mpd version 0.16.4

.. and a minor tweak for libFLAC+libogg detection.

Makefile.am

@@ -11,6 +11,7 @@ noinst_LIBRARIES =
 
 src_mpd_CFLAGS = $(AM_CFLAGS) $(MPD_CFLAGS)
 src_mpd_CPPFLAGS = $(AM_CPPFLAGS) \
+	$(AVAHI_CFLAGS) \
 	$(LIBWRAP_CFLAGS) \
 	$(SQLITE_CFLAGS) \
 	$(ARCHIVE_CFLAGS) \
@@ -21,6 +22,7 @@ src_mpd_CPPFLAGS = $(AM_CPPFLAGS) \
 	$(FILTER_CFLAGS) \
 	$(OUTPUT_CFLAGS)
 src_mpd_LDADD = $(MPD_LIBS) \
+	$(AVAHI_LIBS) \
 	$(LIBWRAP_LDFLAGS) \
 	$(SQLITE_LIBS) \
 	$(ARCHIVE_LIBS) \
@@ -859,6 +861,7 @@ test_run_input_LDADD = $(MPD_LIBS) \
 	$(INPUT_LIBS) \
 	$(GLIB_LIBS)
 test_run_input_SOURCES = test/run_input.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c src/utils.c \
 	src/tag.c src/tag_pool.c src/tag_save.c \
 	src/fd_util.c \
@@ -906,6 +909,7 @@ test_run_decoder_LDADD = $(MPD_LIBS) \
 	$(INPUT_LIBS) $(DECODER_LIBS) \
 	$(GLIB_LIBS)
 test_run_decoder_SOURCES = test/run_decoder.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c src/utils.c src/log.c \
 	src/tag.c src/tag_pool.c \
 	src/replay_gain_info.c \
@@ -946,6 +950,7 @@ test_run_filter_LDADD = $(MPD_LIBS) \
 	$(SAMPLERATE_LIBS) \
 	$(GLIB_LIBS)
 test_run_filter_SOURCES = test/run_filter.c \
+	test/stdbin.h \
 	src/filter_plugin.c \
 	src/filter_registry.c \
 	src/conf.c src/tokenizer.c src/utils.c \
@@ -968,6 +973,7 @@ endif
 if ENABLE_ENCODER
 noinst_PROGRAMS += test/run_encoder
 test_run_encoder_SOURCES = test/run_encoder.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c \
 	src/utils.c \
 	src/tag.c src/tag_pool.c \
@@ -975,12 +981,15 @@ test_run_encoder_SOURCES = test/run_encoder.c \
 	src/audio_format.c \
 	src/audio_parser.c \
 	$(ENCODER_SRC)
+test_run_encoder_CPPFLAGS = $(AM_CPPFLAGS) \
+	$(ENCODER_CFLAGS)
 test_run_encoder_LDADD = $(MPD_LIBS) \
 	$(ENCODER_LIBS) \
 	$(GLIB_LIBS)
 endif
 
 test_software_volume_SOURCES = test/software_volume.c \
+	test/stdbin.h \
 	src/audio_check.c \
 	src/audio_parser.c \
 	src/pcm_volume.c
@@ -988,6 +997,7 @@ test_software_volume_LDADD = \
 	$(GLIB_LIBS)
 
 test_run_normalize_SOURCES = test/run_normalize.c \
+	test/stdbin.h \
 	src/audio_check.c \
 	src/audio_parser.c \
 	src/AudioCompress/compress.c
@@ -1025,6 +1035,7 @@ test_run_output_LDADD = $(MPD_LIBS) \
 	$(OUTPUT_LIBS) \
 	$(GLIB_LIBS)
 test_run_output_SOURCES = test/run_output.c \
+	test/stdbin.h \
 	src/conf.c src/tokenizer.c src/utils.c src/log.c \
 	src/audio_check.c \
 	src/audio_format.c \

NEWS

@@ -1,4 +1,52 @@
-ver 0.16.1 (2010/01/09)
+ver 0.16.4 (2011/09/01)
+* don't abort configure when avahi is not found
+* auto-detect libmad without pkg-config
+* fix memory leaks
+* don't resume playback when seeking to another song while paused
+* apply follow_inside_symlinks to absolute symlinks
+* fix playback discontinuation after seeking
+* input:
+  - curl: limit the receive buffer size
+  - curl: implement a hard-coded timeout of 10 seconds
+* decoder:
+  - ffmpeg: workaround for semantic API change in recent ffmpeg versions
+  - flac: validate the sample rate when scanning the tag
+  - wavpack: obey all decoder commands, stop at CUE track border
+* encoder:
+  - vorbis: don't send end-of-stream on flush
+* output:
+  - alsa: fix SIGFPE when alsa announces a period size of 0
+  - httpd: don't warn on client disconnect
+  - osx: don't drain the buffer when closing
+  - pulse: fix deadlock when resuming the stream
+  - pulse: fix deadlock when the stream was suspended
+
+
+ver 0.16.3 (2011/06/04)
+* fix assertion failure in audio format mask parser
+* fix NULL pointer dereference in playlist parser
+* fix playlist files in base music directory
+* database: allow directories with just playlists
+* decoder:
+  - ffmpeg: support libavcodec 0.7
+
+
+ver 0.16.2 (2011/03/18)
+* configure.ac:
+  - fix bashism in tremor test
+* decoder:
+  - tremor: fix configure test
+  - gme: detect end of song
+* encoder:
+  - vorbis: reset the Ogg stream after flush
+* output:
+  - httpd: fix uninitialized variable
+  - httpd: include sys/socket.h
+  - oss: AFMT_S24_PACKED is little-endian
+  - oss: disable 24 bit playback on FreeBSD
+
+
+ver 0.16.1 (2011/01/09)
 * audio_check: fix parameter in prototype
 * add void casts to suppress "result unused" warnings (clang)
 * input:
@@ -128,9 +176,20 @@ ver 0.16 (2010/12/11)
 * make single mode 'sticky'
 
 
-ver 0.15.16 (2010/??/??)
+ver 0.15.17 (2011/??/??)
+* encoder:
+  - vorbis: reset the Ogg stream after flush
+* decoders:
+  - vorbis: fix tremor support
+
+
+ver 0.15.16 (2011/03/13)
+* output:
+  - ao: initialize the ao_sample_format struct
+  - jack: fix crash with mono playback
 * encoders:
   - lame: explicitly configure the output sample rate
+* update: log all file permission problems
 
 
 ver 0.15.15 (2010/11/08)

configure.ac

@@ -1,5 +1,5 @@
 AC_PREREQ(2.60)
-AC_INIT(mpd, 0.16.1, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.16.4, musicpd-dev-team@lists.sourceforge.net)
 AC_CONFIG_SRCDIR([src/main.c])
 AM_INIT_AUTOMAKE([foreign 1.10 dist-bzip2 subdir-objects])
 AM_CONFIG_HEADER(config.h)
@@ -171,9 +171,9 @@ AC_ARG_ENABLE(fifo,
 	enable_fifo=yes)
 
 AC_ARG_ENABLE(flac,
-	AS_HELP_STRING([--disable-flac],
-		[disable flac support (default: enable)]),,
-	enable_flac=yes)
+	AS_HELP_STRING([--enable-flac],
+		[enable FLAC decoder]),,
+	enable_flac=auto)
 
 AC_ARG_ENABLE(fluidsynth,
 	AS_HELP_STRING([--enable-fluidsynth],
@@ -196,9 +196,9 @@ AC_ARG_ENABLE(httpd-output,
 	[enable_httpd_output=auto])
 
 AC_ARG_ENABLE(id3,
-	AS_HELP_STRING([--disable-id3],
-		[disable id3 support (default: enable)]),,
-	enable_id3=yes)
+	AS_HELP_STRING([--enable-id3],
+		[disable id3 support]),,
+	enable_id3=auto)
 
 AC_ARG_ENABLE(inotify,
 	AS_HELP_STRING([--disable-inotify],
@@ -353,9 +353,9 @@ AC_ARG_ENABLE(un,
 	[enable_un=yes])
 
 AC_ARG_ENABLE(vorbis,
-	AS_HELP_STRING([--disable-vorbis],
-		[disable Ogg Vorbis support (default: enable)]),,
-	enable_vorbis=yes)
+	AS_HELP_STRING([--enable-vorbis],
+		[enable Ogg Vorbis decoder]),,
+	enable_vorbis=auto)
 
 AC_ARG_ENABLE(vorbis-encoder,
 	AS_HELP_STRING([--enable-vorbis-encoder],
@@ -510,13 +510,8 @@ fi
 AM_CONDITIONAL(HAVE_CUE, test x$enable_cue = xyes)
 
 dnl -------------------------------- libid3tag --------------------------------
-if test x$enable_id3 = xyes; then
-	PKG_CHECK_MODULES([ID3TAG], [id3tag],,
-		AC_CHECK_LIB(id3tag, id3_file_open,
-			[ID3TAG_LIBS="-lid3tag -lz" ID3TAG_CFLAGS=""],
-			enable_id3=no))
-fi
-
+MPD_AUTO_PKG_LIB(id3, ID3TAG, id3tag, id3tag, id3_file_open, [-lid3tag -lz], [],
+	[id3tag], [libid3tag not found])
 if test x$enable_id3 = xyes; then
 	AC_DEFINE(HAVE_ID3TAG, 1, [Define to use id3tag])
 fi
@@ -530,36 +525,39 @@ dnl ---------------------------------------------------------------------------
 dnl --------------------------------- zeroconf --------------------------------
 
 case $with_zeroconf in
-no|avahi|bonjour)
+no|bonjour)
+	enable_avahi=no
 	;;
+
+avahi)
+	enable_avahi=yes
+	;;
+
 *)
 	with_zeroconf=auto
+	enable_avahi=auto
 	;;
 esac
 
+MPD_AUTO_PKG(avahi, AVAHI, [avahi-client avahi-glib],
+	[avahi client library], [avahi client+glib not found])
+if test x$enable_avahi = xyes; then
+	AC_DEFINE([HAVE_AVAHI], 1, [Define to enable Avahi Zeroconf support])
+	with_zeroconf=avahi
+fi
+
+AM_CONDITIONAL(HAVE_AVAHI, test x$enable_avahi = xyes)
+
+enable_bounjour=no
 if test x$with_zeroconf != xno; then
-	if test x$with_zeroconf = xavahi || test x$with_zeroconf = xauto; then
-		PKG_CHECK_MODULES([AVAHI], [avahi-client avahi-glib],
-			 [found_avahi=1;AC_DEFINE([HAVE_AVAHI], 1, [Define to enable Avahi Zeroconf support])]
-			 MPD_LIBS="$MPD_LIBS $AVAHI_LIBS" MPD_CFLAGS="$MPD_CFLAGS $AVAHI_CFLAGS",
-			 [found_avahi=0])
-	fi
-
-	if test x$found_avahi = x1; then
-		with_zeroconf=avahi
-	elif test x$with_zeroconf = xavahi; then
-		AC_MSG_ERROR([Avahi support requested but not found])
-	fi
-
 	if test x$with_zeroconf = xbonjour || test x$with_zeroconf = xauto; then
 		AC_CHECK_HEADER(dns_sd.h,
-			[found_bonjour=1;AC_DEFINE([HAVE_BONJOUR], 1, [Define to enable Bonjour Zeroconf support])],
-			[found_bonjour=0])
+			[enable_bonjour=yes;AC_DEFINE([HAVE_BONJOUR], 1, [Define to enable Bonjour Zeroconf support])])
 		AC_CHECK_LIB(dns_sd, DNSServiceRegister,
 			MPD_LIBS="$MPD_LIBS -ldns_sd")
 	fi
 
-	if test x$found_bonjour = x1; then
+	if test x$enable_bonjour = xyes; then
 		with_zeroconf=bonjour
 	elif test x$with_zeroconf = xbonjour; then
 		AC_MSG_ERROR([Bonjour support requested but not found])
@@ -574,7 +572,6 @@ if test x$with_zeroconf != xno; then
 fi
 
