output/alsa, pcm: rename "DSD over USB" to "DoP"

The standard has been renamed since the early draft that was
implemented in MPD.
This commit is contained in:
Max Kellermann 2014-08-31 16:12:26 +02:00
parent 6e04d66a35
commit e5a28bfd8d
10 changed files with 60 additions and 59 deletions

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@ -465,7 +465,7 @@ libpcm_a_SOURCES = \
src/pcm/PcmConvert.cxx src/pcm/PcmConvert.hxx \
src/pcm/dsd2pcm/dsd2pcm.c src/pcm/dsd2pcm/dsd2pcm.h \
src/pcm/PcmDsd.cxx src/pcm/PcmDsd.hxx \
src/pcm/PcmDsdUsb.cxx src/pcm/PcmDsdUsb.hxx \
src/pcm/PcmDop.cxx src/pcm/PcmDop.hxx \
src/pcm/Volume.cxx src/pcm/Volume.hxx \
src/pcm/PcmMix.cxx src/pcm/PcmMix.hxx \
src/pcm/PcmChannels.cxx src/pcm/PcmChannels.hxx \

1
NEWS
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@ -55,6 +55,7 @@ ver 0.19 (not yet released)
- shine: new encoder plugin
* output
- alsa: support native DSD playback
- alsa: rename "DSD over USB" to "DoP"
* threads:
- the update thread runs at "idle" priority
- the output thread runs at "real-time" priority

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@ -1991,15 +1991,15 @@ systemctl start mpd.socket</programlisting>
</row>
<row>
<entry>
<varname>dsd_usb</varname>
<varname>dop</varname>
<parameter>yes|no</parameter>
</entry>
<entry>
If set to <parameter>yes</parameter>, then DSD over
USB according to the <ulink
url="http://www.sonore.us/DoP_openStandard_1v1.pdf">pro
posed standard by dCS and others</ulink> is enabled. This wraps
DSD samples in fake 24 bit PCM, and is understood by
PCM according to the <ulink
url="http://dsd-guide.com/dop-open-standard">DoP
standard others</ulink> is enabled. This wraps DSD
samples in fake 24 bit PCM, and is understood by
some DSD capable products, but may be harmful to
other hardware. Therefore, the default is
<parameter>no</parameter> and you can enable the

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@ -62,12 +62,11 @@ struct AlsaOutput {
bool use_mmap;
/**
* Enable DSD over USB according to the dCS suggested
* standard?
* Enable DSD over PCM according to the DoP standard standard?
*
* @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
* @see http://dsd-guide.com/dop-open-standard
*/
bool dsd_usb;
bool dop;
/** libasound's buffer_time setting (in microseconds) */
unsigned int buffer_time;
@ -153,7 +152,9 @@ AlsaOutput::Configure(const config_param &param, Error &error)
use_mmap = param.GetBlockValue("use_mmap", false);
dsd_usb = param.GetBlockValue("dsd_usb", false);
dop = param.GetBlockValue("dop", false) ||
/* legacy name from MPD 0.18 and older: */
param.GetBlockValue("dsd_usb", false);
buffer_time = param.GetBlockValue("buffer_time",
MPD_ALSA_BUFFER_TIME_US);
@ -639,34 +640,34 @@ alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format,
bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
Error &error)
{
assert(ad->dsd_usb);
assert(ad->dop);
assert(audio_format.format == SampleFormat::DSD);
/* pass 24 bit to alsa_setup() */
AudioFormat usb_format = audio_format;
usb_format.format = SampleFormat::S24_P32;
usb_format.sample_rate /= 2;
AudioFormat dop_format = audio_format;
dop_format.format = SampleFormat::S24_P32;
dop_format.sample_rate /= 2;
const AudioFormat check = usb_format;
const AudioFormat check = dop_format;
if (!alsa_setup(ad, usb_format, packed_r, reverse_endian_r, error))
if (!alsa_setup(ad, dop_format, packed_r, reverse_endian_r, error))
return false;
/* if the device allows only 32 bit, shift all DSD-over-USB
/* if the device allows only 32 bit, shift all DoP
samples left by 8 bit and leave the lower 8 bit cleared;
the DSD-over-USB documentation does not specify whether
this is legal, but there is anecdotical evidence that this
is possible (and the only option for some devices) */
*shift8_r = usb_format.format == SampleFormat::S32;
if (usb_format.format == SampleFormat::S32)
usb_format.format = SampleFormat::S24_P32;
*shift8_r = dop_format.format == SampleFormat::S32;
if (dop_format.format == SampleFormat::S32)
dop_format.format = SampleFormat::S24_P32;
if (usb_format != check) {
if (dop_format != check) {
/* no bit-perfect playback, which is required
for DSD over USB */
error.Format(alsa_output_domain,
"Failed to configure DSD-over-USB on ALSA device \"%s\"",
"Failed to configure DSD-over-PCM on ALSA device \"%s\"",
alsa_device(ad));
delete[] ad->silence;
return false;
@ -681,9 +682,9 @@ alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
{
bool shift8 = false, packed, reverse_endian;
const bool dsd_usb = ad->dsd_usb &&
const bool dop = ad->dop &&
audio_format.format == SampleFormat::DSD;
const bool success = dsd_usb
const bool success = dop
? alsa_setup_dsd(ad, audio_format,
&shift8, &packed, &reverse_endian,
error)
@ -694,7 +695,7 @@ alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
ad->pcm_export->Open(audio_format.format,
audio_format.channels,
dsd_usb, shift8, packed, reverse_endian);
dop, shift8, packed, reverse_endian);
return true;
}

