressor-knee", KneeCallback, p_sys );
+ var_DelCallback( p_aout, "compressor-makeup-gain", MakeupGainCallback,
+ p_sys );
+
+ /* Destroy the mutex */
+ vlc_mutex_destroy( &p_sys->lock );
+
+ /* Destroy the filter parameter structure */
+ free( p_sys );
+}
+
+/*****************************************************************************
+ * DoWork: process samples buffer
+ *****************************************************************************/
+
+static block_t * DoWork( filter_t * p_filter, block_t * p_in_buf )
+{
+ int i_samples = p_in_buf->i_nb_samples;
+ int i_channels = aout_FormatNbChannels( &p_filter->fmt_in.audio );
+ float *pf_buf = (float*)p_in_buf->p_buffer;
+ float f_sample_rate = p_filter->fmt_in.audio.i_rate;
+
+ float f_lo_pregain;
+ float f_mid_pregain;
+ float f_hi_pregain;
+ float f_rms_peak;
+ float f_attack;
+ float f_release;
+ float f_threshold;
+ float f_ratio;
+ float f_knee;
+ float f_makeup_gain;
+
+ float f_amp;
+ float *pf_as;
+ float f_env;
+ float f_env_peak;
+ float f_env_rms;
+ float f_gain;
+ float f_gain_out;
+ rms_env *p_rms;
+ float f_sum;
+ lookahead *p_la;
+ biquad *p_filters;
+
+ float f_ga;
+ float f_gr;
+ float f_rs;
+ float f_mug;
+ float f_knee_min;
+ float f_knee_max;
+ float f_ef_a;
+ float f_ef_ai;
+
+ /* Current parameters */
+ struct filter_sys_t *p_sys = p_filter->p_sys;
+
+ /* Fetch the configurable parameters */
+ vlc_mutex_lock( &p_sys->lock );
+
+ f_lo_pregain = p_sys->f_lo_pregain; /* Lo pre-gain (dB) */
+ f_mid_pregain = p_sys->f_mid_pregain; /* Mid pre-gain (dB) */
+ f_hi_pregain = p_sys->f_hi_pregain; /* High pre-gain (dB) */
+ f_rms_peak = p_sys->f_rms_peak; /* RMS/peak */
+ f_attack = p_sys->f_attack; /* Attack time (ms) */
+ f_release = p_sys->f_release; /* Release time (ms) */
+ f_threshold = p_sys->f_threshold; /* Threshold level (dB) */
+ f_ratio = p_sys->f_ratio; /* Ratio (n:1) */
+ f_knee = p_sys->f_knee; /* Knee radius (dB) */
+ f_makeup_gain = p_sys->f_makeup_gain; /* Makeup gain (dB) */
+
+ vlc_mutex_unlock( &p_sys->lock );
+
+ /* Fetch the internal parameters */
+ f_amp = p_sys->f_amp;
+ pf_as = p_sys->pf_as;
+ f_env = p_sys->f_env;
+ f_env_peak = p_sys->f_env_peak;
+ f_env_rms = p_sys->f_env_rms;
+ f_gain = p_sys->f_gain;
+ f_gain_out = p_sys->f_gain_out;
+ p_rms = &p_sys->rms;
+ f_sum = p_sys->f_sum;
+ p_la = &p_sys->la;
+ p_filters = p_sys->p_filters;
+
+ /* Prepare the equalizer filter info */
+ for( int i_chan = 0; i_chan < i_channels; i_chan++ )
+ {
+ EqSetParams( &p_filters[3 * i_chan + 0], 100.0f, f_lo_pregain,
+ PEAK_BW, f_sample_rate );
+ EqSetParams( &p_filters[3 * i_chan + 1], 1000.0f, f_mid_pregain,
+ PEAK_BW, f_sample_rate );
+ HsSetParams( &p_filters[3 * i_chan + 2], 10000.0f, f_hi_pregain,
+ SHELF_SLOPE, f_sample_rate );
+ }
+
+ /* Prepare other compressor parameters */
+ f_ga = f_attack < 2.0f ? 0.0f :
+ pf_as[Round( f_attack * 0.001f * ( A_TBL - 1 ) )];
+ f_gr = pf_as[Round( f_release * 0.001f * ( A_TBL - 1 ) )];
+ f_rs = ( f_ratio - 1.0f ) / f_ratio;
+ f_mug = Db2Lin( f_makeup_gain, p_sys );
