release v0.19.13

Fixes disappearing duration of remote songs during playback.

See http://bugs.musicpd.org/view.php?id=4492

Makefile.am

@@ -1219,6 +1219,7 @@ liboutput_plugins_a_SOURCES = \
 
 MIXER_LIBS = \
 	libmixer_plugins.a \
+	$(ALSA_LIBS) \
 	$(PULSE_LIBS)
 
 MIXER_API_SRC = \

NEWS

@@ -1,3 +1,27 @@
+ver 0.19.13 (2016/02/23)
+* tags
+  - aiff, riff: fix ID3 chunk padding
+* decoder
+  - ffmpeg: support the TAK codec
+* fix disappearing duration of remote songs during playback
+* initialize supplementary groups with glibc 2.19+
+
+ver 0.19.12 (2015/12/15)
+* fix assertion failure on malformed UTF-8 tag
+* fix build failure on non-Linux systems
+* fix LimitRTTIME in systemd unit file
+
+ver 0.19.11 (2015/10/27)
+* tags
+  - ape: fix buffer overflow
+* decoder
+  - ffmpeg: fix crash due to wrong avio_alloc_context() call
+  - gme: don't loop forever, fall back to GME's default play length
+* encoder
+  - flac: fix crash with 32 bit playback
+* mixer
+  - fix mixer lag after enabling/disabling output
+
 ver 0.19.10 (2015/06/21)
 * input
   - curl: fix deadlock on small responses

configure.ac

@@ -1,10 +1,10 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.19.10, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.19.13, musicpd-dev-team@lists.sourceforge.net)
 
 VERSION_MAJOR=0
 VERSION_MINOR=19
-VERSION_REVISION=10
+VERSION_REVISION=13
 VERSION_EXTRA=0
 
 AC_CONFIG_SRCDIR([src/Main.cxx])
@@ -206,6 +206,7 @@ if test x$host_is_linux = xyes; then
 fi
 
 AC_CHECK_FUNCS(getpwnam_r getpwuid_r)
+AC_CHECK_FUNCS(initgroups)
 AC_CHECK_FUNCS(strndup)
 
 if test x$host_is_linux = xyes; then

doc/user.xml

@@ -1190,6 +1190,58 @@ database {
         plugin).
       </para>
     </section>
+
+    <section id="realtime">
+      <title>Real-Time Scheduling</title>
+
+      <para>
+        On Linux, <application>MPD</application> attempts to configure
+        <ulink
+        url="https://en.wikipedia.org/wiki/Real-time_computing">real-time
+        scheduling</ulink> for some threads that benefit from it.
+      </para>
+
+      <para>
+        This is only possible you allow <application>MPD</application>
+        to do it.  This privilege is controlled by
+        <varname>RLIMIT_RTPRIO</varname>
+        <varname>RLIMIT_RTTIME</varname>.  You can configure this
+        privilege with <command>ulimit</command> before launching
+        <application>MPD</application>:
+      </para>
+
+      <programlisting>ulimit -HS -r 50; mpd</programlisting>
+
+      <para>
+        Or you can use the <command>prlimit</command> program from the
+        <application>util-linux</application> package:
+      </para>
+
+      <programlisting>prlimit --rtprio=50 --rttime=unlimited mpd</programlisting>
+
+      <para>
+        The <application>systemd</application> service file shipped
+        with <application>MPD</application> comes with this setting.
+      </para>
+
+      <para>
+        This works only if the Linux kernel was compiled with
+        <varname>CONFIG_RT_GROUP_SCHED</varname> disabled.  Use the
+        following command to check this option for your current
+        kernel:
+      </para>
+
+      <programlisting>zgrep ^CONFIG_RT_GROUP_SCHED /proc/config.gz</programlisting>
+
+      <note>
+        <para>
+          There is a rumor that real-time scheduling improves audio
+          quality.  That is not true.  All it does is reduce the
+          probability of skipping (audio buffer xruns) when the
+          computer is under heavy load.
+        </para>
+      </note>
+    </section>
   </chapter>
 
