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authorJ. Alexander Treuman <jat@spatialrift.net>2006-07-22 01:14:29 +0000
committerJ. Alexander Treuman <jat@spatialrift.net>2006-07-22 01:14:29 +0000
commitdb76c2a4444ef8108239f9faba2394890bae6c25 (patch)
treedcaea49b9a5d94801edf5dce12c0c621aa6e0d02 /src/normalize.c
parentcf90f8194fdcf4b7d3234453da00f8f3b88e54c9 (diff)
downloadmpd-db76c2a4444ef8108239f9faba2394890bae6c25.tar.gz
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Maybe we should actually commit the normalization code
git-svn-id: https://svn.musicpd.org/mpd/trunk@4425 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src/normalize.c')
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diff --git a/src/normalize.c b/src/normalize.c
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+++ b/src/normalize.c
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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 */
+}