Merge commit 'release-0.15.3' into alt-git

Makefile.am

@@ -599,7 +599,6 @@ noinst_PROGRAMS = \
 	test/run_input \
 	test/run_decoder \
 	test/read_tags \
-	test/run_encoder \
 	test/run_output \
 	test/read_mixer \
 	test/software_volume
@@ -662,6 +661,8 @@ test_read_tags_SOURCES = test/read_tags.c \
 	$(TAG_SRC) \
 	$(DECODER_SRC)
 
+if ENABLE_ENCODER
+noinst_PROGRAMS += test/run_encoder
 test_run_encoder_SOURCES = test/run_encoder.c \
 	src/conf.c src/buffer2array.c \
 	src/utils.c \
@@ -671,6 +672,7 @@ test_run_encoder_SOURCES = test/run_encoder.c \
 test_run_encoder_LDADD = $(MPD_LIBS) \
 	$(ENCODER_LIBS) \
 	$(GLIB_LIBS)
+endif
 
 test_software_volume_SOURCES = test/software_volume.c \
 	src/audio_parser.c \

NEWS

+ver 0.15.3 (2009/08/29)
+* decoders:
+  - vorbis: faster tag scanning with ov_test_callback()
+* output:
+  - fix stuttering due to uninitialized variable
+* update: don't re-read unchanged container files
+
+
+ver 0.15.2 (2009/08/15)
+* tags:
+  - ape: check the tag size (fixes integer underflow)
+  - ape: added protection against large memory allocations
+* decoders:
+  - mad: skip ID3 frames when libid3tag is disabled
+  - flac: parse all replaygain tags
+  - flac: don't allocate cuesheet twice (memleak)
+* output:
+  - shout: fixed stuck pause bug
+  - shout: minimize the unpause latency
+* update: free empty path string (memleak)
+* update: free temporary string in container scan (memleak)
+* directory: free empty directories after removing them (memleak)
+
+
+ver 0.15.1 (2009/07/15)
+* decoders:
+  - flac: fix assertion failure in tag_free() call
+* output:
+  - httpd: include sys/types.h (fixes Mac OS X)
+* commands:
+  - don't resume playback when stopping during pause
+* database: fixed NULL pointer dereference after charset change
+* log: fix double free() bug during shutdown
+
+
 ver 0.15 (2009/06/23)
 * input:
   - parse Icy-Metadata

configure.ac

 AC_PREREQ(2.60)
-AC_INIT(mpd, 0.15, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.15.3, musicpd-dev-team@lists.sourceforge.net)
 AC_CONFIG_SRCDIR([src/main.c])
 AM_INIT_AUTOMAKE([foreign 1.9 dist-bzip2])
 AM_CONFIG_HEADER(config.h)
@@ -397,9 +397,9 @@ AC_ARG_ENABLE(audiofile,
 	enable_audiofile=yes)
 
 AC_ARG_ENABLE(ffmpeg,
-	AS_HELP_STRING([--disable-ffmpeg],
-		[enable FFMPEG support (default: enable)]),,
-	enable_ffmpeg=yes)
+	AS_HELP_STRING([--enable-ffmpeg],
+		[enable FFMPEG support]),,
+	enable_ffmpeg=auto)
 
 AC_ARG_ENABLE(flac,
 	AS_HELP_STRING([--disable-flac],
@@ -591,8 +591,7 @@ dnl audio output plugins
 dnl
 
 AC_ARG_ENABLE(alsa,
-	AS_HELP_STRING([--enable-alsa],
-		[disable ALSA support]),,
+	AS_HELP_STRING([--enable-alsa], [enable ALSA support]),,
 	enable_alsa=auto)
 
 AC_ARG_ENABLE(ao,
@@ -914,10 +913,8 @@ fi
 
 AM_CONDITIONAL(HAVE_AUDIOFILE, test x$enable_audiofile = xyes)
 
