[melted] / src / modules / sox / filter_sox.c

/*
 * filter_sox.c -- apply any number of SOX effects using libst
 * Copyright (C) 2003-2004 Ushodaya Enterprises Limited
 * Author: Dan Dennedy <dan@dennedy.org>
 *
 * 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., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
 */

#include "filter_sox.h"

#include <framework/mlt_frame.h>
#include <framework/mlt_tokeniser.h>

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>

#include <st.h>

#define BUFFER_LEN 8192
#define AMPLITUDE_NORM 0.2511886431509580 /* -12dBFS */
#define AMPLITUDE_MIN 0.00001

/** Compute the mean of a set of doubles skipping unset values flagged as -1
*/
static inline double mean( double *buf, int count )
{
        double mean = 0;
        int i;
        int j = 0;
        
        for ( i = 0; i < count; i++ )
        {
                if ( buf[ i ] != -1.0 )
                {
                        mean += buf[ i ];
                        j ++;
                }
        }
        if ( j > 0 )
                mean /= j;
        
        return mean;
}

/** Create an effect state instance for a channels
*/
static int create_effect( mlt_filter this, char *value, int count, int channel, int frequency )
{
        mlt_tokeniser tokeniser = mlt_tokeniser_init();
        eff_t eff = mlt_pool_alloc( sizeof( struct st_effect ) );
        char id[ 256 ];
        int error = 1;

        // Tokenise the effect specification
        mlt_tokeniser_parse_new( tokeniser, value, " " );

        // Locate the effect
        int opt_count = st_geteffect_opt( eff, tokeniser->count, tokeniser->tokens );
        
        // If valid effect
        if ( opt_count != ST_EOF )
        {
                // Supply the effect parameters
                if ( ( * eff->h->getopts )( eff, opt_count, &tokeniser->tokens[ tokeniser->count - opt_count ] ) == ST_SUCCESS )
                {
                        // Set the sox signal parameters
                        eff->ininfo.rate = frequency;
                        eff->outinfo.rate = frequency;
                        eff->ininfo.channels = 1;
                        eff->outinfo.channels = 1;
                        
                        // Start the effect
                        if ( ( * eff->h->start )( eff ) == ST_SUCCESS )
                        {
                                // Construct id
                                sprintf( id, "_effect_%d_%d", count, channel );

                                // Save the effect state
                                mlt_properties_set_data( MLT_FILTER_PROPERTIES( this ), id, eff, 0, mlt_pool_release, NULL );
                                error = 0;
                        }
                }
        }
        // Some error occurred so delete the temp effect state
        if ( error == 1 )
                mlt_pool_release( eff );
        
        mlt_tokeniser_close( tokeniser );
        
        return error;
}

/** Get the audio.
*/

static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
        // Get the properties of the frame
        mlt_properties properties = MLT_FRAME_PROPERTIES( frame );

        // Get the filter service
        mlt_filter filter = mlt_frame_pop_audio( frame );

        // Get the filter properties
        mlt_properties filter_properties = MLT_FILTER_PROPERTIES( filter );

        // Get the properties
        st_sample_t *input_buffer = mlt_properties_get_data( filter_properties, "input_buffer", NULL );
        st_sample_t *output_buffer = mlt_properties_get_data( filter_properties, "output_buffer", NULL );
        int channels_avail = *channels;
        int i; // channel
        int count = mlt_properties_get_int( filter_properties, "effect_count" );

        // Get the producer's audio
        mlt_frame_get_audio( frame, buffer, format, frequency, &channels_avail, samples );

        // Duplicate channels as necessary
        if ( channels_avail < *channels )
        {
                int size = *channels * *samples * sizeof( int16_t );
                int16_t *new_buffer = mlt_pool_alloc( size );
                int j, k = 0;
                
                // Duplicate the existing channels
                for ( i = 0; i < *samples; i++ )
                {
                        for ( j = 0; j < *channels; j++ )
                        {
                                new_buffer[ ( i * *channels ) + j ] = (*buffer)[ ( i * channels_avail ) + k ];
                                k = ( k + 1 ) % channels_avail;
                        }
                }
                
                // Update the audio buffer now - destroys the old
                mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
                