+ f_knee_min = Db2Lin( f_threshold - f_knee, p_sys );
+ f_knee_max = Db2Lin( f_threshold + f_knee, p_sys );
+ f_ef_a = f_ga * 0.25f;
+ f_ef_ai = 1.0f - f_ef_a;
+
+ /* Process the current buffer */
+ for( int i = 0; i < i_samples; i++ )
+ {
+ float f_lev_in_old, f_lev_in_new;
+
+ /* First, pass the audio through the equalizer (ported from dj_eq_1901
+ * plugin) */
+ for( int i_chan = 0; i_chan < i_channels; i_chan++ )
+ {
+ pf_buf[i_chan] = BiquadRun( &p_filters[3 * i_chan + 0],
+ pf_buf[i_chan] );
+ pf_buf[i_chan] = BiquadRun( &p_filters[3 * i_chan + 1],
+ pf_buf[i_chan] );
+ pf_buf[i_chan] = BiquadRun( &p_filters[3 * i_chan + 2],
+ pf_buf[i_chan] );
+ }
+
+ /* Now, compress the pre-equalized audio (ported from sc4_1882
+ * plugin with a few modifications) */
+
+ /* Fetch the old delayed buffer value */
+ f_lev_in_old = p_la->p_buf[p_la->i_pos].f_lev_in;
+
+ /* Find the peak value of current sample. This becomes the new delayed
+ * buffer value that replaces the old one in the lookahead array */
+ f_lev_in_new = fabs( pf_buf[0] );
+ for( int i_chan = 1; i_chan < i_channels; i_chan++ )
+ {
+ f_lev_in_new = Max( f_lev_in_new, fabs( pf_buf[i_chan] ) );
+ }
+ p_la->p_buf[p_la->i_pos].f_lev_in = f_lev_in_new;
+
+ /* Add the square of the peak value to a running sum */
+ f_sum += f_lev_in_new * f_lev_in_new;
+
+ /* Update the RMS envelope */
+ if( f_amp > f_env_rms )
+ {
+ f_env_rms = f_env_rms * f_ga + f_amp * ( 1.0f - f_ga );
+ }
+ else
+ {
+ f_env_rms = f_env_rms * f_gr + f_amp * ( 1.0f - f_gr );
+ }
+ RoundToZero( &f_env_rms );
+
+ /* Update the peak envelope */
+ if( f_lev_in_old > f_env_peak )
+ {
+ f_env_peak = f_env_peak * f_ga + f_lev_in_old * ( 1.0f - f_ga );
+ }
+ else
+ {
+ f_env_peak = f_env_peak * f_gr + f_lev_in_old * ( 1.0f - f_gr );
+ }
+ RoundToZero( &f_env_peak );
+
+ /* Process the RMS value and update the output gain every 4 samples */
+ if( ( p_sys->i_count++ & 3 ) == 3 )
+ {
+ /* Process the RMS value by placing in the mean square value, and
+ * reset the running sum */
+ f_amp = RmsEnvProcess( p_rms, f_sum * 0.25f );
+ f_sum = 0.0f;
+ if( isnan( f_env_rms ) )
+ {
+ /* This can happen sometimes, but I don't know why. */
+ f_env_rms = 0.0f;
+ }
+
+ /* Find the superposition of the RMS and peak envelopes */
+ f_env = LIN_INTERP( f_rms_peak, f_env_rms, f_env_peak );
+
+ /* Update the output gain */
+ if( f_env <= f_knee_min )
+ {
+ /* Gain below the knee (and below the threshold) */
+ f_gain_out = 1.0f;
+ }
+ else if( f_env < f_knee_max )
+ {
+ /* Gain within the knee */
+ const float f_x = -( f_threshold
+ - f_knee - Lin2Db( f_env, p_sys ) ) / f_knee;
+ f_gain_out = Db2Lin( -f_knee * f_rs * f_x * f_x * 0.25f,
+ p_sys );
+ }
+ else
+ {
+ /* Gain above the knee (and above the threshold) */
+ f_gain_out = Db2Lin( ( f_threshold - Lin2Db( f_env, p_sys ) )
+ * f_rs, p_sys );
+ }
+ }
+
+ /* Find the total gain */
+ f_gain = f_gain * f_ef_a + f_gain_out * f_ef_ai;
+
+ /* Write the resulting buffer to the output */
+ BufferProcess( pf_buf, i_channels, f_gain, f_mug, p_la );
+ pf_buf += i_channels;
+ }
+
