mpd version 0.16.3

Keep those when scanning for empty directories.  The check in
playlist_vector_is_empty() was missing.

Makefile.am

@@ -7,6 +7,8 @@ AM_CPPFLAGS += -DSYSTEM_CONFIG_FILE_LOCATION='"$(sysconfdir)/mpd.conf"'
 
 bin_PROGRAMS = src/mpd
 
+noinst_LIBRARIES =
+
 src_mpd_CFLAGS = $(AM_CFLAGS) $(MPD_CFLAGS)
 src_mpd_CPPFLAGS = $(AM_CPPFLAGS) \
 	$(LIBWRAP_CFLAGS) \
@@ -432,7 +434,6 @@ DECODER_CFLAGS = \
 	$(SNDFILE_CFLAGS) \
 	$(AUDIOFILE_CFLAGS) \
 	$(LIBMIKMOD_CFLAGS) \
-	$(MODPLUG_CFLAGS) \
 	$(GME_CFLAGS) \
 	$(SIDPLAY_CFLAGS) \
 	$(FLUIDSYNTH_CFLAGS) \
@@ -448,7 +449,6 @@ DECODER_LIBS = \
 	$(FLAC_LIBS) \
 	$(SNDFILE_LIBS) \
 	$(AUDIOFILE_LIBS) $(LIBMIKMOD_LIBS) \
-	$(MODPLUG_LIBS) \
 	$(GME_LIBS) \
 	$(SIDPLAY_LIBS) \
 	$(FLUIDSYNTH_LIBS) \
@@ -521,7 +521,11 @@ DECODER_SRC += src/decoder/mikmod_decoder_plugin.c
 endif
 
 if HAVE_MODPLUG
-DECODER_SRC += src/decoder/modplug_decoder_plugin.c
+libmodplug_decoder_plugin_a_SOURCES = src/decoder/modplug_decoder_plugin.c
+libmodplug_decoder_plugin_a_CFLAGS = $(src_mpd_CFLAGS) $(MODPLUG_CFLAGS)
+libmodplug_decoder_plugin_a_CPPFLAGS = $(src_mpd_CPPFLAGS)
+noinst_LIBRARIES += libmodplug_decoder_plugin.a
+DECODER_LIBS += libmodplug_decoder_plugin.a $(MODPLUG_LIBS)
 endif
 
 if ENABLE_SIDPLAY
@@ -855,6 +859,7 @@ 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 \
@@ -902,6 +907,7 @@ 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 \
@@ -942,6 +948,7 @@ 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 \
@@ -964,6 +971,7 @@ 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 \
@@ -971,12 +979,15 @@ 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
@@ -984,6 +995,7 @@ 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
@@ -1021,6 +1033,7 @@ 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

@@ -1,3 +1,43 @@
+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:
+  - ffado: disable by default
+* decoder:
+  - mad: work around build failure on Solaris
+  - resolve modplug vs. libsndfile cflags/headers conflict
+* output:
+  - solaris: add missing parameter to open_cloexec() cal
+  - osx: fix up audio format first, then apply it to device
+* player_thread: discard empty chunks while cross-fading
+* player_thread: fix assertion failure due to early seek
+* output_thread: fix double lock
+
+
 ver 0.16 (2010/12/11)
 * protocol:
   - send song modification time to client
@@ -112,6 +152,22 @@ ver 0.16 (2010/12/11)
 * make single mode 'sticky'
 
 
+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)
 * input:
   - rewind: fix assertion failure

configure.ac

@@ -1,5 +1,5 @@
 AC_PREREQ(2.60)
-AC_INIT(mpd, 0.16, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.16.3, 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)
@@ -13,6 +13,7 @@ dnl Programs
 dnl ---------------------------------------------------------------------------
 AC_PROG_CC_C99
 AC_PROG_CXX
+AC_PROG_RANLIB
 
 HAVE_CXX=yes
 if test x$CXX = xg++; then
@@ -157,7 +158,7 @@ AC_ARG_ENABLE(documentation,
 
