release v0.20.4

Fixes race condition when epoll_ctl() gets called after the socket has
been closed, which may affect a different socket created by another
thread meanwhile.

AUTHORS

@@ -5,7 +5,7 @@ The following people have contributed code to MPD:
 
  Warren Dukes <warren.dukes@gmail.com>
  Avuton Olrich <avuton@gmail.com>
- Max Kellermann <max@duempel.org>
+ Max Kellermann <max.kellermann@gmail.com>
  Laszlo Ashin <kodest@gmail.com>
  Viliam Mateicka <viliam.mateicka@gmail.com>
  Eric Wollesen <encoded@xmtp.net>

Makefile.am

@@ -415,6 +415,7 @@ libutil_a_SOURCES = \
 	src/util/CharUtil.hxx \
 	src/util/NumberParser.hxx \
 	src/util/MimeType.cxx src/util/MimeType.hxx \
+	src/util/StringBuffer.hxx \
 	src/util/StringPointer.hxx \
 	src/util/StringView.cxx src/util/StringView.hxx \
 	src/util/AllocatedString.cxx src/util/AllocatedString.hxx \
@@ -539,6 +540,10 @@ ICU_LDADD = libicu.a $(ICU_LIBS)
 # PCM library
 
 libpcm_a_SOURCES = \
+	src/CheckAudioFormat.cxx src/CheckAudioFormat.hxx \
+	src/AudioFormat.cxx src/AudioFormat.hxx \
+	src/AudioParser.cxx src/AudioParser.hxx \
+	src/pcm/SampleFormat.cxx src/pcm/SampleFormat.hxx \
 	src/pcm/Traits.hxx \
 	src/pcm/Interleave.cxx src/pcm/Interleave.hxx \
 	src/pcm/PcmBuffer.cxx src/pcm/PcmBuffer.hxx \
@@ -575,6 +580,8 @@ PCM_LIBS = \
 
 if ENABLE_DSD
 libpcm_a_SOURCES += \
+	src/pcm/Dsd16.cxx src/pcm/Dsd16.hxx \
+	src/pcm/Dsd32.cxx src/pcm/Dsd32.hxx \
 	src/pcm/PcmDsd.cxx src/pcm/PcmDsd.hxx \
 	src/pcm/dsd2pcm/dsd2pcm.c src/pcm/dsd2pcm/dsd2pcm.h
 endif
@@ -613,7 +620,8 @@ libxiph_a_SOURCES += \
 	src/lib/xiph/OggStreamState.hxx
 endif
 
-XIPH_LIBS = libxiph.a
+XIPH_LIBS = libxiph.a \
+	$(OGG_LIBS)
 
 endif
 
@@ -869,9 +877,6 @@ ARCHIVE_LIBS =
 endif
 
 libbasic_a_SOURCES = \
-	src/CheckAudioFormat.cxx src/CheckAudioFormat.hxx \
-	src/AudioFormat.cxx src/AudioFormat.hxx \
-	src/AudioParser.cxx src/AudioParser.hxx \
 	src/ReplayGainConfig.hxx \
 	src/ReplayGainMode.cxx src/ReplayGainMode.hxx \
 	src/ReplayGainInfo.cxx src/ReplayGainInfo.hxx
@@ -2225,6 +2230,7 @@ test_test_icy_parser_LDADD = \
 endif
 
 test_test_pcm_SOURCES = \
+	test/TestAudioFormat.cxx test/TestAudioFormat.hxx \
 	test/test_pcm_util.hxx \
 	test/test_pcm_dither.cxx \
 	test/test_pcm_pack.cxx \

NEWS

@@ -1,3 +1,40 @@
+ver 0.20.4 (2017/02/01)
+* input
+  - nfs: fix freeze after reconnect
+* output
+  - sndio: work around a libroar C++ incompatibility
+* workaround for GCC 4.9 "constexpr" bug
+* fix FreeBSD build failure
+
+ver 0.20.3 (2017/01/25)
+* protocol
+  - "playlistadd" creates new playlist if it does not exist, as documented
+* database
+  - proxy: fix error "terminate called after throwing ..."
+  - proxy: make connect errors during startup non-fatal
+* neighbor
+  - upnp: fix premature expiry
+* replay gain: don't reset ReplayGain levels when unpausing playback
+* silence surround channels when converting from stereo
+* use shortcuts such as "dsd64" in log messages
+
+ver 0.20.2 (2017/01/15)
+* input
+  - alsa: fix crash bug
+  - alsa: fix buffer overruns
+* decoder
+  - flac: add options "probesize" and "analyzeduration"
+* resampler
+  - libsamplerate: reset state after seeking
+* output
+  - fix static noise after changing to a different audio format
+  - alsa: fix the DSD_U32 sample rate
+  - alsa: fix the DSD_U32 byte order
+  - alsa: support DSD_U16
+  - recorder: fix error "Failed to create : No such file or directory"
+* playlist
+  - cue: fix skipping songs
+
 ver 0.20.1 (2017/01/09)
 * input
   - curl: fix crash bug

configure.ac

@@ -1,10 +1,10 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.20.1, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.20.4, musicpd-dev-team@lists.sourceforge.net)
 
 VERSION_MAJOR=0
 VERSION_MINOR=20
-VERSION_REVISION=1
+VERSION_REVISION=4
 VERSION_EXTRA=0
 
 AC_CONFIG_SRCDIR([src/Main.cxx])

doc/mpd.1

@@ -50,6 +50,6 @@ If you find a bug, please report it at
 .br
 <\fBhttp://bugs.musicpd.org/bug_report_page.php\fP>.
 .SH AUTHORS
-Max Kellermann <max@duempel.org>
+Max Kellermann <max.kellermann@gmail.com>
 
 Special thanks to all the people that provided feedback and patches.

doc/protocol.xml

@@ -573,7 +573,12 @@
               <listitem>
                 <para>
                   <varname>audio</varname>:
-                  <returnvalue>sampleRate:bits:channels</returnvalue>
+                  <returnvalue>
+                    The format emitted by the decoder plugin during
+                    playback, format:
+                    "<replaceable>samplerate:bits:channels</replaceable>".
+                    Check the user manual for a detailed explanation.
+                  </returnvalue>
                 </para>
               </listitem>
               <listitem>

doc/user.xml

@@ -576,10 +576,11 @@ systemctl start mpd.socket</programlisting>
               </entry>
               <entry>
                 <para>
-                  Always open the audio output with the specified audio
+                  Always open the audio output with the specified
-                  format (samplerate:bits:channels), regardless of the
+                  audio format
-                  format of the input file.  This is optional for most
+                  (<replaceable>samplerate:bits:channels</replaceable>),
-                  plugins.
+                  regardless of the format of the input file.  This is
+                  optional for most plugins.
                 </para>
                 <para>
                   Any of the three attributes may be an asterisk to
@@ -596,7 +597,20 @@ systemctl start mpd.socket</programlisting>
                   24 bit integer samples padded to 32 bit),
                   <varname>32</varname> (signed 32 bit integer
                   samples), <varname>f</varname> (32 bit floating
-                  point, -1.0 to 1.0).
+                  point, -1.0 to 1.0), "<varname>dsd</varname>" means
+                  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:
+                  <application>MPD</application> counts the number of
+                  bytes, not bits.  Thus, a DSD "bit" rate of 22.5792
+                  MHz (DSD512) is 2822400 from
+                  <application>MPD</application>'s point of view
+                  (44100*512/8).
                 </para>
               </entry>
             </row>
@@ -742,9 +756,11 @@ systemctl start mpd.socket</programlisting>
       <title>Configuring playlist plugins</title>
 
       <para>
-        Playlist plugins are used to load remote playlists.  This is
+        Playlist plugins are used to load remote playlists (protocol
-        not related to <application>MPD</application>'s playlist
+        commands <command>load</command>,
-        directory.
+        <command>listplaylist</command> and
+        <command>listplaylistinfo</command>).  This is not related to
+        <application>MPD</application>'s playlist directory.
       </para>
 
       <para>
@@ -1800,6 +1816,13 @@ run</programlisting>
           database.
         </para>
 
+        <para>
+          Note that unless overridden by the below settings (e.g. by
+          setting them to a blank value), general curl configuration
+          from environment variables such as http_proxy or specified
+          in ~/.curlrc will be in effect.
+        </para>
+
         <informaltable>
           <tgroup cols="2">
             <thead>
@@ -2206,6 +2229,48 @@ run</programlisting>
           Decodes various codecs using
           <application>FFmpeg</application>.
         </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>analyzeduration</varname>
+                  <parameter>VALUE</parameter>
+                </entry>
+                <entry>
+                  Sets the FFmpeg muxer option
+                  <varname>analyzeduration</varname>, which specifies
+                  how many microseconds are analyzed to probe the
+                  input.  The <ulink
+                  url="https://ffmpeg.org/ffmpeg-formats.html">FFmpeg
+                  formats documentation</ulink> has more information.
+                </entry>
+              </row>
+
+              <row>
+                <entry>
+                  <varname>probesize</varname>
+                  <parameter>VALUE</parameter>
+                </entry>
+                <entry>
+                  Sets the FFmpeg muxer option
+                  <varname>probesize</varname>, which specifies
+                  probing size in bytes, i.e. the size of the data to
+                  analyze to get stream information.  The <ulink
+                  url="https://ffmpeg.org/ffmpeg-formats.html">FFmpeg
+                  formats documentation</ulink> has more information.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
       </section>
 
       <section id="flac_decoder">
@@ -4151,6 +4216,22 @@ run</programlisting>
     <section id="playlist_plugins">
       <title>Playlist plugins</title>
 
+      <section>
+        <title><varname>asx</varname></title>
+
+        <para>
+          Reads <filename>.asx</filename> playlist files.
+        </para>
+      </section>
+
+      <section>
+        <title><varname>cue</varname></title>
+
+        <para>
+          Reads <filename>.cue</filename> files.
+        </para>
+      </section>
+
       <section>
         <title><varname>embcue</varname></title>
 
@@ -4175,6 +4256,15 @@ run</programlisting>
         </para>
       </section>
 
+      <section>
+        <title><varname>flac</varname></title>
+
+        <para>
+          Reads the <varname>cuesheet</varname> metablock from a FLAC
+          file.
+        </para>
+      </section>
+
       <section>
         <title><varname>pls</varname></title>
 
@@ -4183,6 +4273,45 @@ run</programlisting>
         </para>
       </section>
 
+      <section>
+        <title><varname>rss</varname></title>
+
+        <para>
+          Reads music links from <filename>.rss</filename> files.
+        </para>
+      </section>
+
+      <section>
+        <title><varname>soundcloud</varname></title>
+
+        <para>
+          Download playlist from SoundCloud.  It accepts URIs starting
+          with <filename>soundcloud://</filename>.
+        </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>apikey</varname>
+                  <parameter>KEY</parameter>
+                </entry>
+                <entry>
+                  An API key to access the SoundCloud servers.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
+      </section>
+
       <section>
         <title><varname>xspf</varname></title>
 

m4/ucred.m4

@@ -1,6 +1,6 @@
 # Check if "struct ucred" is available.
 #
-# Author: Max Kellermann <max@duempel.org>
+# Author: Max Kellermann <max.kellermann@gmail.com>
 
 AC_DEFUN([STRUCT_UCRED],[
 	AC_MSG_CHECKING([for struct ucred])

python/build/libs.py

@@ -19,8 +19,8 @@ libvorbis = AutotoolsProject(
 )
 
 opus = AutotoolsProject(
-    'http://downloads.xiph.org/releases/opus/opus-1.1.3.tar.gz',
+    'http://downloads.xiph.org/releases/opus/opus-1.1.4.tar.gz',
-    '32bbb6b557fe1b6066adc0ae1f08b629',
+    '9122b6b380081dd2665189f97bfd777f04f92dc3ab6698eea1dbb27ad59d8692',
     'lib/libopus.a',
     ['--disable-shared', '--enable-static'],
 )

src/AudioFormat.cxx

@@ -18,6 +18,7 @@
  */
 
 #include "AudioFormat.hxx"
+#include "util/StringBuffer.hxx"
 
 #include <assert.h>
 #include <stdio.h>
@@ -40,46 +41,24 @@ AudioFormat::ApplyMask(AudioFormat mask)
 	assert(IsValid());
 }
 
-const char *
+StringBuffer<24>
-sample_format_to_string(SampleFormat format)
+ToString(const AudioFormat af)
 {
-	switch (format) {
+	StringBuffer<24> buffer;
-	case SampleFormat::UNDEFINED:
-		return "?";
 
-	case SampleFormat::S8:
+	if (af.format == SampleFormat::DSD && af.sample_rate > 0 &&
-		return "8";
+	    af.sample_rate % 44100 == 0) {
-
+		/* use shortcuts such as "dsd64" which implies the
-	case SampleFormat::S16:
+		   sample rate */
-		return "16";
+		snprintf(buffer.data(), buffer.capacity(), "dsd%u:%u",
-
+			 af.sample_rate * 8 / 44100,
-	case SampleFormat::S24_P32:
+			 af.channels);
-		return "24";
+		return buffer;
-
-	case SampleFormat::S32:
-		return "32";
-
-	case SampleFormat::FLOAT:
-		return "f";
-
-	case SampleFormat::DSD:
-		return "dsd";
 	}
 
