Maybe we should actually commit the normalization code

git-svn-id: https://svn.musicpd.org/mpd/trunk@4425 09075e82-0dd4-0310-85a5-a0d7c8717e4f

1 files changed, 100 insertions, 0 deletions

diff --git a/src/normalize.c b/src/normalize.c
new file mode 100644
index 00000000..8c48f13b
--- /dev/null
+++ b/src/normalize.c
@@ -0,0 +1,100 @@
+/* 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 */
+}