 AM_CONDITIONAL(HAVE_ZEROCONF, test x$with_zeroconf != xno)
-AM_CONDITIONAL(HAVE_AVAHI, test x$with_zeroconf = xavahi)
 AM_CONDITIONAL(HAVE_BONJOUR, test x$with_zeroconf = xbonjour)
 
 dnl ---------------------------------------------------------------------------
@@ -634,11 +631,6 @@ if test x$enable_lastfm = xyes; then
 fi
 AM_CONDITIONAL(ENABLE_LASTFM, test x$enable_lastfm = xyes)
 
-dnl ---------------------------------- libogg ---------------------------------
-if test x$with_tremor == xno || test -z $with_tremor; then
-	PKG_CHECK_MODULES(OGG, [ogg], enable_ogg=yes, enable_ogg=no)
-fi
-
 dnl ---------------------------------- libmms ---------------------------------
 MPD_AUTO_PKG(mms, MMS, [libmms >= 0.4],
 	[libmms mms:// protocol support], [libmms not found])
@@ -753,10 +745,12 @@ fi
 AM_CONDITIONAL(HAVE_FFMPEG, test x$enable_ffmpeg = xyes)
 
 dnl ----------------------------------- FLAC ----------------------------------
+
+MPD_AUTO_PKG(flac, FLAC, [flac >= 1.1],
+	[FLAC decoder], [libFLAC not found])
+
 if test x$enable_flac = xyes; then
-	PKG_CHECK_MODULES(FLAC, [flac >= 1.1],
-		AC_DEFINE(HAVE_FLAC, 1, [Define for FLAC support]),
-		enable_flac=no)
+	AC_DEFINE(HAVE_FLAC, 1, [Define for FLAC support])
 
 	oldcflags="$CFLAGS"
 	oldlibs="$LIBS"
@@ -771,12 +765,10 @@ if test x$enable_flac = xyes; then
 	LIBS="$oldlibs"
 
 	if test x$enable_oggflac = xflac; then
-		if test x$enable_ogg = xyes; then
-			FLAC_LIBS="${FLAC_LIBS} -logg"
-		else
-			enable_oggflac=yes
-			AC_MSG_WARN("FLAC has the ogg API built in, but couldn't find ogg. Disabling oggflac.")
-		fi
+		PKG_CHECK_MODULES(OGG, [ogg],
+			[FLAC_LIBS="${FLAC_LIBS} ${OGG_LIBS}" FLAC_CFLAGS="${FLAC_CFLAGS} ${OGG_CFLAGS}"],
+			[enable_oggflac=yes;
+			AC_MSG_WARN("FLAC has the ogg API built in, but couldn't find ogg. Disabling oggflac.")])
 	fi
 fi
 
@@ -802,7 +794,8 @@ if test x$enable_gme = xyes; then
 fi
 
 dnl ---------------------------------- libmad ---------------------------------
-MPD_AUTO_PKG(mad, MAD, [mad],
+MPD_AUTO_PKG_LIB(mad, MAD, [mad],
+	mad, mad_stream_init, [-lmad], [],
 	[libmad MP3 decoder plugin], [libmad not found])
 if test x$enable_mad = xyes; then
 	AC_DEFINE(HAVE_MAD, 1, [Define to use libmad])
@@ -934,13 +927,19 @@ if test x$enable_tremor = xyes; then
 	ac_save_LIBS="$LIBS"
 	CFLAGS="$CFLAGS $TREMOR_CFLAGS"
 	LIBS="$LIBS $TREMOR_LIBS"
-	AC_CHECK_LIB(vorbisidec,ov_read,enable_vorbis=yes,enable_vorbis=no;
+	AC_CHECK_LIB(vorbisidec,ov_read,,enable_tremor=no;
 		AC_MSG_WARN([vorbisidec lib needed for ogg support with tremor -- disabling ogg support]))
 	CFLAGS="$ac_save_CFLAGS"
 	LIBS="$ac_save_LIBS"
+fi
 
+if test x$enable_tremor = xyes; then
 	AC_DEFINE(HAVE_TREMOR,1,
 		[Define to use tremor (libvorbisidec) for ogg support])
+	AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support])
+else
+	TREMOR_CFLAGS=
+	TREMOR_LIBS=
 fi
 
 AC_SUBST(TREMOR_CFLAGS)
@@ -966,21 +965,21 @@ fi
 AM_CONDITIONAL(HAVE_OGGFLAC, test x$enable_oggflac = xyes)
 
 dnl -------------------------------- Ogg Vorbis -------------------------------
-if test x$enable_vorbis = xyes; then
-	if test x$enable_tremor = xyes; then
+
+if test x$enable_tremor = xyes; then
+	if test x$enable_vorbis = xyes; then
 		AC_MSG_WARN(["OggTremor detected, could not enable Vorbis."])
-		enable_vorbis=no
-	elif test x$enable_ogg = xyes; then
-		PKG_CHECK_MODULES(VORBIS, [vorbis vorbisfile],
-			AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support]),
-			enable_vorbis=no)
-	else
-		AC_MSG_WARN(["Ogg not detected, could not enable Vorbis."])
-		enable_vorbis=no
 	fi
+	enable_vorbis=no
 fi
 
-AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes)
+MPD_AUTO_PKG(vorbis, VORBIS, [vorbis vorbisfile ogg],
+	[Ogg Vorbis decoder], [libvorbis not found])
+if test x$enable_vorbis = xyes; then
+	AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support])
+fi
+
+AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes || test x$enable_tremor = xyes)
 
 dnl --------------------------------- sidplay ---------------------------------
 found_sidplay=$HAVE_CXX
@@ -1075,7 +1074,7 @@ if
 fi
 
 AM_CONDITIONAL(HAVE_OGG_COMMON,
-	  test x$enable_vorbis = xyes || test x$enable_oggflac = xyes || test x$enable_flac = xyes)
+	test x$enable_vorbis = xyes || test x$enable_tremor = xyes || test x$enable_oggflac = xyes || test x$enable_flac = xyes)
 
 AM_CONDITIONAL(HAVE_FLAC_COMMON,
 	  test x$enable_flac = xyes || test x$enable_oggflac = xyes)

doc/developer.xml

@@ -57,7 +57,7 @@
           Some example code:
         </para>
 
-        <programlisting lang="C">static inline bool
+        <programlisting lang="C">static inline int
 foo(const char *abc, int xyz)
 {
         if (abc == NULL) {

doc/protocol.xml

@@ -8,18 +8,54 @@
     <title>General protocol syntax</title>
 
     <section>
-      <title>Requests</title>
+      <title>Protocol overview</title>
 
       <para>
-        If arguments contain spaces, they should be surrounded by double quotation
-        marks.
+        The MPD command protocol exchanges line-based text records
+        between client and server over TCP.  Once the client is
+        connected to the server, they conduct a conversation until the
+        client closes the connection. The conversation flow is always
+        initiated by the client.
       </para>
 
+      <para>
+        The client transmits a command sequence, terminated by the
+        newline character <constant>\n</constant>.  The server will
+        respond with one or more lines, the last of which will be a
+        completion code.
+      </para>
+
+      <para>
+        When the client connects to the server, the server will answer
+        with the following line:
+
+        <synopsis>OK MPD version</synopsis>
+
+        where <varname>version</varname> is a version identifier such as
+        0.12.2.  This version identifier is the version of the protocol
+        spoken, not the real version of the daemon.  (There is no way to
+        retrieve this real version identifier from the connection.)
+      </para>
+    </section>
+
+    <section>
+      <title>Requests</title>
+
       <cmdsynopsis>
         <command>COMMAND</command>
         <arg rep="repeat"><replaceable>ARG</replaceable></arg>
       </cmdsynopsis>
 
+      <para>
+        If arguments contain spaces, they should be surrounded by double
+        quotation marks.
+      </para>
+
+      <para>
+        Argument strings are separated from the command and any other
+        arguments by linear white-space (' ' or '\t').
+      </para>
+
       <para>
         All data between the client and the server is encoded in
         UTF-8. (Note: In UTF-8 all standard ansi characters, 0-127 are
@@ -38,13 +74,97 @@
       <title>Responses</title>
 
       <para>
-        A command returns <returnvalue>OK</returnvalue> on completion
-        or <returnvalue>ACK some error</returnvalue> on failure.
-        These denote the end of command execution.
+        A command returns <returnvalue>OK</returnvalue> on completion or
+        <returnvalue>ACK some error</returnvalue> on failure.  These
+        denote the end of command execution.
       </para>
+
+      <section>
+        <title>Failure responses</title>
+
+        <para>
+          The nature of the error can be gleaned from the information
+          that follows the <returnvalue>ACK</returnvalue>.
+          <returnvalue>ACK</returnvalue> lines are of the form:
+
+          <synopsis>ACK [error@command_listNum] {current_command} message_text\n</synopsis>
+
+          These responses are generated by a call to
+          <function>commandError</function>. They contain four separate
+          terms. Let's look at each of them:
+
+          <itemizedlist>
+            <listitem>
+              <para>
+                <returnvalue>error</returnvalue>: numeric value of one
+                of the <constant>ACK_ERROR</constant> constants defined
+                in <filename>ack.h</filename>.
+              </para>
+            </listitem>
+            <listitem>
+              <para>
+                <returnvalue>command_listNum</returnvalue>:
+                offset of the command that caused the error in a <link
+                  linkend="command_lists">Command List</link>.
+                An error will always cause a command list to terminate
+                at the command that causes the error.
+              </para>
+            </listitem>
+            <listitem>
+              <para>
+                <returnvalue>current_command</returnvalue>:
+                name of the command, in a <link
+                  linkend="command_lists">Command List</link>,
+                that was executing when the error occurred.
+              </para>
+            </listitem>
+            <listitem>
+              <para>
+                <returnvalue>message_text</returnvalue>:
+                some (hopefully) informative text that describes the
+                nature of the error.
+              </para>
+            </listitem>
+          </itemizedlist>
+        </para>
+
+        <example>
+          <title>foo</title>
+          <para>
+            An example might help. Consider the following sequence
+            sent from the client to the server.
+
+            <synopsis>
+              command_list_begin
+              volume 86
+              play 10240
+              status
+              command_list_end
+            </synopsis>
+          </para>
+
+          <para>
+            The server responds with:
+
+            <returnvalue>
+              ACK [50@1] {play} song doesn't exist: "10240"
+            </returnvalue>
+          </para>
+
+          <para>
+            This tells us that the play command, which was the
+            second in the list (the first or only command is
+            numbered 0), failed with error 50.  The number 50
+            translates to <constant>ACK_ERROR_NO_EXIST</constant>--the
+            song doesn't exist.  This is reiterated by the message text
+            which also tells us which song doesn't exist.
+          </para>
+
+        </example>
+      </section>
     </section>
 
-    <section>
+    <section id="command_lists">
       <title>Command lists</title>
 
       <para>

doc/user.xml

@@ -446,7 +446,7 @@ cd mpd-version</programlisting>
       </para>
 
       <para>
-        To configure a filter, add a
+        To configure a playlist plugin, add a
         <varname>playlist_plugin</varname> block to
         <filename>mpd.conf</filename>:
       </para>

m4/mpd_auto.m4

@@ -63,3 +63,18 @@ AC_DEFUN([MPD_AUTO_PKG], [
 
 	MPD_AUTO_RESULT([$1], [$4], [$5])
 ])
+
+dnl Check with pkg-config first, fall back to AC_CHECK_LIB.
+dnl
+dnl Parameters: varname1, varname2, pkgname, libname, symname, libs, cflags, description, errmsg
+AC_DEFUN([MPD_AUTO_PKG_LIB], [
+	if eval "test x`echo '$'enable_$1` != xno"; then
+		PKG_CHECK_MODULES([$2], [$3],
+			[eval "found_$1=yes"],
+			AC_CHECK_LIB($4, $5,
+				[eval "found_$1=yes $2_LIBS='$6' $2_CFLAGS='$7'"],
+				[eval "found_$1=no"]))
+	fi
+
+	MPD_AUTO_RESULT([$1], [$8], [$9])
+])

scripts/makedist.sh

@@ -3,7 +3,7 @@ PWD=`pwd`
 
 ## If we're not in the scripts directory
 ## assume the base directory.
-if test "`basename $PWD`" == "scripts"; then
+if test "`basename $PWD`" = "scripts"; then
 	cd ../
 else
 	MYOLDPWD=`pwd`
@@ -18,7 +18,7 @@ fi
 make
 make dist
 