 AC_ARG_ENABLE(ffado,
 	AS_HELP_STRING([--enable-ffado], [enable libffado (FireWire) support]),,
-	[enable_ffado=auto])
+	[enable_ffado=no])
 
 AC_ARG_ENABLE(ffmpeg,
 	AS_HELP_STRING([--enable-ffmpeg],
@@ -634,7 +635,7 @@ fi
 AM_CONDITIONAL(ENABLE_LASTFM, test x$enable_lastfm = xyes)
 
 dnl ---------------------------------- libogg ---------------------------------
-if test x$with_tremor == xno || test -z $with_tremor; then
+if test x$with_tremor = xno || test -z $with_tremor; then
 	PKG_CHECK_MODULES(OGG, [ogg], enable_ogg=yes, enable_ogg=no)
 fi
 
@@ -849,15 +850,6 @@ if test x$enable_modplug = xyes; then
 fi
 AM_CONDITIONAL(HAVE_MODPLUG, test x$enable_modplug = xyes)
 
-dnl --------------------------- sndfile/modplug test --------------------------
-if test x$enable_sndfile = xauto && test x$enable_modplug = xyes; then
-	dnl If modplug is enabled, enable sndfile only if explicitly
-	dnl requested - modplug's modplug/sndfile.h is known to
-	dnl conflict with libsndfile's sndfile.h.
-	AC_MSG_NOTICE([disabling libsndfile auto-detection, because the modplug decoder is enabled])
-	enable_sndfile=no
-fi
-
 dnl -------------------------------- libsndfile -------------------------------
 dnl See above test, which may disable this.
 MPD_AUTO_PKG(sndfile, SNDFILE, [sndfile],
@@ -942,13 +934,19 @@ 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)
@@ -988,7 +986,7 @@ if test x$enable_vorbis = xyes; then
 	fi
 fi
 
-AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes)
+AM_CONDITIONAL(ENABLE_VORBIS_DECODER, test x$enable_vorbis = xyes || test x$enable_tremor = xyes)
 
 dnl --------------------------------- sidplay ---------------------------------
 found_sidplay=$HAVE_CXX
@@ -1083,7 +1081,7 @@ 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)
@@ -1495,23 +1493,23 @@ dnl ---------------------------------------------------------------------------
 echo ''
 echo '########### MPD CONFIGURATION ############'
 
-echo -ne '\nArchive support:\n\t'
+printf '\nArchive support:\n\t'
 results(bzip2,[bzip2])
 results(iso9660,[ISO9660])
 results(zzip,[ZIP])
 
 if test x$with_zeroconf != xno; then
-	echo -ne '\nAutodiscovery support:\n\t'
+	printf '\nAutodiscovery support:\n\t'
 	results(avahi, [Avahi])
 	results(bonjour, [Bonjour])
 fi
 
-echo -ne '\nClient support:\n\t'
+printf '\nClient support:\n\t'
 results(ipv6, "IPv6")
 results(tcp, "TCP")
 results(un,[UNIX Domain Sockets])
 
-echo -ne '\nFile format support:\n\t'
+printf '\nFile format support:\n\t'
 results(aac, [AAC])
 results(sidplay, [C64 SID])
 results(ffmpeg, [FFMPEG])
@@ -1519,7 +1517,7 @@ results(flac, [FLAC])
 results(fluidsynth, [FluidSynth])
 results(gme, [GME])
 results(sndfile, [libsndfile])
-echo -ne '\n\t'
+printf '\n\t'
 results(mikmod, [MikMod])
 results(modplug, [MODPLUG])
 results(mad, [MAD])
@@ -1527,23 +1525,23 @@ results(mpg123, [MPG123])
 results(mp4, [MP4])
 results(mpc, [Musepack])
 results(oggflac, [OggFLAC], flac)
-echo -ne '\n\t'
+printf '\n\t'
 results(tremor, [OggTremor])
 results(vorbis, [OggVorbis])
 results(audiofile, [WAVE])
 results(wavpack, [WavPack])
 results(wildmidi, [WildMidi])
 
