patch to split DSD decoder into separate decoders for DSF en DFF. Move common

Makefile.am

@@ -496,6 +496,10 @@ libdecoder_plugins_a_SOURCES = \
 	src/decoder/pcm_decoder_plugin.c \
 	src/decoder/dsdiff_decoder_plugin.c \
 	src/decoder/dsdiff_decoder_plugin.h \
+	src/decoder/dsf_decoder_plugin.c \
+	src/decoder/dsf_decoder_plugin.h \
+	src/decoder/dsdlib.c \
+	src/decoder/dsdlib.h \
 	src/decoder_buffer.c \
 	src/decoder_plugin.c \
 	src/decoder_list.c

doc/user.xml

@@ -782,7 +782,7 @@ systemctl start mpd.socket</programlisting>
         <title><varname>dsdiff</varname></title>
 
         <para>
-          Decodes DFF and DSF files containing DSDIFF data (e.g. SACD rips).
+          Decodes DFF files containing DSDIFF data (e.g. SACD rips).
         </para>
 
         <informaltable>
@@ -810,6 +810,15 @@ systemctl start mpd.socket</programlisting>
       </section>
 
       <section>
+        <title><varname>dsf</varname></title>
+
+        <para>
+          Decodes DSF files containing DSDIFF data (e.g. SACD rips).
+        </para>
+
+      </section>
+
+      <section>
         <title><varname>mikmod</varname></title>
 
         <para>

src/decoder/dsdiff_decoder_plugin.h

 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2012 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify

src/decoder/dsdlib.c

+/*
+ * Copyright (C) 2012 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.
+ */
+
+/* \file
+ *
+ * This file contains functions used by the DSF and DSDIFF decoders.
+ *
+ */
+
+#include "config.h"
+#include "dsf_decoder_plugin.h"
+#include "decoder_api.h"
+#include "util/bit_reverse.h"
+#include "dsdlib.h"
+#include "dsdiff_decoder_plugin.h"
+
+#include <unistd.h>
+#include <stdio.h> /* for SEEK_SET, SEEK_CUR */
+
+bool
+dsdlib_id_equals(const struct dsdlib_id *id, const char *s)
+{
+	assert(id != NULL);
+	assert(s != NULL);
+	assert(strlen(s) == sizeof(id->value));
+
+	return memcmp(id->value, s, sizeof(id->value)) == 0;
+}
+
+bool
+dsdlib_read(struct decoder *decoder, struct input_stream *is,
+	    void *data, size_t length)
+{
+	size_t nbytes = decoder_read(decoder, is, data, length);
+	return nbytes == length;
+}
+
+/**
+ * Skip the #input_stream to the specified offset.
+ */
+bool
+dsdlib_skip_to(struct decoder *decoder, struct input_stream *is,
+	       goffset offset)
+{
+	if (is->seekable)
+		return input_stream_seek(is, offset, SEEK_SET, NULL);
+
+	if (is->offset > offset)
+		return false;
+
+	char buffer[8192];
+	while (is->offset < offset) {
+		size_t length = sizeof(buffer);
+		if (offset - is->offset < (goffset)length)
+			length = offset - is->offset;
+
+		size_t nbytes = decoder_read(decoder, is, buffer, length);
+		if (nbytes == 0)
+			return false;
+	}
+
+	assert(is->offset == offset);
+	return true;
+}
+
+/**
+ * Skip some bytes from the #input_stream.
+ */
+bool
+dsdlib_skip(struct decoder *decoder, struct input_stream *is,
+	    goffset delta)
+{
+	assert(delta >= 0);
+
+	if (delta == 0)
+		return true;
+
+	if (is->seekable)
+		return input_stream_seek(is, delta, SEEK_CUR, NULL);
+
+	char buffer[8192];
+	while (delta > 0) {
+		size_t length = sizeof(buffer);
+		if ((goffset)length > delta)
+			length = delta;
+
+		size_t nbytes = decoder_read(decoder, is, buffer, length);
+		if (nbytes == 0)
+			return false;
+
+		delta -= nbytes;
+	}
+
+	return true;
+}
+