-	/* unreachable */
+	snprintf(buffer.data(), buffer.capacity(), "%u:%s:%u",
-	assert(false);
-	gcc_unreachable();
-}
-
-const char *
-audio_format_to_string(const AudioFormat af,
-		       struct audio_format_string *s)
-{
-	assert(s != nullptr);
-
-	snprintf(s->buffer, sizeof(s->buffer), "%u:%s:%u",
 		 af.sample_rate, sample_format_to_string(af.format),
 		 af.channels);
 
-	return s->buffer;
+	return buffer;
 }

src/AudioFormat.hxx

@@ -20,47 +20,14 @@
 #ifndef MPD_AUDIO_FORMAT_HXX
 #define MPD_AUDIO_FORMAT_HXX
 
+#include "pcm/SampleFormat.hxx"
 #include "Compiler.h"
 
-#include <stdint.h>
 #include <assert.h>
+#include <stdint.h>
+#include <stddef.h>
 
-#if defined(WIN32) && GCC_CHECK_VERSION(4,6)
+template<size_t CAPACITY> class StringBuffer;
-/* on WIN32, "FLOAT" is already defined, and this triggers -Wshadow */
-#pragma GCC diagnostic push
-#pragma GCC diagnostic ignored "-Wshadow"
-#endif
-
-enum class SampleFormat : uint8_t {
-	UNDEFINED = 0,
-
-	S8,
-	S16,
-
-	/**
-	 * Signed 24 bit integer samples, packed in 32 bit integers
-	 * (the most significant byte is filled with the sign bit).
-	 */
-	S24_P32,
-
-	S32,
-
-	/**
-	 * 32 bit floating point samples in the host's format.  The
-	 * range is -1.0f to +1.0f.
-	 */
-	FLOAT,
-
-	/**
-	 * Direct Stream Digital.  1-bit samples; each frame has one
-	 * byte (8 samples) per channel.
-	 */
-	DSD,
-};
-
-#if defined(WIN32) && GCC_CHECK_VERSION(4,6)
-#pragma GCC diagnostic pop
-#endif
 
 static constexpr unsigned MAX_CHANNELS = 8;
 
@@ -183,13 +150,6 @@ struct AudioFormat {
 	double GetTimeToSize() const;
 };
 
-/**
- * Buffer for audio_format_string().
- */
-struct audio_format_string {
-	char buffer[24];
-};
-
 /**
  * Checks whether the sample rate is valid.
  *
@@ -201,28 +161,6 @@ audio_valid_sample_rate(unsigned sample_rate)
 	return sample_rate > 0 && sample_rate < (1 << 30);
 }
 
-/**
- * Checks whether the sample format is valid.
- */
-static inline bool
-audio_valid_sample_format(SampleFormat format)
-{
-	switch (format) {
-	case SampleFormat::S8:
-	case SampleFormat::S16:
-	case SampleFormat::S24_P32:
-	case SampleFormat::S32:
-	case SampleFormat::FLOAT:
-	case SampleFormat::DSD:
-		return true;
-
-	case SampleFormat::UNDEFINED:
-		break;
-	}
-
-	return false;
-}
-
 /**
  * Checks whether the number of channels is valid.
  */
@@ -258,34 +196,6 @@ AudioFormat::IsMaskValid() const
 		(channels == 0 || audio_valid_channel_count(channels));
 }
 
-gcc_const
-static inline unsigned
-sample_format_size(SampleFormat format)
-{
-	switch (format) {
-	case SampleFormat::S8:
-		return 1;
-
-	case SampleFormat::S16:
-		return 2;
-
-	case SampleFormat::S24_P32:
-	case SampleFormat::S32:
-	case SampleFormat::FLOAT:
-		return 4;
-
-	case SampleFormat::DSD:
-		/* each frame has 8 samples per channel */
-		return 1;
-
-	case SampleFormat::UNDEFINED:
-		return 0;
-	}
-
-	assert(false);
-	gcc_unreachable();
-}
-
 inline unsigned
 AudioFormat::GetSampleSize() const
 {
@@ -304,28 +214,15 @@ AudioFormat::GetTimeToSize() const
 	return sample_rate * GetFrameSize();
 }
 
-/**
- * Renders a #SampleFormat enum into a string, e.g. for printing it
- * in a log file.
- *
- * @param format a #SampleFormat enum value
- * @return the string
- */
-gcc_pure gcc_malloc
-const char *
-sample_format_to_string(SampleFormat format);
-
 /**
  * Renders the #AudioFormat object into a string, e.g. for printing
  * it in a log file.
  *
  * @param af the #AudioFormat object
- * @param s a buffer to print into
+ * @return the string buffer
- * @return the string, or nullptr if the #AudioFormat object is invalid
  */
-gcc_pure gcc_malloc
+gcc_const
-const char *
+StringBuffer<24>
-audio_format_to_string(AudioFormat af,
+ToString(AudioFormat af);
-		       struct audio_format_string *s);
 
 #endif

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);

src/CommandLine.cxx

@@ -107,7 +107,7 @@ static void version(void)
 	       "\n"
 	       "\n"
 	       "Copyright (C) 2003-2007 Warren Dukes <warren.dukes@gmail.com>\n"
-	       "Copyright (C) 2008-2015 Max Kellermann <max@duempel.org>\n"
+	       "Copyright 2008-2017 Max Kellermann <max.kellermann@gmail.com>\n"
 	       "This is free software; see the source for copying conditions.  There is NO\n"
 	       "warranty; not even MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.\n"
 

src/DetachedSong.hxx

@@ -63,18 +63,18 @@ class DetachedSong {
 
 	Tag tag;
 
-	time_t mtime;
+	time_t mtime = 0;
 
 	/**
 	 * Start of this sub-song within the file.
 	 */
-	SongTime start_time;
+	SongTime start_time = SongTime::zero();
 
 	/**
 	 * End of this sub-song within the file.
 	 * Unused if zero.
 	 */
-	SongTime end_time;
+	SongTime end_time = SongTime::zero();
 
 	explicit DetachedSong(const LightSong &other);
 
@@ -82,26 +82,18 @@ public:
 	explicit DetachedSong(const DetachedSong &) = default;
 
 	explicit DetachedSong(const char *_uri)
-		:uri(_uri),
+		:uri(_uri) {}
-		 mtime(0),
-		 start_time(SongTime::zero()), end_time(SongTime::zero()) {}
 
 	explicit DetachedSong(const std::string &_uri)
-		:uri(_uri),
+		:uri(_uri) {}
-		 mtime(0),
-		 start_time(SongTime::zero()), end_time(SongTime::zero()) {}
 
 	explicit DetachedSong(std::string &&_uri)
-		:uri(std::move(_uri)),
+		:uri(std::move(_uri)) {}
-		 mtime(0),
-		 start_time(SongTime::zero()), end_time(SongTime::zero()) {}
 
 	template<typename U>
 	DetachedSong(U &&_uri, Tag &&_tag)
 		:uri(std::forward<U>(_uri)),
-		 tag(std::move(_tag)),
+		 tag(std::move(_tag)) {}
-		 mtime(0),
-		 start_time(SongTime::zero()), end_time(SongTime::zero()) {}
 
 	DetachedSong(DetachedSong &&) = default;
 
@@ -146,7 +138,9 @@ public:
 	 */
 	gcc_pure
 	bool IsSame(const DetachedSong &other) const {
-		return uri == other.uri;
+		return uri == other.uri &&
+			start_time == other.start_time &&
+			end_time == other.end_time;
 	}
 
 	gcc_pure gcc_nonnull_all

src/Main.cxx

@@ -203,7 +203,11 @@ glue_db_init_and_load(void)
 				   "because the database does not need it");
 	}
 
-	instance->database->Open();
+	try {
+		instance->database->Open();
+	} catch (...) {
+		std::throw_with_nested(std::runtime_error("Failed to open database plugin"));
+	}
 
 	if (!instance->database->IsPlugin(simple_db_plugin))
 		return true;

src/PlaylistFile.cxx

@@ -335,7 +335,7 @@ try {
 	const auto path_fs = spl_map_to_fs(utf8path);
 	assert(!path_fs.IsNull());
 
-	FileOutputStream fos(path_fs, FileOutputStream::Mode::APPEND_EXISTING);
+	FileOutputStream fos(path_fs, FileOutputStream::Mode::APPEND_OR_CREATE);
 
 	if (fos.Tell() / (MPD_PATH_MAX + 1) >= playlist_max_length)
 		throw PlaylistError(PlaylistResult::TOO_LARGE,

src/command/PlayerCommands.cxx

@@ -30,6 +30,7 @@
 #include "Instance.hxx"
 #include "Idle.hxx"
 #include "AudioFormat.hxx"
+#include "util/StringBuffer.hxx"
 #include "util/ScopeExit.hxx"
 #include "util/Exception.hxx"
 
@@ -171,13 +172,9 @@ handle_status(Client &client, gcc_unused Request args, Response &r)
 			r.Format("duration: %1.3f\n",
 				 player_status.total_time.ToDoubleS());
 
-		if (player_status.audio_format.IsDefined()) {
+		if (player_status.audio_format.IsDefined())
-			struct audio_format_string af_string;
-
 			r.Format(COMMAND_STATUS_AUDIO ": %s\n",
-				 audio_format_to_string(player_status.audio_format,
+				 ToString(player_status.audio_format).c_str());
-							&af_string));
-		}
 	}
 
 #ifdef ENABLE_DATABASE

src/db/Count.cxx

@@ -62,7 +62,7 @@ Print(Response &r, TagType group, const TagCountMap &m)
 	}
 }
 
-static bool
+static void
 stats_visitor_song(SearchStats &stats, const LightSong &song)
 {
 	stats.n_songs++;
@@ -70,8 +70,6 @@ stats_visitor_song(SearchStats &stats, const LightSong &song)
 	const auto duration = song.GetDuration();
 	if (!duration.IsNegative())
 		stats.total_duration += duration;
-
-	return true;
 }
 
 static bool
@@ -94,7 +92,7 @@ CollectGroupCounts(TagCountMap &map, TagType group, const Tag &tag)
 	return found;
 }
 
-static bool
+static void
 GroupCountVisitor(TagCountMap &map, TagType group, const LightSong &song)
 {
 	assert(song.tag != nullptr);
@@ -103,8 +101,6 @@ GroupCountVisitor(TagCountMap &map, TagType group, const LightSong &song)
 	if (!CollectGroupCounts(map, group, tag) && group == TAG_ALBUM_ARTIST)
 		/* fall back to "Artist" if no "AlbumArtist" was found */
 		CollectGroupCounts(map, TAG_ARTIST, tag);
-
-	return true;
 }
 
 void

src/db/DatabaseGlue.cxx

@@ -37,5 +37,10 @@ DatabaseGlobalInit(EventLoop &loop, DatabaseListener &listener,
 		throw FormatRuntimeError("No such database plugin: %s",
 					 plugin_name);
 
-	return plugin->create(loop, listener, block);
+	try {
+		return plugin->create(loop, listener, block);
+	} catch (...) {
+		std::throw_with_nested(FormatRuntimeError("Failed to initialize database plugin '%s'",
+							  plugin_name));
+	}
 }

src/db/DatabasePlaylist.cxx

@@ -27,13 +27,12 @@
 
 #include <functional>
 
-static bool
+static void
 AddSong(const Storage &storage, const char *playlist_path_utf8,
 	const LightSong &song)
 {
 	spl_append_song(playlist_path_utf8,
 			DatabaseDetachSong(storage, song));
-	return true;
 }
 
 void

src/db/DatabasePrint.cxx

@@ -49,16 +49,14 @@ PrintDirectoryURI(Response &r, bool base, const LightDirectory &directory)
 		 ApplyBaseFlag(directory.GetPath(), base));
 }
 
-static bool
+static void
 PrintDirectoryBrief(Response &r, bool base, const LightDirectory &directory)
 {
 	if (!directory.IsRoot())
 		PrintDirectoryURI(r, base, directory);
-
-	return true;
 }
 
-static bool
+static void
 PrintDirectoryFull(Response &r, bool base, const LightDirectory &directory)
 {
 	if (!directory.IsRoot()) {
@@ -67,8 +65,6 @@ PrintDirectoryFull(Response &r, bool base, const LightDirectory &directory)
 		if (directory.mtime > 0)
 			time_print(r, "Last-Modified", directory.mtime);
 	}
-
-	return true;
 }
 
 static void
@@ -96,7 +92,7 @@ print_playlist_in_directory(Response &r, bool base,
 			 directory->GetPath(), name_utf8);
 }
 