-echo -en '\nOther features:\n\t'
+printf '\nOther features:\n\t'
 results(lsr, [libsamplerate])
 results(inotify, [inotify])
 results(sqlite, [SQLite])
 
-echo -en '\nMetadata support:\n\t'
+printf '\nMetadata support:\n\t'
 results(cue,[cue])
 results(id3,[ID3])
 
-echo -en '\nPlayback support:\n\t'
+printf '\nPlayback support:\n\t'
 results(alsa,ALSA)
 results(ffado,FFADO)
 results(fifo,FIFO)
@@ -1552,14 +1550,14 @@ results(httpd_output,[HTTP Daemon])
 results(jack,[JACK])
 results(ao,[libao])
 results(oss,[OSS])
-echo -ne '\n\t'
+printf '\n\t'
 results(openal,[OpenAL])
 results(osx, [OS X])
 results(pipe_output, [Pipeline])
 results(pulse, [PulseAudio])
 results(mvp, [Media MVP])
 results(shout, [SHOUTcast])
-echo -ne '\n\t'
+printf '\n\t'
 results(solaris, [Solaris])
 results(winmm_output, [WinMM])
 
@@ -1567,7 +1565,7 @@ if
 	test x$enable_shout = xyes ||
 	test x$enable_recorder = xyes ||
 	test x$enable_httpd_output = xyes; then
-		echo -en '\nStreaming encoder support:\n\t'
+		printf '\nStreaming encoder support:\n\t'
 		results(flac_encoder, [FLAC])
 		results(lame_encoder, [LAME])
 		results(vorbis_encoder, [Ogg Vorbis])
@@ -1575,19 +1573,14 @@ if
 		results(wave_encoder, [WAVE])
 fi
 
-echo -en '\nStreaming support:\n\t'
+printf '\nStreaming support:\n\t'
 results(curl,[CURL])
 results(lastfm,[Last.FM])
 results(mms,[MMS])
 
-echo -ne '\n\n##########################################\n\n'
+printf '\n\n##########################################\n\n'
 
-if test x$enable_sndfile = xyes && test x$enable_modplug = xyes; then
-	AC_MSG_WARN([compilation may fail, because libmodplug conflicts with libsndfile])
-	AC_MSG_WARN([libmodplug ships modplug/sndfile.h, which hides libsndfile's sndfile.h])
-fi
-
-echo -ne 'Generating files needed for compilation\n'
+echo 'Generating files needed for compilation'
 
 dnl ---------------------------------------------------------------------------
 dnl Generate files

m4/faad.m4

@@ -58,7 +58,7 @@ if test x$enable_aac = xyes; then
 	fi
 	if test x$enable_aac = xyes; then
   		AC_MSG_CHECKING(that FAAD2 uses buffer and bufferlen)
-		AC_COMPILE_IFELSE([
+		AC_COMPILE_IFELSE([AC_LANG_SOURCE([
 #include <faad.h>
 
 int main() {
@@ -82,9 +82,9 @@ int main() {
 
 	return 0;
 }
-],[AC_MSG_RESULT(yes);AC_DEFINE(HAVE_FAAD_BUFLEN_FUNCS,1,[Define if FAAD2 uses buflen in function calls])],[AC_MSG_RESULT(no);
+])],[AC_MSG_RESULT(yes);AC_DEFINE(HAVE_FAAD_BUFLEN_FUNCS,1,[Define if FAAD2 uses buflen in function calls])],[AC_MSG_RESULT(no);
 		AC_MSG_CHECKING(that FAAD2 can even be used)
-		AC_COMPILE_IFELSE([
+		AC_COMPILE_IFELSE([AC_LANG_SOURCE([
 #include <faad.h>
 
 int main() {
@@ -113,7 +113,7 @@ int main() {
 
 	return 0;
 }
-],AC_MSG_RESULT(yes),[AC_MSG_RESULT(no);enable_aac=no])
+])],AC_MSG_RESULT(yes),[AC_MSG_RESULT(no);enable_aac=no])
 		])
 	fi
 	if test x$enable_aac = xyes; then
@@ -136,7 +136,7 @@ if test x$enable_aac = xyes; then
 	CPPFLAGS=$CFLAGS
 