-if test x$enable_ffmpeg = xyes; then
-	PKG_CHECK_MODULES(FFMPEG, [libavformat libavcodec libavutil],,
-		enable_ffmpeg=no)
-fi
+MPD_AUTO_PKG(ffmpeg, FFMPEG, [libavformat libavcodec libavutil],
+	[ffmpeg decoder library], [libavformat+libavcodec+libavutil not found])
 
 if test x$enable_ffmpeg = xyes; then
 	old_LIBS=$LIBS

doc/mpd.conf.5

@@ -35,8 +35,8 @@ You must recreate the database after changing this option.
 The default is "yes".
 .TP
 .B follow_inside_symlinks <yes or no>
-Control if MPD will follow symbolic links pointing outside the music dir, potentially
-adding duplicates to the database.
+Control if MPD will follow symbolic links pointing inside the music dir,
+potentially adding duplicates to the database.
 You must recreate the database after changing this option.
 The default is "yes".
 .TP

doc/protocol.xml

@@ -134,7 +134,7 @@
               <listitem>
                 <para>
                   <returnvalue>database</returnvalue>: the song database
-                  has been updated
+                  has been modified after <command>update</command>.
                 </para>
               </listitem>
               <listitem>
@@ -272,6 +272,24 @@
               </listitem>
               <listitem>
                 <para>
+                  <varname>nextsong</varname>:
+                  <footnoteref linkend="since_0_15"/>
+                  <returnvalue> playlist song number of the next
+                  song to be played
+                  </returnvalue>
+                </para>
+              </listitem>
+              <listitem>
+                <para>
+                  <varname>nextsongid</varname>:
+                  <footnoteref linkend="since_0_15"/>
+                  <returnvalue>playlist songid of the next song
+                  to be played
+                  </returnvalue>
+                </para>
+              </listitem>
+              <listitem>
+                <para>
                   <varname>time</varname>:
                   <returnvalue>total time elapsed (of current
                   playing/paused song)</returnvalue>

doc/user.xml

@@ -11,6 +11,30 @@
       This document is work in progress.  Most of it may be incomplete
       yet.  Please help!
     </para>
+
+    <para>
+      MPD (Music Player Daemon) is, as the name suggests, a server
+      software allowing you to remotely play your music, handle
+      playlists, deliver music (HTTP STREAMS with various
+      sub-protocols) and organizze playlists.
+    </para>
+
+    <para>
+      It has been written with minimal resource usage and stability in
+      mind!  Infact, it runs fine on a Pentium 75, allowing you to use
+      your cheap old PC to create a stereo system!
+    </para>
+
+    <para>
+      MPD supports also Gapless playback, buffered audio output, and
+      crossfading!
+    </para>
+
+    <para>
+      The separate client and server design allows users to choose a
+      user interface that best suites their tastes independently of
+      the underlying daemon, which actually plays music!
+    </para>
   </chapter>
 
   <chapter>
@@ -26,10 +50,16 @@
       <title>Installing on Debian/Ubuntu</title>
 
       <para>
-        Install the package <filename>mpd</filename>:
+        Install the package <filename>mpd</filename> via APT:
       </para>
 
       <programlisting>apt-get install mpd</programlisting>
+
+      <para>
+        When installed this way, MPD by default looks for music in
+        /var/lib/mpd/music/; this may not be correct. Look at your
+        /etc/mpd.conf file...
+      </para>
     </section>
 
     <section>
@@ -41,8 +71,8 @@
         page</ulink> and unpack it:
       </para>
 
-      <programlisting>tar xjf mpd-0.14.2.tar.bz
-cd mpd-0.14.2</programlisting>
+      <programlisting>tar xjf mpd-version.tar.bz
+cd mpd-version</programlisting>
 
       <para>
         Make sure that all the required libraries and build tools are
@@ -678,6 +708,28 @@ cd mpd-0.14.2</programlisting>
           The <varname>pipe</varname> plugin starts a program and
           writes raw PCM data into its standard input.
         </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>command</varname>
+                  <parameter>CMD</parameter>
+                </entry>
+                <entry>
+                  This command is invoked with the shell.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
       </section>
 
       <section>
@@ -687,6 +739,40 @@ cd mpd-0.14.2</programlisting>
           The <varname>pulse</varname> plugin connects to a PulseAudio
           server.
         </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>server</varname>
+                  <parameter>HOSTNAME</parameter>
+                </entry>
+                <entry>
+                  Sets the host name of the PulseAudio server.  By
+                  default, MPD connects to the local PulseAudio
+                  server.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>sink</varname>
+                  <parameter>NAME</parameter>
+                </entry>
+                <entry>
+                  Specifies the name of the PulseAudio sink MPD should
+                  play on.
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
       </section>
 
