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-rw-r--r--Makefile.am2
-rw-r--r--NEWS1
-rw-r--r--doc/user.xml10
-rw-r--r--src/output/plugins/AlsaOutputPlugin.cxx41
-rw-r--r--src/pcm/PcmDop.cxx (renamed from src/pcm/PcmDsdUsb.cxx)12
-rw-r--r--src/pcm/PcmDop.hxx (renamed from src/pcm/PcmDsdUsb.hxx)11
-rw-r--r--src/pcm/PcmExport.cxx22
-rw-r--r--src/pcm/PcmExport.hxx14
-rw-r--r--test/test_pcm_all.hxx4
-rw-r--r--test/test_pcm_export.cxx2
10 files changed, 60 insertions, 59 deletions
diff --git a/Makefile.am b/Makefile.am
index 89acbaa8d..606081fab 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -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 \
diff --git a/NEWS b/NEWS
index fb513cd79..94fb234ca 100644
--- a/NEWS
+++ b/NEWS
@@ -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
diff --git a/doc/user.xml b/doc/user.xml
index e5b11927d..8b2782911 100644
--- a/doc/user.xml
+++ b/doc/user.xml
@@ -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
diff --git a/src/output/plugins/AlsaOutputPlugin.cxx b/src/output/plugins/AlsaOutputPlugin.cxx
index 0ebf3ddbe..33a090274 100644
--- a/src/output/plugins/AlsaOutputPlugin.cxx
+++ b/src/output/plugins/AlsaOutputPlugin.cxx
@@ -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;
}
diff --git a/src/pcm/PcmDsdUsb.cxx b/src/pcm/PcmDop.cxx
index 9b854ad07..b2096d9e4 100644
--- a/src/pcm/PcmDsdUsb.cxx
+++ b/src/pcm/PcmDop.cxx
@@ -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 */
diff --git a/src/pcm/PcmDsdUsb.hxx b/src/pcm/PcmDop.hxx
index 5e05c009b..03161c456 100644
--- a/src/pcm/PcmDsdUsb.hxx
+++ b/src/pcm/PcmDop.hxx
@@ -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
diff --git a/src/pcm/PcmExport.cxx b/src/pcm/PcmExport.cxx
index 5f567f3c6..ef099ba71 100644
--- a/src/pcm/PcmExport.cxx
+++ b/src/pcm/PcmExport.cxx
@@ -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;
diff --git a/src/pcm/PcmExport.hxx b/src/pcm/PcmExport.hxx
index 8169dad7b..b99a35835 100644
--- a/src/pcm/PcmExport.hxx
+++ b/src/pcm/PcmExport.hxx
@@ -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.
diff --git a/test/test_pcm_all.hxx b/test/test_pcm_all.hxx
index 64ad82a32..7cdd8b63f 100644
--- a/test/test_pcm_all.hxx
+++ b/test/test_pcm_all.hxx
@@ -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
diff --git a/test/test_pcm_export.cxx b/test/test_pcm_export.cxx
index 88c1974e3..410e64e4d 100644
--- a/test/test_pcm_export.cxx
+++ b/test/test_pcm_export.cxx
@@ -106,7 +106,7 @@ PcmExportTest::TestReverseEndian()
}
void
-PcmExportTest::TestDsdUsb()
+PcmExportTest::TestDop()
{
static constexpr uint8_t src[] = {
0x01, 0x23, 0x45, 0x67,
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-22 04:28:59 +0000