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.
This commit is contained in:
@@ -25,8 +25,6 @@
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#include "audio.h"
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#include "song.h"
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#include "buffer.h"
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#include "normalize.h"
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#include "pipe.h"
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#include "chunk.h"
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@@ -340,8 +338,6 @@ decoder_data(struct decoder *decoder,
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if (replay_gain_mode != REPLAY_GAIN_OFF)
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replay_gain_apply(replay_gain_info, dest, nbytes,
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&dc->out_audio_format);
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else if (normalizationEnabled)
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normalizeData(dest, nbytes, &dc->out_audio_format);
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/* expand the music pipe chunk */
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@@ -49,7 +49,6 @@
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#include "state_file.h"
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#include "tag.h"
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#include "dbUtils.h"
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#include "normalize.h"
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#include "zeroconf.h"
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#include "event_pipe.h"
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#include "dirvec.h"
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@@ -348,7 +347,6 @@ int main(int argc, char *argv[])
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audio_output_all_init();
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client_manager_init();
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replay_gain_global_init();
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initNormalization();
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if (!input_stream_global_init(&error)) {
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g_warning("%s", error->message);
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@@ -426,7 +424,6 @@ int main(int argc, char *argv[])
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playlist_list_global_finish();
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input_stream_global_finish();
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finishNormalization();
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audio_output_all_finish();
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volume_finish();
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mapper_finish();
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@@ -1,54 +0,0 @@
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/*
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* Copyright (C) 2003-2009 The Music Player Daemon Project
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* 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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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "normalize.h"
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#include "AudioCompress/compress.h"
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#include "conf.h"
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#include "audio_format.h"
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#define DEFAULT_VOLUME_NORMALIZATION 0
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int normalizationEnabled;
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static struct Compressor *compressor;
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void initNormalization(void)
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{
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normalizationEnabled = config_get_bool(CONF_VOLUME_NORMALIZATION,
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DEFAULT_VOLUME_NORMALIZATION);
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if (normalizationEnabled)
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compressor = Compressor_new(0);
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}
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void finishNormalization(void)
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{
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if (normalizationEnabled)
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Compressor_delete(compressor);
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}
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void normalizeData(void *buffer, int bufferSize,
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const struct audio_format *format)
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{
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if (format->format != SAMPLE_FORMAT_S16 || format->channels != 2)
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return;
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Compressor_Process_int16(compressor, buffer, bufferSize / 2);
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}
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@@ -1,34 +0,0 @@
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/*
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* Copyright (C) 2003-2009 The Music Player Daemon Project
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* 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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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef MPD_NORMALIZE_H
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#define MPD_NORMALIZE_H
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struct audio_format;
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extern int normalizationEnabled;
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void initNormalization(void);
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void finishNormalization(void);
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void normalizeData(void *buffer, int bufferSize,
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const struct audio_format *format);
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#endif /* !NORMALIZE_H */
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@@ -31,6 +31,7 @@
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#include "filter_registry.h"
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#include "filter_config.h"
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#include "filter/chain_filter_plugin.h"
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#include "filter/autoconvert_filter_plugin.h"
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#include <glib.h>
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@@ -192,6 +193,16 @@ audio_output_init(struct audio_output *ao, const struct config_param *param,
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ao->filter = filter_chain_new();
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assert(ao->filter != NULL);
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if (config_get_bool(CONF_VOLUME_NORMALIZATION, false)) {
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struct filter *normalize_filter =
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filter_new(&normalize_filter_plugin, NULL, NULL);
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assert(normalize_filter != NULL);
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filter_chain_append(ao->filter,
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autoconvert_filter_new(normalize_filter));
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}
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filter_chain_parse(ao->filter,
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config_get_block_string(param, AUDIO_FILTERS, ""),
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&error
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