-if test "`basename $PWD`" == "scripts"; then
+if test "`basename $PWD`" = "scripts"; then
 	cd contrib/
 else
 	cd $MYOLDPWD

src/AudioCompress/compress.c

@@ -16,16 +16,16 @@
 struct Compressor {
         //! The compressor's preferences
         struct CompressorConfig prefs;
-        
+
         //! History of the peak values
         int *peaks;
-                
+
         //! History of the gain values
         int *gain;
-        
+
         //! History of clip amounts
         int *clipped;
-        
+
         unsigned int pos;
         unsigned int bufsz;
 };
@@ -41,9 +41,9 @@ struct Compressor *Compressor_new(unsigned int history)
         obj->peaks = obj->gain = obj->clipped = NULL;
 	obj->bufsz = 0;
         obj->pos = 0;
-        
+
         Compressor_setHistory(obj, history);
-        
+
         return obj;
 }
 
@@ -70,7 +70,7 @@ void Compressor_setHistory(struct Compressor *obj, unsigned int history)
 {
 	if (!history)
                 history = BUCKETS;
-        
+
         obj->peaks = resizeArray(obj->peaks, history, obj->bufsz);
         obj->gain = resizeArray(obj->gain, history, obj->bufsz);
         obj->clipped = resizeArray(obj->clipped, history, obj->bufsz);
@@ -82,7 +82,7 @@ struct CompressorConfig *Compressor_getConfig(struct Compressor *obj)
         return &obj->prefs;
 }
 
-void Compressor_Process_int16(struct Compressor *obj, int16_t *audio, 
+void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
                               unsigned int count)
 {
         struct CompressorConfig *prefs = Compressor_getConfig(obj);
@@ -97,7 +97,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
         int *clipped = obj->clipped + slot;
         unsigned int ramp = count;
         int delta;
-        
+
 	ap = audio;
 	for (i = 0; i < count; i++)
 	{
@@ -124,15 +124,15 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
 
 	//! Determine target gain
 	newGain = (1 << 10)*prefs->target/peakVal;
-        
+
         //! Adjust the gain with inertia from the previous gain value
-        newGain = (curGain*((1 << prefs->smooth) - 1) + newGain) 
+        newGain = (curGain*((1 << prefs->smooth) - 1) + newGain)
                 >> prefs->smooth;
-        
+
         //! Make sure it's no more than the maximum gain value
         if (newGain > (prefs->maxgain << 10))
                 newGain = prefs->maxgain << 10;
-        
+
         //! Make sure it's no less than 1:1
 	if (newGain < (1 << 10))
 		newGain = 1 << 10;
@@ -144,7 +144,7 @@ void Compressor_Process_int16(struct Compressor *obj, int16_t *audio,
                 //! Truncate the ramp time
                 ramp = peakPos;
         }
-        
+
         //! Record the new gain
         obj->gain[slot] = newGain;
 

src/ape.c

@@ -60,8 +60,10 @@ ape_scan_internal(FILE *fp, tag_ape_callback_t callback, void *ctx)
 	assert(remaining > 10);
 
 	char *buffer = g_malloc(remaining);
-	if (fread(buffer, 1, remaining, fp) != remaining)
+	if (fread(buffer, 1, remaining, fp) != remaining) {
+		g_free(buffer);
 		return false;
+	}
 
 	/* read tags */
 	unsigned n = GUINT32_FROM_LE(footer.count);

src/audio_format.h

@@ -22,6 +22,7 @@
 
 #include <stdint.h>
 #include <stdbool.h>
+#include <assert.h>
 
 enum sample_format {
 	SAMPLE_FORMAT_UNDEFINED = 0,
@@ -219,6 +220,9 @@ static inline void
 audio_format_mask_apply(struct audio_format *af,
 			const struct audio_format *mask)
 {
+	assert(audio_format_valid(af));
+	assert(audio_format_mask_valid(mask));
+
 	if (mask->sample_rate != 0)
 		af->sample_rate = mask->sample_rate;
 
@@ -227,6 +231,8 @@ audio_format_mask_apply(struct audio_format *af,
 
 	if (mask->channels != 0)
 		af->channels = mask->channels;
+
+	assert(audio_format_valid(af));
 }
 
 /**

src/audio_parser.c

@@ -192,6 +192,8 @@ audio_format_parse(struct audio_format *dest, const char *src,
 	}
 
 	audio_format_init(dest, rate, sample_format, channels);
+	assert(mask ? audio_format_mask_valid(dest)
+	       : audio_format_valid(dest));
 
 	return true;
 }

src/command.c

@@ -464,7 +464,7 @@ handle_currentsong(struct client *client,
 		   G_GNUC_UNUSED int argc, G_GNUC_UNUSED char *argv[])
 {
 	playlist_print_current(client, &g_playlist);
-	return PLAYLIST_RESULT_SUCCESS;
+	return COMMAND_RETURN_OK;
 }
 
 static enum command_return
@@ -749,7 +749,7 @@ handle_load(struct client *client, G_GNUC_UNUSED int argc, char *argv[])
 
 	result = playlist_open_into_queue(argv[1], &g_playlist);
 	if (result != PLAYLIST_RESULT_NO_SUCH_LIST)
-		return result;
+		return print_playlist_result(client, result);
 
 	result = playlist_load_spl(&g_playlist, argv[1]);
 	return print_playlist_result(client, result);

src/conf.c

@@ -367,6 +367,7 @@ config_read_file(const char *file, GError **error_r)
 			assert(*line != 0);
 			g_propagate_prefixed_error(error_r, error,
 						   "line %i: ", count);
+			fclose(fp);
 			return false;
 		}
 
@@ -378,6 +379,7 @@ config_read_file(const char *file, GError **error_r)
 			g_set_error(error_r, config_quark(), 0,
 				    "unrecognized parameter in config file at "
 				    "line %i: %s\n", count, name);
+			fclose(fp);
 			return false;
 		}
 
@@ -387,6 +389,7 @@ config_read_file(const char *file, GError **error_r)
 				    "config parameter \"%s\" is first defined "
 				    "on line %i and redefined on line %i\n",
 				    name, param->line, count);
+			fclose(fp);
 			return false;
 		}
 
@@ -398,6 +401,7 @@ config_read_file(const char *file, GError **error_r)
 			if (*line != '{') {
 				g_set_error(error_r, config_quark(), 0,
 					    "line %i: '{' expected", count);
+				fclose(fp);
 				return false;
 			}
 
@@ -406,12 +410,15 @@ config_read_file(const char *file, GError **error_r)
 				g_set_error(error_r, config_quark(), 0,
 					    "line %i: Unknown tokens after '{'",
 					    count);
+				fclose(fp);
 				return false;
 			}
 
 			param = config_read_block(fp, &count, string, error_r);
-			if (param == NULL)
+			if (param == NULL) {
+				fclose(fp);
 				return false;
+			}
 		} else {
 			/* a string value */
 
@@ -428,6 +435,7 @@ config_read_file(const char *file, GError **error_r)
 					g_error_free(error);
 				}
 
+				fclose(fp);
 				return false;
 			}
 
@@ -435,6 +443,7 @@ config_read_file(const char *file, GError **error_r)
 				g_set_error(error_r, config_quark(), 0,
 					    "line %i: Unknown tokens after value",
 					    count);
+				fclose(fp);
 				return false;
 			}
 

src/database.c

@@ -180,7 +180,7 @@ db_check(void)
 		}
 
 		/* Check if we can write to the directory */
-		if (access(dirPath, R_OK | W_OK)) {
+		if (access(dirPath, X_OK | W_OK)) {
 			g_warning("Can't create db file in \"%s\": %s",
 				  dirPath, strerror(errno));
 			g_free(dirPath);

src/decoder/audiofile_decoder_plugin.c

@@ -244,7 +244,7 @@ static const char *const audiofile_suffixes[] = {
 static const char *const audiofile_mime_types[] = {
 	"audio/x-wav",
 	"audio/x-aiff",
-	NULL 
+	NULL
 };
 
 const struct decoder_plugin audiofile_decoder_plugin = {

src/decoder/ffmpeg_decoder_plugin.c

@@ -81,11 +81,19 @@ mpd_ffmpeg_log_callback(G_GNUC_UNUSED void *ptr, int level,
 
 #endif /* !OLD_FFMPEG_INCLUDES */
 
+#ifndef AV_VERSION_INT
+#define AV_VERSION_INT(a, b, c) (a<<16 | b<<8 | c)
+#endif
+
 struct mpd_ffmpeg_stream {
 	struct decoder *decoder;
 	struct input_stream *input;
 
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0)
+	AVIOContext *io;
+#else
 	ByteIOContext *io;
+#endif
 	unsigned char buffer[8192];
 };
 
@@ -102,13 +110,11 @@ static int64_t
 mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence)
 {
 	struct mpd_ffmpeg_stream *stream = opaque;
-	bool ret;
 
 	if (whence == AVSEEK_SIZE)
 		return stream->input->size;
 
-	ret = input_stream_seek(stream->input, pos, whence, NULL);
-	if (!ret)
+	if (!input_stream_seek(stream->input, pos, whence, NULL))
 		return -1;
 
 	return stream->input->offset;
@@ -133,6 +139,33 @@ mpd_ffmpeg_stream_open(struct decoder *decoder, struct input_stream *input)
 	return stream;
 }
 
+/**
+ * API compatibility wrapper for av_open_input_stream() and
+ * avformat_open_input().
+ */
+static int
+mpd_ffmpeg_open_input(AVFormatContext **ic_ptr,
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(52,101,0)
+		      AVIOContext *pb,
+#else
+		      ByteIOContext *pb,
+#endif
+		      const char *filename,
+		      AVInputFormat *fmt)
+{
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,1,3)
+	AVFormatContext *context = avformat_alloc_context();
+	if (context == NULL)
+		return AVERROR(ENOMEM);
+
+	context->pb = pb;
+	*ic_ptr = context;
+	return avformat_open_input(ic_ptr, filename, fmt, NULL);
+#else
+	return av_open_input_stream(ic_ptr, pb, filename, fmt, NULL);
+#endif
+}
+
 static void
 mpd_ffmpeg_stream_close(struct mpd_ffmpeg_stream *stream)
 {
@@ -156,7 +189,11 @@ ffmpeg_find_audio_stream(const AVFormatContext *format_context)
 {
 	for (unsigned i = 0; i < format_context->nb_streams; ++i)
 		if (format_context->streams[i]->codec->codec_type ==
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 64, 0)
+		    AVMEDIA_TYPE_AUDIO)
+#else
 		    CODEC_TYPE_AUDIO)
+#endif
 			return i;
 
 	return -1;
@@ -183,30 +220,40 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 		   AVCodecContext *codec_context,
 		   const AVRational *time_base)
 {
-	enum decoder_command cmd = DECODE_COMMAND_NONE;
-	uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
-	int16_t *aligned_buffer;
-	size_t buffer_size;
-	int len, audio_size;
-	uint8_t *packet_data;
-	int packet_size;
-
 	if (packet->pts != (int64_t)AV_NOPTS_VALUE)
 		decoder_timestamp(decoder,
 				  av_rescale_q(packet->pts, *time_base,
 					       (AVRational){1, 1}));
 
-	packet_data = packet->data;
-	packet_size = packet->size;
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
+	AVPacket packet2 = *packet;
+#else
+	const uint8_t *packet_data = packet->data;
+	int packet_size = packet->size;
+#endif
 
-	buffer_size = sizeof(audio_buf);
-	aligned_buffer = align16(audio_buf, &buffer_size);
+	uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2 + 16];
+	size_t buffer_size = sizeof(audio_buf);
+	int16_t *aligned_buffer = align16(audio_buf, &buffer_size);
 
