X-Git-Url: http://research.m1stereo.tv/gitweb?a=blobdiff_plain;f=src%2Fmodules%2Fcore%2Ffilter_volume.c;h=04fb9364ec4bbf0e284d1b7a5ce0a779cfe0ef07;hb=bd208d01a2a792e698a9b4884b43602b2f245a8f;hp=719e3ed7abe4130967730f3b3320e541d655024f;hpb=dfcac1bf1428068f68b700d872452d9aa80a8a1d;p=melted diff --git a/src/modules/core/filter_volume.c b/src/modules/core/filter_volume.c index 719e3ed..04fb936 100644 --- a/src/modules/core/filter_volume.c +++ b/src/modules/core/filter_volume.c @@ -29,9 +29,9 @@ #include #define MAX_CHANNELS 6 -#define SMOOTH_BUFFER_SIZE 50 +#define EPSILON 0.00001 -/* This utilities and limiter function comes from the normalize utility: +/* The normalise functions come from the normalize utility: Copyright (C) 1999--2002 Chris Vaill */ #define samp_width 16 @@ -68,15 +68,16 @@ int strncaseeq(const char *s1, const char *s2, size_t n) */ static inline double limiter( double x, double lmtr_lvl ) { - double xp; + double xp = x; if (x < -lmtr_lvl) xp = tanh((x + lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) - lmtr_lvl; - else if (x <= lmtr_lvl) - xp = x; - else + else if (x > lmtr_lvl) xp = tanh((x - lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) + lmtr_lvl; +// if ( x != xp ) +// fprintf( stderr, "filter_volume: sample %f limited %f\n", x, xp ); + return xp; } @@ -158,6 +159,8 @@ double signal_max_power( int16_t *buffer, int channels, int samples, int16_t *pe return sqrt( maxpow ); } +/* ------ End normalize functions --------------------------------------- */ + /** Get the audio. */ @@ -165,62 +168,111 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format { // Get the properties of the a frame mlt_properties properties = mlt_frame_properties( frame ); - double gain = mlt_properties_get_double( properties, "gain" ); - int use_limiter = mlt_properties_get_int( properties, "volume.use_limiter" ); - double limiter_level = mlt_properties_get_double( properties, "volume.limiter_level" ); + double gain = mlt_properties_get_double( properties, "volume.gain" ); + double max_gain = mlt_properties_get_double( properties, "volume.max_gain" ); + double limiter_level = 0.5; /* -6 dBFS */ int normalise = mlt_properties_get_int( properties, "volume.normalise" ); double amplitude = mlt_properties_get_double( properties, "volume.amplitude" ); - int i; + int i, j; double sample; int16_t peak; + // Get the filter from the frame + mlt_filter this = mlt_properties_get_data( properties, "filter_volume", NULL ); + + // Get the properties from the filter + mlt_properties filter_props = mlt_filter_properties( this ); + + if ( mlt_properties_get( properties, "volume.limiter" ) != NULL ) + limiter_level = mlt_properties_get_double( properties, "volume.limiter" ); + // Restore the original get_audio frame->get_audio = mlt_properties_get_data( properties, "volume.get_audio", NULL ); // Get the producer's audio mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples ); +// fprintf( stderr, "filter_volume: frequency %d\n", *frequency ); // Determine numeric limits int bytes_per_samp = (samp_width - 1) / 8 + 1; int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1; int samplemin = -samplemax - 1; -#if 0 - if ( gain > 1.0 && use_limiter != 0 ) - fprintf(stderr, "filter_volume: limiting samples greater than %f\n", limiter_level ); -#endif - if ( normalise ) { - double *smooth_buffer = mlt_properties_get_data( properties, "volume.smooth_buffer", NULL ); - int *smooth_index = mlt_properties_get_data( properties, "volume.smooth_index", NULL ); + int window = mlt_properties_get_int( filter_props, "window" ); + double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL ); - // Compute the signal power and put into smoothing buffer - smooth_buffer[ *smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak ); - *smooth_index = ( *smooth_index + 1 ) % SMOOTH_BUFFER_SIZE; - - // Smooth the data and compute the gain - gain *= amplitude / get_smoothed_data( smooth_buffer, SMOOTH_BUFFER_SIZE ); + if ( window > 0 && smooth_buffer != NULL ) + { + int smooth_index = mlt_properties_get_int( filter_props, "_smooth_index" ); + + // Compute the signal power and put into smoothing buffer + smooth_buffer[ smooth_index ] = signal_max_power( *buffer, *channels, *samples, &peak ); +// fprintf( stderr, "filter_volume: raw power %f ", smooth_buffer[ smooth_index ] ); + if ( smooth_buffer[ smooth_index ] > EPSILON ) + { + mlt_properties_set_int( filter_props, "_smooth_index", ( smooth_index + 1 ) % window ); + + // Smooth the data and compute the gain +// fprintf( stderr, "smoothed %f over %d frames\n", get_smoothed_data( smooth_buffer, window ), window ); + gain *= amplitude / get_smoothed_data( smooth_buffer, window ); + } + } + else + { + gain *= amplitude / signal_max_power( *buffer, *channels, *samples, &peak ); + } } +// if ( gain > 1.0 && normalise ) +// fprintf(stderr, "filter_volume: limiter level %f gain %f\n", limiter_level, gain ); + + if ( max_gain > 0 && gain > max_gain ) + gain = max_gain; + + // Initialise filter's previous gain value to prevent an inadvertant jump from 0 + if ( mlt_properties_get( filter_props, "previous_gain" ) == NULL ) + mlt_properties_set_double( filter_props, "previous_gain", gain ); + + // Start the gain out at the previous + double previous_gain = mlt_properties_get_double( filter_props, "previous_gain" ); + + // Determine ramp increment + double gain_step = ( gain - previous_gain ) / *samples; +// fprintf( stderr, "filter_volume: previous gain %f current gain %f step %f\n", previous_gain, gain, gain_step ); + + // Save the current gain for the next iteration + mlt_properties_set_double( filter_props, "previous_gain", gain ); + + // Ramp from the previous gain to the current + gain = previous_gain; + + int16_t *p = *buffer; + // Apply the gain - for ( i = 0; i < ( *channels * *samples ); i++ ) + for ( i = 0; i < *samples; i++ ) { - sample = (*buffer)[i] * gain; - (*buffer)[i] = ROUND( sample ); - - if ( gain > 1.0 ) + for ( j = 0; j < *channels; j++ ) { - /* use limiter function instead of clipping */ - if ( use_limiter != 0 ) - (*buffer)[i] = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) ); + sample = *p * gain; + *p = ROUND( sample ); + + if ( gain > 1.0 ) + { + /* use limiter function instead of clipping */ + if ( normalise ) + *p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) ); - /* perform clipping */ - else if ( sample > samplemax ) - (*buffer)[i] = samplemax; - else if ( sample < samplemin ) - (*buffer)[i] = samplemin; + /* perform clipping */ + else if ( sample > samplemax ) + *p = samplemax; + else if ( sample < samplemin ) + *p = samplemin; + } + p++; } + gain += gain_step; } return 0; @@ -234,18 +286,75 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame ) mlt_properties properties = mlt_frame_properties( frame ); mlt_properties filter_props = mlt_filter_properties( this ); - // Propogate the volume/gain property + // Parse the gain property if ( mlt_properties_get( properties, "gain" ) == NULL ) { - double gain = 1.0; // none - if ( mlt_properties_get( filter_props, "volume" ) != NULL ) - gain = mlt_properties_get_double( filter_props, "volume" ); + double gain = 1.0; // no adjustment + if ( mlt_properties_get( filter_props, "gain" ) != NULL ) - gain = mlt_properties_get_double( filter_props, "gain" ); - mlt_properties_set_double( properties, "gain", gain ); + { + char *p = mlt_properties_get( filter_props, "gain" ); + + if ( strncaseeq( p, "normalise", 9 ) ) + mlt_properties_set( filter_props, "normalise", "" ); + else + { + if ( strcmp( p, "" ) != 0 ) + gain = fabs( strtod( p, &p) ); + + while ( isspace( *p ) ) + p++; + + /* check if "dB" is given after number */ + if ( strncaseeq( p, "db", 2 ) ) + gain = DBFSTOAMP( gain ); + + // If there is an end adjust gain to the range + if ( mlt_properties_get( filter_props, "end" ) != NULL ) + { + // Determine the time position of this frame in the transition duration + mlt_position in = mlt_filter_get_in( this ); + mlt_position out = mlt_filter_get_out( this ); + mlt_position time = mlt_frame_get_position( frame ); + double position = ( double )( time - in ) / ( double )( out - in + 1 ); + + double end = -1; + char *p = mlt_properties_get( filter_props, "end" ); + if ( strcmp( p, "" ) != 0 ) + end = fabs( strtod( p, &p) ); + + while ( isspace( *p ) ) + p++; + + /* check if "dB" is given after number */ + if ( strncaseeq( p, "db", 2 ) ) + end = DBFSTOAMP( gain ); + + if ( end != -1 ) + gain += ( end - gain ) * position; + } + } + } + mlt_properties_set_double( properties, "volume.gain", gain ); } - // Parse and propogate the limiter property + // Parse the maximum gain property + if ( mlt_properties_get( filter_props, "max_gain" ) != NULL ) + { + char *p = mlt_properties_get( filter_props, "max_gain" ); + double gain = fabs( strtod( p, &p) ); // 0 = no max + + while ( isspace( *p ) ) + p++; + + /* check if "dB" is given after number */ + if ( strncaseeq( p, "db", 2 ) ) + gain = DBFSTOAMP( gain ); + + mlt_properties_set_double( properties, "volume.max_gain", gain ); + } + + // Parse the limiter property if ( mlt_properties_get( filter_props, "limiter" ) != NULL ) { char *p = mlt_properties_get( filter_props, "limiter" ); @@ -253,10 +362,10 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame ) if ( strcmp( p, "" ) != 0 ) level = strtod( p, &p); - /* check if "dB" is given after number */ while ( isspace( *p ) ) p++; + /* check if "dB" is given after number */ if ( strncaseeq( p, "db", 2 ) ) { if ( level > 0 ) @@ -268,11 +377,10 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame ) if ( level < 0 ) level = -level; } - mlt_properties_set_int( properties, "volume.use_limiter", 1 ); - mlt_properties_set_double( properties, "volume.limiter_level", level ); + mlt_properties_set_double( properties, "volume.limiter", level ); } - // Parse and propogate the normalise property + // Parse the normalise property if ( mlt_properties_get( filter_props, "normalise" ) != NULL ) { char *p = mlt_properties_get( filter_props, "normalise" ); @@ -280,10 +388,10 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame ) if ( strcmp( p, "" ) != 0 ) amplitude = strtod( p, &p); - /* check if "dB" is given after number */ while ( isspace( *p ) ) p++; + /* check if "dB" is given after number */ if ( strncaseeq( p, "db", 2 ) ) { if ( amplitude > 0 ) @@ -297,16 +405,36 @@ static mlt_frame filter_process( mlt_filter this, mlt_frame frame ) if ( amplitude > 1.0 ) amplitude = 1.0; } + + // If there is an end adjust gain to the range + if ( mlt_properties_get( filter_props, "end" ) != NULL ) + { + // Determine the time position of this frame in the transition duration + mlt_position in = mlt_filter_get_in( this ); + mlt_position out = mlt_filter_get_out( this ); + mlt_position time = mlt_frame_get_position( frame ); + double position = ( double )( time - in ) / ( double )( out - in + 1 ); + amplitude *= position; + } mlt_properties_set_int( properties, "volume.normalise", 1 ); mlt_properties_set_double( properties, "volume.amplitude", amplitude ); } - // Propogate the smoothing buffer properties - mlt_properties_set_data( properties, "volume.smooth_buffer", - mlt_properties_get_data( filter_props, "smooth_buffer", NULL ), 0, NULL, NULL ); - mlt_properties_set_data( properties, "volume.smooth_index", - mlt_properties_get_data( filter_props, "smooth_index", NULL ), 0, NULL, NULL ); - + // Parse the window property and allocate smoothing buffer if needed + int window = mlt_properties_get_int( filter_props, "window" ); + if ( mlt_properties_get( filter_props, "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( filter_props, "smooth_buffer", smooth_buffer, 0, free, NULL ); + } + + // Put a filter reference onto the frame + mlt_properties_set_data( properties, "filter_volume", this, 0, NULL, NULL ); + // Backup the original get_audio (it's still needed) mlt_properties_set_data( properties, "volume.get_audio", frame->get_audio, 0, NULL, NULL ); @@ -324,19 +452,13 @@ mlt_filter filter_volume_init( char *arg ) mlt_filter this = calloc( sizeof( struct mlt_filter_s ), 1 ); if ( this != NULL && mlt_filter_init( this, NULL ) == 0 ) { + mlt_properties properties = mlt_filter_properties( this ); this->process = filter_process; if ( arg != NULL ) - mlt_properties_set_double( mlt_filter_properties( this ), "volume", atof( arg ) ); + mlt_properties_set( properties, "gain", arg ); - // Create a smoothing buffer for the calculated "max power" of frame of audio used in normalisation - double *smooth_buffer = (double*) calloc( SMOOTH_BUFFER_SIZE, sizeof( double ) ); - int i; - for ( i = 0; i < SMOOTH_BUFFER_SIZE; i++ ) - smooth_buffer[ i ] = -1.0; - mlt_properties_set_data( mlt_filter_properties( this ), "smooth_buffer", smooth_buffer, 0, free, NULL ); - int *smooth_index = calloc( 1, sizeof( int ) ); - mlt_properties_set_data( mlt_filter_properties( this ), "smooth_index", smooth_index, 0, free, NULL ); + mlt_properties_set_int( properties, "window", 75 ); + mlt_properties_set( properties, "max_gain", "20dB" ); } return this; } -