release v0.18.13

The "current" attribute is a "song order", not a "song position".
This is usually the same - except in random mode.  Fixes Mantis ticket
0004073.

.gitignore

@@ -40,7 +40,7 @@ tags
 .#*
 .stgit*
 src/dsd2pcm/dsd2pcm
-src/win/mpd_win32_rc.rc
+src/win32/mpd_win32_rc.rc
 doc/doxygen.conf
 doc/protocol.html
 doc/protocol

Makefile.am

@@ -151,7 +151,7 @@ src_mpd_SOURCES = \
 	src/IOThread.cxx src/IOThread.hxx \
 	src/Main.cxx src/Main.hxx \
 	src/Instance.cxx src/Instance.hxx \
-	src/Win32Main.cxx \
+	src/win32/Win32Main.cxx \
 	src/GlobalEvents.cxx src/GlobalEvents.hxx \
 	src/Daemon.cxx src/Daemon.hxx \
 	src/AudioCompress/compress.c \
@@ -181,6 +181,7 @@ src_mpd_SOURCES = \
 	src/PlaylistInfo.hxx \
 	src/PlaylistDatabase.cxx src/PlaylistDatabase.hxx \
 	src/PlaylistUpdate.cxx \
+	src/BulkEdit.hxx \
 	src/IdTable.hxx \
 	src/Queue.cxx src/Queue.hxx \
 	src/QueuePrint.cxx src/QueuePrint.hxx \
@@ -210,14 +211,14 @@ src_mpd_SOURCES = \
 # Windows resource file
 #
 
-src/win/mpd_win32_rc.$(OBJEXT): src/win/mpd_win32_rc.rc
+src/win32/mpd_win32_rc.$(OBJEXT): src/win32/mpd_win32_rc.rc
 	$(WINDRES) -i $< -o $@
 
 if HAVE_WINDOWS
-noinst_DATA = src/win/mpd_win32_rc.rc
+noinst_DATA = src/win32/mpd_win32_rc.rc
 
-src_mpd_DEPENDENCIES = src/win/mpd_win32_rc.$(OBJEXT)
+EXTRA_src_mpd_DEPENDENCIES = src/win32/mpd_win32_rc.$(OBJEXT)
-src_mpd_LDFLAGS = -Wl,src/win/mpd_win32_rc.$(OBJEXT)
+src_mpd_LDFLAGS = -Wl,src/win32/mpd_win32_rc.$(OBJEXT)
 endif
 
 if ENABLE_DESPOTIFY
@@ -1218,6 +1219,7 @@ test_dump_playlist_LDADD = \
 	libpcm.a \
 	$(GLIB_LIBS)
 test_dump_playlist_SOURCES = test/dump_playlist.cxx \
+	test/FakeDecoderAPI.cxx \
 	$(DECODER_SRC) \
 	src/Log.cxx \
 	src/IOThread.cxx \
@@ -1271,6 +1273,7 @@ test_read_tags_LDADD = \
 	libutil.a \
 	$(GLIB_LIBS)
 test_read_tags_SOURCES = test/read_tags.cxx \
+	test/FakeDecoderAPI.cxx \
 	src/Log.cxx \
 	src/IOThread.cxx \
 	src/ReplayGainInfo.cxx \
@@ -1630,4 +1633,4 @@ EXTRA_DIST = $(doc_DATA) autogen.sh \
 	test/test_archive_zzip.sh \
 	$(wildcard scripts/*.sh) \
 	$(man_MANS) $(DOCBOOK_FILES) doc/mpdconf.example doc/doxygen.conf \
-	src/win/mpd_win32_rc.rc.in src/win/mpd.ico
+	src/win32/mpd_win32_rc.rc.in src/win32/mpd.ico

NEWS

@@ -1,3 +1,55 @@
+ver 0.18.13 (2014/08/31)
+* protocol
+  - don't change song on "seekcur" in random mode
+* decoder
+  - dsdiff, dsf: fix endless loop on malformed file
+  - ffmpeg: support ffmpeg/libav version 11
+  - gme: fix song duration
+* output
+  - alsa: fix endless loop at end of file in dsd_usb mode
+* fix state file saver
+* fix build failure on Darwin
+
+ver 0.18.12 (2014/07/30)
+* database
+  - proxy: fix build failure with libmpdclient 2.2
+  - proxy: fix add/search and other commands with libmpdclient < 2.9
+* decoder
+  - audiofile: improve responsiveness
+  - audiofile: fix WAV stream playback
+  - dsdiff, dsf: fix stream playback
+  - dsdiff: fix metadata parser bug (uninitialized variables)
+  - faad: estimate song duration for remote files
+  - sndfile: improve responsiveness
+* randomize next song when enabling "random" mode while not playing
+* randomize next song when adding to single-song queue
+
+ver 0.18.11 (2014/05/12)
+* decoder
+  - opus: fix missing song length on high-latency files
+* fix race condition when using GLib event loop (non-Linux)
+
+ver 0.18.10 (2014/04/10)
+* decoder
+  - ffmpeg: fix seeking bug
+  - ffmpeg: handle unknown stream start time
+  - gme: fix memory leak
+  - sndfile: work around libsndfile bug on partial read
+* don't interrupt playback when current song gets deleted
+
+ver 0.18.9 (2014/03/02)
+* protocol
+  - "findadd" requires the "add" permission
+* output
+  - alsa: improved workaround for noise after manual song change
+* decoder
+  - vorbis: fix linker failure when libvorbis/libogg are static
+* encoder
+  - vorbis: fix another linker failure
+* output
+  - pipe: fix hanging child process due to blocked signals
+* fix build failure due to missing signal.h include
+
 ver 0.18.8 (2014/02/07)
 * decoder
   - ffmpeg: support libav v10_alpha1

configure.ac

@@ -1,6 +1,6 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.18.8, mpd-devel@musicpd.org)
+AC_INIT(mpd, 0.18.13, mpd-devel@musicpd.org)
 
 VERSION_MAJOR=0
 VERSION_MINOR=18
@@ -70,7 +70,7 @@ host_is_darwin=no
 case "$host_os" in
 mingw32* | windows*)
 	AC_CONFIG_FILES([
-		src/win/mpd_win32_rc.rc
+		src/win32/mpd_win32_rc.rc
 	])
 	AC_CHECK_TOOL(WINDRES, windres)
 	AM_CPPFLAGS="$AM_CPPFLAGS -DWIN32_LEAN_AND_MEAN"
@@ -1023,7 +1023,7 @@ if test x$enable_tremor = xyes; then
 	fi
 fi
 
-MPD_AUTO_PKG(vorbis, VORBIS, [vorbis vorbisfile ogg],
+MPD_AUTO_PKG(vorbis, VORBIS, [vorbisfile vorbis ogg],
 	[Ogg Vorbis decoder], [libvorbis not found])
 if test x$enable_vorbis = xyes; then
 	AC_DEFINE(ENABLE_VORBIS_DECODER, 1, [Define for Ogg Vorbis support])
@@ -1139,7 +1139,7 @@ fi
 AM_CONDITIONAL(ENABLE_FLAC_ENCODER, test x$enable_flac_encoder = xyes)
 
 dnl ---------------------------- Ogg Vorbis Encoder ---------------------------
-MPD_AUTO_PKG(vorbis_encoder, VORBISENC, [vorbisenc vorbis],
+MPD_AUTO_PKG(vorbis_encoder, VORBISENC, [vorbisenc vorbis ogg],
 	[Ogg Vorbis encoder], [libvorbisenc not found])
 
 if test x$enable_vorbis_encoder = xyes; then

doc/protocol.xml

@@ -576,7 +576,12 @@
               </listitem>
               <listitem>
                 <para>
-                  <varname>songs</varname>: number of albums
+                  <varname>albums</varname>: number of albums
+                </para>
+              </listitem>
+              <listitem>
+                <para>
+                  <varname>songs</varname>: number of songs
                 </para>
               </listitem>
               <listitem>

src/BulkEdit.hxx

@@ -0,0 +1,41 @@
+/*
+ * Copyright (C) 2003-2014 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_BULK_EDIT_HXX
+#define MPD_BULK_EDIT_HXX
+
+#include "Partition.hxx"
+
+/**
+ * Begin a "bulk edit" and commit it automatically.
+ */
+class ScopeBulkEdit {
+	Partition &partition;
+
+public:
+	ScopeBulkEdit(Partition &_partition):partition(_partition) {
+		partition.playlist.BeginBulk();
+	}
+
+	~ScopeBulkEdit() {
+		partition.playlist.CommitBulk(partition.pc);
+	}
+};
+
+#endif

src/DatabaseSelection.cxx

@@ -30,6 +30,18 @@ DatabaseSelection::DatabaseSelection(const char *_uri, bool _recursive,
 		uri = filter->GetBase();
 }
 
+bool
+DatabaseSelection::IsEmpty() const
+{
+	return uri.empty() && (filter == nullptr || filter->IsEmpty());
+}
+
+bool
+DatabaseSelection::HasOtherThanBase() const
+{
+	return filter != nullptr && filter->HasOtherThanBase();
+}
+
 bool
 DatabaseSelection::Match(const Song &song) const
 {

src/DatabaseSelection.hxx

@@ -44,6 +44,15 @@ struct DatabaseSelection {
 	DatabaseSelection(const char *_uri, bool _recursive,
 			  const SongFilter *_filter=nullptr);
 
+	gcc_pure
+	bool IsEmpty() const;
+
+	/**
+	 * Does this selection contain constraints other than "base"?
+	 */
+	gcc_pure
+	bool HasOtherThanBase() const;
+
 	gcc_pure
 	bool Match(const Song &song) const;
 };

src/DecoderAPI.cxx

@@ -292,6 +292,40 @@ decoder_read(Decoder *decoder,
 	return nbytes;
 }
 
+bool
+decoder_read_full(Decoder *decoder, InputStream &is,
+		  void *_buffer, size_t size)
+{
+	uint8_t *buffer = (uint8_t *)_buffer;
+
+	while (size > 0) {
+		size_t nbytes = decoder_read(decoder, is, buffer, size);
+		if (nbytes == 0)
+			return false;
+
+		buffer += nbytes;
+		size -= nbytes;
+	}
+
+	return true;
+}
+
+bool
+decoder_skip(Decoder *decoder, InputStream &is, size_t size)
+{
+	while (size > 0) {
+		char buffer[1024];
+		size_t nbytes = decoder_read(decoder, is, buffer,
+					     std::min(sizeof(buffer), size));
+		if (nbytes == 0)
+			return false;
+
+		size -= nbytes;
+	}
+
+	return true;
+}
+
 void
 decoder_timestamp(Decoder &decoder, double t)
 {

src/DecoderAPI.hxx

@@ -112,6 +112,25 @@ decoder_read(Decoder &decoder, InputStream &is,
 	return decoder_read(&decoder, is, buffer, length);
 }
 
