void print_power (char *filename)
{
AFfilehandle file;
double *sums, *frames;
int channelCount, windowSize, frameCount;
int i, c;
struct smooth *powsmooth;
int winStart, winEnd;
int lastWindow = FALSE;
double pow, maxpow;
double level, peak, minSample = 1, maxSample = -1;
file = afOpenFile(filename, "r", NULL);
if (file == AF_NULL_FILEHANDLE)
{
fprintf(stderr, "Could not open file %s.\n", filename);
return;
}
channelCount = afGetChannels(file, AF_DEFAULT_TRACK);
windowSize = afGetRate(file, AF_DEFAULT_TRACK) / 100;
frameCount = afGetFrameCount(file, AF_DEFAULT_TRACK);
sums = calloc(channelCount, sizeof (double));
for (c=0; c<channelCount; c++)
sums[c] = 0;
frames = calloc(channelCount * windowSize, sizeof (double));
afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_DOUBLE,
sizeof (double));
powsmooth = calloc(channelCount, sizeof (struct smooth));
for (c=0; c<channelCount; c++)
{
/* Use a 100-element (1 second) window. */
powsmooth[c].length = 100;
powsmooth[c].buf = calloc(powsmooth[c].length, sizeof (double));
powsmooth[c].start = 0;
powsmooth[c].n = 0;
}
winStart = 0;
winEnd = 0;
lastWindow = FALSE;
maxpow = 0;
do
{
winEnd = winStart + windowSize;
if (winEnd >= frameCount)
{
winEnd = frameCount;
lastWindow = TRUE;
}
afReadFrames(file, AF_DEFAULT_TRACK, frames, windowSize);
for (c=0; c<channelCount; c++)
{
sums[c] = 0;
for (i=0; i < winEnd - winStart; i++)
{
double sample;
sample = frames[i*channelCount + c];
sums[c] += sample*sample;
if (sample > maxSample)
maxSample = sample;
if (sample < minSample)
minSample = sample;
}
}
/* Compute power for each channel. */
for (c=0; c<channelCount; c++)
{
double pow;
int end;
pow = sums[c] / (winEnd - winStart);
end = (powsmooth[c].start + powsmooth[c].n) %
powsmooth[c].length;
powsmooth[c].buf[end] = pow;
if (powsmooth[c].n == powsmooth[c].length)
{
powsmooth[c].start = (powsmooth[c].start + 1) % powsmooth[c].length;
pow = get_smoothed_data(&powsmooth[c]);
if (pow > maxpow)
maxpow = pow;
}
else
{
powsmooth[c].n++;
}
}
winStart += windowSize;
} while (!lastWindow);
for (c = 0; c < channelCount; c++)
{
pow = get_smoothed_data(&powsmooth[c]);
if (pow > maxpow)
maxpow = pow;
}
free(sums);
free(frames);
for (c=0; c<channelCount; c++)
free(powsmooth[c].buf);
free(powsmooth);
level = sqrt(maxpow);
afCloseFile(file);
printf("file: %s\n", filename);
printf("level (dB): %f\n", 20 * log10(level));
printf("peak-: %f\n", minSample);
printf("peak+: %f\n", maxSample);
peak = abs(minSample);
if (peak < abs(maxSample))
peak = abs(maxSample);
printf("peak (dB): %f\n", 20 * log10(peak));
}
static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
// Get the filter from the frame
mlt_filter this = mlt_frame_pop_audio( frame );
// Get the properties from the filter
mlt_properties filter_props = MLT_FILTER_PROPERTIES( this );
// Get the frame's filter instance properties
mlt_properties instance_props = mlt_frame_unique_properties( frame, MLT_FILTER_SERVICE( this ) );
// Get the parameters
double gain = mlt_properties_get_double( instance_props, "gain" );
double max_gain = mlt_properties_get_double( instance_props, "max_gain" );
double limiter_level = 0.5; /* -6 dBFS */
int normalise = mlt_properties_get_int( instance_props, "normalise" );
double amplitude = mlt_properties_get_double( instance_props, "amplitude" );
int i, j;
double sample;
int16_t peak;
if ( mlt_properties_get( instance_props, "limiter" ) != NULL )
limiter_level = mlt_properties_get_double( instance_props, "limiter" );
// Get the producer's audio
*format = mlt_audio_s16;
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;
mlt_service_lock( MLT_FILTER_SERVICE( this ) );
if ( normalise )
{
int window = mlt_properties_get_int( filter_props, "window" );
double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
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( (int16_t*) *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
mlt_position last_position = mlt_properties_get_position( filter_props, "_last_position" );
mlt_position current_position = mlt_frame_get_position( frame );