-static bool
+static void
 PrintSongBrief(Response &r, Partition &partition,
 	       bool base, const LightSong &song)
 {
@@ -106,11 +102,9 @@ PrintSongBrief(Response &r, Partition &partition,
 		/* this song file has an embedded CUE sheet */
 		print_playlist_in_directory(r, base,
 					    song.directory, song.uri);
-
-	return true;
 }
 
-static bool
+static void
 PrintSongFull(Response &r, Partition &partition,
 	      bool base, const LightSong &song)
 {
@@ -120,21 +114,18 @@ PrintSongFull(Response &r, Partition &partition,
 		/* this song file has an embedded CUE sheet */
 		print_playlist_in_directory(r, base,
 					    song.directory, song.uri);
-
-	return true;
 }
 
-static bool
+static void
 PrintPlaylistBrief(Response &r, bool base,
 		   const PlaylistInfo &playlist,
 		   const LightDirectory &directory)
 {
 	print_playlist_in_directory(r, base,
 				    &directory, playlist.name.c_str());
-	return true;
 }
 
-static bool
+static void
 PrintPlaylistFull(Response &r, bool base,
 		  const PlaylistInfo &playlist,
 		  const LightDirectory &directory)
@@ -144,8 +135,6 @@ PrintPlaylistFull(Response &r, bool base,
 
 	if (playlist.mtime > 0)
 		time_print(r, "Last-Modified", playlist.mtime);
-
-	return true;
 }
 
 void
@@ -191,15 +180,13 @@ db_selection_print(Response &r, Partition &partition,
 			   0, std::numeric_limits<int>::max());
 }
 
-static bool
+static void
 PrintSongURIVisitor(Response &r, Partition &partition, const LightSong &song)
 {
 	song_print_uri(r, partition, song);
-
-	return true;
 }
 
-static bool
+static void
 PrintUniqueTag(Response &r, TagType tag_type,
 	       const Tag &tag)
 {
@@ -211,8 +198,6 @@ PrintUniqueTag(Response &r, TagType tag_type,
 		if (item.type != tag_type)
 			r.Format("%s: %s\n",
 				 tag_item_names[item.type], item.value);
-
-	return true;
 }
 
 void

src/db/DatabaseQueue.cxx

@@ -27,14 +27,13 @@
 
 #include <functional>
 
-static bool
+static void
 AddToQueue(Partition &partition, const LightSong &song)
 {
 	const Storage &storage = *partition.instance.storage;
 	partition.playlist.AppendSong(partition.pc,
 				      DatabaseDetachSong(storage,
 							 song));
-	return true;
 }
 
 void

src/db/Helpers.cxx

@@ -67,15 +67,13 @@ StatsVisitTag(DatabaseStats &stats, StringSet &artists, StringSet &albums,
 	}
 }
 
-static bool
+static void
 StatsVisitSong(DatabaseStats &stats, StringSet &artists, StringSet &albums,
 	       const LightSong &song)
 {
 	++stats.song_count;
 
 	StatsVisitTag(stats, artists, albums, *song.tag);
-
-	return true;
 }
 
 DatabaseStats

src/db/plugins/ProxyDatabasePlugin.cxx

@@ -34,6 +34,7 @@
 #include "tag/TagBuilder.hxx"
 #include "tag/Tag.hxx"
 #include "util/ScopeExit.hxx"
+#include "util/RuntimeError.hxx"
 #include "protocol/Ack.hxx"
 #include "event/SocketMonitor.hxx"
 #include "event/IdleMonitor.hxx"
@@ -46,7 +47,7 @@
 #include <string>
 #include <list>
 
-class LibmpdclientError final : std::runtime_error {
+class LibmpdclientError final : public std::runtime_error {
 	enum mpd_error code;
 
 public:
@@ -108,8 +109,8 @@ public:
 	static Database *Create(EventLoop &loop, DatabaseListener &listener,
 				const ConfigBlock &block);
 
-	virtual void Open() override;
+	void Open() override;
-	virtual void Close() override;
+	void Close() override;
 	const LightSong *GetSong(const char *uri_utf8) const override;
 	void ReturnSong(const LightSong *song) const override;
 
@@ -126,7 +127,7 @@ public:
 
 	unsigned Update(const char *uri_utf8, bool discard) override;
 
-	virtual time_t GetUpdateStamp() const override {
+	time_t GetUpdateStamp() const override {
 		return update_stamp;
 	}
 
@@ -138,10 +139,10 @@ private:
 	void Disconnect();
 
 	/* virtual methods from SocketMonitor */
-	virtual bool OnSocketReady(unsigned flags) override;
+	bool OnSocketReady(unsigned flags) override;
 
 	/* virtual methods from IdleMonitor */
-	virtual void OnIdle() override;
+	void OnIdle() override;
 };
 
 static constexpr struct {
@@ -345,9 +346,15 @@ ProxyDatabase::Create(EventLoop &loop, DatabaseListener &listener,
 void
 ProxyDatabase::Open()
 {
-	Connect();
-
 	update_stamp = 0;
+
+	try {
+		Connect();
+	} catch (const std::runtime_error &error) {
+		/* this error is non-fatal, because this plugin will
+		   attempt to reconnect again automatically */
+		LogError(error);
+	}
 }
 
 void
@@ -371,7 +378,10 @@ ProxyDatabase::Connect()
 		mpd_connection_free(connection);
 		connection = nullptr;
 
-		throw;
+		std::throw_with_nested(host.empty()
+				       ? std::runtime_error("Failed to connect to remote MPD")
+				       : FormatRuntimeError("Failed to connect to remote MPD '%s'",
+							    host.c_str()));
 	}
 
 #if LIBMPDCLIENT_CHECK_VERSION(2, 10, 0)

src/db/plugins/simple/SimpleDatabasePlugin.hxx

@@ -108,8 +108,8 @@ public:
 	bool Unmount(const char *uri);
 
 	/* virtual methods from class Database */
-	virtual void Open() override;
+	void Open() override;
-	virtual void Close() override;
+	void Close() override;
 
 	const LightSong *GetSong(const char *uri_utf8) const override;
 	void ReturnSong(const LightSong *song) const override;
@@ -125,7 +125,7 @@ public:
 
 	DatabaseStats GetStats(const DatabaseSelection &selection) const override;
 
-	virtual time_t GetUpdateStamp() const override {
+	time_t GetUpdateStamp() const override {
 		return mtime;
 	}
 

src/db/plugins/upnp/UpnpDatabasePlugin.cxx

@@ -180,7 +180,7 @@ UpnpDatabase::ReturnSong(const LightSong *_song) const
 const LightSong *
 UpnpDatabase::GetSong(const char *uri) const
 {
-	auto vpath = stringToTokens(uri, "/", true);
+	auto vpath = stringToTokens(uri, '/');
 	if (vpath.size() < 2)
 		throw DatabaseError(DatabaseErrorCode::NOT_FOUND,
 				    "No such song");
@@ -577,7 +577,7 @@ UpnpDatabase::Visit(const DatabaseSelection &selection,
 		    VisitSong visit_song,
 		    VisitPlaylist visit_playlist) const
 {
-	auto vpath = stringToTokens(selection.uri, "/", true);
+	auto vpath = stringToTokens(selection.uri, '/');
 	if (vpath.empty()) {
 		for (const auto &server : discovery->GetDirectories()) {
 			if (visit_directory) {

src/decoder/Bridge.cxx

@@ -32,6 +32,7 @@
 #include "Log.hxx"
 #include "input/InputStream.hxx"
 #include "util/ConstBuffer.hxx"
+#include "util/StringBuffer.hxx"
 
 #include <assert.h>
 #include <string.h>
@@ -246,15 +247,13 @@ void
 DecoderBridge::Ready(const AudioFormat audio_format,
 		     bool seekable, SignedSongTime duration)
 {
-	struct audio_format_string af_string;
-
 	assert(convert == nullptr);
 	assert(stream_tag == nullptr);
 	assert(decoder_tag == nullptr);
 	assert(!seeking);
 
 	FormatDebug(decoder_domain, "audio_format=%s, seekable=%s",
-		    audio_format_to_string(audio_format, &af_string),
+		    ToString(audio_format).c_str(),
 		    seekable ? "true" : "false");
 
 	{
@@ -264,8 +263,7 @@ DecoderBridge::Ready(const AudioFormat audio_format,
 
 	if (dc.in_audio_format != dc.out_audio_format) {
 		FormatDebug(decoder_domain, "converting to %s",
-			    audio_format_to_string(dc.out_audio_format,
+			    ToString(dc.out_audio_format).c_str());
-						   &af_string));
 
 		convert = new PcmConvert();
 
@@ -317,6 +315,9 @@ DecoderBridge::CommandFinished()
 
 		dc.pipe->Clear(*dc.buffer);
 
+		if (convert != nullptr)
+			convert->Reset();
+
 		timestamp = dc.seek_time.ToDoubleS();
 	}
 

src/decoder/plugins/FfmpegDecoderPlugin.cxx

@@ -56,6 +56,11 @@ extern "C" {
 #include <assert.h>
 #include <string.h>
 
+/**
+ * Muxer options to be passed to avformat_open_input().
+ */
+static AVDictionary *avformat_options = nullptr;
+
 static AVFormatContext *
 FfmpegOpenInput(AVIOContext *pb,
 		const char *filename,
@@ -67,7 +72,11 @@ FfmpegOpenInput(AVIOContext *pb,
 
 	context->pb = pb;
 
-	int err = avformat_open_input(&context, filename, fmt, nullptr);
+	AVDictionary *options = nullptr;
+	AtScopeExit(&options) { av_dict_free(&options); };
+	av_dict_copy(&options, avformat_options, 0);
+
+	int err = avformat_open_input(&context, filename, fmt, &options);
 	if (err < 0)
 		throw MakeFfmpegError(err, "avformat_open_input() failed");
 
@@ -75,12 +84,30 @@ FfmpegOpenInput(AVIOContext *pb,
 }
 
 static bool
-ffmpeg_init(gcc_unused const ConfigBlock &block)
+ffmpeg_init(const ConfigBlock &block)
 {
 	FfmpegInit();
+
+	static constexpr const char *option_names[] = {
+		"probesize",
+		"analyzeduration",
+	};
+
+	for (const char *name : option_names) {
+		const char *value = block.GetBlockValue(name);
+		if (value != nullptr)
+			av_dict_set(&avformat_options, name, value, 0);
+	}
+
 	return true;
 }
 
+static void
+ffmpeg_finish()
+{
+	av_dict_free(&avformat_options);
+}
+
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(57, 25, 0) /* FFmpeg 3.1 */
 
 gcc_pure
@@ -967,7 +994,7 @@ static const char *const ffmpeg_mime_types[] = {
 const struct DecoderPlugin ffmpeg_decoder_plugin = {
 	"ffmpeg",
 	ffmpeg_init,
-	nullptr,
+	ffmpeg_finish,
 	ffmpeg_decode,
 	nullptr,
 	nullptr,

src/filter/FilterInternal.hxx

@@ -52,6 +52,12 @@ public:
 		return out_audio_format;
 	}
 
+	/**
+	 * Reset the filter's state, e.g. drop/flush buffers.
+	 */
+	virtual void Reset() {
+	}
+
 	/**
 	 * Filters a block of PCM data.
 	 *
@@ -60,7 +66,7 @@ public:
 	 * @param src the input buffer
 	 * @return the destination buffer on success (will be
 	 * invalidated by deleting this object or the next FilterPCM()
-	 * call)
+	 * or Reset() call)
 	 */
 	virtual ConstBuffer<void> FilterPCM(ConstBuffer<void> src) = 0;
 };

src/filter/plugins/AutoConvertFilterPlugin.cxx

@@ -48,6 +48,13 @@ public:
 		:Filter(_filter->GetOutAudioFormat()),
 		 filter(std::move(_filter)), convert(std::move(_convert)) {}
 
+	void Reset() override {
+		filter->Reset();
+
+		if (convert)
+			convert->Reset();
+	}
+
 	ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
 };
 

src/filter/plugins/ChainFilterPlugin.cxx

@@ -24,6 +24,7 @@
 #include "filter/FilterRegistry.hxx"
 #include "AudioFormat.hxx"
 #include "util/ConstBuffer.hxx"
+#include "util/StringBuffer.hxx"
 #include "util/RuntimeError.hxx"
 
 #include <memory>
@@ -61,6 +62,7 @@ public:
 	}
 
 	/* virtual methods from class Filter */
+	void Reset() override;
 	ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
 };
 
@@ -107,10 +109,9 @@ PreparedChainFilter::Child::Open(const AudioFormat &prev_audio_format)
 	if (conv_audio_format != prev_audio_format) {
 		delete new_filter;
 
-		struct audio_format_string s;
 		throw FormatRuntimeError("Audio format not supported by filter '%s': %s",
 					 name,
-					 audio_format_to_string(prev_audio_format, &s));
+					 ToString(prev_audio_format).c_str());
 	}
 
 	return new_filter;
@@ -130,6 +131,13 @@ PreparedChainFilter::Open(AudioFormat &in_audio_format)
 	return chain.release();
 }
 