+/**
+ * Blocking read from the input stream.  Attempts to fill the buffer
+ * completely; there is no partial result.
+ *
+ * @return true on success, false on error or command or not enough
+ * data
+ */
+bool
+decoder_read_full(Decoder *decoder, InputStream &is,
+		  void *buffer, size_t size);
+
+/**
+ * Skip data on the #InputStream.
+ *
+ * @return true on success, false on error or command
+ */
+bool
+decoder_skip(Decoder *decoder, InputStream &is, size_t size);
+
 /**
  * Sets the time stamp for the next data chunk [seconds].  The MPD
  * core automatically counts it up, and a decoder plugin only needs to

src/DecoderBuffer.cxx

@@ -70,6 +70,12 @@ decoder_buffer_free(DecoderBuffer *buffer)
 	g_free(buffer);
 }
 
+const InputStream &
+decoder_buffer_get_stream(const DecoderBuffer *buffer)
+{
+	return *buffer->is;
+}
+
 bool
 decoder_buffer_is_empty(const DecoderBuffer *buffer)
 {
@@ -82,6 +88,12 @@ decoder_buffer_is_full(const DecoderBuffer *buffer)
 	return buffer->consumed == 0 && buffer->length == buffer->size;
 }
 
+void
+decoder_buffer_clear(DecoderBuffer *buffer)
+{
+	buffer->length = buffer->consumed = 0;
+}
+
 static void
 decoder_buffer_shift(DecoderBuffer *buffer)
 {
@@ -118,6 +130,12 @@ decoder_buffer_fill(DecoderBuffer *buffer)
 	return true;
 }
 
+size_t
+decoder_buffer_available(const DecoderBuffer *buffer)
+{
+	return buffer->length - buffer->consumed;;
+}
+
 const void *
 decoder_buffer_read(const DecoderBuffer *buffer, size_t *length_r)
 {

src/DecoderBuffer.hxx

@@ -20,6 +20,8 @@
 #ifndef MPD_DECODER_BUFFER_HXX
 #define MPD_DECODER_BUFFER_HXX
 
+#include "Compiler.h"
+
 #include <stddef.h>
 
 /**
@@ -50,12 +52,21 @@ decoder_buffer_new(Decoder *decoder, InputStream &is,
 void
 decoder_buffer_free(DecoderBuffer *buffer);
 
+gcc_pure
+const InputStream &
+decoder_buffer_get_stream(const DecoderBuffer *buffer);
+
+gcc_pure
 bool
 decoder_buffer_is_empty(const DecoderBuffer *buffer);
 
+gcc_pure
 bool
 decoder_buffer_is_full(const DecoderBuffer *buffer);
 
+void
+decoder_buffer_clear(DecoderBuffer *buffer);
+
 /**
  * Read data from the input_stream and append it to the buffer.
  *
@@ -66,6 +77,13 @@ decoder_buffer_is_full(const DecoderBuffer *buffer);
 bool
 decoder_buffer_fill(DecoderBuffer *buffer);
 
+/**
+ * How many bytes are stored in the buffer?
+ */
+gcc_pure
+size_t
+decoder_buffer_available(const DecoderBuffer *buffer);
+
 /**
  * Reads data from the buffer.  This data is not yet consumed, you
  * have to call decoder_buffer_consume() to do that.  The returned

src/Playlist.cxx

@@ -103,6 +103,12 @@ playlist::UpdateQueuedSong(PlayerControl &pc, const Song *prev)
 	if (!playing)
 		return;
 
+	if (prev == nullptr && bulk_edit)
+		/* postponed until CommitBulk() to avoid always
+		   queueing the first song that is being added (in
+		   random mode) */
+		return;
+
 	assert(!queue.IsEmpty());
 	assert((queued < 0) == (prev == nullptr));
 
@@ -294,7 +300,9 @@ playlist::SetRandom(PlayerControl &pc, bool status)
 	if (queue.random) {
 		/* shuffle the queue order, but preserve current */
 
-		const int current_position = GetCurrentPosition();
+		const int current_position = playing
+			? GetCurrentPosition()
+			: -1;
 
 		queue.ShuffleOrder();
 

src/Playlist.hxx

@@ -45,6 +45,18 @@ struct playlist {
 	 */
 	bool stop_on_error;
 
+	/**
+	 * If true, then a bulk edit has been initiated by
+	 * BeginBulk(), and UpdateQueuedSong() and OnModified() will
+	 * be postponed until CommitBulk()
+	 */
+	bool bulk_edit;
+
+	/**
+	 * Has the queue been modified during bulk edit mode?
+	 */
+	bool bulk_modified;
+
 	/**
 	 * Number of errors since playback was started.  If this
 	 * number exceeds the length of the playlist, MPD gives up,
@@ -69,7 +81,9 @@ struct playlist {
 	int queued;
 
 	playlist(unsigned max_length)
-		:queue(max_length), playing(false), current(-1), queued(-1) {
+		:queue(max_length), playing(false),
+		 bulk_edit(false),
+		 current(-1), queued(-1) {
 	}
 
 	~playlist() {
@@ -126,6 +140,9 @@ protected:
 	void UpdateQueuedSong(PlayerControl &pc, const Song *prev);
 
 public:
+	void BeginBulk();
+	void CommitBulk(PlayerControl &pc);
+
 	void Clear(PlayerControl &pc);
 
 	/**
@@ -217,6 +234,10 @@ public:
 
 	void PlayPrevious(PlayerControl &pc);
 
+	PlaylistResult SeekSongOrder(PlayerControl &pc,
+				     unsigned song_order,
+				     float seek_time);
+
 	PlaylistResult SeekSongPosition(PlayerControl &pc,
 					unsigned song_position,
 					float seek_time);

src/PlaylistControl.cxx

@@ -153,7 +153,7 @@ playlist::PlayNext(PlayerControl &pc)
 			queue.ShuffleOrder();
 
 			/* note that current and queued are
-			   now invalid, but playlist_play_order() will
+			   now invalid, but PlayOrder() will
 			   discard them anyway */
 		}
 
@@ -190,17 +190,12 @@ playlist::PlayPrevious(PlayerControl &pc)
 }
 
 PlaylistResult
-playlist::SeekSongPosition(PlayerControl &pc, unsigned song, float seek_time)
+playlist::SeekSongOrder(PlayerControl &pc, unsigned i, float seek_time)
 {
-	if (!queue.IsValidPosition(song))
+	assert(queue.IsValidOrder(i));
-		return PlaylistResult::BAD_RANGE;
 
 	const Song *queued_song = GetQueuedSong();
 
-	unsigned i = queue.random
-		? queue.PositionToOrder(song)
-		: song;
-
 	pc.ClearError();
 	stop_on_error = true;
 	error_count = 0;
@@ -228,6 +223,19 @@ playlist::SeekSongPosition(PlayerControl &pc, unsigned song, float seek_time)
 	return PlaylistResult::SUCCESS;
 }
 
+PlaylistResult
+playlist::SeekSongPosition(PlayerControl &pc, unsigned song, float seek_time)
+{
+	if (!queue.IsValidPosition(song))
+		return PlaylistResult::BAD_RANGE;
+
+	unsigned i = queue.random
+		? queue.PositionToOrder(song)
+		: song;
+
+	return SeekSongOrder(pc, i, seek_time);
+}
+
 PlaylistResult
 playlist::SeekSongId(PlayerControl &pc, unsigned id, float seek_time)
 {
@@ -257,5 +265,5 @@ playlist::SeekCurrent(PlayerControl &pc, float seek_time, bool relative)
 	if (seek_time < 0)
 		seek_time = 0;
 
-	return SeekSongPosition(pc, current, seek_time);
+	return SeekSongOrder(pc, current, seek_time);
 }

src/PlaylistEdit.cxx

@@ -40,6 +40,12 @@
 void
 playlist::OnModified()
 {
+	if (bulk_edit) {
+		/* postponed to CommitBulk() */
+		bulk_modified = true;
+		return;
+	}
+
 	queue.IncrementVersion();
 
 	idle_add(IDLE_PLAYLIST);
@@ -56,6 +62,35 @@ playlist::Clear(PlayerControl &pc)
 	OnModified();
 }
 
+void
+playlist::BeginBulk()
+{
+	assert(!bulk_edit);
+
+	bulk_edit = true;
+	bulk_modified = false;
+}
+
+void
+playlist::CommitBulk(PlayerControl &pc)
+{
+	assert(bulk_edit);
+
+	bulk_edit = false;
+	if (!bulk_modified)
+		return;
+
+	if (queued < 0)
+		/* if no song was queued, UpdateQueuedSong() is being
+		   ignored in "bulk" edit mode; now that we have
+		   shuffled all new songs, we can pick a random one
+		   (instead of always picking the first one that was
+		   added) */
+		UpdateQueuedSong(pc, nullptr);
+
+	OnModified();
+}
+
 PlaylistResult
 playlist::AppendFile(PlayerControl &pc,
 		     const char *path_utf8, unsigned *added_id)
@@ -234,12 +269,8 @@ playlist::DeleteInternal(PlayerControl &pc,
 	if (playing && current == (int)songOrder) {
 		const bool paused = pc.GetState() == PlayerState::PAUSE;
 
-		/* the current song is going to be deleted: stop the player */
+		/* the current song is going to be deleted: see which
-
+		   song is going to be played instead */
-		pc.Stop();
-		playing = false;
-
-		/* see which song is going to be played instead */
 
 		current = queue.GetNextOrder(current);
 		if (current == (int)songOrder)
@@ -248,10 +279,12 @@ playlist::DeleteInternal(PlayerControl &pc,
 		if (current >= 0 && !paused)
 			/* play the song after the deleted one */
 			PlayOrder(pc, current);
-		else
+		else {
-			/* no songs left to play, stop playback
+			/* stop the player */
-			   completely */
+
-			Stop(pc);
+			pc.Stop();
+			playing = false;
+		}
 
