mpd version 0.17.3

Track whether _finish() has been called, and deliver all partial
results then.  Fixes Mantis ticket 0003621.

.gitignore

@@ -67,3 +67,7 @@ test/run_ntp_server
 test/run_resolver
 test/run_tcp_connect
 test/test_pcm
+test/dump_rva2
+test/dump_text_file
+test/test_byte_reverse
+test/test_vorbis_encoder

INSTALL

@@ -103,7 +103,7 @@ libsidplay2 - http://sidplay2.sourceforge.net/
 For C64 SID support.
 
 libfluidsynth - http://fluidsynth.resonance.org/
-For MIDI support (DO NOT USE - use libwildmidi instead)
+For MIDI support.
 
 libwildmidi - http://wildmidi.sourceforge.net/
 For MIDI support.

Makefile.am

@@ -227,6 +227,7 @@ src_mpd_SOURCES = \
 	$(OUTPUT_API_SRC) \
 	$(MIXER_API_SRC) \
 	src/glib_socket.h \
+	src/clock.c src/clock.h \
 	src/notify.c \
 	src/audio_config.c src/audio_config.h \
 	src/audio_check.c \
@@ -1069,7 +1070,6 @@ test_dump_playlist_SOURCES = test/dump_playlist.c \
 	src/audio_check.c src/pcm_buffer.c \
 	src/text_input_stream.c src/fifo_buffer.c \
 	src/cue/cue_parser.c src/cue/cue_parser.h \
-	src/timer.c \
 	src/fd_util.c
 
 if HAVE_FLAC
@@ -1096,7 +1096,6 @@ test_run_decoder_SOURCES = test/run_decoder.c \
 	src/fd_util.c \
 	src/audio_check.c \
 	src/audio_format.c \
-	src/timer.c \
 	$(ARCHIVE_SRC) \
 	$(INPUT_SRC) \
 	$(TAG_SRC) \
@@ -1118,7 +1117,6 @@ test_read_tags_SOURCES = test/read_tags.c \
 	src/uri.c \
 	src/fd_util.c \
 	src/audio_check.c \
-	src/timer.c \
 	$(DECODER_SRC)
 
 if HAVE_ID3TAG
@@ -1240,7 +1238,7 @@ test_run_output_SOURCES = test/run_output.c \
 	src/audio_check.c \
 	src/audio_format.c \
 	src/audio_parser.c \
-	src/timer.c \
+	src/timer.c src/clock.c \
 	src/tag.c src/tag_pool.c \
 	src/fifo_buffer.c src/growing_fifo.c \
 	src/page.c \

NEWS

@@ -1,3 +1,40 @@
+ver 0.17.3 (2013/01/06)
+* output:
+  - osx: fix pops during playback
+  - recorder: fix I/O error check
+  - shout: fix memory leak in error handler
+  - recorder, shout: support Ogg packets that span more than one page
+* decoder:
+  - ffmpeg: ignore negative time stamps
+  - ffmpeg: support planar audio
+* playlist:
+  - cue: fix memory leak
+  - cue: fix CUE files with only one track
+
+ver 0.17.2 (2012/09/30)
+* protocol:
+  - fix crash in local file check
+* decoder:
+  - fluidsynth: remove throttle (requires libfluidsynth 1.1)
+  - fluidsynth: stop playback at end of file
+  - fluidsynth: check MIDI file format while scanning
+  - fluidsynth: add sample rate setting
+  - wavpack: support all APEv2 tags
+* output:
+  - httpd: use monotonic clock, avoid hiccups after system clock adjustment
+  - httpd: fix throttling bug after resuming playback
+* playlist:
+  - cue: map "PERFORMER" to "artist" or "album artist"
+* mapper: fix non-UTF8 music directory name
+* mapper: fix potential crash in file permission check
+* playlist: fix use-after-free bug
+* playlist: fix memory leak
+* state_file: save song priorities
+* player: disable cross-fading in "single" mode
+* update: fix unsafe readlink() usage
+* configure.ac:
+  - don't auto-detect the vorbis encoder when Tremor is enabled
+
 ver 0.17.1 (2012/07/31)
 * protocol:
   - require appropriate permissions for searchadd{,pl}

configure.ac

@@ -1,6 +1,6 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.17.1, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.17.3, musicpd-dev-team@lists.sourceforge.net)
 
 VERSION_MAJOR=0
 VERSION_MINOR=17
@@ -217,8 +217,8 @@ AC_ARG_ENABLE(flac,
 
 AC_ARG_ENABLE(fluidsynth,
 	AS_HELP_STRING([--enable-fluidsynth],
-		[enable MIDI support via fluidsynth (default: disable)]),,
-	enable_fluidsynth=no)
+		[enable MIDI support via fluidsynth (default: auto)]),,
+	enable_fluidsynth=auto)
 
 AC_ARG_ENABLE(gme,
 	AS_HELP_STRING([--enable-gme],
@@ -845,7 +845,7 @@ enable_flac_encoder=$enable_flac
 
 dnl -------------------------------- FluidSynth -------------------------------
 if test x$enable_fluidsynth = xyes; then
-	PKG_CHECK_MODULES(FLUIDSYNTH, [fluidsynth],
+	PKG_CHECK_MODULES(FLUIDSYNTH, [fluidsynth >= 1.1],
 		AC_DEFINE(ENABLE_FLUIDSYNTH, 1, [Define for fluidsynth support]),
 		enable_fluidsynth=no)
 fi
@@ -999,6 +999,11 @@ if test x$enable_tremor = xyes; then
 		AC_MSG_WARN(["OggTremor detected, could not enable Vorbis."])
 	fi
 	enable_vorbis=no
+
+	if test x$enable_vorbis_encoder = xauto; then
+		AC_MSG_WARN([OggTremor detected, disabling the Vorbis encoder plugin.])
+		enable_vorbis_encoder=no
+	fi
 fi
 
 MPD_AUTO_PKG(vorbis, VORBIS, [vorbis vorbisfile ogg],

doc/mpd.conf.5

@@ -216,7 +216,7 @@ default is 5.
 .TP
 .B max_playlist_length <number>
 This specifies the maximum number of songs that can be in the playlist.  The
-default is 4096.
+default is 16384.
 .TP
 .B max_command_list_size <size in KiB>
 This specifies the maximum size a command list can be.  The default is 2048.
@@ -252,11 +252,12 @@ when saving playlists.  The default is "no".
 This specifies the tag types that will be scanned for and made available to
 clients.  Note that you must recreate (not update) your database for changes to
 this parameter to take effect.  Possible values are artist, album, title,
-track, name, genre, date, composer, performer, comment, and disc.  Multiple
-tags may be specified as a comma separated list.  An example value is
-"artist,album,title,track".  The special value "none" may be used alone to
-disable all metadata.  The default is to use all known tag types except for
-comments.
+track, name, genre, date, composer, performer, comment, disc,
+musicbrainz_artistid, musicbrainz_albumid, musicbrainz_albumartistid,
+musicbrainz_trackid.  Multiple tags may be specified as a comma separated list.
+An example value is "artist,album,title,track".  The special value "none" may
+be used alone to disable all metadata.  The default is to use all known tag
+types except for comments and those starting with "musicbrainz".
 .TP
 .B auto_update <yes or no>
 This specifies the wheter to support automatic update of music database when

doc/mpdconf.example

@@ -113,10 +113,9 @@
 #
 #save_absolute_paths_in_playlists	"no"
 #
-# This setting defines a list of tag types that will be extracted during the 
-# audio file discovery process. Optionally, 'comment' can be added to this
-# list.
-#
+# This setting defines a list of tag types that will be extracted during the
+# audio file discovery process. The complete list of possible values can be
+# found in the mpd.conf man page.
 #metadata_to_use	"artist,album,title,track,name,genre,date,composer,performer,disc"
 #
 # This setting enables automatic update of MPD's database when files in 

doc/user.xml

@@ -818,6 +818,46 @@ systemctl start mpd.socket</programlisting>
 
       </section>
 
+      <section>
+        <title><varname>fluidsynth</varname></title>
+
+        <para>
+          MIDI decoder based on libfluidsynth.
+        </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>sample_rate</varname>
+                </entry>
+                <entry>
+                  The sample rate that shall be synthesized by the
+                  plugin.  Defaults to 48000.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>soundfont</varname>
+                </entry>
+                <entry>
+                  The absolute path of the soundfont file.  Defaults
+                  to
+                  <filename>/usr/share/sounds/sf2/FluidR3_GM.sf2</filename>.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
+      </section>
+
       <section>
         <title><varname>mikmod</varname></title>
 
@@ -847,6 +887,37 @@ systemctl start mpd.socket</programlisting>
           </tgroup>
         </informaltable>
       </section>
+
+      <section>
+        <title><varname>wildmidi</varname></title>
+
+        <para>
+          MIDI decoder based on libwildmidi.
+        </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>config_file</varname>
+                </entry>
+                <entry>
+                  The absolute path of the timidity config file.  Defaults
+                  to
+                  <filename>/etc/timidity/timidity.cfg</filename>.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
+      </section>
     </section>
 
     <section>

m4/mpd_auto.m4

@@ -73,7 +73,8 @@ AC_DEFUN([MPD_AUTO_PKG_LIB], [
 			[eval "found_$1=yes"],
 			AC_CHECK_LIB($4, $5,
 				[eval "found_$1=yes $2_LIBS='$6' $2_CFLAGS='$7'"],
-				[eval "found_$1=no"]))
+				[eval "found_$1=no"],
+				[$6]))
 	fi
 
 	MPD_AUTO_RESULT([$1], [$8], [$9])

mpd.service.in

@@ -1,6 +1,6 @@
 [Unit]
 Description=Music Player Daemon
-After=sound.target
+After=network.target sound.target
 
 [Service]
 ExecStart=@prefix@/bin/mpd --no-daemon

src/AudioCompress/compress.c

@@ -33,6 +33,9 @@ struct Compressor {
 struct Compressor *Compressor_new(unsigned int history)
 {
 	struct Compressor *obj = malloc(sizeof(struct Compressor));
+	if (obj == NULL)
+		/* out of memory, not much we can do */
+		abort();
 
 	obj->prefs.target = TARGET;
 	obj->prefs.maxgain = GAINMAX;
@@ -61,6 +64,10 @@ void Compressor_delete(struct Compressor *obj)
 static int *resizeArray(int *data, int newsz, int oldsz)
 {
         data = realloc(data, newsz*sizeof(int));
+	if (data == NULL)
+		/* out of memory, not much we can do */
+		abort();
+
         if (newsz > oldsz)
                 memset(data + oldsz, 0, sizeof(int)*(newsz - oldsz));
         return data;

src/client_file.h

@@ -35,7 +35,6 @@ struct client;
  * @param path_fs the absolute path name in filesystem encoding
  * @return true if access is allowed
  */
-G_GNUC_PURE
 bool
 client_allow_file(const struct client *client, const char *path_fs,
 		  GError **error_r);

src/clock.c

@@ -0,0 +1,95 @@
+/*
+ * Copyright (C) 2003-2012 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include "clock.h"
+
+#ifdef WIN32
+#include <windows.h>
+#elif defined(__APPLE__)
+#include <mach/mach_time.h>
+#else
+#include <time.h>
+#endif
+
+unsigned
+monotonic_clock_ms(void)
+{
+#ifdef WIN32
+	return GetTickCount();
+#elif defined(__APPLE__) /* OS X does not define CLOCK_MONOTONIC */
+	static mach_timebase_info_data_t base;
+	if (base.denom == 0)
+		(void)mach_timebase_info(&base);
+
+	return (unsigned)((mach_absolute_time() * base.numer)
+			  / (1000000 * base.denom));
+#elif defined(CLOCK_MONOTONIC)
+	struct timespec ts;
+	clock_gettime(CLOCK_MONOTONIC, &ts);
+	return ts.tv_sec * 1000 + ts.tv_nsec / 1000000;
+#else
+	/* we have no monotonic clock, fall back to gettimeofday() */
+	struct timeval tv;
+	gettimeofday(&tv, 0);
+	return tv.tv_sec * 1000 + tv.tv_usec / 1000;
+#endif
+}
+
+uint64_t
+monotonic_clock_us(void)
+{
+#ifdef WIN32
+	LARGE_INTEGER l_value, l_frequency;
+
+	if (!QueryPerformanceCounter(&l_value) ||
+	    !QueryPerformanceFrequency(&l_frequency))
+		return 0;
+
+	uint64_t value = l_value.QuadPart;
+	uint64_t frequency = l_frequency.QuadPart;
+
+	if (frequency > 1000000) {
+		value *= 10000;
+		value /= frequency / 100;
+	} else if (frequency < 1000000) {
+		value *= 10000;
+		value /= frequency;
+		value *= 100;
+	}
+
+	return value;
+#elif defined(__APPLE__) /* OS X does not define CLOCK_MONOTONIC */
+	static mach_timebase_info_data_t base;
+	if (base.denom == 0)
+		(void)mach_timebase_info(&base);
+
+	return ((uint64_t)mach_absolute_time() * (uint64_t)base.numer)
+		/ (1000 * (uint64_t)base.denom);
+#elif defined(CLOCK_MONOTONIC)
+	struct timespec ts;
+	clock_gettime(CLOCK_MONOTONIC, &ts);
+	return (uint64_t)ts.tv_sec * 1000000 + (uint64_t)(ts.tv_nsec / 1000);
+#else
+	/* we have no monotonic clock, fall back to gettimeofday() */
+	struct timeval tv;
+	gettimeofday(&tv, 0);
+	return (uint64_t)tv.tv_sec * 1000 + (uint64_t)(tv.tv_usec) / 1000(;
+#endif
+}
+

