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Use AudioCompress for volume normalization

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git-svn-id: https://svn.musicpd.org/mpd/trunk@4474 09075e82-0dd4-0310-85a5-a0d7c8717e4f
This commit is contained in:
J. Alexander Treuman
2006-07-27 00:50:59 +00:00
parent bf5336cebd
commit 22b16884a2
10 changed files with 463 additions and 100 deletions
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91
src/normalize.c
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@@ -16,85 +16,32 @@
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <math.h>
#include <limits.h>
#include "compress.h"
#include "conf.h"
#include "normalize.h"
#include "playlist.h"
/* silence level, apparently this is Wrong (tm) */
#define SILENCE_LEVEL (SHRT_MAX * 0.01)
/* not sure what this is :) */
#define MID (SHRT_MAX * 0.25)
#include <stdlib.h>
#define MUL_MIN 0.1
#define MUL_MAX 5.0
#define NSAMPLES 128
#define MIN_SAMPLE_SIZE 32000
int normalizationEnabled;
#define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
void initNormalization()
{
normalizationEnabled = getBoolConfigParam(CONF_VOLUME_NORMALIZATION);
if (normalizationEnabled == -1) normalizationEnabled = 0;
else if (normalizationEnabled < 0) exit(EXIT_FAILURE);
if (normalizationEnabled)
CompressCfg(0, ANTICLIP, TARGET, GAINMAX, GAINSMOOTH, BUCKETS);
}
void finishNormalization()
{
if (normalizationEnabled) CompressFree();
}
void normalizeData(char *buffer, int bufferSize, AudioFormat *format)
{
static float multiplier = 1.0;
static int current_id = 0;
float average = 0.0;
static int old_song = 0;
int new_song = 0;
int total_length = 0;
int temp = 0;
int i = 0;
float root_mean_square = 0.0; /* the rms of the data */
mpd_sint16 *data = (mpd_sint16 *) buffer; /* the audio data */
int length = bufferSize / 2; /* the number of samples */
static struct {
float avg; /* average sample 'level' */
int len; /* sample size (used to weigh sample) */
} mem[NSAMPLES];
if ((format->bits != 16) || (format->channels != 2)) return;
/* operate only on 16 bit, 2 channel audio */
if (format->bits != 16 && format->channels != 2) return;
/* calculate the root mean square of the data */
for (i = 0; i < length; i++)
root_mean_square += (float)(data[i] * data[i]);
root_mean_square = sqrt(root_mean_square / (float)length);
/* reset the multiplier if the song has changed */
if (old_song != (new_song = getPlaylistCurrentSong())) {
old_song = new_song;
/* re-zero 'mem' */
for (i = 0; i < NSAMPLES; i++) {
mem[i].avg = 0.0;
mem[i].len = 0;
}
current_id = 0;
}
/* and now do magic tricks */
for (i = 0; i < NSAMPLES; i++) {
average += mem[i].avg * (float)mem[i].len;
total_length += mem[i].len;
}
if (total_length > MIN_SAMPLE_SIZE) {
average /= (float) total_length;
if (average >= SILENCE_LEVEL) {
multiplier = MID / average;
/* clamp multiplier */
multiplier = clamp(multiplier, MUL_MIN, MUL_MAX);
}
}
/* scale and clamp the samples */
for (i = 0; i < length; i++) {
temp = data[i] * multiplier;
data[i] = clamp(temp, SHRT_MIN, SHRT_MAX);
}
mem[current_id].len = bufferSize / 2;
mem[current_id].avg = multiplier * root_mean_square;
current_id = (current_id + 1) % NSAMPLES; /* increment current_id */
CompressDo(buffer, bufferSize);
}