 	AC_MSG_CHECKING(for broken libfaad headers)
-	AC_COMPILE_IFELSE([
+	AC_COMPILE_IFELSE([AC_LANG_SOURCE([
 #include <faad.h>
 #include <stddef.h>
 #include <stdint.h>
@@ -148,7 +148,7 @@ int main() {
 	faacDecInit2(NULL, NULL, 0, &sample_rate, &channels);
 	return 0;
 }
-	],
+	])],
 		[AC_MSG_RESULT(correct)],
 		[AC_MSG_RESULT(broken);
 		AC_DEFINE(HAVE_FAAD_LONG, 1, [Define if faad.h uses the broken "unsigned long" pointers])])

m4/mpd_check_cflag.m4

@@ -4,9 +4,9 @@ AC_DEFUN([MPD_CHECK_FLAG],[
     [mpd_check_cflag_$var],[
     save_CFLAGS="$CFLAGS"
     CFLAGS="$CFLAGS $1"
-    AC_COMPILE_IFELSE([
+    AC_COMPILE_IFELSE([AC_LANG_SOURCE([
      int main(void) { return 0; }
-     ], [ eval "mpd_check_cflag_$var=yes"
+     ])], [ eval "mpd_check_cflag_$var=yes"
      ], [ eval "mpd_check_cflag_$var=no" ])
     CFLAGS="$save_CFLAGS"
   ])

m4/pretty_print.m4

@@ -1,19 +1,19 @@
 AC_DEFUN([results], [
 	dnl This is a hack to allow "with" names, otherwise "enable".
-	num=`expr match $1 'with'`
+	num=`expr $1 : 'with'`
 	if test "$num" != "0"; then
 		var="`echo '$'$1`"
 	else
 		var="`echo '$'enable_$1`"
 	fi
 
-	echo -n '('
+	printf '('
 	if eval "test x$var = xyes"; then
-		echo -n '+'
+		printf '+'
 	elif test -n "$3" && eval "test x$var = x$3"; then
-		echo -n '+' 
+		printf '+'
 	else
-		echo -n '-'
+		printf '-'
 	fi
-	echo -n "$2) "
+	printf '%s) ' "$2"
 ])

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/audio_check.h

@@ -38,7 +38,7 @@ bool
 audio_check_sample_rate(unsigned long sample_rate, GError **error_r);
 
 bool
-audio_check_sample_format(unsigned sample_format, GError **error_r);
+audio_check_sample_format(enum sample_format, GError **error_r);
 
 bool
 audio_check_channel_count(unsigned sample_format, GError **error_r);

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/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

@@ -81,6 +81,10 @@ 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;
@@ -102,13 +106,11 @@ 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;
@@ -156,7 +158,11 @@ 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 +189,40 @@ 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 +230,13 @@ 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 +251,7 @@ 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;
@@ -299,12 +320,8 @@ 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
+	AVFormatContext *format_context;
 	if (av_open_input_stream(&format_context, stream->io, input->uri,
 				 input_format, NULL) != 0) {
 		g_warning("Open failed\n");
@@ -319,7 +336,7 @@ 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 +344,12 @@ 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");

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/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/mad_decoder_plugin.c

@@ -547,14 +547,14 @@ enum {
 	XING_SCALE  = 0x00000008L
 };
 
-struct version {
+struct lame_version {
 	unsigned major;
 	unsigned minor;
 };
 
 struct lame {
 	char encoder[10];       /* 9 byte encoder name/version ("LAME3.97b") */
-	struct version version; /* struct containing just the version */
+	struct lame_version version; /* struct containing just the version */
 	float peak;             /* replaygain peak */
 	float track_gain;       /* replaygain track gain */
 	float album_gain;       /* replaygain album gain */

src/directory.h

@@ -30,8 +30,8 @@
 
 #define DIRECTORY_DIR		"directory: "
 
-#define DEVICE_INARCHIVE	(dev_t)(-1)
-#define DEVICE_CONTAINER	(dev_t)(-2)
+#define DEVICE_INARCHIVE (dev_t)(-1)
+#define DEVICE_CONTAINER (dev_t)(-2)
 
 struct directory {
 	struct dirvec children;
@@ -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/lame_encoder.c

