From 22b16884a2fb054ed219f07774cc16ec3af34c26 Mon Sep 17 00:00:00 2001
From: "J. Alexander Treuman" <jat@spatialrift.net>
Date: Thu, 27 Jul 2006 00:50:59 +0000
Subject: Use AudioCompress for volume normalization

git-svn-id: https://svn.musicpd.org/mpd/trunk@4474 09075e82-0dd4-0310-85a5-a0d7c8717e4f
---
 src/normalize.c | 89 ++++++++++++---------------------------------------------
 1 file changed, 18 insertions(+), 71 deletions(-)

(limited to 'src/normalize.c')

diff --git a/src/normalize.c b/src/normalize.c
index 8c48f13bd..31394cbfb 100644
--- a/src/normalize.c
+++ b/src/normalize.c
@@ -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)
 
-#define MUL_MIN 0.1
-#define MUL_MAX 5.0
-#define NSAMPLES 128
-#define MIN_SAMPLE_SIZE 32000
+#include <stdlib.h>
 
-#define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
+int normalizationEnabled;
 
-void normalizeData(char *buffer, int bufferSize, AudioFormat *format)
+void initNormalization()
 {
-	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);
+	normalizationEnabled = getBoolConfigParam(CONF_VOLUME_NORMALIZATION);
+	if (normalizationEnabled == -1) normalizationEnabled = 0;
+	else if (normalizationEnabled < 0) exit(EXIT_FAILURE);
 
-	/* 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 (normalizationEnabled)
+		CompressCfg(0, ANTICLIP, TARGET, GAINMAX, GAINSMOOTH, BUCKETS);
+}
 
-	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);
-		}
-	}
+void finishNormalization()
+{
+	if (normalizationEnabled) CompressFree();
+}
 
-	/* scale and clamp the samples */
-	for (i = 0; i < length; i++) {
-		temp = data[i] * multiplier;
-		data[i] = clamp(temp, SHRT_MIN, SHRT_MAX);
-	}
+void normalizeData(char *buffer, int bufferSize, AudioFormat *format)
+{
+	if ((format->bits != 16) || (format->channels != 2)) return;
 
-	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);
 }
-- 
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