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authorMax Kellermann <max@duempel.org>2009-12-02 21:59:06 +0100
committerMax Kellermann <max@duempel.org>2009-12-14 22:24:00 +0100
commit400600ffffe64b85e7be14c978c2d0803a34097d (patch)
treee6d5c1a67220b5c553e5b4ab31f6c893c36f120c /src/filter
parent940e66bb89cc2ebe05ecc71f78851460e5df35ce (diff)
downloadmpd-400600ffffe64b85e7be14c978c2d0803a34097d.tar.gz
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filter: added normalize filter plugin
Wrap the AudioCompress library in a filter plugin.
Diffstat (limited to 'src/filter')
-rw-r--r--src/filter/normalize_filter_plugin.c123
1 files changed, 123 insertions, 0 deletions
diff --git a/src/filter/normalize_filter_plugin.c b/src/filter/normalize_filter_plugin.c
new file mode 100644
index 000000000..2694646ee
--- /dev/null
+++ b/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,
+};