101 lines
3.0 KiB
C
101 lines
3.0 KiB
C
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/* the Music Player Daemon (MPD)
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* (c)2003-2006 by Warren Dukes (warren.dukes@gmail.com)
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* This project's homepage is: http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include <math.h>
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#include <limits.h>
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#include "conf.h"
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#include "normalize.h"
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#include "playlist.h"
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/* silence level, apparently this is Wrong (tm) */
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#define SILENCE_LEVEL (SHRT_MAX * 0.01)
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/* not sure what this is :) */
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#define MID (SHRT_MAX * 0.25)
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#define MUL_MIN 0.1
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#define MUL_MAX 5.0
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#define NSAMPLES 128
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#define MIN_SAMPLE_SIZE 32000
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#define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
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void normalizeData(char *buffer, int bufferSize, AudioFormat *format)
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{
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static float multiplier = 1.0;
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static int current_id = 0;
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float average = 0.0;
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static int old_song = 0;
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int new_song = 0;
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int total_length = 0;
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int temp = 0;
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int i = 0;
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float root_mean_square = 0.0; /* the rms of the data */
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mpd_sint16 *data = (mpd_sint16 *) buffer; /* the audio data */
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int length = bufferSize / 2; /* the number of samples */
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static struct {
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float avg; /* average sample 'level' */
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int len; /* sample size (used to weigh sample) */
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} mem[NSAMPLES];
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/* operate only on 16 bit, 2 channel audio */
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if (format->bits != 16 && format->channels != 2) return;
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/* calculate the root mean square of the data */
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for (i = 0; i < length; i++)
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root_mean_square += (float)(data[i] * data[i]);
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root_mean_square = sqrt(root_mean_square / (float)length);
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/* reset the multiplier if the song has changed */
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if (old_song != (new_song = getPlaylistCurrentSong())) {
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old_song = new_song;
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/* re-zero 'mem' */
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for (i = 0; i < NSAMPLES; i++) {
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mem[i].avg = 0.0;
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mem[i].len = 0;
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}
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current_id = 0;
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}
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/* and now do magic tricks */
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for (i = 0; i < NSAMPLES; i++) {
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average += mem[i].avg * (float)mem[i].len;
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total_length += mem[i].len;
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}
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if (total_length > MIN_SAMPLE_SIZE) {
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average /= (float) total_length;
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if (average >= SILENCE_LEVEL) {
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multiplier = MID / average;
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/* clamp multiplier */
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multiplier = clamp(multiplier, MUL_MIN, MUL_MAX);
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}
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}
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/* scale and clamp the samples */
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for (i = 0; i < length; i++) {
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temp = data[i] * multiplier;
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data[i] = clamp(temp, SHRT_MIN, SHRT_MAX);
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}
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mem[current_id].len = bufferSize / 2;
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mem[current_id].avg = multiplier * root_mean_square;
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current_id = (current_id + 1) % NSAMPLES; /* increment current_id */
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}
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