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@ -18,7 +18,7 @@
*/
#include "config.h"
#include "PcmDsdUsb.hxx"
#include "PcmDop.hxx"
#include "PcmBuffer.hxx"
#include "AudioFormat.hxx"
#include "util/ConstBuffer.hxx"
@ -27,20 +27,20 @@
constexpr
static inline uint32_t
pcm_two_dsd_to_usb_marker1(uint8_t a, uint8_t b)
pcm_two_dsd_to_dop_marker1(uint8_t a, uint8_t b)
{
return 0xff050000 | (a << 8) | b;
}
constexpr
static inline uint32_t
pcm_two_dsd_to_usb_marker2(uint8_t a, uint8_t b)
pcm_two_dsd_to_dop_marker2(uint8_t a, uint8_t b)
{
return 0xfffa0000 | (a << 8) | b;
}
ConstBuffer<uint32_t>
pcm_dsd_to_usb(PcmBuffer &buffer, unsigned channels,
pcm_dsd_to_dop(PcmBuffer &buffer, unsigned channels,
ConstBuffer<uint8_t> _src)
{
assert(audio_valid_channel_count(channels));
@ -65,7 +65,7 @@ pcm_dsd_to_usb(PcmBuffer &buffer, unsigned channels,
/* each 24 bit sample has 16 DSD sample bits
plus the magic 0x05 marker */
*dest++ = pcm_two_dsd_to_usb_marker1(src[0], src[channels]);
*dest++ = pcm_two_dsd_to_dop_marker1(src[0], src[channels]);
/* seek the source pointer to the next
channel */
@ -80,7 +80,7 @@ pcm_dsd_to_usb(PcmBuffer &buffer, unsigned channels,
/* each 24 bit sample has 16 DSD sample bits
plus the magic 0xfa marker */
*dest++ = pcm_two_dsd_to_usb_marker2(src[0], src[channels]);
*dest++ = pcm_two_dsd_to_dop_marker2(src[0], src[channels]);
/* seek the source pointer to the next
channel */

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@ -17,8 +17,8 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef MPD_PCM_DSD_USB_HXX
#define MPD_PCM_DSD_USB_HXX
#ifndef MPD_PCM_DOP_HXX
#define MPD_PCM_DOP_HXX
#include "check.h"
@ -30,12 +30,11 @@ template<typename T> struct ConstBuffer;
/**
* Pack DSD 1 bit samples into (padded) 24 bit PCM samples for
* playback over USB, according to the proposed standard by
* dCS and others:
* http://www.sonore.us/DoP_openStandard_1v1.pdf
* playback over USB, according to the DoP standard:
* http://dsd-guide.com/dop-open-standard
*/
ConstBuffer<uint32_t>
pcm_dsd_to_usb(PcmBuffer &buffer, unsigned channels,
pcm_dsd_to_dop(PcmBuffer &buffer, unsigned channels,
ConstBuffer<uint8_t> src);
#endif

