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/* the Music Player Daemon (MPD)
* (c)2003-2006 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: http://www.musicpd.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 <math.h>
#include <limits.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)
#define MUL_MIN 0.1
#define MUL_MAX 5.0
#define NSAMPLES 128
#define MIN_SAMPLE_SIZE 32000
#define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
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];
/* 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 */
}
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