From: Andre Noll <maan@systemlinux.org>
Date: Sun, 30 Nov 2008 20:45:38 +0000 (+0100)
Subject: Rename filter source files.
X-Git-Tag: v0.3.4~97
X-Git-Url: http://git.tuebingen.mpg.de/?p=paraslash.git;a=commitdiff_plain;h=dd0b1f12758ac877de6834b34f8c7f9291d4311d

Rename filter source files.

Some filters already had the _filter postfix, others had not. Rename
the latter just to be consistent.
---

diff --git a/Makefile.in b/Makefile.in
index 5dd08d6d..8ac8b65b 100644
--- a/Makefile.in
+++ b/Makefile.in
@@ -161,15 +161,15 @@ ortp_recv.o: ortp_recv.c
 ortp_send.o: ortp_send.c
 	$(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @ortp_cppflags@ $<
 
-oggdec.o: oggdec.c
+oggdec_filter.o: oggdec_filter.c
 	$(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @oggvorbis_cppflags@ $<
 ogg_afh.o: ogg_afh.c
 	$(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @oggvorbis_cppflags@ $<
 
-mp3dec.o: mp3dec.c
+mp3dec_filter.o: mp3dec_filter.c
 	$(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @mad_cppflags@ $<
 
-aacdec.o: aacdec.c
+aacdec_filter.o: aacdec_filter.c
 	$(CC) -c $(CPPFLAGS) $(DEBUG_CPPFLAGS) @faad_cppflags@ $<
 
