Audio Normalize seems not to work...

[kbeartx]

...or mebbee I do not understand how to make it work.

I frequently watch commercially produced movies on my Mac and find that I must constantly 'ride' the volume control; if the volume is set to an appropriate volume for the the music and special effects, then the dialog is so soft that it's virtually inaudible; conversely, if the dialog is clearly audible, then the music and special audio effects are teeth-rattle-ing-ly loud.

In my understanding of audio, I expect 'normalization' to even-out the loud and soft parts of the movie's audio track, attenuating the loud and amplifying the soft so that a single volume setting will provide an acceptable compromise between too-soft dialog and too-loud music and sound effects. In pre-digital days, this functionality was provided by a compressor-limiter.

Am I 'asking' for a function that VLC does not provide, or is it that the normalize function requires esoteric knowledge of programming and such?

If not VLC, is there some other freeware/open source software that does what I want/need?

TIA,

KBeartx

[ph0t0n]

Normalization in audio processing is about adjusting overall audio signal amplitude over a timeframe, not evening out the differences in volume among the individual signal components, which is what dynamic compression is doing.

See http://en.wikipedia.org/wiki/Audio_normalization

To achieve the effect you are looking for, the audio part of the commercials would have to be remixed.
You just can't fix everything 'in post (-production, that is)'...

VLC has some normalizing options, even if they won't help in your case:

Volume normalizer
--norm-buff-size <integer> Number of audio buffers
This is the number of audio buffers on which
the power measurement is made. A higher
number of buffers will increase the response
time of the filter to a spike but will make
it less sensitive to short variations.
--norm-max-level <float> Max level
If the average power over the last N buffers
is higher than this value, the volume will be
normalized. This value is a positive floating
point number. A value between 0.5 and 10
seems sensible.

[kbeartx]

Thanks for your response.

Are there any 3rd-party solutions that can extend VLC's audio processing capabilities to include comp-limiting, or is there some kind of Core Audio 'plug-in' that would provide dynamic compression to all system audio?

In other words, if I can't get this done 'at the source' [that is, within the media player, in this case, VLC] I envision some software that would allow me to apply compression/limiting at the system level, that I could enable/disable as needed/wanted.

I'm thinking that this functionality could even out the dynamics of a movie's audio track and solve this recurring problem/issue.

- KBeartx

[|)orian]

Nice! So how does the buffer correspond with milliseconds? Is it one buffer per millisecond? Or is there another variable that defines their relationship?

We should add a knob for --norm-buff-size in the Normalize section.

And a "distance" control that affects bass and treble (but not mid) EQs in a /\/\ pattern, as well as the compression ratio of the normalizer. Or whichever variable is most analogous to ratio.

'Cause lots of bass and treble, as well as heavy ratio, makes sounds seem closer and louder than they are. Distance can also be simulated while maintaining volume when more mid-heavy EQ and audio expansion are combined.

[life036]

So does anyone have any new updates on possible solutions for this?

I experience the same thing. Volume on almost every movie I watch is completely out of control and ridiculous. Music and sound effects are WAY TOO LOUD and speech is unbelievably soft.

I find it extremely hard to believe that I'm in a minority here, being only one of a handful of people who can't stand this...

[gregcss]

I have the same issue. I have been experimenting with the Volume/Audio Normalization in VLC 0.9.9 and it seems it does work but not completely the way i want. There is a delay of the 'normalize' of loud volume.

I went to the advanced options into audio filters and turned on Normalization, then went to the Normalization tab and set the audio buffers to 50 and the filter amount to 1.60 after playing with it awhile. It seems less audio buffers you have the faster it will reduce the loud sounds (ex. 10 buffers takes 1sec and 400 buffers takes ~4seconds to reduce loud sound). Also with the buffers the lower the number the faster the filter 'resets'. In other words if the audio track has dialogue, loud noise, dialogue, then loud noise the normalization will not help unless the loud noise is minimum a few seconds. I watched an anime video and it sounded like someone was taking the volume knob turning it up and down non stop...wierd.


Havent played with filter amount too much yet.


What I am really looking for is something to make the quiet dialogue louder keeping the loud noise the same.

[life036]

Yes, this is crazy. Why the hell does Hollywood do this? Who is sitting in their office thinking that such a crazy disparity in sound volume is a good thing?

[torerik]

I guess it's a long shot for most of you, but.. If your source material is multichannel, and you're using a stereo speaker setup, make sure you have VLC set to output in stereo (Audio -> Audio device -> Stereo). I've seen users who have this set to 5.1 on a stereo setup, which means they're losing all the information in the rear and center (voice) channels.

[bencoder]

This is something that really bothers me as well, I often watch films late at night and don't want to disturb others, but if i turn it down so that the loud sounds are quiet enough, I can't hear anything that's being spoken.

Anyway, I have solved this at least on linux by using Jack. I use the Jack plugin for VLC to output to jack and with qjackcontrol, have the output go to a program called calf, which hosts a dynamic range compression plugin, and then this is routed out to the speakers. I get full control over the compression settings this way.

Edit: apologies, just noticed this was in the Mac forum, I got to this thread through a google search earlier so I didn't notice. Not sure if anything similar to jack exists on macs, but considering they are quite highly regarded for media work, I'd be surprised if there wasn't some way to route audio.

