release v0.18.6

The "mode" of the second ReplayGainFilter was never set, and thus
replay gain was never applied to the new song during cross-fade.

.gitignore

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

NEWS

@@ -1,3 +1,40 @@
+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)
 

configure.ac

@@ -1,6 +1,6 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.18.3, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.18.6, musicpd-dev-team@lists.sourceforge.net)
 
 VERSION_MAJOR=0
 VERSION_MINOR=18
@@ -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)

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

@@ -210,7 +210,6 @@ input {
 #	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
@@ -223,7 +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
@@ -327,12 +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

@@ -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>
@@ -1883,6 +1968,18 @@ 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/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/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,

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/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/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

@@ -1081,8 +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/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/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;
 }
@@ -631,6 +639,8 @@ 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/DsdLib.hxx

@@ -47,6 +47,17 @@ public:
 	}
 };
 
+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_read(Decoder *decoder, InputStream &is,
 	    void *data, size_t length);

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
@@ -294,7 +294,6 @@ dsdiff_read_metadata_extra(Decoder *decoder, InputStream &is,
 			if (!dsdiff_read_chunk_header(decoder, is, chunk_header))
 				return false;
 		}
-		chunk_size = 0;
 	}
 	/* done processing chunk headers, process tags if any */
 
@@ -385,10 +384,10 @@ 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;
 		}
 

src/decoder/DsfDecoderPlugin.cxx

@@ -240,10 +240,10 @@ 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;
 		}
 

src/decoder/FfmpegDecoderPlugin.cxx

@@ -120,8 +120,7 @@ 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;
@@ -252,13 +251,14 @@ 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 - stream->start_time,
+				  stream->time_base));
 
 	AVPacket packet2 = *packet;
 
@@ -342,11 +342,9 @@ 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
@@ -473,7 +471,7 @@ 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
@@ -484,7 +482,8 @@ 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) +
+				av_stream->start_time;
 
 			if (av_seek_frame(format_context, audio_stream, where,
 					  AV_TIME_BASE) < 0)

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/encoder/OpusEncoderPlugin.cxx

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

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/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/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/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/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/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;

test/run_convert.cxx

@@ -36,6 +36,7 @@
 
 #include <assert.h>
 #include <stddef.h>
+#include <stdlib.h>
 #include <unistd.h>
 
 static void