 aac_common.o: aac_common.c
diff --git a/aacdec.c b/aacdec.c
deleted file mode 100644
index 38f69e88..00000000
--- a/aacdec.c
+++ /dev/null
@@ -1,191 +0,0 @@
-/*
- * Copyright (C) 2006-2008 Andre Noll <maan@systemlinux.org>
- *
- * Licensed under the GPL v2. For licencing details see COPYING.
- */
-/*
- * based in parts on libfaad, Copyright (C) 2003-2005 M. Bakker,
- * Ahead Software AG
- */
-
-/** \file aacdec.c paraslash's aac (m4a) decoder */
-
-#include "para.h"
-
-#include "list.h"
-#include "sched.h"
-#include "filter.h"
-#include "error.h"
-#include "string.h"
-#include "aac.h"
-
-/** the output buffer size */
-#define AAC_OUTBUF_SIZE (32 * 1024)
-
-/** give up decoding after that many errors */
-#define MAX_ERRORS 20
-
-/**
- * data specific to the aacdec filter
- *
- * \sa filter, filter_node
- */
-struct private_aacdec_data {
-	/** the return value of aac_open */
-	NeAACDecHandle handle;
-	/** info about the currently decoded frame */
-	NeAACDecFrameInfo frame_info;
-	/** whether this instance of the aac decoder is already initialized */
-	int initialized;
-	/**
-	 * return value of aac_find_esds(). Used to call the right aacdec
-	 * init function
-	 */
-	unsigned long decoder_length;
-	/** number of times the decoder returned an error */
-	unsigned error_count;
-	/** number of bytes already consumed from the imput stream */
-	size_t consumed_total;
-	/** return value of aac_find_entry_point */
-	size_t entry;
-};
-
-static ssize_t aacdec(char *input_buffer, size_t len, struct filter_node *fn)
-{
-	struct private_aacdec_data *padd = fn->private_data;
-	struct filter_chain *fc = fn->fc;
-	int i, ret;
-	unsigned char *p, *outbuffer;
-	unsigned char *inbuf = (unsigned char*)input_buffer;
-	size_t skip, consumed = 0;
-
-	if (fn->loaded > fn->bufsize * 3 / 5)
-		return 0;
-	if (len < 2048 && !*fc->input_error)
-		return 0;
-
-	if (!padd->initialized) {
-		unsigned long rate = 0;
-		unsigned char channels = 0;
-		ret = aac_find_esds(inbuf, len, &skip, &padd->decoder_length);
-		if (ret < 0) {
-			PARA_INFO_LOG("%s\n", para_strerror(-ret));
-			ret = NeAACDecInit(padd->handle, inbuf,
-				len, &rate, &channels);
-			PARA_INFO_LOG("decoder init: %d\n", ret);
-			if (ret < 0) {
-				ret = -E_AACDEC_INIT;
-				goto out;
-			}
-			consumed = ret;
-		} else {
-			PARA_INFO_LOG("decoder len: %lu\n",
-				padd->decoder_length);
-			consumed += skip;
-			p = inbuf + consumed;
-			ret = -E_AACDEC_INIT;
-			if (NeAACDecInit2(padd->handle, p,
-					padd->decoder_length, &rate,
-					&channels) < 0)
-				goto out;
-		}
-		fc->samplerate = rate;
-		fc->channels = channels;
-		PARA_INFO_LOG("rate: %u, channels: %d\n",
-			fc->samplerate, fc->channels);
-		padd->initialized = 1;
-	}
-	if (padd->decoder_length > 0) {
-		consumed = 0;
-		if (!padd->entry) {
-			ret = aac_find_entry_point(inbuf + consumed,
-				len - consumed, &skip);
-			if (ret < 0) {
-				ret = len;
-				goto out;
-			}
-			consumed += skip;
-			padd->entry = ret;
-			PARA_INFO_LOG("entry: %zu\n", padd->entry);
-		}
-		ret = len;
-		if (padd->consumed_total + len < padd->entry)
-			goto out;
-		if (padd->consumed_total < padd->entry)
-			consumed = padd->entry - padd->consumed_total;
-	}
-	for (; consumed < len; consumed++)
-		if ((inbuf[consumed] & 0xfe) == 0x20)
-			break;
-	if (consumed >= len)
-		goto success;
-	p = inbuf + consumed;
-	outbuffer = NeAACDecDecode(padd->handle, &padd->frame_info, p,
-		len - consumed);
-	if (padd->frame_info.error) {
-		ret = -E_AAC_DECODE;
-		if (padd->error_count++ > MAX_ERRORS)
-			goto out;
-		PARA_ERROR_LOG("frame_error: %d, consumed: %zu + %zd + %lu\n",
-			padd->frame_info.error, padd->consumed_total,
-			consumed, padd->frame_info.bytesconsumed);
-		PARA_ERROR_LOG("%s\n", NeAACDecGetErrorMessage(
-			padd->frame_info.error));
-		consumed++; /* catch 21 */
-		goto success;
-	}
-	padd->error_count = 0;
-	consumed += padd->frame_info.bytesconsumed;
-	ret = consumed;
-	if (!padd->frame_info.samples)
-		goto out;
-	ret = -E_AAC_OVERRUN;
-	if (padd->frame_info.samples * 2 + fn->loaded > fn->bufsize)
-		goto out;
-	for (i = 0; i < padd->frame_info.samples; i++) {
-		short *s = (short *)outbuffer;
-		write_int16_host_endian(fn->buf + fn->loaded, s[i]);
-		fn->loaded += 2;
-	}
-success:
-	ret = consumed;
-out:
-	if (ret > 0)
-		padd->consumed_total += ret;
-	return ret;
-}
-
-static void aacdec_open(struct filter_node *fn)
-{
-	struct private_aacdec_data *padd = para_calloc(sizeof(*padd));
-
-	fn->private_data = padd;
-	fn->bufsize = AAC_OUTBUF_SIZE;
-	fn->buf = para_calloc(fn->bufsize);
-	padd->handle = aac_open();
-}
-
-static void aacdec_close(struct filter_node *fn)
-{
-	struct private_aacdec_data *padd = fn->private_data;
-
-	NeAACDecClose(padd->handle);
-	free(fn->buf);
-	fn->buf = NULL;
-	free(padd);
-	fn->private_data = NULL;
-}
-
-/**
- * the init function of the aacdec filter
- *
- * \param f pointer to the filter struct to initialize
- *
- * \sa filter::init
- */
-void aacdec_filter_init(struct filter *f)
-{
-	f->open = aacdec_open;
-	f->convert = aacdec;
-	f->close = aacdec_close;
-}
diff --git a/aacdec_filter.c b/aacdec_filter.c
new file mode 100644
index 00000000..956f338d
--- /dev/null
+++ b/aacdec_filter.c
@@ -0,0 +1,191 @@
+/*
+ * Copyright (C) 2006-2008 Andre Noll <maan@systemlinux.org>
+ *
+ * Licensed under the GPL v2. For licencing details see COPYING.
+ */
+/*
+ * based in parts on libfaad, Copyright (C) 2003-2005 M. Bakker,
+ * Ahead Software AG
+ */
+
+/** \file aacdec_filter.c paraslash's aac (m4a) decoder. */
+
+#include "para.h"
+
+#include "list.h"
+#include "sched.h"
+#include "filter.h"
+#include "error.h"
+#include "string.h"
+#include "aac.h"
+
+/** the output buffer size */
+#define AAC_OUTBUF_SIZE (32 * 1024)
+
+/** give up decoding after that many errors */
+#define MAX_ERRORS 20
+
+/**
+ * data specific to the aacdec filter
+ *
+ * \sa filter, filter_node
+ */
+struct private_aacdec_data {
+	/** the return value of aac_open */
+	NeAACDecHandle handle;
+	/** info about the currently decoded frame */
+	NeAACDecFrameInfo frame_info;
+	/** whether this instance of the aac decoder is already initialized */
+	int initialized;
+	/**
+	 * return value of aac_find_esds(). Used to call the right aacdec
+	 * init function
+	 */
+	unsigned long decoder_length;
+	/** number of times the decoder returned an error */
+	unsigned error_count;
+	/** number of bytes already consumed from the imput stream */
+	size_t consumed_total;
+	/** return value of aac_find_entry_point */
+	size_t entry;
+};
+
+static ssize_t aacdec(char *input_buffer, size_t len, struct filter_node *fn)
+{
+	struct private_aacdec_data *padd = fn->private_data;
+	struct filter_chain *fc = fn->fc;
+	int i, ret;
+	unsigned char *p, *outbuffer;
+	unsigned char *inbuf = (unsigned char*)input_buffer;
+	size_t skip, consumed = 0;
+
+	if (fn->loaded > fn->bufsize * 3 / 5)
+		return 0;
+	if (len < 2048 && !*fc->input_error)
+		return 0;
+
+	if (!padd->initialized) {
+		unsigned long rate = 0;
+		unsigned char channels = 0;
+		ret = aac_find_esds(inbuf, len, &skip, &padd->decoder_length);
+		if (ret < 0) {
+			PARA_INFO_LOG("%s\n", para_strerror(-ret));
+			ret = NeAACDecInit(padd->handle, inbuf,
+				len, &rate, &channels);
+			PARA_INFO_LOG("decoder init: %d\n", ret);
+			if (ret < 0) {
+				ret = -E_AACDEC_INIT;
+				goto out;
+			}
+			consumed = ret;
+		} else {
+			PARA_INFO_LOG("decoder len: %lu\n",
+				padd->decoder_length);
+			consumed += skip;
+			p = inbuf + consumed;
+			ret = -E_AACDEC_INIT;
+			if (NeAACDecInit2(padd->handle, p,
+					padd->decoder_length, &rate,
+					&channels) < 0)
+				goto out;
+		}
+		fc->samplerate = rate;
+		fc->channels = channels;
+		PARA_INFO_LOG("rate: %u, channels: %d\n",
+			fc->samplerate, fc->channels);
+		padd->initialized = 1;
+	}
+	if (padd->decoder_length > 0) {
+		consumed = 0;
+		if (!padd->entry) {
+			ret = aac_find_entry_point(inbuf + consumed,
+				len - consumed, &skip);
+			if (ret < 0) {
+				ret = len;
+				goto out;
+			}
+			consumed += skip;
+			padd->entry = ret;
+			PARA_INFO_LOG("entry: %zu\n", padd->entry);
+		}
+		ret = len;
+		if (padd->consumed_total + len < padd->entry)
+			goto out;
+		if (padd->consumed_total < padd->entry)
+			consumed = padd->entry - padd->consumed_total;
+	}
+	for (; consumed < len; consumed++)
+		if ((inbuf[consumed] & 0xfe) == 0x20)
+			break;
+	if (consumed >= len)
+		goto success;
+	p = inbuf + consumed;
+	outbuffer = NeAACDecDecode(padd->handle, &padd->frame_info, p,
+		len - consumed);
+	if (padd->frame_info.error) {
+		ret = -E_AAC_DECODE;
+		if (padd->error_count++ > MAX_ERRORS)
+			goto out;
+		PARA_ERROR_LOG("frame_error: %d, consumed: %zu + %zd + %lu\n",
+			padd->frame_info.error, padd->consumed_total,
+			consumed, padd->frame_info.bytesconsumed);
+		PARA_ERROR_LOG("%s\n", NeAACDecGetErrorMessage(
+			padd->frame_info.error));
+		consumed++; /* catch 21 */
+		goto success;
+	}
+	padd->error_count = 0;
+	consumed += padd->frame_info.bytesconsumed;
+	ret = consumed;
+	if (!padd->frame_info.samples)
+		goto out;
+	ret = -E_AAC_OVERRUN;
+	if (padd->frame_info.samples * 2 + fn->loaded > fn->bufsize)
+		goto out;
+	for (i = 0; i < padd->frame_info.samples; i++) {
+		short *s = (short *)outbuffer;
+		write_int16_host_endian(fn->buf + fn->loaded, s[i]);
+		fn->loaded += 2;
+	}
+success:
+	ret = consumed;
+out:
+	if (ret > 0)