[fixmacs]

Limiter

I need the same function for VLC. A limiter would be most welcome. Set the volume appropriate for the softer dialogue, and then "tell" VLC to not allow the soundtrack to go much higher than that. This is called a limiter (clipping?). Limiters can be found on MP3 players. Surely, this function could be programmed into VLC.

[Jakk Frost]

@Torerik ~ It's not really a long shot, you're pretty much on the money. I strongly suspect most of these people are watching avi's downloaded via torrents, and the reason you guys are getting quiet dialogue and loud SFX on some movies has nothing to do with Hollywood. For those of you with the problem on legal DVD's, it's exactly what Torerik said, you're missing the center channel, either because you don't have a center speaker or because you don't have VLC setup properly.

For those of you watching avi's, the reason is the same but harder to fix, because different rippers use different encoding techniques. Some, for instance, (and i'm not a sound guy so i may mix up the terminology) mix all the 5.1 channels into a stereo channel, and you have no problem, some encode both stereo and 5.1 feeds and encode both options under audio device, so you can manually switch between channels, and some encode all the 5.1 channels into stereo output without mixing them into a stereo channel, or encoding a 5.1 channel option, so you're stuck "riding the volume control", nothing to do about it but look for another copy by another ripper and hope they encoded right or go out and buy the DVD.

I know this is probably not entirely accurate, like I said I'm not a sound guy, I just have a basic understanding of why the sound is funky, and I hope I've given you the gist of it.

Myself, I was just looking for a way to normalize different movies in a playlist, some are so low i have to crank the volume up (all the sound, not just voice), but then the next movie will blast a hole in my eardrums. Hopefully the normalization settings posted in this thread will help, so thanks guys.

[thockin]

Come on. We're not all morons. Like the rest of this thread, I need a compressor/limiter. I don't always have a pristine listening environment (baby crying, on a plane, etc). I need the RMS power of the audio track boosted without blowing my headphones or speakers out or distorting all to hell on loud segments.

Audio compression. See http://en.wikipedia.org/wiki/Dynamic_range_compression

VLC pretty much owns the competition in all areas of media playback. It has a slew of plugins. Does anyone know if there is an audio compressor/limiter plugin ? If not, it would make a really great addition to VLC? I don't think we need virtual analog or tube-amp emulation or anything silly. Just a simple compressor - the code already exists in a million and one LADSPA, VST, etc plugins. All it needs is for someone intimately familiar with VLC to say "I can do that".

Heck, I would donate money to VLC devs for that feature? I just don't have time/energy/expertise to hack it together myself.

Tim (frustrated after riding the volume on yet another movie in VLC).

[monaco]

Come on. We're not all morons. Like the rest of this thread, I need a compressor/limiter.

I find it interesting that these people don't get it but the Apple fan boys (who know next to nothing) do. Compression is not some revolutionary technology, it's on every record for christ's sake, but these idiots on this forum are acting stupid, almost as what we're talking about is absurd.

Knowing that eventually, some day, every player will have compressors,limiters, dither, (you know, basic sh*t we use in our DAW's) makes me feel like we're living in the stone age. It's funny how some people can know so much about video, but not a damn thing about audio. fuken morons here.

On a side note, I wonder if you can watch them in iTunes with the Sound Check feature enabled.

[logiconcepts819]

Good news. I happened to have ported the SC4 compressor from LADSPA SWH to VLC, and it apparently works well with the right settings. Here is the patch that adds real dynamic range compression support to VLC:

