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

The standard has been renamed since the early draft that was
implemented in MPD.

5 files changed, 50 insertions, 50 deletions

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.