+		padd->consumed_total += ret;
+	return ret;
+}
+
+static void aacdec_open(struct filter_node *fn)
+{
+	struct private_aacdec_data *padd = para_calloc(sizeof(*padd));
+
+	fn->private_data = padd;
+	fn->bufsize = AAC_OUTBUF_SIZE;
+	fn->buf = para_calloc(fn->bufsize);
+	padd->handle = aac_open();
+}
+
+static void aacdec_close(struct filter_node *fn)
+{
+	struct private_aacdec_data *padd = fn->private_data;
+
+	NeAACDecClose(padd->handle);
+	free(fn->buf);
+	fn->buf = NULL;
+	free(padd);
+	fn->private_data = NULL;
+}
+
+/**
+ * the init function of the aacdec filter
+ *
+ * \param f pointer to the filter struct to initialize
+ *
+ * \sa filter::init
+ */
+void aacdec_filter_init(struct filter *f)
+{
+	f->open = aacdec_open;
+	f->convert = aacdec;
+	f->close = aacdec_close;
+}
diff --git a/compress.c b/compress.c
deleted file mode 100644
index d6f7520d..00000000
--- a/compress.c
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
- * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
- *
- * Licensed under the GPL v2. For licencing details see COPYING.
- */
-
-/** \file compress.c Paraslash's dynamic audio range compressor. */
-
-/*
- * Uses ideas of AudioCompress, (C) 2002-2004  M. Hari Nezumi <magenta@trikuare.cx>
- */
-
-#include "para.h"
-#include "compress_filter.cmdline.h"
-#include "list.h"
-#include "sched.h"
-#include "filter.h"
-#include "string.h"
-#include "error.h"
-
-/** The size of the output data buffer. */
-#define COMPRESS_CHUNK_SIZE 40960
-
-extern char *stat_item_values[NUM_STAT_ITEMS];
-
-/** Data specific to the compress filter. */
-struct private_compress_data {
-	/** The current multiplier. */
-	unsigned current_gain;
-	/** Points to the configuration data for this instance of the compress filter. */
-	struct compress_filter_args_info *conf;
-	/** Maximal admissible gain. */
-	unsigned max_gain;
-	/** Number of samples already seen. */
-	unsigned num_samples;
-	/** Absolute value of the maximal sample in the current block. */
-	unsigned peak;
-};
-
-static ssize_t compress(char *inbuf, size_t inbuf_len, struct filter_node *fn)
-{
-	size_t i, length = PARA_MIN((inbuf_len / 2) * 2,
-		(fn->bufsize - fn->loaded) / 2 * 2);
-	struct private_compress_data *pcd = fn->private_data;
-	int16_t *ip = (int16_t *)inbuf, *op = (int16_t *)(fn->buf + fn->loaded);
-	unsigned gain_shift = pcd->conf->inertia_arg + pcd->conf->damp_arg,
-		mask = (1 << pcd->conf->blocksize_arg) - 1;
-
-	if (!length)
-		return 0;
-	for (i = 0; i < length / 2; i++) {
-		/* be careful in that heat, my dear */
-		int sample = *ip++, adjusted_sample = (PARA_ABS(sample) *
-			pcd->current_gain) >> gain_shift;
-		if (unlikely(adjusted_sample > 32767)) { /* clip */
-			PARA_NOTICE_LOG("clip: sample: %d, adjusted sample: %d\n",
-				sample, adjusted_sample);
-			adjusted_sample = 32767;
-			pcd->current_gain = (3 * pcd->current_gain +
-				(1 << pcd->conf->inertia_arg)) / 4;
-			pcd->peak = 0;
-		} else
-			pcd->peak = PARA_MAX(pcd->peak, adjusted_sample);
-		*op++ = sample >= 0? adjusted_sample : -adjusted_sample;
-		if (likely(++pcd->num_samples & mask))
-			continue;
-//		PARA_DEBUG_LOG("gain: %u, peak: %u\n", pcd->current_gain,
-//			pcd->peak);
-		if (pcd->peak < pcd->conf->target_level_arg) {
-			if (pcd->current_gain < pcd->max_gain)
-				pcd->current_gain++;
-		} else
-			pcd->current_gain = PARA_MAX(pcd->current_gain - 2,
-				1 << pcd->conf->inertia_arg);
-		pcd->peak = 0;
-	}
-	fn->loaded += length;
-	return length;
-}
-
-static void close_compress(struct filter_node *fn)
-{
-	free(fn->private_data);
-	free(fn->buf);
-}
-
-/** TODO: Add sanity checks */
-static int compress_parse_config(int argc, char **argv, void **config)
-{
-	int ret;
-	struct compress_filter_args_info *compress_conf
-		= para_calloc(sizeof(*compress_conf));
-
-	ret = -E_COMPRESS_SYNTAX;
-	if (compress_cmdline_parser(argc, argv, compress_conf))
-		goto err;
-	*config = compress_conf;
-	return 1;
-err:
-	free(compress_conf);
-	return  ret;
-}
-
-static void open_compress(struct filter_node *fn)
-{
-	struct private_compress_data *pcd = para_calloc(
-		sizeof(struct private_compress_data));
-	pcd->conf = fn->conf;
-	fn->private_data = pcd;
-	fn->bufsize = COMPRESS_CHUNK_SIZE;
-	fn->buf = para_malloc(fn->bufsize);
-	pcd->current_gain = 1 << pcd->conf->inertia_arg;
-	pcd->max_gain = 1 << (pcd->conf->inertia_arg + pcd->conf->aggressiveness_arg);
-}
-
-/**
- * The init function of the compress filter.
- *
- * \param f Pointer to the struct to initialize.
- */
-void compress_filter_init(struct filter *f)
-{
-	f->open = open_compress;
-	f->close = close_compress;
-	f->convert = compress;
-	f->print_help = compress_cmdline_parser_print_help;
-	f->parse_config = compress_parse_config;
-}
diff --git a/compress_filter.c b/compress_filter.c
new file mode 100644
index 00000000..6034ce79
--- /dev/null
+++ b/compress_filter.c
@@ -0,0 +1,128 @@
+/*
+ * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
+ *
+ * Licensed under the GPL v2. For licencing details see COPYING.
+ */
+
+/** \file compress_filter.c Paraslash's dynamic audio range compressor. */
+
+/*
+ * Uses ideas of AudioCompress, (C) 2002-2004  M. Hari Nezumi <magenta@trikuare.cx>
+ */
+
+#include "para.h"
+#include "compress_filter.cmdline.h"
+#include "list.h"
+#include "sched.h"
+#include "filter.h"
+#include "string.h"
+#include "error.h"
+
+/** The size of the output data buffer. */
+#define COMPRESS_CHUNK_SIZE 40960
+
+extern char *stat_item_values[NUM_STAT_ITEMS];
+
+/** Data specific to the compress filter. */
+struct private_compress_data {
+	/** The current multiplier. */
+	unsigned current_gain;
+	/** Points to the configuration data for this instance of the compress filter. */
+	struct compress_filter_args_info *conf;
+	/** Maximal admissible gain. */
+	unsigned max_gain;
+	/** Number of samples already seen. */
+	unsigned num_samples;
+	/** Absolute value of the maximal sample in the current block. */
+	unsigned peak;
+};
+
+static ssize_t compress(char *inbuf, size_t inbuf_len, struct filter_node *fn)
+{
+	size_t i, length = PARA_MIN((inbuf_len / 2) * 2,
+		(fn->bufsize - fn->loaded) / 2 * 2);
+	struct private_compress_data *pcd = fn->private_data;
+	int16_t *ip = (int16_t *)inbuf, *op = (int16_t *)(fn->buf + fn->loaded);
+	unsigned gain_shift = pcd->conf->inertia_arg + pcd->conf->damp_arg,
+		mask = (1 << pcd->conf->blocksize_arg) - 1;
+
+	if (!length)
+		return 0;
+	for (i = 0; i < length / 2; i++) {
+		/* be careful in that heat, my dear */
+		int sample = *ip++, adjusted_sample = (PARA_ABS(sample) *
+			pcd->current_gain) >> gain_shift;
+		if (unlikely(adjusted_sample > 32767)) { /* clip */
+			PARA_NOTICE_LOG("clip: sample: %d, adjusted sample: %d\n",
+				sample, adjusted_sample);
+			adjusted_sample = 32767;
+			pcd->current_gain = (3 * pcd->current_gain +
+				(1 << pcd->conf->inertia_arg)) / 4;
+			pcd->peak = 0;
+		} else
+			pcd->peak = PARA_MAX(pcd->peak, adjusted_sample);
+		*op++ = sample >= 0? adjusted_sample : -adjusted_sample;
+		if (likely(++pcd->num_samples & mask))
+			continue;
+//		PARA_DEBUG_LOG("gain: %u, peak: %u\n", pcd->current_gain,
+//			pcd->peak);
+		if (pcd->peak < pcd->conf->target_level_arg) {
+			if (pcd->current_gain < pcd->max_gain)
+				pcd->current_gain++;
+		} else
+			pcd->current_gain = PARA_MAX(pcd->current_gain - 2,
+				1 << pcd->conf->inertia_arg);
+		pcd->peak = 0;
+	}
+	fn->loaded += length;
+	return length;
+}
+
+static void close_compress(struct filter_node *fn)
+{
+	free(fn->private_data);
+	free(fn->buf);
+}
+
+/** TODO: Add sanity checks */
+static int compress_parse_config(int argc, char **argv, void **config)
+{
+	int ret;
+	struct compress_filter_args_info *compress_conf
+		= para_calloc(sizeof(*compress_conf));
+
+	ret = -E_COMPRESS_SYNTAX;
+	if (compress_cmdline_parser(argc, argv, compress_conf))
+		goto err;
+	*config = compress_conf;
+	return 1;
+err:
+	free(compress_conf);
+	return  ret;
+}
+
+static void open_compress(struct filter_node *fn)
+{
+	struct private_compress_data *pcd = para_calloc(
+		sizeof(struct private_compress_data));
+	pcd->conf = fn->conf;
+	fn->private_data = pcd;
+	fn->bufsize = COMPRESS_CHUNK_SIZE;
+	fn->buf = para_malloc(fn->bufsize);
+	pcd->current_gain = 1 << pcd->conf->inertia_arg;
+	pcd->max_gain = 1 << (pcd->conf->inertia_arg + pcd->conf->aggressiveness_arg);
+}
+
+/**
+ * The init function of the compress filter.
+ *
+ * \param f Pointer to the struct to initialize.
+ */
+void compress_filter_init(struct filter *f)
+{
+	f->open = open_compress;
+	f->close = close_compress;
+	f->convert = compress;
+	f->print_help = compress_cmdline_parser_print_help;
+	f->parse_config = compress_parse_config;
+}
diff --git a/configure.ac b/configure.ac
index cb6ecd65..7cba9450 100644
--- a/configure.ac
+++ b/configure.ac
@@ -81,7 +81,7 @@ AC_CHECK_FUNCS([atexit dup2 memchr memmove memset \
 all_errlist_objs="server mp3_afh afh_common vss command net string signal time
 daemon stat crypt http_send close_on_fork ipc acl afh fade amp_filter
 dccp_send fd user_list chunk_queue afs osl aft mood score attribute blob ringbuffer
-playlist sha1 rbtree sched audiod grab_client filter_chain wav compress
+playlist sha1 rbtree sched audiod grab_client filter_chain wav_filter compress_filter
 http_recv dccp_recv recv_common write_common file_write audiod_command
 client_common recv stdout filter stdin audioc write client fsck exec send_common"
 