-	while ((packet_size > 0) && (cmd == DECODE_COMMAND_NONE)) {
-		audio_size = buffer_size;
-		len = avcodec_decode_audio2(codec_context,
-					    aligned_buffer, &audio_size,
-					    packet_data, packet_size);
+	enum decoder_command cmd = DECODE_COMMAND_NONE;
+	while (
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
+	       packet2.size > 0 &&
+#else
+	       packet_size > 0 &&
+#endif
+	       cmd == DECODE_COMMAND_NONE) {
+		int audio_size = buffer_size;
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
+		int len = avcodec_decode_audio3(codec_context,
+						aligned_buffer, &audio_size,
+						&packet2);
+#else
+		int len = avcodec_decode_audio2(codec_context,
+						aligned_buffer, &audio_size,
+						packet_data, packet_size);
+#endif
 
 		if (len < 0) {
 			/* if error, we skip the frame */
@@ -214,8 +261,13 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 			break;
 		}
 
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52,25,0)
+		packet2.data += len;
+		packet2.size -= len;
+#else
 		packet_data += len;
 		packet_size -= len;
+#endif
 
 		if (audio_size <= 0)
 			continue;
@@ -230,7 +282,7 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 static enum sample_format
 ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
 {
-#if LIBAVCODEC_VERSION_INT >= ((51<<16)+(41<<8)+0)
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(51, 41, 0)
 	switch (codec_context->sample_fmt) {
 	case SAMPLE_FMT_S16:
 		return SAMPLE_FORMAT_S16;
@@ -299,14 +351,10 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 		return;
 	}
 
-	AVFormatContext *format_context;
-	AVCodecContext *codec_context;
-	AVCodec *codec;
-	int audio_stream;
-
 	//ffmpeg works with ours "fileops" helper
-	if (av_open_input_stream(&format_context, stream->io, input->uri,
-				 input_format, NULL) != 0) {
+	AVFormatContext *format_context = NULL;
+	if (mpd_ffmpeg_open_input(&format_context, stream->io, input->uri,
+				  input_format) != 0) {
 		g_warning("Open failed\n");
 		mpd_ffmpeg_stream_close(stream);
 		return;
@@ -319,7 +367,7 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 		return;
 	}
 
-	audio_stream = ffmpeg_find_audio_stream(format_context);
+	int audio_stream = ffmpeg_find_audio_stream(format_context);
 	if (audio_stream == -1) {
 		g_warning("No audio stream inside\n");
 		av_close_input_stream(format_context);
@@ -327,11 +375,12 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 		return;
 	}
 
-	codec_context = format_context->streams[audio_stream]->codec;
+	AVCodecContext *codec_context =
+		format_context->streams[audio_stream]->codec;
 	if (codec_context->codec_name[0] != 0)
 		g_debug("codec '%s'", codec_context->codec_name);
 
-	codec = avcodec_find_decoder(codec_context->codec_id);
+	AVCodec *codec = avcodec_find_decoder(codec_context->codec_id);
 
 	if (!codec) {
 		g_warning("Unsupported audio codec\n");
@@ -428,13 +477,26 @@ static const ffmpeg_tag_map ffmpeg_tag_maps[] = {
 };
 
 static bool
-ffmpeg_copy_metadata(struct tag *tag, AVMetadata *m,
+ffmpeg_copy_metadata(struct tag *tag,
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,1,0)
+		     AVDictionary *m,
+#else
+		     AVMetadata *m,
+#endif
 		     const ffmpeg_tag_map tag_map)
 {
+#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(53,1,0)
+	AVDictionaryEntry *mt = NULL;
+
+	while ((mt = av_dict_get(m, tag_map.name, mt, 0)) != NULL)
+		tag_add_item(tag, tag_map.type, mt->value);
+#else
 	AVMetadataTag *mt = NULL;
 
 	while ((mt = av_metadata_get(m, tag_map.name, mt, 0)) != NULL)
 		tag_add_item(tag, tag_map.type, mt->value);
+#endif
+
 	return mt != NULL;
 }
 
@@ -452,9 +514,9 @@ ffmpeg_stream_tag(struct input_stream *is)
 	if (stream == NULL)
 		return NULL;
 
-	AVFormatContext *f;
-	if (av_open_input_stream(&f, stream->io, is->uri,
-				 input_format, NULL) != 0) {
+	AVFormatContext *f = NULL;
+	if (mpd_ffmpeg_open_input(&f, stream->io, is->uri,
+				  input_format) != 0) {
 		mpd_ffmpeg_stream_close(stream);
 		return NULL;
 	}

src/decoder/flac_decoder_plugin.c

@@ -81,7 +81,7 @@ flac_tell_cb(G_GNUC_UNUSED const FLAC__StreamDecoder *fd,
 	struct flac_data *data = (struct flac_data *) fdata;
 
 	if (!data->input_stream->seekable)
-		return FLAC__STREAM_DECODER_SEEK_STATUS_UNSUPPORTED;
+		return FLAC__STREAM_DECODER_TELL_STATUS_UNSUPPORTED;
 
 	*offset = (long)(data->input_stream->offset);
 

src/decoder/flac_metadata.c

@@ -224,7 +224,8 @@ flac_tag_apply_metadata(struct tag *tag, const char *track,
 		break;
 
 	case FLAC__METADATA_TYPE_STREAMINFO:
-		tag->time = flac_duration(&block->data.stream_info);
+		if (block->data.stream_info.sample_rate > 0)
+			tag->time = flac_duration(&block->data.stream_info);
 		break;
 
 	default:

src/decoder/flac_metadata.h

@@ -20,6 +20,7 @@
 #ifndef MPD_FLAC_METADATA_H
 #define MPD_FLAC_METADATA_H
 
+#include <assert.h>
 #include <stdbool.h>
 #include <FLAC/metadata.h>
 
@@ -29,6 +30,8 @@ struct replay_gain_info;
 static inline unsigned
 flac_duration(const FLAC__StreamMetadata_StreamInfo *stream_info)
 {
+	assert(stream_info->sample_rate > 0);
+
 	return (stream_info->total_samples + stream_info->sample_rate - 1) /
 		stream_info->sample_rate;
 }

src/decoder/gme_decoder_plugin.c

@@ -153,6 +153,9 @@ gme_file_decode(struct decoder *decoder, const char *path_fs)
 	if((gme_err = gme_start_track(emu, song_num)) != NULL)
 		g_warning("%s", gme_err);
 
+	if(ti->length > 0)
+		gme_set_fade(emu, ti->length);
+
 	/* play */
 	do {
 		gme_err = gme_play(emu, GME_BUFFER_SAMPLES, buf);

src/decoder/mpcdec_decoder_plugin.c

@@ -153,7 +153,6 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 	mpc_uint32_t ret;
 	int32_t chunk[G_N_ELEMENTS(sample_buffer)];
 	long bit_rate = 0;
-	mpc_uint32_t vbr_update_acc;
 	mpc_uint32_t vbr_update_bits;
 	enum decoder_command cmd = DECODE_COMMAND_NONE;
 
@@ -243,10 +242,11 @@ mpcdec_decode(struct decoder *mpd_decoder, struct input_stream *is)
 				decoder_seek_error(mpd_decoder);
 		}
 
-		vbr_update_acc = 0;
 		vbr_update_bits = 0;
 
 #ifdef MPC_IS_OLD_API
+		mpc_uint32_t vbr_update_acc = 0;
+
 		ret = mpc_decoder_decode(&decoder, sample_buffer,
 					 &vbr_update_acc, &vbr_update_bits);
 		if (ret == 0 || ret == (mpc_uint32_t)-1)

src/decoder/wavpack_decoder_plugin.c

@@ -34,9 +34,6 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "wavpack"
 
-/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
-#define CHUNK_SIZE		1020
-
 #define ERRORLEN 80
 
 static struct {
@@ -162,8 +159,6 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
 	enum sample_format sample_format;
 	struct audio_format audio_format;
 	format_samples_t format_samples;
-	char chunk[CHUNK_SIZE];
-	int samples_requested, samples_got;
 	float total_time;
 	int bytes_per_sample, output_sample_size;
 
@@ -193,12 +188,15 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
 	output_sample_size = audio_format_frame_size(&audio_format);
 
 	/* wavpack gives us all kind of samples in a 32-bit space */
-	samples_requested = sizeof(chunk) / (4 * audio_format.channels);
+	int32_t chunk[1024];
+	const uint32_t samples_requested = G_N_ELEMENTS(chunk) /
+		audio_format.channels;
 
 	decoder_initialized(decoder, &audio_format, can_seek, total_time);
 
-	do {
-		if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
+	enum decoder_command cmd = decoder_get_command(decoder);
+	while (cmd != DECODE_COMMAND_STOP) {
+		if (cmd == DECODE_COMMAND_SEEK) {
 			if (can_seek) {
 				unsigned where = decoder_seek_where(decoder) *
 					audio_format.sample_rate;
@@ -213,29 +211,20 @@ wavpack_decode(struct decoder *decoder, WavpackContext *wpc, bool can_seek)
 			}
 		}
 
-		if (decoder_get_command(decoder) == DECODE_COMMAND_STOP) {
+		uint32_t samples_got = WavpackUnpackSamples(wpc, chunk,
+							    samples_requested);
+		if (samples_got == 0)
 			break;
-		}
 
-		samples_got = WavpackUnpackSamples(
-			wpc, (int32_t *)chunk, samples_requested
-		);
-		if (samples_got > 0) {
-			int bitrate = (int)(WavpackGetInstantBitrate(wpc) /
-			              1000 + 0.5);
+		int bitrate = (int)(WavpackGetInstantBitrate(wpc) / 1000 +
+				    0.5);
+		format_samples(bytes_per_sample, chunk,
+			       samples_got * audio_format.channels);
 
-			format_samples(
-				bytes_per_sample, chunk,
-				samples_got * audio_format.channels
-			);
-
-			decoder_data(
-				decoder, NULL, chunk,
-				samples_got * output_sample_size,
-				bitrate
-			);
-		}
-	} while (samples_got > 0);
+		cmd = decoder_data(decoder, NULL, chunk,
+				   samples_got * output_sample_size,
+				   bitrate);
+	}
 }
 
 /**

src/directory.h

@@ -62,7 +62,8 @@ directory_free(struct directory *directory);
 static inline bool
 directory_is_empty(const struct directory *directory)
 {
-	return directory->children.nr == 0 && directory->songs.nr == 0;
+	return directory->children.nr == 0 && directory->songs.nr == 0 &&
+		playlist_vector_is_empty(&directory->playlists);
 }
 
 static inline const char *

src/encoder/flac_encoder.c

@@ -55,7 +55,7 @@ static bool
 flac_encoder_configure(struct flac_encoder *encoder,
 		const struct config_param *param, G_GNUC_UNUSED GError **error)
 {
-	encoder->compression = config_get_block_unsigned(param, 
+	encoder->compression = config_get_block_unsigned(param,
 						"compression", 5);
 
 	return true;
@@ -218,7 +218,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 
 		if (init_status != FLAC__STREAM_ENCODER_OK) {
 			g_set_error(error, flac_encoder_quark(), 0,
-			    "failed to initialize encoder: %s\n", 
+			    "failed to initialize encoder: %s\n",
 			    FLAC__StreamEncoderStateString[init_status]);
 			flac_encoder_close(_encoder);
 			return false;
@@ -234,7 +234,7 @@ flac_encoder_open(struct encoder *_encoder, struct audio_format *audio_format,
 
 		if(init_status != FLAC__STREAM_ENCODER_INIT_STATUS_OK) {
 			g_set_error(error, flac_encoder_quark(), 0,
-			    "failed to initialize encoder: %s\n", 
+			    "failed to initialize encoder: %s\n",
 			    FLAC__StreamEncoderInitStatusString[init_status]);
 			flac_encoder_close(_encoder);
 			return false;

src/encoder/vorbis_encoder.c

@@ -266,6 +266,15 @@ vorbis_encoder_flush(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
 
