/*
* 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 ENABLE_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 ENABLE_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 ENABLE_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 ENABLE_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,
};