@@ -96,7 +96,7 @@ receivers=" http dccp"
 senders=" http dccp"
 
 filter_cmdline_objs="filter.cmdline compress_filter.cmdline amp_filter.cmdline"
-filter_errlist_objs="filter_chain wav compress filter string stdin stdout sched fd amp_filter"
+filter_errlist_objs="filter_chain wav_filter compress_filter filter string stdin stdout sched fd amp_filter"
 filter_ldflags=""
 filters=" compress wav amp"
 
@@ -108,7 +108,7 @@ audiod_cmdline_objs="audiod.cmdline grab_client.cmdline compress_filter.cmdline
 	http_recv.cmdline dccp_recv.cmdline file_write.cmdline client.cmdline
 	audiod_command_list amp_filter.cmdline"
 audiod_errlist_objs="audiod signal string daemon stat net
-	time grab_client filter_chain wav compress amp_filter http_recv dccp_recv
+	time grab_client filter_chain wav_filter compress_filter amp_filter http_recv dccp_recv
 	recv_common fd sched write_common file_write audiod_command crypt
 	client_common"
 audiod_ldflags=""
@@ -316,7 +316,7 @@ AC_CHECK_LIB([ogg], [ogg_stream_init], [], [ have_ogg="no" ])
 AC_CHECK_LIB([vorbis], [vorbis_info_init], [], [ have_ogg="no" ])
 AC_CHECK_HEADERS([ogg/ogg.h vorbis/codec.h], [], [ have_ogg="no" ])
 if test "$have_ogg" = "yes"; then
-	all_errlist_objs="$all_errlist_objs oggdec ogg_afh"
+	all_errlist_objs="$all_errlist_objs oggdec_filter ogg_afh"
 	AC_DEFINE(HAVE_OGGVORBIS, 1, define to 1 to turn on ogg vorbis support)
 	filters="$filters oggdec"
 	if test "$OSTYPE" = "Darwin"; then
@@ -331,8 +331,8 @@ if test "$have_ogg" = "yes"; then
 	audiod_cmdline_objs="$audiod_cmdline_objs oggdec_filter.cmdline"
 
 	server_errlist_objs="$server_errlist_objs ogg_afh"
-	filter_errlist_objs="$filter_errlist_objs oggdec"
-	audiod_errlist_objs="$audiod_errlist_objs oggdec"
+	filter_errlist_objs="$filter_errlist_objs oggdec_filter"
+	audiod_errlist_objs="$audiod_errlist_objs oggdec_filter"
 	afh_errlist_objs="$afh_errlist_objs ogg_afh"
 
 	audiod_audio_formats="ogg"
@@ -366,10 +366,10 @@ AC_CHECK_HEADER(neaacdec.h, [], have_faad=no)
 AC_CHECK_LIB([faad], [NeAACDecOpen], [], have_faad=no)
 if test "$have_faad" = "yes"; then
 	AC_DEFINE(HAVE_FAAD, 1, define to 1 if you want to build the aacdec filter)
-	all_errlist_objs="$all_errlist_objs aac_common aacdec aac_afh"
-	filter_errlist_objs="$filter_errlist_objs aacdec aac_common"
+	all_errlist_objs="$all_errlist_objs aac_common aacdec_filter aac_afh"
+	filter_errlist_objs="$filter_errlist_objs aacdec_filter aac_common"
 	afh_errlist_objs="$afh_errlist_objs aac_common aac_afh"
-	audiod_errlist_objs="$audiod_errlist_objs aacdec aac_common"
+	audiod_errlist_objs="$audiod_errlist_objs aacdec_filter aac_common"
 	server_errlist_objs="$server_errlist_objs aac_afh aac_common"
 	server_ldflags="$server_ldflags $faad_libs -lfaad"
 	filter_ldflags="$filter_ldflags $faad_libs -lfaad"
@@ -412,9 +412,9 @@ AC_CHECK_LIB([mad], [mad_stream_init], [], [
 ])
 if test "$have_mad" = "yes"; then
 	AC_DEFINE(HAVE_MAD, 1, define to 1 if you want to build the mp3dec filter)
-	all_errlist_objs="$all_errlist_objs mp3dec"
-	filter_errlist_objs="$filter_errlist_objs mp3dec"
-	audiod_errlist_objs="$audiod_errlist_objs mp3dec"
+	all_errlist_objs="$all_errlist_objs mp3dec_filter"
+	filter_errlist_objs="$filter_errlist_objs mp3dec_filter"
+	audiod_errlist_objs="$audiod_errlist_objs mp3dec_filter"
 	filter_ldflags="$filter_ldflags $mad_libs -lmad"
 	audiod_ldflags="$audiod_ldflags $mad_libs -lmad"
 	audiod_audio_formats="$audiod_audio_formats mp3"
diff --git a/error.h b/error.h
index cacc83e9..92aaff5d 100644
--- a/error.h
+++ b/error.h
@@ -13,7 +13,7 @@ DEFINE_ERRLIST_OBJECT_ENUM;
 
 /* these do not need error handling (yet) */
 #define SERVER_ERRORS
-#define WAV_ERRORS
+#define WAV_FILTER_ERRORS
 #define TIME_ERRORS
 #define CLOSE_ON_FORK_ERRORS
 #define DAEMON_ERRORS
@@ -32,7 +32,7 @@ DEFINE_ERRLIST_OBJECT_ENUM;
 
 extern const char **para_errlist[];
 
-#define COMPRESS_ERRORS \
+#define COMPRESS_FILTER_ERRORS \
 	PARA_ERROR(COMPRESS_SYNTAX, "syntax error in compress filter config"), \
 
 
@@ -251,7 +251,7 @@ extern const char **para_errlist[];
 	PARA_ERROR(UNKNOWN_STAT_ITEM, "status item not recognized"), \
 
 
-#define OGGDEC_ERRORS \
+#define OGGDEC_FILTER_ERRORS \
 	PARA_ERROR(OGGDEC_READ, "read from media returned an error"), \
 	PARA_ERROR(OGGDEC_NOTVORBIS, "bitstream is not vorbis data"), \
 	PARA_ERROR(OGGDEC_VERSION, "vorbis version mismatch"), \
@@ -271,7 +271,7 @@ extern const char **para_errlist[];
 	PARA_ERROR(GC_VERSION_GIVEN, ""), /* not really an error */ \
 
 
-#define MP3DEC_ERRORS \
+#define MP3DEC_FILTER_ERRORS \
 	PARA_ERROR(MAD_FRAME_DECODE, "mad frame decode error"), \
 	PARA_ERROR(MP3DEC_OVERRUN, "mp3 output buffer overrun"), \
 
@@ -398,7 +398,7 @@ extern const char **para_errlist[];
 	PARA_ERROR(WRITE_COMMON_SYNTAX, "syntax error in write option"), \
 