Code: Select all

diff -purN vlc.orig/modules/audio_filter/compressor.c vlc/modules/audio_filter/compressor.c --- vlc.orig/modules/audio_filter/compressor.c 1969-12-31 18:00:00.000000000 -0600 +++ vlc/modules/audio_filter/compressor.c 2010-06-19 13:17:53.934683044 -0500 @@ -0,0 +1,826 @@ +/***************************************************************************** + * compressor.c: dynamic range compressor, ported from SC4 plugin + ***************************************************************************** + * + * Authors: Ronald Wright <logiconcepts819@gmail.com> + * Original authors: Steve Harris <steve@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 + +#ifdef DB_DEFAULT_CUBE +#define db2lin(a,b) (f_db2lin_cube(a,b)) +#define lin2db(a,b) (f_lin2db_cube(a,b)) +#else +#define db2lin(a,b) (f_db2lin_lerp(a,b)) +#define lin2db(a,b) (f_lin2db_lerp(a,b)) +#endif + +#define RMS_BUF_SIZE (64) + +#define LIN_INTERP(f,a,b) ((a) + (f) * ( (b) - (a) )) + +typedef struct +{ + float buffer[RMS_BUF_SIZE]; + unsigned int pos; + float sum; +} rms_env; + +struct filter_sys_t +{ + float rms_peak; + float attack; + float release; + float threshold; + float ratio; + float knee; + float makeup_gain; + float amplitude; + float gain_red; + + float amp; + float *as; + unsigned int count; + float env; + float env_peak; + float env_rms; + float gain; + float gain_t; + rms_env* rms; + float sum; + + float db_data[DB_TABLE_SIZE]; + float lin_data[LIN_TABLE_SIZE]; + + vlc_mutex_t lock; +}; + +typedef union +{ + float f; + int32_t i; +} ls_pcast32; + +static int Open ( vlc_object_t * ); +static void Close ( vlc_object_t * ); +static void db_init ( filter_sys_t * p_sys ); +static float f_db2lin_cube ( float db, filter_sys_t * p_sys ); +static float f_db2lin_lerp ( float db, filter_sys_t * p_sys ); +static float f_lin2db_cube ( float lin, filter_sys_t * p_sys ); +static float f_lin2db_lerp ( float lin, filter_sys_t * p_sys ); +static void round_to_zero ( volatile float *f ); +static float f_max ( float x, float a ); +static int f_round ( float f ); +static float cube_interp ( const float fr, const float inm1, + const float in, + const float inp1, + const float inp2 ); +static rms_env *rms_env_new ( void ); +static float rms_env_process ( rms_env * r, const float x ); +static void rms_env_free ( rms_env * r ); +static block_t *DoWork ( filter_t *, block_t * ); + +static int RMSPeakCallback ( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, + void * p_data ); +static int AttackCallback ( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, + void * p_data ); +static int ReleaseCallback ( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, + void * p_data ); +static int ThresholdCallback ( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, + void * p_data ); +static int RatioCallback ( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, + void * p_data ); +static int KneeCallback ( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, + void * p_data ); +static int MakeupGainCallback ( vlc_object_t *p_this, char const *psz_cmd, + vlc_value_t oldval, vlc_value_t newval, + void * p_data ); + +/***************************************************************************** + * Module descriptor + *****************************************************************************/ + +#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", 0 ) + set_category( CAT_AUDIO ) + set_subcategory( SUBCAT_AUDIO_AFILTER ) + + add_float( "compressor-rms-peak", 0.0, NULL, RMS_PEAK_TEXT, + RMS_PEAK_LONGTEXT, false ) + add_float( "compressor-attack", 1.5, NULL, ATTACK_TEXT, + ATTACK_LONGTEXT, false ) + add_float( "compressor-release", 800.0, NULL, RELEASE_TEXT, + RELEASE_LONGTEXT, false ) + add_float( "compressor-threshold", -11.0, NULL, THRESHOLD_TEXT, + THRESHOLD_LONGTEXT, false ) + add_float( "compressor-ratio", 10.0, NULL, RATIO_TEXT, + RATIO_LONGTEXT, false ) + add_float( "compressor-knee", 1.0, NULL, KNEE_TEXT, + KNEE_LONGTEXT, false ) + add_float( "compressor-makeup-gain", 10.0, 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; + struct filter_sys_t *p_sys; + float amp; + float *as = NULL; + unsigned int count; + float env; + float env_peak; + float env_rms; + float gain; + float gain_t; + rms_env *rms = NULL; + float sum; + + unsigned int i; + float sample_rate; + + 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 ) ) + { + memcpy( &p_filter->fmt_out.audio, &p_filter->fmt_in.audio, + sizeof(audio_sample_format_t) ); + msg_Warn( p_filter, "input and output formats are not similar" ); + return VLC_EGENERIC; + } + + sample_rate = (float)p_filter->fmt_in.audio.i_rate; + + p_filter->pf_audio_filter = DoWork; + + p_sys = p_filter->p_sys = malloc( sizeof( *p_sys ) ); + if ( !p_sys ) + { + return VLC_ENOMEM; + } + + rms = rms_env_new(); + if ( !rms ) + { + free( p_sys ); + return VLC_ENOMEM; + } + + sum = 0.0f; + amp = 0.0f; + gain = 0.0f; + gain_t = 0.0f; + env = 0.0f; + env_rms = 0.0f; + env_peak = 0.0f; + count = 0; + + as = malloc( A_TBL * sizeof(float) ); + if ( !as ) + { + rms_env_free( rms ); + free( p_sys ); + return VLC_ENOMEM; + } + + as[0] = 1.0f; + for ( i = 1; i < A_TBL; i++ ) { + as[i] = expf( -1.0f / ( sample_rate * (float)i / (float)A_TBL ) ); + } + + db_init( p_sys ); + + p_sys->amp = amp; + p_sys->as = as; + p_sys->count = count; + p_sys->env = env; + p_sys->env_peak = env_peak; + p_sys->env_rms = env_rms; + p_sys->gain = gain; + p_sys->gain_t = gain_t; + p_sys->rms = rms; + p_sys->sum = sum; + + vlc_mutex_init( &p_sys->lock ); + + p_sys->rms_peak = var_CreateGetFloat( p_aout, "compressor-rms-peak" ); + p_sys->attack = var_CreateGetFloat( p_aout, "compressor-attack" ); + p_sys->release = var_CreateGetFloat( p_aout, "compressor-release" ); + p_sys->threshold = var_CreateGetFloat( p_aout, "compressor-threshold" ); + p_sys->ratio = var_CreateGetFloat( p_aout, "compressor-ratio" ); + p_sys->knee = var_CreateGetFloat( p_aout, "compressor-knee" ); + p_sys->makeup_gain = var_CreateGetFloat( p_aout, "compressor-makeup-gain" ); + p_sys->amplitude = 0; + p_sys->gain_red = 0; + + /* Add our own callbacks */ + 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 ); + + 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; + + 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 ); + + vlc_mutex_destroy( &p_sys->lock ); + + rms_env_free( p_sys->rms ); + free( p_sys->as ); + free( p_sys ); +} + +/***************************************************************************** + * Helper functions for compressor + *****************************************************************************/ + +static void db_init( filter_sys_t * p_sys ) +{ + unsigned int i; + float *lin_data = p_sys->lin_data; + float *db_data = p_sys->db_data; + + for ( i = 0; i < LIN_TABLE_SIZE; i++ ) + { + lin_data[i] = powf( 10.0f, ( ( DB_MAX - DB_MIN ) * + (float)i / (float)LIN_TABLE_SIZE + DB_MIN ) / 20.0f ); + } + + for ( i = 0; i < DB_TABLE_SIZE; i++ ) + { + db_data[i] = 20.0f * log10f( ( LIN_MAX - LIN_MIN ) * + (float)i / (float)DB_TABLE_SIZE + LIN_MIN ); + } +} + +static float f_db2lin_cube( float db, filter_sys_t * p_sys ) +{ + float scale = ( db - DB_MIN ) * (float)LIN_TABLE_SIZE / ( DB_MAX - DB_MIN ); + int base = f_round( scale - 0.5f ); + float ofs = scale - base; + float *lin_data = p_sys->lin_data; + + if ( base < 1 ) + { + return 0.0f; + } + else if ( base > LIN_TABLE_SIZE - 3 ) + { + return lin_data[LIN_TABLE_SIZE - 2]; + } + + return cube_interp( ofs, lin_data[base - 1], + lin_data[base], + lin_data[base + 1], + lin_data[base + 2] ); +} + +static float f_db2lin_lerp( float db, filter_sys_t * p_sys ) +{ + float scale = ( db - DB_MIN ) * (float)LIN_TABLE_SIZE / ( DB_MAX - DB_MIN ); + int base = f_round( scale - 0.5f ); + float ofs = scale - base; + float *lin_data = p_sys->lin_data; + + if ( base < 1 ) + { + return 0.0f; + } + else if ( base > LIN_TABLE_SIZE - 3 ) + { + return lin_data[LIN_TABLE_SIZE - 2]; + } + + return ( 1.0f - ofs ) * lin_data[base] + ofs * lin_data[base + 1]; +} + +static float f_lin2db_cube( float lin, filter_sys_t * p_sys ) +{ + float scale = ( lin - LIN_MIN ) * (float)DB_TABLE_SIZE / + ( LIN_MAX - LIN_MIN ); + int base = f_round( scale - 0.5f ); + float ofs = scale - base; + float *db_data = p_sys->db_data; + + if ( base < 2 ) + { + return db_data[2] * scale * 0.5f - 23.0f * ( 2.0f - scale ); + } + else if ( base > DB_TABLE_SIZE - 3 ) + { + return db_data[DB_TABLE_SIZE - 2]; + } + + return cube_interp( ofs, db_data[base - 1], + db_data[base], + db_data[base + 1], + db_data[base + 2] ); +} + +static float f_lin2db_lerp( float lin, filter_sys_t * p_sys ) +{ + float scale = ( lin - LIN_MIN ) * (float)DB_TABLE_SIZE / + ( LIN_MAX - LIN_MIN ); + int base = f_round( scale - 0.5f ); + float ofs = scale - base; + float *db_data = p_sys->db_data; + + if ( base < 2 ) + { + return db_data[2] * scale * 0.5f - 23.0f * ( 2.0f - scale ); + } + else if ( base > DB_TABLE_SIZE - 2 ) + { + return db_data[DB_TABLE_SIZE - 1]; + } + + return ( 1.0f - ofs ) * db_data[base] + ofs * db_data[base + 1]; +} + +static void round_to_zero( volatile float *f ) +{ + *f += 1e-18; + *f -= 1e-18; +} + +static float f_max( float x, float a ) +{ + x -= a; + x += fabs( x ); + x *= 0.5; + x += a; + + return