                *buffer = new_buffer;
        }
        else if ( channels_avail == 6 && *channels == 2 )
        {
                // Nasty hack for ac3 5.1 audio - may be a cause of failure?
                int size = *channels * *samples * sizeof( int16_t );
                int16_t *new_buffer = mlt_pool_alloc( size );
                
                // Drop all but the first *channels
                for ( i = 0; i < *samples; i++ )
                {
                        new_buffer[ ( i * *channels ) + 0 ] = (*buffer)[ ( i * channels_avail ) + 2 ];
                        new_buffer[ ( i * *channels ) + 1 ] = (*buffer)[ ( i * channels_avail ) + 3 ];
                }

                // Update the audio buffer now - destroys the old
                mlt_properties_set_data( properties, "audio", new_buffer, size, ( mlt_destructor )mlt_pool_release, NULL );
                
                *buffer = new_buffer;
        }

        // Even though some effects are multi-channel aware, it is not reliable
        // We must maintain a separate effect state for each channel
        for ( i = 0; i < *channels; i++ )
        {
                char id[ 256 ];
                sprintf( id, "_effect_0_%d", i );
                
                // Get an existing effect state
                eff_t e = mlt_properties_get_data( filter_properties, id, NULL );
                
                // Validate the existing effect state
                if ( e != NULL && ( e->ininfo.rate != *frequency || 
                                                        e->outinfo.rate != *frequency ) )
                        e = NULL;
                
                // (Re)Create the effect state
                if ( e == NULL )
                {
                        int j = 0;
                        
                        // Reset the count
                        count = 0;
        
                        // Loop over all properties
                        for ( j = 0; j < mlt_properties_count( filter_properties ); j ++ )
                        {
                                // Get the name of this property
                                char *name = mlt_properties_get_name( filter_properties, j );
        
                                // If the name does not contain a . and matches effect
                                if ( !strncmp( name, "effect", 6 ) )
                                {
                                        // Get the effect specification
                                        char *value = mlt_properties_get( filter_properties, name );
        
                                        // Create an instance
                                        if ( create_effect( filter, value, count, i, *frequency ) == 0 )
                                                count ++;
                                }
                        }
                        
                        // Save the number of filters
                        mlt_properties_set_int( filter_properties, "effect_count", count );
                        
                }
                if ( *samples > 0 && count > 0 )
                {
                        st_sample_t *p = input_buffer;
                        st_sample_t *end = p + *samples;
                        int16_t *q = *buffer + i;
                        st_size_t isamp = *samples;
                        st_size_t osamp = *samples;
                        double rms = 0;
                        int j;
                        char *normalise = mlt_properties_get( filter_properties, "normalise" );
                        double normalised_gain = 1.0;
                        
                        // Convert to sox encoding
                        while( p != end )
                        {
                                *p = ST_SIGNED_WORD_TO_SAMPLE( *q );
                                
                                // Compute rms amplitude while we are accessing each sample
                                rms += ( double )*p * ( double )*p;
                                
                                p ++;
                                q += *channels;
                        }
                        
                        // Compute final rms amplitude
                        rms = sqrt( rms / *samples / ST_SSIZE_MIN / ST_SSIZE_MIN );
                        
                        if ( normalise )
                        {
                                int window = mlt_properties_get_int( filter_properties, "window" );
                                double *smooth_buffer = mlt_properties_get_data( filter_properties, "smooth_buffer", NULL );
                                double max_gain = mlt_properties_get_double( filter_properties, "max_gain" );
                                
                                // Default the maximum gain factor to 20dBFS
                                if ( max_gain == 0 )
                                        max_gain = 10.0;
                                
                                // The smoothing buffer prevents radical shifts in the gain level
                                if ( window > 0 && smooth_buffer != NULL )
                                {
                                        int smooth_index = mlt_properties_get_int( filter_properties, "_smooth_index" );
                                        smooth_buffer[ smooth_index ] = rms;
                                        
                                        // Ignore very small values that adversely affect the mean
                                        if ( rms > AMPLITUDE_MIN )
                                                mlt_properties_set_int( filter_properties, "_smooth_index", ( smooth_index + 1 ) % window );
                                        
                                        // Smoothing is really just a mean over the past N values
                                        normalised_gain = AMPLITUDE_NORM / mean( smooth_buffer, window );
                                }
                                else if ( rms > 0 )
                                {
                                        // Determine gain to apply as current amplitude
                                        normalised_gain = AMPLITUDE_NORM / rms;
                                }
                                        
