release v0.18.12

Failure to read another chunk header is not fatal.  Continue to read
metadata.

.gitignore

@@ -31,6 +31,7 @@ libtool
 ltmain.sh
 missing
 mkinstalldirs
+/test-driver
 mpd
 mpd.service
 stamp-h1

INSTALL

@@ -3,9 +3,11 @@
 
 Introduction
 ------------
+
 This document is a very small amount of documentation about what is needed to 
-install MPD. If more information is desired see the community wiki at 
-http://mpd.wikia.com.
+install MPD.  If more information is desired, read the user manual:
+
+ http://www.musicpd.org/doc/user/
 
 Dependencies
 ------------
@@ -161,13 +163,9 @@ Get the latest release from of MPD from <http://www.musicpd.org/>.
 Compile
 -------
 
-1) unzip and untar the archive
+1) unpack the archive
 
-$ tar zxvf mpd-x.x.x.tar.gz
-
-or
-
-$ tar jxvf mpd-x.x.x.tar.bz2
+$ tar xf mpd-x.x.x.tar.xz
 
 2) change to directory created
 
@@ -208,6 +206,5 @@ Using MPD
 ---------
 
 You can download many different interfaces for MPD at 
-	<http://mpd.wikia.com/wiki/Clients>
 
-MPD can be interfaced directly using telnet (see COMMANDS, if you are brave).
+ http://www.musicpd.org/clients/

Makefile.am

@@ -1,5 +1,5 @@
 ACLOCAL_AMFLAGS = -I m4
-AUTOMAKE_OPTIONS = foreign 1.11 dist-bzip2 dist-xz subdir-objects
+AUTOMAKE_OPTIONS = foreign 1.11 dist-xz subdir-objects
 
 AM_CPPFLAGS += -I$(srcdir)/src $(GLIB_CFLAGS)
 
@@ -181,6 +181,7 @@ src_mpd_SOURCES = \
 	src/PlaylistInfo.hxx \
 	src/PlaylistDatabase.cxx src/PlaylistDatabase.hxx \
 	src/PlaylistUpdate.cxx \
+	src/BulkEdit.hxx \
 	src/IdTable.hxx \
 	src/Queue.cxx src/Queue.hxx \
 	src/QueuePrint.cxx src/QueuePrint.hxx \
@@ -1061,6 +1062,7 @@ C_TESTS = \
 	test/test_util \
 	test/test_byte_reverse \
 	test/test_mixramp \
+	test/test_icy_parser \
 	test/test_pcm \
 	test/test_queue_priority
 
@@ -1217,6 +1219,7 @@ test_dump_playlist_LDADD = \
 	libpcm.a \
 	$(GLIB_LIBS)
 test_dump_playlist_SOURCES = test/dump_playlist.cxx \
+	test/FakeDecoderAPI.cxx \
 	$(DECODER_SRC) \
 	src/Log.cxx \
 	src/IOThread.cxx \
@@ -1270,6 +1273,7 @@ test_read_tags_LDADD = \
 	libutil.a \
 	$(GLIB_LIBS)
 test_read_tags_SOURCES = test/read_tags.cxx \
+	test/FakeDecoderAPI.cxx \
 	src/Log.cxx \
 	src/IOThread.cxx \
 	src/ReplayGainInfo.cxx \
@@ -1496,6 +1500,16 @@ test_test_mixramp_LDADD = \
 	$(GLIB_LIBS) \
 	$(CPPUNIT_LIBS)
 
+test_test_icy_parser_SOURCES = \
+	src/Log.cxx \
+	test/test_icy_parser.cxx
+test_test_icy_parser_CPPFLAGS = $(AM_CPPFLAGS) $(CPPUNIT_CFLAGS) -DCPPUNIT_HAVE_RTTI=0
+test_test_icy_parser_CXXFLAGS = $(AM_CXXFLAGS) -Wno-error=deprecated-declarations
+test_test_icy_parser_LDADD = \
+	libtag.a \
+	$(GLIB_LIBS) \
+	$(CPPUNIT_LIBS)
+
 test_test_pcm_SOURCES = \
 	test/test_pcm_util.hxx \
 	test/test_pcm_dither.cxx \

NEWS

@@ -1,3 +1,103 @@
+ver 0.18.12 (2014/07/30)
+* database
+  - proxy: fix build failure with libmpdclient 2.2
+  - proxy: fix add/search and other commands with libmpdclient < 2.9
+* decoder
+  - audiofile: improve responsiveness
+  - audiofile: fix WAV stream playback
+  - dsdiff, dsf: fix stream playback
+  - dsdiff: fix metadata parser bug (uninitialized variables)
+  - faad: estimate song duration for remote files
+  - sndfile: improve responsiveness
+* randomize next song when enabling "random" mode while not playing
+* randomize next song when adding to single-song queue
+
+ver 0.18.11 (2014/05/12)
+* decoder
+  - opus: fix missing song length on high-latency files
+* fix race condition when using GLib event loop (non-Linux)
+
+ver 0.18.10 (2014/04/10)
+* decoder
+  - ffmpeg: fix seeking bug
+  - ffmpeg: handle unknown stream start time
+  - gme: fix memory leak
+  - sndfile: work around libsndfile bug on partial read
+* don't interrupt playback when current song gets deleted
+
+ver 0.18.9 (2014/03/02)
+* protocol
+  - "findadd" requires the "add" permission
+* output
+  - alsa: improved workaround for noise after manual song change
+* decoder
+  - vorbis: fix linker failure when libvorbis/libogg are static
+* encoder
+  - vorbis: fix another linker failure
+* output
+  - pipe: fix hanging child process due to blocked signals
+* fix build failure due to missing signal.h include
+
+ver 0.18.8 (2014/02/07)
+* decoder
+  - ffmpeg: support libav v10_alpha1
+* encoder
+  - vorbis: fix linker failure
+* output
+  - roar: documentation
+* more robust Icy-Metadata parser
+* fix Solaris build failure
+
+ver 0.18.7 (2014/01/13)
+* playlist
+  - pls: fix crash after parser error
+  - soundcloud: fix build failure with libyajl 2.0.1
+* decoder
+  - faad: fix memory leak
+  - mpcdec: reject libmpcdec SV7 in configure script
+* daemon: don't initialize supplementary groups when already running
+  as the configured user
+
+ver 0.18.6 (2013/12/24)
+* input
+  - cdio_paranoia: support libcdio-paranoia 0.90
+* tags
+  - riff: recognize upper-case "ID3" chunk name
+* decoder
+  - ffmpeg: use relative timestamps
+* output
+  - openal: fix build failure on Mac OS X
+  - osx: fix build failure
+* mixer
+  - alsa: fix build failure with uClibc
+* fix replay gain during cross-fade
+* accept files without metadata
+
+ver 0.18.5 (2013/11/23)
+* configuration
+  - fix crash when db_file is configured without music_directory
+  - fix crash on "stats" without db_file/music_directory
+* database
+  - proxy: auto-reload statistics
+  - proxy: provide "db_update" in "stats" response
+* input
+  - curl: work around stream resume bug (fixed in libcurl 7.32.0)
+* decoder
+  - fluidsynth: auto-detect by default
+* clip 24 bit data from libsamplerate
+* fix ia64, mipsel and other little-endian architectures
+* fix build failures due to missing includes
+* fix build failure with static libmpdclient
+
+ver 0.18.4 (2013/11/13)
+* decoder
+  - dsdiff: fix byte order bug
+* fix build failures due to missing includes
+* libc++ compatibility
+
+ver 0.18.3 (2013/11/08)
+* fix stuck MPD after song change (0.18.2 regression)
+
 ver 0.18.2 (2013/11/07)
 * protocol:
   - "close" flushes the output buffer

configure.ac

@@ -1,6 +1,6 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.18.2, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.18.12, mpd-devel@musicpd.org)
 
 VERSION_MAJOR=0
 VERSION_MINOR=18
@@ -8,7 +8,7 @@ VERSION_REVISION=0
 VERSION_EXTRA=0
 
 AC_CONFIG_SRCDIR([src/Main.cxx])
-AM_INIT_AUTOMAKE([foreign 1.11 dist-bzip2 dist-xz subdir-objects])
+AM_INIT_AUTOMAKE([foreign 1.11 dist-xz subdir-objects])
 AM_SILENT_RULES
 AC_CONFIG_HEADERS(config.h)
 AC_CONFIG_MACRO_DIR([m4])
@@ -732,7 +732,7 @@ dnl --------------------------------- Soundcloud ------------------------------
 if test x$enable_soundcloud != xno; then
 	PKG_CHECK_MODULES([YAJL], [yajl >= 2.0],
 		[found_soundcloud=yes],
-		AC_CHECK_LIB([yajl], [yajl_alloc],
+		AC_CHECK_LIB([yajl], [yajl_parse_complete],
 			[found_soundcloud=yes YAJL_CFLAGS=-DHAVE_YAJL1 YAJL_LIBS=-lyajl],
 			[found_soundcloud=no]))
 fi
@@ -749,6 +749,7 @@ MPD_AUTO_PKG(cdio_paranoia, CDIO_PARANOIA, [libcdio_paranoia],
 if test x$enable_cdio_paranoia = xyes; then
 	AC_DEFINE([ENABLE_CDIO_PARANOIA], 1,
 		[Define to enable libcdio_paranoia support])
+	AC_CHECK_HEADERS(cdio/paranoia/paranoia.h)
 fi
 
 AM_CONDITIONAL(ENABLE_CDIO_PARANOIA, test x$enable_cdio_paranoia = xyes)
@@ -874,10 +875,12 @@ AM_CONDITIONAL(HAVE_FLAC, test x$enable_flac = xyes)
 enable_flac_encoder=$enable_flac
 
 dnl -------------------------------- FluidSynth -------------------------------
+
+MPD_AUTO_PKG(fluidsynth, FLUIDSYNTH, [fluidsynth >= 1.1],
+	[fluidsynth decoder], [fluidsynth not found])
+
 if test x$enable_fluidsynth = xyes; then
-	PKG_CHECK_MODULES(FLUIDSYNTH, [fluidsynth >= 1.1],
-		AC_DEFINE(ENABLE_FLUIDSYNTH, 1, [Define for fluidsynth support]),
-		enable_fluidsynth=no)
+	AC_DEFINE(ENABLE_FLUIDSYNTH, 1, [Define for fluidsynth support])
 fi
 
 AM_CONDITIONAL(ENABLE_FLUIDSYNTH, test x$enable_fluidsynth = xyes)
@@ -957,7 +960,7 @@ AM_CONDITIONAL(ENABLE_SNDFILE, test x$enable_sndfile = xyes)
 
 dnl --------------------------------- musepack --------------------------------
 
-MPD_AUTO_LIB(mpc, MPCDEC, mpcdec, main, [-lmpcdec], [],
+MPD_AUTO_LIB(mpc, MPCDEC, mpcdec, mpc_demux_init, [-lmpcdec], [],
 	[mpcdec], [libmpcdec not found])
 if test x$enable_mpc = xyes; then
 	AC_DEFINE(HAVE_MPCDEC, 1, [Define to use libmpcdec for MPC decoding])
@@ -1020,7 +1023,7 @@ if test x$enable_tremor = xyes; then
 	fi
 fi
 
-MPD_AUTO_PKG(vorbis, VORBIS, [vorbis vorbisfile ogg],
+MPD_AUTO_PKG(vorbis, VORBIS, [vorbisfile vorbis ogg],
 	[Ogg Vorbis decoder], [libvorbis not found])
 if test x$enable_vorbis = xyes; then
 	AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support])
@@ -1136,7 +1139,7 @@ fi
 AM_CONDITIONAL(ENABLE_FLAC_ENCODER, test x$enable_flac_encoder = xyes)
 
 dnl ---------------------------- Ogg Vorbis Encoder ---------------------------
-MPD_AUTO_PKG(vorbis_encoder, VORBISENC, [vorbisenc],
+MPD_AUTO_PKG(vorbis_encoder, VORBISENC, [vorbisenc vorbis ogg],
 	[Ogg Vorbis encoder], [libvorbisenc not found])
 
 if test x$enable_vorbis_encoder = xyes; then

doc/developer.xml

@@ -68,12 +68,28 @@ foo(const char *abc, int xyz)
   <chapter>
     <title>Hacking The Source</title>
 
+    <para>
+      MPD sources are managed in a git repository on <ulink
+      url="http://git.musicpd.org/">git.musicpd.org</ulink>.
+    </para>
+
     <para>
       Always write your code against the latest git:
     </para>
 
     <programlisting>git clone git://git.musicpd.org/master/mpd.git</programlisting>
 
+    <para>
+      If you already have a clone, update it:
+    </para>
+
+    <programlisting>git pull --rebase git://git.musicpd.org/master/mpd.git master</programlisting>
+
+    <para>
+      You can do without "--rebase", but we recommend that you rebase
+      your repository on the "master" repository all the time.
+    </para>
+
     <para>
       Configure with the options <option>--enable-debug
       --enable-werror</option>.  Enable as many plugins as possible,
@@ -83,8 +99,55 @@ foo(const char *abc, int xyz)
     <para>
       Don't mix several changes in one single patch.  Create a
       separate patch for every change.  Tools like
-      <application>stgit</application> help you with that.
+      <application>stgit</application> help you with that.  This way,
+      we can review your patches more easily, and we can pick the
+      patches we like most first.
     </para>
+
+
+    <section>
+      <title> Basic stgit usage</title>
+
+      <para>
+        stgit allows you to create a set of patches and refine all of
+        them: you can go back to any patch at any time, and re-edit it
+        (both the code and the commit message). You can reorder
+        patches and insert new patches at any position. It encourages
+        creating separate patches for tiny changes.
+      </para>
+
+      <para>
+        stgit needs to be initialized on a git repository: stg init
+      </para>
+
+      <para>
+        Before you edit the code, create a patch: stg new
+        my-patch-name (stgit now asks you for the commit message).
+      </para>
+
+      <para>
+        Now edit the code. Once you're finished, you have to "refresh"
+        the patch, i.e. your edits are incorporated into the patch you
+        have created: stg refresh
+      </para>
+
+      <para>
+        You may now continue editing the same patch, and refresh it as
+        often as you like. Create more patches, edit and refresh them.
+      </para>
+
+      <para>
+        To view the list of patches, type stg series. To go back to a
+        specific patch, type stg goto my-patch-name; now you can
+        re-edit it (don't forget stg refresh when you're finished with
+        that patch).
+      </para>
+
+      <para>
+        When the whole patch series is finished, convert stgit patches
+        to git commits: stg commit
+      </para>
+    </section>
   </chapter>
 
   <chapter>
@@ -92,7 +155,7 @@ foo(const char *abc, int xyz)
 
     <para>
       Send your patches to the mailing list:
-      musicpd-dev-team@lists.sourceforge.net
+      mpd-devel@musicpd.org
     </para>
   </chapter>
 </book>