 		*queued_p = nullptr;
 	} else if (current == (int)songOrder)

src/SongFilter.cxx

@@ -203,6 +203,16 @@ SongFilter::Match(const Song &song) const
 	return true;
 }
 
+bool
+SongFilter::HasOtherThanBase() const
+{
+	for (const auto &i : items)
+		if (i.GetTag() != LOCATE_TAG_BASE_TYPE)
+			return true;
+
+	return false;
+}
+
 std::string
 SongFilter::GetBase() const
 {

src/SongFilter.hxx

@@ -109,6 +109,11 @@ public:
 		return items;
 	}
 
+	gcc_pure
+	bool IsEmpty() const {
+		return items.empty();
+	}
+
 	/**
 	 * Is there at least one item with "fold case" enabled?
 	 */
@@ -121,6 +126,12 @@ public:
 		return false;
 	}
 
+	/**
+	 * Does this filter contain constraints other than "base"?
+	 */
+	gcc_pure
+	bool HasOtherThanBase() const;
+
 	/**
 	 * Returns the "base" specification (if there is one) or an
 	 * empty string.

src/command/AllCommands.cxx

@@ -90,7 +90,7 @@ static const struct command commands[] = {
 	{ "disableoutput", PERMISSION_ADMIN, 1, 1, handle_disableoutput },
 	{ "enableoutput", PERMISSION_ADMIN, 1, 1, handle_enableoutput },
 	{ "find", PERMISSION_READ, 2, -1, handle_find },
-	{ "findadd", PERMISSION_READ, 2, -1, handle_findadd},
+	{ "findadd", PERMISSION_ADD, 2, -1, handle_findadd},
 	{ "idle", PERMISSION_READ, 0, -1, handle_idle },
 	{ "kill", PERMISSION_ADMIN, -1, -1, handle_kill },
 	{ "list", PERMISSION_READ, 1, -1, handle_list },

src/command/DatabaseCommands.cxx

@@ -30,6 +30,7 @@
 #include "util/Error.hxx"
 #include "SongFilter.hxx"
 #include "protocol/Result.hxx"
+#include "BulkEdit.hxx"
 
 #include <assert.h>
 #include <string.h>
@@ -92,6 +93,8 @@ handle_match_add(Client &client, int argc, char *argv[], bool fold_case)
 		return CommandResult::ERROR;
 	}
 
+	const ScopeBulkEdit bulk_edit(client.partition);
+
 	const DatabaseSelection selection("", true, &filter);
 	Error error;
 	return AddFromDatabase(client.partition, selection, error)

src/command/PlaylistCommands.cxx

@@ -26,6 +26,7 @@
 #include "PlaylistFile.hxx"
 #include "PlaylistVector.hxx"
 #include "PlaylistQueue.hxx"
+#include "BulkEdit.hxx"
 #include "TimePrint.hxx"
 #include "Client.hxx"
 #include "protocol/ArgParser.hxx"
@@ -67,6 +68,8 @@ handle_load(Client &client, int argc, char *argv[])
 	} else if (!check_range(client, &start_index, &end_index, argv[2]))
 		return CommandResult::ERROR;
 
+	const ScopeBulkEdit bulk_edit(client.partition);
+
 	const PlaylistResult result =
 		playlist_open_into_queue(argv[1],
 					 start_index, end_index,

src/command/QueueCommands.cxx

@@ -28,6 +28,7 @@
 #include "ClientFile.hxx"
 #include "Client.hxx"
 #include "Partition.hxx"
+#include "BulkEdit.hxx"
 #include "protocol/ArgParser.hxx"
 #include "protocol/Result.hxx"
 #include "ls.hxx"
@@ -43,7 +44,6 @@ CommandResult
 handle_add(Client &client, gcc_unused int argc, char *argv[])
 {
 	char *uri = argv[1];
-	PlaylistResult result;
 
 	if (memcmp(uri, "file:///", 8) == 0) {
 		const char *path_utf8 = uri + 7;
@@ -59,7 +59,7 @@ handle_add(Client &client, gcc_unused int argc, char *argv[])
 		if (!client_allow_file(client, path_fs, error))
 			return print_error(client, error);
 
-		result = client.partition.AppendFile(path_utf8);
+		auto result = client.partition.AppendFile(path_utf8);
 		return print_playlist_result(client, result);
 	}
 
@@ -70,10 +70,12 @@ handle_add(Client &client, gcc_unused int argc, char *argv[])
 			return CommandResult::ERROR;
 		}
 
-		result = client.partition.AppendURI(uri);
+		auto result = client.partition.AppendURI(uri);
 		return print_playlist_result(client, result);
 	}
 
+	const ScopeBulkEdit bulk_edit(client.partition);
+
 	const DatabaseSelection selection(uri, true);
 	Error error;
 	return AddFromDatabase(client.partition, selection, error)

src/db/ProxyDatabasePlugin.cxx

@@ -398,8 +398,13 @@ Convert(const struct mpd_song *song)
 	Song *s = Song::NewDetached(mpd_song_get_uri(song));
 
 	s->mtime = mpd_song_get_last_modified(song);
+
+#if LIBMPDCLIENT_CHECK_VERSION(2,3,0)
 	s->start_ms = mpd_song_get_start(song) * 1000;
 	s->end_ms = mpd_song_get_end(song) * 1000;
+#else
+	s->start_ms = s->end_ms = 0;
+#endif
 
 	TagBuilder tag;
 	tag.SetTime(mpd_song_get_duration(song));
@@ -561,6 +566,23 @@ SearchSongs(struct mpd_connection *connection,
 	return result && CheckError(connection, error);
 }
 
+/**
+ * Check whether we can use the "base" constraint.  Requires
+ * libmpdclient 2.9 and MPD 0.18.
+ */
+gcc_pure
+static bool
+ServerSupportsSearchBase(const struct mpd_connection *connection)
+{
+#if LIBMPDCLIENT_CHECK_VERSION(2,9,0)
+	return mpd_connection_cmp_server_version(connection, 0, 18, 0) >= 0;
+#else
+	(void)connection;
+
+	return false;
+#endif
+}
+
 bool
 ProxyDatabase::Visit(const DatabaseSelection &selection,
 		     VisitDirectory visit_directory,
@@ -572,7 +594,10 @@ ProxyDatabase::Visit(const DatabaseSelection &selection,
 	if (!const_cast<ProxyDatabase *>(this)->EnsureConnected(error))
 		return nullptr;
 
-	if (!visit_directory && !visit_playlist && selection.recursive)
+	if (!visit_directory && !visit_playlist && selection.recursive &&
+	    (ServerSupportsSearchBase(connection)
+	     ? !selection.IsEmpty()
+	     : selection.HasOtherThanBase()))
 		/* this optimized code path can only be used under
 		   certain conditions */
 		return ::SearchSongs(connection, selection, visit_song, error);

src/decoder/AudiofileDecoderPlugin.cxx

@@ -31,12 +31,27 @@
 #include <af_vfs.h>
 
 #include <assert.h>
+#include <stdio.h>
 
 /* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
 #define CHUNK_SIZE		1020
 
 static constexpr Domain audiofile_domain("audiofile");
 
+struct AudioFileInputStream {
+	Decoder *const decoder;
+	InputStream &is;
+
+	size_t Read(void *buffer, size_t size) {
+		/* libaudiofile does not like partial reads at all,
+		   and will abort playback; therefore always force full
+		   reads */
+		return decoder_read_full(decoder, is, buffer, size)
+			? size
+			: 0;
+	}
+};
+
 static int audiofile_get_duration(const char *file)
 {
 	int total_time;
@@ -54,29 +69,26 @@ static int audiofile_get_duration(const char *file)
 static ssize_t
 audiofile_file_read(AFvirtualfile *vfile, void *data, size_t length)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
 
-	Error error;
+	return afis.Read(data, length);
-	size_t nbytes = is.LockRead(data, length, error);
-	if (nbytes == 0 && error.IsDefined()) {
-		LogError(error);
-		return -1;
-	}
-
-	return nbytes;
 }
 
 static AFfileoffset
 audiofile_file_length(AFvirtualfile *vfile)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
+	InputStream &is = afis.is;
+
 	return is.GetSize();
 }
 
 static AFfileoffset
 audiofile_file_tell(AFvirtualfile *vfile)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
+	InputStream &is = afis.is;
+
 	return is.GetOffset();
 }
 
@@ -91,11 +103,14 @@ audiofile_file_destroy(AFvirtualfile *vfile)
 static AFfileoffset
 audiofile_file_seek(AFvirtualfile *vfile, AFfileoffset offset, int is_relative)
 {
-	InputStream &is = *(InputStream *)vfile->closure;
+	AudioFileInputStream &afis = *(AudioFileInputStream *)vfile->closure;
+	InputStream &is = afis.is;
+
 	int whence = (is_relative ? SEEK_CUR : SEEK_SET);
 
 	Error error;
 	if (is.LockSeek(offset, whence, error)) {
+		LogError(error, "Seek failed");
 		return is.GetOffset();
 	} else {
 		return -1;
@@ -103,10 +118,10 @@ audiofile_file_seek(AFvirtualfile *vfile, AFfileoffset offset, int is_relative)
 }
 
 static AFvirtualfile *
-setup_virtual_fops(InputStream &stream)
+setup_virtual_fops(AudioFileInputStream &afis)
 {
 	AFvirtualfile *vf = new AFvirtualfile();
-	vf->closure = &stream;
+	vf->closure = &afis;
 	vf->write = nullptr;
 	vf->read    = audiofile_file_read;
 	vf->length  = audiofile_file_length;
@@ -173,7 +188,8 @@ audiofile_stream_decode(Decoder &decoder, InputStream &is)
 		return;
 	}
 
-	vf = setup_virtual_fops(is);
+	AudioFileInputStream afis{&decoder, is};
+	vf = setup_virtual_fops(afis);
 
 	af_fp = afOpenVirtualFile(vf, "r", nullptr);
 	if (af_fp == AF_NULL_FILEHANDLE) {

src/decoder/DsdLib.cxx

@@ -49,14 +49,6 @@ DsdId::Equals(const char *s) const
 	return memcmp(value, s, sizeof(value)) == 0;
 }
 
