From dfcac1bf1428068f68b700d872452d9aa80a8a1d Mon Sep 17 00:00:00 2001 From: ddennedy Date: Tue, 27 Jan 2004 22:46:16 +0000 Subject: [PATCH] westley bugfixes and audio normalisation git-svn-id: https://mlt.svn.sourceforge.net/svnroot/mlt/trunk/mlt@98 d19143bc-622f-0410-bfdd-b5b2a6649095 --- src/modules/core/filter_volume.c | 272 +++++++++++++++++++++++++++++++- src/modules/westley/consumer_westley.c | 2 +- src/modules/westley/producer_westley.c | 19 ++- 3 files changed, 280 insertions(+), 13 deletions(-) diff --git a/src/modules/core/filter_volume.c b/src/modules/core/filter_volume.c index c0ad4dd..719e3ed 100644 --- a/src/modules/core/filter_volume.c +++ b/src/modules/core/filter_volume.c @@ -24,6 +24,139 @@ #include #include +#include +#include +#include + +#define MAX_CHANNELS 6 +#define SMOOTH_BUFFER_SIZE 50 + +/* This utilities and limiter function comes from the normalize utility: + Copyright (C) 1999--2002 Chris Vaill */ + +#define samp_width 16 + +#ifndef ROUND +# define ROUND(x) floor((x) + 0.5) +#endif + +#define DBFSTOAMP(x) pow(10,(x)/20.0) + +/** Return nonzero if the two strings are equal, ignoring case, up to + the first n characters. +*/ +int strncaseeq(const char *s1, const char *s2, size_t n) +{ + for ( ; n > 0; n--) + { + if (tolower(*s1++) != tolower(*s2++)) + return 0; + } + return 1; +} + +/** Limiter function. + + / tanh((x + lev) / (1-lev)) * (1-lev) - lev (for x < -lev) + | + x' = | x (for |x| <= lev) + | + \ tanh((x - lev) / (1-lev)) * (1-lev) + lev (for x > lev) + + With limiter level = 0, this is equivalent to a tanh() function; + with limiter level = 1, this is equivalent to clipping. +*/ +static inline double limiter( double x, double lmtr_lvl ) +{ + double xp; + + 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 + xp = tanh((x - lmtr_lvl) / (1-lmtr_lvl)) * (1-lmtr_lvl) + lmtr_lvl; + + return xp; +} + + +/** Takes a full smoothing window, and returns the value of the center + element, smoothed. + + Currently, just does a mean filter, but we could do a median or + gaussian filter here instead. +*/ +static inline double get_smoothed_data( double *buf, int count ) +{ + int i, j; + double smoothed = 0; + + for ( i = 0, j = 0; i < count; i++ ) + { + if ( buf[ i ] != -1.0 ) + { + smoothed += buf[ i ]; + j++; + } + } + smoothed /= j; +// fprintf( stderr, "smoothed over %d values, result %f\n", j, smoothed ); + + return smoothed; +} + +/** Get the max power level (using RMS) and peak level of the audio segment. + */ +double signal_max_power( int16_t *buffer, int channels, int samples, int16_t *peak ) +{ + // Determine numeric limits + int bytes_per_samp = (samp_width - 1) / 8 + 1; + int16_t max = (1 << (bytes_per_samp * 8 - 1)) - 1; + int16_t min = -max - 1; + + double *sums = (double *) calloc( channels, sizeof(double) ); + int c, i; + int16_t sample; + double pow, maxpow = 0; + + /* initialize peaks to effectively -inf and +inf */ + int16_t max_sample = min; + int16_t min_sample = max; + + for ( i = 0; i < samples; i++ ) + { + for ( c = 0; c < channels; c++ ) + { + sample = *buffer++; + sums[ c ] += (double) sample * (double) sample; + + /* track peak */ + if ( sample > max_sample ) + max_sample = sample; + else if ( sample < min_sample ) + min_sample = sample; + } + } + for ( c = 0; c < channels; c++ ) + { + pow = sums[ c ] / (double) samples; + if ( pow > maxpow ) + maxpow = pow; + } + + free( sums ); + + /* scale the pow value to be in the range 0.0 -- 1.0 */ + maxpow /= ( (double) min * (double) min); + + if ( -min_sample > max_sample ) + *peak = min_sample / (double) min; + else + *peak = max_sample / (double) max; + + return sqrt( maxpow ); +} /** Get the audio. */ @@ -32,7 +165,14 @@ 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 volume = mlt_properties_get_double( properties, "volume" ); + 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" ); + int normalise = mlt_properties_get_int( properties, "volume.normalise" ); + double amplitude = mlt_properties_get_double( properties, "volume.amplitude" ); + int i; + double sample; + int16_t peak; // Restore the original get_audio frame->get_audio = mlt_properties_get_data( properties, "volume.get_audio", NULL ); @@ -40,10 +180,48 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format // Get the producer's audio mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples ); - // Apply the volume - int i; + // 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 ); + + // 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 ); + } + + // Apply the gain for ( i = 0; i < ( *channels * *samples ); i++ ) - (*buffer)[i] *= volume; + { + sample = (*buffer)[i] * gain; + (*buffer)[i] = ROUND( sample ); + + if ( gain > 1.0 ) + { + /* use limiter function instead of clipping */ + if ( use_limiter != 0 ) + (*buffer)[i] = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) ); + + /* perform