+void
+ChainFilter::Reset()
+{
+	for (auto &child : children)
+		child.filter->Reset();
+}
+
 ConstBuffer<void>
 ChainFilter::FilterPCM(ConstBuffer<void> src)
 {

src/filter/plugins/ConvertFilterPlugin.cxx

@@ -52,6 +52,10 @@ public:
 
 	void Set(const AudioFormat &_out_audio_format);
 
+	void Reset() override {
+		state.Reset();
+	}
+
 	ConstBuffer<void> FilterPCM(ConstBuffer<void> src) override;
 };
 

src/input/plugins/AlsaInputPlugin.cxx

@@ -28,11 +28,11 @@
 #include "AlsaInputPlugin.hxx"
 #include "../InputPlugin.hxx"
 #include "../AsyncInputStream.hxx"
+#include "event/Call.hxx"
 #include "util/Domain.hxx"
 #include "util/RuntimeError.hxx"
 #include "util/StringCompare.hxx"
 #include "util/ReusableArray.hxx"
-#include "util/ScopeExit.hxx"
 
 #include "Log.hxx"
 #include "event/MultiSocketMonitor.hxx"
@@ -56,31 +56,30 @@ static constexpr unsigned int default_rate = 44100; // cd quality
 static constexpr size_t ALSA_MAX_BUFFERED = default_rate * default_channels * 2;
 static constexpr size_t ALSA_RESUME_AT = ALSA_MAX_BUFFERED / 2;
 
-/**
- * This value should be the same as the read buffer size defined in
- * PcmDecoderPlugin.cxx:pcm_stream_decode().
- * We use it to calculate how many audio frames to buffer in the alsa driver
- * before reading from the device. snd_pcm_readi() blocks until that many
- * frames are ready.
- */
-static constexpr size_t read_buffer_size = 4096;
-
 class AlsaInputStream final
 	: public AsyncInputStream,
 	  MultiSocketMonitor, DeferredMonitor {
-	snd_pcm_t *capture_handle;
+
-	size_t frame_size;
+	/**
+	 * The configured name of the ALSA device.
+	 */
+	const std::string device;
+
+	snd_pcm_t *const capture_handle;
+	const size_t frame_size;
 
 	ReusableArray<pollfd> pfd_buffer;
 
 public:
 	AlsaInputStream(EventLoop &loop,
 			const char *_uri, Mutex &_mutex, Cond &_cond,
+			const char *_device,
 			snd_pcm_t *_handle, int _frame_size)
 		:AsyncInputStream(_uri, _mutex, _cond,
 				  ALSA_MAX_BUFFERED, ALSA_RESUME_AT),
 		 MultiSocketMonitor(loop),
 		 DeferredMonitor(loop),
+		 device(_device),
 		 capture_handle(_handle),
 		 frame_size(_frame_size)
 	{
@@ -99,6 +98,14 @@ public:
 	}
 
 	~AlsaInputStream() {
+		/* ClearSocketList must be called from within the
+		   IOThread; if we don't do it manually here, the
+		   ~MultiSocketMonitor() will do it in the current
+		   thread */
+		BlockingCall(MultiSocketMonitor::GetEventLoop(), [this](){
+				ClearSocketList();
+			});
+
 		snd_pcm_close(capture_handle);
 	}
 
@@ -162,7 +169,7 @@ AlsaInputStream::Create(const char *uri, Mutex &mutex, Cond &cond)
 	int frame_size = snd_pcm_format_width(format) / 8 * channels;
 	return new AlsaInputStream(io_thread_get(),
 				   uri, mutex, cond,
-				   handle, frame_size);
+				   device, handle, frame_size);
 }
 
 std::chrono::steady_clock::duration
@@ -205,6 +212,9 @@ AlsaInputStream::DispatchSockets()
 	snd_pcm_sframes_t n_frames;
 	while ((n_frames = snd_pcm_readi(capture_handle,
 					 w.data, w_frames)) < 0) {
+		if (n_frames == -EAGAIN)
+			return;
+
 		if (Recover(n_frames) < 0) {
 			postponed_exception = std::make_exception_ptr(std::runtime_error("PCM error - stream aborted"));
 			cond.broadcast();
@@ -221,20 +231,51 @@ AlsaInputStream::Recover(int err)
 {
 	switch(err) {
 	case -EPIPE:
-		LogDebug(alsa_input_domain, "Buffer Overrun");
+		FormatDebug(alsa_input_domain,
-		// drop through
+			    "Overrun on ALSA capture device \"%s\"",
+			    device.c_str());
+		break;
+
+	case -ESTRPIPE:
+		FormatDebug(alsa_input_domain,
+			    "ALSA capture device \"%s\" was suspended",
+			    device.c_str());
+		break;
+	}
+
+	switch (snd_pcm_state(capture_handle)) {
+	case SND_PCM_STATE_PAUSED:
+		err = snd_pcm_pause(capture_handle, /* disable */ 0);
+		break;
+
+	case SND_PCM_STATE_SUSPENDED:
+		err = snd_pcm_resume(capture_handle);
+		if (err == -EAGAIN)
+			return 0;
+		/* fall-through to snd_pcm_prepare: */
 #if GCC_CHECK_VERSION(7,0)
 		[[fallthrough]];
 #endif
-
+	case SND_PCM_STATE_OPEN:
-	case -ESTRPIPE:
+	case SND_PCM_STATE_SETUP:
-	case -EINTR:
+	case SND_PCM_STATE_XRUN:
-		err = snd_pcm_recover(capture_handle, err, 1);
+		err = snd_pcm_prepare(capture_handle);
+		if (err == 0)
+			err = snd_pcm_start(capture_handle);
+		break;
+
+	case SND_PCM_STATE_DISCONNECTED:
+		break;
+
+	case SND_PCM_STATE_PREPARED:
+	case SND_PCM_STATE_RUNNING:
+	case SND_PCM_STATE_DRAINING:
+		/* this is no error, so just keep running */
+		err = 0;
 		break;
-	default:
-		// something broken somewhere, give up
-		err = -1;
 	}
+
+
 	return err;
 }
 
@@ -245,13 +286,7 @@ ConfigureCapture(snd_pcm_t *capture_handle,
 	int err;
 
 	snd_pcm_hw_params_t *hw_params;
-	if ((err = snd_pcm_hw_params_malloc(&hw_params)) < 0)
+	snd_pcm_hw_params_alloca(&hw_params);
-		throw FormatRuntimeError("Cannot allocate hardware parameter structure (%s)",
-					 snd_strerror(err));
-
-	AtScopeExit(hw_params) {
-		snd_pcm_hw_params_free(hw_params);
-	};
 
 	if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0)
 		throw FormatRuntimeError("Cannot initialize hardware parameter structure (%s)",
@@ -273,37 +308,68 @@ ConfigureCapture(snd_pcm_t *capture_handle,
 		throw FormatRuntimeError("Cannot set sample rate (%s)",
 					 snd_strerror(err));
 
-	/* period needs to be big enough so that poll() doesn't fire too often,
+	snd_pcm_uframes_t buffer_size_min, buffer_size_max;
-	 * but small enough that buffer overruns don't occur if Read() is not
+	snd_pcm_hw_params_get_buffer_size_min(hw_params, &buffer_size_min);
-	 * invoked often enough.
+	snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size_max);
-	 * the calculation here is empirical; however all measurements were
+	unsigned buffer_time_min, buffer_time_max;
-	 * done using 44100:16:2. When we extend this plugin to support
+	snd_pcm_hw_params_get_buffer_time_min(hw_params, &buffer_time_min, 0);
-	 * other audio formats then this may need to be revisited */
+	snd_pcm_hw_params_get_buffer_time_max(hw_params, &buffer_time_max, 0);
-	snd_pcm_uframes_t period = read_buffer_size * 2;
+	FormatDebug(alsa_input_domain, "buffer: size=%u..%u time=%u..%u",
-	int direction = -1;
+		    (unsigned)buffer_size_min, (unsigned)buffer_size_max,
-	if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params,
+		    buffer_time_min, buffer_time_max);
-							  &period, &direction)) < 0)
+
-		throw FormatRuntimeError("Cannot set period size (%s)",
+	snd_pcm_uframes_t period_size_min, period_size_max;
-					 snd_strerror(err));
+	snd_pcm_hw_params_get_period_size_min(hw_params, &period_size_min, 0);
+	snd_pcm_hw_params_get_period_size_max(hw_params, &period_size_max, 0);
+	unsigned period_time_min, period_time_max;
+	snd_pcm_hw_params_get_period_time_min(hw_params, &period_time_min, 0);
+	snd_pcm_hw_params_get_period_time_max(hw_params, &period_time_max, 0);
+	FormatDebug(alsa_input_domain, "period: size=%u..%u time=%u..%u",
+		    (unsigned)period_size_min, (unsigned)period_size_max,
+		    period_time_min, period_time_max);
+
+	/* choose the maximum possible buffer_size ... */
+	snd_pcm_hw_params_set_buffer_size(capture_handle, hw_params,
+					  buffer_size_max);
+
+	/* ... and calculate the period_size to have four periods in
+	   one buffer; this way, we get woken up often enough to avoid
+	   buffer overruns, but not too often */
+	snd_pcm_uframes_t buffer_size;
+	if (snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size) == 0) {
+		snd_pcm_uframes_t period_size = buffer_size / 4;
+		int direction = -1;
+		if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle, hw_params,
+								  &period_size, &direction)) < 0)
+			throw FormatRuntimeError("Cannot set period size (%s)",
+						 snd_strerror(err));
+	}
 
 	if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0)
 		throw FormatRuntimeError("Cannot set parameters (%s)",
 					 snd_strerror(err));
 
+	snd_pcm_uframes_t alsa_buffer_size;
+	err = snd_pcm_hw_params_get_buffer_size(hw_params, &alsa_buffer_size);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_hw_params_get_buffer_size() failed: %s",
+					 snd_strerror(-err));
+
+	snd_pcm_uframes_t alsa_period_size;
+	err = snd_pcm_hw_params_get_period_size(hw_params, &alsa_period_size,
+						nullptr);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_hw_params_get_period_size() failed: %s",
+					 snd_strerror(-err));
+
+	FormatDebug(alsa_input_domain, "buffer_size=%u period_size=%u",
+		    (unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
+
 	snd_pcm_sw_params_t *sw_params;
+	snd_pcm_sw_params_alloca(&sw_params);
 
-	snd_pcm_sw_params_malloc(&sw_params);
 	snd_pcm_sw_params_current(capture_handle, sw_params);
 
-	AtScopeExit(sw_params) {
-		snd_pcm_sw_params_free(sw_params);
-	};
-
-	if ((err = snd_pcm_sw_params_set_start_threshold(capture_handle, sw_params,
-							 period)) < 0)
-		throw FormatRuntimeError("unable to set start threshold (%s)",
-					 snd_strerror(err));
-
 	if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0)
 		throw FormatRuntimeError("unable to install sw params (%s)",
 					 snd_strerror(err));
@@ -316,7 +382,8 @@ AlsaInputStream::OpenDevice(const char *device,
 	snd_pcm_t *capture_handle;
 	int err;
 	if ((err = snd_pcm_open(&capture_handle, device,
-				SND_PCM_STREAM_CAPTURE, 0)) < 0)
+				SND_PCM_STREAM_CAPTURE,
+				SND_PCM_NONBLOCK)) < 0)
 		throw FormatRuntimeError("Failed to open device: %s (%s)",
 					 device, snd_strerror(err));
 

src/java/Class.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2011 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2011 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/Exception.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2012 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2012 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/File.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2014 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2014 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/File.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2014 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2014 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/Global.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2011 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2011 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/Global.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2011 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2011 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/Object.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2011 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2011 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/Ref.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2011 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2011 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/String.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2011 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2011 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/java/String.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2010-2011 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2010-2011 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Easy.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2016 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2016 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Global.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2008-2016 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2008-2016 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Global.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2008-2016 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2008-2016 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Handler.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2008-2017 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2008-2017 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Multi.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2016 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2016 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Request.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2008-2017 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2008-2017 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Request.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2008-2017 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2008-2017 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Slist.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2008-2017 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2008-2017 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Version.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2017 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2017 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/curl/Version.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2017 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2017 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/lib/nfs/Callback.hxx

@@ -24,9 +24,21 @@
 
 #include <exception>
 
+/**
+ * Callbacks for an asynchronous libnfs operation.
+ *
+ * Note that no callback is invoked for cancelled operations.
+ */
 class NfsCallback {
 public:
+	/**
+	 * The operation completed successfully.
+	 */
 	virtual void OnNfsCallback(unsigned status, void *data) = 0;
+
+	/**
+	 * An error has occurred.
+	 */
 	virtual void OnNfsError(std::exception_ptr &&e) = 0;
 };
 

src/lib/nfs/Connection.cxx

@@ -396,6 +396,17 @@ NfsConnection::ScheduleSocket()
 	assert(GetEventLoop().IsInside());
 	assert(context != nullptr);
 
