output_init: use the normalize filter plugin

Use the plugin instead of the glue code in normalize.c.  This is used
wrapped inside a "autoconv" filter, to enable normalization for all
input file formats.

Makefile.am

@@ -120,7 +120,6 @@ mpd_headers = \
 	src/mixer/software_mixer_plugin.h \
 	src/mixer/pulse_mixer_plugin.h \
 	src/daemon.h \
-	src/normalize.h \
 	src/AudioCompress/config.h \
 	src/AudioCompress/compress.h \
 	src/buffer.h \
@@ -264,7 +263,6 @@ src_mpd_SOURCES = \
 	src/main.c \
 	src/event_pipe.c \
 	src/daemon.c \
-	src/normalize.c \
 	src/AudioCompress/compress.c \
 	src/buffer.c \
 	src/pipe.c \
@@ -902,8 +900,7 @@ test_software_volume_LDADD = \
 test_run_normalize_SOURCES = test/run_normalize.c \
 	src/audio_check.c \
 	src/audio_parser.c \
-	src/AudioCompress/compress.c \
-	src/normalize.c
+	src/AudioCompress/compress.c
 test_run_normalize_LDADD = \
 	$(GLIB_LIBS)
 
@@ -952,9 +949,12 @@ test_run_output_SOURCES = test/run_output.c \
 	$(MIXER_SRC) \
 	src/filter_plugin.c src/filter/chain_filter_plugin.c \
 	src/filter_config.c \
+	src/filter/autoconvert_filter_plugin.c \
 	src/filter/convert_filter_plugin.c \
+	src/filter/normalize_filter_plugin.c \
 	src/filter/volume_filter_plugin.c \
 	src/pcm_volume.c \
+	src/AudioCompress/compress.c \
 	src/fd_util.c \
 	$(OUTPUT_SRC)
 

NEWS

@@ -66,6 +66,7 @@ ver 0.16 (20??/??/??)
   - sort songs by album name first, then disc/track number
   - rescan after metadata_to_use change
 * normalize: upgraded to AudioCompress 2.0
+  - automatically convert to 16 bit samples
 * log unused/unknown block parameters
 * removed the deprecated "error_file" option
 * save state when stopped

src/decoder_api.c

@@ -25,8 +25,6 @@
 #include "audio.h"
 #include "song.h"
 #include "buffer.h"
-
-#include "normalize.h"
 #include "pipe.h"
 #include "chunk.h"
 
@@ -340,8 +338,6 @@ decoder_data(struct decoder *decoder,
 		if (replay_gain_mode != REPLAY_GAIN_OFF)
 			replay_gain_apply(replay_gain_info, dest, nbytes,
 					  &dc->out_audio_format);
-		else if (normalizationEnabled)
-			normalizeData(dest, nbytes, &dc->out_audio_format);
 
 		/* expand the music pipe chunk */
 

src/main.c

@@ -49,7 +49,6 @@
 #include "state_file.h"
 #include "tag.h"
 #include "dbUtils.h"
-#include "normalize.h"
 #include "zeroconf.h"
 #include "event_pipe.h"
 #include "dirvec.h"
@@ -348,7 +347,6 @@ int main(int argc, char *argv[])
 	audio_output_all_init();
 	client_manager_init();
 	replay_gain_global_init();
-	initNormalization();
 
 	if (!input_stream_global_init(&error)) {
 		g_warning("%s", error->message);
@@ -426,7 +424,6 @@ int main(int argc, char *argv[])
 
 	playlist_list_global_finish();
 	input_stream_global_finish();
-	finishNormalization();
 	audio_output_all_finish();
 	volume_finish();
 	mapper_finish();

src/normalize.c

@@ -1,54 +0,0 @@
-/*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
- * 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.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include "config.h"
-#include "normalize.h"
-#include "AudioCompress/compress.h"
-#include "conf.h"
-#include "audio_format.h"
-
-#define DEFAULT_VOLUME_NORMALIZATION 0
-
-int normalizationEnabled;
-
-static struct Compressor *compressor;
-
-void initNormalization(void)
-{
-	normalizationEnabled = config_get_bool(CONF_VOLUME_NORMALIZATION,
-					       DEFAULT_VOLUME_NORMALIZATION);
-
-	if (normalizationEnabled)
-		compressor = Compressor_new(0);
-}
-
-void finishNormalization(void)
-{
-	if (normalizationEnabled)
-		Compressor_delete(compressor);
-}
-
-void normalizeData(void *buffer, int bufferSize,
-		   const struct audio_format *format)
-{
-	if (format->format != SAMPLE_FORMAT_S16 || format->channels != 2)
-		return;
-
-	Compressor_Process_int16(compressor, buffer, bufferSize / 2);
-}

src/normalize.h

@@ -1,34 +0,0 @@
-/*
- * Copyright (C) 2003-2009 The Music Player Daemon Project
- * 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.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#ifndef MPD_NORMALIZE_H
-#define MPD_NORMALIZE_H
-
-struct audio_format;
-
-extern int normalizationEnabled;
-
-void initNormalization(void);
-
-void finishNormalization(void);
-
-void normalizeData(void *buffer, int bufferSize,
-		   const struct audio_format *format);
-
-#endif /* !NORMALIZE_H */

src/output_init.c

@@ -31,6 +31,7 @@
 #include "filter_registry.h"
 #include "filter_config.h"
 #include "filter/chain_filter_plugin.h"
+#include "filter/autoconvert_filter_plugin.h"
 
 #include <glib.h>
 
@@ -192,6 +193,16 @@ audio_output_init(struct audio_output *ao, const struct config_param *param,
 
 	ao->filter = filter_chain_new();
 	assert(ao->filter != NULL);
+
+	if (config_get_bool(CONF_VOLUME_NORMALIZATION, false)) {
+		struct filter *normalize_filter =
+			filter_new(&normalize_filter_plugin, NULL, NULL);
+		assert(normalize_filter != NULL);
+
+		filter_chain_append(ao->filter,
+				    autoconvert_filter_new(normalize_filter));
+	}
+
 	filter_chain_parse(ao->filter,
 	                   config_get_block_string(param, AUDIO_FILTERS, ""),
 	                   &error

test/run_normalize.c

@@ -24,10 +24,9 @@
  */
 
 #include "config.h"
-#include "normalize.h"
+#include "AudioCompress/compress.h"
 #include "audio_parser.h"
 #include "audio_format.h"
-#include "conf.h"
 
 #include <glib.h>
 
@@ -35,20 +34,12 @@
 #include <unistd.h>
 #include <string.h>
 
-bool
-config_get_bool(const char *name, bool default_value)
-{
-	if (strcmp(name, CONF_VOLUME_NORMALIZATION) == 0)
-		return true;
-	else
-		return default_value;
-}
-
 int main(int argc, char **argv)
 {
 	GError *error = NULL;
 	struct audio_format audio_format;
 	bool ret;
+	struct Compressor *compressor;
 	static char buffer[4096];
 	ssize_t nbytes;
 
@@ -68,12 +59,13 @@ int main(int argc, char **argv)
 	} else
 		audio_format_init(&audio_format, 48000, 16, 2);
 
-	initNormalization();
+	compressor = Compressor_new(0);
 
 	while ((nbytes = read(0, buffer, sizeof(buffer))) > 0) {
-		normalizeData(buffer, nbytes, &audio_format);
+		Compressor_Process_int16(compressor,
+					 (int16_t *)buffer, nbytes / 2);
 		write(1, buffer, nbytes);
 	}
 
-	finishNormalization();
+	Compressor_delete(compressor);
 }