   <chapter id="use">
@@ -2595,7 +2647,8 @@ buffer_size: 16384</programlisting>
           / <ulink
           url="http://icecast.org/"><application>IceCast</application></ulink>.
           HTTP streaming clients like
-          <application>mplayer</application> can connect to it.
+          <application>mplayer</application>, <application>VLC</application>,
+          and <application>mpv</application> can connect to it.
         </para>
 
         <para>

src/DetachedSong.hxx

@@ -188,6 +188,14 @@ public:
 		tag = std::move(other.tag);
 	}
 
+	/**
+	 * Similar to the MoveTagFrom(), but move only the #TagItem
+	 * array.
+	 */
+	void MoveTagItemsFrom(DetachedSong &&other) {
+		tag.MoveItemsFrom(std::move(other.tag));
+	}
+
 	time_t GetLastModified() const {
 		return mtime;
 	}

src/Main.cxx

@@ -54,7 +54,6 @@
 #include "system/FatalError.hxx"
 #include "util/UriUtil.hxx"
 #include "util/Error.hxx"
-#include "util/Domain.hxx"
 #include "thread/Id.hxx"
 #include "thread/Slack.hxx"
 #include "lib/icu/Init.hxx"
@@ -123,8 +122,6 @@
 static constexpr unsigned DEFAULT_BUFFER_SIZE = 4096;
 static constexpr unsigned DEFAULT_BUFFER_BEFORE_PLAY = 10;
 
-static constexpr Domain main_domain("main");
-
 #ifdef ANDROID
 Context *context;
 #endif
@@ -633,7 +630,7 @@ static int mpd_main_after_fork(struct options options)
 					 config_get_unsigned(CONF_AUTO_UPDATE_DEPTH,
 							     INT_MAX));
 #else
-		FormatWarning(main_domain,
+		FormatWarning(config_domain,
 			      "inotify: auto_update was disabled. enable during compilation phase");
 #endif
 	}

src/decoder/plugins/FfmpegDecoderPlugin.cxx

@@ -92,14 +92,14 @@ struct AvioStream {
 
 	AVIOContext *io;
 
-	unsigned char buffer[8192];
-
 	AvioStream(Decoder *_decoder, InputStream &_input)
 		:decoder(_decoder), input(_input), io(nullptr) {}
 
 	~AvioStream() {
-		if (io != nullptr)
+		if (io != nullptr) {
+			av_free(io->buffer);
 			av_free(io);
+		}
 	}
 
 	bool Open();
@@ -153,11 +153,20 @@ mpd_ffmpeg_stream_seek(void *opaque, int64_t pos, int whence)
 bool
 AvioStream::Open()
 {
-	io = avio_alloc_context(buffer, sizeof(buffer),
+	constexpr size_t BUFFER_SIZE = 8192;
+	auto buffer = (unsigned char *)av_malloc(BUFFER_SIZE);
+	if (buffer == nullptr)
+		return false;
+
+	io = avio_alloc_context(buffer, BUFFER_SIZE,
 				false, this,
 				mpd_ffmpeg_stream_read, nullptr,
 				input.IsSeekable()
 				? mpd_ffmpeg_stream_seek : nullptr);
+	/* If avio_alloc_context() fails, who frees the buffer?  The
+	   libavformat API documentation does not specify this, it
+	   only says that AVIOContext.buffer must be freed in the end,
+	   however no AVIOContext exists in that failure code path. */
 	return io != nullptr;
 }
 
@@ -729,7 +738,7 @@ static const char *const ffmpeg_suffixes[] = {
 	"mve", "mvi", "mxf", "nc", "nsv", "nut", "nuv", "oga", "ogm", "ogv",
 	"ogx", "oma", "ogg", "omg", "opus", "psp", "pva", "qcp", "qt", "r3d", "ra",
 	"ram", "rl2", "rm", "rmvb", "roq", "rpl", "rvc", "shn", "smk", "snd",
-	"sol", "son", "spx", "str", "swf", "tgi", "tgq", "tgv", "thp", "ts",
+	"sol", "son", "spx", "str", "swf", "tak", "tgi", "tgq", "tgv", "thp", "ts",
 	"tsp", "tta", "xa", "xvid", "uv", "uv2", "vb", "vid", "vob", "voc",
 	"vp6", "vmd", "wav", "webm", "wma", "wmv", "wsaud", "wsvga", "wv",
 	"wve",

src/decoder/plugins/GmeDecoderPlugin.cxx

@@ -156,8 +156,11 @@ gme_file_decode(Decoder &decoder, Path path_fs)
 		return;
 	}
 