doc/mpd.conf.5

@@ -363,137 +363,6 @@ errors on bandwidth-limited devices.  Some users have reported good results
 with this set to 50000, but not all devices support values this high.  Most
 users do not need to change this.  The default is 256000000 / sample_rate(kHz),
 or 5804 microseconds for CD-quality audio.
-.SH OPTIONAL OSS OUTPUT PARAMETERS
-.TP
-.B device <dev>
-This specifies the device to use for audio output.  The default is "/dev/dsp".
-.TP
-.B mixer_device <mixer dev>
-This specifies which mixer to use.  The default is "/dev/mixer".
-.TP
-.B mixer_control <mixer ctrl>
-This specifies which mixer control to use (sometimes referred to as the
-"device").  The default is to use the main PCM mixer.  An example is "Pcm".
-.SH OPTIONAL PULSE OUTPUT PARAMETERS
-.TP
-.B server <server list>
-A space separated list of servers to try to connect to.  See
-<\fBhttp://www.pulseaudio.org/wiki/ServerStrings\fP> for more details.  The
-default is to let PulseAudio choose a server.
-If you specify more than one server name, MPD tries to connect to one
-after another until it successfully establishes a connection.
-.TP
-.B sink <sink>
-The sink to output to.  The default is to let PulseAudio choose a sink.
-.SH OPTIONAL JACK OUTPUT PARAMETERS
-.TP
-.B client_name <name>
-The client name to use when connecting to JACK.  The output ports <name>:left
-and <name>:right will also be created for the left and right channels,
-respectively.
-.TP
-.B ports <left_port,right_port>
-This specifies the left and right ports to connect to for the left and right
-channels, respectively.  The default is to let JACK choose a pair of ports.
-.TP
-.B ringbuffer_size <size in bytes>
-This specifies the size of the ringbuffer in bytes.  The default is 32768.
-.SH OPTIONAL AO OUTPUT PARAMETERS
-.TP
-.B driver <driver>
-This specifies the libao driver to use for audio output.  Possible values
-depend on what libao drivers are available.  See
-<\fBhttp://www.xiph.org/ao/doc/drivers.html\fP> for information on some
-commonly used drivers.  Typical values for Linux include "oss" and "alsa09".
-The default is "default", which causes libao to select an appropriate plugin.
-.TP
-.B options <opts>
-This specifies the options to use for the selected libao driver.  For oss, the
-only option available is "dsp".  For alsa09, the available options are: "dev",
-"buf_size", and "periods".  See <\fBhttp://www.xiph.org/ao/doc/drivers.html\fP>
-for available options for some commonly used drivers.  Options are assigned
-using "=", and ";" is used to separate options.  An example for oss:
-"dsp=/dev/dsp".  An example for alsa09: "dev=hw:0,0;buf_size=4096".  The
-default is "".
-.TP
-.B write_size <size in bytes>
-This specifies how many bytes to write to the audio device at once.  This
-parameter is to work around a bug in older versions of libao on sound cards
-with very small buffers.  The default is 1024.
-.SH REQUIRED FIFO OUTPUT PARAMETERS
-.TP
-.B path <path>
-This specifies the path of the FIFO to output to.  Must be an absolute path.
-If the path does not exist it will be created when mpd is started, and removed
-when mpd is stopped.  The FIFO will be created with the same user and group as
-mpd is running as.  Default permissions can be modified by using the builtin
-shell command "umask".  If a FIFO already exists at the specified path it will
-be reused, and will \fBnot\fP be removed when mpd is stopped.  You can use the
-"mkfifo" command to create this, and then you may modify the permissions to
-your liking.
-.SH REQUIRED SHOUT OUTPUT PARAMETERS
-.TP
-.B name <name>
-This specifies not only the unique audio output name, but also the stream
-title.
-.TP
-.B host <hostname>
-This specifies the hostname of the icecast server to connect to.
-.TP
-.B port <port>
-This specifies the port of the icecast server to connect to.
-.TP
-.B mount <mountpoint>
-This specifies the icecast mountpoint to use.
-.TP
-.B password <password>
-This specifies the password to use when logging in to the icecast server.
-.TP
-.B quality <quality>
-This specifies the encoding quality to use.  The value must be between 0
-and 10.  Fractional values, such as 2.5, are permitted.  Either the quality or
-the bitrate parameter must be specified, but not both.  For Ogg, a
-higher quality number produces higher quality output.  For MP3, it's
-just the opposite, with lower numbers producing higher quality output.
-.TP
-.B bitrate <kbps>
-This specifies the bitrate to use for encoding.  Either the quality or the
-bitrate parameter must be specified, but not both.
-.TP
-.B format <sample_rate:bits:channels>
-This specifies the sample rate, bits per sample, and number of channels to use
-for encoding.
-.SH OPTIONAL SHOUT OUTPUT PARAMETERS
-.TP
-.B encoding <encoding>
-This specifies which output encoding to use.  Should be either "ogg"
-or "mp3",  "mp3" is needed for shoutcast streaming. The default is "ogg".
-.TP
-.B protocol <protocol>
-This specifies the protocol that wil be used to connect to the
-icecast/shoutcast server. The options are "shoutcast", "icecast1" and
-"icecast2". The default is "icecast2".
-.TP
-.B user <username>
-This specifies the username to use when logging in to the icecast server.  The
-default is "source".
-.TP
-.B public <yes or no>
-This specifies whether to request that the stream be listed in all public
-stream directories that the icecast server knows about.  The default is no.
-.TP
-.B timeout <seconds>
-This specifies the number of seconds to wait before giving up on trying to
-connect to the icecast server.  The default is 2 seconds.
-.TP
-.B description <description>
-This specifies a description of the stream.
-.TP
-.B url <url>
-This specifies a URL associated with the stream.
-.TP
-.B genre <genre>
-This specifies the genre(s) of the stream.
 .SH FILES
 .TP
 .BI ~/.mpdconf

doc/mpdconf.example

@@ -1,5 +1,5 @@
-# An example configuration file for MPD
-# See the mpd.conf man page for a more detailed description of each parameter.
+# An example configuration file for MPD.
+# Read the user manual for documentation: http://www.musicpd.org/doc/user/
 
 
 # Files and directories #######################################################
@@ -204,16 +204,12 @@ input {
 # blocks. Setting this block is optional, though the server will only attempt
 # autodetection for one sound card.
 #
-# See <http://mpd.wikia.com/wiki/Configuration#Audio_Outputs> for examples of 
-# other audio outputs.
-#
 # An example of an ALSA output:
 #
 #audio_output {
 #	type		"alsa"
 #	name		"My ALSA Device"
 ##	device		"hw:0,0"	# optional
-##	format		"44100:16:2"	# optional
 ##	mixer_type      "hardware"	# optional
 ##	mixer_device	"default"	# optional
 ##	mixer_control	"PCM"		# optional
@@ -226,7 +222,6 @@ input {
 #	type		"oss"
 #	name		"My OSS Device"
 ##	device		"/dev/dsp"	# optional
-##	format		"44100:16:2"	# optional
 ##	mixer_type      "hardware"	# optional
 ##	mixer_device	"/dev/mixer"	# optional
 ##	mixer_control	"PCM"		# optional
@@ -330,12 +325,6 @@ input {
 #	mixer_type      "none"			# optional
 #}
 #
-# This setting will change all decoded audio to be converted to the specified
-# format before being passed to the audio outputs. By default, this setting is
-# disabled.
-#
-#audio_output_format		"44100:16:2"
-#
 # If MPD has been compiled with libsamplerate support, this setting specifies 
 # the sample rate converter to use.  Possible values can be found in the 
 # mpd.conf man page or the libsamplerate documentation. By default, this is

doc/user.xml

@@ -67,11 +67,11 @@
 
       <para>
         Download the source tarball from <ulink
-        url="http://mpd.wikia.com/wiki/Server">the MPD home
+        url="http://www.musicpd.org/download.html">the MPD home
         page</ulink> and unpack it:
       </para>
 
-      <programlisting>tar xjf mpd-version.tar.bz
+      <programlisting>tar xf mpd-version.tar.xz
 cd mpd-version</programlisting>
 
       <para>
@@ -630,7 +630,7 @@ systemctl start mpd.socket</programlisting>
       </para>
 
       <para>
-        The <ulink url="http://mpd.wikia.com/wiki/Clients">MPD
+        The <ulink url="http://www.musicpd.org/clients/">MPD
         Wiki</ulink> contains an extensive list of clients to choose
         from.
       </para>
@@ -829,7 +829,7 @@ systemctl start mpd.socket</programlisting>
             <tbody>
               <row>
                 <entry>
-                  <varname>default_bute_order</varname>
+                  <varname>default_byte_order</varname>
                 <parameter>little_endian|big_endian</parameter>
                 </entry>
                 <entry>
@@ -1398,6 +1398,59 @@ systemctl start mpd.socket</programlisting>
           The <varname>ao</varname> plugin uses the portable
           <filename>libao</filename> library.
         </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>driver</varname>
+                  <parameter>D</parameter>
+                </entry>
+                <entry>
+                  The <filename>libao</filename> driver to use for
+                  audio output.  Possible values depend on what libao
+                  drivers are available.  See <ulink
+                  url="http://www.xiph.org/ao/doc/drivers.html">http://www.xiph.org/ao/doc/drivers.html</ulink>
+                  for information on some commonly used drivers.
+                  Typical values for Linux include "oss" and "alsa09".
+                  The default is "default", which causes libao to
+                  select an appropriate plugin.
+                </entry>
+              </row>
+
+              <row>
+                <entry>
+                  <varname>options</varname>
+                  <parameter>O</parameter>
+                </entry>
+                <entry>
+                  Options to pass to the selected
+                  <filename>libao</filename> driver.
+                </entry>
+              </row>
+
+              <row>
+                <entry>
+                  <varname>write_size</varname>
+                  <parameter>O</parameter>
+                </entry>
+                <entry>
+                  This specifies how many bytes to write to the audio
+                  device at once.  This parameter is to work around a
+                  bug in older versions of libao on sound cards with
+                  very small buffers.  The default is 1024.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
       </section>
 
       <section>
@@ -1408,6 +1461,38 @@ systemctl start mpd.socket</programlisting>
           FIFO (First In, First Out) file.  The data can be read by
           another program.
         </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>path</varname>
+                  <parameter>P</parameter>
+                </entry>
+                <entry>
+                  This specifies the path of the FIFO to write to.
+                  Must be an absolute path.  If the path does not
+                  exist, it will be created when MPD is started, and
+                  removed when MPD is stopped.  The FIFO will be
+                  created with the same user and group as MPD is
+                  running as.  Default permissions can be modified by
+                  using the builtin shell command "umask".  If a FIFO
+                  already exists at the specified path it will be
+                  reused, and will not be removed when MPD is stopped.
+                  You can use the "mkfifo" command to create this, and
+                  then you may modify the permissions to your liking.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
       </section>
 
       <section>
@@ -1767,6 +1852,51 @@ systemctl start mpd.socket</programlisting>
         </informaltable>
       </section>
 
+      <section>
+        <title><varname>roar</varname></title>
+
+        <para>
+          The <varname>roar</varname> plugin connects to a <ulink
+          url="http://roaraudio.keep-cool.org/">RoarAudio</ulink>
+          server.
+        </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>server</varname>
+                  <parameter>HOSTNAME</parameter>
+                </entry>
+                <entry>
+                  The host name of the RoarAudio server.  If not
+                  specified, then MPD will connect to the default
+                  locations.
+                </entry>
+              </row>
+
+              <row>
+                <entry>
+                  <varname>role</varname>
+                  <parameter>ROLE</parameter>
+                </entry>
+                <entry>
+                  The "role" that MPD registers itself as in the
+                  RoarAudio server.  The default is "music".
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
+      </section>
+
       <section>
         <title><varname>recorder</varname></title>
 
@@ -1883,6 +2013,18 @@ systemctl start mpd.socket</programlisting>
                   Defaults to 2 seconds.
                 </entry>
               </row>
+              <row>
+                <entry>
+                  <varname>protocol</varname>
+                  <parameter>icecast2|icecast1|shoutcast</parameter>
+                </entry>
+                <entry>
+                  Specifies the protocol that wil be used to connect
+                  to the icecast/shoutcast server.  The default
+                  is "<parameter>icecast2</parameter>".
+
+                </entry>
+              </row>
               <row>
                 <entry>
                   <varname>mount</varname>

src/BulkEdit.hxx

@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2003-2014 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_BULK_EDIT_HXX
+#define MPD_BULK_EDIT_HXX
+
+#include "Partition.hxx"
+
+/**
+ * Begin a "bulk edit" and commit it automatically.
+ */
+class ScopeBulkEdit {
+	Partition &partition;
+
+public:
+	ScopeBulkEdit(Partition &_partition):partition(_partition) {
+		partition.playlist.BeginBulk();
+	}
+
+	~ScopeBulkEdit() {
+		partition.playlist.CommitBulk(partition.pc);
+	}
+};
+
+#endif

src/Daemon.cxx

@@ -52,7 +52,7 @@ static char *user_name;
 static uid_t user_uid = (uid_t)-1;
 
 /** the Unix group id which MPD runs as */
-static gid_t user_gid = (pid_t)-1;
+static gid_t user_gid = (gid_t)-1;
 
 /** the absolute path of the pidfile */
 static AllocatedPath pidfile = AllocatedPath::Null();
@@ -106,18 +106,21 @@ daemonize_set_user(void)
 		return;
 
 	/* set gid */
-	if (user_gid != (gid_t)-1 && user_gid != getgid()) {
-		if (setgid(user_gid) == -1) {
-			FormatFatalSystemError("Failed to set group %d",
-					       (int)user_gid);
-		}
+	if (user_gid != (gid_t)-1 && user_gid != getgid() &&
+	    setgid(user_gid) == -1) {
+		FormatFatalSystemError("Failed to set group %d",
+				       (int)user_gid);
 	}
 
 #ifdef _BSD_SOURCE
-	/* init suplementary groups
+	/* init supplementary groups
 	 * (must be done before we change our uid)
 	 */
-	if (!had_group && initgroups(user_name, user_gid) == -1) {
+	if (!had_group &&
+	    /* no need to set the new user's supplementary groups if
+	       we are already this user */
+	    user_uid != getuid() &&
+	    initgroups(user_name, user_gid) == -1) {
 		FormatFatalSystemError("Failed to set supplementary groups "
 				       "of user \"%s\"",
 				       user_name);

src/DatabaseGlue.cxx

@@ -148,12 +148,12 @@ DatabaseGlobalOpen(Error &error)
 	return true;
 }
 
-time_t
-db_get_mtime(void)
+bool
+db_exists()
 {
 	assert(db != nullptr);
 	assert(db_is_open);
 	assert(db_is_simple());
 
-	return ((SimpleDatabase *)db)->GetLastModified();
+	return ((SimpleDatabase *)db)->GetUpdateStamp() > 0;
 }

src/DatabasePlugin.hxx

@@ -30,6 +30,8 @@
 #include "tag/TagType.h"
 #include "Compiler.h"
 
+#include <time.h>
+
 struct config_param;
 struct DatabaseSelection;
 struct db_visitor;
@@ -132,6 +134,13 @@ public:
 	virtual bool GetStats(const DatabaseSelection &selection,
 			      DatabaseStats &stats,
 			      Error &error) const = 0;
+
+	/**
+	 * Returns the time stamp of the last database update.
+	 * Returns 0 if that is not not known/available.
+	 */
+	gcc_pure
+	virtual time_t GetUpdateStamp() const = 0;
 };
 
 struct DatabasePlugin {

src/DatabaseSelection.cxx

@@ -30,6 +30,18 @@ DatabaseSelection::DatabaseSelection(const char *_uri, bool _recursive,
 		uri = filter->GetBase();
 }
 
+bool
+DatabaseSelection::IsEmpty() const
+{
+	return uri.empty() && (filter == nullptr || filter->IsEmpty());
+}
+
+bool
+DatabaseSelection::HasOtherThanBase() const
+{
+	return filter != nullptr && filter->HasOtherThanBase();
+}
+
 bool
 DatabaseSelection::Match(const Song &song) const
 {

src/DatabaseSelection.hxx

@@ -44,6 +44,15 @@ struct DatabaseSelection {
 	DatabaseSelection(const char *_uri, bool _recursive,
 			  const SongFilter *_filter=nullptr);
 
+	gcc_pure
+	bool IsEmpty() const;
+
+	/**
+	 * Does this selection contain constraints other than "base"?
+	 */
+	gcc_pure
+	bool HasOtherThanBase() const;
+
 	gcc_pure
 	bool Match(const Song &song) const;
 };

src/DatabaseSimple.hxx

@@ -62,25 +62,13 @@ db_get_directory(const char *name);
 bool
 db_save(Error &error);
 
-/**
- * May only be used if db_is_simple() returns true.
- */
-gcc_pure
-time_t
-db_get_mtime(void);
-
 /**
  * Returns true if there is a valid database file on the disk.
  *
  * May only be used if db_is_simple() returns true.
  */
 gcc_pure
-static inline bool
-db_exists(void)
-{
-	/* mtime is set only if the database file was loaded or saved
-	   successfully */
-	return db_get_mtime() > 0;
-}
+bool
+db_exists();
 
 #endif

src/DecoderAPI.cxx

@@ -35,6 +35,7 @@
 #include <assert.h>
 #include <stdlib.h>
 #include <string.h>
+#include <math.h>
 
 void
 decoder_initialized(Decoder &decoder,
@@ -283,14 +284,48 @@ decoder_read(Decoder *decoder,
 	assert(nbytes == 0 || !error.IsDefined());
 	assert(nbytes > 0 || error.IsDefined() || is.IsEOF());
 
+	is.Unlock();
+
 	if (gcc_unlikely(nbytes == 0 && error.IsDefined()))
 		LogError(error);
 
-	is.Unlock();
-
 	return nbytes;
 }
 
+bool
+decoder_read_full(Decoder *decoder, InputStream &is,
+		  void *_buffer, size_t size)
+{
+	uint8_t *buffer = (uint8_t *)_buffer;
+
+	while (size > 0) {
+		size_t nbytes = decoder_read(decoder, is, buffer, size);
+		if (nbytes == 0)
+			return false;
+
+		buffer += nbytes;
+		size -= nbytes;
+	}
+
+	return true;
+}
+
+bool
+decoder_skip(Decoder *decoder, InputStream &is, size_t size)
+{
+	while (size > 0) {
+		char buffer[1024];
+		size_t nbytes = decoder_read(decoder, is, buffer,
+					     std::min(sizeof(buffer), size));
+		if (nbytes == 0)
+			return false;
+
+		size -= nbytes;
+	}
+
+	return true;
+}
+
 void
 decoder_timestamp(Decoder &decoder, double t)
 {

src/DecoderAPI.hxx

@@ -112,6 +112,25 @@ decoder_read(Decoder &decoder, InputStream &is,
 	return decoder_read(&decoder, is, buffer, length);
 }
 
+/**
+ * Blocking read from the input stream.  Attempts to fill the buffer
+ * completely; there is no partial result.
+ *
+ * @return true on success, false on error or command or not enough
+ * data
+ */
+bool
+decoder_read_full(Decoder *decoder, InputStream &is,
+		  void *buffer, size_t size);
+
+/**
+ * Skip data on the #InputStream.
+ *
+ * @return true on success, false on error or command
+ */
+bool
+decoder_skip(Decoder *decoder, InputStream &is, size_t size);
+
 /**
  * Sets the time stamp for the next data chunk [seconds].  The MPD
  * core automatically counts it up, and a decoder plugin only needs to

src/DecoderBuffer.cxx

@@ -70,6 +70,12 @@ decoder_buffer_free(DecoderBuffer *buffer)
 	g_free(buffer);
 }
 
