[vlc-devel] Dynamic range compressor patch
Laurent Aimar
fenrir at elivagar.org
Sat Jun 26 21:51:58 CEST 2010
On Fri, Jun 25, 2010 at 03:39:30PM -0500, Ron Wright wrote:
> Hello,
>
> I have made many drastic changes in the compressor.c file and some minor
> changes in the UI code. Here are some of the changes:
>
> 1. Added VLC copyright and $id$ tag to the banner.
> 2. Made the filter structure, RMS envelope structure, and attack lookup table
> contiguous in memory so that only one memory allocation and one memory
> deallocation is needed.
> 3. Reorganized the functions so that they don't appear out of place.
> 4. Removed two wasteful members (amplitude and gain reduction) from the filter
> structure (this is probably used to implement output meters in certain LADSPA
> frontends).
> 5. The release of the mutex lock in the DoWork function now occurs immediately
> after retrieving the shared values.
> 6. Replaced the logic in the callback functions with clipping functions.
> 7. Removed useless casts.
> 8. Changed default values.
>
> Here are some notes to take into consideration:
>
> 1. The f_max and f_clamp functions work as expected. Proof:
>
> f_max (with arguments x and a):
>
> If x < a, then 0.5 * ((x - a) + fabs(x - a)) + a = 0.5 * ((x - a) - (x - a)) +
> a = a
> If x >= a, then 0.5 * ((x - a) + fabs(x - a)) + a = 0.5 * ((x - a) + (x - a)) +
> a = 0.5*2*(x - a) + a = x
>
> f_clamp (with arguments x, a, and b):
>
> If x < a, then 0.5 * (fabs(x - a) + a + b - fabs(x - b)) = 0.5 * (-(x - a) + a
> + b + (x - b)) = 0.5 * (2*a - x + b + x - b) = a
> If a <= x < b, then 0.5 * (fabs(x - a) + a + b - fabs(x - b)) = 0.5 * ((x - a)
> + a + b + (x - b)) = 0.5 * (x + b + x - b) = x
> If x >= b, then 0.5 * (fabs(x - a) + a + b - fabs(x - b)) = 0.5 * ((x - a) + a
> + b - (x - b)) = 0.5 * (x + b - x + b) = b
>
> These are branchless clipping functions from Laurent de Soras.
>
> 2. A buffer overflow will not occur when either attack or release converted to
> seconds is multiplied by (A_TBL - 1), with the resulting value used as an index
> to retrieve the attack (or release) value from the lookup table. The maximum
> of the maximum values of attack and release is 800, and A_TBL is 256, so 800 *
> 0.001 * 255 = 204. This means that the index will neither go beyond 255 nor
> even reach 255. The minimum of the minimum values of attack and release is
> 1.5. However, the real minimum value used in the code is 2. Nevertheless,
> there will also be no buffer underflow, since this value is positive.
Thanks,
> 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 \
I am not sure, but seeing that you use mathematics functions, I think you also
need to modify configure.ac to link with -lm.
> diff --git a/modules/audio_filter/compressor.c b/modules/audio_filter/compressor.c
> new file mode 100644
> index 0000000..daa260a
> --- /dev/null
> +++ b/modules/audio_filter/compressor.c
> @@ -0,0 +1,683 @@
> +/*****************************************************************************
> + * compressor.c: dynamic range compressor, ported from SC4 plugin
> + *****************************************************************************
> + * Copyright (C) 2010 the VideoLAN team
Please use your name instead of 'the VideoLAN team' (as the code is ported, dunno
if the original copyright shouldn't be used too).
> +#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 (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 amp;
> + float *as;
> + unsigned int count;
> + float env;
> + float env_peak;
> + float env_rms;
> + float gain;
> + float gain_thres;
> + rms_env* rms;
> + float sum;
> +
> + float db_data[DB_TABLE_SIZE];
> + float lin_data[LIN_TABLE_SIZE];
> +
> + vlc_mutex_t lock;
> +};
I meant to use directly
rms_env rms;
float as[A_TBL].
to simplify the allocation.
Also, if you could move the variables protected by the lock below the
declaration of the lock, it would make it easier to maintain.
> +/*****************************************************************************
> + * 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 sample_rate = p_filter->fmt_in.audio.i_rate;
> + struct filter_sys_t *p_sys;
> + unsigned int i;
> +
> + 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) );
p_filter->fmt_out.audio = p_filter->fmt_in.audio;
is simpler.
> + msg_Warn( p_filter, "input and output formats are not similar" );
> + return VLC_EGENERIC;
> + }
> +
> + p_sys = p_filter->p_sys = calloc( 1, sizeof(struct filter_sys_t)
> + + sizeof(rms_env)
> + + A_TBL * sizeof(float) );
> + if( !p_sys )
> + {
> + return VLC_ENOMEM;
> + }
> + p_sys->rms = (rms_env*)( p_sys + 1 );
> + p_sys->as = (float *)( p_sys->rms + 1 );
See remarks above at filter_sys_t.
> +/*****************************************************************************
> + * 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 );
> +
> + 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 *p_buf = (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_thres, 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;
> +
> + vlc_mutex_unlock( &p_sys->lock );
> +
> + 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_thres = p_sys->gain_thres;
> + rms = p_sys->rms;
> + sum = p_sys->sum;
> +
> + ga = attack < 2.0f ? 0.0f
> + : as[f_round( attack * 0.001f * ( A_TBL - 1 ) )];
> + gr = as[f_round( release * 0.001f * ( 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++ )
If you want, you can also declare a variable inside code (we accept c99) or in for()
statement.
Regards,
--
fenrir
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