                                //printf("filter_sox: rms %.3f gain %.3f\n", rms, normalised_gain );
                                
                                // Govern the maximum gain
                                if ( normalised_gain > max_gain )
                                        normalised_gain = max_gain;
                        }
                        
                        // For each effect
                        for ( j = 0; j < count; j++ )
                        {
                                sprintf( id, "_effect_%d_%d", j, i );
                                e = mlt_properties_get_data( filter_properties, id, NULL );
                                
                                // We better have this guy
                                if ( e != NULL )
                                {
                                        float saved_gain = 1.0;
                                        
                                        // XXX: hack to apply the normalised gain level to the vol effect
                                        if ( normalise && strcmp( e->name, "vol" ) == 0 )
                                        {
                                                float *f = ( float * )( e->priv );
                                                saved_gain = *f;
                                                *f = saved_gain * normalised_gain;
                                        }
                                        
                                        // Apply the effect
                                        if ( ( * e->h->flow )( e, input_buffer, output_buffer, &isamp, &osamp ) == ST_SUCCESS )
                                        {
                                                // Swap input and output buffer pointers for subsequent effects
                                                p = input_buffer;
                                                input_buffer = output_buffer;
                                                output_buffer = p;
                                        }
                                        
                                        // XXX: hack to restore the original vol gain to prevent accumulation
                                        if ( normalise && strcmp( e->name, "vol" ) == 0 )
                                        {
                                                float *f = ( float * )( e->priv );
                                                *f = saved_gain;
                                        }
                                }
                        }
                        
                        // Convert back to signed 16bit
                        p = input_buffer;
                        q = *buffer + i;
                        end = p + *samples;
                        while ( p != end )
                        {
                                *q = ST_SAMPLE_TO_SIGNED_WORD( *p ++ );
                                q += *channels;
                        }
                }
        }

        return 0;
}

/** Filter processing.
*/

static mlt_frame filter_process( mlt_filter this, mlt_frame frame )
{
        if ( mlt_frame_is_test_audio( frame ) != 0 )
        {
                // Add the filter to the frame
                mlt_frame_push_audio( frame, this );
                mlt_frame_push_audio( frame, filter_get_audio );
                
                // Parse the window property and allocate smoothing buffer if needed
                mlt_properties properties = MLT_FILTER_PROPERTIES( this );
                int window = mlt_properties_get_int( properties, "window" );
                if ( mlt_properties_get( properties, "smooth_buffer" ) == NULL && window > 1 )
                {
                        // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation
                        double *smooth_buffer = (double*) calloc( window, sizeof( double ) );
                        int i;
                        for ( i = 0; i < window; i++ )
                                smooth_buffer[ i ] = -1.0;
                        mlt_properties_set_data( properties, "smooth_buffer", smooth_buffer, 0, free, NULL );
                }
        }

        return frame;
}

/** Constructor for the filter.
*/

mlt_filter filter_sox_init( char *arg )
{
        mlt_filter this = mlt_filter_new( );
        if ( this != NULL )
        {
                void *input_buffer = mlt_pool_alloc( BUFFER_LEN );
                void *output_buffer = mlt_pool_alloc( BUFFER_LEN );
                mlt_properties properties = MLT_FILTER_PROPERTIES( this );
                
                this->process = filter_process;
                
                if ( arg != NULL )
                        mlt_properties_set( properties, "effect", arg );
                mlt_properties_set_data( properties, "input_buffer", input_buffer, BUFFER_LEN, mlt_pool_release, NULL );
                mlt_properties_set_data( properties, "output_buffer", output_buffer, BUFFER_LEN, mlt_pool_release, NULL );
                mlt_properties_set_int( properties, "window", 75 );
        }
        return this;
}

// What to do when a libst internal failure occurs
void cleanup(void){}

// Is there a build problem with my sox-devel package?
#ifndef gsm_create
void gsm_create(void){}
#endif
#ifndef gsm_decode
void gsm_decode(void){}
#endif
#ifndef gdm_encode
void gsm_encode(void){}
#endif
#ifndef gsm_destroy
void gsm_destroy(void){}
#endif
#ifndef gsm_option
void gsm_option(void){}
#endif