+const InputStream &
+decoder_buffer_get_stream(const DecoderBuffer *buffer)
+{
+	return *buffer->is;
+}
+
 bool
 decoder_buffer_is_empty(const DecoderBuffer *buffer)
 {
@@ -82,6 +88,12 @@ decoder_buffer_is_full(const DecoderBuffer *buffer)
 	return buffer->consumed == 0 && buffer->length == buffer->size;
 }
 
+void
+decoder_buffer_clear(DecoderBuffer *buffer)
+{
+	buffer->length = buffer->consumed = 0;
+}
+
 static void
 decoder_buffer_shift(DecoderBuffer *buffer)
 {
@@ -118,6 +130,12 @@ decoder_buffer_fill(DecoderBuffer *buffer)
 	return true;
 }
 
+size_t
+decoder_buffer_available(const DecoderBuffer *buffer)
+{
+	return buffer->length - buffer->consumed;;
+}
+
 const void *
 decoder_buffer_read(const DecoderBuffer *buffer, size_t *length_r)
 {

src/DecoderBuffer.hxx

@@ -20,6 +20,8 @@
 #ifndef MPD_DECODER_BUFFER_HXX
 #define MPD_DECODER_BUFFER_HXX
 
+#include "Compiler.h"
+
 #include <stddef.h>
 
 /**
@@ -50,12 +52,21 @@ decoder_buffer_new(Decoder *decoder, InputStream &is,
 void
 decoder_buffer_free(DecoderBuffer *buffer);
 
+gcc_pure
+const InputStream &
+decoder_buffer_get_stream(const DecoderBuffer *buffer);
+
+gcc_pure
 bool
 decoder_buffer_is_empty(const DecoderBuffer *buffer);
 
+gcc_pure
 bool
 decoder_buffer_is_full(const DecoderBuffer *buffer);
 
+void
+decoder_buffer_clear(DecoderBuffer *buffer);
+
 /**
  * Read data from the input_stream and append it to the buffer.
  *
@@ -66,6 +77,13 @@ decoder_buffer_is_full(const DecoderBuffer *buffer);
 bool
 decoder_buffer_fill(DecoderBuffer *buffer);
 
+/**
+ * How many bytes are stored in the buffer?
+ */
+gcc_pure
+size_t
+decoder_buffer_available(const DecoderBuffer *buffer);
+
 /**
  * Reads data from the buffer.  This data is not yet consumed, you
  * have to call decoder_buffer_consume() to do that.  The returned

src/DecoderInternal.cxx

@@ -42,20 +42,12 @@ Decoder::~Decoder()
  * one.
  */
 static DecoderCommand
-need_chunks(DecoderControl &dc, bool do_wait)
+need_chunks(DecoderControl &dc)
 {
-	if (dc.command == DecoderCommand::STOP ||
-	    dc.command == DecoderCommand::SEEK)
-		return dc.command;
-
-	if (do_wait) {
+	if (dc.command == DecoderCommand::NONE)
 		dc.Wait();
-		dc.client_cond.signal();
 
-		return dc.command;
-	}
-
-	return DecoderCommand::NONE;
+	return dc.command;
 }
 
 struct music_chunk *
@@ -80,7 +72,7 @@ decoder_get_chunk(Decoder &decoder)
 		}
 
 		dc.Lock();
-		cmd = need_chunks(dc, true);
+		cmd = need_chunks(dc);
 		dc.Unlock();
 	} while (cmd == DecoderCommand::NONE);
 

src/DecoderThread.cxx

@@ -38,6 +38,8 @@
 #include "tag/ApeReplayGain.hxx"
 #include "Log.hxx"
 
+#include <functional>
+
 static constexpr Domain decoder_thread_domain("decoder_thread");
 
 /**

src/DespotifyUtils.cxx

@@ -28,6 +28,8 @@ extern "C" {
 #include <despotify.h>
 }
 
+#include <stdio.h>
+
 const Domain despotify_domain("despotify");
 
 static struct despotify_session *g_session;

src/IcyMetaDataParser.cxx

@@ -81,31 +81,85 @@ icy_add_item(Tag &tag, TagType type, const char *value)
 }
 
 static void
-icy_parse_tag_item(Tag &tag, const char *item)
+icy_parse_tag_item(Tag &tag, const char *name, const char *value)
 {
-	gchar **p = g_strsplit(item, "=", 0);
+	if (strcmp(name, "StreamTitle") == 0)
+		icy_add_item(tag, TAG_TITLE, value);
+	else
+		FormatDebug(icy_metadata_domain,
+			    "unknown icy-tag: '%s'", name);
+}
 
-	if (p[0] != nullptr && p[1] != nullptr) {
-		if (strcmp(p[0], "StreamTitle") == 0)
-			icy_add_item(tag, TAG_TITLE, p[1]);
-		else
-			FormatDebug(icy_metadata_domain,
-				    "unknown icy-tag: '%s'", p[0]);
+/**
+ * Find a single quote that is followed by a semicolon (or by the end
+ * of the string).  If that fails, return the first single quote.  If
+ * that also fails, return #end.
+ */
+static char *
+find_end_quote(char *p, char *const end)
+{
+	char *fallback = std::find(p, end, '\'');
+	if (fallback >= end - 1 || fallback[1] == ';')
+		return fallback;
+
+	p = fallback + 1;
+	while (true) {
+		p = std::find(p, end, '\'');
+		if (p == end)
+			return fallback;
+
+		if (p == end - 1 || p[1] == ';')
+			return p;
+
+		++p;
 	}
-
-	g_strfreev(p);
 }
 
 static Tag *
-icy_parse_tag(const char *p)
+icy_parse_tag(char *p, char *const end)
 {
+	assert(p != nullptr);
+	assert(end != nullptr);
+	assert(p <= end);
+
 	Tag *tag = new Tag();
-	gchar **items = g_strsplit(p, ";", 0);
 
-	for (unsigned i = 0; items[i] != nullptr; ++i)
-		icy_parse_tag_item(*tag, items[i]);
+	while (p != end) {
+		const char *const name = p;
+		char *eq = std::find(p, end, '=');
+		if (eq == end)
+			break;
 
-	g_strfreev(items);
+		*eq = 0;
+		p = eq + 1;
+
+		if (*p != '\'') {
+			/* syntax error; skip to the next semicolon,
+			   try to recover */
+			char *semicolon = std::find(p, end, ';');
+			if (semicolon == end)
+				break;
+			p = semicolon + 1;
+			continue;
+		}
+
+		++p;
+
+		const char *const value = p;
+		char *quote = find_end_quote(p, end);
+		if (quote == end)
+			break;
+
+		*quote = 0;
+		p = quote + 1;
+
+		icy_parse_tag_item(*tag, name, value);
+
+		char *semicolon = std::find(p, end, ';');
+		if (semicolon == end)
+			break;
+		p = semicolon + 1;
+	}
 
 	return tag;
 }
@@ -152,15 +206,11 @@ IcyMetaDataParser::Meta(const void *data, size_t length)
 		++length;
 
 	if (meta_position == meta_size) {
-		/* null-terminate the string */
-
-		meta_data[meta_size] = 0;
-
 		/* parse */
 
 		delete tag;
 
-		tag = icy_parse_tag(meta_data);
+		tag = icy_parse_tag(meta_data, meta_data + meta_size);
 		g_free(meta_data);
 
 		/* change back to normal data mode */

src/Mapper.cxx

@@ -133,7 +133,9 @@ void mapper_finish(void)
 const char *
 mapper_get_music_directory_utf8(void)
 {
-	return music_dir_utf8.c_str();
+	return music_dir_utf8.empty()
+		? nullptr
+		: music_dir_utf8.c_str();
 }
 
 const AllocatedPath &

src/Mapper.hxx

@@ -41,7 +41,8 @@ mapper_init(AllocatedPath &&music_dir, AllocatedPath &&playlist_dir);
 void mapper_finish(void);
 
 /**
- * Return the absolute path of the music directory encoded in UTF-8.
+ * Return the absolute path of the music directory encoded in UTF-8 or
+ * nullptr if no music directory was configured.
  */
 gcc_const
 const char *

src/OutputAPI.hxx

@@ -18,7 +18,7 @@
  */
 
 #ifndef MPD_OUTPUT_API_HXX
-#define MPD_OUTPUT_API_HxX
+#define MPD_OUTPUT_API_HXX
 
 #include "OutputPlugin.hxx"
 #include "OutputInternal.hxx"

src/OutputControl.cxx

@@ -101,6 +101,8 @@ audio_output_set_replay_gain_mode(struct audio_output *ao,
 {
 	if (ao->replay_gain_filter != nullptr)
 		replay_gain_filter_set_mode(ao->replay_gain_filter, mode);
+	if (ao->other_replay_gain_filter != nullptr)
+		replay_gain_filter_set_mode(ao->other_replay_gain_filter, mode);
 }
 
 void

src/PlayerThread.cxx

@@ -36,8 +36,6 @@
 #include "util/Domain.hxx"
 #include "Log.hxx"
 
-#include <glib.h>
-
 #include <string.h>
 
 static constexpr Domain player_domain("player");
@@ -1043,8 +1041,14 @@ Player::Run()
 			   output thread is still busy, so it's
 			   okay */
 
-			/* XXX synchronize in a better way */
-			g_usleep(10000);
+			pc.Lock();
+
+			/* wake up the decoder (just in case it's
+			   waiting for space in the MusicBuffer) and
+			   wait for it */
+			dc.Signal();
+			dc.WaitForDecoder();
+			continue;
 		} else if (IsDecoderAtNextSong()) {
 			/* at the beginning of a new song */
 
@@ -1077,8 +1081,11 @@ Player::Run()
 
 	delete cross_fade_tag;
 
-	if (song != nullptr)
+	if (song != nullptr) {
+		const auto uri = song->GetURI();
+		FormatDefault(player_domain, "played \"%s\"", uri.c_str());
 		song->Free();
+	}
 
 	pc.Lock();
 

src/Playlist.cxx

@@ -103,6 +103,12 @@ playlist::UpdateQueuedSong(PlayerControl &pc, const Song *prev)
 	if (!playing)
 		return;
 
+	if (prev == nullptr && bulk_edit)
+		/* postponed until CommitBulk() to avoid always
+		   queueing the first song that is being added (in
+		   random mode) */
+		return;
+
 	assert(!queue.IsEmpty());
 	assert((queued < 0) == (prev == nullptr));
 
@@ -294,7 +300,9 @@ playlist::SetRandom(PlayerControl &pc, bool status)
 	if (queue.random) {
 		/* shuffle the queue order, but preserve current */
 
-		const int current_position = GetCurrentPosition();
+		const int current_position = playing
+			? GetCurrentPosition()
+			: -1;
 
 		queue.ShuffleOrder();
 

src/Playlist.hxx

@@ -30,7 +30,7 @@ struct playlist {
 	/**
 	 * The song queue - it contains the "real" playlist.
 	 */
-	struct queue queue;
+	struct Queue queue;
 
 	/**
 	 * This value is true if the player is currently playing (or
@@ -45,6 +45,18 @@ struct playlist {
 	 */
 	bool stop_on_error;
 
+	/**
+	 * If true, then a bulk edit has been initiated by
+	 * BeginBulk(), and UpdateQueuedSong() and OnModified() will
+	 * be postponed until CommitBulk()
+	 */
+	bool bulk_edit;
+
+	/**
+	 * Has the queue been modified during bulk edit mode?
+	 */
+	bool bulk_modified;
+
 	/**
 	 * Number of errors since playback was started.  If this
 	 * number exceeds the length of the playlist, MPD gives up,
@@ -69,7 +81,9 @@ struct playlist {
 	int queued;
 
 	playlist(unsigned max_length)
-		:queue(max_length), playing(false), current(-1), queued(-1) {
+		:queue(max_length), playing(false),
+		 bulk_edit(false),
+		 current(-1), queued(-1) {
 	}
 
 	~playlist() {
@@ -126,6 +140,9 @@ protected:
 	void UpdateQueuedSong(PlayerControl &pc, const Song *prev);
 
 public:
+	void BeginBulk();
+	void CommitBulk(PlayerControl &pc);
+
 	void Clear(PlayerControl &pc);
 
 	/**

src/PlaylistControl.cxx

@@ -153,7 +153,7 @@ playlist::PlayNext(PlayerControl &pc)
 			queue.ShuffleOrder();
 
 			/* note that current and queued are
-			   now invalid, but playlist_play_order() will
+			   now invalid, but PlayOrder() will
 			   discard them anyway */
 		}
 

src/PlaylistEdit.cxx

@@ -40,6 +40,12 @@
 void
 playlist::OnModified()
 {
+	if (bulk_edit) {
+		/* postponed to CommitBulk() */
+		bulk_modified = true;
+		return;
+	}
+
 	queue.IncrementVersion();
 
 	idle_add(IDLE_PLAYLIST);
@@ -56,6 +62,35 @@ playlist::Clear(PlayerControl &pc)
 	OnModified();
 }
 
+void
+playlist::BeginBulk()
+{
+	assert(!bulk_edit);
+
+	bulk_edit = true;
+	bulk_modified = false;
+}
+
+void
+playlist::CommitBulk(PlayerControl &pc)
+{
+	assert(bulk_edit);
+
+	bulk_edit = false;
+	if (!bulk_modified)
+		return;
+
+	if (queued < 0)
+		/* if no song was queued, UpdateQueuedSong() is being
+		   ignored in "bulk" edit mode; now that we have
+		   shuffled all new songs, we can pick a random one
+		   (instead of always picking the first one that was
+		   added) */
+		UpdateQueuedSong(pc, nullptr);
+
+	OnModified();
+}
+
 PlaylistResult
 playlist::AppendFile(PlayerControl &pc,
 		     const char *path_utf8, unsigned *added_id)
@@ -234,12 +269,8 @@ playlist::DeleteInternal(PlayerControl &pc,
 	if (playing && current == (int)songOrder) {
 		const bool paused = pc.GetState() == PlayerState::PAUSE;
 
-		/* the current song is going to be deleted: stop the player */
-
-		pc.Stop();
-		playing = false;
-
-		/* see which song is going to be played instead */
+		/* the current song is going to be deleted: see which
+		   song is going to be played instead */
 
 		current = queue.GetNextOrder(current);
 		if (current == (int)songOrder)
@@ -248,10 +279,12 @@ playlist::DeleteInternal(PlayerControl &pc,
 		if (current >= 0 && !paused)
 			/* play the song after the deleted one */
 			PlayOrder(pc, current);
-		else
-			/* no songs left to play, stop playback
-			   completely */
-			Stop(pc);
+		else {
+			/* stop the player */
+
+			pc.Stop();
+			playing = false;
+		}
 
 		*queued_p = nullptr;
 	} else if (current == (int)songOrder)

src/PlaylistPrint.cxx

@@ -40,7 +40,7 @@
 void
 playlist_print_uris(Client &client, const playlist &playlist)
 {
-	const queue &queue = playlist.queue;
+	const Queue &queue = playlist.queue;
 
 	queue_print_uris(client, queue, 0, queue.GetLength());
 }
@@ -49,7 +49,7 @@ bool
 playlist_print_info(Client &client, const playlist &playlist,
 		    unsigned start, unsigned end)
 {
-	const queue &queue = playlist.queue;
+	const Queue &queue = playlist.queue;
 
 	if (end > queue.GetLength())
 		/* correct the "end" offset */

src/PlaylistSave.cxx

@@ -65,7 +65,7 @@ playlist_print_uri(FILE *file, const char *uri)
 }
 
 PlaylistResult
-spl_save_queue(const char *name_utf8, const queue &queue)
+spl_save_queue(const char *name_utf8, const Queue &queue)
 {
 	if (map_spl_path().IsNull())
 		return PlaylistResult::DISABLED;

src/PlaylistSave.hxx

@@ -25,7 +25,7 @@
 #include <stdio.h>
 
 struct Song;
-struct queue;
+struct Queue;
 struct playlist;
 struct PlayerControl;
 class Error;
@@ -40,7 +40,7 @@ playlist_print_uri(FILE *fp, const char *uri);
  * Saves a queue object into a stored playlist file.
  */
 PlaylistResult
-spl_save_queue(const char *name_utf8, const queue &queue);
+spl_save_queue(const char *name_utf8, const Queue &queue);
 
 /**
  * Saves a playlist object into a stored playlist file.

src/Queue.cxx

@@ -23,7 +23,7 @@
 
 #include <stdlib.h>
 
-queue::queue(unsigned _max_length)
+Queue::Queue(unsigned _max_length)
 	:max_length(_max_length), length(0),
 	 version(1),
 	 items(new Item[max_length]),
@@ -36,7 +36,7 @@ queue::queue(unsigned _max_length)
 {
 }
 
-queue::~queue()
+Queue::~Queue()
 {
 	Clear();
 
@@ -45,7 +45,7 @@ queue::~queue()
 }
 
 int
-queue::GetNextOrder(unsigned _order) const
+Queue::GetNextOrder(unsigned _order) const
 {
 	assert(_order < length);
 
@@ -62,7 +62,7 @@ queue::GetNextOrder(unsigned _order) const
 }
 
 void
-queue::IncrementVersion()
+Queue::IncrementVersion()
 {
 	static unsigned long max = ((uint32_t) 1 << 31) - 1;
 
@@ -77,7 +77,7 @@ queue::IncrementVersion()
 }
 
 void
-queue::ModifyAtOrder(unsigned _order)
+Queue::ModifyAtOrder(unsigned _order)
 {
 	assert(_order < length);
 