src/decoder/dsdlib.h

+/*
+ * Copyright (C) 2012 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_DECODER_DSDLIB_H
+#define MPD_DECODER_DSDLIB_H
+
+struct dsdlib_id {
+	char value[4];
+};
+
+bool
+dsdlib_id_equals(const struct dsdlib_id *id, const char *s);
+
+bool
+dsdlib_read(struct decoder *decoder, struct input_stream *is,
+	    void *data, size_t length);
+
+bool
+dsdlib_skip_to(struct decoder *decoder, struct input_stream *is,
+	       goffset offset);
+
+bool
+dsdlib_skip(struct decoder *decoder, struct input_stream *is,
+	    goffset delta);
+
+#endif

src/decoder/dsf_decoder_plugin.c

+/*
+ * Copyright (C) 2012 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.
+ */
+
+/* \file
+ *
+ * This plugin decodes DSDIFF data (SACD) embedded in DSF files.
+ *
+ * The DSF code was created using the specification found here:
+ * http://dsd-guide.com/sonys-dsf-file-format-spec
+ *
+ * All functions common to both DSD decoders have been moved to dsdlib
+ */
+
+#include "config.h"
+#include "dsf_decoder_plugin.h"
+#include "decoder_api.h"
+#include "audio_check.h"
+#include "util/bit_reverse.h"
+#include "dsdlib.h"
+
+#include <unistd.h>
+#include <stdio.h> /* for SEEK_SET, SEEK_CUR */
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "dsf"
+
+struct dsf_metadata {
+	unsigned sample_rate, channels;
+	bool bitreverse;
+	uint64_t chunk_size;
+};
+
+struct dsf_header {
+	/** DSF header id: "DSD " */
+	struct dsdlib_id id;
+	/** DSD chunk size, including id = 28 */
+	uint32_t size_low, size_high;
+	/** total file size */
+	uint32_t fsize_low, fsize_high;
+	/** pointer to id3v2 metadata, should be at the end of the file */
+	uint32_t pmeta_low, pmeta_high;
+};
+/** DSF file fmt chunk */
+struct dsf_fmt_chunk {
+
+	/** id: "fmt " */
+	struct dsdlib_id id;
+	/** fmt chunk size, including id, normally 52 */
+	uint32_t size_low, size_high;
+	/** version of this format = 1 */
+	uint32_t version;
+	/** 0: DSD raw */
+	uint32_t formatid;
+	/** channel type, 1 = mono, 2 = stereo, 3 = 3 channels, etc */
+	uint32_t channeltype;
+	/** Channel number, 1 = mono, 2 = stereo, ... 6 = 6 channels */
+	uint32_t channelnum;
+	/** sample frequency: 2822400, 5644800 */
+	uint32_t sample_freq;
+	/** bits per sample 1 or 8 */
+	uint32_t bitssample;
+	/** Sample count per channel in bytes */
+	uint32_t scnt_low, scnt_high;
+	/** block size per channel = 4096 */
+	uint32_t block_size;
+	/** reserved, should be all zero */
+	uint32_t reserved;
+};
+
+struct dsf_data_chunk {
+	struct dsdlib_id id;
+	/** "data" chunk size, includes header (id+size) */
+	uint32_t size_low, size_high;
+};
+
+/**
+ * Read and parse all needed metadata chunks for DSF files.
+ */
+static bool
+dsf_read_metadata(struct decoder *decoder, struct input_stream *is,
+		  struct dsf_metadata *metadata)
+{
+	uint64_t chunk_size;
+	struct dsf_header dsf_header;
+	if (!dsdlib_read(decoder, is, &dsf_header, sizeof(dsf_header)) ||
+	    !dsdlib_id_equals(&dsf_header.id, "DSD "))
+		return false;
+
+	chunk_size = (((uint64_t)GUINT32_FROM_LE(dsf_header.size_high)) << 32) |
+		((uint64_t)GUINT32_FROM_LE(dsf_header.size_low));
+
+	if (sizeof(dsf_header) != chunk_size)
+		return false;
+
+	/* read the 'fmt ' chunk of the DSF file */
+	struct dsf_fmt_chunk dsf_fmt_chunk;
+	if (!dsdlib_read(decoder, is, &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) ||
