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author | Jurgen Kramer <gtmkramer@xs4all.nl> | 2012-06-16 13:46:42 +0200 |
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committer | Max Kellermann <max@duempel.org> | 2012-06-27 09:54:24 +0200 |
commit | 7235dbadfd113bc3f851dc4f58b74d01d9910af5 (patch) | |
tree | 92ea99f8248f23068d60618f74ab1612ec278202 /src/decoder/dsf_decoder_plugin.c | |
parent | ecec41025f35ea8957dd3b59caeb3f4dfee37821 (diff) | |
download | mpd-7235dbadfd113bc3f851dc4f58b74d01d9910af5.tar.gz mpd-7235dbadfd113bc3f851dc4f58b74d01d9910af5.tar.xz mpd-7235dbadfd113bc3f851dc4f58b74d01d9910af5.zip |
patch to split DSD decoder into separate decoders for DSF en DFF. Move common
functions to new dsdlib. Update user doc.
Diffstat (limited to 'src/decoder/dsf_decoder_plugin.c')
-rw-r--r-- | src/decoder/dsf_decoder_plugin.c | 328 |
1 files changed, 328 insertions, 0 deletions
diff --git a/src/decoder/dsf_decoder_plugin.c b/src/decoder/dsf_decoder_plugin.c new file mode 100644 index 000000000..b66ed75e9 --- /dev/null +++ b/src/decoder/dsf_decoder_plugin.c @@ -0,0 +1,328 @@ +/* + * 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, +}; |