src/clock.h

@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2003-2012 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_CLOCK_H
+#define MPD_CLOCK_H
+
+#include <glib.h>
+
+#include <stdint.h>
+
+/**
+ * Returns the value of a monotonic clock in milliseconds.
+ */
+G_GNUC_PURE
+unsigned
+monotonic_clock_ms(void);
+
+/**
+ * Returns the value of a monotonic clock in microseconds.
+ */
+G_GNUC_PURE
+uint64_t
+monotonic_clock_us(void);
+
+#endif

src/cmdline.c

@@ -213,12 +213,12 @@ parse_cmdline(int argc, char **argv, struct options *options,
 				if(g_file_test(system_path,
 						G_FILE_TEST_IS_REGULAR)) {
 					ret = config_read_file(system_path,error_r);
+					g_free(system_path);
 					break;
-				}
-				++i;;
+				} else
+					g_free(system_path);
+				++i;
 			}
-			g_free(system_path);
-			g_free(&system_config_dirs);
 		}
 #else /* G_OS_WIN32 */
 		char *path2;

src/command.c

@@ -1069,7 +1069,7 @@ handle_next(G_GNUC_UNUSED struct client *client,
 {
 	/* single mode is not considered when this is user who
 	 * wants to change song. */
-	int single = g_playlist.queue.single;
+	const bool single = g_playlist.queue.single;
 	g_playlist.queue.single = false;
 
 	playlist_next(&g_playlist, client->player_control);
@@ -1544,7 +1544,7 @@ handle_config(struct client *client,
 		return COMMAND_RETURN_ERROR;
 	}
 
-	const char *path = mapper_get_music_directory();
+	const char *path = mapper_get_music_directory_utf8();
 	if (path != NULL)
 		client_printf(client, "music_directory: %s\n", path);
 

src/cue/cue_parser.c

@@ -58,9 +58,35 @@ struct cue_parser {
 
 	char *filename;
 
-	struct song *current, *previous, *finished;
+	/**
+	 * The song currently being edited.
+	 */
+	struct song *current;
 
+	/**
+	 * The previous song.  It is remembered because its end_time
+	 * will be set to the current song's start time.
+	 */
+	struct song *previous;
+
+	/**
+	 * A song that is completely finished and can be returned to
+	 * the caller via cue_parser_get().
+	 */
+	struct song *finished;
+
+	/**
+	 * Set to true after previous.end_time has been updated to the
+	 * start time of the current song.
+	 */
 	bool last_updated;
+
+	/**
+	 * Tracks whether cue_parser_finish() has been called.  If
+	 * true, then all remaining (partial) results will be
+	 * delivered by cue_parser_get().
+	 */
+	bool end;
 };
 
 struct cue_parser *
@@ -73,6 +99,7 @@ cue_parser_new(void)
 	parser->current = NULL;
 	parser->previous = NULL;
 	parser->finished = NULL;
+	parser->end = false;
 	return parser;
 }
 
@@ -85,6 +112,9 @@ cue_parser_free(struct cue_parser *parser)
 	if (parser->current != NULL)
 		song_free(parser->current);
 
+	if (parser->previous != NULL)
+		song_free(parser->previous);
+
 	if (parser->finished != NULL)
 		song_free(parser->finished);
 
@@ -201,10 +231,32 @@ cue_parse_position(const char *p)
 	return minutes * 60000 + seconds * 1000 + frames * 1000 / 75;
 }
 
+/**
+ * Commit the current song.  It will be moved to "previous", so the
+ * next song may soon edit its end time (using the next song's start
+ * time).
+ */
+static void
+cue_parser_commit(struct cue_parser *parser)
+{
+	/* the caller of this library must call cue_parser_get() often
+	   enough */
+	assert(parser->finished == NULL);
+	assert(!parser->end);
+
+	if (parser->current == NULL)
+		return;
+
+	parser->finished = parser->previous;
+	parser->previous = parser->current;
+	parser->current = NULL;
+}
+
 static void
 cue_parser_feed2(struct cue_parser *parser, char *p)
 {
 	assert(parser != NULL);
+	assert(!parser->end);
 	assert(p != NULL);
 
 	const char *command = cue_next_token(&p);
@@ -216,16 +268,26 @@ cue_parser_feed2(struct cue_parser *parser, char *p)
 		if (tag != NULL)
 			cue_parse_rem(p, tag);
 	} else if (strcmp(command, "PERFORMER") == 0) {
+		/* MPD knows a "performer" tag, but it is not a good
+		   match for this CUE tag; from the Hydrogenaudio
+		   Knowledgebase: "At top-level this will specify the
+		   CD artist, while at track-level it specifies the
+		   track artist." */
+
+		enum tag_type type = parser->state == TRACK
+			? TAG_ARTIST
+			: TAG_ALBUM_ARTIST;
+
 		struct tag *tag = cue_current_tag(parser);
 		if (tag != NULL)
-			cue_add_tag(tag, TAG_PERFORMER, p);
+			cue_add_tag(tag, type, p);
 	} else if (strcmp(command, "TITLE") == 0) {
 		if (parser->state == HEADER)
 			cue_add_tag(parser->tag, TAG_ALBUM, p);
 		else if (parser->state == TRACK)
 			cue_add_tag(parser->current->tag, TAG_TITLE, p);
 	} else if (strcmp(command, "FILE") == 0) {
-		cue_parser_finish(parser);
+		cue_parser_commit(parser);
 
 		const char *filename = cue_next_value(&p);
 		if (filename == NULL)
@@ -248,7 +310,7 @@ cue_parser_feed2(struct cue_parser *parser, char *p)
 	} else if (parser->state == IGNORE_FILE) {
 		return;
 	} else if (strcmp(command, "TRACK") == 0) {
-		cue_parser_finish(parser);
+		cue_parser_commit(parser);
 
 		const char *nr = cue_next_token(&p);
 		if (nr == NULL)
@@ -300,6 +362,7 @@ void
 cue_parser_feed(struct cue_parser *parser, const char *line)
 {
 	assert(parser != NULL);
+	assert(!parser->end);
 	assert(line != NULL);
 
 	char *allocated = g_strdup(line);
@@ -310,12 +373,12 @@ cue_parser_feed(struct cue_parser *parser, const char *line)
 void
 cue_parser_finish(struct cue_parser *parser)
 {
-	if (parser->finished != NULL)
-		song_free(parser->finished);
+	if (parser->end)
+		/* has already been called, ignore */
+		return;
 
-	parser->finished = parser->previous;
-	parser->previous = parser->current;
-	parser->current = NULL;
+	cue_parser_commit(parser);
+	parser->end = true;
 }
 
 struct song *
@@ -323,6 +386,15 @@ cue_parser_get(struct cue_parser *parser)
 {
 	assert(parser != NULL);
 
+	if (parser->finished == NULL && parser->end) {
+		/* cue_parser_finish() has been called already:
+		   deliver all remaining (partial) results */
+		assert(parser->current == NULL);
+
+		parser->finished = parser->previous;
+		parser->previous = NULL;
+	}
+
 	struct song *song = parser->finished;
 	parser->finished = NULL;
 	return song;

src/decoder/_flac_common.c

@@ -124,11 +124,10 @@ void flac_metadata_common_cb(const FLAC__StreamMetadata * block,
 	case FLAC__METADATA_TYPE_VORBIS_COMMENT:
 		if (flac_parse_replay_gain(&rgi, block))
 			replay_gain_db = decoder_replay_gain(data->decoder, &rgi);
-		if (flac_parse_mixramp(&mixramp_start, &mixramp_end, block)) {
-			g_debug("setting mixramp_tags");
+
+		if (flac_parse_mixramp(&mixramp_start, &mixramp_end, block))
 			decoder_mixramp(data->decoder, replay_gain_db,
 					mixramp_start, mixramp_end);
-		}
 
 		if (data->tag != NULL)
 			flac_vorbis_comments_to_tag(data->tag, NULL,

src/decoder/_ogg_common.c

@@ -33,12 +33,14 @@ ogg_stream_type ogg_stream_type_detect(struct input_stream *inStream)
 	size_t r;
 
 	r = decoder_read(NULL, inStream, buf, sizeof(buf));
-	if (r >= 32 && memcmp(buf, "OggS", 4) == 0 && (
-				(memcmp(buf+29, "FLAC", 4) == 0
-					&& memcmp(buf+37, "fLaC", 4) == 0)
-				|| (memcmp(buf+28, "FLAC", 4) == 0)
-				|| (memcmp(buf+28, "fLaC", 4) == 0))) {
+	if (r < sizeof(buf) || memcmp(buf, "OggS", 4) != 0)
+		return VORBIS;
+
+	if ((memcmp(buf + 29, "FLAC", 4) == 0 &&
+	     memcmp(buf + 37, "fLaC", 4) == 0) ||
+	    memcmp(buf + 28, "FLAC", 4) == 0 ||
+	    memcmp(buf + 28, "fLaC", 4) == 0)
 		return FLAC;
-	}
+
 	return VORBIS;
 }

src/decoder/ffmpeg_decoder_plugin.c

@@ -234,6 +234,21 @@ time_to_ffmpeg(double t, const AVRational time_base)
 }
 
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(53,25,0)
+
+static void
+copy_interleave_frame2(uint8_t *dest, uint8_t **src,
+		       unsigned nframes, unsigned nchannels,
+		       unsigned sample_size)
+{
+	for (unsigned frame = 0; frame < nframes; ++frame) {
+		for (unsigned channel = 0; channel < nchannels; ++channel) {
+			memcpy(dest, src[channel] + frame * sample_size,
+			       sample_size);
+			dest += sample_size;
+		}
+	}
+}
+
 /**
  * Copy PCM data from a AVFrame to an interleaved buffer.
  */
@@ -254,11 +269,10 @@ copy_interleave_frame(const AVCodecContext *codec_context,
 
 	if (av_sample_fmt_is_planar(codec_context->sample_fmt) &&
 	    codec_context->channels > 1) {
-		for (int i = 0, channels = codec_context->channels;
-		     i < channels; i++) {
-			memcpy(buffer, frame->extended_data[i], plane_size);
-			buffer += plane_size;
-		}
+		copy_interleave_frame2(buffer, frame->extended_data,
+				       frame->nb_samples,
+				       codec_context->channels,
+				       av_get_bytes_per_sample(codec_context->sample_fmt));
 	} else {
 		memcpy(buffer, frame->extended_data[0], data_size);
 	}
@@ -273,7 +287,7 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 		   AVCodecContext *codec_context,
 		   const AVRational *time_base)
 {
-	if (packet->pts != (int64_t)AV_NOPTS_VALUE)
+	if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE)
 		decoder_timestamp(decoder,
 				  time_from_ffmpeg(packet->pts, *time_base));
 
@@ -352,12 +366,20 @@ ffmpeg_send_packet(struct decoder *decoder, struct input_stream *is,
 	return cmd;
 }
 
+#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(52, 94, 1)
+#define AVSampleFormat SampleFormat
+#endif
+
+G_GNUC_CONST
 static enum sample_format
-ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
+ffmpeg_sample_format(enum AVSampleFormat sample_fmt)
 {
-	switch (codec_context->sample_fmt) {
+	switch (sample_fmt) {
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
 	case AV_SAMPLE_FMT_S16:
+#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0)
+	case AV_SAMPLE_FMT_S16P:
+#endif
 #else
 	case SAMPLE_FMT_S16:
 #endif
@@ -365,16 +387,30 @@ ffmpeg_sample_format(G_GNUC_UNUSED const AVCodecContext *codec_context)
 