-bool
-dsdlib_read(Decoder *decoder, InputStream &is,
-	    void *data, size_t length)
-{
-	size_t nbytes = decoder_read(decoder, is, data, length);
-	return nbytes == length;
-}
-
 /**
  * Skip the #input_stream to the specified offset.
  */
@@ -149,7 +141,7 @@ dsdlib_tag_id3(InputStream &is,
 	id3_byte_t *dsdid3data;
 	dsdid3data = dsdid3;
 
-	if (!dsdlib_read(nullptr, is, dsdid3data, count))
+	if (!decoder_read_full(nullptr, is, dsdid3data, count))
 		return;
 
 	id3_tag = id3_tag_parse(dsdid3data, count);

src/decoder/DsdLib.hxx

@@ -58,10 +58,6 @@ public:
 	}
 };
 
-bool
-dsdlib_read(Decoder *decoder, InputStream &is,
-	    void *data, size_t length);
-
 bool
 dsdlib_skip_to(Decoder *decoder, InputStream &is,
 	       int64_t offset);

src/decoder/DsdiffDecoderPlugin.cxx

@@ -93,14 +93,14 @@ static bool
 dsdiff_read_id(Decoder *decoder, InputStream &is,
 	       DsdId *id)
 {
-	return dsdlib_read(decoder, is, id, sizeof(*id));
+	return decoder_read_full(decoder, is, id, sizeof(*id));
 }
 
 static bool
 dsdiff_read_chunk_header(Decoder *decoder, InputStream &is,
 			 DsdiffChunkHeader *header)
 {
-	return dsdlib_read(decoder, is, header, sizeof(*header));
+	return decoder_read_full(decoder, is, header, sizeof(*header));
 }
 
 static bool
@@ -112,8 +112,7 @@ dsdiff_read_payload(Decoder *decoder, InputStream &is,
 	if (size != (uint64_t)length)
 		return false;
 
-	size_t nbytes = decoder_read(decoder, is, data, length);
+	return decoder_read_full(decoder, is, data, length);
-	return nbytes == length;
 }
 
 /**
@@ -145,8 +144,8 @@ dsdiff_read_prop_snd(Decoder *decoder, InputStream &is,
 		} else if (header.id.Equals("CHNL")) {
 			uint16_t channels;
 			if (header.GetSize() < sizeof(channels) ||
-			    !dsdlib_read(decoder, is,
+			    !decoder_read_full(decoder, is,
-					 &channels, sizeof(channels)) ||
+					       &channels, sizeof(channels)) ||
 			    !dsdlib_skip_to(decoder, is, chunk_end_offset))
 				return false;
 
@@ -154,8 +153,8 @@ dsdiff_read_prop_snd(Decoder *decoder, InputStream &is,
 		} else if (header.id.Equals("CMPR")) {
 			DsdId type;
 			if (header.GetSize() < sizeof(type) ||
-			    !dsdlib_read(decoder, is,
+			    !decoder_read_full(decoder, is,
-					 &type, sizeof(type)) ||
+					       &type, sizeof(type)) ||
 			    !dsdlib_skip_to(decoder, is, chunk_end_offset))
 				return false;
 
@@ -208,7 +207,7 @@ dsdiff_handle_native_tag(InputStream &is,
 
 	struct dsdiff_native_tag metatag;
 
-	if (!dsdlib_read(nullptr, is, &metatag, sizeof(metatag)))
+	if (!decoder_read_full(nullptr, is, &metatag, sizeof(metatag)))
 		return;
 
 	uint32_t length = FromBE32(metatag.size);
@@ -221,7 +220,7 @@ dsdiff_handle_native_tag(InputStream &is,
 	char *label;
 	label = string;
 
-	if (!dsdlib_read(nullptr, is, label, (size_t)length))
+	if (!decoder_read_full(nullptr, is, label, (size_t)length))
 		return;
 
 	string[length] = '\0';
@@ -251,15 +250,17 @@ dsdiff_read_metadata_extra(Decoder *decoder, InputStream &is,
 	if (!dsdiff_read_chunk_header(decoder, is, chunk_header))
 		return false;
 
+	metadata->diar_offset = 0;
+	metadata->diti_offset = 0;
+
 #ifdef HAVE_ID3TAG
-	metadata->id3_size = 0;
+	metadata->id3_offset = 0;
 #endif
 
 	/* Now process all the remaining chunk headers in the stream
 	   and record their position and size */
 
-	const auto size = is.GetSize();
+	do {
-	while (is.GetOffset() < size) {
 		uint64_t chunk_size = chunk_header->GetSize();
 
 		/* DIIN chunk, is directly followed by other chunks  */
@@ -285,16 +286,11 @@ dsdiff_read_metadata_extra(Decoder *decoder, InputStream &is,
 			metadata->id3_size = chunk_size;
 		}
 #endif
-		if (chunk_size != 0) {
-			if (!dsdlib_skip(decoder, is, chunk_size))
-				break;
-		}
 
-		if (is.GetOffset() < size) {
+		if (!dsdlib_skip(decoder, is, chunk_size))
-			if (!dsdiff_read_chunk_header(decoder, is, chunk_header))
+			break;
-				return false;
+	} while (dsdiff_read_chunk_header(decoder, is, chunk_header));
-		}
+
-	}
 	/* done processing chunk headers, process tags if any */
 
 #ifdef HAVE_ID3TAG
@@ -328,7 +324,7 @@ dsdiff_read_metadata(Decoder *decoder, InputStream &is,
 		     DsdiffChunkHeader *chunk_header)
 {
 	DsdiffHeader header;
-	if (!dsdlib_read(decoder, is, &header, sizeof(header)) ||
+	if (!decoder_read_full(decoder, is, &header, sizeof(header)) ||
 	    !header.id.Equals("FRM8") ||
 	    !header.format.Equals("DSD "))
 		return false;
@@ -381,7 +377,7 @@ dsdiff_decode_chunk(Decoder &decoder, InputStream &is,
 	const unsigned buffer_samples = buffer_frames * frame_size;
 	const size_t buffer_size = buffer_samples * sample_size;
 
-	while (chunk_size > 0) {
+	while (chunk_size >= frame_size) {
 		/* see how much aligned data from the remaining chunk
 		   fits into the local buffer */
 		size_t now_size = buffer_size;
@@ -391,10 +387,10 @@ dsdiff_decode_chunk(Decoder &decoder, InputStream &is,
 			now_size = now_frames * frame_size;
 		}
 
-		size_t nbytes = decoder_read(decoder, is, buffer, now_size);
+		if (!decoder_read_full(&decoder, is, buffer, now_size))
-		if (nbytes != now_size)
 			return false;
 
+		const size_t nbytes = now_size;
 		chunk_size -= nbytes;
 
 		if (lsbitfirst)

src/decoder/DsfDecoderPlugin.cxx

@@ -103,7 +103,7 @@ dsf_read_metadata(Decoder *decoder, InputStream &is,
 		  DsfMetaData *metadata)
 {
 	DsfHeader dsf_header;
-	if (!dsdlib_read(decoder, is, &dsf_header, sizeof(dsf_header)) ||
+	if (!decoder_read_full(decoder, is, &dsf_header, sizeof(dsf_header)) ||
 	    !dsf_header.id.Equals("DSD "))
 		return false;
 
@@ -117,7 +117,8 @@ dsf_read_metadata(Decoder *decoder, InputStream &is,
 
 	/* read the 'fmt ' chunk of the DSF file */
 	DsfFmtChunk dsf_fmt_chunk;
-	if (!dsdlib_read(decoder, is, &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) ||
+	if (!decoder_read_full(decoder, is,
+			       &dsf_fmt_chunk, sizeof(dsf_fmt_chunk)) ||
 	    !dsf_fmt_chunk.id.Equals("fmt "))
 		return false;
 
@@ -143,7 +144,7 @@ dsf_read_metadata(Decoder *decoder, InputStream &is,
 
 	/* read the 'data' chunk of the DSF file */
 	DsfDataChunk data_chunk;
-	if (!dsdlib_read(decoder, is, &data_chunk, sizeof(data_chunk)) ||
+	if (!decoder_read_full(decoder, is, &data_chunk, sizeof(data_chunk)) ||
 	    !data_chunk.id.Equals("data"))
 		return false;
 
@@ -237,7 +238,7 @@ dsf_decode_chunk(Decoder &decoder, InputStream &is,
 	const unsigned buffer_samples = buffer_frames * frame_size;
 	const size_t buffer_size = buffer_samples * sample_size;
 
-	while (chunk_size > 0) {
+	while (chunk_size >= frame_size) {
 		/* see how much aligned data from the remaining chunk
 		   fits into the local buffer */
 		size_t now_size = buffer_size;
@@ -247,10 +248,10 @@ dsf_decode_chunk(Decoder &decoder, InputStream &is,
 			now_size = now_frames * frame_size;
 		}
 
-		size_t nbytes = decoder_read(&decoder, is, buffer, now_size);
+		if (!decoder_read_full(&decoder, is, buffer, now_size))
-		if (nbytes != now_size)
 			return false;
 
+		const size_t nbytes = now_size;
 		chunk_size -= nbytes;
 
 		if (bitreverse)

src/decoder/FaadDecoderPlugin.cxx

@@ -34,8 +34,6 @@
 #include <string.h>
 #include <unistd.h>
 
-#define AAC_MAX_CHANNELS	6
-
 static const unsigned adts_sample_rates[] =
     { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
 	16000, 12000, 11025, 8000, 7350, 0, 0, 0
@@ -66,16 +64,13 @@ adts_check_frame(const unsigned char *data)
 static size_t
 adts_find_frame(DecoderBuffer *buffer)
 {
-	size_t length, frame_length;
-	bool ret;
-
 	while (true) {
+		size_t length;
 		const uint8_t *data = (const uint8_t *)
 			decoder_buffer_read(buffer, &length);
 		if (data == nullptr || length < 8) {
 			/* not enough data yet */
-			ret = decoder_buffer_fill(buffer);
+			if (!decoder_buffer_fill(buffer))
-			if (!ret)
 				/* failed */
 				return 0;
 
@@ -86,7 +81,7 @@ adts_find_frame(DecoderBuffer *buffer)
 		const uint8_t *p = (const uint8_t *)memchr(data, 0xff, length);
 		if (p == nullptr) {
 			/* no marker - discard the buffer */
-			decoder_buffer_consume(buffer, length);
+			decoder_buffer_clear(buffer);
 			continue;
 		}
 