       <section>
@@ -696,6 +782,129 @@ cd mpd-0.14.2</programlisting>
           The <varname>shout</varname> plugin connects to a ShoutCast
           or IceCast server.  It forwards tags to this server.
         </para>
+
+        <para>
+          You must set a <varname>format</varname>.
+        </para>
+
+        <informaltable>
+          <tgroup cols="2">
+            <thead>
+              <row>
+                <entry>Setting</entry>
+                <entry>Description</entry>
+              </row>
+            </thead>
+            <tbody>
+              <row>
+                <entry>
+                  <varname>host</varname>
+                  <parameter>HOSTNAME</parameter>
+                </entry>
+                <entry>
+                  Sets the host name of the Shoutcast/Icecast server.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>port</varname>
+                  <parameter>PORTNUMBER</parameter>
+                </entry>
+                <entry>
+                  Connect to this port number on the specified host.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>timeout</varname>
+                  <parameter>SECONDS</parameter>
+                </entry>
+                <entry>
+                  Set the timeout for the shout connection in seconds.
+                  Defaults to 2 seconds.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>mount</varname>
+                  <parameter>URI</parameter>
+                </entry>
+                <entry>
+                  Mounts the MPD stream in the specified URI.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>user</varname>
+                  <parameter>USERNAME</parameter>
+                </entry>
+                <entry>
+                  Sets the user name for submitting the stream to the
+                  server.  Default is "source".
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>password</varname>
+                  <parameter>PWD</parameter>
+                </entry>
+                <entry>
+                  Sets the password for submitting the stream to the
+                  server.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>name</varname>
+                  <parameter>NAME</parameter>
+                </entry>
+                <entry>
+                  Sets the name of the stream.
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>genre</varname>
+                  <parameter>GENRE</parameter>
+                </entry>
+                <entry>
+                  Sets the genre of the stream (optional).
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>description</varname>
+                  <parameter>DESCRIPTION</parameter>
+                </entry>
+                <entry>
+                  Sets a short description of the stream (optional).
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>public</varname>
+                  <parameter>yes|no</parameter>
+                </entry>
+                <entry>
+                  Specifies whether the stream should be "public".
+                  Default is "no".
+                </entry>
+              </row>
+              <row>
+                <entry>
+                  <varname>encoder</varname>
+                  <parameter>PLUGIN</parameter>
+                </entry>
+                <entry>
+                  Sets the name of the encoder plugin.  Default is
+                  "vorbis".  "vorbis" and "lame" are valid encoder
+                  plugins (provided that you enabled them at compile
+                  time).
+                </entry>
+              </row>
+            </tbody>
+          </tgroup>
+        </informaltable>
       </section>
 
       <section>

src/database.c

@@ -318,10 +318,11 @@ db_load(GError **error)
 			if (old_charset != NULL
 			    && strcmp(new_charset, old_charset)) {
 				fclose(fp);
-				g_message("Existing database has charset \"%s\" "
-					  "instead of \"%s\"; "
-					  "discarding database file",
-					  new_charset, old_charset);
+				g_set_error(error, db_quark(), 0,
+					    "Existing database has charset "
+					    "\"%s\" instead of \"%s\"; "
+					    "discarding database file",
+					    new_charset, old_charset);
 				return false;
 			}
 		} else {

src/decoder/_flac_common.c

@@ -40,53 +40,57 @@ flac_data_init(struct flac_data *data, struct decoder * decoder,
 	data->tag = NULL;
 }
 
-static bool
+static void
 flac_find_float_comment(const FLAC__StreamMetadata *block,
-			const char *cmnt, float *fl)
+			const char *cmnt, float *fl, bool *found_r)
 {
-	int offset =
-	    FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0, cmnt);
-
-	if (offset >= 0) {
-		size_t pos = strlen(cmnt) + 1;	/* 1 is for '=' */
-		int len = block->data.vorbis_comment.comments[offset].length
-		    - pos;
-		if (len > 0) {
-			unsigned char tmp;
-			unsigned char *p = &(block->data.vorbis_comment.
-					     comments[offset].entry[pos]);
-			tmp = p[len];
-			p[len] = '\0';
-			*fl = (float)atof((char *)p);
-			p[len] = tmp;
-
-			return true;
-		}
-	}
+	int offset;
+	size_t pos;
+	int len;
+	unsigned char tmp, *p;
+
+	offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, 0,
+								     cmnt);
+	if (offset < 0)
+		return;
 