if ( mlt_properties_get( filter_props, "_previous_gain" ) == NULL
|| current_position != last_position + 1 )
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 );
mlt_properties_set_position( filter_props, "_last_position", current_position );
mlt_service_unlock( MLT_FILTER_SERVICE( this ) );
// Ramp from the previous gain to the current
gain = previous_gain;
int16_t *p = (int16_t*) *buffer;
// Apply the gain
for ( i = 0; i < *samples; i++ )
{
for ( j = 0; j < *channels; j++ )
{
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 )
*p = samplemax;
else if ( sample < samplemin )
*p = samplemin;
}
p++;
}
gain += gain_step;
}
return 0;
}
static int filter_get_audio( mlt_frame frame, void **buffer, mlt_audio_format *format, int *frequency, int *channels, int *samples )
{
// Get the filter from the frame
mlt_filter filter = mlt_frame_pop_audio( frame );
// Get the properties from the filter
mlt_properties filter_props = MLT_FILTER_PROPERTIES( filter );
// Get the frame's filter instance properties
mlt_properties instance_props = mlt_frame_unique_properties( frame, MLT_FILTER_SERVICE( filter ) );
// Get the parameters
double gain = mlt_properties_get_double( instance_props, "gain" );
double max_gain = mlt_properties_get_double( instance_props, "max_gain" );
double limiter_level = 0.5; /* -6 dBFS */
int normalise = mlt_properties_get_int( instance_props, "normalise" );
double amplitude = mlt_properties_get_double( instance_props, "amplitude" );
int i, j;
double sample;
int16_t peak;
// Use animated value for gain if "level" property is set
char* level_property = mlt_properties_get( filter_props, "level" );
if ( level_property != NULL )
{
mlt_position position = mlt_filter_get_position( filter, frame );
mlt_position length = mlt_filter_get_length2( filter, frame );
gain = mlt_properties_anim_get_double( filter_props, "level", position, length );
gain = DBFSTOAMP( gain );
}
if ( mlt_properties_get( instance_props, "limiter" ) != NULL )
limiter_level = mlt_properties_get_double( instance_props, "limiter" );
// Get the producer's audio
*format = normalise? mlt_audio_s16 : mlt_audio_f32le;
mlt_frame_get_audio( frame, buffer, format, frequency, channels, samples );
mlt_service_lock( MLT_FILTER_SERVICE( filter ) );
if ( normalise )
{
int window = mlt_properties_get_int( filter_props, "window" );
double *smooth_buffer = mlt_properties_get_data( filter_props, "smooth_buffer", NULL );
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 );
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
gain *= amplitude / get_smoothed_data( smooth_buffer, window );
}
}
else
{
gain *= amplitude / signal_max_power( *buffer, *channels, *samples, &peak );
}
}
if ( max_gain > 0 && gain > max_gain )
gain = max_gain;
// Initialise filter's previous gain value to prevent an inadvertant jump from 0
mlt_position last_position = mlt_properties_get_position( filter_props, "_last_position" );
mlt_position current_position = mlt_frame_get_position( frame );
if ( mlt_properties_get( filter_props, "_previous_gain" ) == NULL
|| current_position != last_position + 1 )
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;
// Save the current gain for the next iteration
mlt_properties_set_double( filter_props, "_previous_gain", gain );
mlt_properties_set_position( filter_props, "_last_position", current_position );
mlt_service_unlock( MLT_FILTER_SERVICE( filter ) );
// Ramp from the previous gain to the current
gain = previous_gain;
// Apply the gain
if ( normalise )
{
int16_t *p = *buffer;
// Determine numeric limits
int bytes_per_samp = (samp_width - 1) / 8 + 1;
int samplemax = (1 << (bytes_per_samp * 8 - 1)) - 1;
for ( i = 0; i < *samples; i++, gain += gain_step ) {
for ( j = 0; j < *channels; j++ ) {
sample = *p * gain;
*p = ROUND( sample );
if ( gain > 1.0 && normalise ) {
/* use limiter function instead of clipping */
*p = ROUND( samplemax * limiter( sample / (double) samplemax, limiter_level ) );
}
p++;
}
}
}
else
{
float *p = *buffer;
for ( i = 0; i < *samples; i++, gain += gain_step ) {
for ( j = 0; j < *channels; j++, p++ ) {
p[0] *= gain;
}
}
}
return 0;
}
