diff --git a/NEWS b/NEWS
index c17b563f2..bf6e1a6e3 100644
--- a/NEWS
+++ b/NEWS
@@ -1,6 +1,7 @@
 ver 0.20.3 (not yet released)
 * protocol
   - "playlistadd" creates new playlist if it does not exist, as documented
+* use shortcuts such as "dsd64" in log messages
 
 ver 0.20.2 (2017/01/15)
 * input
diff --git a/doc/user.xml b/doc/user.xml
index 8124462ed..e75777448 100644
--- a/doc/user.xml
+++ b/doc/user.xml
@@ -598,7 +598,11 @@ systemctl start mpd.socket</programlisting>
                   <varname>32</varname> (signed 32 bit integer
                   samples), <varname>f</varname> (32 bit floating
                   point, -1.0 to 1.0), "<varname>dsd</varname>" means
-                  DSD (Direct Stream Digital).
+                  DSD (Direct Stream Digital).  For DSD, there are
+                  special cases such as "<varname>dsd64</varname>",
+                  which allows you to omit the sample rate
+                  (e.g. <parameter>dsd512:2</parameter> for stereo
+                  DSD512, i.e. 22.5792 MHz).
                 </para>
                 <para>
                   The sample rate is special for DSD:
diff --git a/src/AudioFormat.cxx b/src/AudioFormat.cxx
index c3ebb2e6f..9a9f8ab5e 100644
--- a/src/AudioFormat.cxx
+++ b/src/AudioFormat.cxx
@@ -45,6 +45,17 @@ StringBuffer<24>
 ToString(const AudioFormat af)
 {
 	StringBuffer<24> buffer;
+
+	if (af.format == SampleFormat::DSD && af.sample_rate > 0 &&
+	    af.sample_rate % 44100 == 0) {
+		/* use shortcuts such as "dsd64" which implies the
+		   sample rate */
+		snprintf(buffer.data(), buffer.capacity(), "dsd%u:%u",
+			 af.sample_rate * 8 / 44100,
+			 af.channels);
+		return buffer;
+	}
+
 	snprintf(buffer.data(), buffer.capacity(), "%u:%s:%u",
 		 af.sample_rate, sample_format_to_string(af.format),
 		 af.channels);
diff --git a/src/AudioParser.cxx b/src/AudioParser.cxx
index 55e01b0cc..b2811cd94 100644
--- a/src/AudioParser.cxx
+++ b/src/AudioParser.cxx
@@ -137,6 +137,26 @@ ParseAudioFormat(const char *src, bool mask)
 	AudioFormat dest;
 	dest.Clear();
 
+	if (strncmp(src, "dsd", 3) == 0) {
+		/* allow format specifications such as "dsd64" which
+		   implies the sample rate */
+
+		char *endptr;
+		auto dsd = strtoul(src + 3, &endptr, 10);
+		if (endptr > src + 3 && *endptr == ':' &&
+		    dsd >= 32 && dsd <= 4096 && dsd % 2 == 0) {
+			dest.sample_rate = dsd * 44100 / 8;
+			dest.format = SampleFormat::DSD;
+
+			src = endptr + 1;
+			dest.channels = ParseChannelCount(src, mask, &src);
+			if (*src != 0)
+				throw FormatRuntimeError("Extra data after channel count: %s", src);
+
+			return dest;
+		}
+	}
+
 	/* parse sample rate */
 
 	dest.sample_rate = ParseSampleRate(src, mask, &src);
diff --git a/test/TestAudioFormat.cxx b/test/TestAudioFormat.cxx
index a29bf1b9d..e5a5041b9 100644
--- a/test/TestAudioFormat.cxx
+++ b/test/TestAudioFormat.cxx
@@ -67,7 +67,8 @@ static constexpr AudioFormatStringTest af_string_tests[] = {
 	{ AudioFormat(44100, SampleFormat::S16, 2), "44100:16:2" },
 	{ AudioFormat(48000, SampleFormat::S24_P32, 6), "48000:24:6" },
 	{ AudioFormat(192000, SampleFormat::FLOAT, 2), "192000:f:2" },
-	{ AudioFormat(352800, SampleFormat::DSD, 2), "352800:dsd:2" },
+	{ AudioFormat(352801, SampleFormat::DSD, 2), "352801:dsd:2" },
+	{ AudioFormat(352800, SampleFormat::DSD, 2), "dsd64:2" },
 };
 
 static constexpr AudioFormatStringTest af_mask_tests[] = {