-	return false;
+	pos = strlen(cmnt) + 1; /* 1 is for '=' */
+	len = block->data.vorbis_comment.comments[offset].length - pos;
+	if (len <= 0)
+		return;
+
+	p = &block->data.vorbis_comment.comments[offset].entry[pos];
+	tmp = p[len];
+	p[len] = '\0';
+	*fl = (float)atof((char *)p);
+	p[len] = tmp;
+
+	*found_r = true;
 }
 
-/* replaygain stuff by AliasMrJones */
 static void
 flac_parse_replay_gain(const FLAC__StreamMetadata *block,
 		       struct flac_data *data)
 {
-	bool found;
+	bool found = false;
 
 	if (data->replay_gain_info)
 		replay_gain_info_free(data->replay_gain_info);
 
 	data->replay_gain_info = replay_gain_info_new();
 
-	found = flac_find_float_comment(block, "replaygain_album_gain",
-					&data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain) ||
-		flac_find_float_comment(block, "replaygain_album_peak",
-					&data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak) ||
-		flac_find_float_comment(block, "replaygain_track_gain",
-					&data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain) ||
-		flac_find_float_comment(block, "replaygain_track_peak",
-					&data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak);
+	flac_find_float_comment(block, "replaygain_album_gain",
+				&data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].gain,
+				&found);
+	flac_find_float_comment(block, "replaygain_album_peak",
+				&data->replay_gain_info->tuples[REPLAY_GAIN_ALBUM].peak,
+				&found);
+	flac_find_float_comment(block, "replaygain_track_gain",
+				&data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].gain,
+				&found);
+	flac_find_float_comment(block, "replaygain_track_peak",
+				&data->replay_gain_info->tuples[REPLAY_GAIN_TRACK].peak,
+				&found);
 
 	if (!found) {
 		replay_gain_info_free(data->replay_gain_info);
@@ -106,25 +110,27 @@ flac_comment_value(const FLAC__StreamMetadata_VorbisComment_Entry *entry,
 	size_t char_tnum_length = 0;
 	const char *comment = (const char*)entry->entry;
 
-	if (entry->length > name_length &&
-	    g_ascii_strncasecmp(comment, name, name_length) == 0) {
-	        if (char_tnum != NULL) {
-	            char_tnum_length = strlen(char_tnum);
-		    if (entry->length > name_length + char_tnum_length + 2 &&
-		        comment[name_length] == '[' &&
-		        g_ascii_strncasecmp(comment + name_length + 1,
-			char_tnum, char_tnum_length) == 0 &&
-			comment[name_length + char_tnum_length + 1] == ']')
-			    name_length = name_length + char_tnum_length + 2;
-		    else if (entry->length > name_length + char_tnum_length &&
-		        g_ascii_strncasecmp(comment + name_length,
-		        char_tnum, char_tnum_length) == 0)
-			    name_length = name_length + char_tnum_length;
-	        }
-	        if (comment[name_length] == '=') {
-		    *length_r = entry->length - name_length - 1;
-		    return comment + name_length + 1;
-		}
+	if (entry->length <= name_length ||
+	    g_ascii_strncasecmp(comment, name, name_length) != 0)
+		return NULL;
+
+	if (char_tnum != NULL) {
+		char_tnum_length = strlen(char_tnum);
+		if (entry->length > name_length + char_tnum_length + 2 &&
+		    comment[name_length] == '[' &&
+		    g_ascii_strncasecmp(comment + name_length + 1,
+					char_tnum, char_tnum_length) == 0 &&
+		    comment[name_length + char_tnum_length + 1] == ']')
+			name_length = name_length + char_tnum_length + 2;
+		else if (entry->length > name_length + char_tnum_length &&
+			 g_ascii_strncasecmp(comment + name_length,
+					     char_tnum, char_tnum_length) == 0)
+			name_length = name_length + char_tnum_length;
+	}
+
+	if (comment[name_length] == '=') {
+		*length_r = entry->length - name_length - 1;
+		return comment + name_length + 1;
 	}
 
 	return NULL;
@@ -371,13 +377,15 @@ char*
 flac_cue_track(	const char* pathname,
 		const unsigned int tnum)
 {
-	FLAC__StreamMetadata* cs = FLAC__metadata_object_new(FLAC__METADATA_TYPE_CUESHEET);
+	FLAC__bool success;
+	FLAC__StreamMetadata* cs;
 
