1 files changed, 383 insertions, 0 deletions
diff --git a/src/decoder/plugins/DsfDecoderPlugin.cxx b/src/decoder/plugins/DsfDecoderPlugin.cxx
new file mode 100644
index 000000000..690616d15
--- /dev/null
+++ b/src/decoder/plugins/DsfDecoderPlugin.cxx
@@ -0,0 +1,383 @@
+/*
+ * Copyright (C) 2003-2014 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 "DsfDecoderPlugin.hxx"
+#include "../DecoderAPI.hxx"
+#include "input/InputStream.hxx"
+#include "CheckAudioFormat.hxx"
+#include "util/bit_reverse.h"
+#include "util/Error.hxx"
+#include "system/ByteOrder.hxx"
+#include "DsdLib.hxx"
+#include "tag/TagHandler.hxx"
+#include "Log.hxx"
+
+#include <string.h>
+
+static constexpr unsigned DSF_BLOCK_SIZE = 4096;
+
+struct DsfMetaData {
+	unsigned sample_rate, channels;
+	bool bitreverse;
+	offset_type n_blocks;
+#ifdef HAVE_ID3TAG
+	offset_type id3_offset;
+#endif
+};
+
+struct DsfHeader {
+	/** DSF header id: "DSD " */
+	DsdId id;
+	/** DSD chunk size, including id = 28 */
+	DsdUint64 size;
+	/** total file size */
+	DsdUint64 fsize;
+	/** pointer to id3v2 metadata, should be at the end of the file */
+	DsdUint64 pmeta;
+};
+
+/** DSF file fmt chunk */
+struct DsfFmtChunk {
+	/** id: "fmt " */
+	DsdId id;
+	/** fmt chunk size, including id, normally 52 */
+	DsdUint64 size;
+	/** 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 */
+	DsdUint64 scnt;
+	/** block size per channel = 4096 */
+	uint32_t block_size;
+	/** reserved, should be all zero */
+	uint32_t reserved;
+};
+
+struct DsfDataChunk {
+	DsdId id;
+	/** "data" chunk size, includes header (id+size) */
+	DsdUint64 size;
+};
+
+/**
+ * Read and parse all needed metadata chunks for DSF files.
+ */
+static bool
+dsf_read_metadata(Decoder *decoder, InputStream &is,
+		  DsfMetaData *metadata)
+{
+	DsfHeader dsf_header;
+	if (!decoder_read_full(decoder, is, &dsf_header, sizeof(dsf_header)) ||
+	    !dsf_header.id.Equals("DSD "))
+		return false;
+
+	const offset_type chunk_size = dsf_header.size.Read();
+	if (sizeof(dsf_header) != chunk_size)
+		return false;
+
+#ifdef HAVE_ID3TAG
+	const offset_type metadata_offset = dsf_header.pmeta.Read();
+#endif
+
+	/* read the 'fmt ' chunk of the DSF file */
+	DsfFmtChunk dsf_fmt_chunk;
+	if (!decoder_read_full(decoder, is,
+			       &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) ||
+	    !dsf_fmt_chunk.id.Equals("fmt "))
+		return false;
+
+	const uint64_t fmt_chunk_size = dsf_fmt_chunk.size.Read();
+	if (fmt_chunk_size != sizeof(dsf_fmt_chunk))
+		return false;
+
+	uint32_t samplefreq = FromLE32(dsf_fmt_chunk.sample_freq);
+	const unsigned channels = FromLE32(dsf_fmt_chunk.channelnum);
+
+	/* for now, only support version 1 of the standard, DSD raw stereo
+	   files with a sample freq of 2822400 or 5644800 Hz */
+
+	if (FromLE32(dsf_fmt_chunk.version) != 1 ||
+	    FromLE32(dsf_fmt_chunk.formatid) != 0 ||
+	    !audio_valid_channel_count(channels) ||
+	    !dsdlib_valid_freq(samplefreq))
+		return false;
+
+	uint32_t chblksize = FromLE32(dsf_fmt_chunk.block_size);
+	/* according to the spec block size should always be 4096 */
+	if (chblksize != DSF_BLOCK_SIZE)