-	const SignedSongTime song_len = ti->length > 0
+	const int length = ti->play_length;
-		? SignedSongTime::FromMS(ti->length)
+	gme_free_info(ti);
+
+	const SignedSongTime song_len = length > 0
+		? SignedSongTime::FromMS(length)
 		: SignedSongTime::Negative();
 
 	/* initialize the MPD decoder */
@@ -168,7 +171,6 @@ gme_file_decode(Decoder &decoder, Path path_fs)
 				       SampleFormat::S16, GME_CHANNELS,
 				       error)) {
 		LogError(error);
-		gme_free_info(ti);
 		gme_delete(emu);
 		return;
 	}
@@ -179,8 +181,8 @@ gme_file_decode(Decoder &decoder, Path path_fs)
 	if (gme_err != nullptr)
 		LogWarning(gme_domain, gme_err);
 
-	if (ti->length > 0)
+	if (length > 0)
-		gme_set_fade(emu, ti->length);
+		gme_set_fade(emu, length);
 
 	/* play */
 	DecoderCommand cmd;
@@ -196,16 +198,17 @@ gme_file_decode(Decoder &decoder, Path path_fs)
 		if (cmd == DecoderCommand::SEEK) {
 			unsigned where = decoder_seek_time(decoder).ToMS();
 			gme_err = gme_seek(emu, where);
-			if (gme_err != nullptr)
+			if (gme_err != nullptr) {
 				LogWarning(gme_domain, gme_err);
-			decoder_command_finished(decoder);
+				decoder_seek_error(decoder);
+			} else
+				decoder_command_finished(decoder);
 		}
 
 		if (gme_track_ended(emu))
 			break;
 	} while (cmd != DecoderCommand::STOP);
 
-	gme_free_info(ti);
 	gme_delete(emu);
 }
 
@@ -236,9 +239,9 @@ gme_scan_file(Path path_fs,
 
 	assert(ti != nullptr);
 
-	if (ti->length > 0)
+	if (ti->play_length > 0)
 		tag_handler_invoke_duration(handler, handler_ctx,
-					    SongTime::FromMS(ti->length));
+					    SongTime::FromMS(ti->play_length));
 
 	if (ti->song != nullptr) {
 		if (gme_track_count(emu) > 1) {

src/decoder/plugins/MpcdecDecoderPlugin.cxx

@@ -22,10 +22,12 @@
 #include "../DecoderAPI.hxx"
 #include "input/InputStream.hxx"
 #include "CheckAudioFormat.hxx"
+#include "pcm/Traits.hxx"
 #include "tag/TagHandler.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "util/Macros.hxx"
+#include "util/Clamp.hxx"
 #include "Log.hxx"
 
 #include <mpc/mpcdec.h>
@@ -42,6 +44,9 @@ struct mpc_decoder_data {
 
 static constexpr Domain mpcdec_domain("mpcdec");
 
+static constexpr SampleFormat mpcdec_sample_format = SampleFormat::S24_P32;
+typedef SampleTraits<mpcdec_sample_format> MpcdecSampleTraits;
+
 static mpc_int32_t
 mpc_read_cb(mpc_reader *reader, void *ptr, mpc_int32_t size)
 {
@@ -91,18 +96,15 @@ mpc_getsize_cb(mpc_reader *reader)
 }
 
 /* this _looks_ performance-critical, don't de-inline -- eric */
-static inline int32_t
+static inline MpcdecSampleTraits::value_type
 mpc_to_mpd_sample(MPC_SAMPLE_FORMAT sample)
 {
 	/* only doing 16-bit audio for now */
-	int32_t val;
+	MpcdecSampleTraits::value_type val;
 