 
-#define AACDEC_ERRORS \
+#define AACDEC_FILTER_ERRORS \
 	PARA_ERROR(AACDEC_INIT, "failed to init aac decoder"), \
 	PARA_ERROR(AAC_DECODE, "aac decode error"), \
 	PARA_ERROR(AAC_OVERRUN, "aac output buffer overrun"), \
diff --git a/filter.h b/filter.h
index 2a75b186..3425ae85 100644
--- a/filter.h
+++ b/filter.h
@@ -143,7 +143,7 @@ struct filter_callback {
  * Note: As several instances of the same filter may be running at the same
  * time, all these filter functions must be reentrant; no static non-constant
  * variables may be used.
- * \sa mp3dec.c, oggdec.c, wav.c, compress.c, filter_node
+ * \sa mp3dec_filter.c, oggdec_filter.c, wav_filter.c, compress_filter.c, filter_node
  */
 struct filter {
 	/** The name of the filter. */
diff --git a/mp3dec.c b/mp3dec.c
deleted file mode 100644
index 599d8a90..00000000
--- a/mp3dec.c
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
- *
- * Licensed under the GPL v2. For licencing details see COPYING.
- */
-
-/** \file mp3dec.c Paraslash's mp3 decoder. */
-
-#include "para.h"
-#include "list.h"
-#include "sched.h"
-#include "filter.h"
-#include "error.h"
-#include <mad.h>
-#include "string.h"
-
-/** The output buffer size. */
-#define MP3_OUTBUF_SIZE (128 * 1024)
-
-/** Convert a sample value from libmad to a signed short. */
-#define MAD_TO_SHORT(f) (f) >= MAD_F_ONE? SHRT_MAX :\
-	(f) <= -MAD_F_ONE? -SHRT_MAX : (signed short) ((f) >> (MAD_F_FRACBITS - 15))
-
-/** Data specific to the mp3dec filter. */
-struct private_mp3dec_data {
-	/** Information on the current mp3 stream. */
-	struct mad_stream stream;
-	/** Information about the frame which is currently decoded. */
-	struct mad_frame frame;
-	/** Contains the PCM output. */
-	struct mad_synth synth;
-};
-
-static ssize_t mp3dec(char *inbuffer, size_t len, struct filter_node *fn)
-{
-	int i, ret;
-	struct private_mp3dec_data *pmd = fn->private_data;
-	size_t copy = PARA_MIN(len, 4096);
-
-	if (fn->loaded > fn->bufsize * 4 / 5)
-		return 0;
-	mad_stream_buffer(&pmd->stream, (unsigned char *) inbuffer, copy);
-	pmd->stream.error = 0;
-next_frame:
-	ret = mad_header_decode(&pmd->frame.header, &pmd->stream);
-	if (ret < 0) {
-		if (pmd->stream.error != MAD_ERROR_BUFLEN &&
-			pmd->stream.error != MAD_ERROR_LOSTSYNC)
-			PARA_DEBUG_LOG("header decode: %s\n",
-				mad_stream_errorstr(&pmd->stream));
-		goto out;
-	}
-	fn->fc->samplerate = pmd->frame.header.samplerate;
-	fn->fc->channels = MAD_NCHANNELS(&pmd->frame.header);
-	ret = mad_frame_decode(&pmd->frame, &pmd->stream);
-	if (ret) {
-		if (MAD_RECOVERABLE(pmd->stream.error) ||
-			pmd->stream.error == MAD_ERROR_BUFLEN) {
-			PARA_DEBUG_LOG("frame decode: %s\n",
-				mad_stream_errorstr(&pmd->stream));
-			goto out;
-		}
-		PARA_ERROR_LOG("frame decode: %s\n",
-			mad_stream_errorstr(&pmd->stream));
-		return -E_MAD_FRAME_DECODE;
-	}
-	mad_synth_frame(&pmd->synth, &pmd->frame);
-
-	for (i = 0; i < pmd->synth.pcm.length; i++) {
-		int s = MAD_TO_SHORT(pmd->synth.pcm.samples[0][i]);
-		write_int16_host_endian(fn->buf + fn->loaded, s);
-		fn->loaded += 2;
-		if (MAD_NCHANNELS(&pmd->frame.header) == 2) { /* stereo */
-			s = MAD_TO_SHORT(pmd->synth.pcm.samples[1][i]);
-			write_int16_host_endian(fn->buf + fn->loaded, s);
-			fn->loaded += 2;
-		}
-		if (fn->loaded != fn->bufsize) /* output buffer not full */
-			continue;
-		PARA_ERROR_LOG("output buffer full: %zd\n", fn->loaded);
-			return -E_MP3DEC_OVERRUN;
-	}
-	if (fn->loaded <= fn->bufsize * 4 / 5)
-		goto next_frame;
-out:
-	if (pmd->stream.next_frame) { /* we still have some data */
-		size_t off = pmd->stream.bufend - pmd->stream.next_frame;
-//		PARA_INFO_LOG("off: %zd, rate: %u, returning %zd\n", off,
-//			fn->fc->samplerate, copy - off);
-		return copy - off;
-	}
-	return copy;
-}
-
-static void mp3dec_close(struct filter_node *fn)
-{
-	struct private_mp3dec_data *pmd = fn->private_data;
-
-	mad_synth_finish(&pmd->synth);
-	mad_frame_finish(&pmd->frame);
-	mad_stream_finish(&pmd->stream);
-
-	free(fn->buf);
-	fn->buf = NULL;
-	free(pmd);
-	fn->private_data = NULL;
-}
-
-static void mp3dec_open(struct filter_node *fn)
-{
-	struct private_mp3dec_data *pmd = para_calloc(sizeof(*pmd));
-
-	fn->private_data = pmd;
-	mad_stream_init(&pmd->stream);
-	mad_frame_init(&pmd->frame);
-	mad_synth_init(&pmd->synth);
-	fn->loaded = 0;
-	fn->bufsize = MP3_OUTBUF_SIZE;
-	fn->buf = para_calloc(fn->bufsize);
-}
-
-/**
- * The init function of the mp3dec filter.
- *
- * \param f Pointer to the filter struct to initialize.
- *
- * \sa filter::init.
- */
-void mp3dec_filter_init(struct filter *f)
-{
-	f->open = mp3dec_open;
-	f->convert = mp3dec;
-	f->close = mp3dec_close;
-}
diff --git a/mp3dec_filter.c b/mp3dec_filter.c
new file mode 100644
index 00000000..50e530e1
--- /dev/null
+++ b/mp3dec_filter.c
@@ -0,0 +1,134 @@
+/*
+ * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
+ *
+ * Licensed under the GPL v2. For licencing details see COPYING.
+ */
+
+/** \file mp3dec_filter.c Paraslash's mp3 decoder. */
+
+#include "para.h"
+#include "list.h"
+#include "sched.h"
+#include "filter.h"
+#include "error.h"
+#include <mad.h>
+#include "string.h"
+
+/** The output buffer size. */
+#define MP3_OUTBUF_SIZE (128 * 1024)
+
+/** Convert a sample value from libmad to a signed short. */
+#define MAD_TO_SHORT(f) (f) >= MAD_F_ONE? SHRT_MAX :\
+	(f) <= -MAD_F_ONE? -SHRT_MAX : (signed short) ((f) >> (MAD_F_FRACBITS - 15))
+
+/** Data specific to the mp3dec filter. */
+struct private_mp3dec_data {
+	/** Information on the current mp3 stream. */
+	struct mad_stream stream;
+	/** Information about the frame which is currently decoded. */
+	struct mad_frame frame;
+	/** Contains the PCM output. */
+	struct mad_synth synth;
+};
+
+static ssize_t mp3dec(char *inbuffer, size_t len, struct filter_node *fn)
+{
+	int i, ret;
+	struct private_mp3dec_data *pmd = fn->private_data;
+	size_t copy = PARA_MIN(len, 4096);
+
+	if (fn->loaded > fn->bufsize * 4 / 5)
+		return 0;
+	mad_stream_buffer(&pmd->stream, (unsigned char *) inbuffer, copy);
+	pmd->stream.error = 0;
+next_frame:
+	ret = mad_header_decode(&pmd->frame.header, &pmd->stream);
+	if (ret < 0) {
+		if (pmd->stream.error != MAD_ERROR_BUFLEN &&
+			pmd->stream.error != MAD_ERROR_LOSTSYNC)
+			PARA_DEBUG_LOG("header decode: %s\n",
+				mad_stream_errorstr(&pmd->stream));
+		goto out;
+	}
+	fn->fc->samplerate = pmd->frame.header.samplerate;
+	fn->fc->channels = MAD_NCHANNELS(&pmd->frame.header);
+	ret = mad_frame_decode(&pmd->frame, &pmd->stream);
+	if (ret) {
+		if (MAD_RECOVERABLE(pmd->stream.error) ||
+			pmd->stream.error == MAD_ERROR_BUFLEN) {
+			PARA_DEBUG_LOG("frame decode: %s\n",
+				mad_stream_errorstr(&pmd->stream));
+			goto out;