 #if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
 	case AV_SAMPLE_FMT_S32:
+#if LIBAVUTIL_VERSION_INT >= AV_VERSION_INT(51,17,0)
+	case AV_SAMPLE_FMT_S32P:
+#endif
 #else
 	case SAMPLE_FMT_S32:
 #endif
 		return SAMPLE_FORMAT_S32;
 
 	default:
-		g_warning("Unsupported libavcodec SampleFormat value: %d",
-			  codec_context->sample_fmt);
-		return SAMPLE_FORMAT_UNDEFINED;
+		break;
 	}
+
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(52, 94, 1)
+	char buffer[64];
+	const char *name = av_get_sample_fmt_string(buffer, sizeof(buffer),
+						    sample_fmt);
+	if (name != NULL)
+		g_warning("Unsupported libavcodec SampleFormat value: %s (%d)",
+			  name, sample_fmt);
+	else
+#endif
+		g_warning("Unsupported libavcodec SampleFormat value: %d",
+			  sample_fmt);
+	return SAMPLE_FORMAT_UNDEFINED;
 }
 
 static AVInputFormat *
@@ -485,11 +521,16 @@ ffmpeg_decode(struct decoder *decoder, struct input_stream *input)
 		return;
 	}
 
+	const enum sample_format sample_format =
+		ffmpeg_sample_format(codec_context->sample_fmt);
+	if (sample_format == SAMPLE_FORMAT_UNDEFINED)
+		return;
+
 	GError *error = NULL;
 	struct audio_format audio_format;
 	if (!audio_format_init_checked(&audio_format,
 				       codec_context->sample_rate,
-				       ffmpeg_sample_format(codec_context),
+				       sample_format,
 				       codec_context->channels, &error)) {
 		g_warning("%s", error->message);
 		g_error_free(error);

src/decoder/fluidsynth_decoder_plugin.c

@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2012 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,18 +17,9 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-/*
- * WARNING!  This plugin suffers from major shortcomings in the
- * libfluidsynth API, which render it practically unusable.  For a
- * discussion, see the post on the fluidsynth mailing list:
- *
- * http://www.mail-archive.com/fluid-dev@nongnu.org/msg01099.html
- *
- */
-
 #include "config.h"
 #include "decoder_api.h"
-#include "timer.h"
+#include "audio_check.h"
 #include "conf.h"
 
 #include <glib.h>
@@ -38,6 +29,9 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "fluidsynth"
 
+static unsigned sample_rate;
+static const char *soundfont_path;
+
 /**
  * Convert a fluidsynth log level to a GLib log level.
  */
@@ -75,8 +69,21 @@ fluidsynth_mpd_log_function(int level, char *message, G_GNUC_UNUSED void *data)
 }
 
 static bool
-fluidsynth_init(G_GNUC_UNUSED const struct config_param *param)
+fluidsynth_init(const struct config_param *param)
 {
+	GError *error = NULL;
+
+	sample_rate = config_get_block_unsigned(param, "sample_rate", 48000);
+	if (!audio_check_sample_rate(sample_rate, &error)) {
+		g_warning("%s\n", error->message);
+		g_error_free(error);
+		return false;
+	}
+
+	soundfont_path =
+		config_get_block_string(param, "soundfont",
+					"/usr/share/sounds/sf2/FluidR3_GM.sf2");
+
 	fluid_set_log_function(LAST_LOG_LEVEL,
 			       fluidsynth_mpd_log_function, NULL);
 
@@ -86,36 +93,24 @@ fluidsynth_init(G_GNUC_UNUSED const struct config_param *param)
 static void
 fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
 {
-	static const struct audio_format audio_format = {
-		.sample_rate = 48000,
-		.format = SAMPLE_FORMAT_S16,
-		.channels = 2,
-	};
 	char setting_sample_rate[] = "synth.sample-rate";
 	/*
 	char setting_verbose[] = "synth.verbose";
 	char setting_yes[] = "yes";
 	*/
-	const char *soundfont_path;
 	fluid_settings_t *settings;
 	fluid_synth_t *synth;
 	fluid_player_t *player;
-	char *path_dup;
 	int ret;
-	struct timer *timer;
 	enum decoder_command cmd;
 
-	soundfont_path =
-		config_get_string("soundfont",
-				  "/usr/share/sounds/sf2/FluidR3_GM.sf2");
-
 	/* set up fluid settings */
 
 	settings = new_fluid_settings();
 	if (settings == NULL)
 		return;
 
-	fluid_settings_setnum(settings, setting_sample_rate, 48000);
+	fluid_settings_setnum(settings, setting_sample_rate, sample_rate);
 
 	/*
 	fluid_settings_setstr(settings, setting_verbose, setting_yes);
@@ -146,11 +141,7 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
 		return;
 	}
 
-	/* temporarily duplicate the path_fs string, because
-	   fluidsynth wants a writable string */
-	path_dup = g_strdup(path_fs);
-	ret = fluid_player_add(player, path_dup);
-	g_free(path_dup);
+	ret = fluid_player_add(player, path_fs);
 	if (ret != 0) {
 		g_warning("fluid_player_add() failed");
 		delete_fluid_player(player);
@@ -170,47 +161,34 @@ fluidsynth_file_decode(struct decoder *decoder, const char *path_fs)
 		return;
 	}
 
-	/* set up a timer for synchronization; fluidsynth always
-	   decodes in real time, which forces us to synchronize */
-	/* XXX is there any way to switch off real-time decoding? */
-
-	timer = timer_new(&audio_format);
-	timer_start(timer);
-
 	/* initialization complete - announce the audio format to the
 	   MPD core */
 
+	struct audio_format audio_format;
+	audio_format_init(&audio_format, sample_rate, SAMPLE_FORMAT_S16, 2);
 	decoder_initialized(decoder, &audio_format, false, -1);
 
-	do {
+	while (fluid_player_get_status(player) == FLUID_PLAYER_PLAYING) {
 		int16_t buffer[2048];
 		const unsigned max_frames = G_N_ELEMENTS(buffer) / 2;
 
-		/* synchronize with the fluid player */
-
-		timer_add(timer, sizeof(buffer));
-		timer_sync(timer);
-
 		/* read samples from fluidsynth and send them to the
 		   MPD core */
 
 		ret = fluid_synth_write_s16(synth, max_frames,
 					    buffer, 0, 2,
 					    buffer, 1, 2);
-		/* XXX how do we see whether the player is done?  We
-		   can't access the private attribute
-		   player->status */
 		if (ret != 0)
 			break;
 
 		cmd = decoder_data(decoder, NULL, buffer, sizeof(buffer),
 				   0);
-	} while (cmd == DECODE_COMMAND_NONE);
+		if (cmd != DECODE_COMMAND_NONE)
+			break;
+	}
 
 	/* clean up */
 
-	timer_free(timer);
-
 	fluid_player_stop(player);
 	fluid_player_join(player);
 
@@ -224,10 +202,7 @@ fluidsynth_scan_file(const char *file,
 		     G_GNUC_UNUSED const struct tag_handler *handler,
 		     G_GNUC_UNUSED void *handler_ctx)
 {
-	/* to be implemented */
-	(void)file;
-
-	return true;
+	return fluid_is_midifile(file);
 }
 
 static const char *const fluidsynth_suffixes[] = {

src/decoder/mad_decoder_plugin.c

@@ -365,11 +365,10 @@ static void mp3_parse_id3(struct mp3_data *data, size_t tagsize,
 			replay_gain_db = decoder_replay_gain(data->decoder, &rgi);
 			data->found_replay_gain = true;
 		}
-		if (parse_id3_mixramp(&mixramp_start, &mixramp_end, id3_tag)) {
-			g_debug("setting mixramp_tags");
+
+		if (parse_id3_mixramp(&mixramp_start, &mixramp_end, id3_tag))
 			decoder_mixramp(data->decoder, replay_gain_db,
 					mixramp_start, mixramp_end);
-		}
 	}
 
 	id3_tag_delete(id3_tag);

src/decoder/wavpack_decoder_plugin.c

@@ -24,6 +24,7 @@
 #include "utils.h"
 #include "tag_table.h"
 #include "tag_handler.h"
+#include "tag_ape.h"
 
 #include <wavpack/wavpack.h>
 #include <glib.h>
@@ -38,21 +39,6 @@
 
 #define ERRORLEN 80
 
-static const struct tag_table wavpack_tags[] = {
-	{ "artist", TAG_ARTIST },
-	{ "album", TAG_ALBUM },
-	{ "title", TAG_TITLE },
-	{ "track", TAG_TRACK },
-	{ "name", TAG_NAME },
-	{ "genre", TAG_GENRE },
-	{ "date", TAG_DATE },
-	{ "composer", TAG_COMPOSER },
-	{ "performer", TAG_PERFORMER },
-	{ "comment", TAG_COMMENT },
-	{ "disc", TAG_DISC },
-	{ NULL, TAG_NUM_OF_ITEM_TYPES }
-};
-
 /** A pointer type for format converter function. */
 typedef void (*format_samples_t)(
 	int bytes_per_sample,
@@ -321,7 +307,17 @@ wavpack_scan_file(const char *fname,
 				    WavpackGetNumSamples(wpc) /
 				    WavpackGetSampleRate(wpc));
 
-	for (const struct tag_table *i = wavpack_tags; i->name != NULL; ++i)
+	/* the WavPack format implies APEv2 tags, which means we can
+	   reuse the mapping from tag_ape.c */
+
+	for (unsigned i = 0; i < TAG_NUM_OF_ITEM_TYPES; ++i) {
+		const char *name = tag_item_names[i];
+		if (name != NULL)
+			wavpack_scan_tag_item(wpc, name, (enum tag_type)i,
+					      handler, handler_ctx);
+	}
+
+	for (const struct tag_table *i = ape_tags; i->name != NULL; ++i)
 		wavpack_scan_tag_item(wpc, i->name, i->type,
 				      handler, handler_ctx);
 

src/decoder_control.c

@@ -169,7 +169,6 @@ dc_mixramp_start(struct decoder_control *dc, char *mixramp_start)
 
 	g_free(dc->mixramp_start);
 	dc->mixramp_start = mixramp_start;
-	g_debug("mixramp_start = %s", mixramp_start ? mixramp_start : "NULL");
 }
 
 void
@@ -179,7 +178,6 @@ dc_mixramp_end(struct decoder_control *dc, char *mixramp_end)
 
 	g_free(dc->mixramp_end);
 	dc->mixramp_end = mixramp_end;
-	g_debug("mixramp_end = %s", mixramp_end ? mixramp_end : "NULL");
 }
 
 void
@@ -189,5 +187,4 @@ dc_mixramp_prev_end(struct decoder_control *dc, char *mixramp_prev_end)
 
 	g_free(dc->mixramp_prev_end);
 	dc->mixramp_prev_end = mixramp_prev_end;
-	g_debug("mixramp_prev_end = %s", mixramp_prev_end ? mixramp_prev_end : "NULL");
 }

src/decoder_thread.c

@@ -468,7 +468,6 @@ decoder_task(gpointer arg)
 
 		switch (dc->command) {
 		case DECODE_COMMAND_START:
-			g_debug("clearing mixramp tags");
 			dc_mixramp_start(dc, NULL);
 			dc_mixramp_prev_end(dc, dc->mixramp_end);
 			dc->mixramp_end = NULL; /* Don't free, it's copied above. */

src/encoder/vorbis_encoder.c

@@ -65,13 +65,11 @@ static bool
 vorbis_encoder_configure(struct vorbis_encoder *encoder,
 			 const struct config_param *param, GError **error)
 {
-	const char *value;
-	char *endptr;
-
-	value = config_get_block_string(param, "quality", NULL);
+	const char *value = config_get_block_string(param, "quality", NULL);
 	if (value != NULL) {
 		/* a quality was configured (VBR) */
 
+		char *endptr;
 		encoder->quality = g_ascii_strtod(value, &endptr);
 
 		if (*endptr != '\0' || encoder->quality < -1.0 ||
@@ -103,8 +101,9 @@ vorbis_encoder_configure(struct vorbis_encoder *encoder,
 		}
 
 		encoder->quality = -2.0;
-		encoder->bitrate = g_ascii_strtoll(value, &endptr, 10);
 