+ /* Update the internal parameters */
+ p_sys->f_sum = f_sum;
+ p_sys->f_amp = f_amp;
+ p_sys->f_gain = f_gain;
+ p_sys->f_gain_out = f_gain_out;
+ p_sys->f_env = f_env;
+ p_sys->f_env_rms = f_env_rms;
+ p_sys->f_env_peak = f_env_peak;
+
+ return p_in_buf;
+}
+
+/*****************************************************************************
+ * Helper functions for compressor
+ *****************************************************************************/
+
+static void DbInit( filter_sys_t * p_sys )
+{
+ float *pf_lin_data = p_sys->pf_lin_data;
+ float *pf_db_data = p_sys->pf_db_data;
+
+ /* Fill linear lookup table */
+ for( int i = 0; i < LIN_TABLE_SIZE; i++ )
+ {
+ pf_lin_data[i] = powf( 10.0f, ( ( DB_MAX - DB_MIN ) *
+ (float)i / LIN_TABLE_SIZE + DB_MIN ) / 20.0f );
+ }
+
+ /* Fill logarithmic lookup table */
+ for( int i = 0; i < DB_TABLE_SIZE; i++ )
+ {
+ pf_db_data[i] = 20.0f * log10f( ( LIN_MAX - LIN_MIN ) *
+ (float)i / DB_TABLE_SIZE + LIN_MIN );
+ }
+}
+
+static float Db2Lin( float f_db, filter_sys_t * p_sys )
+{
+ float f_scale = ( f_db - DB_MIN ) * LIN_TABLE_SIZE / ( DB_MAX - DB_MIN );
+ int i_base = Round( f_scale - 0.5f );
+ float f_ofs = f_scale - i_base;
+ float *pf_lin_data = p_sys->pf_lin_data;
+
+ if( i_base < 1 )
+ {
+ return 0.0f;
+ }
+ else if( i_base > LIN_TABLE_SIZE - 3 )
+ {
+ return pf_lin_data[LIN_TABLE_SIZE - 2];
+ }
+
+#ifdef DB_DEFAULT_CUBE
+ return CubeInterp( f_ofs, pf_lin_data[i_base - 1],
+ pf_lin_data[i_base],
+ pf_lin_data[i_base + 1],
+ pf_lin_data[i_base + 2] );
+#else
+ return ( 1.0f - f_ofs ) * pf_lin_data[i_base]
+ + f_ofs * pf_lin_data[i_base + 1];
+#endif
+}
+
+static float Lin2Db( float f_lin, filter_sys_t * p_sys )
+{
+ float f_scale = ( f_lin - LIN_MIN ) * DB_TABLE_SIZE / ( LIN_MAX - LIN_MIN );
+ int i_base = Round( f_scale - 0.5f );
+ float f_ofs = f_scale - i_base;
+ float *pf_db_data = p_sys->pf_db_data;
+
+ if( i_base < 2 )
+ {
+ return pf_db_data[2] * f_scale * 0.5f - 23.0f * ( 2.0f - f_scale );
+ }
+ else if( i_base > DB_TABLE_SIZE - 3 )
+ {
+ return pf_db_data[DB_TABLE_SIZE - 2];
+ }
+
+#ifdef DB_DEFAULT_CUBE
+ return CubeInterp( f_ofs, pf_db_data[i_base - 1],
+ pf_db_data[i_base],
+ pf_db_data[i_base + 1],
+ pf_db_data[i_base + 2] );
+#else
+ return ( 1.0f - f_ofs ) * pf_db_data[i_base]
+ + f_ofs * pf_db_data[i_base + 1];
+#endif
+}
+
+#ifdef DB_DEFAULT_CUBE
+/* Cubic interpolation function */
+static float CubeInterp( const float f_fr, const float f_inm1,
+ const float f_in,
+ const float f_inp1,
+ const float f_inp2 )
+{
+ return f_in + 0.5f * f_fr * ( f_inp1 - f_inm1 +
+ f_fr * ( 4.0f * f_inp1 + 2.0f * f_inm1 - 5.0f * f_in - f_inp2 +
+ f_fr * ( 3.0f * ( f_in - f_inp1 ) - f_inm1 + f_inp2 ) ) );
+}
+#endif
+
+static float FlushToZero( float f_x )
+{
+ ls_pcast32 v;
+
+ v.f = f_x;
+
+ /* original: return ( v.i & 0x7f800000 ) == 0 ? 0.0f : f_x;
+ * version from Tim Blechmann */
+ return ( v.i & 0x7f800000 ) < 0x08000000 ? 0.0f : f_x;
+}
+
+/* Zero out denormals by adding and subtracting a small number, from Laurent
+ * de Soras */