+		}
+		PARA_ERROR_LOG("frame decode: %s\n",
+			mad_stream_errorstr(&pmd->stream));
+		return -E_MAD_FRAME_DECODE;
+	}
+	mad_synth_frame(&pmd->synth, &pmd->frame);
+
+	for (i = 0; i < pmd->synth.pcm.length; i++) {
+		int s = MAD_TO_SHORT(pmd->synth.pcm.samples[0][i]);
+		write_int16_host_endian(fn->buf + fn->loaded, s);
+		fn->loaded += 2;
+		if (MAD_NCHANNELS(&pmd->frame.header) == 2) { /* stereo */
+			s = MAD_TO_SHORT(pmd->synth.pcm.samples[1][i]);
+			write_int16_host_endian(fn->buf + fn->loaded, s);
+			fn->loaded += 2;
+		}
+		if (fn->loaded != fn->bufsize) /* output buffer not full */
+			continue;
+		PARA_ERROR_LOG("output buffer full: %zd\n", fn->loaded);
+			return -E_MP3DEC_OVERRUN;
+	}
+	if (fn->loaded <= fn->bufsize * 4 / 5)
+		goto next_frame;
+out:
+	if (pmd->stream.next_frame) { /* we still have some data */
+		size_t off = pmd->stream.bufend - pmd->stream.next_frame;
+//		PARA_INFO_LOG("off: %zd, rate: %u, returning %zd\n", off,
+//			fn->fc->samplerate, copy - off);
+		return copy - off;
+	}
+	return copy;
+}
+
+static void mp3dec_close(struct filter_node *fn)
+{
+	struct private_mp3dec_data *pmd = fn->private_data;
+
+	mad_synth_finish(&pmd->synth);
+	mad_frame_finish(&pmd->frame);
+	mad_stream_finish(&pmd->stream);
+
+	free(fn->buf);
+	fn->buf = NULL;
+	free(pmd);
+	fn->private_data = NULL;
+}
+
+static void mp3dec_open(struct filter_node *fn)
+{
+	struct private_mp3dec_data *pmd = para_calloc(sizeof(*pmd));
+
+	fn->private_data = pmd;
+	mad_stream_init(&pmd->stream);
+	mad_frame_init(&pmd->frame);
+	mad_synth_init(&pmd->synth);
+	fn->loaded = 0;
+	fn->bufsize = MP3_OUTBUF_SIZE;
+	fn->buf = para_calloc(fn->bufsize);
+}
+
+/**
+ * The init function of the mp3dec filter.
+ *
+ * \param f Pointer to the filter struct to initialize.
+ *
+ * \sa filter::init.
+ */
+void mp3dec_filter_init(struct filter *f)
+{
+	f->open = mp3dec_open;
+	f->convert = mp3dec;
+	f->close = mp3dec_close;
+}
diff --git a/oggdec.c b/oggdec.c
deleted file mode 100644
index b4befd01..00000000
--- a/oggdec.c
+++ /dev/null
@@ -1,203 +0,0 @@
-/*
- * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
- *
- * Licensed under the GPL v2. For licencing details see COPYING.
- */
-
-/** \file oggdec.c Paraslash's ogg vorbis decoder. */
-
-#include "para.h"
-
-#include "oggdec_filter.cmdline.h"
-#include "list.h"
-#include "sched.h"
-#include "filter.h"
-#include "error.h"
-#include "string.h"
-
-#include <vorbis/vorbisfile.h>
-
-/** Determine byte sex. */
-#ifdef WORDS_BIGENDIAN
-#define ENDIAN 1
-#else
-#define ENDIAN 0
-#endif
-
-/** Data specific to the oggdec filter. */
-struct private_oggdec_data {
-	/** Describes an ogg vorbis file. */
-	OggVorbis_File *vf;
-	/** The input buffer. */
-	char *inbuf;
-	/** The length of \a inbuf. */
-	size_t inbuf_len;
-	/** The number of bytes consumed from the input buffer. */
-	size_t converted;
-};
-
-static size_t cb_read(void *buf, size_t size, size_t nmemb, void *datasource)
-{
-	struct filter_node *fn = datasource;
-	struct private_oggdec_data *pod = fn->private_data;
-	size_t ret, have = pod->inbuf_len - pod->converted;
-	char *p = pod->inbuf + pod->converted;
-
-//	PARA_DEBUG_LOG("pod = %p\n", pod);
-//	PARA_DEBUG_LOG("vorbis requests %d bytes, have %d\n", size * nmemb, have);
-	if (pod->inbuf_len < size) {
-		if (*fn->fc->input_error)
-			return 0;
-		errno = EAGAIN;
-		return (size_t)-1;
-	}
-	ret = PARA_MIN(nmemb, have / size) * size;
-	memcpy(buf, p, ret);
-	pod->converted += ret;
-	return ret;
-}
-
-/*
- * Custom data seeking function.
- *
- * Since we want the data source to be treated as unseekable at all
- * times, the provided seek callback always returns -1 (failure).
- */
-static int cb_seek(__a_unused void *datasource, __a_unused ogg_int64_t offset,
-		__a_unused int whence)
-{
-	return -1;
-}
-
-static int cb_close(__a_unused void *datasource)
-{
-	return 0;
-}
-
-static const ov_callbacks ovc = {
-	.read_func = cb_read,
-	.seek_func = cb_seek,
-	.close_func = cb_close,
-	/*
-	 * The tell function need not be provided if the data IO abstraction is
-	 * not seekable
-	 */
-	.tell_func = NULL
-};
-
-static void ogg_open(struct filter_node *fn)
-{
-	struct private_oggdec_data *pod = para_calloc(
-		sizeof(struct private_oggdec_data));
-	struct oggdec_filter_args_info *conf = fn->conf;
-
-	fn->private_data = pod;
-	fn->bufsize = conf->bufsize_arg * 1024;
-	fn->buf = para_malloc(fn->bufsize);
-}
-
-static void ogg_close(struct filter_node *fn)
-{
-	struct private_oggdec_data *pod = fn->private_data;
-	if (pod->vf) {
-		PARA_DEBUG_LOG("ov_clearing %p, pod = %p\n", pod->vf, pod);
-		ov_clear(pod->vf);
-		free(pod->vf);
-		pod->vf = NULL;
-	} else
-		PARA_DEBUG_LOG("nothing to close in fc %p, pod = %p\n", pod->vf, pod);
-	free(fn->buf);
-	fn->buf = NULL;
-	free(fn->private_data);
-	fn->private_data = NULL;
-}
-
-static ssize_t ogg_convert(char *inbuffer, size_t len, struct filter_node *fn)
-{
-	ssize_t ret;
-	struct private_oggdec_data *pod = fn->private_data;
-	struct oggdec_filter_args_info *conf = fn->conf;
-	/* make the buffer known to the read callback cb_read() */
-	pod->inbuf = inbuffer;
-	pod->inbuf_len = len;
-	pod->converted = 0;
-
-	if (!pod->vf) {
-		int ib = 1024 * conf->initial_buffer_arg; /* initial buffer */
-		if (len <ib && !*fn->fc->input_error) {
-			PARA_DEBUG_LOG("initial input buffer %zd/%d, "
-				"waiting for more data\n", len, ib);
-			return 0;
-		}
-		pod->vf = para_malloc(sizeof(struct OggVorbis_File));
-		PARA_NOTICE_LOG("input buffer: %zd, opening ov callbacks\n", len);
-		ret = ov_open_callbacks(fn, pod->vf,
-			NULL, /* no initial buffer */
-			0, /* no initial bytes */
-			ovc); /* the ov_open_callbacks */
-		if (ret == OV_EREAD)
-			return -E_OGGDEC_READ;
-		if (ret == OV_ENOTVORBIS)
-			return -E_OGGDEC_NOTVORBIS;
-		if (ret == OV_EVERSION)
-			return -E_OGGDEC_VERSION;
-		if (ret == OV_EBADHEADER)
-			return -E_OGGDEC_BADHEADER;
-		if (ret < 0)
-			return -E_OGGDEC_FAULT;
-		fn->fc->channels = ov_info(pod->vf, 0)->channels;
-		fn->fc->samplerate = ov_info(pod->vf, 0)->rate;
-		PARA_NOTICE_LOG("%d channels, %d Hz\n", fn->fc->channels,
-			fn->fc->samplerate);
-	}
-	while (fn->loaded < fn->bufsize) {
-		int length = fn->bufsize - fn->loaded;
-		long read_ret = ov_read(pod->vf, fn->buf + fn->loaded, length,
-			ENDIAN, 2 /* 16 bit */, 1 /* signed */, NULL);
-		if (read_ret == OV_HOLE || !read_ret)
-			return pod->converted;
-		if (read_ret < 0)
-			return -E_OGGDEC_BADLINK;
-		fn->loaded += read_ret;
-	}
-	return pod->converted;
-}
-
-static int oggdec_parse_config(int argc, char **argv, void **config)
-{
-	int ret;
-	struct oggdec_filter_args_info *ogg_conf;
-
-	ogg_conf = para_calloc(sizeof(*ogg_conf));