+		char *endptr;
+		encoder->bitrate = g_ascii_strtoll(value, &endptr, 10);
 		if (*endptr != '\0' || encoder->bitrate <= 0) {
 			g_set_error(error, vorbis_encoder_quark(), 0,
 				    "bitrate at line %i should be a positive integer",
@@ -119,9 +118,7 @@ vorbis_encoder_configure(struct vorbis_encoder *encoder,
 static struct encoder *
 vorbis_encoder_init(const struct config_param *param, GError **error)
 {
-	struct vorbis_encoder *encoder;
-
-	encoder = g_new(struct vorbis_encoder, 1);
+	struct vorbis_encoder *encoder = g_new(struct vorbis_encoder, 1);
 	encoder_struct_init(&encoder->encoder, &vorbis_encoder_plugin);
 
 	/* load configuration from "param" */
@@ -211,14 +208,12 @@ vorbis_encoder_open(struct encoder *_encoder,
 		    GError **error)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
-	bool ret;
 
 	audio_format->format = SAMPLE_FORMAT_S16;
 
 	encoder->audio_format = *audio_format;
 
-	ret = vorbis_encoder_reinit(encoder, error);
-	if (!ret)
+	if (!vorbis_encoder_reinit(encoder, error))
 		return false;
 
 	vorbis_encoder_send_header(encoder);
@@ -251,11 +246,10 @@ static void
 vorbis_encoder_blockout(struct vorbis_encoder *encoder)
 {
 	while (vorbis_analysis_blockout(&encoder->vd, &encoder->vb) == 1) {
-		ogg_packet packet;
-
 		vorbis_analysis(&encoder->vb, NULL);
 		vorbis_bitrate_addblock(&encoder->vb);
 
+		ogg_packet packet;
 		while (vorbis_bitrate_flushpacket(&encoder->vd, &packet))
 			ogg_stream_packetin(&encoder->os, &packet);
 	}
@@ -344,9 +338,9 @@ vorbis_encoder_write(struct encoder *_encoder,
 		     G_GNUC_UNUSED GError **error)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
-	unsigned num_frames;
 
-	num_frames = length / audio_format_frame_size(&encoder->audio_format);
+	unsigned num_frames = length
+		/ audio_format_frame_size(&encoder->audio_format);
 
 	/* this is for only 16-bit audio */
 
@@ -364,12 +358,10 @@ static size_t
 vorbis_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
-	ogg_page page;
-	int ret;
 	unsigned char *dest = _dest;
-	size_t nbytes;
 
-	ret = ogg_stream_pageout(&encoder->os, &page);
+	ogg_page page;
+	int ret = ogg_stream_pageout(&encoder->os, &page);
 	if (ret == 0 && encoder->flush) {
 		encoder->flush = false;
 		ret = ogg_stream_flush(&encoder->os, &page);
@@ -381,7 +373,7 @@ vorbis_encoder_read(struct encoder *_encoder, void *_dest, size_t length)
 
 	assert(page.header_len > 0 || page.body_len > 0);
 
-	nbytes = (size_t)page.header_len + (size_t)page.body_len;
+	size_t nbytes = (size_t)page.header_len + (size_t)page.body_len;
 
 	if (nbytes > length)
 		/* XXX better error handling */

src/encoder_plugin.h

@@ -119,6 +119,10 @@ encoder_finish(struct encoder *encoder)
  * Before you free it, you must call encoder_close().  You may open
  * and close (reuse) one encoder any number of times.
  *
+ * After this function returns successfully and before the first
+ * encoder_write() call, you should invoke encoder_read() to obtain
+ * the file header.
+ *
  * @param encoder the encoder
  * @param audio_format the encoder's input audio format; the plugin
  * may modify the struct to adapt it to its abilities
@@ -291,6 +295,8 @@ encoder_write(struct encoder *encoder, const void *data, size_t length,
 /**
  * Reads encoded data from the encoder.
  *
+ * Call this repeatedly until no more data is returned.
+ *
  * @param encoder the encoder
  * @param dest the destination buffer to copy to
  * @param length the maximum length of the destination buffer

src/filter/volume_filter_plugin.c

@@ -26,7 +26,6 @@
 #include "pcm_buffer.h"
 #include "pcm_volume.h"
 #include "audio_format.h"
-#include "player_control.h"
 
 #include <assert.h>
 #include <string.h>

src/inotify_update.c

@@ -266,7 +266,7 @@ mpd_inotify_callback(int wd, unsigned mask,
 	    (mask & IN_ISDIR) != 0) {
 		/* a sub directory was changed: register those in
 		   inotify */
-		const char *root = mapper_get_music_directory();
+		const char *root = mapper_get_music_directory_fs();
 		const char *path_fs;
 		char *allocated = NULL;
 
@@ -308,7 +308,7 @@ mpd_inotify_init(unsigned max_depth)
 
 	g_debug("initializing inotify");
 
-	const char *path = mapper_get_music_directory();
+	const char *path = mapper_get_music_directory_fs();
 	if (path == NULL) {
 		g_debug("no music directory configured");
 		return;

src/input/ffmpeg_input_plugin.c

@@ -22,16 +22,13 @@
 #include "input_internal.h"
 #include "input_plugin.h"
 
+#include <libavutil/avutil.h>
 #include <libavformat/avio.h>
 #include <libavformat/avformat.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "input_ffmpeg"
 
-#ifndef AV_VERSION_INT
-#define AV_VERSION_INT(a, b, c) (a<<16 | b<<8 | c)
-#endif
-
 struct input_ffmpeg {
 	struct input_stream base;
 

src/input_stream.h

@@ -29,10 +29,6 @@
 #include <stdbool.h>
 #include <sys/types.h>
 
-#if !GLIB_CHECK_VERSION(2,14,0)
-typedef gint64 goffset;
-#endif
-
 struct input_stream {
 	/**
 	 * the plugin which implements this input stream

src/locate.c

@@ -127,7 +127,7 @@ locate_tag_search(const struct song *song, enum tag_type type, const char *str)
 {
 	bool ret = false;
 
-	if (type == LOCATE_TAG_FILE_TYPE || type == LOCATE_TAG_ANY_TYPE) {
+	if (type == LOCATE_TAG_FILE_TYPE || (int)type == LOCATE_TAG_ANY_TYPE) {
 		char *uri = song_get_uri(song);
 		char *p = g_utf8_casefold(uri, -1);
 		g_free(uri);
@@ -147,7 +147,7 @@ locate_tag_search(const struct song *song, enum tag_type type, const char *str)
 
 	for (unsigned i = 0; i < song->tag->num_items && !ret; i++) {
 		visited_types[song->tag->items[i]->type] = true;
-		if (type != LOCATE_TAG_ANY_TYPE &&
+		if ((int)type != LOCATE_TAG_ANY_TYPE &&
 		    song->tag->items[i]->type != type) {
 			continue;
 		}
@@ -185,7 +185,7 @@ locate_song_search(const struct song *song,
 static bool
 locate_tag_match(const struct song *song, enum tag_type type, const char *str)
 {
-	if (type == LOCATE_TAG_FILE_TYPE || type == LOCATE_TAG_ANY_TYPE) {
+	if (type == LOCATE_TAG_FILE_TYPE || (int)type == LOCATE_TAG_ANY_TYPE) {
 		char *uri = song_get_uri(song);
 		bool matches = strcmp(str, uri) == 0;
 		g_free(uri);
@@ -205,7 +205,7 @@ locate_tag_match(const struct song *song, enum tag_type type, const char *str)
 
 	for (unsigned i = 0; i < song->tag->num_items; i++) {
 		visited_types[song->tag->items[i]->type] = true;
-		if (type != LOCATE_TAG_ANY_TYPE &&
+		if ((int)type != LOCATE_TAG_ANY_TYPE &&
 		    song->tag->items[i]->type != type) {
 			continue;
 		}

src/log.c

@@ -55,7 +55,7 @@ static const char *log_charset;
 
 static bool stdout_mode = true;
 static int out_fd;
-static const char *out_filename;
+static char *out_filename;
 
 static void redirect_logs(int fd)
 {
@@ -134,14 +134,15 @@ open_log_file(void)
 }
 
 static bool
-log_init_file(const char *path, unsigned line, GError **error_r)
+log_init_file(unsigned line, GError **error_r)
 {
-	out_filename = path;
+	assert(out_filename != NULL);
+
 	out_fd = open_log_file();
 	if (out_fd < 0) {
 		g_set_error(error_r, log_quark(), errno,
 			    "failed to open log file \"%s\" (config line %u): %s",
-			    path, line, g_strerror(errno));
+			    out_filename, line, g_strerror(errno));
 		return false;
 	}
 
@@ -271,22 +272,33 @@ log_init(bool verbose, bool use_stdout, GError **error_r)
 			return true;
 #endif
 		} else {
-			GError *error = NULL;
-			char *path = config_dup_path(CONF_LOG_FILE, &error);
-			if (path == NULL) {
-				assert(error != NULL);
-				g_propagate_error(error_r, error);
-				return false;
-			}
-
-			bool success = log_init_file(path, param->line,
-						     error_r);
-			g_free(path);
-			return success;
+			out_filename = config_dup_path(CONF_LOG_FILE, error_r);
+			return out_filename != NULL &&
+				log_init_file(param->line, error_r);
 		}
 	}
 }
 
+static void
+close_log_files(void)
+{
+	if (stdout_mode)
+		return;
+
+#ifdef HAVE_SYSLOG
+	if (out_filename == NULL)
+		closelog();
+#endif
+}
+
+void
+log_deinit(void)
+{
+	close_log_files();
+	g_free(out_filename);
+}
+
+
 void setup_log_output(bool use_stdout)
 {
 	fflush(NULL);
@@ -327,15 +339,3 @@ int cycle_log_files(void)
 	g_debug("Done cycling log files\n");
 	return 0;
 }
-
-void close_log_files(void)
-{
-	if (stdout_mode)
-		return;
-
-#ifdef HAVE_SYSLOG
-	if (out_filename == NULL)
-		closelog();
-#endif
-}
-

src/log.h

@@ -44,10 +44,11 @@ log_early_init(bool verbose);
 bool
 log_init(bool verbose, bool use_stdout, GError **error_r);
 
+void
+log_deinit(void);
+
 void setup_log_output(bool use_stdout);
 
 int cycle_log_files(void);
 
-void close_log_files(void);
-
 #endif /* LOG_H */

src/main.c

@@ -135,10 +135,8 @@ glue_mapper_init(GError **error_r)
 		return false;
 	}
 
-#if GLIB_CHECK_VERSION(2,14,0)
 	if (music_dir == NULL)
 		music_dir = g_strdup(g_get_user_special_dir(G_USER_DIRECTORY_MUSIC));
-#endif
 
 	mapper_init(music_dir, playlist_dir);
 
@@ -538,6 +536,6 @@ int mpd_main(int argc, char *argv[])
 	WSACleanup();
 #endif
 
-	close_log_files();
+	log_deinit();
 	return EXIT_SUCCESS;
 }

src/mapper.c

@@ -36,10 +36,24 @@
 #include <errno.h>
 #include <dirent.h>
 
-static char *music_dir;
-static size_t music_dir_length;
+/**
+ * The absolute path of the music directory encoded in UTF-8.
+ */
+static char *music_dir_utf8;
+static size_t music_dir_utf8_length;
 
-static char *playlist_dir;
+/**
+ * The absolute path of the music directory encoded in the filesystem
+ * character set.
+ */
+static char *music_dir_fs;
+static size_t music_dir_fs_length;
+
+/**
+ * The absolute path of the playlist directory encoded in the
+ * filesystem character set.
+ */
+static char *playlist_dir_fs;
 
 /**
  * Duplicate a string, chop all trailing slashes.
@@ -79,27 +93,28 @@ check_directory(const char *path)
 #endif
 
 	DIR *dir = opendir(path);
-	if (dir == NULL && errno == EACCES)
-		g_warning("No permission to read directory: %s", path);
-	else
+	if (dir != NULL)
 		closedir(dir);
+	else if (errno == EACCES)
+		g_warning("No permission to read directory: %s", path);
 }
 
 static void
-mapper_set_music_dir(const char *path)
+mapper_set_music_dir(const char *path_utf8)
 {
-	check_directory(path);
+	music_dir_utf8 = strdup_chop_slash(path_utf8);
+	music_dir_utf8_length = strlen(music_dir_utf8);
 