-	FLAC__metadata_get_cuesheet(pathname, &cs);
-
-	if (cs == NULL)
+	success = FLAC__metadata_get_cuesheet(pathname, &cs);
+	if (!success)
 		return NULL;
 
+	assert(cs != NULL);
+
 	if (cs->data.cue_sheet.num_tracks <= 1)
 	{
 		FLAC__metadata_object_delete(cs);

src/decoder/flac_plugin.c

@@ -394,6 +394,7 @@ flac_decode_internal(struct decoder * decoder,
 	if (!(flac_dec = flac_new()))
 		return;
 	flac_data_init(&data, decoder, input_stream);
+	data.tag = tag_new();
 
 #if defined(FLAC_API_VERSION_CURRENT) && FLAC_API_VERSION_CURRENT > 7
         if(!FLAC__stream_decoder_set_metadata_respond(flac_dec, FLAC__METADATA_TYPE_VORBIS_COMMENT))
@@ -422,8 +423,6 @@ flac_decode_internal(struct decoder * decoder,
 		}
 	}
 
-	data.tag = tag_new();
-
 	if (!flac_process_metadata(flac_dec)) {
 		err = "problem reading metadata";
 		goto fail;

src/decoder/mad_plugin.c

@@ -350,11 +350,11 @@ parse_id3_replay_gain_info(struct id3_tag *tag)
 }
 #endif
 
-#ifdef HAVE_ID3TAG
 static void mp3_parse_id3(struct mp3_data *data, size_t tagsize,
 			  struct tag **mpd_tag,
 			  struct replay_gain_info **replay_gain_info_r)
 {
+#ifdef HAVE_ID3TAG
 	struct id3_tag *id3_tag = NULL;
 	id3_length_t count;
 	id3_byte_t const *id3_data;
@@ -418,8 +418,34 @@ static void mp3_parse_id3(struct mp3_data *data, size_t tagsize,
 	id3_tag_delete(id3_tag);
 
 	g_free(allocated);
-}
+#else /* !HAVE_ID3TAG */
+	(void)mpd_tag;
+	(void)replay_gain_info_r;
+
+	/* This code is enabled when libid3tag is disabled.  Instead
+	   of parsing the ID3 frame, it just skips it. */
+
+	mad_stream_skip(&data->stream, tagsize);
 #endif
+}
+
+#ifndef HAVE_ID3TAG
+/**
+ * This function emulates libid3tag when it is disabled.  Instead of
+ * doing a real analyzation of the frame, it just checks whether the
+ * frame begins with the string "ID3".  If so, it returns the full
+ * length.
+ */
+static signed long
+id3_tag_query(const void *p0, size_t length)
+{
+	const char *p = p0;
+
+	return length > 3 && memcmp(p, "ID3", 3) == 0
+		? length
+		: 0;
+}
+#endif /* !HAVE_ID3TAG */
 
 static enum mp3_action
 decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag,
@@ -433,7 +459,6 @@ decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag,
 			return DECODE_BREAK;
 	}
 	if (mad_header_decode(&data->frame.header, &data->stream)) {
-#ifdef HAVE_ID3TAG
 		if ((data->stream).error == MAD_ERROR_LOSTSYNC &&
 		    (data->stream).this_frame) {
 			signed long tagsize = id3_tag_query((data->stream).
@@ -454,7 +479,6 @@ decode_next_frame_header(struct mp3_data *data, G_GNUC_UNUSED struct tag **tag,
 				return DECODE_CONT;
 			}
 		}
-#endif
 		if (MAD_RECOVERABLE((data->stream).error)) {
 			return DECODE_SKIP;
 		} else {
@@ -493,7 +517,6 @@ decodeNextFrame(struct mp3_data *data)
 			return DECODE_BREAK;
 	}
 	if (mad_frame_decode(&data->frame, &data->stream)) {
-#ifdef HAVE_ID3TAG
 		if ((data->stream).error == MAD_ERROR_LOSTSYNC) {
 			signed long tagsize = id3_tag_query((data->stream).
 							    this_frame,
@@ -506,7 +529,6 @@ decodeNextFrame(struct mp3_data *data)
 				return DECODE_CONT;
 			}
 		}
-#endif
 		if (MAD_RECOVERABLE((data->stream).error)) {
 			return DECODE_SKIP;
 		} else {

