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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 "PcmDsdUsb.hxx"
#include "PcmPack.hxx"
#include "util/ByteReverse.hxx"
void
PcmExport::Open(SampleFormat sample_format, unsigned _channels,
bool _dsd_usb, bool _shift8, bool _pack, bool _reverse_endian)
{
assert(audio_valid_sample_format(sample_format));
assert(!_dsd_usb || audio_valid_channel_count(_channels));
channels = _channels;
dsd_usb = _dsd_usb && sample_format == SampleFormat::DSD;
if (dsd_usb)
/* after the conversion to DSD-over-USB, 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 (dsd_usb)
/* 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();
}
const void *
PcmExport::Export(const void *data, size_t size, size_t &dest_size_r)
{
if (dsd_usb)
data = pcm_dsd_to_usb(dsd_buffer, channels,
(const uint8_t *)data, size, &size);
if (pack24) {
assert(size % 4 == 0);
const size_t num_samples = size / 4;
const size_t dest_size = num_samples * 3;
const uint8_t *src8 = (const uint8_t *)data;
const uint8_t *src_end8 = src8 + size;
uint8_t *dest = (uint8_t *)pack_buffer.Get(dest_size);
assert(dest != nullptr);
pcm_pack_24(dest, (const int32_t *)src8,
(const int32_t *)src_end8);
data = dest;
size = dest_size;
} else if (shift8) {
assert(size % 4 == 0);
const uint8_t *src8 = (const uint8_t *)data;
const uint8_t *src_end8 = src8 + size;
const uint32_t *src = (const uint32_t *)src8;
const uint32_t *const src_end = (const uint32_t *)src_end8;
uint32_t *dest = (uint32_t *)pack_buffer.Get(size);
data = dest;
while (src < src_end)
*dest++ = *src++ << 8;
}
if (reverse_endian > 0) {
assert(reverse_endian >= 2);
uint8_t *dest = (uint8_t *)reverse_buffer.Get(size);
assert(dest != nullptr);
const uint8_t *src = (const uint8_t *)data;
const uint8_t *src_end = src + size;
reverse_bytes(dest, src, src_end, reverse_endian);
data = dest;
}
dest_size_r = size;
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 (dsd_usb)
/* DSD over USB doubles the transport size */
size /= 2;
return size;
}
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