-	music_dir = strdup_chop_slash(path);
-	music_dir_length = strlen(music_dir);
+	music_dir_fs = utf8_to_fs_charset(music_dir_utf8);
+	check_directory(music_dir_fs);
+	music_dir_fs_length = strlen(music_dir_fs);
 }
 
 static void
-mapper_set_playlist_dir(const char *path)
+mapper_set_playlist_dir(const char *path_utf8)
 {
-	check_directory(path);
-
-	playlist_dir = g_strdup(path);
+	playlist_dir_fs = utf8_to_fs_charset(path_utf8);
+	check_directory(playlist_dir_fs);
 }
 
 void mapper_init(const char *_music_dir, const char *_playlist_dir)
@@ -113,23 +128,31 @@ void mapper_init(const char *_music_dir, const char *_playlist_dir)
 
 void mapper_finish(void)
 {
-	g_free(music_dir);
-	g_free(playlist_dir);
+	g_free(music_dir_utf8);
+	g_free(music_dir_fs);
+	g_free(playlist_dir_fs);
 }
 
 const char *
-mapper_get_music_directory(void)
+mapper_get_music_directory_utf8(void)
 {
-	return music_dir;
+	return music_dir_utf8;
+}
+
+const char *
+mapper_get_music_directory_fs(void)
+{
+	return music_dir_fs;
 }
 
 const char *
 map_to_relative_path(const char *path_utf8)
 {
-	return music_dir != NULL &&
-		memcmp(path_utf8, music_dir, music_dir_length) == 0 &&
-		G_IS_DIR_SEPARATOR(path_utf8[music_dir_length])
-		? path_utf8 + music_dir_length + 1
+	return music_dir_utf8 != NULL &&
+		memcmp(path_utf8, music_dir_utf8,
+		       music_dir_utf8_length) == 0 &&
+		G_IS_DIR_SEPARATOR(path_utf8[music_dir_utf8_length])
+		? path_utf8 + music_dir_utf8_length + 1
 		: path_utf8;
 }
 
@@ -141,14 +164,14 @@ map_uri_fs(const char *uri)
 	assert(uri != NULL);
 	assert(*uri != '/');
 
-	if (music_dir == NULL)
+	if (music_dir_fs == NULL)
 		return NULL;
 
 	uri_fs = utf8_to_fs_charset(uri);
 	if (uri_fs == NULL)
 		return NULL;
 
-	path_fs = g_build_filename(music_dir, uri_fs, NULL);
+	path_fs = g_build_filename(music_dir_fs, uri_fs, NULL);
 	g_free(uri_fs);
 
 	return path_fs;
@@ -157,10 +180,11 @@ map_uri_fs(const char *uri)
 char *
 map_directory_fs(const struct directory *directory)
 {
-	assert(music_dir != NULL);
+	assert(music_dir_utf8 != NULL);
+	assert(music_dir_fs != NULL);
 
 	if (directory_is_root(directory))
-		return g_strdup(music_dir);
+		return g_strdup(music_dir_fs);
 
 	return map_uri_fs(directory_get_path(directory));
 }
@@ -168,9 +192,10 @@ map_directory_fs(const struct directory *directory)
 char *
 map_directory_child_fs(const struct directory *directory, const char *name)
 {
-	char *name_fs, *parent_fs, *path;
+	assert(music_dir_utf8 != NULL);
+	assert(music_dir_fs != NULL);
 
-	assert(music_dir != NULL);
+	char *name_fs, *parent_fs, *path;
 
 	/* check for invalid or unauthorized base names */
 	if (*name == 0 || strchr(name, '/') != NULL ||
@@ -208,11 +233,11 @@ map_song_fs(const struct song *song)
 char *
 map_fs_to_utf8(const char *path_fs)
 {
-	if (music_dir != NULL &&
-	    strncmp(path_fs, music_dir, music_dir_length) == 0 &&
-	    G_IS_DIR_SEPARATOR(path_fs[music_dir_length]))
+	if (music_dir_fs != NULL &&
+	    strncmp(path_fs, music_dir_fs, music_dir_fs_length) == 0 &&
+	    G_IS_DIR_SEPARATOR(path_fs[music_dir_fs_length]))
 		/* remove musicDir prefix */
-		path_fs += music_dir_length + 1;
+		path_fs += music_dir_fs_length + 1;
 	else if (G_IS_DIR_SEPARATOR(path_fs[0]))
 		/* not within musicDir */
 		return NULL;
@@ -226,7 +251,7 @@ map_fs_to_utf8(const char *path_fs)
 const char *
 map_spl_path(void)
 {
-	return playlist_dir;
+	return playlist_dir_fs;
 }
 
 char *
@@ -234,7 +259,7 @@ map_spl_utf8_to_fs(const char *name)
 {
 	char *filename_utf8, *filename_fs, *path;
 
-	if (playlist_dir == NULL)
+	if (playlist_dir_fs == NULL)
 		return NULL;
 
 	filename_utf8 = g_strconcat(name, PLAYLIST_FILE_SUFFIX, NULL);
@@ -243,7 +268,7 @@ map_spl_utf8_to_fs(const char *name)
 	if (filename_fs == NULL)
 		return NULL;
 
-	path = g_build_filename(playlist_dir, filename_fs, NULL);
+	path = g_build_filename(playlist_dir_fs, filename_fs, NULL);
 	g_free(filename_fs);
 
 	return path;

src/mapper.h

@@ -36,9 +36,20 @@ void mapper_init(const char *_music_dir, const char *_playlist_dir);
 
 void mapper_finish(void);
 
+/**
+ * Return the absolute path of the music directory encoded in UTF-8.
+ */
 G_GNUC_CONST
 const char *
-mapper_get_music_directory(void);
+mapper_get_music_directory_utf8(void);
+
+/**
+ * Return the absolute path of the music directory encoded in the
+ * filesystem character set.
+ */
+G_GNUC_CONST
+const char *
+mapper_get_music_directory_fs(void);
 
 /**
  * Returns true if a music directory was configured.
@@ -47,7 +58,7 @@ G_GNUC_CONST
 static inline bool
 mapper_has_music_directory(void)
 {
-	return mapper_get_music_directory() != NULL;
+	return mapper_get_music_directory_utf8() != NULL;
 }
 
 /**

src/output/httpd_output_plugin.c

@@ -53,6 +53,31 @@ httpd_output_quark(void)
 	return g_quark_from_static_string("httpd_output");
 }
 
+/**
+ * Check whether there is at least one client.
+ *
+ * Caller must lock the mutex.
+ */
+G_GNUC_PURE
+static bool
+httpd_output_has_clients(const struct httpd_output *httpd)
+{
+	return httpd->clients != NULL;
+}
+
+/**
+ * Check whether there is at least one client.
+ */
+G_GNUC_PURE
+static bool
+httpd_output_lock_has_clients(const struct httpd_output *httpd)
+{
+	g_mutex_lock(httpd->mutex);
+	bool result = httpd_output_has_clients(httpd);
+	g_mutex_unlock(httpd->mutex);
+	return result;
+}
+
 static void
 httpd_listen_in_event(int fd, const struct sockaddr *address,
 		      size_t address_length, int uid, void *ctx);
@@ -89,10 +114,6 @@ httpd_output_init(const struct config_param *param,
 		return NULL;
 	}
 
-	const char *encoder_name, *bind_to_address;
-	const struct encoder_plugin *encoder_plugin;
-	guint port;
-
 	/* read configuration */
 	httpd->name =
 		config_get_block_string(param, "name", "Set name in config");
@@ -101,10 +122,12 @@ httpd_output_init(const struct config_param *param,
 	httpd->website =
 		config_get_block_string(param, "website", "Set website in config");
 
-	port = config_get_block_unsigned(param, "port", 8000);
+	guint port = config_get_block_unsigned(param, "port", 8000);
 
-	encoder_name = config_get_block_string(param, "encoder", "vorbis");
-	encoder_plugin = encoder_plugin_get(encoder_name);
+	const char *encoder_name =
+		config_get_block_string(param, "encoder", "vorbis");
+	const struct encoder_plugin *encoder_plugin =
+		encoder_plugin_get(encoder_name);
 	if (encoder_plugin == NULL) {
 		g_set_error(error, httpd_output_quark(), 0,
 			    "No such encoder: %s", encoder_name);
@@ -119,7 +142,7 @@ httpd_output_init(const struct config_param *param,
 
 	httpd->server_socket = server_socket_new(httpd_listen_in_event, httpd);
 
-	bind_to_address =
+	const char *bind_to_address =
 		config_get_block_string(param, "bind_to_address", NULL);
 	bool success = bind_to_address != NULL &&
 		strcmp(bind_to_address, "any") != 0
@@ -250,8 +273,6 @@ httpd_listen_in_event(int fd, const struct sockaddr *address,
 static struct page *
 httpd_output_read_page(struct httpd_output *httpd)
 {
-	size_t size = 0, nbytes;
-
 	if (httpd->unflushed_input >= 65536) {
 		/* we have fed a lot of input into the encoder, but it
 		   didn't give anything back yet - flush now to avoid
@@ -260,9 +281,11 @@ httpd_output_read_page(struct httpd_output *httpd)
 		httpd->unflushed_input = 0;
 	}
 
+	size_t size = 0;
 	do {
-		nbytes = encoder_read(httpd->encoder, httpd->buffer + size,
-				      sizeof(httpd->buffer) - size);
+		size_t nbytes = encoder_read(httpd->encoder,
+					     httpd->buffer + size,
+					     sizeof(httpd->buffer) - size);
 		if (nbytes == 0)
 			break;
 
@@ -282,10 +305,7 @@ httpd_output_encoder_open(struct httpd_output *httpd,
 			  struct audio_format *audio_format,
 			  GError **error)
 {
-	bool success;
-
-	success = encoder_open(httpd->encoder, audio_format, error);
-	if (!success)
+	if (!encoder_open(httpd->encoder, audio_format, error))
 		return false;
 
 	/* we have to remember the encoder header, i.e. the first
@@ -319,14 +339,12 @@ httpd_output_open(struct audio_output *ao, struct audio_format *audio_format,
 		  GError **error)
 {
 	struct httpd_output *httpd = (struct httpd_output *)ao;
-	bool success;
 
 	g_mutex_lock(httpd->mutex);
 
 	/* open the encoder */
 
-	success = httpd_output_encoder_open(httpd, audio_format, error);
-	if (!success) {
+	if (!httpd_output_encoder_open(httpd, audio_format, error)) {
 		g_mutex_unlock(httpd->mutex);
 		return false;
 	}
@@ -397,6 +415,19 @@ httpd_output_delay(struct audio_output *ao)
 {
 	struct httpd_output *httpd = (struct httpd_output *)ao;
 
+	if (!httpd_output_lock_has_clients(httpd) && httpd->base.pause) {
+		/* if there's no client and this output is paused,
+		   then httpd_output_pause() will not do anything, it
+		   will not fill the buffer and it will not update the
+		   timer; therefore, we reset the timer here */
+		timer_reset(httpd->timer);
+
+		/* some arbitrary delay that is long enough to avoid
+		   consuming too much CPU, and short enough to notice
+		   new clients quickly enough */
+		return 1000;
+	}
+
 	return httpd->timer->started
 		? timer_delay(httpd->timer)
 		: 0;
@@ -457,10 +488,7 @@ static bool
 httpd_output_encode_and_play(struct httpd_output *httpd,
 			     const void *chunk, size_t size, GError **error)
 {
-	bool success;
-
-	success = encoder_write(httpd->encoder, chunk, size, error);
-	if (!success)
+	if (!encoder_write(httpd->encoder, chunk, size, error))
 		return false;
 
 	httpd->unflushed_input += size;
@@ -472,21 +500,12 @@ httpd_output_encode_and_play(struct httpd_output *httpd,
 
 static size_t
 httpd_output_play(struct audio_output *ao, const void *chunk, size_t size,
-		  GError **error)
+		  GError **error_r)
 {
 	struct httpd_output *httpd = (struct httpd_output *)ao;
-	bool has_clients;
 
-	g_mutex_lock(httpd->mutex);
-	has_clients = httpd->clients != NULL;
-	g_mutex_unlock(httpd->mutex);
-
-	if (has_clients) {
-		bool success;
-
-		success = httpd_output_encode_and_play(httpd, chunk, size,
-						       error);
-		if (!success)
+	if (httpd_output_lock_has_clients(httpd)) {
+		if (!httpd_output_encode_and_play(httpd, chunk, size, error_r))
 			return 0;
 	}
 
@@ -502,16 +521,11 @@ httpd_output_pause(struct audio_output *ao)
 {
 	struct httpd_output *httpd = (struct httpd_output *)ao;
 
-	g_mutex_lock(httpd->mutex);
-	bool has_clients = httpd->clients != NULL;
-	g_mutex_unlock(httpd->mutex);
-
-	if (has_clients) {
+	if (httpd_output_lock_has_clients(httpd)) {
 		static const char silence[1020];
 		return httpd_output_play(ao, silence, sizeof(silence),
 					 NULL) > 0;
 	} else {
-		g_usleep(100000);
 		return true;
 	}
 }
@@ -534,7 +548,6 @@ httpd_output_tag(struct audio_output *ao, const struct tag *tag)
 
 	if (httpd->encoder->plugin->tag != NULL) {
 		/* embed encoder tags */
-		struct page *page;
 