+	const int which_events = nfs_which_events(context);
+
+	if (which_events == POLLOUT && SocketMonitor::IsDefined())
+		/* kludge: if libnfs asks only for POLLOUT, it means
+		   that it is currently waiting for the connect() to
+		   finish - rpc_reconnect_requeue() may have been
+		   called from inside nfs_service(); we must now
+		   unregister the old socket and register the new one
+		   instead */
+		SocketMonitor::Steal();
+
 	if (!SocketMonitor::IsDefined()) {
 		int _fd = nfs_get_fd(context);
 		if (_fd < 0)
@@ -405,7 +416,8 @@ NfsConnection::ScheduleSocket()
 		SocketMonitor::Open(_fd);
 	}
 
-	SocketMonitor::Schedule(libnfs_to_events(nfs_which_events(context)));
+	SocketMonitor::Schedule(libnfs_to_events(which_events)
+				| SocketMonitor::HANGUP);
 }
 
 inline int
@@ -442,10 +454,14 @@ NfsConnection::OnSocketReady(unsigned flags)
 	bool closed = false;
 
 	const bool was_mounted = mount_finished;
-	if (!mount_finished)
+	if (!mount_finished || (flags & SocketMonitor::HANGUP) != 0)
 		/* until the mount is finished, the NFS client may use
 		   various sockets, therefore we unregister and
 		   re-register it each time */
+		/* also re-register the socket if we got a HANGUP,
+		   which is a sure sign that libnfs will close the
+		   socket, which can lead to a race condition if
+		   epoll_ctl() is called later */
 		SocketMonitor::Steal();
 
 	const int result = Service(flags);

src/lib/nfs/FileReader.hxx

@@ -35,6 +35,14 @@
 struct nfsfh;
 class NfsConnection;
 
+/**
+ * A helper class which helps with reading from a file.  It obtains a
+ * connection lease (#NfsLease), opens the given file, "stats" the
+ * file, and finally allos you to read its contents.
+ *
+ * To get started, derive your class from it and implement the pure
+ * virtual methods, construct an instance, and call Open().
+ */
 class NfsFileReader : NfsLease, NfsCallback, DeferredMonitor {
 	enum class State {
 		INITIAL,
@@ -63,14 +71,30 @@ public:
 	void DeferClose();
 
 	/**
+	 * Open the file.  This method is thread-safe.
+	 *
 	 * Throws std::runtime_error on error.
 	 */
 	void Open(const char *uri);
 
 	/**
+	 * Attempt to read from the file.  This may only be done after
+	 * OnNfsFileOpen() has been called.  Only one read operation
+	 * may be performed at a time.
+	 *
+	 * This method is not thread-safe and must be called from
+	 * within the I/O thread.
+	 *
 	 * Throws std::runtime_error on error.
 	 */
 	void Read(uint64_t offset, size_t size);
+
+	/**
+	 * Cancel the most recent Read() call.
+	 *
+	 * This method is not thread-safe and must be called from
+	 * within the I/O thread.
+	 */
 	void CancelRead();
 
 	bool IsIdle() const {
@@ -78,8 +102,27 @@ public:
 	}
 
 protected:
+	/**
+	 * The file has been opened successfully.  It is a regular
+	 * file, and its size is known.  It is ready to be read from
+	 * using Read().
+	 *
+	 * This method will be called from within the I/O thread.
+	 */
 	virtual void OnNfsFileOpen(uint64_t size) = 0;
+
+	/**
+	 * A Read() has completed successfully.
+	 *
+	 * This method will be called from within the I/O thread.
+	 */
 	virtual void OnNfsFileRead(const void *data, size_t size) = 0;
+
+	/**
+	 * An error has occurred, which can be either while waiting
+	 * for OnNfsFileOpen(), or while waiting for OnNfsFileRead(),
+	 * or if disconnected while idle.
+	 */
 	virtual void OnNfsFileError(std::exception_ptr &&e) = 0;
 
 private:

src/lib/upnp/Discovery.hxx

@@ -62,7 +62,7 @@ class UPnPDeviceDirectory final : UpnpCallback {
 		DiscoveredTask(const Upnp_Discovery *disco)
 			:url(disco->Location),
 			 device_id(disco->DeviceId),
-			 expires(disco->Expires) {}
+			 expires(std::chrono::seconds(disco->Expires)) {}
 	};
 
 	/**

src/lib/upnp/Util.cxx

@@ -71,20 +71,19 @@ path_getfather(const std::string &s)
 
 std::list<std::string>
 stringToTokens(const std::string &str,
-	       const char *delims, bool skipinit)
+	       const char delim)
 {
 	std::list<std::string> tokens;
 
-	std::string::size_type startPos = 0;
+	std::string::size_type startPos = str.find_first_not_of(delim, 0);
 
 	// Skip initial delims, return empty if this eats all.
-	if (skipinit &&
+	if (startPos == std::string::npos)
-	    (startPos = str.find_first_not_of(delims, 0)) == std::string::npos)
 		return tokens;
 
 	while (startPos < str.size()) {
 		// Find next delimiter or end of string (end of token)
-		auto pos = str.find_first_of(delims, startPos);
+		auto pos = str.find_first_of(delim, startPos);
 
 		// Add token to the vector and adjust start
 		if (pos == std::string::npos) {

src/lib/upnp/Util.hxx

@@ -33,8 +33,7 @@ path_getfather(const std::string &s);
 
 gcc_pure
 std::list<std::string>
-stringToTokens(const std::string &str,
+stringToTokens(const std::string &str, char delim);
-	       const char *delims = "/", bool skipinit = true);
 
 template <class T>
 bool

src/lib/upnp/WorkQueue.hxx

@@ -52,11 +52,11 @@ class WorkQueue {
 
 	// Status
 	// Worker threads having called exit
-	unsigned n_workers_exited;
+	unsigned n_workers_exited = 0;
-	bool ok;
+	bool ok = false;
 
-	unsigned n_threads;
+	unsigned n_threads = 0;
-	pthread_t *threads;
+	pthread_t *threads = nullptr;
 
 	// Synchronization
 	std::queue<T> queue;
@@ -68,11 +68,8 @@ public:
 	/** Create a WorkQueue
 	 * @param _name for message printing
 	 */
-	WorkQueue(const char *_name)
+	explicit WorkQueue(const char *_name)
-		:name(_name),
+		:name(_name)
-		 n_workers_exited(0),
-		 ok(false),
-		 n_threads(0), threads(nullptr)
 	{
 	}
 
@@ -80,6 +77,9 @@ public:
 		setTerminateAndWait();
 	}
 
+	WorkQueue(const WorkQueue &) = delete;
+	WorkQueue &operator=(const WorkQueue &) = delete;
+
 	/** Start the worker threads.
 	 *
 	 * @param nworkers number of threads copies to start.
@@ -97,6 +97,7 @@ public:
 		assert(n_threads == 0);
 		assert(threads == nullptr);
 
+		ok = true;
 		n_threads = nworkers;
 		threads = new pthread_t[n_threads];
 
@@ -109,7 +110,6 @@ public:
 			}
 		}
 
-		ok = true;
 		return true;
 	}
 

src/net/AllocatedSocketAddress.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2012-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/net/AllocatedSocketAddress.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2012-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/net/SocketAddress.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2012-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/net/SocketAddress.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2012-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/net/StaticSocketAddress.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2012-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/net/StaticSocketAddress.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2012-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2012-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/net/ToString.cxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2011-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2011-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/net/ToString.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2011-2015 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2011-2015 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/output/OutputThread.cxx

@@ -35,6 +35,7 @@
 #include "thread/Slack.hxx"
 #include "thread/Name.hxx"
 #include "util/ConstBuffer.hxx"
+#include "util/StringBuffer.hxx"
 #include "util/ScopeExit.hxx"
 #include "util/RuntimeError.hxx"
 #include "Log.hxx"
@@ -111,8 +112,7 @@ AudioOutput::Open()
 		f = source.Open(request.audio_format, *request.pipe,
 				prepared_replay_gain_filter,
 				prepared_other_replay_gain_filter,
-				prepared_filter)
+				prepared_filter);
-			.WithMask(config_audio_format);
 
 		if (mixer != nullptr && mixer->IsPlugin(software_mixer_plugin))
 			software_mixer_set_filter(*mixer, volume_filter.Get());
@@ -121,14 +121,16 @@ AudioOutput::Open()
 							  name, plugin.name));
 	}
 
-	if (open && f != filter_audio_format) {
+	const auto cf = f.WithMask(config_audio_format);
+
+	if (open && cf != filter_audio_format) {
 		/* if the filter's output format changes, the output
 		   must be reopened as well */
 		CloseOutput(true);
 		open = false;
 	}
 
-	filter_audio_format = f;
+	filter_audio_format = cf;
 
 	if (!open) {
 		try {
@@ -139,7 +141,25 @@ AudioOutput::Open()
 		}
 
 		open = true;
+	} else if (f != out_audio_format) {
+		/* reconfigure the final ConvertFilter for its new
+		   input AudioFormat */
+
+		try {
+			convert_filter_set(convert_filter.Get(),
+					   out_audio_format);
+		} catch (const std::runtime_error &e) {
+			Close(false);
+			std::throw_with_nested(FormatRuntimeError("Failed to convert for \"%s\" [%s]",
+								  name, plugin.name));
+		}
 	}
+
+	if (f != source.GetInputAudioFormat() || f != out_audio_format)
+		FormatDebug(output_domain, "converting in=%s -> f=%s -> out=%s",
+			    ToString(source.GetInputAudioFormat()).c_str(),
+			    ToString(f).c_str(),
+			    ToString(out_audio_format).c_str());
 }
 
 void
@@ -154,6 +174,11 @@ AudioOutput::OpenOutputAndConvert(AudioFormat desired_audio_format)
 							  name, plugin.name));
 	}
 
+	FormatDebug(output_domain,
+		    "opened plugin=%s name=\"%s\" audio_format=%s",
+		    plugin.name, name,
+		    ToString(out_audio_format).c_str());
+
 	try {
 		convert_filter_set(convert_filter.Get(), out_audio_format);
 	} catch (const std::runtime_error &e) {
@@ -177,17 +202,6 @@ AudioOutput::OpenOutputAndConvert(AudioFormat desired_audio_format)
 		std::throw_with_nested(FormatRuntimeError("Failed to convert for \"%s\" [%s]",
 							  name, plugin.name));
 	}
-
-	struct audio_format_string af_string;
-	FormatDebug(output_domain,
-		    "opened plugin=%s name=\"%s\" audio_format=%s",
-		    plugin.name, name,
-		    audio_format_to_string(out_audio_format, &af_string));
-
-	if (source.GetInputAudioFormat() != out_audio_format)
-		FormatDebug(output_domain, "converting from %s",
-			    audio_format_to_string(source.GetInputAudioFormat(),
-						   &af_string));
 }
 
 void

src/output/Source.cxx

@@ -70,6 +70,22 @@ AudioOutputSource::Close()
 	CloseFilter();
 }
 
+void
+AudioOutputSource::Cancel()
+{
+	current_chunk = nullptr;
+	pipe.Cancel();
+
+	if (replay_gain_filter_instance != nullptr)
+		replay_gain_filter_instance->Reset();
+
+	if (other_replay_gain_filter_instance != nullptr)
+		other_replay_gain_filter_instance->Reset();
+
+	if (filter_instance != nullptr)
+		filter_instance->Reset();
+}
+
 void
 AudioOutputSource::OpenFilter(AudioFormat audio_format,
 			      PreparedFilter *prepared_replay_gain_filter,
@@ -79,13 +95,17 @@ try {
 	assert(audio_format.IsValid());
 
 	/* the replay_gain filter cannot fail here */
-	if (prepared_replay_gain_filter != nullptr)
+	if (prepared_replay_gain_filter != nullptr) {
+		replay_gain_serial = 0;
 		replay_gain_filter_instance =
 			prepared_replay_gain_filter->Open(audio_format);
+	}
 
-	if (prepared_other_replay_gain_filter != nullptr)
+	if (prepared_other_replay_gain_filter != nullptr) {
+		other_replay_gain_serial = 0;
 		other_replay_gain_filter_instance =
 			prepared_other_replay_gain_filter->Open(audio_format);
+	}
 
 	filter_instance = prepared_filter->Open(audio_format);
 } catch (...) {

src/output/Source.hxx

@@ -64,13 +64,13 @@ class AudioOutputSource {
 	 * The serial number of the last replay gain info.  0 means no
 	 * replay gain info was available.
 	 */
-	unsigned replay_gain_serial = 0;
+	unsigned replay_gain_serial;
 
 	/**
 	 * The serial number of the last replay gain info by the
 	 * "other" chunk during cross-fading.
 	 */
-	unsigned other_replay_gain_serial = 0;
+	unsigned other_replay_gain_serial;
 
