68 files changed, 5472 insertions, 649 deletions

diff --git a/src/output/Domain.cxx b/src/output/Domain.cxx
new file mode 100644
index 000000000..878e5f3c5
--- /dev/null
+++ b/src/output/Domain.cxx
@@ -0,0 +1,23 @@
+/*
+ * 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.
+ */
+
+#include "Domain.hxx"
+#include "util/Domain.hxx"
+
+const Domain output_domain("output");
diff --git a/src/output/Domain.hxx b/src/output/Domain.hxx
new file mode 100644
index 000000000..e3a20142f
--- /dev/null
+++ b/src/output/Domain.hxx
@@ -0,0 +1,25 @@
+/*
+ * 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_OUTPUT_ERROR_HXX
+#define MPD_OUTPUT_ERROR_HXX
+
+extern const class Domain output_domain;
+
+#endif
diff --git a/src/output/Finish.cxx b/src/output/Finish.cxx
new file mode 100644
index 000000000..be2ca463e
--- /dev/null
+++ b/src/output/Finish.cxx
@@ -0,0 +1,50 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "Internal.hxx"
+#include "OutputPlugin.hxx"
+#include "mixer/MixerControl.hxx"
+#include "filter/FilterInternal.hxx"
+
+#include <assert.h>
+
+AudioOutput::~AudioOutput()
+{
+	assert(!open);
+	assert(!fail_timer.IsDefined());
+	assert(!thread.IsDefined());
+
+	if (mixer != nullptr)
+		mixer_free(mixer);
+
+	delete replay_gain_filter;
+	delete other_replay_gain_filter;
+	delete filter;
+}
+
+void
+audio_output_free(AudioOutput *ao)
+{
+	assert(!ao->open);
+	assert(!ao->fail_timer.IsDefined());
+	assert(!ao->thread.IsDefined());
+
+	ao_plugin_finish(ao);
+}
diff --git a/src/output/Init.cxx b/src/output/Init.cxx
new file mode 100644
index 000000000..eafcec432
--- /dev/null
+++ b/src/output/Init.cxx
@@ -0,0 +1,335 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "Internal.hxx"
+#include "Registry.hxx"
+#include "Domain.hxx"
+#include "OutputAPI.hxx"
+#include "filter/FilterConfig.hxx"
+#include "AudioParser.hxx"
+#include "mixer/MixerList.hxx"
+#include "mixer/MixerType.hxx"
+#include "mixer/MixerControl.hxx"
+#include "mixer/plugins/SoftwareMixerPlugin.hxx"
+#include "filter/FilterPlugin.hxx"
+#include "filter/FilterRegistry.hxx"
+#include "filter/plugins/AutoConvertFilterPlugin.hxx"
+#include "filter/plugins/ReplayGainFilterPlugin.hxx"
+#include "filter/plugins/ChainFilterPlugin.hxx"
+#include "config/ConfigError.hxx"
+#include "config/ConfigGlobal.hxx"
+#include "util/Error.hxx"
+#include "Log.hxx"
+
+#include <assert.h>
+#include <string.h>
+
+#define AUDIO_OUTPUT_TYPE	"type"
+#define AUDIO_OUTPUT_NAME	"name"
+#define AUDIO_OUTPUT_FORMAT	"format"
+#define AUDIO_FILTERS		"filters"
+
+AudioOutput::AudioOutput(const AudioOutputPlugin &_plugin)
+	:plugin(_plugin),
+	 enabled(true), really_enabled(false),
+	 open(false),
+	 pause(false),
+	 allow_play(true),
+	 in_playback_loop(false),
+	 woken_for_play(false),
+	 filter(nullptr),
+	 replay_gain_filter(nullptr),
+	 other_replay_gain_filter(nullptr),
+	 command(AO_COMMAND_NONE)
+{
+	assert(plugin.finish != nullptr);
+	assert(plugin.open != nullptr);
+	assert(plugin.close != nullptr);
+	assert(plugin.play != nullptr);
+}
+
+static const AudioOutputPlugin *
+audio_output_detect(Error &error)
+{
+	LogDefault(output_domain, "Attempt to detect audio output device");
+
+	audio_output_plugins_for_each(plugin) {
+		if (plugin->test_default_device == nullptr)
+			continue;
+
+		FormatDefault(output_domain,
+			      "Attempting to detect a %s audio device",
+			      plugin->name);
+		if (ao_plugin_test_default_device(plugin))
+			return plugin;
+	}
+
+	error.Set(output_domain, "Unable to detect an audio device");
+	return nullptr;
+}
+
+/**
+ * Determines the mixer type which should be used for the specified
+ * configuration block.
+ *
+ * This handles the deprecated options mixer_type (global) and
+ * mixer_enabled, if the mixer_type setting is not configured.