 		/* flush the current stream, and end it */
 
@@ -550,7 +563,7 @@ httpd_output_tag(struct audio_output *ao, const struct tag *tag)
 		   used as the new "header" page, which is sent to all
 		   new clients */
 
-		page = httpd_output_read_page(httpd);
+		struct page *page = httpd_output_read_page(httpd);
 		if (page != NULL) {
 			if (httpd->header != NULL)
 				page_unref(httpd->header);

src/output/jack_output_plugin.c

@@ -608,6 +608,16 @@ mpd_jack_close(G_GNUC_UNUSED struct audio_output *ao)
 	mpd_jack_stop(jd);
 }
 
+static unsigned
+mpd_jack_delay(struct audio_output *ao)
+{
+	struct jack_data *jd = (struct jack_data *)ao;
+
+	return jd->base.pause && jd->pause && !jd->shutdown
+		? 1000
+		: 0;
+}
+
 static inline jack_default_audio_sample_t
 sample_16_to_jack(int16_t sample)
 {
@@ -727,10 +737,6 @@ mpd_jack_pause(struct audio_output *ao)
 
 	jd->pause = true;
 
-	/* due to a MPD API limitation, we have to sleep a little bit
-	   here, to avoid hogging the CPU */
-	g_usleep(50000);
-
 	return true;
 }
 
@@ -742,6 +748,7 @@ const struct audio_output_plugin jack_output_plugin = {
 	.enable = mpd_jack_enable,
 	.disable = mpd_jack_disable,
 	.open = mpd_jack_open,
+	.delay = mpd_jack_delay,
 	.play = mpd_jack_play,
 	.pause = mpd_jack_pause,
 	.close = mpd_jack_close,

src/output/osx_output_plugin.c

@@ -228,9 +228,13 @@ osx_render(void *vdata,
 	g_cond_signal(od->condition);
 	g_mutex_unlock(od->mutex);
 
-	if (nbytes < buffer_size)
-		memset((unsigned char*)buffer->mData + nbytes, 0,
-		       buffer_size - nbytes);
+	buffer->mDataByteSize = nbytes;
+
+	unsigned i;
+	for (i = 1; i < buffer_list->mNumberBuffers; ++i) {
+		buffer = &buffer_list->mBuffers[i];
+		buffer->mDataByteSize = 0;
+	}
 
 	return 0;
 }

src/output/pulse_output_plugin.c

@@ -681,35 +681,6 @@ pulse_output_close(struct audio_output *ao)
 	pa_threaded_mainloop_unlock(po->mainloop);
 }
 
-/**
- * Check if the stream is (already) connected, and waits for a signal
- * if not.  The mainloop must be locked before calling this function.
- *
- * @return the current stream state
- */
-static pa_stream_state_t
-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:
-	case PA_STREAM_TERMINATED:
-	case PA_STREAM_UNCONNECTED:
-		break;
-
-	case PA_STREAM_CREATING:
-		pa_threaded_mainloop_wait(po->mainloop);
-		state = pa_stream_get_state(po->stream);
-		break;
-	}
-
-	return state;
-}
-
 /**
  * Check if the stream is (already) connected, and waits if not.  The
  * mainloop must be locked before calling this function.
@@ -719,35 +690,25 @@ pulse_output_check_stream(struct pulse_output *po)
 static bool
 pulse_output_wait_stream(struct pulse_output *po, GError **error_r)
 {
-	pa_stream_state_t state = pa_stream_get_state(po->stream);
+	while (true) {
+		switch (pa_stream_get_state(po->stream)) {
+		case PA_STREAM_READY:
+			return true;
 
-	switch (state) {
-	case PA_STREAM_READY:
-		return true;
+		case PA_STREAM_FAILED:
+		case PA_STREAM_TERMINATED:
+		case PA_STREAM_UNCONNECTED:
+			g_set_error(error_r, pulse_output_quark(),
+				    pa_context_errno(po->context),
+				    "failed to connect the stream: %s",
+				    pa_strerror(pa_context_errno(po->context)));
+			return false;
 
-	case PA_STREAM_FAILED:
-	case PA_STREAM_TERMINATED:
-	case PA_STREAM_UNCONNECTED:
-		g_set_error(error_r, pulse_output_quark(), 0,
-			    "disconnected");
-		return false;
-
-	case PA_STREAM_CREATING:
-		break;
+		case PA_STREAM_CREATING:
+			pa_threaded_mainloop_wait(po->mainloop);
+			break;
+		}
 	}
-
-	do {
-		state = pulse_output_check_stream(po);
-	} while (state == PA_STREAM_CREATING);
-
-	if (state != PA_STREAM_READY) {
-		g_set_error(error_r, pulse_output_quark(), 0,
-			    "failed to connect the stream: %s",
-			    pa_strerror(pa_context_errno(po->context)));
-		return false;
-	}
-
-	return true;
 }
 
 /**
@@ -801,6 +762,24 @@ pulse_output_stream_pause(struct pulse_output *po, bool pause,
 	return true;
 }
 
+static unsigned
+pulse_output_delay(struct audio_output *ao)
+{
+	struct pulse_output *po = (struct pulse_output *)ao;
+	unsigned result = 0;
+
+	pa_threaded_mainloop_lock(po->mainloop);
+
+	if (po->base.pause && pulse_output_stream_is_paused(po) &&
+	    pa_stream_get_state(po->stream) == PA_STREAM_READY)
+		/* idle while paused */
+		result = 1000;
+
+	pa_threaded_mainloop_unlock(po->mainloop);
+
+	return result;
+}
+
 static size_t
 pulse_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		  GError **error_r)
@@ -928,13 +907,8 @@ pulse_output_pause(struct audio_output *ao)
 
 	/* cork the stream */
 
-	if (pulse_output_stream_is_paused(po)) {
-		/* already paused; due to a MPD API limitation, we
-		   have to sleep a little bit here, to avoid hogging
-		   the CPU */
-
-		g_usleep(50000);
-	} else if (!pulse_output_stream_pause(po, true, &error)) {
+	if (!pulse_output_stream_is_paused(po) &&
+	    !pulse_output_stream_pause(po, true, &error)) {
 		pa_threaded_mainloop_unlock(po->mainloop);
 		g_warning("%s", error->message);
 		g_error_free(error);
@@ -971,6 +945,7 @@ const struct audio_output_plugin pulse_output_plugin = {
 	.enable = pulse_output_enable,
 	.disable = pulse_output_disable,
 	.open = pulse_output_open,
+	.delay = pulse_output_delay,
 	.play = pulse_output_play,
 	.cancel = pulse_output_cancel,
 	.pause = pulse_output_pause,

src/output/recorder_output_plugin.c

@@ -77,13 +77,12 @@ recorder_output_init(const struct config_param *param, GError **error_r)
 		return NULL;
 	}
 
-	const char *encoder_name;
-	const struct encoder_plugin *encoder_plugin;
-
 	/* read configuration */
 
-	encoder_name = config_get_block_string(param, "encoder", "vorbis");
-	encoder_plugin = encoder_plugin_get(encoder_name);
+	const char *encoder_name =
+		config_get_block_string(param, "encoder", "vorbis");
+	const struct encoder_plugin *encoder_plugin =
+		encoder_plugin_get(encoder_name);
 	if (encoder_plugin == NULL) {
 		g_set_error(error_r, recorder_output_quark(), 0,
 			    "No such encoder: %s", encoder_name);
@@ -121,33 +120,22 @@ recorder_output_finish(struct audio_output *ao)
 	g_free(recorder);
 }
 
-/**
- * Writes pending data from the encoder to the output file.
- */
 static bool
-recorder_output_encoder_to_file(struct recorder_output *recorder,
-			      GError **error_r)
+recorder_write_to_file(struct recorder_output *recorder,
+		       const void *_data, size_t length,
+		       GError **error_r)
 {
-	size_t size = 0, position, nbytes;
+	assert(length > 0);
 
-	assert(recorder->fd >= 0);
+	const int fd = recorder->fd;
 
-	/* read from the encoder */
+	const uint8_t *data = (const uint8_t *)_data, *end = data + length;
 
-	size = encoder_read(recorder->encoder, recorder->buffer,
-			    sizeof(recorder->buffer));
-	if (size == 0)
-		return true;
-
-	/* write everything into the file */
-
-	position = 0;
 	while (true) {
-		nbytes = write(recorder->fd, recorder->buffer + position,
-			       size - position);
+		ssize_t nbytes = write(fd, data, end - data);
 		if (nbytes > 0) {
-			position += (size_t)nbytes;
-			if (position >= size)
+			data += nbytes;
+			if (data == end)
 				return true;
 		} else if (nbytes == 0) {
 			/* shouldn't happen for files */
@@ -163,13 +151,37 @@ recorder_output_encoder_to_file(struct recorder_output *recorder,
 	}
 }
 
+/**
+ * Writes pending data from the encoder to the output file.
+ */
+static bool
+recorder_output_encoder_to_file(struct recorder_output *recorder,
+				GError **error_r)
+{
+	assert(recorder->fd >= 0);
+
+	while (true) {
+		/* read from the encoder */
+
+		size_t size = encoder_read(recorder->encoder, recorder->buffer,
+					   sizeof(recorder->buffer));
+		if (size == 0)
+			return true;
+
+		/* write everything into the file */
+
+		if (!recorder_write_to_file(recorder, recorder->buffer, size,
+					    error_r))
+			return false;
+	}
+}
+
 static bool
 recorder_output_open(struct audio_output *ao,
 		     struct audio_format *audio_format,
 		     GError **error_r)
 {
 	struct recorder_output *recorder = (struct recorder_output *)ao;
-	bool success;
 
 	/* create the output file */
 
@@ -185,8 +197,14 @@ recorder_output_open(struct audio_output *ao,
 
 	/* open the encoder */
 
-	success = encoder_open(recorder->encoder, audio_format, error_r);
-	if (!success) {
+	if (!encoder_open(recorder->encoder, audio_format, error_r)) {
+		close(recorder->fd);
+		unlink(recorder->path);
+		return false;
+	}
+
+	if (!recorder_output_encoder_to_file(recorder, error_r)) {
+		encoder_close(recorder->encoder);
 		close(recorder->fd);
 		unlink(recorder->path);
 		return false;

src/output/shout_output_plugin.c

@@ -36,11 +36,6 @@
 
 #define DEFAULT_CONN_TIMEOUT  2
 
-struct shout_buffer {
-	unsigned char data[32768];
-	size_t len;
-};
-
 struct shout_data {
 	struct audio_output base;
 
@@ -54,7 +49,7 @@ struct shout_data {
 
 	int timeout;
 
-	struct shout_buffer buf;
+	uint8_t buffer[32768];
 };
 
 static int shout_init_count;
@@ -114,24 +109,7 @@ static struct audio_output *
 my_shout_init_driver(const struct config_param *param,
 		     GError **error)
 {
-	struct shout_data *sd;
-	char *test;
-	unsigned port;
-	char *host;
-	char *mount;
-	char *passwd;
-	const char *encoding;
-	const struct encoder_plugin *encoder_plugin;
-	unsigned shout_format;
-	unsigned protocol;
-	const char *user;
-	char *name;
-	const char *value;
-	const struct block_param *block_param;
-	int public;
-
-	sd = new_shout_data();
-
+	struct shout_data *sd = new_shout_data();
 	if (!ao_base_init(&sd->base, &shout_output_plugin, param, error)) {
 		free_shout_data(sd);
 		return NULL;
@@ -152,13 +130,14 @@ my_shout_init_driver(const struct config_param *param,
 
 	shout_init_count++;
 
+	const struct block_param *block_param;
 	check_block_param("host");
-	host = block_param->value;
+	char *host = block_param->value;
 
 	check_block_param("mount");
-	mount = block_param->value;
+	char *mount = block_param->value;
 
-	port = config_get_block_unsigned(param, "port", 0);
+	unsigned port = config_get_block_unsigned(param, "port", 0);
 	if (port == 0) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "shout port must be configured");
@@ -166,17 +145,18 @@ my_shout_init_driver(const struct config_param *param,
 	}
 
 	check_block_param("password");
-	passwd = block_param->value;
+	const char *passwd = block_param->value;
 
 	check_block_param("name");
-	name = block_param->value;
+	const char *name = block_param->value;
 
-	public = config_get_block_bool(param, "public", false);
+	bool public = config_get_block_bool(param, "public", false);
 
-	user = config_get_block_string(param, "user", "source");
+	const char *user = config_get_block_string(param, "user", "source");
 