@@ -97,7 +92,7 @@ adts_find_frame(DecoderBuffer *buffer)
 		}
 
 		/* is it a frame? */
-		frame_length = adts_check_frame(data);
+		const size_t frame_length = adts_check_frame(data);
 		if (frame_length == 0) {
 			/* it's just some random 0xff byte; discard it
 			   and continue searching */
@@ -109,15 +104,11 @@ adts_find_frame(DecoderBuffer *buffer)
 			/* available buffer size is smaller than the
 			   frame will be - attempt to read more
 			   data */
-			ret = decoder_buffer_fill(buffer);
+			if (!decoder_buffer_fill(buffer)) {
-			if (!ret) {
 				/* not enough data; discard this frame
 				   to prevent a possible buffer
 				   overflow */
-				data = (const uint8_t *)
+				decoder_buffer_clear(buffer);
-					decoder_buffer_read(buffer, &length);
-				if (data != nullptr)
-					decoder_buffer_consume(buffer, length);
 			}
 
 			continue;
@@ -131,13 +122,18 @@ adts_find_frame(DecoderBuffer *buffer)
 static float
 adts_song_duration(DecoderBuffer *buffer)
 {
-	unsigned int frames, frame_length;
+	const InputStream &is = decoder_buffer_get_stream(buffer);
+	const bool estimate = !is.CheapSeeking();
+	const auto file_size = is.GetSize();
+	if (estimate && file_size <= 0)
+		return -1;
+
 	unsigned sample_rate = 0;
-	float frames_per_second;
 
 	/* Read all frames to ensure correct time and bitrate */
+	unsigned frames;
 	for (frames = 0;; frames++) {
-		frame_length = adts_find_frame(buffer);
+		const unsigned frame_length = adts_find_frame(buffer);
 		if (frame_length == 0)
 			break;
 
@@ -150,36 +146,52 @@ adts_song_duration(DecoderBuffer *buffer)
 			assert(frame_length <= buffer_length);
 
 			sample_rate = adts_sample_rates[(data[2] & 0x3c) >> 2];
+			if (sample_rate == 0)
+				break;
 		}
 
 		decoder_buffer_consume(buffer, frame_length);
+
+		if (estimate && frames == 128) {
+			/* if this is a remote file, don't slurp the
+			   whole file just for checking the song
+			   duration; instead, stop after some time and
+			   extrapolate the song duration from what we
+			   have until now */
+
+			const auto offset = is.GetOffset()
+				- decoder_buffer_available(buffer);
+			if (offset <= 0)
+				return -1;
+
+			frames = (frames * file_size) / offset;
+			break;
+		}
 	}
 
-	frames_per_second = (float)sample_rate / 1024.0;
+	if (sample_rate == 0)
-	if (frames_per_second <= 0)
 		return -1;
 
+	float frames_per_second = (float)sample_rate / 1024.0;
+	assert(frames_per_second > 0);
+
 	return (float)frames / frames_per_second;
 }
 
 static float
 faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 {
-	size_t fileread;
-	size_t tagsize;
-	size_t length;
-	bool success;
-
 	const auto size = is.GetSize();
-	fileread = size >= 0 ? size : 0;
+	const size_t fileread = size >= 0 ? size : 0;
 
 	decoder_buffer_fill(buffer);
+	size_t length;
 	const uint8_t *data = (const uint8_t *)
 		decoder_buffer_read(buffer, &length);
 	if (data == nullptr)
 		return -1;
 
-	tagsize = 0;
+	size_t tagsize = 0;
 	if (length >= 10 && !memcmp(data, "ID3", 3)) {
 		/* skip the ID3 tag */
 
@@ -188,7 +200,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 
 		tagsize += 10;
 
-		success = decoder_buffer_skip(buffer, tagsize) &&
+		const bool success = decoder_buffer_skip(buffer, tagsize) &&
 			decoder_buffer_fill(buffer);
 		if (!success)
 			return -1;
@@ -198,22 +210,20 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 			return -1;
 	}
 
-	if (is.IsSeekable() && length >= 2 &&
+	if (length >= 8 && adts_check_frame(data) > 0) {
-	    data[0] == 0xFF && ((data[1] & 0xF6) == 0xF0)) {
 		/* obtain the duration from the ADTS header */
+
+		if (!is.IsSeekable())
+			return -1;
+
 		float song_length = adts_song_duration(buffer);
 
 		is.LockSeek(tagsize, SEEK_SET, IgnoreError());
-
+		decoder_buffer_clear(buffer);
-		data = (const uint8_t *)decoder_buffer_read(buffer, &length);
-		if (data != nullptr)
-			decoder_buffer_consume(buffer, length);
-		decoder_buffer_fill(buffer);
 
 		return song_length;
 	} else if (length >= 5 && memcmp(data, "ADIF", 4) == 0) {
 		/* obtain the duration from the ADIF header */
-		unsigned bit_rate;
 		size_t skip_size = (data[4] & 0x80) ? 9 : 0;
 
 		if (8 + skip_size > length)
@@ -221,7 +231,7 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 			   header */
 			return -1;
 
-		bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
+		unsigned bit_rate = ((data[4 + skip_size] & 0x0F) << 19) |
 			(data[5 + skip_size] << 11) |
 			(data[6 + skip_size] << 3) |
 			(data[7 + skip_size] & 0xE0);
@@ -234,6 +244,21 @@ faad_song_duration(DecoderBuffer *buffer, InputStream &is)
 		return -1;
 }
 
+static NeAACDecHandle
+faad_decoder_new()
+{
+	const NeAACDecHandle decoder = NeAACDecOpen();
+
+	NeAACDecConfigurationPtr config =
+		NeAACDecGetCurrentConfiguration(decoder);
+	config->outputFormat = FAAD_FMT_16BIT;
+	config->downMatrix = 1;
+	config->dontUpSampleImplicitSBR = 0;
+	NeAACDecSetConfiguration(decoder, config);
+
+	return decoder;
+}
+
 /**
  * Wrapper for NeAACDecInit() which works around some API
  * inconsistencies in libfaad.
@@ -242,17 +267,6 @@ static bool
 faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
 		  AudioFormat &audio_format, Error &error)
 {
-	int32_t nbytes;
-	uint32_t sample_rate;
-	uint8_t channels;
-#ifdef HAVE_FAAD_LONG
-	/* neaacdec.h declares all arguments as "unsigned long", but
-	   internally expects uint32_t pointers.  To avoid gcc
-	   warnings, use this workaround. */
-	unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
-#else
-	uint32_t *sample_rate_p = &sample_rate;
-#endif
 
 	size_t length;
 	const unsigned char *data = (const unsigned char *)
@@ -262,11 +276,21 @@ faad_decoder_init(NeAACDecHandle decoder, DecoderBuffer *buffer,
 		return false;
 	}
 
-	nbytes = NeAACDecInit(decoder,
+	uint8_t channels;
-			      /* deconst hack, libfaad requires this */
+	uint32_t sample_rate;
-			      const_cast<unsigned char *>(data),
+#ifdef HAVE_FAAD_LONG
-			     length,
+	/* neaacdec.h declares all arguments as "unsigned long", but
-			     sample_rate_p, &channels);
+	   internally expects uint32_t pointers.  To avoid gcc
+	   warnings, use this workaround. */
+	unsigned long *sample_rate_p = (unsigned long *)(void *)&sample_rate;
+#else
+	uint32_t *sample_rate_p = &sample_rate;
+#endif
+	long nbytes = NeAACDecInit(decoder,
+				   /* deconst hack, libfaad requires this */
+				   const_cast<unsigned char *>(data),
+				   length,
+				   sample_rate_p, &channels);
 	if (nbytes < 0) {
 		error.Set(faad_decoder_domain, "Not an AAC stream");
 		return false;
@@ -306,29 +330,21 @@ faad_decoder_decode(NeAACDecHandle decoder, DecoderBuffer *buffer,
 static float
 faad_get_file_time_float(InputStream &is)
 {
-	DecoderBuffer *buffer;
+	DecoderBuffer *const buffer =
-	float length;
+		decoder_buffer_new(nullptr, is,
+				   FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
 
-	buffer = decoder_buffer_new(nullptr, is,
+	float length = faad_song_duration(buffer, is);
-				    FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
-	length = faad_song_duration(buffer, is);
 
 	if (length < 0) {
-		bool ret;
 		AudioFormat audio_format;
 
-		NeAACDecHandle decoder = NeAACDecOpen();
+		NeAACDecHandle decoder = faad_decoder_new();
-
-		NeAACDecConfigurationPtr config =
-			NeAACDecGetCurrentConfiguration(decoder);
-		config->outputFormat = FAAD_FMT_16BIT;
-		NeAACDecSetConfiguration(decoder, config);
 
 		decoder_buffer_fill(buffer);
 
-		ret = faad_decoder_init(decoder, buffer, audio_format,
+		if (faad_decoder_init(decoder, buffer, audio_format,
-					IgnoreError());
+				      IgnoreError()))
-		if (ret)
 			length = 0;
 
 		NeAACDecClose(decoder);
@@ -347,38 +363,25 @@ faad_get_file_time_float(InputStream &is)
 static int
 faad_get_file_time(InputStream &is)
 {
-	int file_time = -1;
+	float length = faad_get_file_time_float(is);
-	float length;
+	if (length < 0)
+		return -1;
 
-	if ((length = faad_get_file_time_float(is)) >= 0)
+	return int(length + 0.5);
-		file_time = length + 0.5;
-
-	return file_time;
 }
 
 static void
 faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
 {
-	float total_time = 0;
+	DecoderBuffer *const buffer =
-	AudioFormat audio_format;
+		decoder_buffer_new(&mpd_decoder, is,
-	bool ret;
+				   FAAD_MIN_STREAMSIZE * MAX_CHANNELS);
-	uint16_t bit_rate = 0;
-	DecoderBuffer *buffer;
 
-	buffer = decoder_buffer_new(&mpd_decoder, is,
+	const float total_time = faad_song_duration(buffer, is);
-				    FAAD_MIN_STREAMSIZE * AAC_MAX_CHANNELS);
-	total_time = faad_song_duration(buffer, is);
 