-	enum {
+	constexpr int bits = MpcdecSampleTraits::BITS;
-		bits = 24,
+	constexpr auto clip_min = MpcdecSampleTraits::MIN;
-	};
+	constexpr auto clip_max = MpcdecSampleTraits::MAX;
-
-	const int clip_min = -1 << (bits - 1);
-	const int clip_max = (1 << (bits - 1)) - 1;
 
 #ifdef MPC_FIXED_POINT
 	const int shift = bits - MPC_FIXED_POINT_SCALE_SHIFT;
@@ -117,16 +119,12 @@ mpc_to_mpd_sample(MPC_SAMPLE_FORMAT sample)
 	val = sample * float_scale;
 #endif
 
-	if (val < clip_min)
+	return Clamp(val, clip_min, clip_max);
-		val = clip_min;
-	else if (val > clip_max)
-		val = clip_max;
-
-	return val;
 }
 
 static void
-mpc_to_mpd_buffer(int32_t *dest, const MPC_SAMPLE_FORMAT *src,
+mpc_to_mpd_buffer(MpcdecSampleTraits::pointer_type dest,
+		  const MPC_SAMPLE_FORMAT *src,
 		  unsigned num_samples)
 {
 	while (num_samples-- > 0)
@@ -162,7 +160,7 @@ mpcdec_decode(Decoder &mpd_decoder, InputStream &is)
 	Error error;
 	AudioFormat audio_format;
 	if (!audio_format_init_checked(audio_format, info.sample_freq,
-				       SampleFormat::S24_P32,
+				       mpcdec_sample_format,
 				       info.channels, error)) {
 		LogError(error);
 		mpc_demux_exit(demux);
@@ -214,7 +212,7 @@ mpcdec_decode(Decoder &mpd_decoder, InputStream &is)
 		mpc_uint32_t ret = frame.samples;
 		ret *= info.channels;
 
-		int32_t chunk[ARRAY_SIZE(sample_buffer)];
+		MpcdecSampleTraits::value_type chunk[ARRAY_SIZE(sample_buffer)];
 		mpc_to_mpd_buffer(chunk, sample_buffer, ret);
 
 		long bit_rate = vbr_update_bits * audio_format.sample_rate

src/encoder/plugins/FlacEncoderPlugin.cxx

@@ -157,8 +157,6 @@ flac_encoder_open(Encoder *_encoder, AudioFormat &audio_format, Error &error)
 	struct flac_encoder *encoder = (struct flac_encoder *)_encoder;
 	unsigned bits_per_sample;
 
-	encoder->audio_format = audio_format;
-
 	/* FIXME: flac should support 32bit as well */
 	switch (audio_format.format) {
 	case SampleFormat::S8:
@@ -178,6 +176,8 @@ flac_encoder_open(Encoder *_encoder, AudioFormat &audio_format, Error &error)
 		audio_format.format = SampleFormat::S24_P32;
 	}
 
+	encoder->audio_format = audio_format;
+
 	/* allocate the encoder */
 	encoder->fse = FLAC__stream_encoder_new();
 	if (encoder->fse == nullptr) {

src/encoder/plugins/ShineEncoderPlugin.cxx

@@ -26,7 +26,6 @@
 #include "util/NumberParser.hxx"
 #include "util/DynamicFifoBuffer.hxx"
 #include "util/Error.hxx"
-#include "util/Domain.hxx"
 
 extern "C"
 {
@@ -60,8 +59,6 @@ struct ShineEncoder {
 	bool WriteChunk(bool flush);
 };
 
-static constexpr Domain shine_encoder_domain("shine_encoder");
-
 inline bool
 ShineEncoder::Configure(const config_param &param,
 			 gcc_unused Error &error)

src/filter/plugins/VolumeFilterPlugin.cxx

@@ -26,7 +26,6 @@
 #include "AudioFormat.hxx"
 #include "util/ConstBuffer.hxx"
 #include "util/Error.hxx"
-#include "util/Domain.hxx"
 