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@ -19,7 +19,7 @@
#include "config.h"
#include "PcmExport.hxx"
#include "PcmDsdUsb.hxx"
#include "PcmDop.hxx"
#include "PcmPack.hxx"
#include "util/ByteReverse.hxx"
#include "util/ConstBuffer.hxx"
@ -28,15 +28,15 @@
void
PcmExport::Open(SampleFormat sample_format, unsigned _channels,
bool _dsd_usb, bool _shift8, bool _pack, bool _reverse_endian)
bool _dop, bool _shift8, bool _pack, bool _reverse_endian)
{
assert(audio_valid_sample_format(sample_format));
assert(!_dsd_usb || audio_valid_channel_count(_channels));
assert(!_dop || 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
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;
@ -64,7 +64,7 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
/* packed 24 bit samples (3 bytes per sample) */
return audio_format.channels * 3;
if (dsd_usb)
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
@ -77,8 +77,8 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
ConstBuffer<void>
PcmExport::Export(ConstBuffer<void> data)
{
if (dsd_usb)
data = pcm_dsd_to_usb(dsd_buffer, channels,
if (dop)
data = pcm_dsd_to_dop(dop_buffer, channels,
ConstBuffer<uint8_t>::FromVoid(data))
.ToVoid();
@ -125,8 +125,8 @@ PcmExport::CalcSourceSize(size_t size) const
/* 32 bit to 24 bit conversion (4 to 3 bytes) */
size = (size / 3) * 4;
if (dsd_usb)
/* DSD over USB doubles the transport size */
if (dop)
/* DoP doubles the transport size */
size /= 2;
return size;

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@ -35,11 +35,11 @@ template<typename T> struct ConstBuffer;
struct PcmExport {
/**
* The buffer is used to convert DSD samples to the
* DSD-over-USB format.
* DoP format.
*
* @see #dsd_usb
* @see #dop
*/
PcmBuffer dsd_buffer;
PcmBuffer dop_buffer;
/**
* The buffer is used to pack samples, removing padding.
@ -61,11 +61,11 @@ struct PcmExport {
uint8_t channels;
/**
* Convert DSD to DSD-over-USB? Input format must be
* Convert DSD to DSD-over-PCM (DoP)? Input format must be
* SampleFormat::DSD and output format must be
* SampleFormat::S24_P32.
*/
bool dsd_usb;
bool dop;
/**
* Convert (padded) 24 bit samples to 32 bit by shifting 8
@ -93,10 +93,10 @@ struct PcmExport {
*
* This function cannot fail.
*
* @param channels the number of channels; ignored unless dsd_usb is set
* @param channels the number of channels; ignored unless dop is set
*/
void Open(SampleFormat sample_format, unsigned channels,
bool dsd_usb, bool shift8, bool pack, bool reverse_endian);
bool dop, bool shift8, bool pack, bool reverse_endian);
/**
* Calculate the size of one output frame.

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@ -108,14 +108,14 @@ class PcmExportTest : public CppUnit::TestFixture {
CPPUNIT_TEST(TestShift8);
CPPUNIT_TEST(TestPack24);
CPPUNIT_TEST(TestReverseEndian);
CPPUNIT_TEST(TestDsdUsb);
CPPUNIT_TEST(TestDop);
CPPUNIT_TEST_SUITE_END();
public:
void TestShift8();
void TestPack24();
void TestReverseEndian();
void TestDsdUsb();
void TestDop();
};
#endif

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@ -106,7 +106,7 @@ PcmExportTest::TestReverseEndian()
}
void
PcmExportTest::TestDsdUsb()
PcmExportTest::TestDop()
{
static constexpr uint8_t src[] = {
0x01, 0x23, 0x45, 0x67,