@@ -86,7 +86,7 @@ queue::ModifyAtOrder(unsigned _order)
 }
 
 unsigned
-queue::Append(Song *song, uint8_t priority)
+Queue::Append(Song *song, uint8_t priority)
 {
 	assert(!IsFull());
 
@@ -105,7 +105,7 @@ queue::Append(Song *song, uint8_t priority)
 }
 
 void
-queue::SwapPositions(unsigned position1, unsigned position2)
+Queue::SwapPositions(unsigned position1, unsigned position2)
 {
 	unsigned id1 = items[position1].id;
 	unsigned id2 = items[position2].id;
@@ -120,7 +120,7 @@ queue::SwapPositions(unsigned position1, unsigned position2)
 }
 
 void
-queue::MovePostion(unsigned from, unsigned to)
+Queue::MovePostion(unsigned from, unsigned to)
 {
 	const Item tmp = items[from];
 
@@ -156,7 +156,7 @@ queue::MovePostion(unsigned from, unsigned to)
 }
 
 void
-queue::MoveRange(unsigned start, unsigned end, unsigned to)
+Queue::MoveRange(unsigned start, unsigned end, unsigned to)
 {
 	Item tmp[end - start];
 	// Copy the original block [start,end-1]
@@ -198,7 +198,7 @@ queue::MoveRange(unsigned start, unsigned end, unsigned to)
 }
 
 void
-queue::MoveOrder(unsigned from_order, unsigned to_order)
+Queue::MoveOrder(unsigned from_order, unsigned to_order)
 {
 	assert(from_order < length);
 	assert(to_order <= length);
@@ -217,7 +217,7 @@ queue::MoveOrder(unsigned from_order, unsigned to_order)
 }
 
 void
-queue::DeletePosition(unsigned position)
+Queue::DeletePosition(unsigned position)
 {
 	assert(position < length);
 
@@ -254,7 +254,7 @@ queue::DeletePosition(unsigned position)
 }
 
 void
-queue::Clear()
+Queue::Clear()
 {
 	for (unsigned i = 0; i < length; i++) {
 		Item *item = &items[i];
@@ -270,7 +270,7 @@ queue::Clear()
 }
 
 static void
-queue_sort_order_by_priority(struct queue *queue, unsigned start, unsigned end)
+queue_sort_order_by_priority(Queue *queue, unsigned start, unsigned end)
 {
 	assert(queue != nullptr);
 	assert(queue->random);
@@ -278,8 +278,8 @@ queue_sort_order_by_priority(struct queue *queue, unsigned start, unsigned end)
 	assert(end <= queue->length);
 
 	auto cmp = [queue](unsigned a_pos, unsigned b_pos){
-		const queue::Item &a = queue->items[a_pos];
-		const queue::Item &b = queue->items[b_pos];
+		const Queue::Item &a = queue->items[a_pos];
+		const Queue::Item &b = queue->items[b_pos];
 
 		return a.priority > b.priority;
 	};
@@ -288,7 +288,7 @@ queue_sort_order_by_priority(struct queue *queue, unsigned start, unsigned end)
 }
 
 void
-queue::ShuffleOrderRange(unsigned start, unsigned end)
+Queue::ShuffleOrderRange(unsigned start, unsigned end)
 {
 	assert(random);
 	assert(start <= end);
@@ -303,7 +303,7 @@ queue::ShuffleOrderRange(unsigned start, unsigned end)
  * priority group.
  */
 void
-queue::ShuffleOrderRangeWithPriority(unsigned start, unsigned end)
+Queue::ShuffleOrderRangeWithPriority(unsigned start, unsigned end)
 {
 	assert(random);
 	assert(start <= end);
@@ -337,13 +337,13 @@ queue::ShuffleOrderRangeWithPriority(unsigned start, unsigned end)
 }
 
 void
-queue::ShuffleOrder()
+Queue::ShuffleOrder()
 {
 	ShuffleOrderRangeWithPriority(0, length);
 }
 
 void
-queue::ShuffleOrderFirst(unsigned start, unsigned end)
+Queue::ShuffleOrderFirst(unsigned start, unsigned end)
 {
 	rand.AutoCreate();
 
@@ -352,7 +352,7 @@ queue::ShuffleOrderFirst(unsigned start, unsigned end)
 }
 
 void
-queue::ShuffleOrderLast(unsigned start, unsigned end)
+Queue::ShuffleOrderLast(unsigned start, unsigned end)
 {
 	rand.AutoCreate();
 
@@ -361,7 +361,7 @@ queue::ShuffleOrderLast(unsigned start, unsigned end)
 }
 
 void
-queue::ShuffleRange(unsigned start, unsigned end)
+Queue::ShuffleRange(unsigned start, unsigned end)
 {
 	assert(start <= end);
 	assert(end <= length);
@@ -377,7 +377,7 @@ queue::ShuffleRange(unsigned start, unsigned end)
 }
 
 unsigned
-queue::FindPriorityOrder(unsigned start_order, uint8_t priority,
+Queue::FindPriorityOrder(unsigned start_order, uint8_t priority,
 			 unsigned exclude_order) const
 {
 	assert(random);
@@ -394,7 +394,7 @@ queue::FindPriorityOrder(unsigned start_order, uint8_t priority,
 }
 
 unsigned
-queue::CountSamePriority(unsigned start_order, uint8_t priority) const
+Queue::CountSamePriority(unsigned start_order, uint8_t priority) const
 {
 	assert(random);
 	assert(start_order <= length);
@@ -410,7 +410,7 @@ queue::CountSamePriority(unsigned start_order, uint8_t priority) const
 }
 
 bool
-queue::SetPriority(unsigned position, uint8_t priority, int after_order)
+Queue::SetPriority(unsigned position, uint8_t priority, int after_order)
 {
 	assert(position < length);
 
@@ -468,7 +468,7 @@ queue::SetPriority(unsigned position, uint8_t priority, int after_order)
 }
 
 bool
-queue::SetPriorityRange(unsigned start_position, unsigned end_position,
+Queue::SetPriorityRange(unsigned start_position, unsigned end_position,
 			uint8_t priority, int after_order)
 {
 	assert(start_position <= end_position);

src/Queue.hxx

@@ -41,7 +41,7 @@ struct Song;
  * - the unique id (which stays the same, regardless of moves)
  * - the order number (which only differs from "position" in random mode)
  */
-struct queue {
+struct Queue {
 	/**
 	 * reserve max_length * HASH_MULT elements in the id
 	 * number space
@@ -103,16 +103,16 @@ struct queue {
 	/** random number generator for shuffle and random mode */
 	LazyRandomEngine rand;
 
-	queue(unsigned max_length);
+	explicit Queue(unsigned max_length);
 
 	/**
 	 * Deinitializes a queue object.  It does not free the queue
 	 * pointer itself.
 	 */
-	~queue();
+	~Queue();
 
-	queue(const queue &other) = delete;
-	queue &operator=(const queue &other) = delete;
+	Queue(const Queue &) = delete;
+	Queue &operator=(const Queue &) = delete;
 
 	unsigned GetLength() const {
 		assert(length <= max_length);

src/QueuePrint.cxx

@@ -38,7 +38,7 @@ extern "C" {
  * @param end the index of the last song (excluding)
  */
 static void
-queue_print_song_info(Client &client, const queue &queue,
+queue_print_song_info(Client &client, const Queue &queue,
 		      unsigned position)
 {
 	song_print_info(client, queue.Get(position));
@@ -51,7 +51,7 @@ queue_print_song_info(Client &client, const queue &queue,
 }
 
 void
-queue_print_info(Client &client, const queue &queue,
+queue_print_info(Client &client, const Queue &queue,
 		 unsigned start, unsigned end)
 {
 	assert(start <= end);
@@ -62,7 +62,7 @@ queue_print_info(Client &client, const queue &queue,
 }
 
 void
-queue_print_uris(Client &client, const queue &queue,
+queue_print_uris(Client &client, const Queue &queue,
 		 unsigned start, unsigned end)
 {
 	assert(start <= end);
@@ -75,7 +75,7 @@ queue_print_uris(Client &client, const queue &queue,
 }
 
 void
-queue_print_changes_info(Client &client, const queue &queue,
+queue_print_changes_info(Client &client, const Queue &queue,
 			 uint32_t version)
 {
 	for (unsigned i = 0; i < queue.GetLength(); i++) {
@@ -85,7 +85,7 @@ queue_print_changes_info(Client &client, const queue &queue,
 }
 
 void
-queue_print_changes_position(Client &client, const queue &queue,
+queue_print_changes_position(Client &client, const Queue &queue,
 			     uint32_t version)
 {
 	for (unsigned i = 0; i < queue.GetLength(); i++)
@@ -95,7 +95,7 @@ queue_print_changes_position(Client &client, const queue &queue,
 }
 
 void
-queue_find(Client &client, const queue &queue,
+queue_find(Client &client, const Queue &queue,
 	   const SongFilter &filter)
 {
 	for (unsigned i = 0; i < queue.GetLength(); i++) {

src/QueuePrint.hxx

@@ -27,28 +27,28 @@
 
 #include <stdint.h>
 
-struct queue;
+struct Queue;
 class SongFilter;
 class Client;
 
 void
-queue_print_info(Client &client, const queue &queue,
+queue_print_info(Client &client, const Queue &queue,
 		 unsigned start, unsigned end);
 
 void
-queue_print_uris(Client &client, const queue &queue,
+queue_print_uris(Client &client, const Queue &queue,
 		 unsigned start, unsigned end);
 
 void
-queue_print_changes_info(Client &client, const queue &queue,
+queue_print_changes_info(Client &client, const Queue &queue,
 			 uint32_t version);
 
 void
-queue_print_changes_position(Client &client, const queue &queue,
+queue_print_changes_position(Client &client, const Queue &queue,
 			     uint32_t version);
 
 void
-queue_find(Client &client, const queue &queue,
+queue_find(Client &client, const Queue &queue,
 	   const SongFilter &filter);
 
 #endif

src/QueueSave.cxx

@@ -60,7 +60,7 @@ queue_save_song(FILE *fp, int idx, const Song &song)
 }
 
 void
-queue_save(FILE *fp, const queue &queue)
+queue_save(FILE *fp, const Queue &queue)
 {
 	for (unsigned i = 0; i < queue.GetLength(); i++) {
 		uint8_t prio = queue.GetPriorityAtPosition(i);
@@ -72,7 +72,7 @@ queue_save(FILE *fp, const queue &queue)
 }
 
 void
-queue_load_song(TextFile &file, const char *line, queue &queue)
+queue_load_song(TextFile &file, const char *line, Queue &queue)
 {
 	if (queue.IsFull())
 		return;

src/QueueSave.hxx

@@ -27,16 +27,16 @@
 
 #include <stdio.h>
 
-struct queue;
+struct Queue;
 class TextFile;
 
 void
-queue_save(FILE *fp, const queue &queue);
+queue_save(FILE *fp, const Queue &queue);
 
 /**
  * Loads one song from the state file and appends it to the queue.
  */
 void
-queue_load_song(TextFile &file, const char *line, queue &queue);
+queue_load_song(TextFile &file, const char *line, Queue &queue);
 
 #endif

src/SongFilter.cxx

@@ -101,7 +101,7 @@ bool
 SongFilter::Item::Match(const Tag &_tag) const
 {
 	bool visited_types[TAG_NUM_OF_ITEM_TYPES];
-	std::fill_n(visited_types, TAG_NUM_OF_ITEM_TYPES, false);
+	std::fill_n(visited_types, size_t(TAG_NUM_OF_ITEM_TYPES), false);
 
 	for (unsigned i = 0; i < _tag.num_items; i++) {
 		visited_types[_tag.items[i]->type] = true;
@@ -203,6 +203,16 @@ SongFilter::Match(const Song &song) const
 	return true;
 }
 
+bool
+SongFilter::HasOtherThanBase() const
+{
+	for (const auto &i : items)
+		if (i.GetTag() != LOCATE_TAG_BASE_TYPE)
+			return true;
+
+	return false;
+}
+
 std::string
 SongFilter::GetBase() const
 {

src/SongFilter.hxx

@@ -109,6 +109,11 @@ public:
 		return items;
 	}
 
+	gcc_pure
+	bool IsEmpty() const {
+		return items.empty();
+	}
+
 	/**
 	 * Is there at least one item with "fold case" enabled?
 	 */
@@ -121,6 +126,12 @@ public:
 		return false;
 	}
 
+	/**
+	 * Does this filter contain constraints other than "base"?
+	 */
+	gcc_pure
+	bool HasOtherThanBase() const;
+
 	/**
 	 * Returns the "base" specification (if there is one) or an
 	 * empty string.

src/SongSave.cxx

@@ -30,6 +30,7 @@
 #include "util/Domain.hxx"
 
 #include <string.h>
+#include <stdlib.h>
 
 #define SONG_MTIME "mtime"
 #define SONG_END "song_end"

src/SongUpdate.cxx

@@ -95,8 +95,7 @@ Song::UpdateFile()
 
 	TagBuilder tag_builder;
 	if (!tag_file_scan(path_fs,
-			   &full_tag_handler, &tag_builder) ||
-	    !tag_builder.IsDefined())
+			   &full_tag_handler, &tag_builder))
 		return false;
 
 	if (tag_builder.IsEmpty())

src/Stats.cxx

@@ -30,59 +30,75 @@
 
 #include <glib.h>
 
-struct stats stats;
+static GTimer *uptime;
+static DatabaseStats stats;
 
 void stats_global_init(void)
 {
-	stats.timer = g_timer_new();
+	uptime = g_timer_new();
 }
 
 void stats_global_finish(void)
 {
-	g_timer_destroy(stats.timer);
+	g_timer_destroy(uptime);
 }
 
 void stats_update(void)
 {
+	assert(GetDatabase() != nullptr);
+
 	Error error;
 
 	DatabaseStats stats2;
 
 	const DatabaseSelection selection("", true);
 	if (GetDatabase()->GetStats(selection, stats2, error)) {
-		stats.song_count = stats2.song_count;
-		stats.song_duration = stats2.total_duration;
-		stats.artist_count = stats2.artist_count;
-		stats.album_count = stats2.album_count;
+		stats = stats2;
 	} else {
 		LogError(error);
 
-		stats.song_count = 0;
-		stats.song_duration = 0;
-		stats.artist_count = 0;
-		stats.album_count = 0;
+		stats.Clear();
 	}
 }
 
+static void
+db_stats_print(Client &client)
+{
+	assert(GetDatabase() != nullptr);
+
+	if (!db_is_simple())
+		/* reload statistics if we're using the "proxy"
+		   database plugin */
+		/* TODO: move this into the "proxy" database plugin as
+		   an "idle" handler */
+		stats_update();
+
+	client_printf(client,
+		      "artists: %u\n"
+		      "albums: %u\n"
+		      "songs: %u\n"
+		      "db_playtime: %lu\n",
+		      stats.artist_count,
+		      stats.album_count,
+		      stats.song_count,
+		      stats.total_duration);
+
+	const time_t update_stamp = GetDatabase()->GetUpdateStamp();
+	if (update_stamp > 0)
+		client_printf(client,
+			      "db_update: %lu\n",
+			      (unsigned long)update_stamp);
+}
+
 void
 stats_print(Client &client)
 {
 	client_printf(client,
-		      "artists: %u\n"
-		      "albums: %u\n"
-		      "songs: %i\n"
-		      "uptime: %li\n"
-		      "playtime: %li\n"
-		      "db_playtime: %li\n",
-		      stats.artist_count,
-		      stats.album_count,
-		      stats.song_count,
-		      (long)g_timer_elapsed(stats.timer, NULL),
-		      (long)(client.player_control.GetTotalPlayTime() + 0.5),
-		      stats.song_duration);
+		      "uptime: %lu\n"
+		      "playtime: %lu\n",
+		      (unsigned long)g_timer_elapsed(uptime, NULL),
+		      (unsigned long)(client.player_control.GetTotalPlayTime() + 0.5));
 