+	encoder->flush = true;
+	return true;
+}
+
+static bool
+vorbis_encoder_pre_tag(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
+{
+	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
+
 	vorbis_analysis_wrote(&encoder->vd, 0);
 	vorbis_encoder_blockout(encoder);
 
@@ -276,6 +285,8 @@ vorbis_encoder_flush(struct encoder *_encoder, G_GNUC_UNUSED GError **error)
 	vorbis_analysis_init(&encoder->vd, &encoder->vi);
 	vorbis_block_init(&encoder->vd, &encoder->vb);
 
+	ogg_stream_reset(&encoder->os);
+
 	encoder->flush = true;
 	return true;
 }
@@ -364,6 +375,7 @@ vorbis_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
 	if (ret == 0 && encoder->flush) {
 		encoder->flush = false;
 		ret = ogg_stream_flush(&encoder->os, &page);
+
 	}
 
 	if (ret == 0)
@@ -396,6 +408,7 @@ const struct encoder_plugin vorbis_encoder_plugin = {
 	.open = vorbis_encoder_open,
 	.close = vorbis_encoder_close,
 	.flush = vorbis_encoder_flush,
+	.pre_tag = vorbis_encoder_pre_tag,
 	.tag = vorbis_encoder_tag,
 	.write = vorbis_encoder_write,
 	.read = vorbis_encoder_read,

src/encoder/wave_encoder.c

@@ -58,7 +58,7 @@ wave_encoder_quark(void)
 }
 
 static void
-fill_wave_header(struct wave_header *header, int channels, int bits, 
+fill_wave_header(struct wave_header *header, int channels, int bits,
 		int freq, int block_size)
 {
 	int data_size = 0x0FFFFFFF;
@@ -142,7 +142,7 @@ wave_encoder_open(struct encoder *_encoder,
 	buffer = pcm_buffer_get(&encoder->buffer, sizeof(struct wave_header) );
 
 	/* create PCM wave header in initial buffer */
-	fill_wave_header((struct wave_header *) buffer, 
+	fill_wave_header((struct wave_header *) buffer,
 			audio_format->channels,
 			 encoder->bits,
 			audio_format->sample_rate,

src/encoder_plugin.h

@@ -50,6 +50,8 @@ struct encoder_plugin {
 
 	bool (*flush)(struct encoder *encoder, GError **error);
 
+	bool (*pre_tag)(struct encoder *encoder, GError **error);
+
 	bool (*tag)(struct encoder *encoder, const struct tag *tag,
 		    GError **error);
 
@@ -147,9 +149,31 @@ encoder_flush(struct encoder *encoder, GError **error)
 		: true;
 }
 
+/**
+ * Prepare for sending a tag to the encoder.  This is used by some
+ * encoders to flush the previous sub-stream, in preparation to begin
+ * a new one.
+ *
+ * @param encoder the encoder
+ * @param tag the tag object
+ * @param error location to store the error occuring, or NULL to ignore errors.
+ * @return true on success
+ */
+static inline bool
+encoder_pre_tag(struct encoder *encoder, GError **error)
+{
+	/* this method is optional */
+	return encoder->plugin->pre_tag != NULL
+		? encoder->plugin->pre_tag(encoder, error)
+		: true;
+}
+
 /**
  * Sends a tag to the encoder.
  *
+ * Instructions: call encoder_pre_tag(); then obtain flushed data with
+ * encoder_read(); finally call encoder_tag().
+ *
  * @param encoder the encoder
  * @param tag the tag object
  * @param error location to store the error occuring, or NULL to ignore errors.

src/input/curl_input_plugin.c

@@ -42,6 +42,13 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "input_curl"
 
+/**
+ * Do not buffer more than this number of bytes.  It should be a
+ * reasonable limit that doesn't make low-end machines suffer too
+ * much, but doesn't cause stuttering on high-latency lines.
+ */
+static const size_t CURL_MAX_BUFFERED = 512 * 1024;
+
 /**
  * Buffers created by input_curl_writefunction().
  */
@@ -144,6 +151,25 @@ input_curl_finish(void)
 	curl_global_cleanup();
 }
 
+/**
+ * Determine the total sizes of all buffers, including portions that
+ * have already been consumed.
+ */
+G_GNUC_PURE
+static size_t
+curl_total_buffer_size(const struct input_curl *c)
+{
+	size_t total = 0;
+
+	for (GList *i = g_queue_peek_head_link(c->buffers);
+	     i != NULL; i = g_list_next(i)) {
+		struct buffer *buffer = i->data;
+		total += buffer->size;
+	}
+
+	return total;
+}
+
 static void
 buffer_free_callback(gpointer data, G_GNUC_UNUSED gpointer user_data)
 {
@@ -473,6 +499,10 @@ static int
 input_curl_buffer(struct input_stream *is, GError **error_r)
 {
 	struct input_curl *c = (struct input_curl *)is;
+
+	if (curl_total_buffer_size(c) >= CURL_MAX_BUFFERED)
+		return 0;
+
 	CURLMcode mcode;
 	int running_handles;
 	bool ret;
@@ -650,6 +680,9 @@ input_curl_easy_init(struct input_curl *c, GError **error_r)
 	curl_easy_setopt(c->easy, CURLOPT_MAXREDIRS, 5);
 	curl_easy_setopt(c->easy, CURLOPT_FAILONERROR, true);
 	curl_easy_setopt(c->easy, CURLOPT_ERRORBUFFER, c->error);
+	curl_easy_setopt(c->easy, CURLOPT_NOPROGRESS, 1l);
+	curl_easy_setopt(c->easy, CURLOPT_NOSIGNAL, 1l);
+	curl_easy_setopt(c->easy, CURLOPT_CONNECTTIMEOUT, 10l);
 
 	if (proxy != NULL)
 		curl_easy_setopt(c->easy, CURLOPT_PROXY, proxy);

src/mixer/winmm_mixer_plugin.c

@@ -58,11 +58,11 @@ winmm_mixer_init(void *ao, G_GNUC_UNUSED const struct config_param *param,
 		 G_GNUC_UNUSED GError **error_r)
 {
 	assert(ao != NULL);
-	
+
 	struct winmm_mixer *wm = g_new(struct winmm_mixer, 1);
 	mixer_init(&wm->base, &winmm_mixer_plugin);
 	wm->output = (struct winmm_output *) ao;
-	
+
 	return &wm->base;
 }
 
@@ -79,13 +79,13 @@ winmm_mixer_get_volume(struct mixer *mixer, GError **error_r)
 	DWORD volume;
 	HWAVEOUT handle = winmm_output_get_handle(wm->output);
 	MMRESULT result = waveOutGetVolume(handle, &volume);
-	
+
 	if (result != MMSYSERR_NOERROR) {
 		g_set_error(error_r, 0, winmm_mixer_quark(),
 			    "Failed to get winmm volume");
 		return -1;
 	}
-	
+
 	return winmm_volume_decode(volume);
 }
 
@@ -102,7 +102,7 @@ winmm_mixer_set_volume(struct mixer *mixer, unsigned volume, GError **error_r)
 			    "Failed to set winmm volume");
 		return false;
 	}
-	
+
 	return true;
 }
 

src/output/alsa_plugin.c

@@ -508,6 +508,14 @@ configure_hw:
 	g_debug("buffer_size=%u period_size=%u",
 		(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
 
+	if (alsa_period_size == 0)
+		/* this works around a SIGFPE bug that occurred when
+		   an ALSA driver indicated period_size==0; this
+		   caused a division by zero in alsa_play().  By using
+		   the fallback "1", we make sure that this won't
+		   happen again. */
+		alsa_period_size = 1;
+
 	ad->period_frames = alsa_period_size;
 	ad->period_position = 0;
 

src/output/ao_plugin.c

@@ -26,6 +26,9 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "ao"
 
+/* An ao_sample_format, with all fields set to zero: */
+static const ao_sample_format OUR_AO_FORMAT_INITIALIZER;
+
 static unsigned ao_output_ref;
 
 struct ao_data {
@@ -103,12 +106,14 @@ ao_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 		g_set_error(error, ao_output_quark(), 0,
 			    "\"%s\" is not a valid ao driver",
 			    value);
+		g_free(ad);
 		return NULL;
 	}
 
 	if ((ai = ao_driver_info(ad->driver)) == NULL) {
 		g_set_error(error, ao_output_quark(), 0,
 			    "problems getting driver info");
+		g_free(ad);
 		return NULL;
 	}
 
@@ -126,6 +131,7 @@ ao_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 				g_set_error(error, ao_output_quark(), 0,
 					    "problems parsing options \"%s\"",
 					    options[i]);
+				g_free(ad);
 				return NULL;
 			}
 
@@ -167,7 +173,7 @@ static bool
 ao_output_open(void *data, struct audio_format *audio_format,
 	       GError **error)
 {
-	ao_sample_format format;
+	ao_sample_format format = OUR_AO_FORMAT_INITIALIZER;
 	struct ao_data *ad = (struct ao_data *)data;
 
 	switch (audio_format->format) {

src/output/httpd_client.c

@@ -143,6 +143,8 @@ httpd_client_unref_page(gpointer data, G_GNUC_UNUSED gpointer user_data)
 void
 httpd_client_free(struct httpd_client *client)
 {
+	assert(client != NULL);
+
 	if (client->state == RESPONSE) {
 		if (client->write_source_id != 0)
 			g_source_remove(client->write_source_id);
@@ -169,6 +171,8 @@ httpd_client_free(struct httpd_client *client)
 static void
 httpd_client_close(struct httpd_client *client)
 {
+	assert(client != NULL);
+
 	httpd_output_remove_client(client->httpd, client);
 	httpd_client_free(client);
 }
@@ -179,6 +183,9 @@ httpd_client_close(struct httpd_client *client)
 static void
 httpd_client_begin_response(struct httpd_client *client)
 {
+	assert(client != NULL);
+	assert(client->state != RESPONSE);
+
 	client->state = RESPONSE;
 	client->write_source_id = 0;
 	client->pages = g_queue_new();
@@ -239,6 +246,9 @@ httpd_client_handle_line(struct httpd_client *client, const char *line)
 static char *
 httpd_client_read_line(struct httpd_client *client)
 {
+	assert(client != NULL);
+	assert(client->state != RESPONSE);
+
 	const char *p, *newline;
 	size_t length;
 	char *line;
@@ -271,6 +281,7 @@ httpd_client_send_response(struct httpd_client *client)
 	GIOStatus status;
 	gsize bytes_written;
 
+	assert(client != NULL);
 	assert(client->state == RESPONSE);
 
 	if (!client->metadata_requested) {
@@ -334,14 +345,19 @@ httpd_client_send_response(struct httpd_client *client)
 static bool
 httpd_client_received(struct httpd_client *client)
 {
+	assert(client != NULL);
+	assert(client->state != RESPONSE);
+
 	char *line;
 	bool success;
 
 	while ((line = httpd_client_read_line(client)) != NULL) {
 		success = httpd_client_handle_line(client, line);
 		g_free(line);
-		if (!success)
+		if (!success) {
+			assert(client->state != RESPONSE);
 			return false;
+		}
 
 		if (client->state == RESPONSE) {
 			if (!fifo_buffer_is_empty(client->input)) {
@@ -370,7 +386,14 @@ httpd_client_read(struct httpd_client *client)
 	if (client->state == RESPONSE) {
 		/* the client has already sent the request, and he
 		   must not send more */
-		g_warning("unexpected input from client");
+		char buffer[1];
+
+		status = g_io_channel_read_chars(client->channel, buffer,
+						 sizeof(buffer), &bytes_read,
+						 NULL);
+		if (status == G_IO_STATUS_NORMAL)
+			g_warning("unexpected input from client");
+
 		return false;
 	}
 

src/output/httpd_internal.h

@@ -111,7 +111,7 @@ struct httpd_output {
 	char buffer[32768];
 
 	/**
-	 * The maximum and current number of clients connected 
+	 * The maximum and current number of clients connected
 	 * at the same time.
 	 */
 	guint clients_max, clients_cnt;

src/output/httpd_output_plugin.c

@@ -36,6 +36,7 @@
 #include <errno.h>
 
 #ifdef HAVE_LIBWRAP
+#include <sys/socket.h> /* needed for AF_UNIX */
 #include <tcpd.h>
 #endif
 
