filter: added normalize filter plugin

Wrap the AudioCompress library in a filter plugin.

src/filter/normalize_filter_plugin.c

@@ -0,0 +1,123 @@
+/*
+ * 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 "filter_plugin.h"
+#include "filter_internal.h"
+#include "filter_registry.h"
+#include "pcm_buffer.h"
+#include "audio_format.h"
+#include "AudioCompress/compress.h"
+
+#include <assert.h>
+#include <string.h>
+
+struct normalize_filter {
+	struct filter filter;
+
+	struct Compressor *compressor;
+
+	struct pcm_buffer buffer;
+};
+
+static inline GQuark
+normalize_quark(void)
+{
+	return g_quark_from_static_string("normalize");
+}
+
+static struct filter *
+normalize_filter_init(G_GNUC_UNUSED const struct config_param *param,
+		      G_GNUC_UNUSED GError **error_r)
+{
+	struct normalize_filter *filter = g_new(struct normalize_filter, 1);
+
+	filter_init(&filter->filter, &normalize_filter_plugin);
+
+	return &filter->filter;
+}
+
+static void
+normalize_filter_finish(struct filter *filter)
+{
+	g_free(filter);
+}
+
+static const struct audio_format *
+normalize_filter_open(struct filter *_filter,
+		      const struct audio_format *audio_format,
+		      GError **error_r)
+{
+	struct normalize_filter *filter = (struct normalize_filter *)_filter;
+
+	if (audio_format->format != SAMPLE_FORMAT_S16) {
+		g_set_error(error_r, normalize_quark(), 0,
+			    "Unsupported audio format");
+		return false;
+	}
+
+	if (audio_format->reverse_endian) {
+		g_set_error(error_r, normalize_quark(), 0,
+			    "Normalize for reverse endian "
+			    "samples is not implemented");
+		return false;
+	}
+
+	filter->compressor = Compressor_new(0);
+
+	pcm_buffer_init(&filter->buffer);
+
+	return audio_format;
+}
+
+static void
+normalize_filter_close(struct filter *_filter)
+{
+	struct normalize_filter *filter = (struct normalize_filter *)_filter;
+
+	pcm_buffer_deinit(&filter->buffer);
+	Compressor_delete(filter->compressor);
+}
+
+static const void *
+normalize_filter_filter(struct filter *_filter,
+			const void *src, size_t src_size, size_t *dest_size_r,
+			G_GNUC_UNUSED GError **error_r)
+{
+	struct normalize_filter *filter = (struct normalize_filter *)_filter;
+	void *dest;
+
+	dest = pcm_buffer_get(&filter->buffer, src_size);
+
+	memcpy(dest, src, src_size);
+
+	Compressor_Process_int16(filter->compressor, dest, src_size / 2);
+
+	*dest_size_r = src_size;
+	return dest;
+}
+
+const struct filter_plugin normalize_filter_plugin = {
+	.name = "normalize",
+	.init = normalize_filter_init,
+	.finish = normalize_filter_finish,
+	.open = normalize_filter_open,
+	.close = normalize_filter_close,
+	.filter = normalize_filter_filter,
+};