+	    !dsdlib_id_equals(&dsf_fmt_chunk.id, "fmt "))
+		return false;
+
+	uint64_t fmt_chunk_size;
+	fmt_chunk_size = (((uint64_t)GUINT32_FROM_LE(dsf_fmt_chunk.size_high)) << 32) |
+			  ((uint64_t)GUINT32_FROM_LE(dsf_fmt_chunk.size_low));
+
+	if (fmt_chunk_size != sizeof(dsf_fmt_chunk))
+		return false;
+
+	uint32_t samplefreq = (uint32_t)GUINT32_FROM_LE(dsf_fmt_chunk.sample_freq);
+
+	/* for now, only support version 1 of the standard, DSD raw stereo
+	   files with a sample freq of 2822400 Hz */
+
+	if (dsf_fmt_chunk.version != 1 || dsf_fmt_chunk.formatid != 0
+	    || dsf_fmt_chunk.channeltype != 2
+	    || dsf_fmt_chunk.channelnum != 2
+	    || samplefreq != 2822400)
+		return false;
+
+	uint32_t chblksize = (uint32_t)GUINT32_FROM_LE(dsf_fmt_chunk.block_size);
+	/* according to the spec block size should always be 4096 */
+	if (chblksize != 4096)
+		return false;
+
+	/* read the 'data' chunk of the DSF file */
+	struct dsf_data_chunk data_chunk;
+	if (!dsdlib_read(decoder, is, &data_chunk, sizeof(data_chunk)) ||
+	    !dsdlib_id_equals(&data_chunk.id, "data"))
+		return false;
+
+	/* data size of DSF files are padded to multiple of 4096,
+	   we use the actual data size as chunk size */
+
+	uint64_t data_size;
+	data_size = (((uint64_t)GUINT32_FROM_LE(data_chunk.size_high)) << 32) |
+		((uint64_t)GUINT32_FROM_LE(data_chunk.size_low));
+	data_size -= sizeof(data_chunk);
+
+	metadata->chunk_size = data_size;
+	metadata->channels = (unsigned) dsf_fmt_chunk.channelnum;
+	metadata->sample_rate = samplefreq;
+
+	/* check bits per sample format, determine if bitreverse is needed */
+	metadata->bitreverse = dsf_fmt_chunk.bitssample == 1;
+	return true;
+}
+
+static void
+bit_reverse_buffer(uint8_t *p, uint8_t *end)
+{
+	for (; p < end; ++p)
+		*p = bit_reverse(*p);
+}
+
+/**
+ * DSF data is build up of alternating 4096 blocks of DSD samples for left and
+ * right. Convert the buffer holding 1 block of 4096 DSD left samples and 1
+ * block of 4096 DSD right samples to 8k of samples in normal PCM left/right
+ * order.
+ */
+static void
+dsf_to_pcm_order(uint8_t *dest, uint8_t *scratch, size_t nrbytes)
+{
+	for (unsigned i = 0, j = 0; i < (unsigned)nrbytes; i += 2) {
+		scratch[i] = *(dest+j);
+		j++;
+	}
+
+	for (unsigned i = 1, j = 0; i < (unsigned) nrbytes; i += 2) {
+		scratch[i] = *(dest+4096+j);
+		j++;
+	}
+
+	for (unsigned i = 0; i < (unsigned)nrbytes; i++) {
+		*dest = scratch[i];
+		dest++;
+	}
+}
+
+/**
+ * Decode one complete DSF 'data' chunk i.e. a complete song
+ */
+static bool
+dsf_decode_chunk(struct decoder *decoder, struct input_stream *is,
+		    unsigned channels,
+		    uint64_t chunk_size,
+		    bool bitreverse)
+{
+	uint8_t buffer[8192];
+
+	/* scratch buffer for DSF samples to convert to the needed
+	   normal left/right regime of samples */
+	uint8_t dsf_scratch_buffer[8192];
+
+	const size_t sample_size = sizeof(buffer[0]);
+	const size_t frame_size = channels * sample_size;
+	const unsigned buffer_frames = sizeof(buffer) / frame_size;
+	const unsigned buffer_samples = buffer_frames * frame_size;
+	const size_t buffer_size = buffer_samples * sample_size;
+
+	while (chunk_size > 0) {
+		/* see how much aligned data from the remaining chunk
+		   fits into the local buffer */
+		unsigned now_frames = buffer_frames;