-	ret = -E_OGGDEC_SYNTAX;
-	if (oggdec_cmdline_parser(argc, argv, ogg_conf))
-		goto err;
-	ret = -ERRNO_TO_PARA_ERROR(EINVAL);
-	if (ogg_conf->bufsize_arg < 0)
-		goto err;
-	if (ogg_conf->bufsize_arg >= INT_MAX / 1024)
-		goto err;
-	if (ogg_conf->initial_buffer_arg < 0)
-		goto err;
-	if (ogg_conf->initial_buffer_arg >= INT_MAX / 1024)
-		goto err;
-	*config = ogg_conf;
-	return 1;
-err:
-	free(ogg_conf);
-	return ret;
-}
-
-/**
- * The init function of the ogg vorbis decoder.
- *
- * \param f Its fields are filled in by the function.
- */
-void oggdec_filter_init(struct filter *f)
-{
-	f->open = ogg_open;
-	f->close = ogg_close;
-	f->convert = ogg_convert;
-	f->print_help = oggdec_cmdline_parser_print_help;
-	f->parse_config = oggdec_parse_config;
-}
diff --git a/oggdec_filter.c b/oggdec_filter.c
new file mode 100644
index 00000000..689fc7aa
--- /dev/null
+++ b/oggdec_filter.c
@@ -0,0 +1,203 @@
+/*
+ * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
+ *
+ * Licensed under the GPL v2. For licencing details see COPYING.
+ */
+
+/** \file oggdec_filter.c Paraslash's ogg vorbis decoder. */
+
+#include "para.h"
+
+#include "oggdec_filter.cmdline.h"
+#include "list.h"
+#include "sched.h"
+#include "filter.h"
+#include "error.h"
+#include "string.h"
+
+#include <vorbis/vorbisfile.h>
+
+/** Determine byte sex. */
+#ifdef WORDS_BIGENDIAN
+#define ENDIAN 1
+#else
+#define ENDIAN 0
+#endif
+
+/** Data specific to the oggdec filter. */
+struct private_oggdec_data {
+	/** Describes an ogg vorbis file. */
+	OggVorbis_File *vf;
+	/** The input buffer. */
+	char *inbuf;
+	/** The length of \a inbuf. */
+	size_t inbuf_len;
+	/** The number of bytes consumed from the input buffer. */
+	size_t converted;
+};
+
+static size_t cb_read(void *buf, size_t size, size_t nmemb, void *datasource)
+{
+	struct filter_node *fn = datasource;
+	struct private_oggdec_data *pod = fn->private_data;
+	size_t ret, have = pod->inbuf_len - pod->converted;
+	char *p = pod->inbuf + pod->converted;
+
+//	PARA_DEBUG_LOG("pod = %p\n", pod);
+//	PARA_DEBUG_LOG("vorbis requests %d bytes, have %d\n", size * nmemb, have);
+	if (pod->inbuf_len < size) {
+		if (*fn->fc->input_error)
+			return 0;
+		errno = EAGAIN;
+		return (size_t)-1;
+	}
+	ret = PARA_MIN(nmemb, have / size) * size;
+	memcpy(buf, p, ret);
+	pod->converted += ret;
+	return ret;
+}
+
+/*
+ * Custom data seeking function.
+ *
+ * Since we want the data source to be treated as unseekable at all
+ * times, the provided seek callback always returns -1 (failure).
+ */
+static int cb_seek(__a_unused void *datasource, __a_unused ogg_int64_t offset,
+		__a_unused int whence)
+{
+	return -1;
+}
+
+static int cb_close(__a_unused void *datasource)
+{
+	return 0;
+}
+
+static const ov_callbacks ovc = {
+	.read_func = cb_read,
+	.seek_func = cb_seek,
+	.close_func = cb_close,
+	/*
+	 * The tell function need not be provided if the data IO abstraction is
+	 * not seekable
+	 */
+	.tell_func = NULL
+};
+
+static void ogg_open(struct filter_node *fn)
+{
+	struct private_oggdec_data *pod = para_calloc(
+		sizeof(struct private_oggdec_data));
+	struct oggdec_filter_args_info *conf = fn->conf;
+
+	fn->private_data = pod;
+	fn->bufsize = conf->bufsize_arg * 1024;
+	fn->buf = para_malloc(fn->bufsize);
+}
+
+static void ogg_close(struct filter_node *fn)
+{
+	struct private_oggdec_data *pod = fn->private_data;
+	if (pod->vf) {
+		PARA_DEBUG_LOG("ov_clearing %p, pod = %p\n", pod->vf, pod);
+		ov_clear(pod->vf);
+		free(pod->vf);
+		pod->vf = NULL;
+	} else
+		PARA_DEBUG_LOG("nothing to close in fc %p, pod = %p\n", pod->vf, pod);
+	free(fn->buf);
+	fn->buf = NULL;
+	free(fn->private_data);
+	fn->private_data = NULL;
+}
+
+static ssize_t ogg_convert(char *inbuffer, size_t len, struct filter_node *fn)
+{
+	ssize_t ret;
+	struct private_oggdec_data *pod = fn->private_data;
+	struct oggdec_filter_args_info *conf = fn->conf;
+	/* make the buffer known to the read callback cb_read() */
+	pod->inbuf = inbuffer;
+	pod->inbuf_len = len;
+	pod->converted = 0;
+
+	if (!pod->vf) {
+		int ib = 1024 * conf->initial_buffer_arg; /* initial buffer */
+		if (len <ib && !*fn->fc->input_error) {
+			PARA_DEBUG_LOG("initial input buffer %zd/%d, "
+				"waiting for more data\n", len, ib);
+			return 0;
+		}
+		pod->vf = para_malloc(sizeof(struct OggVorbis_File));
+		PARA_NOTICE_LOG("input buffer: %zd, opening ov callbacks\n", len);
+		ret = ov_open_callbacks(fn, pod->vf,
+			NULL, /* no initial buffer */
+			0, /* no initial bytes */
+			ovc); /* the ov_open_callbacks */
+		if (ret == OV_EREAD)
+			return -E_OGGDEC_READ;
+		if (ret == OV_ENOTVORBIS)
+			return -E_OGGDEC_NOTVORBIS;
+		if (ret == OV_EVERSION)
+			return -E_OGGDEC_VERSION;
+		if (ret == OV_EBADHEADER)
+			return -E_OGGDEC_BADHEADER;
+		if (ret < 0)
+			return -E_OGGDEC_FAULT;
+		fn->fc->channels = ov_info(pod->vf, 0)->channels;
+		fn->fc->samplerate = ov_info(pod->vf, 0)->rate;
+		PARA_NOTICE_LOG("%d channels, %d Hz\n", fn->fc->channels,
+			fn->fc->samplerate);
+	}
+	while (fn->loaded < fn->bufsize) {
+		int length = fn->bufsize - fn->loaded;
+		long read_ret = ov_read(pod->vf, fn->buf + fn->loaded, length,
+			ENDIAN, 2 /* 16 bit */, 1 /* signed */, NULL);
+		if (read_ret == OV_HOLE || !read_ret)
+			return pod->converted;
+		if (read_ret < 0)
+			return -E_OGGDEC_BADLINK;
+		fn->loaded += read_ret;
+	}
+	return pod->converted;
+}
+
+static int oggdec_parse_config(int argc, char **argv, void **config)
+{
+	int ret;
+	struct oggdec_filter_args_info *ogg_conf;
+
+	ogg_conf = para_calloc(sizeof(*ogg_conf));
+	ret = -E_OGGDEC_SYNTAX;
+	if (oggdec_cmdline_parser(argc, argv, ogg_conf))
+		goto err;
+	ret = -ERRNO_TO_PARA_ERROR(EINVAL);
+	if (ogg_conf->bufsize_arg < 0)
+		goto err;
+	if (ogg_conf->bufsize_arg >= INT_MAX / 1024)
+		goto err;
+	if (ogg_conf->initial_buffer_arg < 0)
+		goto err;
+	if (ogg_conf->initial_buffer_arg >= INT_MAX / 1024)
+		goto err;
+	*config = ogg_conf;
+	return 1;
+err:
+	free(ogg_conf);
+	return ret;
+}
+
+/**
+ * The init function of the ogg vorbis decoder.
+ *
+ * \param f Its fields are filled in by the function.
+ */
+void oggdec_filter_init(struct filter *f)
+{
+	f->open = ogg_open;
+	f->close = ogg_close;
+	f->convert = ogg_convert;
+	f->print_help = oggdec_cmdline_parser_print_help;
+	f->parse_config = oggdec_parse_config;
+}
diff --git a/server.c b/server.c
index 91568764..459f4462 100644
--- a/server.c
+++ b/server.c
@@ -23,8 +23,8 @@
  * The gory details, listed by topic:
  *
  * 	- Audio format handlers: \ref send_common.c \ref mp3_afh.c, \ref ogg_afh.c, \ref aac_afh.c,
- * 	- Decoders: \ref mp3dec.c, \ref oggdec.c, \ref aacdec.c,
- *	- Volume normalizer: \ref compress.c,