src/decoder/vorbis_plugin.c

@@ -383,7 +383,7 @@ vorbis_tag_dup(const char *file)
 		return NULL;
 	}
 
-	if (ov_open(fp, &vf, NULL, 0) < 0) {
+	if (ov_test_callbacks(fp, &vf, NULL, 0, OV_CALLBACKS_STREAMONLY) < 0) {
 		fclose(fp);
 		return NULL;
 	}

src/directory.c

@@ -73,9 +73,14 @@ directory_prune_empty(struct directory *directory)
 	struct dirvec *dv = &directory->children;
 
 	for (i = dv->nr; --i >= 0; ) {
-		directory_prune_empty(dv->base[i]);
-		if (directory_is_empty(dv->base[i]))
-			dirvec_delete(dv, dv->base[i]);
+		struct directory *child = dv->base[i];
+
+		directory_prune_empty(child);
+
+		if (directory_is_empty(child)) {
+			dirvec_delete(dv, child);
+			directory_free(child);
+		}
 	}
 	if (!dv->nr)
 		dirvec_destroy(dv);

src/log.c

@@ -259,12 +259,8 @@ void log_init(bool verbose, bool use_stdout)
 			log_init_syslog();
 #endif
 		} else {
-			char *path = parsePath(param->value);
-			g_free(param->value);
-
-			if (path == NULL)
-				g_error("error parsing \"%s\" at line %i\n",
-					CONF_LOG_FILE, param->line);
+			const char *path = config_get_path(CONF_LOG_FILE);
+			assert(path != NULL);
 
 			log_init_file(path, param->line);
 		}

src/output/httpd_output_plugin.c

@@ -28,15 +28,12 @@
 
 #include <assert.h>
 
+#include <sys/types.h>
 #include <netinet/in.h>
 #include <netdb.h>
 #include <unistd.h>
 #include <errno.h>
 
-#ifdef HAVE_OSX
-#include <sys/types.h>
-#endif
-
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "httpd_output"
 

src/output/shout_plugin.c

@@ -448,8 +448,15 @@ my_shout_play(void *data, const void *chunk, size_t size, GError **error)
 static bool
 my_shout_pause(void *data)
 {
+	struct shout_data *sd = (struct shout_data *)data;
 	static const char silence[1020];
 
+	if (shout_delay(sd->shout_conn) > 500) {
+		/* cap the latency for unpause */
+		g_usleep(500000);
+		return true;
+	}
+
 	return my_shout_play(data, silence, sizeof(silence), NULL);
 }
 

src/output_all.c

@@ -439,8 +439,6 @@ audio_output_all_cancel(void)
 {
 	unsigned int i;
 
-	audio_output_all_update();
-
 	/* send the cancel() command to all audio outputs */
 
 	for (i = 0; i < num_audio_outputs; ++i) {

src/output_control.c

@@ -76,6 +76,17 @@ audio_output_open(struct audio_output *ao,
 	    audio_format_equals(audio_format, &ao->in_audio_format)) {
 		assert(ao->pipe == mp);
 
+		if (ao->pause) {
+			/* unpause with the CANCEL command; this is a
+			   hack, but suits well for forcing the thread
+			   to leave the ao_pause() thread, and we need
+			   to flush the device buffer anyway */
+
+			/* we're not using audio_output_cancel() here,
+			   because that function is asynchronous */
+			ao_command(ao, AO_COMMAND_CANCEL);
+		}
+
 		return true;
 	}
 

src/output_init.c

@@ -109,6 +109,7 @@ audio_output_init(struct audio_output *ao, const struct config_param *param,
 	ao->plugin = plugin;
 	ao->enabled = config_get_block_bool(param, "enabled", true);
 	ao->open = false;
+	ao->pause = false;
 	ao->fail_timer = NULL;
 
 	pcm_convert_init(&ao->convert_state);

src/output_internal.h

@@ -81,6 +81,12 @@ struct audio_output {
 	bool open;
 
 	/**
+	 * Is the device paused?  i.e. the output thread is in the
+	 * ao_pause() loop.
+	 */
+	bool pause;
+
+	/**
 	 * If not NULL, the device has failed, and this timer is used
 	 * to estimate how long it should stay disabled (unless
 	 * explicitly reopened with "play").

