release v0.19.12

These are other popular clients. In particular, VLC is available on
mobile devices.

Signed-off-by: Ben Boeckel <mathstuf@gmail.com>

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 +1,34 @@
+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
+  - smbclient: fix DFF playback
+* decoder
+  - ffmpeg: improve seeking accuracy
+  - fix stuck stream tags
+* encoder
+  - opus: fix bogus granulepos
+* output
+  - fix failure to open device right after booting
+* neighbor
+  - nfs: fix deadlock when connecting
+* fix "single" mode breakage due to queue edits
+
 ver 0.19.9 (2015/02/06)
 * decoder
   - dsdiff, dsf: raise ID3 tag limit to 1 MB

configure.ac

@@ -1,10 +1,10 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.19.9, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.19.12, musicpd-dev-team@lists.sourceforge.net)
 
 VERSION_MAJOR=0
 VERSION_MINOR=19
-VERSION_REVISION=9
+VERSION_REVISION=12
 VERSION_EXTRA=0
 
 AC_CONFIG_SRCDIR([src/Main.cxx])

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/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/PlayerThread.cxx

@@ -612,6 +612,12 @@ Player::ProcessCommand()
 
 		queued = true;
 		pc.CommandFinished();
+
+		pc.Unlock();
+		if (dc.LockIsIdle())
+			StartDecoder(*new MusicPipe());
+		pc.Lock();
+
 		break;
 
 	case PlayerCommand::PAUSE:

src/decoder/DecoderAPI.cxx

@@ -433,8 +433,11 @@ update_stream_tag(Decoder &decoder, InputStream *is)
 
 		/* no stream tag present - submit the song tag
 		   instead */
-		decoder.song_tag = nullptr;
-	}
+	} else
+		/* discard the song tag; we don't need it */
+		delete decoder.song_tag;
+
+	decoder.song_tag = nullptr;
 
 	delete decoder.stream_tag;
 	decoder.stream_tag = tag;
@@ -566,7 +569,7 @@ decoder_tag(Decoder &decoder, InputStream *is,
 	/* save the tag */
 
 	delete decoder.decoder_tag;
-	decoder.decoder_tag = new Tag(tag);
+	decoder.decoder_tag = new Tag(std::move(tag));
 
 	/* check for a new stream tag */
 

src/decoder/DecoderThread.cxx

@@ -380,7 +380,11 @@ decoder_run_song(DecoderControl &dc,
 		 const DetachedSong &song, const char *uri, Path path_fs)
 {
 	Decoder decoder(dc, dc.start_time.IsPositive(),
-			new Tag(song.GetTag()));
+			/* pass the song tag only if it's
+			   authoritative, i.e. if it's a local file -
+			   tags on "stream" songs are just remembered
+			   from the last time we played it*/
+			song.IsFile() ? new Tag(song.GetTag()) : nullptr);
 	int ret;
 
 	dc.state = DecoderState::START;

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;
 }
 
@@ -305,20 +314,60 @@ copy_interleave_frame(const AVCodecContext *codec_context,
 	return data_size;
 }
 
+/**
+ * Convert AVPacket::pts to a stream-relative time stamp (still in
+ * AVStream::time_base units).  Returns a negative value on error.
+ */
+gcc_pure
+static int64_t
+StreamRelativePts(const AVPacket &packet, const AVStream &stream)
+{
+	auto pts = packet.pts;
+	if (pts < 0 || pts == int64_t(AV_NOPTS_VALUE))
+		return -1;
+
+	auto start = start_time_fallback(stream);
+	return pts - start;
+}
+
+/**
+ * Convert a non-negative stream-relative time stamp in
+ * AVStream::time_base units to a PCM frame number.
+ */
+gcc_pure
+static uint64_t
+PtsToPcmFrame(uint64_t pts, const AVStream &stream,
+	      const AVCodecContext &codec_context)
+{
+	return av_rescale_q(pts, stream.time_base, codec_context.time_base);
+}
+
+/**
+ * @param min_frame skip all data before this PCM frame number; this
+ * is used after seeking to skip data in an AVPacket until the exact
+ * desired time stamp has been reached
+ */
 static DecoderCommand
 ffmpeg_send_packet(Decoder &decoder, InputStream &is,
 		   const AVPacket *packet,
 		   AVCodecContext *codec_context,
 		   const AVStream *stream,
 		   AVFrame *frame,
+		   uint64_t min_frame, size_t pcm_frame_size,
 		   uint8_t **buffer, int *buffer_size)
 {
-	if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE) {
-		auto start = start_time_fallback(*stream);
-		if (packet->pts >= start)
+	size_t skip_bytes = 0;
+
+	const auto pts = StreamRelativePts(*packet, *stream);
+	if (pts >= 0) {
+		if (min_frame > 0) {
+			auto cur_frame = PtsToPcmFrame(pts, *stream,
+						       *codec_context);
+			if (cur_frame < min_frame)
+				skip_bytes = pcm_frame_size * (min_frame - cur_frame);
+		} else
 			decoder_timestamp(decoder,
-					  time_from_ffmpeg(packet->pts - start,
-							   stream->time_base));
+					  time_from_ffmpeg(pts, stream->time_base));
 	}
 
 	AVPacket packet2 = *packet;
@@ -354,8 +403,20 @@ ffmpeg_send_packet(Decoder &decoder, InputStream &is,
 		if (audio_size <= 0)
 			continue;
 
