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authorJurgen Kramer <gtmkramer@xs4all.nl>2012-06-16 13:46:42 +0200
committerMax Kellermann <max@duempel.org>2012-06-27 09:54:24 +0200
commit7235dbadfd113bc3f851dc4f58b74d01d9910af5 (patch)
tree92ea99f8248f23068d60618f74ab1612ec278202 /src/decoder/dsf_decoder_plugin.c
parentecec41025f35ea8957dd3b59caeb3f4dfee37821 (diff)
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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.c328
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,
+};