@@ -170,6 +170,13 @@ lame_encoder_setup(struct lame_encoder *encoder, GError **error)
 		return false;
 	}
 
+	if (0 != lame_set_out_samplerate(encoder->gfp,
+					 encoder->audio_format.sample_rate)) {
+		g_set_error(error, lame_encoder_quark(), 0,
+			    "error setting lame out sample rate");
+		return false;
+	}
+
 	if (0 > lame_init_params(encoder->gfp)) {
 		g_set_error(error, lame_encoder_quark(), 0,
 			    "error initializing lame params");

src/encoder/vorbis_encoder.c

@@ -276,6 +276,8 @@ 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;
 }

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/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/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 {
@@ -167,7 +170,7 @@ 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

@@ -29,6 +29,9 @@
 #include <assert.h>
 #include <string.h>
 
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "httpd_output"
+
 struct httpd_client {
 	/**
 	 * The httpd output object this client is connected to.

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
 
@@ -123,6 +124,7 @@ httpd_output_init(G_GNUC_UNUSED const struct audio_format *audio_format,
 
 	/* initialize metadata */
 	httpd->metadata = NULL;
+	httpd->unflushed_input = 0;
 
 	/* initialize encoder */
 

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

@@ -214,15 +214,6 @@ osx_output_open(void *data, struct audio_format *audio_format, GError **error)
 	stream_description.mSampleRate = audio_format->sample_rate;
 	stream_description.mFormatID = kAudioFormatLinearPCM;
 	stream_description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
-#if G_BYTE_ORDER == G_BIG_ENDIAN
-	stream_description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-#endif
-
-	stream_description.mBytesPerPacket =
-		audio_format_frame_size(audio_format);
-	stream_description.mFramesPerPacket = 1;
-	stream_description.mBytesPerFrame = stream_description.mBytesPerPacket;
-	stream_description.mChannelsPerFrame = audio_format->channels;
 
 	switch (audio_format->format) {
 	case SAMPLE_FORMAT_S8:
@@ -239,6 +230,16 @@ osx_output_open(void *data, struct audio_format *audio_format, GError **error)
 		break;
 	}
 
+#if G_BYTE_ORDER == G_BIG_ENDIAN
+	stream_description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+
+	stream_description.mBytesPerPacket =
+		audio_format_frame_size(audio_format);
+	stream_description.mFramesPerPacket = 1;
+	stream_description.mBytesPerFrame = stream_description.mBytesPerPacket;
+	stream_description.mChannelsPerFrame = audio_format->channels;
+
 	result = AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
 				      kAudioUnitScope_Input, 0,
 				      &stream_description,

src/output/solaris_output_plugin.c

@@ -93,7 +93,7 @@ solaris_output_open(void *data, struct audio_format *audio_format,
 
 	/* open the device in non-blocking mode */
 
-	so->fd = open_cloexec(so->device, O_WRONLY|O_NONBLOCK);
+	so->fd = open_cloexec(so->device, O_WRONLY|O_NONBLOCK, 0);
 	if (so->fd < 0) {
 		g_set_error(error, solaris_output_quark(), errno,
 			    "Failed to open %s: %s",

src/output_control.c

@@ -115,6 +115,7 @@ audio_output_open(struct audio_output *ao,
 {
 	bool open;
 
+	assert(audio_format_valid(audio_format));
 	assert(mp != NULL);
 
 	if (ao->fail_timer != NULL) {

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);
@@ -303,7 +308,7 @@ ao_wait(struct audio_output *ao)
 		GTimeVal tv;
 		g_get_current_time(&tv);
 		g_time_val_add(&tv, delay * 1000);
-		g_cond_timed_wait(ao->cond, ao->mutex, &tv);
+		(void)g_cond_timed_wait(ao->cond, ao->mutex, &tv);
 
 		if (ao->command != AO_COMMAND_NONE)
 			return false;
@@ -463,12 +468,9 @@ ao_play_chunk(struct audio_output *ao, const struct music_chunk *chunk)
 