 	/* create the libfaad decoder */
 
-	NeAACDecHandle decoder = NeAACDecOpen();
+	const NeAACDecHandle decoder = faad_decoder_new();
-
-	NeAACDecConfigurationPtr config =
-		NeAACDecGetCurrentConfiguration(decoder);
-	config->outputFormat = FAAD_FMT_16BIT;
-	config->downMatrix = 1;
-	config->dontUpSampleImplicitSBR = 0;
-	NeAACDecSetConfiguration(decoder, config);
 
 	while (!decoder_buffer_is_full(buffer) && !is.LockIsEOF() &&
 	       decoder_get_command(mpd_decoder) == DecoderCommand::NONE) {
@@ -389,8 +392,8 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
 	/* initialize it */
 
 	Error error;
-	ret = faad_decoder_init(decoder, buffer, audio_format, error);
+	AudioFormat audio_format;
-	if (!ret) {
+	if (!faad_decoder_init(decoder, buffer, audio_format, error)) {
 		LogError(error);
 		NeAACDecClose(decoder);
 		decoder_buffer_free(buffer);
@@ -404,21 +407,20 @@ faad_stream_decode(Decoder &mpd_decoder, InputStream &is)
 	/* the decoder loop */
 
 	DecoderCommand cmd;
+	uint16_t bit_rate = 0;
 	do {
-		size_t frame_size;
-		const void *decoded;
-		NeAACDecFrameInfo frame_info;
-
 		/* find the next frame */
 
-		frame_size = adts_find_frame(buffer);
+		const size_t frame_size = adts_find_frame(buffer);
 		if (frame_size == 0)
 			/* end of file */
 			break;
 
 		/* decode it */
 
-		decoded = faad_decoder_decode(decoder, buffer, &frame_info);
+		NeAACDecFrameInfo frame_info;
+		const void *const decoded =
+			faad_decoder_decode(decoder, buffer, &frame_info);
 
 		if (frame_info.error > 0) {
 			FormatWarning(faad_decoder_domain,
@@ -470,7 +472,6 @@ faad_scan_stream(InputStream &is,
 		 const struct tag_handler *handler, void *handler_ctx)
 {
 	int file_time = faad_get_file_time(is);
-
 	if (file_time < 0)
 		return false;
 

src/decoder/FfmpegDecoderPlugin.cxx

@@ -197,6 +197,29 @@ time_to_ffmpeg(double t, const AVRational time_base)
 			    time_base);
 }
 
+/**
+ * Replace #AV_NOPTS_VALUE with the given fallback.
+ */
+static constexpr int64_t
+timestamp_fallback(int64_t t, int64_t fallback)
+{
+	return gcc_likely(t != int64_t(AV_NOPTS_VALUE))
+		? t
+		: fallback;
+}
+
+/**
+ * Accessor for AVStream::start_time that replaces AV_NOPTS_VALUE with
+ * zero.  We can't use AV_NOPTS_VALUE in calculations, and we simply
+ * assume that the stream's start time is zero, which appears to be
+ * the best way out of that situation.
+ */
+static int64_t
+start_time_fallback(const AVStream &stream)
+{
+	return timestamp_fallback(stream.start_time, 0);
+}
+
 static void
 copy_interleave_frame2(uint8_t *dest, uint8_t **src,
 		       unsigned nframes, unsigned nchannels,
@@ -263,7 +286,7 @@ ffmpeg_send_packet(Decoder &decoder, InputStream &is,
 {
 	if (packet->pts >= 0 && packet->pts != (int64_t)AV_NOPTS_VALUE)
 		decoder_timestamp(decoder,
-				  time_from_ffmpeg(packet->pts - stream->start_time,
+				  time_from_ffmpeg(packet->pts - start_time_fallback(*stream),
 				  stream->time_base));
 
 	AVPacket packet2 = *packet;
@@ -410,9 +433,18 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 	AVStream *av_stream = format_context->streams[audio_stream];
 
 	AVCodecContext *codec_context = av_stream->codec;
+
+#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(54, 25, 0)
+	const AVCodecDescriptor *codec_descriptor =
+		avcodec_descriptor_get(codec_context->codec_id);
+	if (codec_descriptor != nullptr)
+		FormatDebug(ffmpeg_domain, "codec '%s'",
+			    codec_descriptor->name);
+#else
 	if (codec_context->codec_name[0] != 0)
 		FormatDebug(ffmpeg_domain, "codec '%s'",
 			    codec_context->codec_name);
+#endif
 
 	AVCodec *codec = avcodec_find_decoder(codec_context->codec_id);
 
@@ -493,10 +525,10 @@ ffmpeg_decode(Decoder &decoder, InputStream &input)
 			int64_t where =
 				time_to_ffmpeg(decoder_seek_where(decoder),
 					       av_stream->time_base) +
-				av_stream->start_time;
+				start_time_fallback(*av_stream);
 
 			if (av_seek_frame(format_context, audio_stream, where,
-					  AV_TIME_BASE) < 0)
+					  AVSEEK_FLAG_ANY) < 0)
 				decoder_seek_error(decoder);
 			else {
 				avcodec_flush_buffers(codec_context);

src/decoder/GmeDecoderPlugin.cxx

@@ -117,6 +117,7 @@ gme_container_scan(const char *path_fs, const unsigned int tnum)
 	}
 
 	const unsigned num_songs = gme_track_count(emu);
+	gme_delete(emu);
 	/* if it only contains a single tune, don't treat as container */
 	if (num_songs < 2)
 		return nullptr;
@@ -234,7 +235,7 @@ gme_scan_file(const char *path_fs,
 
 	if (ti->length > 0)
 		tag_handler_invoke_duration(handler, handler_ctx,
-					    ti->length / 100);
+					    ti->length / 1000);
 
 	if (ti->song != nullptr) {
 		if (gme_track_count(emu) > 1) {

src/decoder/MadDecoderPlugin.cxx

@@ -353,18 +353,8 @@ MadDecoder::ParseId3(size_t tagsize, Tag **mpd_tag)
 		memcpy(allocated, stream.this_frame, count);
 		mad_stream_skip(&(stream), count);
 
-		while (count < tagsize) {
+		if (!decoder_read_full(decoder, input_stream,
-			size_t len;
+				       allocated + count, tagsize - count)) {
-
-			len = decoder_read(decoder, input_stream,
-					   allocated + count, tagsize - count);
-			if (len == 0)
-				break;
-			else
-				count += len;
-		}
-
-		if (count != tagsize) {
 			LogDebug(mad_domain, "error parsing ID3 tag");
 			g_free(allocated);
 			return;
@@ -413,20 +403,7 @@ MadDecoder::ParseId3(size_t tagsize, Tag **mpd_tag)
 		mad_stream_skip(&stream, tagsize);
 	} else {
 		mad_stream_skip(&stream, count);
-
+		decoder_skip(decoder, input_stream, tagsize - count);
-		while (count < tagsize) {
-			size_t len = tagsize - count;
-			char ignored[1024];
-			if (len > sizeof(ignored))
-				len = sizeof(ignored);
-
-			len = decoder_read(decoder, input_stream,
-					   ignored, len);
-			if (len == 0)
-				break;
-			else
-				count += len;
-		}
 	}
 #endif
 }

src/decoder/OggUtil.cxx

@@ -81,7 +81,7 @@ bool
 OggExpectPageSeek(ogg_sync_state &oy, ogg_page &page,
 		  Decoder *decoder, InputStream &input_stream)
 {
-	size_t remaining_skipped = 16384;
+	size_t remaining_skipped = 32768;
 
 	while (true) {
 		int r = ogg_sync_pageseek(&oy, &page);

src/decoder/OpusDecoderPlugin.cxx

@@ -40,6 +40,7 @@
 #include <glib.h>
 
 #include <string.h>
+#include <stdio.h>
 
 static constexpr opus_int32 opus_sample_rate = 48000;
 

src/decoder/SndfileDecoderPlugin.cxx

@@ -31,10 +31,24 @@
 
 static constexpr Domain sndfile_domain("sndfile");
 
+struct SndfileInputStream {
+	Decoder *const decoder;
+	InputStream &is;
+
+	size_t Read(void *buffer, size_t size) {
+		/* libsndfile chokes on partial reads; therefore
+		   always force full reads */
+		return decoder_read_full(decoder, is, buffer, size)
+			? size
+			: 0;
+	}
+};
+
 static sf_count_t
 sndfile_vio_get_filelen(void *user_data)
 {
-	const InputStream &is = *(const InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
+	const InputStream &is = sis.is;
 
 	return is.GetSize();
 }
@@ -42,10 +56,14 @@ sndfile_vio_get_filelen(void *user_data)
 static sf_count_t
 sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
 {
-	InputStream &is = *(InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
+	InputStream &is = sis.is;
 
-	if (!is.LockSeek(offset, whence, IgnoreError()))
+	Error error;
+	if (!is.LockSeek(offset, whence, error)) {
+		LogError(error, "Seek failed");
 		return -1;
+	}
 
 	return is.GetOffset();
 }
@@ -53,16 +71,9 @@ sndfile_vio_seek(sf_count_t offset, int whence, void *user_data)
 static sf_count_t
 sndfile_vio_read(void *ptr, sf_count_t count, void *user_data)
 {
-	InputStream &is = *(InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
 
-	Error error;
+	return sis.Read(ptr, count);
-	size_t nbytes = is.LockRead(ptr, count, error);
-	if (nbytes == 0 && error.IsDefined()) {
-		LogError(error);
-		return -1;
-	}
-
-	return nbytes;
 }
 
 static sf_count_t
@@ -77,7 +88,8 @@ sndfile_vio_write(gcc_unused const void *ptr,
 static sf_count_t
 sndfile_vio_tell(void *user_data)
 {
-	const InputStream &is = *(const InputStream *)user_data;
+	SndfileInputStream &sis = *(SndfileInputStream *)user_data;
+	const InputStream &is = sis.is;
 
 	return is.GetOffset();
 }
@@ -123,7 +135,8 @@ sndfile_stream_decode(Decoder &decoder, InputStream &is)
 
 	info.format = 0;
 
-	sf = sf_open_virtual(&vio, SFM_READ, &info, &is);
+	SndfileInputStream sis{&decoder, is};
+	sf = sf_open_virtual(&vio, SFM_READ, &info, &sis);
 	if (sf == nullptr) {
 		LogWarning(sndfile_domain, "sf_open_virtual() failed");
 		return;

src/event/BufferedSocket.cxx

@@ -118,7 +118,7 @@ BufferedSocket::OnSocketReady(unsigned flags)
 		if (!ReadToBuffer() || !ResumeInput())
 			return false;
 
