/* * Copyright (C) 2003-2013 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 "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 #include /* for SEEK_SET, SEEK_CUR */ struct DsfMetaData { unsigned sample_rate, channels; bool bitreverse; uint64_t chunk_size; #ifdef HAVE_ID3TAG InputStream::offset_type id3_offset; uint64_t id3_size; #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 (!dsdlib_read(decoder, is, &dsf_header, sizeof(dsf_header)) || !dsf_header.id.Equals("DSD ")) return false; const uint64_t chunk_size = dsf_header.size.Read(); if (sizeof(dsf_header) != chunk_size) return false; #ifdef HAVE_ID3TAG const uint64_t metadata_offset = dsf_header.pmeta.Read(); #endif /* read the 'fmt ' chunk of the DSF file */ DsfFmtChunk dsf_fmt_chunk; if (!dsdlib_read(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); /* 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 = FromLE32(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 */ DsfDataChunk data_chunk; if (!dsdlib_read(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 */ uint64_t data_size = data_chunk.size.Read(); if (data_size < sizeof(data_chunk)) return false; data_size -= sizeof(data_chunk); metadata->chunk_size = data_size; /* data_size cannot be bigger or equal to total file size */ const uint64_t size = (uint64_t)is.GetSize(); if (data_size >= size) return false; metadata->channels = (unsigned) dsf_fmt_chunk.channelnum; metadata->sample_rate = samplefreq; #ifdef HAVE_ID3TAG /* metada_offset cannot be bigger then or equal to total file size */ if (metadata_offset >= size) metadata->id3_offset = 0; else metadata->id3_offset = (InputStream::offset_type)metadata_offset; #endif /* 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(Decoder &decoder, InputStream &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); const auto cmd = decoder_data(decoder, is, buffer, nbytes, 0); switch (cmd) { case DecoderCommand::NONE: break; case DecoderCommand::START: case DecoderCommand::STOP: return false; case DecoderCommand::SEEK: /* not implemented yet */ decoder_seek_error(decoder); break; } } return dsdlib_skip(&decoder, is, chunk_size); } 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 */ uint64_t chunk_size = metadata.chunk_size; float songtime = ((chunk_size / metadata.channels) * 8) / (float) metadata.sample_rate; /* success: file was recognized */ decoder_initialized(decoder, audio_format, false, songtime); if (!dsf_decode_chunk(decoder, is, metadata.channels, chunk_size, metadata.bitreverse)) return; } 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(NULL, 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 */ unsigned songtime = ((metadata.chunk_size / metadata.channels) * 8) / metadata.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", NULL }; static const char *const dsf_mime_types[] = { "application/x-dsf", NULL }; const struct DecoderPlugin dsf_decoder_plugin = { "dsf", nullptr, nullptr, dsf_stream_decode, nullptr, nullptr, dsf_scan_stream, nullptr, dsf_suffixes, dsf_mime_types, };