+		const uint8_t *data = output_buffer;
+		if (skip_bytes > 0) {
+			if (skip_bytes >= size_t(audio_size)) {
+				skip_bytes -= audio_size;
+				continue;
+			}
+
+			data += skip_bytes;
+			audio_size -= skip_bytes;
+			skip_bytes = 0;
+		}
+
 		cmd = decoder_data(decoder, is,
-				   output_buffer, audio_size,
+				   data, audio_size,
 				   codec_context->bit_rate / 1000);
 	}
 	return cmd;
@@ -559,6 +620,8 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 	uint8_t *interleaved_buffer = nullptr;
 	int interleaved_buffer_size = 0;
 
+	uint64_t min_frame = 0;
+
 	DecoderCommand cmd;
 	do {
 		AVPacket packet;
@@ -566,13 +629,15 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 			/* end of file */
 			break;
 
-		if (packet.stream_index == audio_stream)
+		if (packet.stream_index == audio_stream) {
 			cmd = ffmpeg_send_packet(decoder, input,
 						 &packet, codec_context,
 						 av_stream,
 						 frame,
+						 min_frame, audio_format.GetFrameSize(),
 						 &interleaved_buffer, &interleaved_buffer_size);
-		else
+			min_frame = 0;
+		} else
 			cmd = decoder_get_command(decoder);
 
 		av_free_packet(&packet);
@@ -583,11 +648,16 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 					       av_stream->time_base) +
 				start_time_fallback(*av_stream);
 
+			/* AVSEEK_FLAG_BACKWARD asks FFmpeg to seek to
+			   the packet boundary before the seek time
+			   stamp, not after */
+
 			if (av_seek_frame(format_context, audio_stream, where,
-					  AVSEEK_FLAG_ANY) < 0)
+					  AVSEEK_FLAG_ANY|AVSEEK_FLAG_BACKWARD) < 0)
 				decoder_seek_error(decoder);
 			else {
 				avcodec_flush_buffers(codec_context);
+				min_frame = decoder_seek_where_frame(decoder);
 				decoder_command_finished(decoder);
 			}
 		}

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
-		? SignedSongTime::FromMS(ti->length)
+	const int length = ti->play_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)
-		gme_set_fade(emu, ti->length);
+	if (length > 0)
+		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 {
-		bits = 24,
-	};
-
-	const int clip_min = -1 << (bits - 1);
-	const int clip_max = (1 << (bits - 1)) - 1;
+	constexpr int bits = MpcdecSampleTraits::BITS;
+	constexpr auto clip_min = MpcdecSampleTraits::MIN;
+	constexpr auto clip_max = MpcdecSampleTraits::MAX;
 
 #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)
-		val = clip_min;
-	else if (val > clip_max)
-		val = clip_max;
-
-	return val;
+	return Clamp(val, clip_min, clip_max);
 }
 
 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/OpusEncoderPlugin.cxx

@@ -66,7 +66,7 @@ struct opus_encoder {
 
 	ogg_int64_t granulepos;
 
-	opus_encoder():encoder(opus_encoder_plugin) {}
+	opus_encoder():encoder(opus_encoder_plugin), granulepos(0) {}
 };
 
 static constexpr Domain opus_encoder_domain("opus_encoder");

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/input/plugins/CurlInputPlugin.cxx

@@ -453,6 +453,8 @@ CurlInputStream::RequestDone(CURLcode result, long status)
 		SeekDone();
 	else if (!IsReady())
 		SetReady();
+	else
+		cond.broadcast();
 }
 
 static void

src/input/plugins/SmbclientInputPlugin.cxx

@@ -132,6 +132,7 @@ SmbclientInputStream::Read(void *ptr, size_t read_size, Error &error)
 		nbytes = 0;
 	}
 
+	offset += nbytes;
 	return nbytes;
 }
 

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/OutputControl.cxx

@@ -184,7 +184,8 @@ AudioOutput::LockUpdate(const AudioFormat audio_format,
 	const ScopeLock protect(mutex);
 
 	if (enabled && really_enabled) {
-		if (fail_timer.Check(REOPEN_AFTER * 1000)) {
+		if (!fail_timer.IsDefined() ||
+		    fail_timer.Check(REOPEN_AFTER * 1000)) {
 			return Open(audio_format, mp);
 		}
 	} else if (IsOpen())

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/storage/plugins/NfsStorage.cxx

@@ -177,6 +177,8 @@ private:
 				mutex.unlock();
 				DeferredMonitor::Schedule();
 				mutex.lock();
+				if (state == State::INITIAL)
+					cond.wait(mutex);
 				break;
 
 			case State::CONNECTING:
@@ -188,8 +190,6 @@ private:
 				error.Set(last_error);
 				return false;
 			}
-
-			cond.wait(mutex);
 		}
 	}
 

src/system/PeriodClock.hxx

@@ -63,7 +63,7 @@ protected:
   }
 
 public:
-  bool IsDefined() const {
+  constexpr bool IsDefined() const {
     return last != 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') {
-			p++;
-			remaining--;
-		}
-		p++;
-		remaining--;
+		const char *key_end = (const char *)memchr(p, '\0', remaining);
+		if (key_end == nullptr)
+			break;
+
+		p = key_end + 1;
+		remaining -= p - key;
 
 		/* get the value */
 		if (remaining < size)

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 */

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
-
-# 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
+LimitRTTIME=infinity
 
 [Install]
 WantedBy=multi-user.target