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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.
*/
#include "config.h"
#include "PcmExport.hxx"
#include "PcmDop.hxx"
#include "PcmPack.hxx"
#include "util/ByteReverse.hxx"
#include "util/ConstBuffer.hxx"
#include <iterator>
void
PcmExport::Open(SampleFormat sample_format, unsigned _channels,
bool _dop, bool _shift8, bool _pack, bool _reverse_endian)
{
assert(audio_valid_sample_format(sample_format));
assert(!_dop || audio_valid_channel_count(_channels));
channels = _channels;
dop = _dop && sample_format == SampleFormat::DSD;
if (dop)
/* after the conversion to DoP, the DSD
samples are stuffed inside fake 24 bit samples */
sample_format = SampleFormat::S24_P32;
shift8 = _shift8 && sample_format == SampleFormat::S24_P32;
pack24 = _pack && sample_format == SampleFormat::S24_P32;
assert(!shift8 || !pack24);
reverse_endian = 0;
if (_reverse_endian) {
size_t sample_size = pack24
? 3
: sample_format_size(sample_format);
assert(sample_size <= 0xff);
if (sample_size > 1)
reverse_endian = sample_size;
}
}
size_t
PcmExport::GetFrameSize(const AudioFormat &audio_format) const
{
if (pack24)
/* packed 24 bit samples (3 bytes per sample) */
return audio_format.channels * 3;
if (dop)
/* the DSD-over-USB draft says that DSD 1-bit samples
are enclosed within 24 bit samples, and MPD's
representation of 24 bit is padded to 32 bit (4
bytes per sample) */
return channels * 4;
return audio_format.GetFrameSize();
}
ConstBuffer<void>
PcmExport::Export(ConstBuffer<void> data)
{
if (dop)
data = pcm_dsd_to_dop(dop_buffer, channels,
ConstBuffer<uint8_t>::FromVoid(data))
.ToVoid();
if (pack24) {
const auto src = ConstBuffer<int32_t>::FromVoid(data);
const size_t num_samples = src.size;
const size_t dest_size = num_samples * 3;
uint8_t *dest = (uint8_t *)pack_buffer.Get(dest_size);
assert(dest != nullptr);
pcm_pack_24(dest, src.begin(), src.end());
data.data = dest;
data.size = dest_size;
} else if (shift8) {
const auto src = ConstBuffer<int32_t>::FromVoid(data);
uint32_t *dest = (uint32_t *)pack_buffer.Get(data.size);
data.data = dest;
for (auto i : src)
*dest++ = i << 8;
}
if (reverse_endian > 0) {
assert(reverse_endian >= 2);
const auto src = ConstBuffer<uint8_t>::FromVoid(data);
uint8_t *dest = (uint8_t *)reverse_buffer.Get(data.size);
assert(dest != nullptr);
data.data = dest;
reverse_bytes(dest, src.begin(), src.end(), reverse_endian);
}
return data;
}
size_t
PcmExport::CalcSourceSize(size_t size) const
{
if (pack24)
/* 32 bit to 24 bit conversion (4 to 3 bytes) */
size = (size / 3) * 4;
if (dop)
/* DoP doubles the transport size */
size /= 2;
return size;
}
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