clipping */ + else if ( sample > samplemax ) + (*buffer)[i] = samplemax; + else if ( sample < samplemin ) + (*buffer)[i] = samplemin; + } + } return 0; } @@ -54,12 +232,81 @@ static int filter_get_audio( mlt_frame frame, int16_t **buffer, mlt_audio_format 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 level property - if ( mlt_properties_get( mlt_filter_properties( this ), "volume" ) != NULL ) - mlt_properties_set_double( properties, "volume", - mlt_properties_get_double( mlt_filter_properties( this ), "volume" ) ); + // Propogate the volume/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" ); + if ( mlt_properties_get( filter_props, "gain" ) != NULL ) + gain = mlt_properties_get_double( filter_props, "gain" ); + mlt_properties_set_double( properties, "gain", gain ); + } + // Parse and propogate the limiter property + if ( mlt_properties_get( filter_props, "limiter" ) != NULL ) + { + char *p = mlt_properties_get( filter_props, "limiter" ); + double level = 0.5; /* -6dBFS */ + if ( strcmp( p, "" ) != 0 ) + level = strtod( p, &p); + + /* check if "dB" is given after number */ + while ( isspace( *p ) ) + p++; + + if ( strncaseeq( p, "db", 2 ) ) + { + if ( level > 0 ) + level = -level; + level = DBFSTOAMP( level ); + } + else + { + if ( level < 0 ) + level = -level; + } + mlt_properties_set_int( properties, "volume.use_limiter", 1 ); + mlt_properties_set_double( properties, "volume.limiter_level", level ); + } + + // Parse and propogate the normalise property + if ( mlt_properties_get( filter_props, "normalise" ) != NULL ) + { + char *p = mlt_properties_get( filter_props, "normalise" ); + double amplitude = 0.2511886431509580; /* -12dBFS */ + if ( strcmp( p, "" ) != 0 ) + amplitude = strtod( p, &p); + + /* check if "dB" is given after number */ + while ( isspace( *p ) ) + p++; + + if ( strncaseeq( p, "db", 2 ) ) + { + if ( amplitude > 0 ) + amplitude = -amplitude; + amplitude = DBFSTOAMP( amplitude ); + } + else + { + if ( amplitude < 0 ) + amplitude = -amplitude; + if ( amplitude > 1.0 ) + amplitude = 1.0; + } + 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 ); + // Backup the original get_audio (it's still needed) mlt_properties_set_data( properties, "volume.get_audio", frame->get_audio, 0, NULL, NULL ); @@ -80,6 +327,15 @@ mlt_filter filter_volume_init( char *arg ) this->process = filter_process; if ( arg != NULL ) mlt_properties_set_double( mlt_filter_properties( this ), "volume", atof( 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 ); } return this; } diff --git a/src/modules/westley/consumer_westley.c b/src/modules/westley/consumer_westley.c index d196ffb..75932a5 100644 --- a/src/modules/westley/consumer_westley.c +++ b/src/modules/westley/consumer_westley.c @@ -355,7 +355,7 @@ static int consumer_start( mlt_consumer this ) free( context ); if ( mlt_properties_get( mlt_consumer_properties( this ), "resource" ) == NULL ) - xmlDocFormatDump( stderr, doc, 1 ); + xmlDocFormatDump( stdout, doc, 1 ); else xmlSaveFormatFile( mlt_properties_get( mlt_consumer_properties( this ), "resource" ), doc, 1 ); } diff --git a/src/modules/westley/producer_westley.c b/src/modules/westley/producer_westley.c index ff5ba41..2bd3204 100644 --- a/src/modules/westley/producer_westley.c +++ b/src/modules/westley/producer_westley.c @@ -342,6 +342,18 @@ static void on_end_entry( deserialise_context context, const xmlChar *name ) context_push_service( context, service ); } +static void on_end_tractor( deserialise_context context, const xmlChar *name ) +{ + // Discard the last producer + mlt_producer multitrack = MLT_PRODUCER( context_pop_service( context ) ); + + // Inherit the producer's properties + mlt_properties properties = mlt_producer_properties( multitrack ); + mlt_properties_set_position( properties, "length", mlt_producer_get_out( multitrack ) + 1 ); + mlt_producer_set_in_and_out( multitrack, 0, mlt_producer_get_out( multitrack ) ); + mlt_properties_set_double( properties, "fps", mlt_producer_get_fps( multitrack ) ); +} + static void on_start_element( void *ctx, const xmlChar *name, const xmlChar **atts) { deserialise_context context = ( deserialise_context ) ctx; @@ -377,10 +389,7 @@ static void on_end_element( void *ctx, const xmlChar *name ) else if ( strcmp( name, "entry" ) == 0 ) on_end_entry( context, name ); else if ( strcmp( name, "tractor" ) == 0 ) - { - // Discard the last producer - context_pop_service( context ); - } + on_end_tractor( context, name ); } @@ -408,6 +417,8 @@ mlt_producer producer_westley_init( char *filename ) mlt_properties_set_data( mlt_service_properties( service ), "__destructors__", context->destructors, 0, (mlt_destructor) mlt_properties_close, NULL ); free( context ); + mlt_properties_set( mlt_service_properties( service ), "resource", filename ); + return MLT_PRODUCER( service ); } -- 1.7.4.4