-	if (db_is_simple())
-		client_printf(client,
-			      "db_update: %li\n",
-			      (long)db_get_mtime());
+	if (GetDatabase() != nullptr)
+		db_stats_print(client);
 }

src/Stats.hxx

@@ -21,26 +21,6 @@
 #define MPD_STATS_HXX
 
 class Client;
-typedef struct _GTimer GTimer;
-
-struct stats {
-	GTimer *timer;
-
-	/** number of song files in the music directory */
-	unsigned song_count;
-
-	/** sum of all song durations in the music directory (in
-	    seconds) */
-	unsigned long song_duration;
-
-	/** number of distinct artist names in the music directory */
-	unsigned artist_count;
-
-	/** number of distinct album names in the music directory */
-	unsigned album_count;
-};
-
-extern struct stats stats;
 
 void stats_global_init(void);
 

src/TagFile.hxx

@@ -28,6 +28,9 @@ struct tag_handler;
 /**
  * Scan the tags of a song file.  Invokes matching decoder plugins,
  * but does not invoke the special "APE" and "ID3" scanners.
+ *
+ * @return true if the file was recognized (even if no metadata was
+ * found)
  */
 bool
 tag_file_scan(Path path,

src/archive/Iso9660ArchivePlugin.cxx

@@ -190,18 +190,16 @@ iso9660_input_read(InputStream *is, void *ptr, size_t size,
 		   Error &error)
 {
 	Iso9660InputStream *iis = (Iso9660InputStream *)is;
-	int toread, readed = 0;
+	int readed = 0;
 	int no_blocks, cur_block;
 	size_t left_bytes = iis->statbuf->size - is->offset;
 
 	size = (size * ISO_BLOCKSIZE) / ISO_BLOCKSIZE;
 
 	if (left_bytes < size) {
-		toread = left_bytes;
 		no_blocks = CEILING(left_bytes,ISO_BLOCKSIZE);
 	} else {
-		toread = size;
-		no_blocks = toread / ISO_BLOCKSIZE;
+		no_blocks = size / ISO_BLOCKSIZE;
 	}
 	if (no_blocks > 0) {
 

src/command/AllCommands.cxx

@@ -90,7 +90,7 @@ static const struct command commands[] = {
 	{ "disableoutput", PERMISSION_ADMIN, 1, 1, handle_disableoutput },
 	{ "enableoutput", PERMISSION_ADMIN, 1, 1, handle_enableoutput },
 	{ "find", PERMISSION_READ, 2, -1, handle_find },
-	{ "findadd", PERMISSION_READ, 2, -1, handle_findadd},
+	{ "findadd", PERMISSION_ADD, 2, -1, handle_findadd},
 	{ "idle", PERMISSION_READ, 0, -1, handle_idle },
 	{ "kill", PERMISSION_ADMIN, -1, -1, handle_kill },
 	{ "list", PERMISSION_READ, 1, -1, handle_list },

src/command/DatabaseCommands.cxx

@@ -30,6 +30,7 @@
 #include "util/Error.hxx"
 #include "SongFilter.hxx"
 #include "protocol/Result.hxx"
+#include "BulkEdit.hxx"
 
 #include <assert.h>
 #include <string.h>
@@ -92,6 +93,8 @@ handle_match_add(Client &client, int argc, char *argv[], bool fold_case)
 		return CommandResult::ERROR;
 	}
 
+	const ScopeBulkEdit bulk_edit(client.partition);
+
 	const DatabaseSelection selection("", true, &filter);
 	Error error;
 	return AddFromDatabase(client.partition, selection, error)

src/command/PlaylistCommands.cxx

@@ -26,6 +26,7 @@
 #include "PlaylistFile.hxx"
 #include "PlaylistVector.hxx"
 #include "PlaylistQueue.hxx"
+#include "BulkEdit.hxx"
 #include "TimePrint.hxx"
 #include "Client.hxx"
 #include "protocol/ArgParser.hxx"
@@ -67,6 +68,8 @@ handle_load(Client &client, int argc, char *argv[])
 	} else if (!check_range(client, &start_index, &end_index, argv[2]))
 		return CommandResult::ERROR;
 
+	const ScopeBulkEdit bulk_edit(client.partition);
+
 	const PlaylistResult result =
 		playlist_open_into_queue(argv[1],
 					 start_index, end_index,

src/command/QueueCommands.cxx

@@ -28,6 +28,7 @@
 #include "ClientFile.hxx"
 #include "Client.hxx"
 #include "Partition.hxx"
+#include "BulkEdit.hxx"
 #include "protocol/ArgParser.hxx"
 #include "protocol/Result.hxx"
 #include "ls.hxx"
@@ -43,7 +44,6 @@ CommandResult
 handle_add(Client &client, gcc_unused int argc, char *argv[])
 {
 	char *uri = argv[1];
-	PlaylistResult result;
 
 	if (memcmp(uri, "file:///", 8) == 0) {
 		const char *path_utf8 = uri + 7;
@@ -59,7 +59,7 @@ handle_add(Client &client, gcc_unused int argc, char *argv[])
 		if (!client_allow_file(client, path_fs, error))
 			return print_error(client, error);
 
-		result = client.partition.AppendFile(path_utf8);
+		auto result = client.partition.AppendFile(path_utf8);
 		return print_playlist_result(client, result);
 	}
 
@@ -70,10 +70,12 @@ handle_add(Client &client, gcc_unused int argc, char *argv[])
 			return CommandResult::ERROR;
 		}
 
-		result = client.partition.AppendURI(uri);
+		auto result = client.partition.AppendURI(uri);
 		return print_playlist_result(client, result);
 	}
 
+	const ScopeBulkEdit bulk_edit(client.partition);
+
 	const DatabaseSelection selection(uri, true);
 	Error error;
 	return AddFromDatabase(client.partition, selection, error)

src/db/ProxyDatabasePlugin.cxx

@@ -46,6 +46,9 @@ class ProxyDatabase : public Database {
 	struct mpd_connection *connection;
 	Directory *root;
 
+	/* this is mutable because GetStats() must be "const" */
+	mutable time_t update_stamp;
+
 public:
 	static Database *Create(const config_param &param,
 				Error &error);
@@ -71,6 +74,10 @@ public:
 			      DatabaseStats &stats,
 			      Error &error) const override;
 
+	virtual time_t GetUpdateStamp() const override {
+		return update_stamp;
+	}
+
 private:
 	bool Configure(const config_param &param, Error &error);
 
@@ -237,6 +244,7 @@ ProxyDatabase::Open(Error &error)
 		return false;
 
 	root = Directory::NewRoot();
+	update_stamp = 0;
 
 	return true;
 }
@@ -390,8 +398,13 @@ Convert(const struct mpd_song *song)
 	Song *s = Song::NewDetached(mpd_song_get_uri(song));
 
 	s->mtime = mpd_song_get_last_modified(song);
+
+#if LIBMPDCLIENT_CHECK_VERSION(2,3,0)
 	s->start_ms = mpd_song_get_start(song) * 1000;
 	s->end_ms = mpd_song_get_end(song) * 1000;
+#else
+	s->start_ms = s->end_ms = 0;
+#endif
 
 	TagBuilder tag;
 	tag.SetTime(mpd_song_get_duration(song));
@@ -553,6 +566,23 @@ SearchSongs(struct mpd_connection *connection,
 	return result && CheckError(connection, error);
 }
 
+/**
+ * Check whether we can use the "base" constraint.  Requires
+ * libmpdclient 2.9 and MPD 0.18.
+ */
+gcc_pure
+static bool
+ServerSupportsSearchBase(const struct mpd_connection *connection)
+{
+#if LIBMPDCLIENT_CHECK_VERSION(2,9,0)
+	return mpd_connection_cmp_server_version(connection, 0, 18, 0) >= 0;
+#else
+	(void)connection;
+
+	return false;
+#endif
+}
+
 bool
 ProxyDatabase::Visit(const DatabaseSelection &selection,
 		     VisitDirectory visit_directory,
@@ -564,7 +594,10 @@ ProxyDatabase::Visit(const DatabaseSelection &selection,
 	if (!const_cast<ProxyDatabase *>(this)->EnsureConnected(error))
 		return nullptr;
 
-	if (!visit_directory && !visit_playlist && selection.recursive)
+	if (!visit_directory && !visit_playlist && selection.recursive &&
+	    (ServerSupportsSearchBase(connection)
+	     ? !selection.IsEmpty()
+	     : selection.HasOtherThanBase()))
 		/* this optimized code path can only be used under
 		   certain conditions */
 		return ::SearchSongs(connection, selection, visit_song, error);
@@ -631,6 +664,8 @@ ProxyDatabase::GetStats(const DatabaseSelection &selection,
 	if (stats2 == nullptr)
 		return CheckError(connection, error);
 
+	update_stamp = (time_t)mpd_stats_get_db_update_time(stats2);
+
 	stats.song_count = mpd_stats_get_number_of_songs(stats2);
 	stats.total_duration = mpd_stats_get_db_play_time(stats2);
 	stats.artist_count = mpd_stats_get_number_of_artists(stats2);

src/db/SimpleDatabasePlugin.hxx

@@ -26,8 +26,6 @@
 
 #include <cassert>
 
-#include <time.h>
-
 struct Directory;
 
 class SimpleDatabase : public Database {
@@ -55,11 +53,6 @@ public:
 
 	bool Save(Error &error);
 
-	gcc_pure
-	time_t GetLastModified() const {
-		return mtime;
-	}
-
 	static Database *Create(const config_param &param,
 				Error &error);
 
@@ -85,6 +78,10 @@ public:
 			      DatabaseStats &stats,
 			      Error &error) const override;
 
+	virtual time_t GetUpdateStamp() const override {
+		return mtime;
+	}
+
 protected:
 	bool Configure(const config_param &param, Error &error);
 

src/decoder/AudiofileDecoderPlugin.cxx

@@ -31,12 +31,27 @@
 #include <af_vfs.h>
 
 #include <assert.h>
+#include <stdio.h>
 
 /* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
 #define CHUNK_SIZE		1020
 
 static constexpr Domain audiofile_domain("audiofile");
 
+struct AudioFileInputStream {
+	Decoder *const decoder;
+	InputStream &is;
+
+	size_t Read(void *buffer, size_t size) {
+		/* libaudiofile does not like partial reads at all,
+		   and will abort playback; therefore always force full
+		   reads */
+		return decoder_read_full(decoder, is, buffer, size)
+			? size
+			: 0;
+	}
+};
+
 static int audiofile_get_duration(const char *file)
 {
 	int total_time;
@@ -54,29 +69,26 @@ static int audiofile_get_duration(const char *file)
 static ssize_t
 audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
 
-	Error error;
-	size_t nbytes = is.LockRead(data, length, error);
-	if (nbytes == 0 && error.IsDefined()) {
-		LogError(error);
-		return -1;
-	}
-
-	return nbytes;
+	return afis.Read(data, length);
 }
 
 static AFfileoffset
 audiofile_file_length(AFvirtualfile *vfile)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
+	InputStream &is = afis.is;
+
 	return is.GetSize();
 }
 
 static AFfileoffset
 audiofile_file_tell(AFvirtualfile *vfile)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
+	InputStream &is = afis.is;
+
 	return is.GetOffset();
 }
 
@@ -91,11 +103,14 @@ audiofile_file_destroy(AFvirtualfile *vfile)
 static AFfileoffset
 audiofile_file_seek(AFvirtualfile *vfile, AFfileoffset offset, int is_relative)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
+	InputStream &is = afis.is;
+
 	int whence = (is_relative ? SEEK_CUR : SEEK_SET);
 
 	Error error;
 	if (is.LockSeek(offset, whence, error)) {
+		LogError(error, "Seek failed");
 		return is.GetOffset();
 	} else {
 		return -1;
@@ -103,10 +118,10 @@ audiofile_file_seek(AFvirtualfile *vfile, AFfileoffset offset, int is_relative)
 }
 
 static AFvirtualfile *
-setup_virtual_fops(InputStream &stream)
+setup_virtual_fops(AudioFileInputStream &afis)
 {
 	AFvirtualfile *vf = new AFvirtualfile();
-	vf->closure = &stream;
+	vf->closure = &afis;
 	vf->write = nullptr;
 	vf->read    = audiofile_file_read;
 	vf->length  = audiofile_file_length;
@@ -173,7 +188,8 @@ audiofile_stream_decode(Decoder &decoder, InputStream &is)
 		return;
 	}
 
-	vf = setup_virtual_fops(is);
+	AudioFileInputStream afis{&decoder, is};
+	vf = setup_virtual_fops(afis);
 
 	af_fp = afOpenVirtualFile(vf, "r", nullptr);
 	if (af_fp == AF_NULL_FILEHANDLE) {

src/decoder/DsdLib.cxx

@@ -49,14 +49,6 @@ DsdId::Equals(const char *s) const
 	return memcmp(value, s, sizeof(value)) == 0;
 }
 
-bool
-dsdlib_read(Decoder *decoder, InputStream &is,
-	    void *data, size_t length)
-{
-	size_t nbytes = decoder_read(decoder, is, data, length);
-	return nbytes == length;
-}
-
 /**
  * Skip the #input_stream to the specified offset.
  */
@@ -149,7 +141,7 @@ dsdlib_tag_id3(InputStream &is,
 	id3_byte_t *dsdid3data;
 	dsdid3data = dsdid3;
 
-	if (!dsdlib_read(nullptr, is, dsdid3data, count))
+	if (!decoder_read_full(nullptr, is, dsdid3data, count))
 		return;
 
 	id3_tag = id3_tag_parse(dsdid3data, count);

src/decoder/DsdLib.hxx

@@ -47,9 +47,16 @@ public:
 	}
 };
 
-bool
-dsdlib_read(Decoder *decoder, InputStream &is,
-	    void *data, size_t length);
+class DffDsdUint64 {
+	uint32_t hi;
+	uint32_t lo;
+
+public:
+	constexpr uint64_t Read() const {
+		return (uint64_t(FromBE32(hi)) << 32) |
+			uint64_t(FromBE32(lo));
+	}
+};
 
 bool
 dsdlib_skip_to(Decoder *decoder, InputStream &is,

src/decoder/DsdiffDecoderPlugin.cxx

@@ -43,13 +43,13 @@
 
 struct DsdiffHeader {
 	DsdId id;
-	DsdUint64 size;
+	DffDsdUint64 size;
 	DsdId format;
 };
 
 struct DsdiffChunkHeader {
 	DsdId id;
-	DsdUint64 size;
+	DffDsdUint64 size;
 
 	/**
 	 * Read the "size" attribute from the specified header, converting it
@@ -93,14 +93,14 @@ static bool
 dsdiff_read_id(Decoder *decoder, InputStream &is,
 	       DsdId *id)
 {
-	return dsdlib_read(decoder, is, id, sizeof(*id));
+	return decoder_read_full(decoder, is, id, sizeof(*id));
 }
 
 static bool
 dsdiff_read_chunk_header(Decoder *decoder, InputStream &is,
 			 DsdiffChunkHeader *header)
 {
-	return dsdlib_read(decoder, is, header, sizeof(*header));
+	return decoder_read_full(decoder, is, header, sizeof(*header));
 }
 
 static bool
@@ -112,8 +112,7 @@ dsdiff_read_payload(Decoder *decoder, InputStream &is,
 	if (size != (uint64_t)length)
 		return false;
 
-	size_t nbytes = decoder_read(decoder, is, data, length);
-	return nbytes == length;
+	return decoder_read_full(decoder, is, data, length);
 }
 
 /**
@@ -145,8 +144,8 @@ dsdiff_read_prop_snd(Decoder *decoder, InputStream &is,
 		} else if (header.id.Equals("CHNL")) {
 			uint16_t channels;
 			if (header.GetSize() < sizeof(channels) ||
-			    !dsdlib_read(decoder, is,
-					 &channels, sizeof(channels)) ||
+			    !decoder_read_full(decoder, is,
+					       &channels, sizeof(channels)) ||
 			    !dsdlib_skip_to(decoder, is, chunk_end_offset))
 				return false;
 
@@ -154,8 +153,8 @@ dsdiff_read_prop_snd(Decoder *decoder, InputStream &is,
 		} else if (header.id.Equals("CMPR")) {
 			DsdId type;
 			if (header.GetSize() < sizeof(type) ||
-			    !dsdlib_read(decoder, is,
-					 &type, sizeof(type)) ||
+			    !decoder_read_full(decoder, is,
+					       &type, sizeof(type)) ||
 			    !dsdlib_skip_to(decoder, is, chunk_end_offset))
 				return false;
 
@@ -208,7 +207,7 @@ dsdiff_handle_native_tag(InputStream &is,
 
 	struct dsdiff_native_tag metatag;
 
-	if (!dsdlib_read(nullptr, is, &metatag, sizeof(metatag)))
+	if (!decoder_read_full(nullptr, is, &metatag, sizeof(metatag)))
 		return;
 
 	uint32_t length = FromBE32(metatag.size);
@@ -221,7 +220,7 @@ dsdiff_handle_native_tag(InputStream &is,
 	char *label;
 	label = string;
 
-	if (!dsdlib_read(nullptr, is, label, (size_t)length))
+	if (!decoder_read_full(nullptr, is, label, (size_t)length))
 		return;
 
 	string[length] = '\0';
@@ -251,15 +250,17 @@ dsdiff_read_metadata_extra(Decoder *decoder, InputStream &is,
 	if (!dsdiff_read_chunk_header(decoder, is, chunk_header))
 		return false;
 
+	metadata->diar_offset = 0;
+	metadata->diti_offset = 0;
+
 #ifdef HAVE_ID3TAG
-	metadata->id3_size = 0;
+	metadata->id3_offset = 0;
 #endif
 
 	/* Now process all the remaining chunk headers in the stream
 	   and record their position and size */
 
-	const auto size = is.GetSize();
-	while (is.GetOffset() < size) {
+	do {
 		uint64_t chunk_size = chunk_header->GetSize();
 
 		/* DIIN chunk, is directly followed by other chunks  */
@@ -285,17 +286,11 @@ dsdiff_read_metadata_extra(Decoder *decoder, InputStream &is,
 			metadata->id3_size = chunk_size;
 		}
 #endif
-		if (chunk_size != 0) {
-			if (!dsdlib_skip(decoder, is, chunk_size))
-				break;
-		}
 
-		if (is.GetOffset() < size) {
-			if (!dsdiff_read_chunk_header(decoder, is, chunk_header))
-				return false;
-		}
-		chunk_size = 0;
-	}
+		if (!dsdlib_skip(decoder, is, chunk_size))
+			break;
+	} while (dsdiff_read_chunk_header(decoder, is, chunk_header));
+
 	/* done processing chunk headers, process tags if any */
 