-	value = config_get_block_string(param, "quality", NULL);
+	const char *value = config_get_block_string(param, "quality", NULL);
 	if (value != NULL) {
+		char *test;
 		sd->quality = strtod(value, &test);
 
 		if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
@@ -201,6 +181,7 @@ my_shout_init_driver(const struct config_param *param,
 			goto failure;
 		}
 
+		char *test;
 		sd->bitrate = strtol(value, &test, 10);
 
 		if (*test != '\0' || sd->bitrate <= 0) {
@@ -210,8 +191,10 @@ my_shout_init_driver(const struct config_param *param,
 		}
 	}
 
-	encoding = config_get_block_string(param, "encoding", "ogg");
-	encoder_plugin = shout_encoder_plugin_get(encoding);
+	const char *encoding = config_get_block_string(param, "encoding",
+						       "ogg");
+	const struct encoder_plugin *encoder_plugin =
+		shout_encoder_plugin_get(encoding);
 	if (encoder_plugin == NULL) {
 		g_set_error(error, shout_output_quark(), 0,
 			    "couldn't find shout encoder plugin \"%s\"",
@@ -223,11 +206,13 @@ my_shout_init_driver(const struct config_param *param,
 	if (sd->encoder == NULL)
 		goto failure;
 
+	unsigned shout_format;
 	if (strcmp(encoding, "mp3") == 0 || strcmp(encoding, "lame") == 0)
 		shout_format = SHOUT_FORMAT_MP3;
 	else
 		shout_format = SHOUT_FORMAT_OGG;
 
+	unsigned protocol;
 	value = config_get_block_string(param, "protocol", NULL);
 	if (value != NULL) {
 		if (0 == strcmp(value, "shoutcast") &&
@@ -355,26 +340,24 @@ handle_shout_error(struct shout_data *sd, int err, GError **error)
 static bool
 write_page(struct shout_data *sd, GError **error)
 {
-	int err;
-
 	assert(sd->encoder != NULL);
 
-	sd->buf.len = encoder_read(sd->encoder,
-				   sd->buf.data, sizeof(sd->buf.data));
-	if (sd->buf.len == 0)
-		return true;
+	while (true) {
+		size_t nbytes = encoder_read(sd->encoder,
+					     sd->buffer, sizeof(sd->buffer));
+		if (nbytes == 0)
+			return true;
 
-	err = shout_send(sd->shout_conn, sd->buf.data, sd->buf.len);
-	if (!handle_shout_error(sd, err, error))
-		return false;
+		int err = shout_send(sd->shout_conn, sd->buffer, nbytes);
+		if (!handle_shout_error(sd, err, error))
+			return false;
+	}
 
 	return true;
 }
 
 static void close_shout_conn(struct shout_data * sd)
 {
-	sd->buf.len = 0;
-
 	if (sd->encoder != NULL) {
 		if (encoder_end(sd->encoder, NULL))
 			write_page(sd, NULL);
@@ -425,10 +408,7 @@ my_shout_close_device(struct audio_output *ao)
 static bool
 shout_connect(struct shout_data *sd, GError **error)
 {
-	int state;
-
-	state = shout_open(sd->shout_conn);
-	switch (state) {
+	switch (shout_open(sd->shout_conn)) {
 	case SHOUTERR_SUCCESS:
 	case SHOUTERR_CONNECTED:
 		return true;
@@ -448,17 +428,17 @@ my_shout_open_device(struct audio_output *ao, struct audio_format *audio_format,
 		     GError **error)
 {
 	struct shout_data *sd = (struct shout_data *)ao;
-	bool ret;
 
-	ret = shout_connect(sd, error);
-	if (!ret)
+	if (!shout_connect(sd, error))
 		return false;
 
-	sd->buf.len = 0;
+	if (!encoder_open(sd->encoder, audio_format, error)) {
+		shout_close(sd->shout_conn);
+		return false;
+	}
 
-	ret = encoder_open(sd->encoder, audio_format, error) &&
-		write_page(sd, error);
-	if (!ret) {
+	if (!write_page(sd, error)) {
+		encoder_close(sd->encoder);
 		shout_close(sd->shout_conn);
 		return false;
 	}
@@ -528,32 +508,27 @@ static void my_shout_set_tag(struct audio_output *ao,
 			     const struct tag *tag)
 {
 	struct shout_data *sd = (struct shout_data *)ao;
-	bool ret;
 	GError *error = NULL;
 
 	if (sd->encoder->plugin->tag != NULL) {
 		/* encoder plugin supports stream tags */
 
-		ret = encoder_pre_tag(sd->encoder, &error);
-		if (!ret) {
+		if (!encoder_pre_tag(sd->encoder, &error)) {
 			g_warning("%s", error->message);
 			g_error_free(error);
 			return;
 		}
 
-		ret = write_page(sd, NULL);
-		if (!ret)
+		if (!write_page(sd, NULL))
 			return;
 
-		ret = encoder_tag(sd->encoder, tag, &error);
-		if (!ret) {
+		if (!encoder_tag(sd->encoder, tag, &error)) {
 			g_warning("%s", error->message);
 			g_error_free(error);
 		}
 	} else {
 		/* no stream tag support: fall back to icy-metadata */
 		char song[1024];
-
 		shout_tag_to_metadata(tag, song, sizeof(song));
 
 		shout_metadata_add(sd->shout_meta, "song", song);

src/player_control.c

@@ -210,6 +210,14 @@ pc_set_pause(struct player_control *pc, bool pause_flag)
 	player_unlock(pc);
 }
 
+void
+pc_set_border_pause(struct player_control *pc, bool border_pause)
+{
+	player_lock(pc);
+	pc->border_pause = border_pause;
+	player_unlock(pc);
+}
+
 void
 pc_get_status(struct player_control *pc, struct player_status *status)
 {

src/player_control.h

@@ -115,6 +115,15 @@ struct player_control {
 	float mixramp_db;
 	float mixramp_delay_seconds;
 	double total_play_time;
+
+	/**
+	 * If this flag is set, then the player will be auto-paused at
+	 * the end of the song, before the next song starts to play.
+	 *
+	 * This is a copy of the queue's "single" flag most of the
+	 * time.
+	 */
+	bool border_pause;
 };
 
 struct player_control *
@@ -207,6 +216,12 @@ pc_set_pause(struct player_control *pc, bool pause_flag);
 void
 pc_pause(struct player_control *pc);
 
+/**
+ * Set the player's #border_pause flag.
+ */
+void
+pc_set_border_pause(struct player_control *pc, bool border_pause);
+
 void
 pc_kill(struct player_control *pc);
 

src/player_thread.c

@@ -450,6 +450,8 @@ static bool player_seek_decoder(struct player *player)
 
 	assert(pc->next_song != NULL);
 
+	const unsigned start_ms = song->start_ms;
+
 	if (decoder_current_song(dc) != song) {
 		/* the decoder is already decoding the "next" song -
 		   stop it and start the previous song again */
@@ -498,7 +500,7 @@ static bool player_seek_decoder(struct player *player)
 	if (where < 0.0)
 		where = 0.0;
 
-	if (!dc_seek(dc, where + song->start_ms / 1000.0)) {
+	if (!dc_seek(dc, where + start_ms / 1000.0)) {
 		/* decoder failure */
 		player_command_finished(pc);
 		return false;
@@ -840,6 +842,16 @@ player_song_border(struct player *player)
 	if (!player_wait_for_decoder(player))
 		return false;
 
+	struct player_control *const pc = player->pc;
+	player_lock(pc);
+
+	if (pc->border_pause) {
+		player->paused = true;
+		pc->state = PLAYER_STATE_PAUSE;
+	}
+
+	player_unlock(pc);
+
 	return true;
 }
 
@@ -955,7 +967,10 @@ static void do_play(struct player_control *pc, struct decoder_control *dc)
 			player_dc_start(&player, music_pipe_new());
 		}
 
-		if (player_dc_at_next_song(&player) &&
+		if (/* no cross-fading if MPD is going to pause at the
+		       end of the current song */
+		    !pc->border_pause &&
+		    player_dc_at_next_song(&player) &&
 		    player.xfade == XFADE_UNKNOWN &&
 		    !decoder_lock_is_starting(dc)) {
 			/* enable cross fading in this song?  if yes,

src/playlist.c

@@ -109,11 +109,6 @@ playlist_song_started(struct playlist *playlist, struct player_control *pc)
 	playlist->current = playlist->queued;
 	playlist->queued = -1;
 
-	/* Pause if we are in single mode. */
-	if(playlist->queue.single && !playlist->queue.repeat) {
-		pc_set_pause(pc, true);
-	}
-
 	if(playlist->queue.consume)
 		playlist_delete(playlist, pc,
 				queue_order_to_position(&playlist->queue,
@@ -239,9 +234,13 @@ playlist_sync(struct playlist *playlist, struct player_control *pc)
 		if (pc_next_song == NULL && playlist->queued != -1)
 			playlist_song_started(playlist, pc);
 
+		player_lock(pc);
+		pc_next_song = pc->next_song;
+		player_unlock(pc);
+
 		/* make sure the queued song is always set (if
 		   possible) */
-		if (pc->next_song == NULL && playlist->queued < 0)
+		if (pc_next_song == NULL && playlist->queued < 0)
 			playlist_update_queued_song(playlist, pc, NULL);
 	}
 }
@@ -306,7 +305,11 @@ playlist_set_repeat(struct playlist *playlist, struct player_control *pc,
 	if (status == playlist->queue.repeat)
 		return;
 
-	playlist->queue.repeat = status;
+	struct queue *queue = &playlist->queue;
+
+	queue->repeat = status;
+
+	pc_set_border_pause(pc, queue->single && !queue->repeat);
 
 	/* if the last song is currently being played, the "next song"
 	   might change when repeat mode is toggled */
@@ -334,7 +337,11 @@ playlist_set_single(struct playlist *playlist, struct player_control *pc,
 	if (status == playlist->queue.single)
 		return;
 
-	playlist->queue.single = status;
+	struct queue *queue = &playlist->queue;
+
+	queue->single = status;
+
+	pc_set_border_pause(pc, queue->single && !queue->repeat);
 
 	/* if the last song is currently being played, the "next song"
 	   might change when single mode is toggled */

src/playlist/lastfm_playlist_plugin.c

@@ -58,12 +58,10 @@ lastfm_init(const struct config_param *param)
 
 	lastfm_config.user = g_uri_escape_string(user, NULL, false);
 
-#if GLIB_CHECK_VERSION(2,16,0)
 	if (strlen(passwd) != 32)
 		lastfm_config.md5 = g_compute_checksum_for_string(G_CHECKSUM_MD5,
 								  passwd, strlen(passwd));
 	else
-#endif
 		lastfm_config.md5 = g_strdup(passwd);
 
 	return true;

src/playlist_edit.c

@@ -83,7 +83,7 @@ playlist_append_song(struct playlist *playlist, struct player_control *pc,
 
 	queued = playlist_get_queued_song(playlist);
 
-	id = queue_append(&playlist->queue, song);
+	id = queue_append(&playlist->queue, song, 0);
 
 	if (playlist->queue.random) {
 		/* shuffle the new song into the list of remaining

src/playlist_print.c

@@ -163,6 +163,9 @@ playlist_provider_print(struct client *client, const char *uri,
 			song_print_info(client, song);
 		else
 			song_print_uri(client, song);
+
+		if (!song_in_database(song))
+			song_free(song);
 	}
 
 	g_free(base_uri);

src/playlist_save.c

@@ -137,7 +137,8 @@ playlist_load_spl(struct playlist *playlist, struct player_control *pc,
 					*p = '/';
 				p++;
 			}
-			if ((playlist_append_uri(playlist, pc, temp, NULL)) != PLAYLIST_RESULT_SUCCESS) {
+			if ((playlist_append_uri(playlist, pc, temp2,
+						 NULL)) != PLAYLIST_RESULT_SUCCESS) {
 				g_warning("can't add file \"%s\"", temp2);
 			}
 			g_free(temp2);

src/playlist_song.c

@@ -23,6 +23,7 @@
 #include "mapper.h"
 #include "song.h"
 #include "uri.h"
+#include "path.h"
 #include "ls.h"
 #include "tag.h"
 
@@ -62,14 +63,19 @@ apply_song_metadata(struct song *dest, const struct song *src)
 		if (path_fs == NULL)
 			return dest;
 