 	/**
 	 * The replay_gain_filter_plugin instance of this audio
@@ -139,11 +139,7 @@ public:
 			 PreparedFilter *prepared_filter);
 
 	void Close();
-
+	void Cancel();
-	void Cancel() {
-		current_chunk = nullptr;
-		pipe.Cancel();
-	}
 
 	/**
 	 * Ensure that ReadTag() or PeekData() return any input.

src/output/plugins/AlsaOutputPlugin.cxx

@@ -50,7 +50,9 @@ static constexpr unsigned MPD_ALSA_BUFFER_TIME_US = 500000;
 
 static constexpr unsigned MPD_ALSA_RETRY_NR = 5;
 
-struct AlsaOutput {
+class AlsaOutput {
+	friend struct AudioOutputWrapper<AlsaOutput>;
+
 	AudioOutput base;
 
 	Manual<PcmExport> pcm_export;
@@ -121,6 +123,7 @@ struct AlsaOutput {
 	 */
 	uint8_t *silence;
 
+public:
 	AlsaOutput(const ConfigBlock &block);
 
 	~AlsaOutput() {
@@ -141,11 +144,20 @@ struct AlsaOutput {
 	void Open(AudioFormat &audio_format);
 	void Close();
 
+	size_t PlayRaw(ConstBuffer<void> data);
 	size_t Play(const void *chunk, size_t size);
 	void Drain();
 	void Cancel();
 
 private:
+	/**
+	 * Set up the snd_pcm_t object which was opened by the caller.
+	 * Set up the configured settings and the audio format.
+	 *
+	 * Throws #std::runtime_error on error.
+	 */
+	void Setup(AudioFormat &audio_format, PcmExport::Params &params);
+
 #ifdef ENABLE_DSD
 	void SetupDop(AudioFormat audio_format,
 		      PcmExport::Params &params);
@@ -384,7 +396,7 @@ AlsaTryFormatOrByteSwap(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 
 /**
  * Attempts to configure the specified sample format.  On DSD_U8
- * failure, attempt to switch to DSD_U32.
+ * failure, attempt to switch to DSD_U32 or DSD_U16.
  */
 static int
 AlsaTryFormatDsd(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
@@ -393,8 +405,10 @@ AlsaTryFormatDsd(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 	int err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
 
 #if defined(ENABLE_DSD) && defined(HAVE_ALSA_DSD_U32)
-	if (err == 0)
+	if (err == 0) {
+		params.dsd_u16 = false;
 		params.dsd_u32 = false;
+	}
 
 	if (err == -EINVAL && fmt == SND_PCM_FORMAT_DSD_U8) {
 		/* attempt to switch to DSD_U32 */
@@ -404,6 +418,20 @@ AlsaTryFormatDsd(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 		err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
 		if (err == 0)
 			params.dsd_u32 = true;
+		else
+			fmt = SND_PCM_FORMAT_DSD_U8;
+	}
+
+	if (err == -EINVAL && fmt == SND_PCM_FORMAT_DSD_U8) {
+		/* attempt to switch to DSD_U16 */
+		fmt = IsLittleEndian()
+			? SND_PCM_FORMAT_DSD_U16_LE
+			: SND_PCM_FORMAT_DSD_U16_BE;
+		err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
+		if (err == 0)
+			params.dsd_u16 = true;
+		else
+			fmt = SND_PCM_FORMAT_DSD_U8;
 	}
 #endif
 
@@ -460,51 +488,44 @@ AlsaSetupFormat(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 }
 
 /**
- * Set up the snd_pcm_t object which was opened by the caller.  Set up
+ * Wrapper for snd_pcm_hw_params().
- * the configured settings and the audio format.
  *
- * Throws #std::runtime_error on error.
+ * @param buffer_time the configured buffer time, or 0 if not configured
+ * @param period_time the configured period time, or 0 if not configured
+ * @param audio_format an #AudioFormat to be configured (or modified)
+ * by this function
+ * @param params to be modified by this function
  */
 static void
-AlsaSetup(AlsaOutput *ad, AudioFormat &audio_format,
+AlsaSetupHw(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
-	  PcmExport::Params &params)
+	    unsigned buffer_time, unsigned period_time,
+	    AudioFormat &audio_format, PcmExport::Params &params)
 {
-	unsigned int sample_rate = audio_format.sample_rate;
-	unsigned int channels = audio_format.channels;
 	int err;
 	unsigned retry = MPD_ALSA_RETRY_NR;
-	unsigned int period_time, period_time_ro;
+	unsigned int period_time_ro = period_time;
-	unsigned int buffer_time;
 
-	period_time_ro = period_time = ad->period_time;
 configure_hw:
 	/* configure HW params */
-	snd_pcm_hw_params_t *hwparams;
+	err = snd_pcm_hw_params_any(pcm, hwparams);
-	snd_pcm_hw_params_alloca(&hwparams);
-	err = snd_pcm_hw_params_any(ad->pcm, hwparams);
 	if (err < 0)
 		throw FormatRuntimeError("snd_pcm_hw_params_any() failed: %s",
 					 snd_strerror(-err));
 
-	err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
+	err = snd_pcm_hw_params_set_access(pcm, hwparams,
 					   SND_PCM_ACCESS_RW_INTERLEAVED);
 	if (err < 0)
 		throw FormatRuntimeError("snd_pcm_hw_params_set_access() failed: %s",
 					 snd_strerror(-err));
 
-	err = AlsaSetupFormat(ad->pcm, hwparams, audio_format, params);
+	err = AlsaSetupFormat(pcm, hwparams, audio_format, params);
 	if (err < 0)
 		throw FormatRuntimeError("Failed to configure format %s: %s",
 					 sample_format_to_string(audio_format.format),
 					 snd_strerror(-err));
 
-	snd_pcm_format_t format;
+	unsigned int channels = audio_format.channels;
-	if (snd_pcm_hw_params_get_format(hwparams, &format) == 0)
+	err = snd_pcm_hw_params_set_channels_near(pcm, hwparams,
-		FormatDebug(alsa_output_domain,
-			    "format=%s (%s)", snd_pcm_format_name(format),
-			    snd_pcm_format_description(format));
-
-	err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
 						  &channels);
 	if (err < 0)
 		throw FormatRuntimeError("Failed to configure %i channels: %s",
@@ -513,18 +534,23 @@ configure_hw:
 
 	audio_format.channels = (int8_t)channels;
 
-	err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
+	const unsigned requested_sample_rate =
-					      &sample_rate, nullptr);
+		params.CalcOutputSampleRate(audio_format.sample_rate);
+	unsigned output_sample_rate = requested_sample_rate;
+
+	err = snd_pcm_hw_params_set_rate_near(pcm, hwparams,
+					      &output_sample_rate, nullptr);
 	if (err < 0)
 		throw FormatRuntimeError("Failed to configure sample rate %u Hz: %s",
-					 audio_format.sample_rate,
+					 requested_sample_rate,
 					 snd_strerror(-err));
 
-	if (sample_rate == 0)
+	if (output_sample_rate == 0)
 		throw FormatRuntimeError("Failed to configure sample rate %u Hz",
 					 audio_format.sample_rate);
 
-	audio_format.sample_rate = sample_rate;
+	if (output_sample_rate != requested_sample_rate)
+		audio_format.sample_rate = params.CalcInputSampleRate(output_sample_rate);
 
 	snd_pcm_uframes_t buffer_size_min, buffer_size_max;
 	snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
@@ -546,9 +572,8 @@ configure_hw:
 		    (unsigned)period_size_min, (unsigned)period_size_max,
 		    period_time_min, period_time_max);
 
-	if (ad->buffer_time > 0) {
+	if (buffer_time > 0) {
-		buffer_time = ad->buffer_time;
+		err = snd_pcm_hw_params_set_buffer_time_near(pcm, hwparams,
-		err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm, hwparams,
 							     &buffer_time, nullptr);
 		if (err < 0)
 			throw FormatRuntimeError("snd_pcm_hw_params_set_buffer_time_near() failed: %s",
@@ -570,14 +595,14 @@ configure_hw:
 
 	if (period_time_ro > 0) {
 		period_time = period_time_ro;
-		err = snd_pcm_hw_params_set_period_time_near(ad->pcm, hwparams,
+		err = snd_pcm_hw_params_set_period_time_near(pcm, hwparams,
 							     &period_time, nullptr);
 		if (err < 0)
 			throw FormatRuntimeError("snd_pcm_hw_params_set_period_time_near() failed: %s",
 						 snd_strerror(-err));
 	}
 
-	err = snd_pcm_hw_params(ad->pcm, hwparams);
+	err = snd_pcm_hw_params(pcm, hwparams);
 	if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
 		period_time_ro = period_time_ro >> 1;
 		goto configure_hw;
@@ -587,9 +612,59 @@ configure_hw:
 	if (retry != MPD_ALSA_RETRY_NR)
 		FormatDebug(alsa_output_domain,
 			    "ALSA period_time set to %d", period_time);
+}
+
+/**
+ * Wrapper for snd_pcm_sw_params().
+ */
+static void
+AlsaSetupSw(snd_pcm_t *pcm, snd_pcm_uframes_t start_threshold,
+	    snd_pcm_uframes_t avail_min)
+{
+	snd_pcm_sw_params_t *swparams;
+	snd_pcm_sw_params_alloca(&swparams);
+
+	int err = snd_pcm_sw_params_current(pcm, swparams);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_sw_params_current() failed: %s",
+					 snd_strerror(-err));
+
+	err = snd_pcm_sw_params_set_start_threshold(pcm, swparams,
+						    start_threshold);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_sw_params_set_start_threshold() failed: %s",
+					 snd_strerror(-err));
+
+	err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_sw_params_set_avail_min() failed: %s",
+					 snd_strerror(-err));
+
+	err = snd_pcm_sw_params(pcm, swparams);
+	if (err < 0)
+		throw FormatRuntimeError("snd_pcm_sw_params() failed: %s",
+					 snd_strerror(-err));
+}
+
+inline void
+AlsaOutput::Setup(AudioFormat &audio_format,
+		  PcmExport::Params &params)
+{
+	snd_pcm_hw_params_t *hwparams;
+	snd_pcm_hw_params_alloca(&hwparams);
+
+	AlsaSetupHw(pcm, hwparams,
+		    buffer_time, period_time,
+		    audio_format, params);
+
+	snd_pcm_format_t format;
+	if (snd_pcm_hw_params_get_format(hwparams, &format) == 0)
+		FormatDebug(alsa_output_domain,
+			    "format=%s (%s)", snd_pcm_format_name(format),
+			    snd_pcm_format_description(format));
 
 	snd_pcm_uframes_t alsa_buffer_size;
-	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
+	int err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
 	if (err < 0)
 		throw FormatRuntimeError("snd_pcm_hw_params_get_buffer_size() failed: %s",
 					 snd_strerror(-err));
@@ -601,32 +676,8 @@ configure_hw:
 		throw FormatRuntimeError("snd_pcm_hw_params_get_period_size() failed: %s",
 					 snd_strerror(-err));
 
-	/* configure SW params */
+	AlsaSetupSw(pcm, alsa_buffer_size - alsa_period_size,
-	snd_pcm_sw_params_t *swparams;
+		    alsa_period_size);
-	snd_pcm_sw_params_alloca(&swparams);
-
-	err = snd_pcm_sw_params_current(ad->pcm, swparams);
-	if (err < 0)
-		throw FormatRuntimeError("snd_pcm_sw_params_current() failed: %s",
-					 snd_strerror(-err));
-
-	err = snd_pcm_sw_params_set_start_threshold(ad->pcm, swparams,
-						    alsa_buffer_size -
-						    alsa_period_size);
-	if (err < 0)
-		throw FormatRuntimeError("snd_pcm_sw_params_set_start_threshold() failed: %s",
-					 snd_strerror(-err));
-
-	err = snd_pcm_sw_params_set_avail_min(ad->pcm, swparams,
-					      alsa_period_size);
-	if (err < 0)
-		throw FormatRuntimeError("snd_pcm_sw_params_set_avail_min() failed: %s",
-					 snd_strerror(-err));
-
-	err = snd_pcm_sw_params(ad->pcm, swparams);
-	if (err < 0)
-		throw FormatRuntimeError("snd_pcm_sw_params() failed: %s",
-					 snd_strerror(-err));
 
 	FormatDebug(alsa_output_domain, "buffer_size=%u period_size=%u",
 		    (unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
@@ -639,13 +690,12 @@ configure_hw:
 		   happen again. */
 		alsa_period_size = 1;
 
-	ad->period_frames = alsa_period_size;
+	period_frames = alsa_period_size;
-	ad->period_position = 0;
+	period_position = 0;
 