+		return false;
+
+	/* read the 'data' chunk of the DSF file */
+	DsfDataChunk data_chunk;
+	if (!decoder_read_full(decoder, is, &data_chunk, sizeof(data_chunk)) ||
+	    !data_chunk.id.Equals("data"))
+		return false;
+
+	/* data size of DSF files are padded to multiple of 4096,
+	   we use the actual data size as chunk size */
+
+	offset_type data_size = data_chunk.size.Read();
+	if (data_size < sizeof(data_chunk))
+		return false;
+
+	data_size -= sizeof(data_chunk);
+
+	/* data_size cannot be bigger or equal to total file size */
+	if (is.KnownSize() && data_size > is.GetRest())
+		return false;
+
+	/* use the sample count from the DSF header as the upper
+	   bound, because some DSF files contain junk at the end of
+	   the "data" chunk */
+	const uint64_t samplecnt = dsf_fmt_chunk.scnt.Read();
+	const offset_type playable_size = samplecnt * channels / 8;
+	if (data_size > playable_size)
+		data_size = playable_size;
+
+	const size_t block_size = channels * DSF_BLOCK_SIZE;
+	metadata->n_blocks = data_size / block_size;
+	metadata->channels = channels;
+	metadata->sample_rate = samplefreq;
+#ifdef HAVE_ID3TAG
+	metadata->id3_offset = metadata_offset;
+#endif
+	/* check bits per sample format, determine if bitreverse is needed */
+	metadata->bitreverse = FromLE32(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);
+}
+
+static void
+InterleaveDsfBlockMono(uint8_t *gcc_restrict dest,
+		       const uint8_t *gcc_restrict src)
+{
+	memcpy(dest, src, DSF_BLOCK_SIZE);
+}
+
+/**
+ * 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
+InterleaveDsfBlockStereo(uint8_t *gcc_restrict dest,
+			 const uint8_t *gcc_restrict src)
+{
+	for (size_t i = 0; i < DSF_BLOCK_SIZE; ++i) {
+		dest[2 * i] = src[i];
+		dest[2 * i + 1] = src[DSF_BLOCK_SIZE + i];
+	}
+}
+
+static void
+InterleaveDsfBlockChannel(uint8_t *gcc_restrict dest,
+			  const uint8_t *gcc_restrict src,
+			  unsigned channels)
+{
+	for (size_t i = 0; i < DSF_BLOCK_SIZE; ++i, dest += channels, ++src)
+		*dest = *src;
+}
+
+static void
+InterleaveDsfBlockGeneric(uint8_t *gcc_restrict dest,
+			  const uint8_t *gcc_restrict src,
+			  unsigned channels)
+{
+	for (unsigned c = 0; c < channels; ++c, ++dest, src += DSF_BLOCK_SIZE)
+		InterleaveDsfBlockChannel(dest, src, channels);
+}
+
+static void
+InterleaveDsfBlock(uint8_t *gcc_restrict dest, const uint8_t *gcc_restrict src,
+		   unsigned channels)
+{
+	if (channels == 1)
+		InterleaveDsfBlockMono(dest, src);
+	else if (channels == 2)
+		InterleaveDsfBlockStereo(dest, src);
+	else
+		InterleaveDsfBlockGeneric(dest, src, channels);
+}
+
+static offset_type
+FrameToBlock(uint64_t frame)
+{
+	return frame / DSF_BLOCK_SIZE;
+}
+
+/**
+ * Decode one complete DSF 'data' chunk i.e. a complete song
+ */
+static bool
+dsf_decode_chunk(Decoder &decoder, InputStream &is,
+		 unsigned channels, unsigned sample_rate,
+		 offset_type n_blocks,
+		 bool bitreverse)
+{
+	const size_t block_size = channels * DSF_BLOCK_SIZE;
+	const offset_type start_offset = is.GetOffset();
+
+	auto cmd = decoder_get_command(decoder);
+	for (offset_type i = 0; i < n_blocks && cmd != DecoderCommand::STOP;) {