src/output_thread.c

@@ -165,6 +165,7 @@ static void ao_pause(struct audio_output *ao)
 	bool ret;
 
 	ao_plugin_cancel(ao->plugin, ao->data);
+	ao->pause = true;
 	ao_command_finished(ao);
 
 	do {
@@ -174,6 +175,8 @@ static void ao_pause(struct audio_output *ao)
 			break;
 		}
 	} while (ao->command == AO_COMMAND_NONE);
+
+	ao->pause = false;
 }
 
 static gpointer audio_output_task(gpointer arg)
@@ -246,7 +249,11 @@ static gpointer audio_output_task(gpointer arg)
 
 		case AO_COMMAND_PAUSE:
 			ao_pause(ao);
-			break;
+			/* don't "break" here: this might cause
+			   ao_play() to be called when command==CLOSE
+			   ends the paused state - "continue" checks
+			   the new command first */
+			continue;
 
 		case AO_COMMAND_CANCEL:
 			ao->chunk = NULL;

src/song.c

@@ -49,6 +49,7 @@ song_alloc(const char *url, struct directory *parent)
 	song->tag = NULL;
 	memcpy(song->url, url, urllen + 1);
 	song->parent = parent;
+	song->mtime = 0;
 
 	return song;
 }

src/tag_ape.c

@@ -22,6 +22,7 @@
 
 #include <glib.h>
 
+#include <assert.h>
 #include <stdio.h>
 
 struct tag *
@@ -86,15 +87,18 @@ tag_ape_load(const char *file)
 
 	/* find beginning of ape tag */
 	tagLen = GUINT32_FROM_LE(footer.length);
-	if (tagLen < sizeof(footer))
+	if (tagLen <= sizeof(footer) + 10)
+		goto fail;
+	if (tagLen > 1024 * 1024)
+		/* refuse to load more than one megabyte of tag data */
 		goto fail;
 	if (fseek(fp, size - tagLen, SEEK_SET))
 		goto fail;
 
 	/* read tag into buffer */
 	tagLen -= sizeof(footer);
-	if (tagLen <= 0)
-		goto fail;
+	assert(tagLen > 10);
+
 	buffer = g_malloc(tagLen);
 	if (fread(buffer, 1, tagLen, fp) != tagLen)
 		goto fail;
@@ -112,7 +116,7 @@ tag_ape_load(const char *file)
 
 		/* get the key */
 		key = p;
-		while (tagLen - size > 0 && *p != '\0') {
+		while (tagLen > size && *p != '\0') {
 			p++;
 			tagLen--;
 		}

src/update.c

@@ -430,7 +430,7 @@ update_container_file(	struct directory* directory,
 {
 	char* vtrack = NULL;
 	unsigned int tnum = 0;
-	const char* pathname = map_directory_child_fs(directory, name);
+	char* pathname = map_directory_child_fs(directory, name);
 	struct directory* contdir = dirvec_find(&directory->children, name);
 
 	// directory exists already
@@ -446,8 +446,10 @@ update_container_file(	struct directory* directory,
 
 			modified = true;
 		}
-		else
+		else {
+			g_free(pathname);
 			return true;
+		}
 	}
 
 	contdir = make_subdir(directory, name);
@@ -473,6 +475,8 @@ update_container_file(	struct directory* directory,
 		g_free(vtrack);
 	}
 
+	g_free(pathname);
+
 	if (tnum == 1)
 	{
 		delete_directory(contdir);
@@ -498,7 +502,8 @@ update_regular_file(struct directory *directory,
 	{
 		struct song* song = songvec_find(&directory->songs, name);
 
-		if (plugin->container_scan != NULL)
+		if (!(song != NULL && st->st_mtime == song->mtime) &&
+			plugin->container_scan != NULL)
 		{
 			if (update_container_file(directory, name, st, plugin))
 			{
@@ -767,7 +772,6 @@ static void * update_task(void *_path)
 {
 	if (_path != NULL && !isRootDirectory(_path)) {
 		updatePath((char *)_path);
-		g_free(_path);
 	} else {
 		struct directory *directory = db_get_root();
 		struct stat st;
@@ -776,6 +780,8 @@ static void * update_task(void *_path)
 			updateDirectory(directory, &st);
 	}
 
+	g_free(_path);
+
 	if (modified || !db_exists())
 		db_save();