@@ -102,6 +103,7 @@ httpd_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 	if (encoder_plugin == NULL) {
 		g_set_error(error, httpd_output_quark(), 0,
 			    "No such encoder: %s", encoder_name);
+		g_free(httpd);
 		return NULL;
 	}
 
@@ -123,6 +125,7 @@ httpd_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 
 	/* initialize metadata */
 	httpd->metadata = NULL;
+	httpd->unflushed_input = 0;
 
 	/* initialize encoder */
 
@@ -490,7 +493,8 @@ httpd_output_pause(void *data)
 
 	if (has_clients) {
 		static const char silence[1020];
-		return httpd_output_play(data, silence, sizeof(silence), NULL);
+		return httpd_output_play(data, silence, sizeof(silence),
+					 NULL) > 0;
 	} else {
 		g_usleep(100000);
 		return true;
@@ -519,7 +523,7 @@ httpd_output_tag(void *data, const struct tag *tag)
 
 		/* flush the current stream, and end it */
 
-		encoder_flush(httpd->encoder, NULL);
+		encoder_pre_tag(httpd->encoder, NULL);
 		httpd_output_encoder_to_clients(httpd);
 
 		/* send the tag to the encoder - which starts a new

src/output/jack_output_plugin.c

@@ -40,7 +40,7 @@ enum {
 	MAX_PORTS = 16,
 };
 
-static const size_t sample_size = sizeof(jack_default_audio_sample_t);
+static const size_t jack_sample_size = sizeof(jack_default_audio_sample_t);
 
 struct jack_data {
 	/**
@@ -103,9 +103,9 @@ mpd_jack_available(const struct jack_data *jd)
 			min = current;
 	}
 
-	assert(min % sample_size == 0);
+	assert(min % jack_sample_size == 0);
 
-	return min / sample_size;
+	return min / jack_sample_size;
 }
 
 static int
@@ -123,7 +123,7 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 		const jack_nframes_t available = mpd_jack_available(jd);
 		for (unsigned i = 0; i < jd->audio_format.channels; ++i)
 			jack_ringbuffer_read_advance(jd->ringbuffer[i],
-						     available * sample_size);
+						     available * jack_sample_size);
 
 		/* generate silence while MPD is paused */
 
@@ -144,7 +144,7 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 	for (unsigned i = 0; i < jd->audio_format.channels; ++i) {
 		out = jack_port_get_buffer(jd->ports[i], nframes);
 		jack_ringbuffer_read(jd->ringbuffer[i],
-				     (char *)out, available * sample_size);
+				     (char *)out, available * jack_sample_size);
 
 		for (jack_nframes_t f = available; f < nframes; ++f)
 			/* ringbuffer underrun, fill with silence */
@@ -675,7 +675,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error_r)
 				space = space1;
 		}
 
-		if (space >= frame_size)
+		if (space >= jack_sample_size)
 			break;
 
 		/* XXX do something more intelligent to
@@ -683,7 +683,7 @@ mpd_jack_play(void *data, const void *chunk, size_t size, GError **error_r)
 		g_usleep(1000);
 	}
 
-	space /= sample_size;
+	space /= jack_sample_size;
 	if (space < size)
 		size = space;
 

src/output/mvp_plugin.c

@@ -17,7 +17,7 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-/* 
+/*
  * Media MVP audio output based on code from MVPMC project:
  * http://mvpmc.sourceforge.net/
  */

src/output/oss_plugin.c

@@ -41,6 +41,15 @@
 # include <sys/soundcard.h>
 #endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
 
+/* We got bug reports from FreeBSD users who said that the two 24 bit
+   formats generate white noise on FreeBSD, but 32 bit works.  This is
+   a workaround until we know what exactly is expected by the kernel
+   audio drivers. */
+#ifndef __linux__
+#undef AFMT_S24_PACKED
+#undef AFMT_S24_NE
+#endif
+
 struct oss_data {
 	int fd;
 	const char *device;
@@ -347,7 +356,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
 		case SUCCESS:
 			if (!audio_valid_sample_rate(sample_rate))
 				break;
-		
+
 			audio_format->sample_rate = sample_rate;
 			return true;
 
@@ -461,6 +470,12 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 			break;
 
 		audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+		if (oss_format == AFMT_S24_PACKED)
+			audio_format->reverse_endian =
+				G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
 		return true;
 
 	case ERROR:
@@ -502,6 +517,12 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 				break;
 
 			audio_format->format = mpd_format;
+
+#ifdef AFMT_S24_PACKED
+			if (oss_format == AFMT_S24_PACKED)
+				audio_format->reverse_endian =
+					G_BYTE_ORDER != G_LITTLE_ENDIAN;
+#endif
 			return true;
 
 		case ERROR:

src/output/osx_plugin.c

@@ -95,12 +95,6 @@ static void osx_output_close(void *data)
 {
 	struct osx_output *od = data;
 
-	g_mutex_lock(od->mutex);
-	while (od->len) {
-		g_cond_wait(od->condition, od->mutex);
-	}
-	g_mutex_unlock(od->mutex);
-
 	AudioOutputUnitStop(od->au);
 	AudioUnitUninitialize(od->au);
 	CloseComponent(od->au);
@@ -143,8 +137,8 @@ osx_render(void *vdata,
 	if (od->pos >= od->buffer_size)
 		od->pos = 0;
 
-	g_mutex_unlock(od->mutex);
 	g_cond_signal(od->condition);
+	g_mutex_unlock(od->mutex);
 
 	buffer->mDataByteSize = buffer_size;
 

src/output/pulse_output_plugin.c

@@ -207,6 +207,9 @@ pulse_output_subscribe_cb(pa_context *context,
 static bool
 pulse_output_connect(struct pulse_output *po, GError **error_r)
 {
+	assert(po != NULL);
+	assert(po->context != NULL);
+
 	int error;
 
 	error = pa_context_connect(po->context, po->server,
@@ -221,6 +224,44 @@ pulse_output_connect(struct pulse_output *po, GError **error_r)
 	return true;
 }
 
+/**
+ * Frees and clears the stream.
+ */
+static void
+pulse_output_delete_stream(struct pulse_output *po)
+{
+	assert(po != NULL);
+	assert(po->stream != NULL);
+
+#if PA_CHECK_VERSION(0,9,8)
+	pa_stream_set_suspended_callback(po->stream, NULL, NULL);
+#endif
+
+	pa_stream_set_state_callback(po->stream, NULL, NULL);
+	pa_stream_set_write_callback(po->stream, NULL, NULL);
+
+	pa_stream_disconnect(po->stream);
+	pa_stream_unref(po->stream);
+	po->stream = NULL;
+}
+
+/**
+ * Frees and clears the context.
+ */
+static void
+pulse_output_delete_context(struct pulse_output *po)
+{
+	assert(po != NULL);
+	assert(po->context != NULL);
+
+	pa_context_set_state_callback(po->context, NULL, NULL);
+	pa_context_set_subscribe_callback(po->context, NULL, NULL);
+
+	pa_context_disconnect(po->context);
+	pa_context_unref(po->context);
+	po->context = NULL;
+}
+
 /**
  * Create, set up and connect a context.
  *
@@ -229,6 +270,9 @@ pulse_output_connect(struct pulse_output *po, GError **error_r)
 static bool
 pulse_output_setup_context(struct pulse_output *po, GError **error_r)
 {
+	assert(po != NULL);
+	assert(po->mainloop != NULL);
+
 	po->context = pa_context_new(pa_threaded_mainloop_get_api(po->mainloop),
 				     MPD_PULSE_NAME);
 	if (po->context == NULL) {
@@ -243,25 +287,13 @@ pulse_output_setup_context(struct pulse_output *po, GError **error_r)
 					  pulse_output_subscribe_cb, po);
 
 	if (!pulse_output_connect(po, error_r)) {
-		pa_context_unref(po->context);
-		po->context = NULL;
+		pulse_output_delete_context(po);
 		return false;
 	}
 
 	return true;
 }
 
-/**
- * Frees and clears the context.
- */
-static void
-pulse_output_delete_context(struct pulse_output *po)
-{
-	pa_context_disconnect(po->context);
-	pa_context_unref(po->context);
-	po->context = NULL;
-}
-
 static void *
 pulse_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 		  const struct config_param *param,
@@ -347,6 +379,8 @@ pulse_output_disable(void *data)
 {
 	struct pulse_output *po = data;
 
+	assert(po->mainloop != NULL);
+
 	pa_threaded_mainloop_stop(po->mainloop);
 	if (po->context != NULL)
 		pulse_output_delete_context(po);
@@ -363,6 +397,8 @@ pulse_output_disable(void *data)
 static bool
 pulse_output_wait_connection(struct pulse_output *po, GError **error_r)
 {
+	assert(po->mainloop != NULL);
+
 	pa_context_state_t state;
 
 	pa_threaded_mainloop_lock(po->mainloop);
@@ -399,11 +435,32 @@ pulse_output_wait_connection(struct pulse_output *po, GError **error_r)
 	}
 }
 
+#if PA_CHECK_VERSION(0,9,8)
+
+static void
+pulse_output_stream_suspended_cb(G_GNUC_UNUSED pa_stream *stream, void *userdata)
+{
+	struct pulse_output *po = userdata;
+
+	assert(stream == po->stream || po->stream == NULL);
+	assert(po->mainloop != NULL);
+
+	/* wake up the main loop to break out of the loop in
+	   pulse_output_play() */
+	pa_threaded_mainloop_signal(po->mainloop, 0);
+}
+
+#endif
+
 static void
 pulse_output_stream_state_cb(pa_stream *stream, void *userdata)
 {
 	struct pulse_output *po = userdata;
 
+	assert(stream == po->stream || po->stream == NULL);
+	assert(po->mainloop != NULL);
+	assert(po->context != NULL);
+
 	switch (pa_stream_get_state(stream)) {
 	case PA_STREAM_READY:
 		if (po->mixer != NULL)
@@ -432,6 +489,8 @@ pulse_output_stream_write_cb(G_GNUC_UNUSED pa_stream *stream, size_t nbytes,
 {
 	struct pulse_output *po = userdata;
 
+	assert(po->mainloop != NULL);
+
 	po->writable = nbytes;
 	pa_threaded_mainloop_signal(po->mainloop, 0);
 }
@@ -444,6 +503,8 @@ pulse_output_open(void *data, struct audio_format *audio_format,
 	pa_sample_spec ss;
 	int error;
 
+	assert(po->mainloop != NULL);
+
 	if (po->context != NULL) {
 		switch (pa_context_get_state(po->context)) {
 		case PA_CONTEXT_UNCONNECTED:
@@ -487,6 +548,11 @@ pulse_output_open(void *data, struct audio_format *audio_format,
 		return false;
 	}
 
+#if PA_CHECK_VERSION(0,9,8)
+	pa_stream_set_suspended_callback(po->stream,
+					 pulse_output_stream_suspended_cb, po);
+#endif
+
 	pa_stream_set_state_callback(po->stream,
 				     pulse_output_stream_state_cb, po);
 	pa_stream_set_write_callback(po->stream,
@@ -497,8 +563,7 @@ pulse_output_open(void *data, struct audio_format *audio_format,
 	error = pa_stream_connect_playback(po->stream, po->sink,
 					   NULL, 0, NULL, NULL);
 	if (error < 0) {
-		pa_stream_unref(po->stream);
-		po->stream = NULL;
+		pulse_output_delete_stream(po);
 
 		g_set_error(error_r, pulse_output_quark(), 0,
 			    "pa_stream_connect_playback() has failed: %s",
@@ -522,6 +587,8 @@ pulse_output_close(void *data)
 	struct pulse_output *po = data;
 	pa_operation *o;
 
+	assert(po->mainloop != NULL);
+
 	pa_threaded_mainloop_lock(po->mainloop);
 
 	if (pa_stream_get_state(po->stream) == PA_STREAM_READY) {
@@ -534,9 +601,7 @@ pulse_output_close(void *data)
 			pulse_wait_for_operation(po->mainloop, o);
 	}
 