x; +} + +static int f_round( float f ) +{ + ls_pcast32 p; + + p.f = f; + p.f += ( 3 << 22 ); + + return p.i - 0x4b400000; +} + +static float cube_interp( const float fr, const float inm1, const float in, + const float inp1, const float inp2 ) +{ + return in + 0.5f * fr * ( inp1 - inm1 + + fr * ( 4.0f * inp1 + 2.0f * inm1 - 5.0f * in - inp2 + + fr * ( 3.0f * ( in - inp1 ) - inm1 + inp2 ) ) ); +} + +static rms_env * rms_env_new( void ) +{ + rms_env *new = (rms_env *)calloc( 1, sizeof(rms_env) ); + return new; +} + +static float rms_env_process( rms_env * r, const float x ) +{ + r->sum -= r->buffer[r->pos]; + r->sum += x; + if ( r->sum < 1.0e-6 ) + { + r->sum = 0.0f; + } + r->buffer[r->pos] = x; + r->pos = ( r->pos + 1 ) & ( RMS_BUF_SIZE - 1 ); + + return sqrt( r->sum / (float)RMS_BUF_SIZE ); +} + +static void rms_env_free( rms_env * r ) +{ + free( r ); +} + +/***************************************************************************** + * 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 *p_out = (float*)p_in_buf->p_buffer; + float *p_in = (float*)p_in_buf->p_buffer; + + float rms_peak, attack, release, threshold, ratio, knee, makeup_gain; + float amp, *as, env, env_peak, env_rms, gain, gain_t, sum; + unsigned int count; + rms_env *rms; + + float ga, gr, rs, mug, knee_min, knee_max, ef_a, ef_ai; + + int pos, pos_chan; + + /* Current configuration */ + struct filter_sys_t *p_sys = p_filter->p_sys; + + vlc_mutex_lock( &p_sys->lock ); + + /* RMS/peak (float value) */ + rms_peak = p_sys->rms_peak; + + /* Attack time (ms) (float value) */ + attack = p_sys->attack; + + /* Release time (ms) (float value) */ + release = p_sys->release; + + /* Threshold level (dB) (float value) */ + threshold = p_sys->threshold; + + /* Ratio (n:1) (float value) */ + ratio = p_sys->ratio; + + /* Knee radius (dB) (float value) */ + knee = p_sys->knee; + + /* Makeup gain (dB) (float value) */ + makeup_gain = p_sys->makeup_gain; + + amp = p_sys->amp; + as = p_sys->as; + count = p_sys->count; + env = p_sys->env; + env_peak = p_sys->env_peak; + env_rms = p_sys->env_rms; + gain = p_sys->gain; + gain_t = p_sys->gain_t; + rms = p_sys->rms; + sum = p_sys->sum; + + ga = attack < 2.0f ? 0.0f + : as[f_round( attack * 0.001f * (float)( A_TBL - 1 ) )]; + gr = as[f_round( release * 0.001f * (float)( A_TBL - 1 ) )]; + rs = ( ratio - 1.0f ) / ratio; + mug = db2lin( makeup_gain, p_sys ); + knee_min = db2lin( threshold - knee, p_sys ); + knee_max = db2lin( threshold + knee, p_sys ); + ef_a = ga * 0.25f; + ef_ai = 1.0f - ef_a; + + for ( pos = 0; pos < i_samples; pos++ ) + { + float lev_in = fabs( p_in[0] ); + for( pos_chan = 1; pos_chan < i_channels; pos_chan++ ) + { + lev_in = f_max( lev_in, fabs( p_in[pos_chan] ) ); + } + sum += lev_in * lev_in; + + if ( amp > env_rms ) + { + env_rms = env_rms * ga + amp * ( 1.0f - ga ); + } + else + { + env_rms = env_rms * gr + amp * ( 1.0f - gr ); + } + round_to_zero( &env_rms ); + if ( lev_in > env_peak ) + { + env_peak = env_peak * ga + lev_in * ( 1.0f - ga ); + } + else + { + env_peak = env_peak * gr + lev_in * ( 1.0f - gr ); + } + round_to_zero( &env_peak ); + if ( ( count++ & 3 ) == 3 ) + { + amp = rms_env_process( rms, sum * 0.25f ); + sum = 0.0f; + if ( isnan( env_rms ) ) + { + /* This can happen sometimes, but I don't know why */ + env_rms = 0.0f; + } + + env = LIN_INTERP( rms_peak, env_rms, env_peak ); + + if ( env <= knee_min ) + { + gain_t = 1.0f; + } + else if ( env < knee_max ) + { + const float x = -( threshold - knee + - lin2db( env, p_sys ) ) / knee; + gain_t = db2lin( -knee * rs * x * x * 0.25f, p_sys ); + } + else + { + gain_t = db2lin( ( threshold - lin2db( env, p_sys ) ) * rs, + p_sys ); + } + } + + gain = gain * ef_a + gain_t * ef_ai; + + for ( pos_chan = 0; pos_chan < i_channels; pos_chan++ ) + { + p_out[pos_chan] = p_in[pos_chan] * gain * mug; + } + + p_in += i_channels; + p_out += i_channels; + } + + p_sys->sum = sum; + p_sys->amp = amp; + p_sys->gain = gain; + p_sys->gain_t = gain_t; + p_sys->env = env; + p_sys->env_rms = env_rms; + p_sys->env_peak = env_peak; + p_sys->count = count; + + p_sys->amplitude = lin2db( env, p_sys ); + p_sys->gain_red = lin2db( gain, p_sys ); + + vlc_mutex_unlock( &p_sys->lock ); + + return p_in_buf; +} + +/***************************************************************************** + * Callback functions + *****************************************************************************/ + +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; + + if ( newval.f_float < 0.0 ) + { + newval.f_float = 0.0; + } + else if ( newval.f_float > 1.0 ) + { + newval.f_float = 