/* * Copyright (C) 2003-2015 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 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(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(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, };