 #ifdef HAVE_ID3TAG
@@ -329,7 +324,7 @@ dsdiff_read_metadata(Decoder *decoder, InputStream &is,
 		     DsdiffChunkHeader *chunk_header)
 {
 	DsdiffHeader header;
-	if (!dsdlib_read(decoder, is, &header, sizeof(header)) ||
+	if (!decoder_read_full(decoder, is, &header, sizeof(header)) ||
 	    !header.id.Equals("FRM8") ||
 	    !header.format.Equals("DSD "))
 		return false;
@@ -385,17 +380,17 @@ dsdiff_decode_chunk(Decoder &decoder, InputStream &is,
 	while (chunk_size > 0) {
 		/* see how much aligned data from the remaining chunk
 		   fits into the local buffer */
-		unsigned now_frames = buffer_frames;
 		size_t now_size = buffer_size;
 		if (chunk_size < (uint64_t)now_size) {
-			now_frames = (unsigned)chunk_size / frame_size;
+			unsigned now_frames =
+				(unsigned)chunk_size / frame_size;
 			now_size = now_frames * frame_size;
 		}
 
-		size_t nbytes = decoder_read(decoder, is, buffer, now_size);
-		if (nbytes != now_size)
+		if (!decoder_read_full(&decoder, is, buffer, now_size))
 			return false;
 
+		const size_t nbytes = now_size;
 		chunk_size -= nbytes;
 
 		if (lsbitfirst)

src/decoder/DsfDecoderPlugin.cxx

@@ -103,7 +103,7 @@ dsf_read_metadata(Decoder *decoder, InputStream &is,
 		  DsfMetaData *metadata)
 {
 	DsfHeader dsf_header;
-	if (!dsdlib_read(decoder, is, &dsf_header, sizeof(dsf_header)) ||
+	if (!decoder_read_full(decoder, is, &dsf_header, sizeof(dsf_header)) ||
 	    !dsf_header.id.Equals("DSD "))
 		return false;
 
@@ -117,7 +117,8 @@ dsf_read_metadata(Decoder *decoder, InputStream &is,
 
 	/* read the 'fmt ' chunk of the DSF file */
 	DsfFmtChunk dsf_fmt_chunk;
-	if (!dsdlib_read(decoder, is, &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) ||
+	if (!decoder_read_full(decoder, is,
+			       &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) ||
 	    !dsf_fmt_chunk.id.Equals("fmt "))
 		return false;
 
@@ -143,7 +144,7 @@ dsf_read_metadata(Decoder *decoder, InputStream &is,
 
 	/* read the 'data' chunk of the DSF file */
 	DsfDataChunk data_chunk;
-	if (!dsdlib_read(decoder, is, &data_chunk, sizeof(data_chunk)) ||
+	if (!decoder_read_full(decoder, is, &data_chunk, sizeof(data_chunk)) ||
 	    !data_chunk.id.Equals("data"))
 		return false;
 
@@ -240,17 +241,17 @@ dsf_decode_chunk(Decoder &decoder, InputStream &is,
 	while (chunk_size > 0) {
 		/* see how much aligned data from the remaining chunk
 		   fits into the local buffer */
-		unsigned now_frames = buffer_frames;
 		size_t now_size = buffer_size;
 		if (chunk_size < (uint64_t)now_size) {
-			now_frames = (unsigned)chunk_size / frame_size;
+			unsigned now_frames =
+				(unsigned)chunk_size / frame_size;
 			now_size = now_frames * frame_size;
 		}
 
-		size_t nbytes = decoder_read(&decoder, is, buffer, now_size);
-		if (nbytes != now_size)
+		if (!decoder_read_full(&decoder, is, buffer, now_size))
 			return false;
 
+		const size_t nbytes = now_size;
 		chunk_size -= nbytes;
 
 		if (bitreverse)

src/decoder/FaadDecoderPlugin.cxx

@@ -34,8 +34,6 @@
 #include <string.h>
 #include <unistd.h>
 
-#define AAC_MAX_CHANNELS	6
-
 static const unsigned adts_sample_rates[] =
     { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
 	16000, 12000, 11025, 8000, 7350, 0, 0, 0
@@ -66,16 +64,13 @@ adts_check_frame(const unsigned char *data)
 static size_t
 adts_find_frame(DecoderBuffer *buffer)
 {
-	size_t length, frame_length;
-	bool ret;
-
 	while (true) {
+		size_t length;
 		const uint8_t *data = (const uint8_t *)
 			decoder_buffer_read(buffer, &length);
 		if (data == nullptr || length < 8) {
 			/* not enough data yet */
-			ret = decoder_buffer_fill(buffer);
-			if (!ret)
+			if (!decoder_buffer_fill(buffer))
 				/* failed */
 				return 0;
 
@@ -86,7 +81,7 @@ adts_find_frame(DecoderBuffer *buffer)
 		const uint8_t *p = (const uint8_t *)memchr(data, 0xff, length);
 		if (p == nullptr) {
 			/* no marker - discard the buffer */
-			decoder_buffer_consume(buffer, length);
+			decoder_buffer_clear(buffer);
 			continue;
 		}
 
@@ -97,7 +92,7 @@ adts_find_frame(DecoderBuffer *buffer)
 		}
 
 		/* is it a frame? */
-		frame_length = adts_check_frame(data);
+		const size_t frame_length = adts_check_frame(data);
 		if (frame_length == 0) {
 			/* it's just some random 0xff byte; discard it
 			   and continue searching */
@@ -109,15 +104,11 @@ adts_find_frame(DecoderBuffer *buffer)
 			/* available buffer size is smaller than the
 			   frame will be - attempt to read more
 			   data */
-			ret = decoder_buffer_fill(buffer);
-			if (!ret) {
+			if (!decoder_buffer_fill(buffer)) {
 				/* not enough data; discard this frame
 				   to prevent a possible buffer
 				   overflow */
-				data = (const uint8_t *)
-					decoder_buffer_read(buffer, &length);
-				if (data != nullptr)
-					decoder_buffer_consume(buffer, length);
+				decoder_buffer_clear(buffer);
 			}
 
 			continue;
@@ -131,13 +122,18 @@ adts_find_frame(DecoderBuffer *buffer)
 static float
 adts_song_duration(DecoderBuffer *buffer)
 {
-	unsigned int frames, frame_length;
+	const InputStream &is = decoder_buffer_get_stream(buffer);
+	const bool estimate = !is.CheapSeeking();
+	const auto file_size = is.GetSize();
+	if (estimate && file_size <= 0)
+		return -1;
+
 	unsigned sample_rate = 0;
-	float frames_per_second;
 
 	/* Read all frames to ensure correct time and bitrate */
+	unsigned frames;
 	for (frames = 0;; frames++) {
-		frame_length = adts_find_frame(buffer);
+		const unsigned frame_length = adts_find_frame(buffer);
 		if (frame_length == 0)
 			break;
 
@@ -150,36 +146,52 @@ adts_song_duration(DecoderBuffer *buffer)
 			assert(frame_length <= buffer_length);
 
 			sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2];
+			if (sample_rate == 0)
+				break;
 		}
 
 		decoder_buffer_consume(buffer, frame_length);
+
+		if (estimate && frames == 128) {
+			/* if this is a remote file, don't slurp the
+			   whole file just for checking the song
+			   duration; instead, stop after some time and
+			   extrapolate the song duration from what we
+			   have until now */
+
+			const auto offset = is.GetOffset()
+				- decoder_buffer_available(buffer);
+			if (offset <= 0)
+				return -1;
+
+			frames = (frames * file_size) / offset;
+			break;
+		}
 	}
 
-	frames_per_second = (float)sample_rate / 1024.0;
-	if (frames_per_second <= 0)
+	if (sample_rate == 0)
 		return -1;
 
+	float frames_per_second = (float)sample_rate / 1024.0;
+	assert(frames_per_second > 0);
+
 	return (float)frames / frames_per_second;
 }
 
 static float
 faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 {
-	size_t fileread;
-	size_t tagsize;
-	size_t length;
-	bool success;
-
 	const auto size = is.GetSize();
-	fileread = size >= 0 ? size : 0;
+	const size_t fileread = size >= 0 ? size : 0;
 
 	decoder_buffer_fill(buffer);
+	size_t length;
 	const uint8_t *data = (const uint8_t *)
 		decoder_buffer_read(buffer, &length);
 	if (data == nullptr)
 		return -1;
 
-	tagsize = 0;
+	size_t tagsize = 0;
 	if (length >= 10 && !memcmp(data, "ID3", 3)) {
 		/* skip the ID3 tag */
 
@@ -188,7 +200,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 
 		tagsize += 10;
 
-		success = decoder_buffer_skip(buffer, tagsize) &&
+		const bool success = decoder_buffer_skip(buffer, tagsize) &&
 			decoder_buffer_fill(buffer);
 		if (!success)
 			return -1;
@@ -198,22 +210,20 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 			return -1;
 	}
 
-	if (is.IsSeekable() && length >= 2 &&
-	    data[0] == 0xFF && ((data[1] & 0xF6) == 0xF0)) {
+	if (length >= 8 && adts_check_frame(data) > 0) {
 		/* obtain the duration from the ADTS header */
+
+		if (!is.IsSeekable())
+			return -1;
+
 		float song_length = adts_song_duration(buffer);
 
 		is.LockSeek(tagsize, SEEK_SET, IgnoreError());
-
-		data = (const uint8_t *)decoder_buffer_read(buffer, &length);
-		if (data != nullptr)
-			decoder_buffer_consume(buffer, length);
-		decoder_buffer_fill(buffer);
+		decoder_buffer_clear(buffer);
 
 		return song_length;
 	} else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) {
 		/* obtain the duration from the ADIF header */
-		unsigned bit_rate;
 		size_t skip_size = (data[4] & 0x80) ? 9 : 0;
 
 		if (8 + skip_size > length)
@@ -221,7 +231,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 			   header */
 			return -1;
 
-		bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
+		unsigned bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
 			(data[5 + skip_size] << 11) |
 			(data[6 + skip_size] << 3) |
 			(data[7 + skip_size] & 0xE0);
@@ -234,6 +244,21 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 		return -1;
 }
 
+static NeAACDecHandle
+faad_decoder_new()
+{
+	const NeAACDecHandle decoder = NeAACDecOpen();
+
+	NeAACDecConfigurationPtr config =
+		NeAACDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+	config->downMatrix = 1;
+	config->dontUpSampleImplicitSBR = 0;
+	NeAACDecSetConfiguration(decoder, config);
+
+	return decoder;
+}
+
 /**
  * Wrapper for NeAACDecInit() which works around some API
  * inconsistencies in libfaad.
@@ -242,17 +267,6 @@ static bool
 faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
 		  AudioFormat &audio_format, Error &error)
 {
-	int32_t nbytes;
-	uint32_t sample_rate;
-	uint8_t channels;
-#ifdef HAVE_FAAD_LONG
-	/* neaacdec.h declares all arguments as "unsigned long", but
-	   internally expects uint32_t pointers.  To avoid gcc
-	   warnings, use this workaround. */
-	unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
-#else
-	uint32_t *sample_rate_p = &sample_rate;
-#endif
 
 	size_t length;
 	const unsigned char *data = (const unsigned char *)
@@ -262,11 +276,21 @@ faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
 		return false;
 	}
 
-	nbytes = NeAACDecInit(decoder,
-			      /* deconst hack, libfaad requires this */
-			      const_cast<unsigned char *>(data),
-			     length,
-			     sample_rate_p, &channels);
+	uint8_t channels;
+	uint32_t sample_rate;
+#ifdef HAVE_FAAD_LONG
+	/* neaacdec.h declares all arguments as "unsigned long", but
+	   internally expects uint32_t pointers.  To avoid gcc
+	   warnings, use this workaround. */
+	unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
+#else
+	uint32_t *sample_rate_p = &sample_rate;
+#endif
+	long nbytes = NeAACDecInit(decoder,
+				   /* deconst hack, libfaad requires this */
+				   const_cast<unsigned char *>(data),
+				   length,
+				   sample_rate_p, &channels);
 	if (nbytes < 0) {
 		error.Set(faad_decoder_domain, "Not an AAC stream");
 		return false;
@@ -306,29 +330,21 @@ faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer *buffer,
 static float
 faad_get_file_time_float(InputStream &is)
 {
-	DecoderBuffer *buffer;
-	float length;
+	DecoderBuffer *const buffer =
+		decoder_buffer_new(nullptr, is,
+				   FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
 
-	buffer = decoder_buffer_new(nullptr, is,
-				    FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
-	length = faad_song_duration(buffer, is);
+	float length = faad_song_duration(buffer, is);
 
 	if (length < 0) {
-		bool ret;
 		AudioFormat audio_format;
 
-		NeAACDecHandle decoder = NeAACDecOpen();
-
-		NeAACDecConfigurationPtr config =
-			NeAACDecGetCurrentConfiguration(decoder);
-		config->outputFormat = FAAD_FMT_16BIT;
-		NeAACDecSetConfiguration(decoder, config);
+		NeAACDecHandle decoder = faad_decoder_new();
 
 		decoder_buffer_fill(buffer);
 
-		ret = faad_decoder_init(decoder, buffer, audio_format,
-					IgnoreError());
-		if (ret)
+		if (faad_decoder_init(decoder, buffer, audio_format,
+				      IgnoreError()))
 			length = 0;
 
 		NeAACDecClose(decoder);
@@ -347,38 +363,25 @@ faad_get_file_time_float(InputStream &is)
 static int
 faad_get_file_time(InputStream &is)
 {
-	int file_time = -1;
-	float length;
+	float length = faad_get_file_time_float(is);
+	if (length < 0)
+		return -1;
 
-	if ((length = faad_get_file_time_float(is)) >= 0)
-		file_time = length + 0.5;
-
-	return file_time;
+	return int(length + 0.5);
 }
 
 static void
 faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
 {
-	float total_time = 0;
-	AudioFormat audio_format;
-	bool ret;
-	uint16_t bit_rate = 0;
-	DecoderBuffer *buffer;
+	DecoderBuffer *const buffer =
+		decoder_buffer_new(&mpd_decoder, is,
+				   FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
 
-	buffer = decoder_buffer_new(&mpd_decoder, is,
-				    FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
-	total_time = faad_song_duration(buffer, is);
+	const float total_time = faad_song_duration(buffer, is);
 
 	/* create the libfaad decoder */
 
-	NeAACDecHandle decoder = NeAACDecOpen();
-
-	NeAACDecConfigurationPtr config =
-		NeAACDecGetCurrentConfiguration(decoder);
-	config->outputFormat = FAAD_FMT_16BIT;
-	config->downMatrix = 1;
-	config->dontUpSampleImplicitSBR = 0;
-	NeAACDecSetConfiguration(decoder, config);
+	const NeAACDecHandle decoder = faad_decoder_new();
 
 	while (!decoder_buffer_is_full(buffer) && !is.LockIsEOF() &&
 	       decoder_get_command(mpd_decoder) == DecoderCommand::NONE) {
@@ -389,10 +392,11 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
 	/* initialize it */
 
 	Error error;
-	ret = faad_decoder_init(decoder, buffer, audio_format, error);
-	if (!ret) {
+	AudioFormat audio_format;
+	if (!faad_decoder_init(decoder, buffer, audio_format, error)) {
 		LogError(error);
 		NeAACDecClose(decoder);
+		decoder_buffer_free(buffer);
 		return;
 	}
 
@@ -403,21 +407,20 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
 	/* the decoder loop */
 
 	DecoderCommand cmd;
+	uint16_t bit_rate = 0;
 	do {
-		size_t frame_size;
-		const void *decoded;
-		NeAACDecFrameInfo frame_info;
-
 		/* find the next frame */
 
-		frame_size = adts_find_frame(buffer);
+		const size_t frame_size = adts_find_frame(buffer);
 		if (frame_size == 0)
 			/* end of file */
 			break;
 
 		/* decode it */
 
-		decoded = faad_decoder_decode(decoder, buffer, &frame_info);
+		NeAACDecFrameInfo frame_info;
+		const void *const decoded =
+			faad_decoder_decode(decoder, buffer, &frame_info);
 
 		if (frame_info.error > 0) {
 			FormatWarning(faad_decoder_domain,
@@ -461,6 +464,7 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
 	/* cleanup */
 
 	NeAACDecClose(decoder);
+	decoder_buffer_free(buffer);
 }
 
 static bool
@@ -468,7 +472,6 @@ faad_scan_stream(InputStream &is,
 		 const struct tag_handler *handler, void *handler_ctx)
 {
 	int file_time = faad_get_file_time(is);
-
 	if (file_time < 0)
 		return false;
 

src/decoder/FfmpegDecoderPlugin.cxx

@@ -38,7 +38,10 @@ extern "C" {
 #include <libavutil/avutil.h>
 #include <libavutil/log.h>
 #include <libavutil/mathematics.h>
-#include <libavutil/dict.h>
+
+#if LIBAVUTIL_VERSION_MAJOR >= 53
+#include <libavutil/frame.h>
+#endif
 }
 
 #include <assert.h>
@@ -120,8 +123,7 @@ mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence)
 	if (whence == AVSEEK_SIZE)
 		return stream->input.size;
 
-	Error error;
-	if (!stream->input.LockSeek(pos, whence, error))
+	if (!stream->input.LockSeek(pos, whence, IgnoreError()))
 		return -1;
 
 	return stream->input.offset;
@@ -195,6 +197,29 @@ time_to_ffmpeg(double t, const AVRational time_base)
 			    time_base);
 }
 