+ * 	- Decoders: \ref mp3dec_filter.c, \ref oggdec_filter.c, \ref aacdec_filter.c,
+ *	- Volume normalizer: \ref compress_filter.c,
  *	- Output: \ref alsa_write.c, \ref osx_write.c,
  * 	- http: \ref http_recv.c, \ref http_send.c,
  *	- ortp: \ref ortp_recv.c, \ref ortp_send.c,
diff --git a/wav.c b/wav.c
deleted file mode 100644
index ad3458d1..00000000
--- a/wav.c
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
- *
- * Licensed under the GPL v2. For licencing details see COPYING.
- */
-
-/** \file wav.c a filter that inserts a wave header */
-
-#include "para.h"
-
-#include "list.h"
-#include "sched.h"
-#include "filter.h"
-#include "string.h"
-
-/** size of the output buffer */
-#define WAV_OUTBUF_SIZE 81920
-/** a wav header is always 44 bytes */
-#define WAV_HEADER_LEN 44
-/** always write 16 bit header */
-#define BITS 16
-
-/**
- * write a 32 bit unsigned value to a buffer
- *
- * \param buf the buffer to write to
- * \param x the value to write
- */
-#define WRITE_U32(buf, x) (buf)[0] = (unsigned char)((x) & 0xff);\
-	(buf)[1] = (unsigned char)(((x) >> 8) & 0xff);\
-	(buf)[2] = (unsigned char)(((x) >> 16) & 0xff);\
-	(buf)[3] = (unsigned char)(((x) >> 24) & 0xff);
-
-/**
- * write a 16 bit unsigned value to a buffer
- *
- * \param buf the buffer to write to
- * \param x the value to write
- */
-#define WRITE_U16(buf, x) (buf)[0] = (unsigned char)((x) & 0xff);
-
-static void make_wav_header(unsigned int channels, unsigned int samplerate,
-		struct filter_node *fn)
-{
-
-	char *headbuf = fn->buf;
-	unsigned int size = 0x7fffffff;
-	int bytespersec = channels * samplerate * BITS / 8;
-	int align = channels * BITS / 8;
-
-	assert(channels);
-	PARA_DEBUG_LOG("writing wave header: %d channels, %d KHz\n", channels, samplerate);
-	memset(headbuf, 0, WAV_HEADER_LEN);
-	memcpy(headbuf, "RIFF", 4);
-	WRITE_U32(headbuf + 4, size - 8);
-	memcpy(headbuf + 8, "WAVE", 4);
-	memcpy(headbuf + 12, "fmt ", 4);
-	WRITE_U32(headbuf + 16, 16);
-	WRITE_U16(headbuf + 20, 1);	/* format */
-	WRITE_U16(headbuf + 22, channels);
-	WRITE_U32(headbuf + 24, samplerate);
-	WRITE_U32(headbuf + 28, bytespersec);
-	WRITE_U16(headbuf + 32, align);
-	WRITE_U16(headbuf + 34, BITS);
-	memcpy(headbuf + 36, "data", 4);
-	WRITE_U32(headbuf + 40, size - 44);
-}
-
-static ssize_t wav_convert(char *inbuf, size_t len, struct filter_node *fn)
-{
-	size_t copy;
-	int *bof = fn->private_data;
-
-	if (*bof) {
-		if (!len)
-			return 0;
-		make_wav_header(fn->fc->channels, fn->fc->samplerate, fn);
-		fn->loaded = WAV_HEADER_LEN;
-		*bof = 0;
-//		return 0;
-	}
-	copy = PARA_MIN(len, fn->bufsize - fn->loaded);
-	memmove(fn->buf + fn->loaded, inbuf, copy);
-	fn->loaded += copy;
-//	PARA_DEBUG_LOG("len = %d, copy = %d\n", len, copy);
-	return copy;
-}
-
-static void wav_close(struct filter_node *fn)
-{
-	free(fn->buf);
-	fn->buf = NULL;
-	free(fn->private_data);
-	fn->private_data = NULL;
-}
-
-static void wav_open(struct filter_node *fn)
-{
-	int *bof;
-
-	fn->bufsize = WAV_OUTBUF_SIZE;
-	fn->buf = para_malloc(fn->bufsize);
-	fn->private_data = para_malloc(sizeof(int));
-	bof = fn->private_data;
-	fn->loaded = 0;
-	*bof = 1;
-	PARA_INFO_LOG("wav filter node: %p, output buffer: %p, loaded: %zd\n",
-		fn, fn->buf, fn->loaded);
-}
-
-/**
- * the init function of the wav filter
- *
- * \param f struct to initialize
- */
-void wav_filter_init(struct filter *f)
-{
-	f->convert = wav_convert;
-	f->close = wav_close;
-	f->open = wav_open;
-}
diff --git a/wav_filter.c b/wav_filter.c
new file mode 100644
index 00000000..242cb4bb
--- /dev/null
+++ b/wav_filter.c
@@ -0,0 +1,121 @@
+/*
+ * Copyright (C) 2005-2008 Andre Noll <maan@systemlinux.org>
+ *
+ * Licensed under the GPL v2. For licencing details see COPYING.
+ */
+
+/** \file wav_filter.c A filter that inserts a wave header. */
+
+#include "para.h"
+
+#include "list.h"
+#include "sched.h"
+#include "filter.h"
+#include "string.h"
+
+/** size of the output buffer */
+#define WAV_OUTBUF_SIZE 81920
+/** a wav header is always 44 bytes */
+#define WAV_HEADER_LEN 44
+/** always write 16 bit header */
+#define BITS 16
+
+/**
+ * write a 32 bit unsigned value to a buffer
+ *
+ * \param buf the buffer to write to
+ * \param x the value to write
+ */
+#define WRITE_U32(buf, x) (buf)[0] = (unsigned char)((x) & 0xff);\
+	(buf)[1] = (unsigned char)(((x) >> 8) & 0xff);\
+	(buf)[2] = (unsigned char)(((x) >> 16) & 0xff);\
+	(buf)[3] = (unsigned char)(((x) >> 24) & 0xff);
+
+/**
+ * write a 16 bit unsigned value to a buffer
+ *
+ * \param buf the buffer to write to
+ * \param x the value to write
+ */
+#define WRITE_U16(buf, x) (buf)[0] = (unsigned char)((x) & 0xff);
+
+static void make_wav_header(unsigned int channels, unsigned int samplerate,
+		struct filter_node *fn)
+{
+
+	char *headbuf = fn->buf;
+	unsigned int size = 0x7fffffff;
+	int bytespersec = channels * samplerate * BITS / 8;
+	int align = channels * BITS / 8;
+
+	assert(channels);
+	PARA_DEBUG_LOG("writing wave header: %d channels, %d KHz\n", channels, samplerate);
+	memset(headbuf, 0, WAV_HEADER_LEN);
+	memcpy(headbuf, "RIFF", 4);
+	WRITE_U32(headbuf + 4, size - 8);
+	memcpy(headbuf + 8, "WAVE", 4);
+	memcpy(headbuf + 12, "fmt ", 4);
+	WRITE_U32(headbuf + 16, 16);
+	WRITE_U16(headbuf + 20, 1);	/* format */
+	WRITE_U16(headbuf + 22, channels);
+	WRITE_U32(headbuf + 24, samplerate);
+	WRITE_U32(headbuf + 28, bytespersec);
+	WRITE_U16(headbuf + 32, align);
+	WRITE_U16(headbuf + 34, BITS);
+	memcpy(headbuf + 36, "data", 4);
+	WRITE_U32(headbuf + 40, size - 44);
+}
+
+static ssize_t wav_convert(char *inbuf, size_t len, struct filter_node *fn)
+{
+	size_t copy;
+	int *bof = fn->private_data;
+
+	if (*bof) {
+		if (!len)
+			return 0;
+		make_wav_header(fn->fc->channels, fn->fc->samplerate, fn);
+		fn->loaded = WAV_HEADER_LEN;
+		*bof = 0;
+//		return 0;
+	}
+	copy = PARA_MIN(len, fn->bufsize - fn->loaded);
+	memmove(fn->buf + fn->loaded, inbuf, copy);
+	fn->loaded += copy;
+//	PARA_DEBUG_LOG("len = %d, copy = %d\n", len, copy);
+	return copy;
+}
+
+static void wav_close(struct filter_node *fn)
+{
+	free(fn->buf);
+	fn->buf = NULL;
+	free(fn->private_data);
+	fn->private_data = NULL;
+}
+
+static void wav_open(struct filter_node *fn)
+{
+	int *bof;
+
+	fn->bufsize = WAV_OUTBUF_SIZE;
+	fn->buf = para_malloc(fn->bufsize);
+	fn->private_data = para_malloc(sizeof(int));
+	bof = fn->private_data;
+	fn->loaded = 0;
+	*bof = 1;
+	PARA_INFO_LOG("wav filter node: %p, output buffer: %p, loaded: %zd\n",
+		fn, fn->buf, fn->loaded);
+}
+
+/**
+ * the init function of the wav filter
+ *
+ * \param f struct to initialize
+ */
+void wav_filter_init(struct filter *f)
+{
+	f->convert = wav_convert;
+	f->close = wav_close;
+	f->open = wav_open;
+}