+		if (cmd == DecoderCommand::SEEK) {
+			uint64_t frame = decoder_seek_where_frame(decoder);
+			offset_type block = FrameToBlock(frame);
+			if (block >= n_blocks) {
+				decoder_command_finished(decoder);
+				break;
+			}
+
+			offset_type offset =
+				start_offset + block * block_size;
+			if (dsdlib_skip_to(&decoder, is, offset)) {
+				decoder_command_finished(decoder);
+				i = block;
+			} else
+				decoder_seek_error(decoder);
+		}
+
+		/* worst-case buffer size */
+		uint8_t buffer[MAX_CHANNELS * DSF_BLOCK_SIZE];
+		if (!decoder_read_full(&decoder, is, buffer, block_size))
+			return false;
+
+		if (bitreverse)
+			bit_reverse_buffer(buffer, buffer + block_size);
+
+		uint8_t interleaved_buffer[MAX_CHANNELS * DSF_BLOCK_SIZE];
+		InterleaveDsfBlock(interleaved_buffer, buffer, channels);
+
+		cmd = decoder_data(decoder, is,
+				   interleaved_buffer, block_size,
+				   sample_rate / 1000);
+		++i;
+	}
+
+	return true;
+}
+
+static void
+dsf_stream_decode(Decoder &decoder, InputStream &is)
+{
+	/* check if it is a proper DSF file */
+	DsfMetaData metadata;
+	if (!dsf_read_metadata(&decoder, is, &metadata))
+		return;
+
+	Error error;
+	AudioFormat audio_format;
+	if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8,
+				       SampleFormat::DSD,
+				       metadata.channels, error)) {
+		LogError(error);
+		return;
+	}
+	/* Calculate song time from DSD chunk size and sample frequency */
+	const auto n_blocks = metadata.n_blocks;
+	auto songtime = SongTime::FromScale<uint64_t>(n_blocks * DSF_BLOCK_SIZE,
+						      audio_format.sample_rate);
+
+	/* success: file was recognized */
+	decoder_initialized(decoder, audio_format, is.IsSeekable(), songtime);
+
+	dsf_decode_chunk(decoder, is, metadata.channels,
+			 metadata.sample_rate,
+			 n_blocks,
+			 metadata.bitreverse);
+}
+
+static bool
+dsf_scan_stream(InputStream &is,
+		gcc_unused const struct tag_handler *handler,
+		gcc_unused void *handler_ctx)
+{
+	/* check DSF metadata */
+	DsfMetaData metadata;
+	if (!dsf_read_metadata(nullptr, is, &metadata))
+		return false;
+
+	AudioFormat audio_format;
+	if (!audio_format_init_checked(audio_format, metadata.sample_rate / 8,
+				       SampleFormat::DSD,
+				       metadata.channels, IgnoreError()))
+		/* refuse to parse files which we cannot play anyway */
+		return false;
+
+	/* calculate song time and add as tag */
+	const auto n_blocks = metadata.n_blocks;
+	auto songtime = SongTime::FromScale<uint64_t>(n_blocks * DSF_BLOCK_SIZE,
+						      audio_format.sample_rate);
+	tag_handler_invoke_duration(handler, handler_ctx, songtime);
+
+#ifdef HAVE_ID3TAG
+	/* Add available tags from the ID3 tag */
+	dsdlib_tag_id3(is, handler, handler_ctx, metadata.id3_offset);
+#endif
+	return true;
+}
+
+static const char *const dsf_suffixes[] = {
+	"dsf",
+	nullptr
+};
+
+static const char *const dsf_mime_types[] = {
+	"application/x-dsf",
+	nullptr
+};
+
+const struct DecoderPlugin dsf_decoder_plugin = {
+	"dsf",
+	nullptr,
+	nullptr,
+	dsf_stream_decode,
+	nullptr,
+	nullptr,
+	dsf_scan_stream,
+	nullptr,
+	dsf_suffixes,
+	dsf_mime_types,
+};