+/**
+ * Replace #AV_NOPTS_VALUE with the given fallback.
+ */
+static constexpr int64_t
+timestamp_fallback(int64_t t, int64_t fallback)
+{
+	return gcc_likely(t != int64_t(AV_NOPTS_VALUE))
+		? t
+		: fallback;
+}
+
+/**
+ * Accessor for AVStream::start_time that replaces AV_NOPTS_VALUE with
+ * zero.  We can't use AV_NOPTS_VALUE in calculations, and we simply
+ * assume that the stream's start time is zero, which appears to be
+ * the best way out of that situation.
+ */
+static int64_t
+start_time_fallback(const AVStream &stream)
+{
+	return timestamp_fallback(stream.start_time, 0);
+}
+
 static void
 copy_interleave_frame2(uint8_t *dest, uint8_t **src,
 		       unsigned nframes, unsigned nchannels,
@@ -224,6 +249,9 @@ copy_interleave_frame(const AVCodecContext *codec_context,
 					   codec_context->channels,
 					   frame->nb_samples,
 					   codec_context->sample_fmt, 1);
+	if (data_size <= 0)
+		return data_size;
+
 	if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
 	    codec_context->channels > 1) {
 		if(*global_buffer_size < data_size) {
@@ -252,13 +280,14 @@ static DecoderCommand
 ffmpeg_send_packet(Decoder &decoder, InputStream &is,
 		   const AVPacket *packet,
 		   AVCodecContext *codec_context,
-		   const AVRational *time_base,
+		   const AVStream *stream,
 		   AVFrame *frame,
 		   uint8_t **buffer, int *buffer_size)
 {
 	if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE)
 		decoder_timestamp(decoder,
-				  time_from_ffmpeg(packet->pts, *time_base));
+				  time_from_ffmpeg(packet->pts - start_time_fallback(*stream),
+				  stream->time_base));
 
 	AVPacket packet2 = *packet;
 
@@ -342,11 +371,9 @@ ffmpeg_probe(Decoder *decoder, InputStream &is)
 		PADDING = 16,
 	};
 
-	Error error;
-
 	unsigned char buffer[BUFFER_SIZE];
 	size_t nbytes = decoder_read(decoder, is, buffer, BUFFER_SIZE);
-	if (nbytes <= PADDING || !is.LockRewind(error))
+	if (nbytes <= PADDING || !is.LockRewind(IgnoreError()))
 		return nullptr;
 
 	/* some ffmpeg parsers (e.g. ac3_parser.c) read a few bytes
@@ -453,7 +480,11 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 	decoder_initialized(decoder, audio_format,
 			    input.seekable, total_time);
 
+#if LIBAVUTIL_VERSION_MAJOR >= 53
+	AVFrame *frame = av_frame_alloc();
+#else
 	AVFrame *frame = avcodec_alloc_frame();
+#endif
 	if (!frame) {
 		LogError(ffmpeg_domain, "Could not allocate frame");
 		avformat_close_input(&format_context);
@@ -473,7 +504,7 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 		if (packet.stream_index == audio_stream)
 			cmd = ffmpeg_send_packet(decoder, input,
 						 &packet, codec_context,
-						 &av_stream->time_base,
+						 av_stream,
 						 frame,
 						 &interleaved_buffer, &interleaved_buffer_size);
 		else
@@ -484,10 +515,11 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 		if (cmd == DecoderCommand::SEEK) {
 			int64_t where =
 				time_to_ffmpeg(decoder_seek_where(decoder),
-					       av_stream->time_base);
+					       av_stream->time_base) +
+				start_time_fallback(*av_stream);
 
 			if (av_seek_frame(format_context, audio_stream, where,
-					  AV_TIME_BASE) < 0)
+					  AVSEEK_FLAG_ANY) < 0)
 				decoder_seek_error(decoder);
 			else {
 				avcodec_flush_buffers(codec_context);
@@ -496,7 +528,9 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 		}
 	} while (cmd != DecoderCommand::STOP);
 
-#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0)
+#if LIBAVUTIL_VERSION_MAJOR >= 53
+	av_frame_free(&frame);
+#elif LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 28, 0)
 	avcodec_free_frame(&frame);
 #else
 	av_freep(&frame);

src/decoder/FfmpegMetaData.hxx

@@ -21,8 +21,6 @@
 #define MPD_FFMPEG_METADATA_HXX
 
 extern "C" {
-#include <libavformat/avformat.h>
-#include <libavutil/avutil.h>
 #include <libavutil/dict.h>
 }
 
@@ -35,6 +33,6 @@ struct tag_handler;
 
 void
 ffmpeg_scan_dictionary(AVDictionary *dict,
-		       const struct tag_handler *handler, void *handler_ctx);
+		       const tag_handler *handler, void *handler_ctx);
 
 #endif

src/decoder/GmeDecoderPlugin.cxx

@@ -117,6 +117,7 @@ gme_container_scan(const char *path_fs, const unsigned int tnum)
 	}
 
 	const unsigned num_songs = gme_track_count(emu);
+	gme_delete(emu);
 	/* if it only contains a single tune, don't treat as container */
 	if (num_songs < 2)
 		return nullptr;

src/decoder/MadDecoderPlugin.cxx

@@ -353,18 +353,8 @@ MadDecoder::ParseId3(size_t tagsize, Tag **mpd_tag)
 		memcpy(allocated, stream.this_frame, count);
 		mad_stream_skip(&(stream), count);
 
-		while (count < tagsize) {
-			size_t len;
-
-			len = decoder_read(decoder, input_stream,
-					   allocated + count, tagsize - count);
-			if (len == 0)
-				break;
-			else
-				count += len;
-		}
-
-		if (count != tagsize) {
+		if (!decoder_read_full(decoder, input_stream,
+				       allocated + count, tagsize - count)) {
 			LogDebug(mad_domain, "error parsing ID3 tag");
 			g_free(allocated);
 			return;
@@ -413,20 +403,7 @@ MadDecoder::ParseId3(size_t tagsize, Tag **mpd_tag)
 		mad_stream_skip(&stream, tagsize);
 	} else {
 		mad_stream_skip(&stream, count);
-
-		while (count < tagsize) {
-			size_t len = tagsize - count;
-			char ignored[1024];
-			if (len > sizeof(ignored))
-				len = sizeof(ignored);
-
-			len = decoder_read(decoder, input_stream,
-					   ignored, len);
-			if (len == 0)
-				break;
-			else
-				count += len;
-		}
+		decoder_skip(decoder, input_stream, tagsize - count);
 	}
 #endif
 }

src/decoder/OggFind.cxx

@@ -22,6 +22,8 @@
 #include "OggSyncState.hxx"
 #include "util/Error.hxx"
 
+#include <stdio.h>
+
 bool
 OggFindEOS(OggSyncState &oy, ogg_stream_state &os, ogg_packet &packet)
 {

src/decoder/OggUtil.cxx

@@ -81,7 +81,7 @@ bool
 OggExpectPageSeek(ogg_sync_state &oy, ogg_page &page,
 		  Decoder *decoder, InputStream &input_stream)
 {
-	size_t remaining_skipped = 16384;
+	size_t remaining_skipped = 32768;
 
 	while (true) {
 		int r = ogg_sync_pageseek(&oy, &page);

src/decoder/OpusDecoderPlugin.cxx

@@ -40,6 +40,7 @@
 #include <glib.h>
 
 #include <string.h>
+#include <stdio.h>
 
 static constexpr opus_int32 opus_sample_rate = 48000;
 

src/decoder/SndfileDecoderPlugin.cxx

@@ -31,10 +31,24 @@
 
 static constexpr Domain sndfile_domain("sndfile");
 
+struct SndfileInputStream {
+	Decoder *const decoder;
+	InputStream &is;
+
+	size_t Read(void *buffer, size_t size) {
+		/* libsndfile chokes on partial reads; therefore
+		   always force full reads */
+		return decoder_read_full(decoder, is, buffer, size)
+			? size
+			: 0;
+	}
+};
+
 static sf_count_t
 sndfile_vio_get_filelen(void *user_data)
 {
-	const InputStream &is = *(const InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
+	const InputStream &is = sis.is;
 
 	return is.GetSize();
 }
@@ -42,10 +56,14 @@ sndfile_vio_get_filelen(void *user_data)
 static sf_count_t
 sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
 {
-	InputStream &is = *(InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
+	InputStream &is = sis.is;
 
-	if (!is.LockSeek(offset, whence, IgnoreError()))
+	Error error;
+	if (!is.LockSeek(offset, whence, error)) {
+		LogError(error, "Seek failed");
 		return -1;
+	}
 
 	return is.GetOffset();
 }
@@ -53,16 +71,9 @@ sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
 static sf_count_t
 sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
 {
-	InputStream &is = *(InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
 
-	Error error;
-	size_t nbytes = is.LockRead(ptr, count, error);
-	if (nbytes == 0 && error.IsDefined()) {
-		LogError(error);
-		return -1;
-	}
-
-	return nbytes;
+	return sis.Read(ptr, count);
 }
 
 static sf_count_t
@@ -77,7 +88,8 @@ sndfile_vio_write(gcc_unused const void *ptr,
 static sf_count_t
 sndfile_vio_tell(void *user_data)
 {
-	const InputStream &is = *(const InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
+	const InputStream &is = sis.is;
 
 	return is.GetOffset();
 }
@@ -123,7 +135,8 @@ sndfile_stream_decode(Decoder &decoder, InputStream &is)
 
 	info.format = 0;
 
-	sf = sf_open_virtual(&vio, SFM_READ, &info, &is);
+	SndfileInputStream sis{&decoder, is};
+	sf = sf_open_virtual(&vio, SFM_READ, &info, &sis);
 	if (sf == nullptr) {
 		LogWarning(sndfile_domain, "sf_open_virtual() failed");
 		return;

src/encoder/OpusEncoderPlugin.cxx

@@ -34,6 +34,7 @@
 #include <glib.h>
 
 #include <assert.h>
+#include <stdlib.h>
 
 struct opus_encoder {
 	/** the base class */

src/event/DeferredMonitor.cxx

@@ -27,9 +27,11 @@ DeferredMonitor::Cancel()
 #ifdef USE_EPOLL
 	pending = false;
 #else
-	const auto id = source_id.exchange(0);
-	if (id != 0)
-		g_source_remove(id);
+	const ScopeLock protect(mutex);
+	if (source_id != 0) {
+		g_source_remove(source_id);
+		source_id = 0;
+	}
 #endif
 }
 
@@ -40,10 +42,9 @@ DeferredMonitor::Schedule()
 	if (!pending.exchange(true))
 		fd.Write();
 #else
-	const unsigned id = loop.AddIdle(Callback, this);
-	const auto old_id = source_id.exchange(id);
-	if (old_id != 0)
-		g_source_remove(old_id);
+	const ScopeLock protect(mutex);
+	if (source_id == 0)
+		source_id = loop.AddIdle(Callback, this);
 #endif
 }
 
@@ -65,9 +66,16 @@ DeferredMonitor::OnSocketReady(unsigned)
 void
 DeferredMonitor::Run()
 {
-	const auto id = source_id.exchange(0);
-	if (id != 0)
-		RunDeferred();
+	{
+		const ScopeLock protect(mutex);
+		if (source_id == 0)
+			/* cancelled */
+			return;
+
+		source_id = 0;
+	}
+
+	RunDeferred();
 }
 
 gboolean

src/event/DeferredMonitor.hxx

@@ -27,6 +27,7 @@
 #include "SocketMonitor.hxx"
 #include "WakeFD.hxx"
 #else
+#include "thread/Mutex.hxx"
 #include <glib.h>
 #endif
 
@@ -48,7 +49,9 @@ class DeferredMonitor
 #else
 	EventLoop &loop;
 
-	std::atomic<guint> source_id;
+	Mutex mutex;
+
+	guint source_id;
 #endif
 
 public:

src/event/SignalMonitor.cxx

@@ -39,6 +39,12 @@
 
 #include <algorithm>
 
+#ifdef USE_SIGNALFD
+#include <pthread.h>
+#endif
+
+#include <signal.h>
+
 class SignalMonitor final : private SocketMonitor {
 #ifdef USE_SIGNALFD
 	SignalFD fd;
@@ -99,7 +105,21 @@ static std::atomic_bool signal_pending[MAX_SIGNAL];
 
 static Manual<SignalMonitor> monitor;
 
-#ifndef USE_SIGNALFD
+#ifdef USE_SIGNALFD
+
+/**
+ * This is a pthread_atfork() callback that unblocks the signals that
+ * were blocked for our signalfd().  Without this, our child processes
+ * would inherit the blocked signals.
+ */
+static void
+at_fork_child()
+{
+	sigprocmask(SIG_UNBLOCK, &signal_mask, nullptr);
+}
+
+#else
+
 static void
 SignalCallback(int signo)
 {
@@ -108,6 +128,7 @@ SignalCallback(int signo)
 	if (!signal_pending[signo].exchange(true))
 		monitor->WakeUp();
 }
+
 #endif
 
 void
@@ -115,6 +136,8 @@ SignalMonitorInit(EventLoop &loop)
 {
 #ifdef USE_SIGNALFD
 	sigemptyset(&signal_mask);
+
+	pthread_atfork(nullptr, nullptr, at_fork_child);
 #endif
 
 	monitor.Construct(loop);

src/filter/RouteFilterPlugin.cxx

@@ -56,6 +56,7 @@
 #include <assert.h>
 #include <string.h>
 #include <stdint.h>
+#include <stdlib.h>
 
 class RouteFilter final : public Filter {
 	/**

src/input/CdioParanoiaInputPlugin.cxx

@@ -41,7 +41,12 @@
 #include <glib.h>
 #include <assert.h>
 
+#ifdef HAVE_CDIO_PARANOIA_PARANOIA_H
+#include <cdio/paranoia/paranoia.h>
+#else
 #include <cdio/paranoia.h>
+#endif
+
 #include <cdio/cd_types.h>
 
 struct CdioParanoiaInputStream {

src/input/CurlInputPlugin.cxx

@@ -273,12 +273,27 @@ public:
 		SocketAction(CURL_SOCKET_TIMEOUT, 0);
 	}
 
+	/**
+	 * This is a kludge to allow pausing/resuming a stream with
+	 * libcurl < 7.32.0.  Read the curl_easy_pause manpage for
+	 * more information.
+	 */
+	void ResumeSockets() {
+		int running_handles;
+		curl_multi_socket_all(multi, &running_handles);
+	}
+
 private:
 	static int TimerFunction(CURLM *multi, long timeout_ms, void *userp);
 
 	virtual void OnTimeout() override;
 };
 
+/**
+ * libcurl version number encoded in a 24 bit integer.
+ */
+static unsigned curl_version_num;
+
 /** libcurl should accept "ICY 200 OK" */
 static struct curl_slist *http_200_aliases;
 
@@ -330,6 +345,13 @@ input_curl_resume(struct input_curl *c)
 	if (c->paused) {
 		c->paused = false;
 		curl_easy_pause(c->easy, CURLPAUSE_CONT);
+
+		if (curl_version_num < 0x072000)
+			/* libcurl older than 7.32.0 does not update
+			   its sockets after curl_easy_pause(); force
+			   libcurl to do it now */
+			curl_multi->ResumeSockets();
+
 		curl_multi->InvalidateSockets();
 	}
 }
@@ -586,6 +608,16 @@ input_curl_init(const config_param &param, Error &error)
 		return false;
 	}
 
+	const auto version_info = curl_version_info(CURLVERSION_FIRST);
+	if (version_info != nullptr) {
+		FormatDebug(curl_domain, "version %s", version_info->version);
+		if (version_info->features & CURL_VERSION_SSL)
+			FormatDebug(curl_domain, "with %s",
+				    version_info->ssl_version);
+
+		curl_version_num = version_info->version_num;
+	}
+
 	http_200_aliases = curl_slist_append(http_200_aliases, "ICY 200 OK");
 
 	proxy = param.GetBlockValue("proxy");

src/mixer/AlsaMixerPlugin.cxx

@@ -24,6 +24,7 @@
 #include "Main.hxx"
 #include "event/MultiSocketMonitor.hxx"
 #include "event/Loop.hxx"
+#include "event/Call.hxx"
 #include "util/ASCII.hxx"
 #include "util/ReusableArray.hxx"
 #include "util/Error.hxx"
@@ -46,10 +47,22 @@ class AlsaMixerMonitor final : private MultiSocketMonitor {
 public:
 	AlsaMixerMonitor(EventLoop &_loop, snd_mixer_t *_mixer)
 		:MultiSocketMonitor(_loop), mixer(_mixer) {
+#ifdef USE_EPOLL
 		_loop.AddCall([this](){ InvalidateSockets(); });
+#else
+		_loop.AddIdle(InitAlsaMixerMonitor, this);
+#endif
 	}
 
 private:
+#ifndef USE_EPOLL
+	static gboolean InitAlsaMixerMonitor(gpointer data) {
+		AlsaMixerMonitor &amm = *(AlsaMixerMonitor *)data;
+		amm.InvalidateSockets();
+		return false;
+	}
+#endif
+
 	virtual int PrepareSockets() override;
 	virtual void DispatchSockets() override;
 };

src/output/AlsaOutputPlugin.cxx

@@ -106,12 +106,16 @@ struct AlsaOutput {
 	snd_pcm_uframes_t period_position;
 