1.0; + } + + vlc_mutex_lock( &p_sys->lock ); + p_sys->rms_peak = newval.f_float; + 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; + + if ( newval.f_float < 1.5 ) + { + newval.f_float = 1.5; + } + else if ( newval.f_float > 400.0 ) + { + newval.f_float = 400.0; + } + + vlc_mutex_lock( &p_sys->lock ); + p_sys->attack = newval.f_float; + 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; + + if ( newval.f_float < 2.0 ) + { + newval.f_float = 2.0; + } + else if ( newval.f_float > 800.0 ) + { + newval.f_float = 800.0; + } + + vlc_mutex_lock( &p_sys->lock ); + p_sys->release = newval.f_float; + 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; + + if ( newval.f_float < -30.0 ) + { + newval.f_float = -30.0; + } + else if ( newval.f_float > 0.0 ) + { + newval.f_float = 0.0; + } + + vlc_mutex_lock( &p_sys->lock ); + p_sys->threshold = newval.f_float; + 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; + + if ( newval.f_float < 1.0 ) + { + newval.f_float = 1.0; + } + else if ( newval.f_float > 20.0 ) + { + newval.f_float = 20.0; + } + + vlc_mutex_lock( &p_sys->lock ); + p_sys->ratio = newval.f_float; + 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; + + if ( newval.f_float < 1.0 ) + { + newval.f_float = 1.0; + } + else if ( newval.f_float > 10.0 ) + { + newval.f_float = 10.0; + } + + vlc_mutex_lock( &p_sys->lock ); + p_sys->knee = newval.f_float; + 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; + + if ( newval.f_float < 0.0 ) + { + newval.f_float = 0.0; + } + else if ( newval.f_float > 24.0 ) + { + newval.f_float = 24.0; + } + + vlc_mutex_lock( &p_sys->lock ); + p_sys->makeup_gain = newval.f_float; + vlc_mutex_unlock( &p_sys->lock ); + + return VLC_SUCCESS; +} diff -purN vlc.orig/modules/audio_filter/Modules.am vlc/modules/audio_filter/Modules.am --- vlc.orig/modules/audio_filter/Modules.am 2010-06-19 13:09:58.074687899 -0500 +++ vlc/modules/audio_filter/Modules.am 2010-06-19 13:10:43.144685059 -0500 @@ -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. libvlc_LTLIBRARIES += \ libaudiobargraph_a_plugin.la \ libchorus_flanger_plugin.la \ + libcompressor_plugin.la \ libequalizer_plugin.la \ libnormvol_plugin.la \ libparam_eq_plugin.la \ diff -purN vlc.orig/modules/gui/qt4/components/extended_panels.cpp vlc/modules/gui/qt4/components/extended_panels.cpp --- vlc.orig/modules/gui/qt4/components/extended_panels.cpp 2010-06-19 13:09:58.144688104 -0500 +++ vlc/modules/gui/qt4/components/extended_panels.cpp 2010-06-19 13:15:36.317535503 -0500 @@ -1117,6 +1117,176 @@ void Equalizer::addCallbacks( aout_insta **********************************************************************/ /********************************************************************** + * Dynamic range compressor + **********************************************************************/ +static const char *psz_comp_control_names[] = +{ + "compressor-rms-peak", "compressor-attack", "compressor-release", + "compressor-threshold", "compressor-ratio", "compressor-knee", + "compressor-makeup-gain" +}; + +static const char *psz_comp_control_descs[] = +{ + "RMS/peak", "Attack", "Release", "Threshold", "Ratio", "Knee radius", + "Makeup gain" +}; + +static const char *psz_comp_control_units[] = +{ + "", " ms", " ms", " dB", ":1", " dB", " dB" +}; + +static const float f_comp_min_max_val_res_data[][4] = +{ + // min max value resolution + // ---- ----- ----- ---------- + { 0.0, 1.0, 0.0, 0.001 }, // RMS/peak + { 1.5, 400.0, 1.5, 0.100 }, // Attack + { 2.0, 800.0, 800.0, 0.100 }, // Release + { -30.0, 0.0, -11.0, 0.010 }, // Threshold + { 1.0, 20.0, 10.0, 0.010 }, // Ratio + { 1.0, 10.0, 1.0, 0.010 }, // Knee radius + { 0.0, 24.0, 10.0, 0.010 } // Makeup gain +}; + +Compressor::Compressor( intf_thread_t *_p_intf, QWidget *_parent ) : + QWidget( _parent ) , p_intf( _p_intf ) +{ + QFont smallFont = QApplication::font( static_cast<QWidget*>( 0 ) ); + smallFont.setPointSize( smallFont.pointSize() - 3 ); + + QGridLayout *layout = new QGridLayout( this ); + layout->setMargin( 0 ); + + enableCheck = new QCheckBox( qtr( "Enable dynamic range compressor" ) ); + layout->addWidget( enableCheck, 0, 0, 1, NUM_CP_CTRL ); + + for( int i = 0 ; i < NUM_CP_CTRL ; i++ ) + { + compCtrl[i] = new QSlider( Qt::Vertical ); + + const int i_min = (int)( f_comp_min_max_val_res_data[i][0] + / f_comp_min_max_val_res_data[i][3] ); + const int i_max = (int)( f_comp_min_max_val_res_data[i][1] + / f_comp_min_max_val_res_data[i][3] ); + const int i_val = (int)( f_comp_min_max_val_res_data[i][2] + / f_comp_min_max_val_res_data[i][3] ); + + compCtrl[i]->setMinimum( i_min ); + compCtrl[i]->setMaximum( i_max ); + compCtrl[i]->setValue( i_val ); + oldControlVars[i] = f_comp_min_max_val_res_data[i][2]; + CONNECT( compCtrl[i], valueChanged( int ), this, setInitValues() ); + ctrl_texts[i] = new QLabel( qtr( psz_comp_control_descs[i] ) + + qtr( "\n" ) ); + ctrl_texts[i]->setFont( smallFont ); + ctrl_readout[i] = new QLabel( qtr( "" ) ); + ctrl_readout[i]->setFont( smallFont ); + layout->addWidget( compCtrl[i], 1, i ); + layout->addWidget( ctrl_readout[i], 2, i ); + layout->addWidget( ctrl_texts[i], 3, i ); + } + + BUTTONACT( enableCheck, enable() ); + + /* Write down initial values */ + aout_instance_t *p_aout = THEMIM->getAout(); + char *psz_af; + + if( p_aout ) + { + psz_af = var_GetNonEmptyString( p_aout, "audio-filter" ); + for( int i = 0; i < NUM_CP_CTRL; i++ ) + { + controlVars[i] = var_GetFloat( p_aout, + psz_comp_control_names[i] ); + } + vlc_object_release( p_aout ); + } + else + { + psz_af = config_GetPsz( p_intf, "audio-filter" ); + for( int i = 0; i < NUM_CP_CTRL; i++ ) + { + controlVars[i] = config_GetFloat( p_intf, + psz_comp_control_names[i] ); + } + } + if( psz_af && strstr( psz_af, "compressor" ) != NULL ) + enableCheck->setChecked( true ); + free( psz_af ); + enable( enableCheck->isChecked() ); + updateSliders( controlVars ); + setValues( controlVars ); +} + +Compressor::~Compressor() +{ +} + +void Compressor::enable() +{ + bool en = enableCheck->isChecked(); + aout_EnableFilter( THEPL, "compressor", en ); + enable( en ); +} + +void Compressor::enable( bool en ) +{ + for( int i = 0 ; i < NUM_CP_CTRL ; i++ ) + { + compCtrl[i]->setEnabled( en ); + ctrl_texts[i]->setEnabled( en ); + ctrl_readout[i]->setEnabled( en ); + } +} + +void Compressor::updateSliders( float * controlVars ) +{ + for( int i = 0 ; i < NUM_CP_CTRL ; i++ ) + { + if ( oldControlVars[i] != controlVars[i] ) + { + const int i_val = (int)( controlVars[i] + / f_comp_min_max_val_res_data[i][3] ); + compCtrl[i]->setValue( i_val ); + } + } +} + +void Compressor::setInitValues() +{ + setValues( controlVars ); +} + +void Compressor::setValues( float * controlVars ) +{ + aout_instance_t *p_aout = THEMIM->getAout(); + + for( int i = 0 ; i < NUM_CP_CTRL ; i++ ) + { + float f = (float)( compCtrl[i]->value() ) + * ( f_comp_min_max_val_res_data[i][3] ); + ctrl_readout[i]->setText( QString::number( f, 'f', 1 ) + + qtr( psz_comp_control_units[i] ) ); + if( oldControlVars[i] != f ) + { + if( p_aout ) + { + var_SetFloat( p_aout, psz_comp_control_names[i], f ); + } + config_PutFloat( p_intf, psz_comp_control_names[i], f ); + oldControlVars[i] = f; + } + } + if( p_aout ) + { + vlc_object_release( p_aout ); + } +} + +/********************************************************************** * Spatializer **********************************************************************/ static const char *psz_control_names[] = diff -purN vlc.orig/modules/gui/qt4/components/extended_panels.hpp vlc/modules/gui/qt4/components/extended_panels.hpp --- vlc.orig/modules/gui/qt4/components/extended_panels.hpp 2010-06-19 13:09:58.154691544 -0500 +++ vlc/modules/gui/qt4/components/extended_panels.hpp 2010-06-19 13:13:22.525934483 -0500 @@ -38,6 +38,7 @@ #include <QTabWidget> #define BANDS 10 +#define NUM_CP_CTRL 7 #define NUM_SP_CTRL 5 class QSignalMapper; @@ -115,6 +116,33 @@ private slots: void setCorePreset(int); }; +class Compressor: public QWidget +{ + Q_OBJECT +public: + Compressor( intf_thread_t *, QWidget * ); + virtual ~Compressor(); + +private: + QSlider *compCtrl[NUM_CP_CTRL]; + QLabel *ctrl_texts[NUM_CP_CTRL]; + QLabel *ctrl_readout[NUM_CP_CTRL]; + float controlVars[7]; + float oldControlVars[7]; + + QCheckBox *enableCheck; + + void delCallbacks( aout_instance_t * ); + void addCallbacks( aout_instance_t * ); + intf_thread_t *p_intf; +private slots: + void enable(bool); + void enable(); + void updateSliders(float *); + void setValues(float *); + void setInitValues(); +}; + class Spatializer: public QWidget { Q_OBJECT diff -purN vlc.orig/modules/gui/qt4/dialogs/extended.cpp vlc/modules/gui/qt4/dialogs/extended.cpp --- vlc.orig/modules/gui/qt4/dialogs/extended.cpp 2010-06-19 13:09:58.144688104 -0500 +++ vlc/modules/gui/qt4/dialogs/extended.cpp 2010-06-19 13:12:35.184683318 -0500 @@ -54,6 +54,9 @@ ExtendedDialog::ExtendedDialog( intf_thr equal = new Equalizer( p_intf, audioTab ); audioTab->addTab( equal, qtr( "Graphic Equalizer" ) ); + Compressor *compres = new Compressor( p_intf, audioTab ); + audioTab->addTab( compres, qtr( "Compressor" ) ); + Spatializer *spatial = new Spatializer( p_intf, audioTab ); audioTab->addTab( spatial, qtr( "Spatializer" ) ); audioLayout->addWidget( audioTab );