-		if (input.IsFull())
+		if (!input.IsFull())
 			ScheduleRead();
 	}
 

src/event/DeferredMonitor.cxx

@@ -27,9 +27,11 @@ DeferredMonitor::Cancel()
 #ifdef USE_EPOLL
 	pending = false;
 #else
-	const auto id = source_id.exchange(0);
+	const ScopeLock protect(mutex);
-	if (id != 0)
+	if (source_id != 0) {
-		g_source_remove(id);
+		g_source_remove(source_id);
+		source_id = 0;
+	}
 #endif
 }
 
@@ -40,10 +42,9 @@ DeferredMonitor::Schedule()
 	if (!pending.exchange(true))
 		fd.Write();
 #else
-	const unsigned id = loop.AddIdle(Callback, this);
+	const ScopeLock protect(mutex);
-	const auto old_id = source_id.exchange(id);
+	if (source_id == 0)
-	if (old_id != 0)
+		source_id = loop.AddIdle(Callback, this);
-		g_source_remove(old_id);
 #endif
 }
 
@@ -65,9 +66,16 @@ DeferredMonitor::OnSocketReady(unsigned)
 void
 DeferredMonitor::Run()
 {
-	const auto id = source_id.exchange(0);
+	{
-	if (id != 0)
+		const ScopeLock protect(mutex);
-		RunDeferred();
+		if (source_id == 0)
+			/* cancelled */
+			return;
+
+		source_id = 0;
+	}
+
+	RunDeferred();
 }
 
 gboolean

src/event/DeferredMonitor.hxx

@@ -27,6 +27,7 @@
 #include "SocketMonitor.hxx"
 #include "WakeFD.hxx"
 #else
+#include "thread/Mutex.hxx"
 #include <glib.h>
 #endif
 
@@ -48,7 +49,9 @@ class DeferredMonitor
 #else
 	EventLoop &loop;
 
-	std::atomic<guint> source_id;
+	Mutex mutex;
+
+	guint source_id;
 #endif
 
 public:

src/event/SignalMonitor.cxx

@@ -39,6 +39,12 @@
 
 #include <algorithm>
 
+#ifdef USE_SIGNALFD
+#include <pthread.h>
+#endif
+
+#include <signal.h>
+
 class SignalMonitor final : private SocketMonitor {
 #ifdef USE_SIGNALFD
 	SignalFD fd;
@@ -99,7 +105,21 @@ static std::atomic_bool signal_pending[MAX_SIGNAL];
 
 static Manual<SignalMonitor> monitor;
 
-#ifndef USE_SIGNALFD
+#ifdef USE_SIGNALFD
+
+/**
+ * This is a pthread_atfork() callback that unblocks the signals that
+ * were blocked for our signalfd().  Without this, our child processes
+ * would inherit the blocked signals.
+ */
+static void
+at_fork_child()
+{
+	sigprocmask(SIG_UNBLOCK, &signal_mask, nullptr);
+}
+
+#else
+
 static void
 SignalCallback(int signo)
 {
@@ -108,6 +128,7 @@ SignalCallback(int signo)
 	if (!signal_pending[signo].exchange(true))
 		monitor->WakeUp();
 }
+
 #endif
 
 void
@@ -115,6 +136,8 @@ SignalMonitorInit(EventLoop &loop)
 {
 #ifdef USE_SIGNALFD
 	sigemptyset(&signal_mask);
+
+	pthread_atfork(nullptr, nullptr, at_fork_child);
 #endif
 
 	monitor.Construct(loop);

src/event/TimeoutMonitor.cxx

@@ -64,7 +64,9 @@ TimeoutMonitor::ScheduleSeconds(unsigned s)
 void
 TimeoutMonitor::Run()
 {
-#ifndef USE_EPOLL
+#ifdef USE_EPOLL
+	active = false;
+#else
 	Cancel();
 #endif
 

src/output/AlsaOutputPlugin.cxx

@@ -106,12 +106,16 @@ struct AlsaOutput {
 	snd_pcm_uframes_t period_position;
 
 	/**
-	 * Set to non-zero when the Raspberry Pi workaround has been
+	 * Do we need to call snd_pcm_prepare() before the next write?
-	 * activated in alsa_recover(); decremented by each write.
+	 * It means that we put the device to SND_PCM_STATE_SETUP by
-	 * This will avoid activating it again, leading to an endless
+	 * calling snd_pcm_drop().
-	 * loop.  This problem was observed with a "RME Digi9636/52".
+	 *
+	 * Without this flag, we could easily recover after a failed
+	 * optimistic write (returning -EBADFD), but the Raspberry Pi
+	 * audio driver is infamous for generating ugly artefacts from
+	 * this.
 	 */
-	unsigned pi_workaround;
+	bool must_prepare;
 
 	/**
 	 * This buffer gets allocated after opening the ALSA device.
@@ -676,8 +680,6 @@ alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
-	ad->pi_workaround = 0;
-
 	int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
 			       SND_PCM_STREAM_PLAYBACK, ad->mode);
 	if (err < 0) {
@@ -699,6 +701,8 @@ alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
 	ad->in_frame_size = audio_format.GetFrameSize();
 	ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format);
 
+	ad->must_prepare = false;
+
 	return true;
 }
 
@@ -736,29 +740,6 @@ alsa_recover(AlsaOutput *ad, int err)
 	case SND_PCM_STATE_XRUN:
 		ad->period_position = 0;
 		err = snd_pcm_prepare(ad->pcm);
-
-		if (err == 0 && ad->pi_workaround == 0) {
-			/* this works around a driver bug observed on
-			   the Raspberry Pi: after snd_pcm_drop(), the
-			   whole ring buffer must be invalidated, but
-			   the snd_pcm_prepare() call above makes the
-			   driver play random data that just happens
-			   to be still in the buffer; by adding and
-			   cancelling some silence, this bug does not
-			   occur */
-			alsa_write_silence(ad, ad->period_frames);
-
-			/* cancel the silence data right away to avoid
-			   increasing latency; even though this
-			   function call invalidates the portion of
-			   silence, the driver seems to avoid the
-			   bug */
-			snd_pcm_reset(ad->pcm);
-
-			/* disable the workaround for some time */
-			ad->pi_workaround = 8;
-		}
-
 		break;
 	case SND_PCM_STATE_DISCONNECTED:
 		break;
@@ -801,6 +782,7 @@ alsa_cancel(struct audio_output *ao)
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
 	ad->period_position = 0;
+	ad->must_prepare = true;
 
 	snd_pcm_drop(ad->pcm);
 }
@@ -820,13 +802,34 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
+	assert(size > 0);
 	assert(size % ad->in_frame_size == 0);
 
+	if (ad->must_prepare) {
+		ad->must_prepare = false;
+
+		int err = snd_pcm_prepare(ad->pcm);
+		if (err < 0) {
+			error.Set(alsa_output_domain, err, snd_strerror(-err));
+			return 0;
+		}
+	}
+
+	const size_t original_size = size;
 	chunk = ad->pcm_export->Export(chunk, size, size);
+	if (size == 0)
+		/* the DoP (DSD over PCM) filter converts two frames
+		   at a time and ignores the last odd frame; if there
+		   was only one frame (e.g. the last frame in the
+		   file), the result is empty; to avoid an endless
+		   loop, bail out here, and pretend the one frame has
+		   been played */
+		return original_size;
 
 	assert(size % ad->out_frame_size == 0);
 
 	size /= ad->out_frame_size;
+	assert(size > 0);
 
 	while (true) {
 		snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
@@ -834,9 +837,6 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 			ad->period_position = (ad->period_position + ret)
 				% ad->period_frames;
 
-			if (ad->pi_workaround > 0)
-				--ad->pi_workaround;
-
 			size_t bytes_written = ret * ad->out_frame_size;
 			return ad->pcm_export->CalcSourceSize(bytes_written);
 		}

src/output/HttpdClient.cxx

@@ -30,6 +30,7 @@
 
 #include <assert.h>
 #include <string.h>
+#include <stdio.h>
 
 HttpdClient::~HttpdClient()
 {

src/output/OssOutputPlugin.cxx

@@ -727,6 +727,8 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	OssOutput *od = (OssOutput *)ao;
 	ssize_t ret;
 
+	assert(size > 0);
+
 	/* reopen the device since it was closed by dropBufferedAudio */
 	if (od->fd < 0 && !oss_reopen(od, error))
 		return 0;
@@ -735,6 +737,8 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	chunk = od->pcm_export->Export(chunk, size, size);
 #endif
 
+	assert(size > 0);
+
 	while (true) {
 		ret = write(od->fd, chunk, size);
 		if (ret > 0) {

src/output/PulseOutputPlugin.cxx

@@ -369,8 +369,6 @@ pulse_output_enable(struct audio_output *ao, Error &error)
 
 	po->mainloop = pa_threaded_mainloop_new();
 	if (po->mainloop == nullptr) {
-		g_free(po);
-
 		error.Set(pulse_output_domain,
 			  "pa_threaded_mainloop_new() has failed");
 		return false;

src/playlist/PlsPlaylistPlugin.cxx

@@ -31,6 +31,7 @@
 #include <glib.h>
 
 #include <string>
+#include <stdio.h>
 
 static constexpr Domain pls_domain("pls");
 

src/system/ByteOrder.hxx

@@ -40,6 +40,16 @@
 /* well-known big-endian */
 #  define IS_LITTLE_ENDIAN false
 #  define IS_BIG_ENDIAN true
+#elif defined(__APPLE__)
+/* compile-time check for MacOS */
+#  include <machine/endian.h>
+#  if BYTE_ORDER == LITTLE_ENDIAN
+#    define IS_LITTLE_ENDIAN true
+#    define IS_BIG_ENDIAN false
+#  else
+#    define IS_LITTLE_ENDIAN false
+#    define IS_BIG_ENDIAN true
+#  endif
 #else
 /* generic compile-time check */
 #  include <endian.h>

src/util/ASCII.hxx

@@ -33,7 +33,7 @@
 #include "Compiler.h"
 
 #include <assert.h>
-#include <string.h>
+#include <strings.h>
 
 /**
  * Determine whether two strings are equal, ignoring case for ASCII

src/win32/mpd_win32_rc.rc.in

@@ -3,7 +3,7 @@
 #define VERSION_NUMBER @VERSION_MAJOR@,@VERSION_MINOR@,@VERSION_REVISION@,@VERSION_EXTRA@
 #define VERSION_NUMBER_STR "@VERSION_MAJOR@,@VERSION_MINOR@,@VERSION_REVISION@,@VERSION_EXTRA@"
 