-	ad->silence = new uint8_t[snd_pcm_frames_to_bytes(ad->pcm,
+	silence = new uint8_t[snd_pcm_frames_to_bytes(pcm, alsa_period_size)];
-							  alsa_period_size)];
+	snd_pcm_format_set_silence(format, silence,
-	snd_pcm_format_set_silence(format, ad->silence,
+				   alsa_period_size * audio_format.channels);
-				   alsa_period_size * channels);
 
 }
 
@@ -662,11 +712,10 @@ AlsaOutput::SetupDop(const AudioFormat audio_format,
 
 	AudioFormat dop_format = audio_format;
 	dop_format.format = SampleFormat::S24_P32;
-	dop_format.sample_rate /= 2;
 
 	const AudioFormat check = dop_format;
 
-	AlsaSetup(this, dop_format, params);
+	Setup(dop_format, params);
 
 	/* if the device allows only 32 bit, shift all DoP
 	   samples left by 8 bit and leave the lower 8 bit cleared;
@@ -694,17 +743,18 @@ AlsaOutput::SetupOrDop(AudioFormat &audio_format, PcmExport::Params &params)
 	std::exception_ptr dop_error;
 	if (dop && audio_format.format == SampleFormat::DSD) {
 		try {
-			SetupDop(audio_format, params);
 			params.dop = true;
+			SetupDop(audio_format, params);
 			return;
 		} catch (...) {
 			dop_error = std::current_exception();
+			params.dop = false;
 		}
 	}
 
 	try {
 #endif
-		AlsaSetup(this, audio_format, params);
+		Setup(audio_format, params);
 #ifdef ENABLE_DSD
 	} catch (...) {
 		if (dop_error)
@@ -742,6 +792,11 @@ AlsaOutput::Open(AudioFormat &audio_format)
 							  GetDevice()));
 	}
 
+#ifdef ENABLE_DSD
+	if (params.dop)
+		FormatDebug(alsa_output_domain, "DoP (DSD over PCM) enabled");
+#endif
+
 	pcm_export->Open(audio_format.format,
 			 audio_format.channels,
 			 params);
@@ -777,6 +832,7 @@ AlsaOutput::Recover(int err)
 #if GCC_CHECK_VERSION(7,0)
 		[[fallthrough]];
 #endif
+	case SND_PCM_STATE_OPEN:
 	case SND_PCM_STATE_SETUP:
 	case SND_PCM_STATE_XRUN:
 		period_position = 0;
@@ -785,12 +841,11 @@ AlsaOutput::Recover(int err)
 	case SND_PCM_STATE_DISCONNECTED:
 		break;
 	/* this is no error, so just keep running */
+	case SND_PCM_STATE_PREPARED:
 	case SND_PCM_STATE_RUNNING:
+	case SND_PCM_STATE_DRAINING:
 		err = 0;
 		break;
-	default:
-		/* unknown state, do nothing */
-		break;
 	}
 
 	return err;
@@ -822,6 +877,8 @@ AlsaOutput::Cancel()
 	must_prepare = true;
 
 	snd_pcm_drop(pcm);
+
+	pcm_export->Reset();
 }
 
 inline void
@@ -831,6 +888,36 @@ AlsaOutput::Close()
 	delete[] silence;
 }
 
+inline size_t
+AlsaOutput::PlayRaw(ConstBuffer<void> data)
+{
+	if (data.IsEmpty())
+		return 0;
+
+	assert(data.size % out_frame_size == 0);
+
+	const size_t n_frames = data.size / out_frame_size;
+	assert(n_frames > 0);
+
+	while (true) {
+		const auto frames_written = snd_pcm_writei(pcm, data.data,
+							   n_frames);
+		if (frames_written > 0) {
+			period_position = (period_position + frames_written)
+				% period_frames;
+
+			return frames_written * out_frame_size;
+		}
+
+		if (frames_written < 0 && frames_written != -EAGAIN &&
+		    frames_written != -EINTR &&
+		    Recover(frames_written) < 0)
+			throw FormatRuntimeError("snd_pcm_writei() failed: %s",
+						 snd_strerror(-frames_written));
+	}
+
+}
+
 inline size_t
 AlsaOutput::Play(const void *chunk, size_t size)
 {
@@ -856,29 +943,8 @@ AlsaOutput::Play(const void *chunk, size_t size)
 		   been played */
 		return size;
 
-	chunk = e.data;
+	const size_t bytes_written = PlayRaw(e);
-	size = e.size;
+	return pcm_export->CalcSourceSize(bytes_written);
-
-	assert(size % out_frame_size == 0);
-
-	size /= out_frame_size;
-	assert(size > 0);
-
-	while (true) {
-		snd_pcm_sframes_t ret = snd_pcm_writei(pcm, chunk, size);
-		if (ret > 0) {
-			period_position = (period_position + ret)
-				% period_frames;
-
-			size_t bytes_written = ret * out_frame_size;
-			return pcm_export->CalcSourceSize(bytes_written);
-		}
-
-		if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
-		    Recover(ret) < 0)
-			throw FormatRuntimeError("snd_pcm_writei() failed: %s",
-						 snd_strerror(-ret));
-	}
 }
 
 typedef AudioOutputWrapper<AlsaOutput> Wrapper;

src/output/plugins/OssOutputPlugin.cxx

@@ -659,6 +659,10 @@ OssOutput::Cancel()
 		ioctl(fd, SNDCTL_DSP_RESET, 0);
 		DoClose();
 	}
+
+#ifdef AFMT_S24_PACKED
+	pcm_export->Reset();
+#endif
 }
 
 inline size_t

src/output/plugins/RecorderOutputPlugin.cxx

@@ -274,7 +274,7 @@ RecorderOutput::ReopenFormat(AllocatedPath &&new_path)
 	assert(path.IsNull());
 	assert(file == nullptr);
 
-	FileOutputStream *new_file = new FileOutputStream(path);
+	FileOutputStream *new_file = new FileOutputStream(new_path);
 
 	AudioFormat new_audio_format = effective_audio_format;
 

src/output/plugins/SndioOutputPlugin.cxx

@@ -24,8 +24,16 @@
 #include "util/Domain.hxx"
 #include "Log.hxx"
 
+/* work around a C++ incompatibility if the sndio API is emulated by
+   libroar: libroar's "struct roar_service_stream" has a member named
+   "new", which is an illegal identifier in C++ */
+#define new new_
+
 #include <sndio.h>
 
+/* undo the libroar workaround */
+#undef new
+
 #include <stdexcept>
 
 #ifndef SIO_DEVANY

src/output/plugins/sles/AndroidSimpleBufferQueue.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2011-2012 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/output/plugins/sles/Engine.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2011-2012 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/output/plugins/sles/Object.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2011-2012 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/output/plugins/sles/Play.hxx

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ * Copyright (C) 2011-2012 Max Kellermann <max.kellermann@gmail.com>
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions

src/pcm/ChannelsConverter.hxx

@@ -21,7 +21,7 @@
 #define MPD_PCM_CHANNELS_CONVERTER_HXX
 
 #include "check.h"
-#include "AudioFormat.hxx"
+#include "SampleFormat.hxx"
 #include "PcmBuffer.hxx"
 
 #ifndef NDEBUG

src/pcm/Dsd16.cxx

@@ -0,0 +1,62 @@
+/*
+ * Copyright 2003-2017 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 "Dsd16.hxx"
+#include "PcmBuffer.hxx"
+#include "util/ConstBuffer.hxx"
+
+/**
+ * Construct a 16 bit integer from two bytes.
+ */
+static constexpr inline uint16_t
+Construct16(uint8_t a, uint8_t b)
+{
+	/* "a" is the oldest byte, which must be in the most
+	   significant byte */
+
+	return uint16_t(b) | (uint16_t(a) << 8);
+}
+
+static constexpr inline uint16_t
+Dsd8To16Sample(const uint8_t *src, unsigned channels)
+{
+	return Construct16(src[0], src[channels]);
+}
+
+ConstBuffer<uint16_t>
+Dsd8To16(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> _src)
+{
+	const size_t in_frames = _src.size / channels;
+	const size_t out_frames = in_frames / 2;
+	const size_t out_samples = out_frames * channels;
+
+	const uint8_t *src = _src.data;
+	uint16_t *const dest0 = buffer.GetT<uint16_t>(out_samples);
+	uint16_t *dest = dest0;
+
+	for (size_t i = 0; i < out_frames; ++i) {
+		for (size_t c = 0; c < channels; ++c)
+			*dest++ = Dsd8To16Sample(src++, channels);
+
+		src += channels;
+	}
+
+	return {dest0, out_samples};
+}

src/pcm/Dsd16.hxx

@@ -0,0 +1,36 @@
+/*
+ * Copyright 2003-2017 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.
+ */
+
+#ifndef MPD_PCM_DSD_16_HXX
+#define MPD_PCM_DSD_16_HXX
+
+#include "check.h"
+
+#include <stdint.h>
+
+template<typename T> struct ConstBuffer;
+class PcmBuffer;
+
+/**
+ * Convert DSD_U8 to DSD_U16 (native endian, oldest bits in MSB).
+ */
+ConstBuffer<uint16_t>
+Dsd8To16(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> src);
+
+#endif

src/pcm/Dsd32.cxx

@@ -0,0 +1,64 @@
+/*
+ * Copyright 2003-2017 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 "Dsd32.hxx"
+#include "PcmBuffer.hxx"
+#include "util/ConstBuffer.hxx"
+
+/**
+ * Construct a 32 bit integer from four bytes.
+ */
+static constexpr inline uint32_t
+Construct32(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
+{
+	/* "a" is the oldest byte, which must be in the most
+	   significant byte */
+
+	return uint32_t(d) | (uint32_t(c) << 8) |
+		(uint32_t(b) << 16) | (uint32_t(a) << 24);
+}
+
+static constexpr inline uint32_t
+Dsd8To32Sample(const uint8_t *src, unsigned channels)
+{
+	return Construct32(src[0], src[channels],
+			   src[2 * channels], src[3 * channels]);
+}
+
+ConstBuffer<uint32_t>
+Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> _src)
+{
+	const size_t in_frames = _src.size / channels;
+	const size_t out_frames = in_frames / 4;
+	const size_t out_samples = out_frames * channels;
+
+	const uint8_t *src = _src.data;
+	uint32_t *const dest0 = buffer.GetT<uint32_t>(out_samples);
+	uint32_t *dest = dest0;
+
+	for (size_t i = 0; i < out_frames; ++i) {
+		for (size_t c = 0; c < channels; ++c)
+			*dest++ = Dsd8To32Sample(src++, channels);
+
+		src += 3 * channels;
+	}
+
+	return {dest0, out_samples};
+}

src/pcm/Dsd32.hxx

@@ -0,0 +1,36 @@
+/*
+ * Copyright 2003-2017 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.
+ */
+
+#ifndef MPD_PCM_DSD_32_HXX
+#define MPD_PCM_DSD_32_HXX
+
+#include "check.h"
+
+#include <stdint.h>
+
+template<typename T> struct ConstBuffer;
+class PcmBuffer;
+
+/**
+ * Convert DSD_U8 to DSD_U32 (native endian, oldest bits in MSB).
+ */
+ConstBuffer<uint32_t>
+Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> src);
+
+#endif

src/pcm/FormatConverter.hxx

@@ -21,7 +21,7 @@
 #define MPD_PCM_FORMAT_CONVERTER_HXX
 
 #include "check.h"
-#include "AudioFormat.hxx"
+#include "SampleFormat.hxx"
 #include "PcmBuffer.hxx"
 #include "PcmDither.hxx"
 

src/pcm/GlueResampler.cxx

@@ -65,6 +65,12 @@ GluePcmResampler::Close()
 	resampler->Close();
 }
 
+void
+GluePcmResampler::Reset()
+{
+	resampler->Reset();
+}
+
 ConstBuffer<void>
 GluePcmResampler::Resample(ConstBuffer<void> src)
 {

src/pcm/GlueResampler.hxx

@@ -55,6 +55,11 @@ public:
 		return output_sample_format;
 	}
 
+	/**
+	 * @see PcmResampler::Reset()
+	 */
+	void Reset();
+
 	ConstBuffer<void> Resample(ConstBuffer<void> src);
 };
 

src/pcm/LibsamplerateResampler.cxx

@@ -114,6 +114,12 @@ LibsampleratePcmResampler::Close()
 	state = src_delete(state);
 }
 
+void
+LibsampleratePcmResampler::Reset()
+{
+	src_reset(state);
+}
+
 inline ConstBuffer<float>
 LibsampleratePcmResampler::Resample2(ConstBuffer<float> src)
 {

src/pcm/LibsamplerateResampler.hxx

@@ -44,6 +44,7 @@ class LibsampleratePcmResampler final : public PcmResampler {
 public:
 	AudioFormat Open(AudioFormat &af, unsigned new_sample_rate) override;
 	void Close() override;
+	void Reset() override;
 	ConstBuffer<void> Resample(ConstBuffer<void> src) override;
 
 private:

src/pcm/Order.hxx

@@ -21,7 +21,7 @@
 #define MPD_PCM_ORDER_HXX
 
 #include "check.h"
-#include "AudioFormat.hxx"
+#include "SampleFormat.hxx"
 
 class PcmBuffer;
 template<typename T> struct ConstBuffer;

src/pcm/PcmBuffer.hxx

@@ -40,7 +40,7 @@ public:
 
