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path: root/src/decoder/plugins/DsfDecoderPlugin.cxx
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Diffstat (limited to 'src/decoder/plugins/DsfDecoderPlugin.cxx')
-rw-r--r--src/decoder/plugins/DsfDecoderPlugin.cxx357
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diff --git a/src/decoder/plugins/DsfDecoderPlugin.cxx b/src/decoder/plugins/DsfDecoderPlugin.cxx
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+/*
+ * 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 "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"
+
+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 or 5644800 Hz */
+
+ if (dsf_fmt_chunk.version != 1 || dsf_fmt_chunk.formatid != 0
+ || dsf_fmt_chunk.channeltype != 2
+ || dsf_fmt_chunk.channelnum != 2
+ || (samplefreq != 2822400 && samplefreq != 5644800))
+ 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);
+
+ /* 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;
+
+ /* 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 uint64_t playable_size = samplecnt * 2 / 8;
+ if (data_size > playable_size)
+ data_size = playable_size;
+
+ metadata->chunk_size = data_size;
+ 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 */
+ size_t now_size = buffer_size;
+ if (chunk_size < (uint64_t)now_size) {
+ unsigned 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(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 */
+ 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",
+ 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,
+};