-MPD_ICON ICON "@top_srcdir@/src/win/mpd.ico"
+MPD_ICON ICON "@top_srcdir@/src/win32/mpd.ico"
 
 1 VERSIONINFO
 FILETYPE VFT_APP

test/FakeDecoderAPI.cxx

@@ -0,0 +1,144 @@
+/*
+ * 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 "config.h"
+#include "DecoderAPI.hxx"
+#include "InputStream.hxx"
+#include "util/Error.hxx"
+#include "Compiler.h"
+
+#include <glib.h>
+
+#include <unistd.h>
+
+void
+decoder_initialized(gcc_unused Decoder &decoder,
+		    gcc_unused const AudioFormat audio_format,
+		    gcc_unused bool seekable,
+		    gcc_unused float total_time)
+{
+}
+
+DecoderCommand
+decoder_get_command(gcc_unused Decoder &decoder)
+{
+	return DecoderCommand::NONE;
+}
+
+void
+decoder_command_finished(gcc_unused Decoder &decoder)
+{
+}
+
+double
+decoder_seek_where(gcc_unused Decoder &decoder)
+{
+	return 1.0;
+}
+
+void
+decoder_seek_error(gcc_unused Decoder &decoder)
+{
+}
+
+size_t
+decoder_read(gcc_unused Decoder *decoder,
+	     InputStream &is,
+	     void *buffer, size_t length)
+{
+	return is.LockRead(buffer, length, IgnoreError());
+}
+
+bool
+decoder_read_full(Decoder *decoder, InputStream &is,
+		  void *_buffer, size_t size)
+{
+	uint8_t *buffer = (uint8_t *)_buffer;
+
+	while (size > 0) {
+		size_t nbytes = decoder_read(decoder, is, buffer, size);
+		if (nbytes == 0)
+			return false;
+
+		buffer += nbytes;
+		size -= nbytes;
+	}
+
+	return true;
+}
+
+bool
+decoder_skip(Decoder *decoder, InputStream &is, size_t size)
+{
+	while (size > 0) {
+		char buffer[1024];
+		size_t nbytes = decoder_read(decoder, is, buffer,
+					     std::min(sizeof(buffer), size));
+		if (nbytes == 0)
+			return false;
+
+		size -= nbytes;
+	}
+
+	return true;
+}
+
+void
+decoder_timestamp(gcc_unused Decoder &decoder,
+		  gcc_unused double t)
+{
+}
+
+DecoderCommand
+decoder_data(gcc_unused Decoder &decoder,
+	     gcc_unused InputStream *is,
+	     const void *data, size_t datalen,
+	     gcc_unused uint16_t kbit_rate)
+{
+	gcc_unused ssize_t nbytes = write(1, data, datalen);
+	return DecoderCommand::NONE;
+}
+
+DecoderCommand
+decoder_tag(gcc_unused Decoder &decoder,
+	    gcc_unused InputStream *is,
+	    gcc_unused Tag &&tag)
+{
+	return DecoderCommand::NONE;
+}
+
+void
+decoder_replay_gain(gcc_unused Decoder &decoder,
+		    const ReplayGainInfo *rgi)
+{
+	const ReplayGainTuple *tuple = &rgi->tuples[REPLAY_GAIN_ALBUM];
+	if (tuple->IsDefined())
+		g_printerr("replay_gain[album]: gain=%f peak=%f\n",
+			   tuple->gain, tuple->peak);
+
+	tuple = &rgi->tuples[REPLAY_GAIN_TRACK];
+	if (tuple->IsDefined())
+		g_printerr("replay_gain[track]: gain=%f peak=%f\n",
+			   tuple->gain, tuple->peak);
+}
+
+void
+decoder_mixramp(gcc_unused Decoder &decoder, gcc_unused MixRampInfo &&mix_ramp)
+{
+}

test/dump_playlist.cxx

@@ -24,7 +24,6 @@
 #include "Directory.hxx"
 #include "InputStream.hxx"
 #include "ConfigGlobal.hxx"
-#include "DecoderAPI.hxx"
 #include "DecoderList.hxx"
 #include "InputInit.hxx"
 #include "IOThread.hxx"
@@ -53,88 +52,6 @@ my_log_func(const gchar *log_domain, gcc_unused GLogLevelFlags log_level,
 		g_printerr("%s\n", message);
 }
 
-void
-decoder_initialized(gcc_unused Decoder &decoder,
-		    gcc_unused const AudioFormat audio_format,
-		    gcc_unused bool seekable,
-		    gcc_unused float total_time)
-{
-}
-
-DecoderCommand
-decoder_get_command(gcc_unused Decoder &decoder)
-{
-	return DecoderCommand::NONE;
-}
-
-void
-decoder_command_finished(gcc_unused Decoder &decoder)
-{
-}
-
-double
-decoder_seek_where(gcc_unused Decoder &decoder)
-{
-	return 1.0;
-}
-
-void
-decoder_seek_error(gcc_unused Decoder &decoder)
-{
-}
-
-size_t
-decoder_read(gcc_unused Decoder *decoder,
-	     InputStream &is,
-	     void *buffer, size_t length)
-{
-	return is.LockRead(buffer, length, IgnoreError());
-}
-
-void
-decoder_timestamp(gcc_unused Decoder &decoder,
-		  gcc_unused double t)
-{
-}
-
-DecoderCommand
-decoder_data(gcc_unused Decoder &decoder,
-	     gcc_unused InputStream *is,
-	     const void *data, size_t datalen,
-	     gcc_unused uint16_t kbit_rate)
-{
-	gcc_unused ssize_t nbytes = write(1, data, datalen);
-	return DecoderCommand::NONE;
-}
-
-DecoderCommand
-decoder_tag(gcc_unused Decoder &decoder,
-	    gcc_unused InputStream *is,
-	    gcc_unused Tag &&tag)
-{
-	return DecoderCommand::NONE;
-}
-
-void
-decoder_replay_gain(gcc_unused Decoder &decoder,
-		    const ReplayGainInfo *rgi)
-{
-	const ReplayGainTuple *tuple = &rgi->tuples[REPLAY_GAIN_ALBUM];
-	if (tuple->IsDefined())
-		g_printerr("replay_gain[album]: gain=%f peak=%f\n",
-			   tuple->gain, tuple->peak);
-
-	tuple = &rgi->tuples[REPLAY_GAIN_TRACK];
-	if (tuple->IsDefined())
-		g_printerr("replay_gain[track]: gain=%f peak=%f\n",
-			   tuple->gain, tuple->peak);
-}
-
-void
-decoder_mixramp(gcc_unused Decoder &decoder, gcc_unused MixRampInfo &&mix_ramp)
-{
-}
-
 int main(int argc, char **argv)
 {
 	const char *uri;

test/read_tags.cxx

@@ -20,7 +20,7 @@
 #include "config.h"
 #include "IOThread.hxx"
 #include "DecoderList.hxx"
-#include "DecoderAPI.hxx"
+#include "DecoderPlugin.hxx"
 #include "InputInit.hxx"
 #include "InputStream.hxx"
 #include "AudioFormat.hxx"
@@ -42,79 +42,6 @@
 #include <locale.h>
 #endif
 
-void
-decoder_initialized(gcc_unused Decoder &decoder,
-		    gcc_unused const AudioFormat audio_format,
-		    gcc_unused bool seekable,
-		    gcc_unused float total_time)
-{
-}
-
-DecoderCommand
-decoder_get_command(gcc_unused Decoder &decoder)
-{
-	return DecoderCommand::NONE;
-}
-
-void
-decoder_command_finished(gcc_unused Decoder &decoder)
-{
-}
-
-double
-decoder_seek_where(gcc_unused Decoder &decoder)
-{
-	return 1.0;
-}
-
-void
-decoder_seek_error(gcc_unused Decoder &decoder)
-{
-}
-
-size_t
-decoder_read(gcc_unused Decoder *decoder,
-	     InputStream &is,
-	     void *buffer, size_t length)
-{
-	return is.LockRead(buffer, length, IgnoreError());
-}
-
-void
-decoder_timestamp(gcc_unused Decoder &decoder,
-		  gcc_unused double t)
-{
-}
-
-DecoderCommand
-decoder_data(gcc_unused Decoder &decoder,
-	     gcc_unused InputStream *is,
-	     const void *data, size_t datalen,
-	     gcc_unused uint16_t kbit_rate)
-{
-	gcc_unused ssize_t nbytes = write(1, data, datalen);
-	return DecoderCommand::NONE;
-}
-
-DecoderCommand
-decoder_tag(gcc_unused Decoder &decoder,
-	    gcc_unused InputStream *is,
-	    gcc_unused Tag &&tag)
-{
-	return DecoderCommand::NONE;
-}
-
-void
-decoder_replay_gain(gcc_unused Decoder &decoder,
-		    gcc_unused const ReplayGainInfo *replay_gain_info)
-{
-}
-
-void
-decoder_mixramp(gcc_unused Decoder &decoder, gcc_unused MixRampInfo &&mix_ramp)
-{
-}
-
 static bool empty = true;
 
 static void

test/run_decoder.cxx

@@ -101,6 +101,40 @@ decoder_read(gcc_unused Decoder *decoder,
 	return is.LockRead(buffer, length, IgnoreError());
 }
 
+bool
+decoder_read_full(Decoder *decoder, InputStream &is,
+		  void *_buffer, size_t size)
+{
+	uint8_t *buffer = (uint8_t *)_buffer;
+
+	while (size > 0) {
+		size_t nbytes = decoder_read(decoder, is, buffer, size);
+		if (nbytes == 0)
+			return false;
+
+		buffer += nbytes;
+		size -= nbytes;
+	}
+
+	return true;
+}
+
+bool
+decoder_skip(Decoder *decoder, InputStream &is, size_t size)
+{
+	while (size > 0) {
+		char buffer[1024];
+		size_t nbytes = decoder_read(decoder, is, buffer,
+					     std::min(sizeof(buffer), size));
+		if (nbytes == 0)
+			return false;
+
+		size -= nbytes;
+	}
+
+	return true;
+}
+
 void
 decoder_timestamp(gcc_unused Decoder &decoder,
 		  gcc_unused double t)