 #include <assert.h>
 #include <string.h>
@@ -50,8 +49,6 @@ public:
 				    Error &error) override;
 };
 
-static constexpr Domain volume_domain("pcm_volume");
-
 static Filter *
 volume_filter_init(gcc_unused const config_param &param,
 		   gcc_unused Error &error)

src/mixer/MixerAll.cxx

@@ -25,13 +25,10 @@
 #include "output/Internal.hxx"
 #include "pcm/Volume.hxx"
 #include "util/Error.hxx"
-#include "util/Domain.hxx"
 #include "Log.hxx"
 
 #include <assert.h>
 
-static constexpr Domain mixer_domain("mixer");
-
 static int
 output_mixer_get_volume(const AudioOutput &ao)
 {

src/output/OutputCommand.cxx

@@ -30,6 +30,7 @@
 #include "Internal.hxx"
 #include "PlayerControl.hxx"
 #include "mixer/MixerControl.hxx"
+#include "mixer/Volume.hxx"
 #include "Idle.hxx"
 
 extern unsigned audio_output_state_version;
@@ -47,6 +48,11 @@ audio_output_enable_index(MultipleOutputs &outputs, unsigned idx)
 	ao.enabled = true;
 	idle_add(IDLE_OUTPUT);
 
+	if (ao.mixer != nullptr) {
+		InvalidateHardwareVolume();
+		idle_add(IDLE_MIXER);
+	}
+
 	ao.player_control->UpdateAudio();
 
 	++audio_output_state_version;
@@ -70,6 +76,7 @@ audio_output_disable_index(MultipleOutputs &outputs, unsigned idx)
 	Mixer *mixer = ao.mixer;
 	if (mixer != nullptr) {
 		mixer_close(mixer);
+		InvalidateHardwareVolume();
 		idle_add(IDLE_MIXER);
 	}
 
@@ -94,6 +101,7 @@ audio_output_toggle_index(MultipleOutputs &outputs, unsigned idx)
 		Mixer *mixer = ao.mixer;
 		if (mixer != nullptr) {
 			mixer_close(mixer);
+			InvalidateHardwareVolume();
 			idle_add(IDLE_MIXER);
 		}
 	}

src/output/plugins/PipeOutputPlugin.cxx

@@ -22,7 +22,6 @@
 #include "../OutputAPI.hxx"
 #include "config/ConfigError.hxx"
 #include "util/Error.hxx"
-#include "util/Domain.hxx"
 
 #include <string>
 
@@ -44,8 +43,6 @@ struct PipeOutput {
 	bool Configure(const config_param &param, Error &error);
 };
 
-static constexpr Domain pipe_output_domain("pipe_output");
-
 inline bool
 PipeOutput::Configure(const config_param &param, Error &error)
 {

src/playlist/plugins/XspfPlaylistPlugin.cxx

@@ -25,14 +25,11 @@
 #include "input/InputStream.hxx"
 #include "tag/TagBuilder.hxx"
 #include "util/Error.hxx"
-#include "util/Domain.hxx"
 #include "lib/expat/ExpatParser.hxx"
 #include "Log.hxx"
 
 #include <string.h>
 
-static constexpr Domain xspf_domain("xspf");
-
 /**
  * This is the state object for the GLib XML parser.
  */

src/queue/Playlist.cxx

@@ -37,7 +37,7 @@ playlist::TagModified(DetachedSong &&song)
 
 	DetachedSong &current_song = queue.GetOrder(current);
 	if (song.IsSame(current_song))
-		current_song.MoveTagFrom(std::move(song));
+		current_song.MoveTagItemsFrom(std::move(song));
 
 	queue.ModifyAtOrder(current);
 	queue.IncrementVersion();

src/tag/Aiff.cxx

@@ -84,14 +84,14 @@ aiff_seek_id3(FILE *file)
 			   underflow when casting to off_t */
 			return 0;
 