 	/**
-	 * Set to non-zero when the Raspberry Pi workaround has been
-	 * activated in alsa_recover(); decremented by each write.
-	 * This will avoid activating it again, leading to an endless
-	 * loop.  This problem was observed with a "RME Digi9636/52".
+	 * Do we need to call snd_pcm_prepare() before the next write?
+	 * It means that we put the device to SND_PCM_STATE_SETUP by
+	 * calling snd_pcm_drop().
+	 *
+	 * Without this flag, we could easily recover after a failed
+	 * optimistic write (returning -EBADFD), but the Raspberry Pi
+	 * audio driver is infamous for generating ugly artefacts from
+	 * this.
 	 */
-	unsigned pi_workaround;
+	bool must_prepare;
 
 	/**
 	 * This buffer gets allocated after opening the ALSA device.
@@ -676,8 +680,6 @@ alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
-	ad->pi_workaround = 0;
-
 	int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
 			       SND_PCM_STREAM_PLAYBACK, ad->mode);
 	if (err < 0) {
@@ -699,6 +701,8 @@ alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
 	ad->in_frame_size = audio_format.GetFrameSize();
 	ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format);
 
+	ad->must_prepare = false;
+
 	return true;
 }
 
@@ -736,29 +740,6 @@ alsa_recover(AlsaOutput *ad, int err)
 	case SND_PCM_STATE_XRUN:
 		ad->period_position = 0;
 		err = snd_pcm_prepare(ad->pcm);
-
-		if (err == 0 && ad->pi_workaround == 0) {
-			/* this works around a driver bug observed on
-			   the Raspberry Pi: after snd_pcm_drop(), the
-			   whole ring buffer must be invalidated, but
-			   the snd_pcm_prepare() call above makes the
-			   driver play random data that just happens
-			   to be still in the buffer; by adding and
-			   cancelling some silence, this bug does not
-			   occur */
-			alsa_write_silence(ad, ad->period_frames);
-
-			/* cancel the silence data right away to avoid
-			   increasing latency; even though this
-			   function call invalidates the portion of
-			   silence, the driver seems to avoid the
-			   bug */
-			snd_pcm_reset(ad->pcm);
-
-			/* disable the workaround for some time */
-			ad->pi_workaround = 8;
-		}
-
 		break;
 	case SND_PCM_STATE_DISCONNECTED:
 		break;
@@ -801,6 +782,7 @@ alsa_cancel(struct audio_output *ao)
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
 	ad->period_position = 0;
+	ad->must_prepare = true;
 
 	snd_pcm_drop(ad->pcm);
 }
@@ -822,6 +804,16 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 
 	assert(size % ad->in_frame_size == 0);
 
+	if (ad->must_prepare) {
+		ad->must_prepare = false;
+
+		int err = snd_pcm_prepare(ad->pcm);
+		if (err < 0) {
+			error.Set(alsa_output_domain, err, snd_strerror(-err));
+			return 0;
+		}
+	}
+
 	chunk = ad->pcm_export->Export(chunk, size, size);
 
 	assert(size % ad->out_frame_size == 0);
@@ -834,9 +826,6 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 			ad->period_position = (ad->period_position + ret)
 				% ad->period_frames;
 
-			if (ad->pi_workaround > 0)
-				--ad->pi_workaround;
-
 			size_t bytes_written = ret * ad->out_frame_size;
 			return ad->pcm_export->CalcSourceSize(bytes_written);
 		}

src/output/HttpdClient.cxx

@@ -30,6 +30,7 @@
 
 #include <assert.h>
 #include <string.h>
+#include <stdio.h>
 
 HttpdClient::~HttpdClient()
 {

src/output/HttpdInternal.hxx

@@ -30,6 +30,11 @@
 #include "thread/Mutex.hxx"
 #include "event/ServerSocket.hxx"
 
+#ifdef _LIBCPP_VERSION
+/* can't use incomplete template arguments with libc++ */
+#include "HttpdClient.hxx"
+#endif
+
 #include <forward_list>
 
 struct config_param;

src/output/OSXOutputPlugin.cxx

@@ -363,9 +363,9 @@ osx_output_open(struct audio_output *ao, AudioFormat &audio_format,
 
 	OSStatus status = AudioUnitInitialize(od->au);
 	if (status != noErr) {
-		error.Set(osx_output_domain, status,
-			  "Unable to initialize OS X audio unit: %s",
-			  GetMacOSStatusCommentString(status));
+		error.Format(osx_output_domain, status,
+			     "Unable to initialize OS X audio unit: %s",
+			     GetMacOSStatusCommentString(status));
 		return false;
 	}
 

src/output/OpenALOutputPlugin.cxx

@@ -25,7 +25,7 @@
 
 #include <glib.h>
 
-#ifndef HAVE_OSX
+#ifndef __APPLE__
 #include <AL/al.h>
 #include <AL/alc.h>
 #else

src/output/PulseOutputPlugin.cxx

@@ -369,8 +369,6 @@ pulse_output_enable(struct audio_output *ao, Error &error)
 
 	po->mainloop = pa_threaded_mainloop_new();
 	if (po->mainloop == nullptr) {
-		g_free(po);
-
 		error.Set(pulse_output_domain,
 			  "pa_threaded_mainloop_new() has failed");
 		return false;

src/pcm/PcmResample.cxx

@@ -148,3 +148,26 @@ PcmResampler::Resample32(unsigned channels, unsigned src_rate,
 					src_rate, src_buffer, src_size,
 					dest_rate, dest_size_r);
 }
+
+const int32_t *
+PcmResampler::Resample24(unsigned channels, unsigned src_rate,
+			 const int32_t *src_buffer, size_t src_size,
+			 unsigned dest_rate, size_t *dest_size_r,
+			 Error &error_r)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	if (pcm_resample_lsr_enabled())
+		return pcm_resample_lsr_24(this, channels,
+					   src_rate, src_buffer, src_size,
+					   dest_rate, dest_size_r,
+					   error_r);
+#else
+	(void)error_r;
+#endif
+
+	/* reuse the 32 bit code - the resampler code doesn't care if
+	   the upper 8 bits are actually used */
+	return pcm_resample_fallback_32(this, channels,
+					src_rate, src_buffer, src_size,
+					dest_rate, dest_size_r);
+}

src/pcm/PcmResample.hxx

@@ -124,13 +124,7 @@ struct PcmResampler {
 	const int32_t *Resample24(unsigned channels, unsigned src_rate,
 				  const int32_t *src_buffer, size_t src_size,
 				  unsigned dest_rate, size_t *dest_size_r,
-				  Error &error_r)
-	{
-		/* reuse the 32 bit code - the resampler code doesn't care if
-		   the upper 8 bits are actually used */
-		return Resample32(channels, src_rate, src_buffer, src_size,
-				  dest_rate, dest_size_r, error_r);
-	}
+				  Error &error_r);
 };
 
 bool

src/pcm/PcmResampleInternal.hxx

@@ -69,6 +69,15 @@ pcm_resample_lsr_32(PcmResampler *state,
 		    unsigned dest_rate, size_t *dest_size_r,
 		    Error &error);
 
+const int32_t *
+pcm_resample_lsr_24(PcmResampler *state,
+		    unsigned channels,
+		    unsigned src_rate,
+		    const int32_t *src_buffer,
+		    size_t src_size,
+		    unsigned dest_rate, size_t *dest_size_r,
+		    Error &error);
+
 #endif
 
 const int16_t *

src/pcm/PcmResampleLibsamplerate.cxx

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "PcmResampleInternal.hxx"
+#include "PcmUtils.hxx"
 #include "util/ASCII.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
@@ -283,3 +284,27 @@ pcm_resample_lsr_32(PcmResampler *state,
 
 	return dest_buffer;
 }
+
+const int32_t *
+pcm_resample_lsr_24(PcmResampler *state,
+		    unsigned channels,
+		    unsigned src_rate,
+		    const int32_t *src_buffer, size_t src_size,
+		    unsigned dest_rate, size_t *dest_size_r,
+		    Error &error)
+{
+	const auto result = pcm_resample_lsr_32(state, channels,
+						src_rate, src_buffer, src_size,
+						dest_rate, dest_size_r,
+						error);
+	if (result != nullptr)
+		/* src_float_to_int_array() clamps for 32 bit
+		   integers; now make sure everything's fine for 24
+		   bit */
+		/* TODO: eliminate the 32 bit clamp to reduce overhead */
+		PcmClampN<int32_t, int32_t, 24>(const_cast<int32_t *>(result),
+						result,
+						*dest_size_r / sizeof(*result));
+
+	return result;
+}

src/pcm/PcmUtils.hxx

@@ -63,4 +63,16 @@ PcmClamp(U x)
 	return T(x);
 }
 
+/**
+ * Check if the values in this buffer are within the range of the
+ * provided bit size, and clamps them whenever necessary.
+ */
+template<typename T, typename U, unsigned bits>
+static inline void
+PcmClampN(T *dest, const U *src, unsigned n)
+{
+	while (n-- > 0)
+		*dest++ = PcmClamp<T, U, bits>(*src++);
+}
+
 #endif

src/playlist/PlsPlaylistPlugin.cxx

@@ -31,6 +31,7 @@
 #include <glib.h>
 
 #include <string>
+#include <stdio.h>
 
 static constexpr Domain pls_domain("pls");
 
@@ -68,7 +69,6 @@ pls_parser(GKeyFile *keyfile, std::forward_list<SongPointer> &songs)
 			FormatError(pls_domain, "Invalid PLS entry %s: '%s'",
 				    key, error->message);
 			g_error_free(error);
-			g_free(key);
 			return;
 		}
 

src/system/ByteOrder.hxx

@@ -33,11 +33,23 @@
 #include <stdint.h>
 
 #if defined(__i386__) || defined(__x86_64__) || defined(__ARMEL__)
-#define IS_LITTLE_ENDIAN true
-#define IS_BIG_ENDIAN false
+/* well-known little-endian */
+#  define IS_LITTLE_ENDIAN true
+#  define IS_BIG_ENDIAN false
+#elif defined(__MIPSEB__)
+/* well-known big-endian */
+#  define IS_LITTLE_ENDIAN false
+#  define IS_BIG_ENDIAN true
 #else
-#define IS_LITTLE_ENDIAN false
-#define IS_BIG_ENDIAN true
+/* generic compile-time check */
+#  include <endian.h>
+#  if __BYTE_ORDER == __LITTLE_ENDIAN
+#    define IS_LITTLE_ENDIAN true
+#    define IS_BIG_ENDIAN false
+#  else
+#    define IS_LITTLE_ENDIAN false
+#    define IS_BIG_ENDIAN true
+#  endif
 #endif
 
 static inline constexpr bool

src/system/fd_util.h

@@ -104,6 +104,11 @@ socketpair_cloexec_nonblock(int domain, int type, int protocol, int sv[2]);
 
 #endif
 
+#ifdef HAVE_LIBMPDCLIENT
+/* Avoid symbol conflict with statically linked libmpdclient */
+#define socket_cloexec_nonblock socket_cloexec_nonblock_noconflict
+#endif
+
 /**
  * Wrapper for socket(), which sets the CLOEXEC and the NONBLOCK flag
  * (atomically if supported by the OS).

src/tag/Riff.cxx

@@ -87,7 +87,8 @@ riff_seek_id3(FILE *file)
 			/* pad byte */
 			++size;
 
-		if (memcmp(chunk.id, "id3 ", 4) == 0)
+		if (memcmp(chunk.id, "id3 ", 4) == 0 ||
+		    memcmp(chunk.id, "ID3 ", 4) == 0)
 			/* found it! */
 			return size;
 

src/tag/TagConfig.cxx

@@ -39,7 +39,7 @@ TagLoadConfig()
 	if (value == nullptr)
 		return;
 
-	std::fill_n(ignore_tag_items, TAG_NUM_OF_ITEM_TYPES, true);
+	std::fill_n(ignore_tag_items, size_t(TAG_NUM_OF_ITEM_TYPES), true);
 
 	if (StringEqualsCaseASCII(value, "none"))
 		return;

src/util/Error.hxx

@@ -24,6 +24,7 @@
 #include "Compiler.h"
 
 #include <string>
+#include <algorithm>
 
 #include <assert.h>
 

src/util/LazyRandomEngine.hxx

@@ -49,12 +49,12 @@ public:
 	 */
 	void AutoCreate();
 
-	result_type min() const {
-		return engine->min();
+	static constexpr result_type min() {
+		return std::mt19937::min();
 	}
 
-	result_type max() const {
-		return engine->max();
+	static constexpr result_type max() {
+		return std::mt19937::max();
 	}
 
 	result_type operator()() {

src/util/PeakBuffer.cxx

@@ -130,8 +130,9 @@ PeakBuffer::Append(const void *data, size_t length)
 			return true;
 	}
 
-	if (peak_buffer == nullptr && peak_size > 0) {
-		peak_buffer = (fifo_buffer *)HugeAllocate(peak_size);
+	if (peak_buffer == nullptr) {
+		if (peak_size > 0)
+			peak_buffer = (fifo_buffer *)HugeAllocate(peak_size);
 		if (peak_buffer == nullptr)
 			return false;
 

src/util/RefCount.hxx

@@ -42,7 +42,12 @@ class RefCount {
 	std::atomic_uint n;
 
 public:
-	constexpr RefCount():n(1) {}
+#ifndef _LIBCPP_VERSION
+	/* the "constexpr" is missing in libc++'s "atomic"
+	   implementation */
+	constexpr
+#endif
+	RefCount():n(1) {}
 
 	void Increment() {
 		++n;

src/util/UriUtil.cxx

@@ -32,7 +32,8 @@ const char *
 uri_get_suffix(const char *uri)
 {
 	const char *suffix = strrchr(uri, '.');
-	if (suffix == nullptr)
+	if (suffix == nullptr || suffix == uri ||
+	    suffix[-1] == '/' || suffix[-1] == '\\')
 		return nullptr;
 
 	++suffix;

test/FakeDecoderAPI.cxx

@@ -0,0 +1,144 @@
+/*
+ * Copyright (C) 2003-2012 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 "DecoderAPI.hxx"
+#include "InputStream.hxx"
+#include "util/Error.hxx"
+#include "Compiler.h"
+
+#include <glib.h>
+
+#include <unistd.h>
+
+void
+decoder_initialized(gcc_unused Decoder &decoder,
+		    gcc_unused const AudioFormat audio_format,
+		    gcc_unused bool seekable,
+		    gcc_unused float total_time)
+{
+}
+
+DecoderCommand
+decoder_get_command(gcc_unused Decoder &decoder)
+{
+	return DecoderCommand::NONE;
+}
+
+void
+decoder_command_finished(gcc_unused Decoder &decoder)
+{
+}
+
+double
+decoder_seek_where(gcc_unused Decoder &decoder)
+{
+	return 1.0;
+}
+
+void
+decoder_seek_error(gcc_unused Decoder &decoder)
+{
+}
+
+size_t
+decoder_read(gcc_unused Decoder *decoder,
+	     InputStream &is,
+	     void *buffer, size_t length)
+{
+	return is.LockRead(buffer, length, IgnoreError());
+}
+
+bool
+decoder_read_full(Decoder *decoder, InputStream &is,
+		  void *_buffer, size_t size)
+{
+	uint8_t *buffer = (uint8_t *)_buffer;
+
+	while (size > 0) {
+		size_t nbytes = decoder_read(decoder, is, buffer, size);
+		if (nbytes == 0)
+			return false;
+
+		buffer += nbytes;
+		size -= nbytes;
+	}
+
+	return true;
+}
+
+bool
+decoder_skip(Decoder *decoder, InputStream &is, size_t size)
+{
+	while (size > 0) {
+		char buffer[1024];
+		size_t nbytes = decoder_read(decoder, is, buffer,
+					     std::min(sizeof(buffer), size));
+		if (nbytes == 0)
+			return false;
+
+		size -= nbytes;
+	}
+
+	return true;
+}
+
+void
+decoder_timestamp(gcc_unused Decoder &decoder,
+		  gcc_unused double t)
+{
+}
+
+DecoderCommand
+decoder_data(gcc_unused Decoder &decoder,
+	     gcc_unused InputStream *is,
+	     const void *data, size_t datalen,
+	     gcc_unused uint16_t kbit_rate)
+{
+	gcc_unused ssize_t nbytes = write(1, data, datalen);
+	return DecoderCommand::NONE;
+}
+
+DecoderCommand
+decoder_tag(gcc_unused Decoder &decoder,
+	    gcc_unused InputStream *is,
+	    gcc_unused Tag &&tag)
+{
+	return DecoderCommand::NONE;
+}
+
+void
+decoder_replay_gain(gcc_unused Decoder &decoder,
+		    const ReplayGainInfo *rgi)
+{
+	const ReplayGainTuple *tuple = &rgi->tuples[REPLAY_GAIN_ALBUM];
+	if (tuple->IsDefined())
+		g_printerr("replay_gain[album]: gain=%f peak=%f\n",
+			   tuple->gain, tuple->peak);
+
+	tuple = &rgi->tuples[REPLAY_GAIN_TRACK];
+	if (tuple->IsDefined())
+		g_printerr("replay_gain[track]: gain=%f peak=%f\n",
+			   tuple->gain, tuple->peak);
+}
+
+void
+decoder_mixramp(gcc_unused Decoder &decoder, gcc_unused MixRampInfo &&mix_ramp)
+{
+}