 			/* don't automatically reopen this device for
 			   10 seconds */
-			g_mutex_lock(ao->mutex);
-
 			assert(ao->fail_timer == NULL);
 			ao->fail_timer = g_timer_new();
 
-			g_mutex_unlock(ao->mutex);
 			return false;
 		}
 

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.h

@@ -20,9 +20,9 @@
 #ifndef MPD_PIPE_H
 #define MPD_PIPE_H
 
-#ifndef NDEBUG
 #include <stdbool.h>
 
+#ifndef NDEBUG
 struct audio_format;
 #endif
 

src/player_thread.c

@@ -356,16 +356,9 @@ player_check_decoder_startup(struct player *player)
 static bool
 player_send_silence(struct player *player)
 {
-	struct music_chunk *chunk;
-	size_t frame_size =
-		audio_format_frame_size(&player->play_audio_format);
-	/* this formula ensures that we don't send
-	   partial frames */
-	unsigned num_frames = sizeof(chunk->data) / frame_size;
-
 	assert(audio_format_defined(&player->play_audio_format));
 
-	chunk = music_buffer_allocate(player_buffer);
+	struct music_chunk *chunk = music_buffer_allocate(player_buffer);
 	if (chunk == NULL) {
 		g_warning("Failed to allocate silence buffer");
 		return false;
@@ -375,6 +368,12 @@ player_send_silence(struct player *player)
 	chunk->audio_format = player->play_audio_format;
 #endif
 
+	size_t frame_size =
+		audio_format_frame_size(&player->play_audio_format);
+	/* this formula ensures that we don't send
+	   partial frames */
+	unsigned num_frames = sizeof(chunk->data) / frame_size;
+
 	chunk->times = -1.0; /* undefined time stamp */
 	chunk->length = num_frames * frame_size;
 	memset(chunk->data, 0, chunk->length);
@@ -396,8 +395,6 @@ static bool player_seek_decoder(struct player *player)
 {
 	struct song *song = pc.next_song;
 	struct decoder_control *dc = player->dc;
-	double where;
-	bool ret;
 
 	assert(pc.next_song != NULL);
 
@@ -413,8 +410,7 @@ static bool player_seek_decoder(struct player *player)
 
 		/* re-start the decoder */
 		player_dc_start(player, player->pipe);
-		ret = player_wait_for_decoder(player);
-		if (!ret) {
+		if (!player_wait_for_decoder(player)) {
 			/* decoder failure */
 			player_command_finished();
 			return false;
@@ -435,8 +431,7 @@ static bool player_seek_decoder(struct player *player)
 	/* wait for the decoder to complete initialization */
 
 	while (player->decoder_starting) {
-		ret = player_check_decoder_startup(player);
-		if (!ret) {
+		if (!player_check_decoder_startup(player)) {
 			/* decoder failure */
 			player_command_finished();
 			return false;
@@ -445,14 +440,13 @@ static bool player_seek_decoder(struct player *player)
 
 	/* send the SEEK command */
 
-	where = pc.seek_where;
+	double where = pc.seek_where;
 	if (where > pc.total_time)
 		where = pc.total_time - 0.1;
 	if (where < 0.0)
 		where = 0.0;
 
-	ret = dc_seek(dc, where + song->start_ms / 1000.0);
-	if (!ret) {
+	if (!dc_seek(dc, where + song->start_ms / 1000.0)) {
 		/* decoder failure */
 		player_command_finished();
 		return false;
@@ -583,14 +577,12 @@ static void player_process_command(struct player *player)
 static void
 update_song_tag(struct song *song, const struct tag *new_tag)
 {
-	struct tag *old_tag;
-
 	if (song_is_file(song))
 		/* don't update tags of local files, only remote
 		   streams may change tags dynamically */
 		return;
 