+		size_t now_size = buffer_size;
+		if (chunk_size < (uint64_t)now_size) {
+			now_frames = (unsigned)chunk_size / frame_size;
+			now_size = now_frames * frame_size;
+		}
+
+		size_t nbytes = decoder_read(decoder, is, buffer, now_size);
+		if (nbytes != now_size)
+			return false;
+
+		chunk_size -= nbytes;
+
+		if (bitreverse)
+			bit_reverse_buffer(buffer, buffer + nbytes);
+
+		dsf_to_pcm_order(buffer, dsf_scratch_buffer, nbytes);
+
+		enum decoder_command cmd =
+			decoder_data(decoder, is, buffer, nbytes, 0);
+		switch (cmd) {
+		case DECODE_COMMAND_NONE:
+			break;
+
+		case DECODE_COMMAND_START:
+		case DECODE_COMMAND_STOP:
+			return false;
+
+		case DECODE_COMMAND_SEEK:
+
+			/* not implemented yet */
+			decoder_seek_error(decoder);
+			break;
+			}
+	}
+	return dsdlib_skip(decoder, is, chunk_size);
+}
+
+static void
+dsf_stream_decode(struct decoder *decoder, struct input_stream *is)
+{
+	struct dsf_metadata metadata = {
+		.sample_rate = 0,
+		.channels = 0,
+	};
+
+	/* check if it is a proper DSF file */
+	if (!dsf_read_metadata(decoder, is, &metadata))
+		return;
+
+	GError *error = NULL;
+	struct audio_format audio_format;
+	if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
+				       SAMPLE_FORMAT_DSD,
+				       metadata.channels, &error)) {
+		g_warning("%s", error->message);
+		g_error_free(error);
+		return;
+	}
+
+	/* success: file was recognized */
+	decoder_initialized(decoder, &audio_format, false, -1);
+
+	if (!dsf_decode_chunk(decoder, is, metadata.channels,
+			      metadata.chunk_size,
+			      metadata.bitreverse))
+		return;
+}
+
+static bool
+dsf_scan_stream(struct input_stream *is,
+		   G_GNUC_UNUSED const struct tag_handler *handler,
+		   G_GNUC_UNUSED void *handler_ctx)
+{
+	struct dsf_metadata metadata = {
+		.sample_rate = 0,
+		.channels = 0,
+	};
+
+	/* check DSF metadata */
+	if (!dsf_read_metadata(NULL, is, &metadata))
+		return false;
+
+	struct audio_format audio_format;
+	if (!audio_format_init_checked(&audio_format, metadata.sample_rate / 8,
+				       SAMPLE_FORMAT_DSD,
+				       metadata.channels, NULL))
+		/* refuse to parse files which we cannot play anyway */
+		return false;
+
+	return true;
+}
+
+static const char *const dsf_suffixes[] = {
+	"dsf",
+	NULL
+};
+
+static const char *const dsf_mime_types[] = {
+	"application/x-dsf",
+	NULL
+};
+
+const struct decoder_plugin dsf_decoder_plugin = {
+	.name = "dsf",
+	.stream_decode = dsf_stream_decode,
+	.scan_stream = dsf_scan_stream,
+	.suffixes = dsf_suffixes,
+	.mime_types = dsf_mime_types,
+};

src/decoder/dsf_decoder_plugin.h

+/*
+ * Copyright (C) 2012 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_DECODER_DSF_H
+#define MPD_DECODER_DSF_H
+
+extern const struct decoder_plugin dsf_decoder_plugin;
+
+#endif

src/decoder_list.c

 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2012 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -25,6 +25,7 @@
 #include "mpd_error.h"
 #include "decoder/pcm_decoder_plugin.h"
 #include "decoder/dsdiff_decoder_plugin.h"
+#include "decoder/dsf_decoder_plugin.h"
 
 #include <glib.h>
 
@@ -72,6 +73,7 @@ const struct decoder_plugin *const decoder_plugins[] = {
 	&audiofile_decoder_plugin,
 #endif
 	&dsdiff_decoder_plugin,
+	&dsf_decoder_plugin,
 #ifdef HAVE_FAAD
 	&faad_decoder_plugin,
 #endif