 	/**
 	 * Get the buffer, and guarantee a minimum size.  This buffer becomes
-	 * invalid with the next pcm_buffer_get() call.
+	 * invalid with the next Get() call.
 	 *
 	 * This function will never return nullptr, even if size is
 	 * zero, because the PCM library uses the nullptr return value

src/pcm/PcmChannels.cxx

@@ -20,9 +20,14 @@
 #include "config.h"
 #include "PcmChannels.hxx"
 #include "PcmBuffer.hxx"
+#include "Silence.hxx"
 #include "Traits.hxx"
 #include "AudioFormat.hxx"
 #include "util/ConstBuffer.hxx"
+#include "util/WritableBuffer.hxx"
+
+#include <array>
+#include <algorithm>
 
 #include <assert.h>
 
@@ -90,6 +95,38 @@ NToStereo(typename Traits::pointer_type dest,
 	return dest;
 }
 
+/**
+ * Convert stereo to N channels (where N > 2).  Left and right map to
+ * the first two channels (front left and front right), and the
+ * remaining (surround) channels are filled with silence.
+ */
+template<SampleFormat F, class Traits=SampleTraits<F>>
+static typename Traits::pointer_type
+StereoToN(typename Traits::pointer_type dest,
+	  unsigned dest_channels,
+	  typename Traits::const_pointer_type src,
+	  typename Traits::const_pointer_type end)
+{
+	assert(dest_channels > 2);
+	assert((end - src) % 2 == 0);
+
+	std::array<typename Traits::value_type, MAX_CHANNELS - 2> silence;
+	PcmSilence({&silence.front(), sizeof(silence)}, F);
+
+	while (src != end) {
+		/* copy left/right to front-left/front-right, which is
+		   the first two channels in all multi-channel
+		   configurations **/
+		*dest++ = *src++;
+		*dest++ = *src++;
+
+		/* all other channels are silent */
+		dest = std::copy_n(silence.begin(), dest_channels - 2, dest);
+	}
+
+	return dest;
+}
+
 template<SampleFormat F, class Traits=SampleTraits<F>>
 static typename Traits::pointer_type
 NToM(typename Traits::pointer_type dest,
@@ -133,6 +170,9 @@ ConvertChannels(PcmBuffer &buffer,
 		StereoToMono<F>(dest, src.begin(), src.end());
 	else if (dest_channels == 2)
 		NToStereo<F>(dest, src_channels, src.begin(), src.end());
+	else if (src_channels == 2 && dest_channels > 2)
+		StereoToN<F, Traits>(dest, dest_channels,
+				     src.begin(), src.end());
 	else
 		NToM<F>(dest, dest_channels,
 			src_channels, src.begin(), src.end());

src/pcm/PcmConvert.cxx

@@ -20,7 +20,6 @@
 #include "config.h"
 #include "PcmConvert.hxx"
 #include "ConfiguredResampler.hxx"
-#include "AudioFormat.hxx"
 #include "util/ConstBuffer.hxx"
 
 #include <assert.h>
@@ -117,6 +116,17 @@ PcmConvert::Close()
 #endif
 }
 
+void
+PcmConvert::Reset()
+{
+	if (enable_resampler)
+		resampler.Reset();
+
+#ifdef ENABLE_DSD
+	dsd.Reset();
+#endif
+}
+
 ConstBuffer<void>
 PcmConvert::Convert(ConstBuffer<void> buffer)
 {

src/pcm/PcmConvert.hxx

@@ -67,6 +67,11 @@ public:
 	 */
 	void Close();
 
+	/**
+	 * Reset the filter's state, e.g. drop/flush buffers.
+	 */
+	void Reset();
+
 	/**
 	 * Converts PCM data between two audio formats.
 	 *

src/pcm/PcmDop.cxx

@@ -44,8 +44,6 @@ pcm_dsd_to_dop(PcmBuffer &buffer, unsigned channels,
 	       ConstBuffer<uint8_t> _src)
 {
 	assert(audio_valid_channel_count(channels));
-	assert(!_src.IsNull());
-	assert(_src.size > 0);
 	assert(_src.size % channels == 0);
 
 	const unsigned num_src_samples = _src.size;

src/pcm/PcmDsd.cxx

@@ -71,41 +71,3 @@ PcmDsd::ToFloat(unsigned channels, ConstBuffer<uint8_t> src)
 
 	return { dest, num_samples };
 }
-
-/**
- * Construct a 32 bit integer from four bytes.
- */
-static constexpr inline uint32_t
-Construct32(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
-{
-	return uint32_t(a) | (uint32_t(b) << 8) |
-		(uint32_t(c) << 16) | (uint32_t(d) << 24);
-}
-
-static constexpr inline uint32_t
-Dsd8To32Sample(const uint8_t *src, unsigned channels)
-{
-	return Construct32(src[0], src[channels],
-			   src[2 * channels], src[3 * channels]);
-}
-
-ConstBuffer<uint32_t>
-Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> _src)
-{
-	const size_t in_frames = _src.size / channels;
-	const size_t out_frames = in_frames / 4;
-	const size_t out_samples = out_frames * channels;
-
-	const uint8_t *src = _src.data;
-	uint32_t *const dest0 = buffer.GetT<uint32_t>(out_samples);
-	uint32_t *dest = dest0;
-
-	for (size_t i = 0; i < out_frames; ++i) {
-		for (size_t c = 0; c < channels; ++c)
-			*dest++ = Dsd8To32Sample(src++, channels);
-
-		src += 3 * channels;
-	}
-
-	return {dest0, out_samples};
-}

src/pcm/PcmDsd.hxx

@@ -48,10 +48,4 @@ public:
 				   ConstBuffer<uint8_t> src);
 };
 
-/**
- * Convert DSD_U8 to DSD_U32 (native endian).
- */
-ConstBuffer<uint32_t>
-Dsd8To32(PcmBuffer &buffer, unsigned channels, ConstBuffer<uint8_t> src);
-
 #endif

src/pcm/PcmExport.cxx

@@ -19,12 +19,15 @@
 
 #include "config.h"
 #include "PcmExport.hxx"
+#include "AudioFormat.hxx"
 #include "Order.hxx"
 #include "PcmPack.hxx"
 #include "util/ByteReverse.hxx"
 #include "util/ConstBuffer.hxx"
 
 #ifdef ENABLE_DSD
+#include "Dsd16.hxx"
+#include "Dsd32.hxx"
 #include "PcmDsd.hxx"
 #include "PcmDop.hxx"
 #endif
@@ -41,9 +44,15 @@ PcmExport::Open(SampleFormat sample_format, unsigned _channels,
 		: SampleFormat::UNDEFINED;
 
 #ifdef ENABLE_DSD
-	assert(!params.dsd_u32 || !params.dop);
+	assert((params.dsd_u16 + params.dsd_u32 + params.dop) <= 1);
 	assert(!params.dop || audio_valid_channel_count(_channels));
 
+	dsd_u16 = params.dsd_u16 && sample_format == SampleFormat::DSD;
+	if (dsd_u16)
+		/* after the conversion to DSD_U16, the DSD samples
+		   are stuffed inside fake 16 bit samples */
+		sample_format = SampleFormat::S16;
+
 	dsd_u32 = params.dsd_u32 && sample_format == SampleFormat::DSD;
 	if (dsd_u32)
 		/* after the conversion to DSD_U32, the DSD samples
@@ -55,8 +64,6 @@ PcmExport::Open(SampleFormat sample_format, unsigned _channels,
 		/* after the conversion to DoP, the DSD
 		   samples are stuffed inside fake 24 bit samples */
 		sample_format = SampleFormat::S24_P32;
-#else
-	(void)_channels;
 #endif
 
 	shift8 = params.shift8 && sample_format == SampleFormat::S24_P32;
@@ -84,6 +91,9 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
 		return audio_format.channels * 3;
 
 #ifdef ENABLE_DSD
+	if (dsd_u16)
+		return channels * 2;
+
 	if (dsd_u32)
 		return channels * 4;
 
@@ -98,6 +108,46 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
 	return audio_format.GetFrameSize();
 }
 
+unsigned
+PcmExport::Params::CalcOutputSampleRate(unsigned sample_rate) const
+{
+#ifdef ENABLE_DSD
+	if (dsd_u16)
+		/* DSD_U16 combines two 8-bit "samples" in one 16-bit
+		   "sample" */
+		sample_rate /= 2;
+
+	if (dsd_u32)
+		/* DSD_U32 combines four 8-bit "samples" in one 32-bit
+		   "sample" */
+		sample_rate /= 4;
+
+	if (dop)
+		/* DoP packs two 8-bit "samples" in one 24-bit
+		   "sample" */
+		sample_rate /= 2;
+#endif
+
+	return sample_rate;
+}
+
+unsigned
+PcmExport::Params::CalcInputSampleRate(unsigned sample_rate) const
+{
+#ifdef ENABLE_DSD
+	if (dsd_u16)
+		sample_rate *= 2;
+
+	if (dsd_u32)
+		sample_rate *= 4;
+
+	if (dop)
+		sample_rate *= 2;
+#endif
+
+	return sample_rate;
+}
+
 ConstBuffer<void>
 PcmExport::Export(ConstBuffer<void> data)
 {
@@ -106,6 +156,11 @@ PcmExport::Export(ConstBuffer<void> data)
 					  alsa_channel_order, channels);
 
 #ifdef ENABLE_DSD
+	if (dsd_u16)
+		data = Dsd8To16(dop_buffer, channels,
+				ConstBuffer<uint8_t>::FromVoid(data))
+			.ToVoid();
+
 	if (dsd_u32)
 		data = Dsd8To32(dop_buffer, channels,
 				ConstBuffer<uint8_t>::FromVoid(data))

src/pcm/PcmExport.hxx

@@ -21,28 +21,18 @@
 #define PCM_EXPORT_HXX
 
 #include "check.h"
+#include "SampleFormat.hxx"
 #include "PcmBuffer.hxx"
-#include "AudioFormat.hxx"
 
 template<typename T> struct ConstBuffer;
+struct AudioFormat;
 
 /**
  * An object that handles export of PCM samples to some instance
  * outside of MPD.  It has a few more options to tweak the binary
  * representation which are not supported by the pcm_convert library.
  */
-struct PcmExport {
+class PcmExport {
-	struct Params {
-		bool alsa_channel_order = false;
-#ifdef ENABLE_DSD
-		bool dsd_u32 = false;
-		bool dop = false;
-#endif
-		bool shift8 = false;
-		bool pack24 = false;
-		bool reverse_endian = false;
-	};
-
 	/**
 	 * This buffer is used to reorder channels.
 	 *
@@ -90,6 +80,11 @@ struct PcmExport {
 	SampleFormat alsa_channel_order;
 
 #ifdef ENABLE_DSD
+	/**
+	 * Convert DSD (U8) to DSD_U16?
+	 */
+	bool dsd_u16;
+
 	/**
 	 * Convert DSD (U8) to DSD_U32?
 	 */
@@ -121,6 +116,34 @@ struct PcmExport {
 	 */
 	uint8_t reverse_endian;
 
+public:
+	struct Params {
+		bool alsa_channel_order = false;
+#ifdef ENABLE_DSD
+		bool dsd_u16 = false;
+		bool dsd_u32 = false;
+		bool dop = false;
+#endif
+		bool shift8 = false;
+		bool pack24 = false;
+		bool reverse_endian = false;
+
+		/**
+		 * Calculate the output sample rate, given a specific input
+		 * sample rate.  Usually, both are the same; however, with
+		 * DSD_U32, four input bytes (= 4 * 8 bits) are combined to
+		 * one output word (32 bits), dividing the sample rate by 4.
+		 */
+		gcc_pure
+		unsigned CalcOutputSampleRate(unsigned input_sample_rate) const;
+
+		/**
+		 * The inverse of CalcOutputSampleRate().
+		 */
+		gcc_pure
+		unsigned CalcInputSampleRate(unsigned output_sample_rate) const;
+	};
+
 	/**
 	 * Open the object.
 	 *
@@ -134,6 +157,12 @@ struct PcmExport {
 	void Open(SampleFormat sample_format, unsigned channels,
 		  Params params);
 
+	/**
+	 * Reset the filter's state, e.g. drop/flush buffers.
+	 */
+	void Reset() {
+	}
+
 	/**
 	 * Calculate the size of one output frame.
 	 */

src/pcm/PcmFormat.hxx

@@ -20,7 +20,7 @@
 #ifndef MPD_PCM_FORMAT_HXX
 #define MPD_PCM_FORMAT_HXX
 
-#include "AudioFormat.hxx"
+#include "SampleFormat.hxx"
 
 #include <stdint.h>