-	old_tag = song->tag;
+	struct tag *old_tag = song->tag;
 	song->tag = tag_dup(new_tag);
 
 	if (old_tag != NULL)
@@ -648,15 +640,14 @@ static bool
 play_next_chunk(struct player *player)
 {
 	struct decoder_control *dc = player->dc;
-	struct music_chunk *chunk = NULL;
-	unsigned cross_fade_position;
-	bool success;
 
 	if (!audio_output_all_wait(64))
 		/* the output pipe is still large enough, don't send
 		   another chunk */
 		return true;
 
+	unsigned cross_fade_position;
+	struct music_chunk *chunk = NULL;
 	if (player->xfade == XFADE_ENABLED &&
 	    player_dc_at_next_song(player) &&
 	    (cross_fade_position = music_pipe_size(player->pipe))
@@ -694,6 +685,19 @@ play_next_chunk(struct player *player)
 				chunk->mix_ratio = nan("");
 			}
 
+			if (music_chunk_is_empty(other_chunk)) {
+				/* the "other" chunk was a music_chunk
+				   which had only a tag, but no music
+				   data - we cannot cross-fade that;
+				   but since this happens only at the
+				   beginning of the new song, we can
+				   easily recover by throwing it away
+				   now */
+				music_buffer_return(player_buffer,
+						    other_chunk);
+				other_chunk = NULL;
+			}
+
 			chunk->other = other_chunk;
 		} else {
 			/* there are not enough decoded chunks yet */
@@ -732,9 +736,7 @@ play_next_chunk(struct player *player)
 
 	/* play the current chunk */
 
-	success = play_chunk(player->song, chunk, &player->play_audio_format);
-
-	if (!success) {
+	if (!play_chunk(player->song, chunk, &player->play_audio_format)) {
 		music_buffer_return(player_buffer, chunk);
 
 		player_lock();
@@ -776,11 +778,9 @@ play_next_chunk(struct player *player)
 static bool
 player_song_border(struct player *player)
 {
-	char *uri;
-
 	player->xfade = XFADE_UNKNOWN;
 
-	uri = song_get_uri(player->song);
+	char *uri = song_get_uri(player->song);
 	g_message("played \"%s\"", uri);
 	g_free(uri);
 
@@ -875,16 +875,17 @@ static void do_play(struct decoder_control *dc)
 
 		if (player.decoder_starting) {
 			/* wait until the decoder is initialized completely */
-			bool success;
-			const struct song *song;
 
-			success = player_check_decoder_startup(&player);
-			if (!success)
+			if (!player_check_decoder_startup(&player))
 				break;
 
 			/* seek to the beginning of the range */
-			song = decoder_current_song(dc);
+			const struct song *song = decoder_current_song(dc);
 			if (song != NULL && song->start_ms > 0 &&
+			    /* we must not send a seek command until
+			       the decoder is initialized
+			       completely */
+			    !player.decoder_starting &&
 			    !dc_seek(dc, song->start_ms / 1000.0))
 				player_dc_stop(&player);
 
@@ -1092,10 +1093,9 @@ static gpointer player_task(G_GNUC_UNUSED gpointer arg)
 
 void player_create(void)
 {
-	GError *e = NULL;
-
 	assert(pc.thread == NULL);
 
+	GError *e = NULL;
 	pc.thread = g_thread_create(player_task, NULL, true, &e);
 	if (pc.thread == NULL)
 		MPD_ERROR("Failed to spawn player task: %s", e->message);

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/poison.h

@@ -47,7 +47,7 @@ poison_noaccess(void *p, size_t length)
 	memset(p, 0x01, length);
 
 #ifdef HAVE_VALGRIND_MEMCHECK_H
-	VALGRIND_MAKE_MEM_NOACCESS(p, length);
+	(void)VALGRIND_MAKE_MEM_NOACCESS(p, length);
 #endif
 #endif
 }
@@ -68,7 +68,7 @@ poison_undefined(void *p, size_t length)
 	memset(p, 0x02, length);
 
 #ifdef HAVE_VALGRIND_MEMCHECK_H
-	VALGRIND_MAKE_MEM_UNDEFINED(p, length);
+	(void)VALGRIND_MAKE_MEM_UNDEFINED(p, length);
 #endif
 #endif
 }

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);