-	pa_stream_disconnect(po->stream);
-	pa_stream_unref(po->stream);
-	po->stream = NULL;
+	pulse_output_delete_stream(po);
 
 	if (po->context != NULL &&
 	    pa_context_get_state(po->context) != PA_CONTEXT_READY)
@@ -556,6 +621,8 @@ pulse_output_check_stream(struct pulse_output *po)
 {
 	pa_stream_state_t state = pa_stream_get_state(po->stream);
 
+	assert(po->mainloop != NULL);
+
 	switch (state) {
 	case PA_STREAM_READY:
 	case PA_STREAM_FAILED:
@@ -637,6 +704,8 @@ pulse_output_stream_pause(struct pulse_output *po, bool pause,
 {
 	pa_operation *o;
 
+	assert(po->mainloop != NULL);
+	assert(po->context != NULL);
 	assert(po->stream != NULL);
 
 	o = pa_stream_cork(po->stream, pause,
@@ -667,6 +736,7 @@ pulse_output_play(void *data, const void *chunk, size_t size, GError **error_r)
 	struct pulse_output *po = data;
 	int error;
 
+	assert(po->mainloop != NULL);
 	assert(po->stream != NULL);
 
 	pa_threaded_mainloop_lock(po->mainloop);
@@ -683,19 +753,30 @@ pulse_output_play(void *data, const void *chunk, size_t size, GError **error_r)
 	/* unpause if previously paused */
 
 	if (pulse_output_stream_is_paused(po) &&
-	    !pulse_output_stream_pause(po, false, error_r))
+	    !pulse_output_stream_pause(po, false, error_r)) {
+		pa_threaded_mainloop_unlock(po->mainloop);
 		return 0;
+	}
 
 	/* wait until the server allows us to write */
 
 	while (po->writable == 0) {
+#if PA_CHECK_VERSION(0,9,8)
+		if (pa_stream_is_suspended(po->stream)) {
+			pa_threaded_mainloop_unlock(po->mainloop);
+			g_set_error(error_r, pulse_output_quark(), 0,
+				    "suspended");
+			return 0;
+		}
+#endif
+
 		pa_threaded_mainloop_wait(po->mainloop);
 
 		if (pa_stream_get_state(po->stream) != PA_STREAM_READY) {
 			pa_threaded_mainloop_unlock(po->mainloop);
 			g_set_error(error_r, pulse_output_quark(), 0,
 				    "disconnected");
-			return false;
+			return 0;
 		}
 	}
 
@@ -725,6 +806,7 @@ pulse_output_cancel(void *data)
 	struct pulse_output *po = data;
 	pa_operation *o;
 
+	assert(po->mainloop != NULL);
 	assert(po->stream != NULL);
 
 	pa_threaded_mainloop_lock(po->mainloop);
@@ -756,6 +838,7 @@ pulse_output_pause(void *data)
 	struct pulse_output *po = data;
 	GError *error = NULL;
 
+	assert(po->mainloop != NULL);
 	assert(po->stream != NULL);
 
 	pa_threaded_mainloop_lock(po->mainloop);

src/output/recorder_output_plugin.c

@@ -79,23 +79,27 @@ recorder_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 	if (encoder_plugin == NULL) {
 		g_set_error(error_r, recorder_output_quark(), 0,
 			    "No such encoder: %s", encoder_name);
-		return NULL;
+		goto failure;
 	}
 
 	recorder->path = config_get_block_string(param, "path", NULL);
 	if (recorder->path == NULL) {
 		g_set_error(error_r, recorder_output_quark(), 0,
 			    "'path' not configured");
-		return NULL;
+		goto failure;
 	}
 
 	/* initialize encoder */
 
 	recorder->encoder = encoder_init(encoder_plugin, param, error_r);
 	if (recorder->encoder == NULL)
-		return NULL;
+		goto failure;
 
 	return recorder;
+
+failure:
+	g_free(recorder);
+	return NULL;
 }
 
 static void

src/output/shout_plugin.c

@@ -152,7 +152,7 @@ my_shout_init_driver(const struct audio_format *audio_format,
 	if (port == 0) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "shout port must be configured");
-		return NULL;
+		goto failure;
 	}
 
 	check_block_param("password");
@@ -174,21 +174,21 @@ my_shout_init_driver(const struct audio_format *audio_format,
 				    "shout quality \"%s\" is not a number in the "
 				    "range -1 to 10, line %i",
 				    value, param->line);
-			return NULL;
+			goto failure;
 		}
 
 		if (config_get_block_string(param, "bitrate", NULL) != NULL) {
 			g_set_error(error, shout_output_quark(), 0,
 				    "quality and bitrate are "
 				    "both defined");
-			return NULL;
+			goto failure;
 		}
 	} else {
 		value = config_get_block_string(param, "bitrate", NULL);
 		if (value == NULL) {
 			g_set_error(error, shout_output_quark(), 0,
 				    "neither bitrate nor quality defined");
-			return NULL;
+			goto failure;
 		}
 
 		sd->bitrate = strtol(value, &test, 10);
@@ -196,7 +196,7 @@ my_shout_init_driver(const struct audio_format *audio_format,
 		if (*test != '\0' || sd->bitrate <= 0) {
 			g_set_error(error, shout_output_quark(), 0,
 				    "bitrate must be a positive integer");
-			return NULL;
+			goto failure;
 		}
 	}
 
@@ -206,12 +206,12 @@ my_shout_init_driver(const struct audio_format *audio_format,
 		g_set_error(error, shout_output_quark(), 0,
 			    "couldn't find shout encoder plugin \"%s\"",
 			    encoding);
-		return NULL;
+		goto failure;
 	}
 
 	sd->encoder = encoder_init(encoder_plugin, param, error);
 	if (sd->encoder == NULL)
-		return NULL;
+		goto failure;
 
 	if (strcmp(encoding, "mp3") == 0 || strcmp(encoding, "lame") == 0)
 		shout_format = SHOUT_FORMAT_MP3;
@@ -225,7 +225,7 @@ my_shout_init_driver(const struct audio_format *audio_format,
 			g_set_error(error, shout_output_quark(), 0,
 				    "you cannot stream \"%s\" to shoutcast, use mp3",
 				    encoding);
-			return NULL;
+			goto failure;
 		} else if (0 == strcmp(value, "shoutcast"))
 			protocol = SHOUT_PROTOCOL_ICY;
 		else if (0 == strcmp(value, "icecast1"))
@@ -237,7 +237,7 @@ my_shout_init_driver(const struct audio_format *audio_format,
 				    "shout protocol \"%s\" is not \"shoutcast\" or "
 				    "\"icecast1\"or \"icecast2\"",
 				    value);
-			return NULL;
+			goto failure;
 		}
 	} else {
 		protocol = SHOUT_PROTOCOL_HTTP;
@@ -256,7 +256,7 @@ my_shout_init_driver(const struct audio_format *audio_format,
 	    shout_set_agent(sd->shout_conn, "MPD") != SHOUTERR_SUCCESS) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "%s", shout_get_error(sd->shout_conn));
-		return NULL;
+		goto failure;
 	}
 
 	/* optional paramters */
@@ -267,14 +267,14 @@ my_shout_init_driver(const struct audio_format *audio_format,
 	if (value != NULL && shout_set_genre(sd->shout_conn, value)) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "%s", shout_get_error(sd->shout_conn));
-		return NULL;
+		goto failure;
 	}
 
 	value = config_get_block_string(param, "description", NULL);
 	if (value != NULL && shout_set_description(sd->shout_conn, value)) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "%s", shout_get_error(sd->shout_conn));
-		return NULL;
+		goto failure;
 	}
 
 	{
@@ -300,6 +300,10 @@ my_shout_init_driver(const struct audio_format *audio_format,
 	}
 
 	return sd;
+
+failure:
+	free_shout_data(sd);
+	return NULL;
 }
 
 static bool
@@ -507,7 +511,7 @@ static void my_shout_set_tag(void *data,
 	if (sd->encoder->plugin->tag != NULL) {
 		/* encoder plugin supports stream tags */
 
-		ret = encoder_flush(sd->encoder, &error);
+		ret = encoder_pre_tag(sd->encoder, &error);
 		if (!ret) {
 			g_warning("%s", error->message);
 			g_error_free(error);

src/output_all.c

@@ -205,27 +205,14 @@ static void audio_output_wait_all(void)
 		notify_wait(&audio_output_client_notify);
 }
 
-/**
- * Signals the audio output if it is open.  This function locks the
- * mutex.
- */
-static void
-audio_output_lock_signal(struct audio_output *ao)
-{
-	g_mutex_lock(ao->mutex);
-	if (audio_output_is_open(ao))
-		g_cond_signal(ao->cond);
-	g_mutex_unlock(ao->mutex);
-}
-
 /**
  * Signals all audio outputs which are open.
  */
 static void
-audio_output_signal_all(void)
+audio_output_allow_play_all(void)
 {
 	for (unsigned i = 0; i < num_audio_outputs; ++i)
-		audio_output_lock_signal(&audio_outputs[i]);
+		audio_output_allow_play(&audio_outputs[i]);
 }
 
 static void
@@ -530,7 +517,7 @@ audio_output_all_cancel(void)
 	/* the audio outputs are now waiting for a signal, to
 	   synchronize the cleared music pipe */
 
-	audio_output_signal_all();
+	audio_output_allow_play_all();
 
 	/* invalidate elapsed_time */
 

src/output_control.c

@@ -115,6 +115,9 @@ audio_output_open(struct audio_output *ao,
 {
 	bool open;
 
+	assert(ao != NULL);
+	assert(ao->allow_play);
+	assert(audio_format_valid(audio_format));
 	assert(mp != NULL);
 
 	if (ao->fail_timer != NULL) {
@@ -139,10 +142,6 @@ audio_output_open(struct audio_output *ao,
 			/* we're not using audio_output_cancel() here,
 			   because that function is asynchronous */
 			ao_command(ao, AO_COMMAND_CANCEL);
-
-			/* the audio output is now waiting for a
-			   signal; wake it up immediately */
-			g_cond_signal(ao->cond);
 		}
 
 		return true;
@@ -180,6 +179,7 @@ static void
 audio_output_close_locked(struct audio_output *ao)
 {
 	assert(ao != NULL);
+	assert(ao->allow_play);
 
 	if (ao->mixer != NULL)
 		mixer_auto_close(ao->mixer);
@@ -222,6 +222,8 @@ audio_output_play(struct audio_output *ao)
 {
 	g_mutex_lock(ao->mutex);
 
+	assert(ao->allow_play);
+
 	if (audio_output_is_open(ao))
 		g_cond_signal(ao->cond);
 
@@ -237,6 +239,7 @@ void audio_output_pause(struct audio_output *ao)
 		mixer_auto_close(ao->mixer);
 
 	g_mutex_lock(ao->mutex);
+	assert(ao->allow_play);
 	if (audio_output_is_open(ao))
 		ao_command_async(ao, AO_COMMAND_PAUSE);
 	g_mutex_unlock(ao->mutex);
@@ -246,6 +249,7 @@ void
 audio_output_drain_async(struct audio_output *ao)
 {
 	g_mutex_lock(ao->mutex);
+	assert(ao->allow_play);
 	if (audio_output_is_open(ao))
 		ao_command_async(ao, AO_COMMAND_DRAIN);
 	g_mutex_unlock(ao->mutex);
@@ -254,8 +258,24 @@ audio_output_drain_async(struct audio_output *ao)
 void audio_output_cancel(struct audio_output *ao)
 {
 	g_mutex_lock(ao->mutex);
-	if (audio_output_is_open(ao))
+
+	if (audio_output_is_open(ao)) {
+		ao->allow_play = false;
 		ao_command_async(ao, AO_COMMAND_CANCEL);
+	}
+
+	g_mutex_unlock(ao->mutex);
+}
+
+void
+audio_output_allow_play(struct audio_output *ao)
+{
+	g_mutex_lock(ao->mutex);
+
+	ao->allow_play = true;
+	if (audio_output_is_open(ao))
+		g_cond_signal(ao->cond);
+
 	g_mutex_unlock(ao->mutex);
 }
 