+static void RoundToZero( float *pf_x )
+{
+ static const float f_anti_denormal = 1e-18;
+
+ *pf_x += f_anti_denormal;
+ *pf_x -= f_anti_denormal;
+}
+
+/* A set of branchless clipping operations from Laurent de Soras */
+
+static float Max( float f_x, float f_a )
+{
+ f_x -= f_a;
+ f_x += fabs( f_x );
+ f_x *= 0.5;
+ f_x += f_a;
+
+ return f_x;
+}
+
+static float Clamp( float f_x, float f_a, float f_b )
+{
+ const float f_x1 = fabs( f_x - f_a );
+ const float f_x2 = fabs( f_x - f_b );
+
+ f_x = f_x1 + f_a + f_b;
+ f_x -= f_x2;
+ f_x *= 0.5;
+
+ return f_x;
+}
+
+/* Round float to int using IEEE int* hack */
+static int Round( float f_x )
+{
+ ls_pcast32 p;
+
+ p.f = f_x;
+ p.f += ( 3 << 22 );
+
+ return p.i - 0x4b400000;
+}
+
+/* Calculate current level from root-mean-squared of circular buffer ("RMS") */
+static float RmsEnvProcess( rms_env * p_r, const float f_x )
+{
+ /* Remove the old term from the sum */
+ p_r->f_sum -= p_r->pf_buf[p_r->i_pos];
+
+ /* Add the new term to the sum */
+ p_r->f_sum += f_x;
+
+ /* If the sum is small enough, make it zero */
+ if( p_r->f_sum < 1.0e-6 )
+ {
+ p_r->f_sum = 0.0f;
+ }
+
+ /* Replace the old term in the array with the new one */
+ p_r->pf_buf[p_r->i_pos] = f_x;
+
+ /* Go to the next position for the next RMS calculation */
+ p_r->i_pos = ( p_r->i_pos + 1 ) % ( p_r->i_count );
+
+ /* Return the RMS value */
+ return sqrt( p_r->f_sum / p_r->i_count );
+}
+
+/* Output the compressed delayed buffer and store the current buffer. Uses a
+ * circular array, just like the one used in calculating the RMS of the buffer
+ */
+static void BufferProcess( float * pf_buf, int i_channels, float f_gain,
+ float f_mug, lookahead * p_la )
+{
+ /* Loop through every channel */
+ for( int i_chan = 0; i_chan < i_channels; i_chan++ )
+ {
+ float f_x = pf_buf[i_chan]; /* Current buffer value */
+
+ /* Output the compressed delayed buffer value */
+ pf_buf[i_chan] = p_la->p_buf[p_la->i_pos].pf_vals[i_chan]
+ * f_gain * f_mug;
+
+ /* Update the delayed buffer value */
+ p_la->p_buf[p_la->i_pos].pf_vals[i_chan] = f_x;
+ }
+
+ /* Go to the next delayed buffer value for the next run */
+ p_la->i_pos = ( p_la->i_pos + 1 ) % ( p_la->i_count );
+}
+
+static void EqSetParams( biquad *p_f, float f_fc, float f_gain, float f_bw,
+ float f_sr )
+{
+ float f_w = 2.0f * M_PI * LIMIT( f_fc, 1.0f, f_sr / 2.0f ) / f_sr;
+ float f_cw = cosf( f_w );
+ float f_sw = sinf( f_w );
+ float f_J = pow( 10.0f, f_gain * 0.025f );
+ float f_g = f_sw * sinhf( LN_2_2 * LIMIT( f_bw, 0.0001f, 4.0f )
+ * f_w / f_sw );
+ float f_a0r = 1.0f / ( 1.0f + ( f_g / f_J ) );
+
+ p_f->f_b0 = ( 1.0f + ( f_g * f_J ) ) * f_a0r;
+ p_f->f_b1 = ( -2.0f * f_cw ) * f_a0r;
+ p_f->f_b2 = ( 1.0f - ( f_g * f_J ) ) * f_a0r;
+ p_f->f_a1 = -( p_f->f_b1 );
+ p_f->f_a2 = ( ( f_g / f_J ) - 1.0f ) * f_a0r;
+}
+
+static void HsSetParams( biquad *p_f, float f_fc, float f_gain, float f_slope,
+ float f_sr )
+{
+ float f_w = 2.0f * M_PI * LIMIT( f_fc, 1.0f, f_sr / 2.0f ) / f_sr;
+ float f_cw = cosf( f_w );
+ float f_sw = sinf( f_w );
+ float f_A = powf( 10.0f, f_gain * 0.025f );