-		tmp = song_file_new(path_fs, NULL);
-		g_free(path_fs);
+		char *path_utf8 = fs_charset_to_utf8(path_fs);
+		if (path_utf8 != NULL)
+			g_free(path_fs);
+		else
+			path_utf8 = path_fs;
+
+		tmp = song_file_new(path_utf8, NULL);
+		g_free(path_utf8);
 
 		merge_song_metadata(tmp, dest, src);
 	} else {
 		tmp = song_file_new(dest->uri, NULL);
 		merge_song_metadata(tmp, dest, src);
-		song_free(dest);
 	}
 
 	if (dest->tag != NULL && dest->tag->time > 0 &&
@@ -80,20 +86,42 @@ apply_song_metadata(struct song *dest, const struct song *src)
 		   (e.g. last track on a CUE file); fix it up here */
 		tmp->tag->time = dest->tag->time - src->start_ms / 1000;
 
+	if (!song_in_database(dest))
+		song_free(dest);
+
 	return tmp;
 }
 
+static struct song *
+playlist_check_load_song(const struct song *song, const char *uri, bool secure)
+{
+	struct song *dest;
+
+	if (uri_has_scheme(uri)) {
+		dest = song_remote_new(uri);
+	} else if (g_path_is_absolute(uri) && secure) {
+		dest = song_file_load(uri, NULL);
+		if (dest == NULL)
+			return NULL;
+	} else {
+		dest = db_get_song(uri);
+		if (dest == NULL)
+			/* not found in database */
+			return NULL;
+	}
+
+	return apply_song_metadata(dest, song);
+}
+
 struct song *
 playlist_check_translate_song(struct song *song, const char *base_uri,
 			      bool secure)
 {
-	struct song *dest;
-
 	if (song_in_database(song))
 		/* already ok */
 		return song;
 
-	char *uri = song->uri;
+	const char *uri = song->uri;
 
 	if (uri_has_scheme(uri)) {
 		if (uri_supported_scheme(uri))
@@ -115,11 +143,11 @@ playlist_check_translate_song(struct song *song, const char *base_uri,
 
 	if (g_path_is_absolute(uri)) {
 		/* XXX fs_charset vs utf8? */
-		const char *prefix = mapper_get_music_directory();
+		const char *suffix = map_to_relative_path(uri);
+		assert(suffix != NULL);
 
-		if (prefix != NULL && g_str_has_prefix(uri, prefix) &&
-		    uri[strlen(prefix)] == '/')
-			uri += strlen(prefix) + 1;
+		if (suffix != uri)
+			uri = suffix;
 		else if (!secure) {
 			/* local files must be relative to the music
 			   directory when "secure" is enabled */
@@ -130,32 +158,12 @@ playlist_check_translate_song(struct song *song, const char *base_uri,
 		base_uri = NULL;
 	}
 
+	char *allocated = NULL;
 	if (base_uri != NULL)
-		uri = g_build_filename(base_uri, uri, NULL);
-	else
-		uri = g_strdup(uri);
+		uri = allocated = g_build_filename(base_uri, uri, NULL);
 
-	if (uri_has_scheme(uri)) {
-		dest = song_remote_new(uri);
-		g_free(uri);
-	} else if (g_path_is_absolute(uri) && secure) {
-		dest = song_file_load(uri, NULL);
-		if (dest == NULL) {
-			song_free(song);
-			return NULL;
-		}
-	} else {
-		dest = db_get_song(uri);
-		g_free(uri);
-		if (dest == NULL) {
-			/* not found in database */
-			song_free(song);
-			return dest;
-		}
-	}
-
-	dest = apply_song_metadata(dest, song);
+	struct song *dest = playlist_check_load_song(song, uri, secure);
 	song_free(song);
-
+	g_free(allocated);
 	return dest;
 }

src/queue.c

@@ -96,7 +96,7 @@ queue_modify_all(struct queue *queue)
 }
 
 unsigned
-queue_append(struct queue *queue, struct song *song)
+queue_append(struct queue *queue, struct song *song, uint8_t priority)
 {
 	unsigned id = queue_generate_id(queue);
 
@@ -106,7 +106,7 @@ queue_append(struct queue *queue, struct song *song)
 		.song = song,
 		.id = id,
 		.version = queue->version,
-		.priority = 0,
+		.priority = priority,
 	};
 
 	queue->order[queue->length] = queue->length;

src/queue.h

@@ -280,9 +280,11 @@ queue_modify_all(struct queue *queue);
  *
  * If a song is not in the database (determined by
  * song_in_database()), it is freed when removed from the queue.
+ *
+ * @param priority the priority of this new queue item
  */
 unsigned
-queue_append(struct queue *queue, struct song *song);
+queue_append(struct queue *queue, struct song *song, uint8_t priority);
 
 /**
  * Swaps two songs, addressed by their position.

src/queue_save.c

@@ -24,9 +24,12 @@
 #include "uri.h"
 #include "database.h"
 #include "song_save.h"
+#include "text_file.h"
 
 #include <stdlib.h>
 
+#define PRIO_LABEL "Prio: "
+
 static void
 queue_save_database_song(FILE *fp, int idx, const struct song *song)
 {
@@ -54,8 +57,13 @@ queue_save_song(FILE *fp, int idx, const struct song *song)
 void
 queue_save(FILE *fp, const struct queue *queue)
 {
-	for (unsigned i = 0; i < queue_length(queue); i++)
+	for (unsigned i = 0; i < queue_length(queue); i++) {
+		uint8_t prio = queue_get_priority_at_position(queue, i);
+		if (prio != 0)
+			fprintf(fp, PRIO_LABEL "%u\n", prio);
+
 		queue_save_song(fp, i, queue_get(queue, i));
+	}
 }
 
 static struct song *
@@ -75,6 +83,15 @@ queue_load_song(FILE *fp, GString *buffer, const char *line,
 	if (queue_is_full(queue))
 		return;
 
+	uint8_t priority = 0;
+	if (g_str_has_prefix(line, PRIO_LABEL)) {
+		priority = strtoul(line + sizeof(PRIO_LABEL) - 1, NULL, 10);
+
+		line = read_text_line(fp, buffer);
+		if (line == NULL)
+			return;
+	}
+
 	if (g_str_has_prefix(line, SONG_BEGIN)) {
 		const char *uri = line + sizeof(SONG_BEGIN) - 1;
 		if (!uri_has_scheme(uri) && !g_path_is_absolute(uri))
@@ -102,5 +119,5 @@ queue_load_song(FILE *fp, GString *buffer, const char *line,
 			return;
 	}
 
-	queue_append(queue, song);
+	queue_append(queue, song, priority);
 }

src/tag_ape.c

@@ -24,7 +24,7 @@
 #include "tag_handler.h"
 #include "ape.h"
 
-static const struct tag_table ape_tags[] = {
+const struct tag_table ape_tags[] = {
 	{ "album artist", TAG_ALBUM_ARTIST },
 	{ "year", TAG_DATE },
 	{ NULL, TAG_NUM_OF_ITEM_TYPES }

src/tag_ape.h

@@ -20,10 +20,14 @@
 #ifndef MPD_TAG_APE_H
 #define MPD_TAG_APE_H
 
+#include "tag_table.h"
+
 #include <stdbool.h>
 
 struct tag_handler;
 
+extern const struct tag_table ape_tags[];
+
 /**
  * Scan the APE tags of a file.
  *

src/timer.c

@@ -20,23 +20,14 @@
 #include "config.h"
 #include "timer.h"
 #include "audio_format.h"
+#include "clock.h"
 
 #include <glib.h>
 
 #include <assert.h>
 #include <limits.h>
-#include <sys/time.h>
 #include <stddef.h>
 
-static uint64_t now(void)
-{
-	struct timeval tv;
-
-	gettimeofday(&tv, NULL);
-
-	return ((uint64_t)tv.tv_sec * 1000000) + tv.tv_usec;
-}
-
 struct timer *timer_new(const struct audio_format *af)
 {
 	struct timer *timer = g_new(struct timer, 1);
@@ -54,7 +45,7 @@ void timer_free(struct timer *timer)
 
 void timer_start(struct timer *timer)
 {
-	timer->time = now();
+	timer->time = monotonic_clock_us();
 	timer->started = 1;
 }
 
@@ -74,7 +65,7 @@ void timer_add(struct timer *timer, int size)
 unsigned
 timer_delay(const struct timer *timer)
 {
-	int64_t delay = (int64_t)(timer->time - now()) / 1000;
+	int64_t delay = (int64_t)(timer->time - monotonic_clock_us()) / 1000;
 	if (delay < 0)
 		return 0;
 
@@ -90,7 +81,7 @@ void timer_sync(struct timer *timer)
 
 	assert(timer->started);
 
-	sleep_duration = timer->time - now();
+	sleep_duration = timer->time - monotonic_clock_us();
 	if (sleep_duration > 0)
 		g_usleep(sleep_duration);
 }

src/update_walk.c

@@ -283,12 +283,20 @@ skip_symlink(const struct directory *directory, const char *utf8_name)
 		return true;
 
 	char buffer[MPD_PATH_MAX];
-	ssize_t ret = readlink(path_fs, buffer, sizeof(buffer));
+	ssize_t length = readlink(path_fs, buffer, sizeof(buffer));
 	g_free(path_fs);
-	if (ret < 0)
+	if (length < 0)
 		/* don't skip if this is not a symlink */
 		return errno != EINVAL;
 
+	if ((size_t)length >= sizeof(buffer))
+		/* skip symlinks when the buffer is too small for the
+		   link target */
+		return true;
+
+	/* null-terminate the buffer, because readlink() will not */
+	buffer[length] = 0;
+
 	if (!follow_inside_symlinks && !follow_outside_symlinks) {
 		/* ignore all symlinks */
 		return true;

src/volume.c

@@ -20,7 +20,6 @@
 #include "config.h"
 #include "volume.h"
 #include "conf.h"
-#include "player_control.h"
 #include "idle.h"
 #include "pcm_volume.h"
 #include "output_all.h"

test/dump_playlist.c

@@ -161,7 +161,7 @@ int main(int argc, char **argv)
 	config_global_init();
 	success = config_read_file(argv[1], &error);
 	if (!success) {
-		g_printerr("%s:", error->message);
+		g_printerr("%s\n", error->message);
 		g_error_free(error);
 		return 1;
 	}

test/run_encoder.c

@@ -99,14 +99,15 @@ int main(int argc, char **argv)
 		}
 	}
 
-	ret = encoder_open(encoder, &audio_format, &error);
-	if (encoder == NULL) {
+	if (!encoder_open(encoder, &audio_format, &error)) {
 		g_printerr("Failed to open encoder: %s\n",
 			   error->message);
 		g_error_free(error);
 		return 1;
 	}
 
+	encoder_to_stdout(encoder);
+
 	/* do it */
 
 	while ((nbytes = read(0, buffer, sizeof(buffer))) > 0) {

test/test_queue_priority.c

@@ -41,7 +41,7 @@ main(G_GNUC_UNUSED int argc, G_GNUC_UNUSED char **argv)
 	queue_init(&queue, 32);
 
 	for (unsigned i = 0; i < G_N_ELEMENTS(songs); ++i)
-		queue_append(&queue, &songs[i]);
+		queue_append(&queue, &songs[i], 0);
 
 	assert(queue_length(&queue) == G_N_ELEMENTS(songs));
 

test/test_vorbis_encoder.c

@@ -67,6 +67,8 @@ main(G_GNUC_UNUSED int argc, G_GNUC_UNUSED char **argv)
 	success = encoder_open(encoder, &audio_format, NULL);
 	assert(success);
 
+	encoder_to_stdout(encoder);
+
 	/* write a block of data */
 
 	success = encoder_write(encoder, zero, sizeof(zero), NULL);

valgrind.suppressions

@@ -39,6 +39,14 @@
    fun:g_main_loop_run
 }
 
+{
+   <insert_a_suppression_name_here>
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:g_once_init_leave
+}
+
 {
    g_log
    Memcheck:Leak
@@ -57,6 +65,14 @@
    fun:g_slice_*
 }
 
+{
+   <insert_a_suppression_name_here>
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:g_static_mutex_get_mutex_impl
+}
+
 {
    g_private
    Memcheck:Leak
@@ -483,3 +499,24 @@
    fun:?alloc
    fun:snd1_dlobj_cache_get
 }
+
+{
+   <insert_a_suppression_name_here>
+   Memcheck:Leak
+   fun:_Znwm
+   ...
+   obj:*/libjack.so*
+   fun:call_init
+   fun:_dl_init
+}
+
+{
+   <insert_a_suppression_name_here>
+   Memcheck:Leak
+   fun:*alloc
+   fun:_dl_allocate_tls
+   ...
+   obj:*/libffado.so*
+   fun:call_init
+   fun:_dl_init
+}