@@ -286,6 +306,7 @@ void audio_output_finish(struct audio_output *ao)
 
 	if (ao->thread != NULL) {
 		g_mutex_lock(ao->mutex);
+		assert(ao->allow_play);
 		ao_command(ao, AO_COMMAND_KILL);
 		g_mutex_unlock(ao->mutex);
 		g_thread_join(ao->thread);

src/output_control.h

@@ -70,8 +70,19 @@ void audio_output_pause(struct audio_output *ao);
 void
 audio_output_drain_async(struct audio_output *ao);
 
+/**
+ * Clear the "allow_play" flag and send the "CANCEL" command
+ * asynchronously.  To finish the operation, the caller has to call
+ * audio_output_allow_play().
+ */
 void audio_output_cancel(struct audio_output *ao);
 
+/**
+ * Set the "allow_play" and signal the thread.
+ */
+void
+audio_output_allow_play(struct audio_output *ao);
+
 void audio_output_close(struct audio_output *ao);
 
 /**

src/output_init.c

@@ -189,6 +189,7 @@ audio_output_init(struct audio_output *ao, const struct config_param *param,
 	ao->really_enabled = false;
 	ao->open = false;
 	ao->pause = false;
+	ao->allow_play = true;
 	ao->fail_timer = NULL;
 
 	pcm_buffer_init(&ao->cross_fade_buffer);

src/output_internal.h

@@ -109,6 +109,15 @@ struct audio_output {
 	 */
 	bool pause;
 
+	/**
+	 * When this flag is set, the output thread will not do any
+	 * playback.  It will wait until the flag is cleared.
+	 *
+	 * This is used to synchronize the "clear" operation on the
+	 * shared music pipe during the CANCEL command.
+	 */
+	bool allow_play;
+
 	/**
 	 * If not NULL, the device has failed, and this timer is used
 	 * to estimate how long it should stay disabled (unless

src/output_thread.c

@@ -95,6 +95,8 @@ ao_filter_open(struct audio_output *ao,
 	       struct audio_format *audio_format,
 	       GError **error_r)
 {
+	assert(audio_format_valid(audio_format));
+
 	/* the replay_gain filter cannot fail here */
 	if (ao->replay_gain_filter != NULL)
 		filter_open(ao->replay_gain_filter, audio_format, error_r);
@@ -136,6 +138,7 @@ ao_open(struct audio_output *ao)
 	assert(!ao->open);
 	assert(ao->pipe != NULL);
 	assert(ao->chunk == NULL);
+	assert(audio_format_valid(&ao->in_audio_format));
 
 	if (ao->fail_timer != NULL) {
 		/* this can only happen when this
@@ -164,6 +167,8 @@ ao_open(struct audio_output *ao)
 		return;
 	}
 
+	assert(audio_format_valid(filter_audio_format));
+
 	ao->out_audio_format = *filter_audio_format;
 	audio_format_mask_apply(&ao->out_audio_format,
 				&ao->config_audio_format);
@@ -635,15 +640,14 @@ static gpointer audio_output_task(gpointer arg)
 
 		case AO_COMMAND_CANCEL:
 			ao->chunk = NULL;
-			if (ao->open)
-				ao_plugin_cancel(ao->plugin, ao->data);
-			ao_command_finished(ao);
 
-			/* the player thread will now clear our music
-			   pipe - wait for a notify, to give it some
-			   time */
-			if (ao->command == AO_COMMAND_NONE)
-				g_cond_wait(ao->cond, ao->mutex);
+			if (ao->open) {
+				g_mutex_unlock(ao->mutex);
+				ao_plugin_cancel(ao->plugin, ao->data);
+				g_mutex_lock(ao->mutex);
+			}
+
+			ao_command_finished(ao);
 			continue;
 
 		case AO_COMMAND_KILL:
@@ -653,7 +657,7 @@ static gpointer audio_output_task(gpointer arg)
 			return NULL;
 		}
 
-		if (ao->open && ao_play(ao))
+		if (ao->open && ao->allow_play && ao_play(ao))
 			/* don't wait for an event if there are more
 			   chunks in the pipe */
 			continue;

src/pcm_byteswap.c

@@ -49,7 +49,7 @@ const int16_t *pcm_byteswap_16(struct pcm_buffer *buffer,
 
 static inline uint32_t swab32(uint32_t x)
 {
-	return (x << 24) | 
+	return (x << 24) |
 		((x & 0xff00) << 8) |
 		((x & 0xff0000) >> 8) |
 		(x >> 24);

src/pipe.c

@@ -187,5 +187,8 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk)
 unsigned
 music_pipe_size(const struct music_pipe *mp)
 {
-	return mp->size;
+	g_mutex_lock(mp->mutex);
+	unsigned size = mp->size;
+	g_mutex_unlock(mp->mutex);
+	return size;
 }

src/pipe.h

@@ -20,9 +20,10 @@
 #ifndef MPD_PIPE_H
 #define MPD_PIPE_H
 
-#ifndef NDEBUG
+#include <glib.h>
 #include <stdbool.h>
 
+#ifndef NDEBUG
 struct audio_format;
 #endif
 
@@ -38,6 +39,7 @@ struct music_pipe;
 /**
  * Creates a new #music_pipe object.  It is empty.
  */
+G_GNUC_MALLOC
 struct music_pipe *
 music_pipe_new(void);
 
@@ -70,6 +72,7 @@ music_pipe_contains(const struct music_pipe *mp,
  * Returns the first #music_chunk from the pipe.  Returns NULL if the
  * pipe is empty.
  */
+G_GNUC_PURE
 const struct music_chunk *
 music_pipe_peek(const struct music_pipe *mp);
 
@@ -96,9 +99,11 @@ music_pipe_push(struct music_pipe *mp, struct music_chunk *chunk);
 /**
  * Returns the number of chunks currently in this pipe.
  */
+G_GNUC_PURE
 unsigned
 music_pipe_size(const struct music_pipe *mp);
 
+G_GNUC_PURE
 static inline bool
 music_pipe_empty(const struct music_pipe *mp)
 {

src/player_thread.c

@@ -618,7 +618,9 @@ play_chunk(struct song *song, struct music_chunk *chunk,
 		return true;
 	}
 
+	player_lock();
 	pc.bit_rate = chunk->bit_rate;
+	player_unlock();
 
 	/* send the chunk to the audio outputs */
 

src/playlist_control.c

@@ -222,10 +222,12 @@ playlist_seek_song(struct playlist *playlist, unsigned song, float seek_time)
 	playlist->error_count = 0;
 
 	if (!playlist->playing || (unsigned)playlist->current != i) {
-		/* seeking is not within the current song - first
-		   start playing the new song */
+		/* seeking is not within the current song - prepare
+		   song change */
+
+		playlist->playing = true;
+		playlist->current = i;
 
-		playlist_play_order(playlist, i);
 		queued = NULL;
 	}
 

src/playlist_song.c

@@ -105,6 +105,13 @@ playlist_check_translate_song(struct song *song, const char *base_uri)
 		}
 	}
 
+	if (base_uri != NULL && strcmp(base_uri, ".") == 0)
+		/* g_path_get_dirname() returns "." when there is no
+		   directory name in the given path; clear that now,
+		   because it would break the database lookup
+		   functions */
+		base_uri = NULL;
+
 	if (g_path_is_absolute(uri)) {
 		/* XXX fs_charset vs utf8? */
 		char *prefix = base_uri != NULL
@@ -129,7 +136,7 @@ playlist_check_translate_song(struct song *song, const char *base_uri)
 	else
 		uri = g_strdup(uri);
 
-	if (uri_has_scheme(base_uri)) {
+	if (uri_has_scheme(uri)) {
 		dest = song_remote_new(uri);
 		g_free(uri);
 	} else {

src/playlist_vector.h

@@ -51,6 +51,12 @@ playlist_vector_init(struct playlist_vector *pv)
 void
 playlist_vector_deinit(struct playlist_vector *pv);
 
+static inline bool
+playlist_vector_is_empty(const struct playlist_vector *pv)
+{
+	return pv->head == NULL;
+}
+
 struct playlist_metadata *
 playlist_vector_find(struct playlist_vector *pv, const char *name);
 

src/sticker.c

@@ -579,8 +579,10 @@ sticker_load(const char *type, const char *uri)
 	bool success;
 
 	success = sticker_list_values(sticker->table, type, uri);
-	if (!success)
+	if (!success) {
+		sticker_free(sticker);
 		return NULL;
+	}
 
 	if (g_hash_table_size(sticker->table) == 0) {
 		/* don't return empty sticker objects */

src/update_walk.c

@@ -300,6 +300,9 @@ stat_directory(const struct directory *directory, struct stat *st)
 	if (path_fs == NULL)
 		return -1;
 	ret = stat(path_fs, st);
+	if (ret < 0)
+		g_warning("Failed to stat %s: %s", path_fs, g_strerror(errno));
+
 	g_free(path_fs);
 	return ret;
 }
@@ -316,6 +319,9 @@ stat_directory_child(const struct directory *parent, const char *name,
 		return -1;
 
 	ret = stat(path_fs, st);
+	if (ret < 0)
+		g_warning("Failed to stat %s: %s", path_fs, g_strerror(errno));
+
 	g_free(path_fs);
 	return ret;
 }
@@ -557,6 +563,7 @@ directory_child_access(const struct directory *directory,
 	/* access() is useless on WIN32 */
 	(void)directory;
 	(void)name;
+	(void)mode;
 	return true;
 #else
 	char *path = map_directory_child_fs(directory, name);
@@ -707,8 +714,14 @@ skip_symlink(const struct directory *directory, const char *utf8_name)
 		return false;
 	}
 
-	if (buffer[0] == '/')
-		return !follow_outside_symlinks;
+	if (g_path_is_absolute(buffer)) {
+		/* if the symlink points to an absolute path, see if
+		   that path is inside the music directory */
+		const char *relative = map_to_relative_path(buffer);
+		return relative > buffer
+			? !follow_inside_symlinks
+			: !follow_outside_symlinks;
+	}
 
 	p = buffer;
 	while (*p == '.') {

test/read_conf.c

@@ -37,30 +37,28 @@ my_log_func(G_GNUC_UNUSED const gchar *log_domain,
 
 int main(int argc, char **argv)
 {
-	const char *path, *name, *value;
-	GError *error = NULL;
-	bool success;
-	int ret;
-
 	if (argc != 3) {
 		g_printerr("Usage: read_conf FILE SETTING\n");
 		return 1;
 	}
 
-	path = argv[1];
-	name = argv[2];
+	const char *path = argv[1];
+	const char *name = argv[2];
 
 	g_log_set_default_handler(my_log_func, NULL);
 
 	config_global_init();
-	success = config_read_file(path, &error);
+
+	GError *error = NULL;
+	bool success = config_read_file(path, &error);
 	if (!success) {
 		g_printerr("%s:", error->message);
 		g_error_free(error);
 		return 1;
 	}
 
-	value = config_get_string(name, NULL);
+	const char *value = config_get_string(name, NULL);
+	int ret;
 	if (value != NULL) {
 		g_print("%s\n", value);
 		ret = 0;
@@ -70,5 +68,5 @@ int main(int argc, char **argv)
 	}
 
 	config_global_finish();
-	return 0;
+	return ret;
 }

test/run_filter.c

@@ -106,7 +106,6 @@ int main(int argc, char **argv)
 	struct filter *filter;
 	const struct audio_format *out_audio_format;
 	char buffer[4096];
-	size_t frame_size;
 
 	if (argc < 3 || argc > 4) {
 		g_printerr("Usage: run_filter CONFIG NAME [FORMAT] <IN\n");
@@ -162,8 +161,6 @@ int main(int argc, char **argv)
 	g_printerr("audio_format=%s\n",
 		   audio_format_to_string(out_audio_format, &af_string));
 
-	frame_size = audio_format_frame_size(&audio_format);
-
 	/* play */
 
 	while (true) {