Of course, this patch was applied to the GIT version of VLC, so I cannot guarantee that this patch will work with the current release. Also, I cannot guarantee that the added code will work with the OS X version. Finally, I cannot seem to force the compressed audio to go into the equalizer so that the equalized audio doesn't go into the compressor. Any assistance is most helpful.

(Oops, I didn't realize this thread is in the OS X forums. I apologize.)

[Jean-Baptiste Kempf]

Can you paste on vlc-devel this patch ?

[logiconcepts819]

Indeed the patch has been posted.

[Jean-Baptiste Kempf]

Many thanks.

[wateras]

So does anyone have any new updates on possible solutions for this?

I experience the same thing. Volume on almost every movie I watch is completely out of control and ridiculous. Music and sound effects are WAY TOO LOUD and speech is unbelievably soft.

I find it extremely hard to believe that I'm in a minority here, being only one of a handful of people who can't stand this...

While its not entirely ridiculous for home theater, it can be for headphones and such setups. I got rude surprises using vlc on headphones a couple times, i wonder how much hearing i've lost:(

mediaplayer classic and other players that use ffdshow or whatever seem to allow for better settings on this. In mixer you can chance each speakers output separately..i boost the center channel. Id still like to see a working solution on vlc though.

Yes, this is crazy. Why the hell does Hollywood do this? Who is sitting in their office thinking that such a crazy disparity in sound volume is a good thing?

Wide dymanic range is more, not less. It is however bad for quiet or non home theater setups that cannot compensate with "night mode" or some such ability to fiddle with audio range compression and or boost the center channel.

[wateras]

VLC pretty much owns the competition in all areas of media playback.

lol
well not in this area, ffdshow audio settings just seem far more well done.

i use vlc for dvd and some mkv with subs that my coreavc+mediaplayer classic bodge up so i do need this ability in vlc as well though:(

[preposterous]

As an amateur DAW user and person with similar frustration it seems to me, correct me if I am wrong, that the first answer photon gave was excellent and what people like us need is a video player vst plugin for our daw, not a compression/limiter plugin for VLC.

[VLC Fan]

Sorry to post this here but it really is elinghtening...
The Dynamic Range Compression (DRC) feature is not esoteric at all: it is the only featre that keeps me watching DVDs on Nero Showtime rather than on VLC. To the developers I'd say, if you missunderstood the other people here, I'd go check how DRC works in Nero Showtime -I too, want to hear (in relative terms) the soft sounds brought up a, and the ridiculous blasts brought down; the fixing of the relatively low volume of voice/dialogues is but an example of that. Even an originla DVD suffers from moronic sound volume settings by the people who recorded it.

[koggit]

I registered to bump and ask- any solution yet?

[roger jonsson]

I use Audio Hijack Pro http://www.rogueamoeba.com/ to insert AU plugins for VLC (VLC as input in Audio Hijack Pro).
I don't use it to compress dynamics, I like dynamics, but it of course can be done with an compressor or limiter plugin of your choice inside Audio hijack Pro.

[aaronsds]

This is something that really bothers me as well, I often watch films late at night and don't want to disturb others, but if i turn it down so that the loud sounds are quiet enough, I can't hear anything that's being spoken.

Anyway, I have solved this at least on linux by using Jack. I use the Jack plugin for VLC to output to jack and with qjackcontrol, have the output go to a program called calf, which hosts a dynamic range compression plugin, and then this is routed out to the speakers. I get full control over the compression settings this way.

Edit: apologies, just noticed this was in the Mac forum, I got to this thread through a google search earlier so I didn't notice. Not sure if anything similar to jack exists on macs, but considering they are quite highly regarded for media work, I'd be surprised if there wasn't some way to route audio.

is there any chance that you have a guide on how to set all of that up?

[aitikin]

I find it interesting that these people don't get it but the Apple fan boys (who know next to nothing) do. Compression is not some revolutionary technology, it's on every record for christ's sake, but these idiots on this forum are acting stupid, almost as what we're talking about is absurd.

Knowing that eventually, some day, every player will have compressors,limiters, dither, (you know, basic sh*t we use in our DAW's) makes me feel like we're living in the stone age. It's funny how some people can know so much about video, but not a damn thing about audio. fuken morons here.

On a side note, I wonder if you can watch them in iTunes with the Sound Check feature enabled.

I just had to register solely to respond to this idiocy. iTunes Sound Check sends tracks with a lower RMS through a brickwall limiter while Lowering the overall gain on tracks with a higher RMS, somewhat giving the illusion of giving them the same level. This can be all well and good, except for the simple fact that this destroys classical albums, and is still a very poor solution, seeing as it's still sending everything through a brickwall limiter.

For those who do not understand what a brickwall limiter is, it basically should only ever be used for preventing distortion and preventing hearing loss, not dynamic control so much. A DNR (Dynamic Range Compressor) will allow for louds and softs to still exist, however they will not have as much of a difference between them. Truly, a dbx compressor emulator would be more functional as it has a wonderful auto setting.

[rant]As for those claiming that this is poor encoding or poor configuration, that may be the case in some, but more likely than not, it's not the case in all. As audio moved from the analog (tape) to digital (data) spectra, there was a loss of inherent compression that comes with tape. Whenever something is recorded to digital it is merely ones and zeros, but when recorded to tape, there is a natural compression. This, ironically, has not been exemplified in CDs because the bands and the record execs want their albums to be "louder" which defeats the purpose of having a huge dynamic range (CDs can theoretically have more dynamic range than a symphony orchestra). This leads to the video world, with many things (especially things recorded in HD) simply going digital, there's no analog tape involved, so the same mixing techniques aren't working the same and the movie doesn't hold up (aurally) to the less than stellar backgrounds that we must deal with. It may hold up fine in a studio or a theater, but often times that's not good enough for the real world.[/rant]

Regardless, I would love to see a somewhat controllable DNR compressor available, even if it is simply a ratio control.