-		if (size % 2 != 0)
-			/* pad byte */
-			++size;
-
 		if (memcmp(chunk.id, "ID3 ", 4) == 0)
 			/* found it! */
 			return size;
 
+		if (size % 2 != 0)
+			/* pad byte */
+			++size;
+
 		if (fseek(file, size, SEEK_CUR) != 0)
 			return 0;
 	}

src/tag/ApeLoader.cxx

@@ -78,12 +78,12 @@ ape_scan_internal(FILE *fp, ApeTagCallback callback)
 
 		/* get the key */
 		const char *key = p;
-		while (remaining > size && *p != '\0') {
+		const char *key_end = (const char *)memchr(p, '\0', remaining);
-			p++;
+		if (key_end == nullptr)
-			remaining--;
+			break;
-		}
+
-		p++;
+		p = key_end + 1;
-		remaining--;
+		remaining -= p - key;
 
 		/* get the value */
 		if (remaining < size)

src/tag/Riff.cxx

@@ -82,15 +82,15 @@ riff_seek_id3(FILE *file)
 			   underflow when casting to off_t */
 			return 0;
 
-		if (size % 2 != 0)
-			/* pad byte */
-			++size;
-
 		if (memcmp(chunk.id, "id3 ", 4) == 0 ||
 		    memcmp(chunk.id, "ID3 ", 4) == 0)
 			/* found it! */
 			return size;
 
+		if (size % 2 != 0)
+			/* pad byte */
+			++size;
+
 		if (fseek(file, size, SEEK_CUR) != 0)
 			return 0;
 	}

src/tag/Tag.hxx

@@ -80,9 +80,17 @@ struct Tag {
 	Tag &operator=(Tag &&other) {
 		duration = other.duration;
 		has_playlist = other.has_playlist;
+		MoveItemsFrom(std::move(other));
+		return *this;
+	}
+
+	/**
+	 * Similar to the move operator, but move only the #TagItem
+	 * array.
+	 */
+	void MoveItemsFrom(Tag &&other) {
 		std::swap(items, other.items);
 		std::swap(num_items, other.num_items);
-		return *this;
 	}
 
 	/**

src/tag/TagString.cxx

@@ -40,9 +40,9 @@ FindInvalidUTF8(const char *p, const char *const end)
 		/* now call the other SequenceLengthUTF8() overload
 		   which also validates the continuations */
 		const size_t t = SequenceLengthUTF8(p);
-		assert(s == t);
 		if (t == 0)
 			return p;
+		assert(s == t);
 
 		p += s;
 	}

src/unix/Daemon.cxx

@@ -22,7 +22,6 @@
 #include "system/FatalError.hxx"
 #include "fs/AllocatedPath.hxx"
 #include "fs/FileSystem.hxx"
-#include "util/Domain.hxx"
 #include "PidFile.hxx"
 #include "Log.hxx"
 
@@ -37,8 +36,6 @@
 #include <grp.h>
 #endif
 
-static constexpr Domain daemon_domain("daemon");
-
 #ifndef WIN32
 
 /** the Unix user name which MPD runs as */
@@ -113,7 +110,7 @@ daemonize_set_user(void)
 				       (int)user_gid);
 	}
 
-#ifdef _BSD_SOURCE
+#ifdef HAVE_INITGROUPS
 	/* init supplementary groups
 	 * (must be done before we change our uid)
 	 */

systemd/mpd.service.in

@@ -7,16 +7,7 @@ ExecStart=@prefix@/bin/mpd --no-daemon
 
 # allow MPD to use real-time priority 50
 LimitRTPRIO=50
-LimitRTTIME=-1
+LimitRTTIME=infinity
-
-# move MPD to a top-level cgroup, as real-time budget assignment fails
-# in cgroup /system/mpd.service, because /system has a zero real-time
-# budget; see
-# http://www.freedesktop.org/wiki/Software/systemd/MyServiceCantGetRealtime/
-ControlGroup=cpu:/mpd
-
-# assign a real-time budget
-ControlGroupAttribute=cpu.rt_runtime_us 500000
 
 [Install]
 WantedBy=multi-user.target