+ float f_b = sqrt( ( ( 1.0f + f_A * f_A ) / LIMIT( f_slope, 0.0001f, 1.0f ) )
+ - ( ( f_A - 1.0f ) * ( f_A - 1.0f ) ) );
+
+ float f_apc = f_cw * ( f_A + 1.0f );
+ float f_amc = f_cw * ( f_A - 1.0f );
+ float f_bs = f_b * f_sw;
+ float f_a0r = 1.0f / ( f_A + 1.0f - f_amc + f_bs );
+
+ p_f->f_b0 = f_a0r * f_A * ( f_A + 1.0f + f_amc + f_bs );
+ p_f->f_b1 = f_a0r * -2.0f * f_A * ( f_A - 1.0f + f_apc );
+ p_f->f_b2 = f_a0r * f_A * ( f_A + 1.0f + f_amc - f_bs );
+ p_f->f_a1 = f_a0r * -2.0f * ( f_A - 1.0f - f_apc );
+ p_f->f_a2 = f_a0r * ( -f_A - 1.0f + f_amc + f_bs );
+}
+
+static float BiquadRun( biquad *p_f, const float f_x )
+{
+ float f_y;
+
+ f_y = p_f->f_b0 * f_x + p_f->f_b1 * p_f->f_x1 + p_f->f_b2 * p_f->f_x2
+ + p_f->f_a1 * p_f->f_y1 + p_f->f_a2 * p_f->f_y2;
+ f_y = FlushToZero( f_y );
+ p_f->f_x2 = p_f->f_x1;
+ p_f->f_x1 = f_x;
+ p_f->f_y2 = p_f->f_y1;
+ p_f->f_y1 = f_y;
+
+ return f_y;
+}
+
+/*****************************************************************************
+ * Callback functions
+ *****************************************************************************/
+
+static int LoPregainCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_lo_pregain = Clamp( newval.f_float, -70.0f, 6.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int MidPregainCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_mid_pregain = Clamp( newval.f_float, -70.0f, 6.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int HiPregainCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_hi_pregain = Clamp( newval.f_float, -70.0f, 6.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int RMSPeakCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_rms_peak = Clamp( newval.f_float, 0.0f, 1.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int AttackCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_attack = Clamp( newval.f_float, 1.5f, 400.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int ReleaseCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_release = Clamp( newval.f_float, 2.0f, 800.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int ThresholdCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_threshold = Clamp( newval.f_float, -30.0f, 0.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int RatioCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_ratio = Clamp( newval.f_float, 1.0f, 20.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int KneeCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_knee = Clamp( newval.f_float, 1.0f, 10.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
+
+static int MakeupGainCallback( vlc_object_t *p_this, char const *psz_cmd,
+ vlc_value_t oldval, vlc_value_t newval,
+ void * p_data )
+{
+ VLC_UNUSED(p_this); VLC_UNUSED(psz_cmd); VLC_UNUSED(oldval);
+ filter_sys_t *p_sys = p_data;
+
+ vlc_mutex_lock( &p_sys->lock );
+ p_sys->f_makeup_gain = Clamp( newval.f_float, 0.0f, 24.0f );
+ vlc_mutex_unlock( &p_sys->lock );
+
+ return VLC_SUCCESS;
+}
--
1.7.0.4
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