+ */
+gcc_pure
+static enum mixer_type
+audio_output_mixer_type(const config_param &param)
+{
+	/* read the local "mixer_type" setting */
+	const char *p = param.GetBlockValue("mixer_type");
+	if (p != nullptr)
+		return mixer_type_parse(p);
+
+	/* try the local "mixer_enabled" setting next (deprecated) */
+	if (!param.GetBlockValue("mixer_enabled", true))
+		return MIXER_TYPE_NONE;
+
+	/* fall back to the global "mixer_type" setting (also
+	   deprecated) */
+	return mixer_type_parse(config_get_string(CONF_MIXER_TYPE,
+						  "hardware"));
+}
+
+static Mixer *
+audio_output_load_mixer(EventLoop &event_loop, AudioOutput &ao,
+			const config_param &param,
+			const MixerPlugin *plugin,
+			Filter &filter_chain,
+			MixerListener &listener,
+			Error &error)
+{
+	Mixer *mixer;
+
+	switch (audio_output_mixer_type(param)) {
+	case MIXER_TYPE_NONE:
+	case MIXER_TYPE_UNKNOWN:
+		return nullptr;
+
+	case MIXER_TYPE_HARDWARE:
+		if (plugin == nullptr)
+			return nullptr;
+
+		return mixer_new(event_loop, *plugin, ao, listener,
+				 param, error);
+
+	case MIXER_TYPE_SOFTWARE:
+		mixer = mixer_new(event_loop, software_mixer_plugin, ao,
+				  listener,
+				  config_param(),
+				  IgnoreError());
+		assert(mixer != nullptr);
+
+		filter_chain_append(filter_chain, "software_mixer",
+				    software_mixer_get_filter(mixer));
+		return mixer;
+	}
+
+	assert(false);
+	gcc_unreachable();
+}
+
+bool
+AudioOutput::Configure(const config_param &param, Error &error)
+{
+	if (!param.IsNull()) {
+		name = param.GetBlockValue(AUDIO_OUTPUT_NAME);
+		if (name == nullptr) {
+			error.Set(config_domain,
+				  "Missing \"name\" configuration");
+			return false;
+		}
+
+		const char *p = param.GetBlockValue(AUDIO_OUTPUT_FORMAT);
+		if (p != nullptr) {
+			bool success =
+				audio_format_parse(config_audio_format,
+						   p, true, error);
+			if (!success)
+				return false;
+		} else
+			config_audio_format.Clear();
+	} else {
+		name = "default detected output";
+
+		config_audio_format.Clear();
+	}
+
+	tags = param.GetBlockValue("tags", true);
+	always_on = param.GetBlockValue("always_on", false);
+	enabled = param.GetBlockValue("enabled", true);
+
+	/* set up the filter chain */
+
+	filter = filter_chain_new();
+	assert(filter != nullptr);
+
+	/* create the normalization filter (if configured) */
+
+	if (config_get_bool(CONF_VOLUME_NORMALIZATION, false)) {
+		Filter *normalize_filter =
+			filter_new(&normalize_filter_plugin, config_param(),
+				   IgnoreError());
+		assert(normalize_filter != nullptr);
+
+		filter_chain_append(*filter, "normalize",
+				    autoconvert_filter_new(normalize_filter));
+	}
+
+	Error filter_error;
+	filter_chain_parse(*filter,
+			   param.GetBlockValue(AUDIO_FILTERS, ""),
+			   filter_error);
+
+	// It's not really fatal - Part of the filter chain has been set up already
+	// and even an empty one will work (if only with unexpected behaviour)
+	if (filter_error.IsDefined())
+		FormatError(filter_error,
+			    "Failed to initialize filter chain for '%s'",
+			    name);
+
+	/* done */
+
+	return true;
+}
+
+static bool
+audio_output_setup(EventLoop &event_loop, AudioOutput &ao,
+		   MixerListener &mixer_listener,
+		   const config_param &param,
+		   Error &error)
+{
+
+	/* create the replay_gain filter */
+
+	const char *replay_gain_handler =
+		param.GetBlockValue("replay_gain_handler", "software");
+
+	if (strcmp(replay_gain_handler, "none") != 0) {
+		ao.replay_gain_filter = filter_new(&replay_gain_filter_plugin,
+						   param, IgnoreError());
+		assert(ao.replay_gain_filter != nullptr);
+
+		ao.replay_gain_serial = 0;
+
+		ao.other_replay_gain_filter = filter_new(&replay_gain_filter_plugin,
+							 param,
+							 IgnoreError());
+		assert(ao.other_replay_gain_filter != nullptr);
+
+		ao.other_replay_gain_serial = 0;
+	} else {
+		ao.replay_gain_filter = nullptr;
+		ao.other_replay_gain_filter = nullptr;
+	}
+
+	/* set up the mixer */
+
+	Error mixer_error;
+	ao.mixer = audio_output_load_mixer(event_loop, ao, param,
+					   ao.plugin.mixer_plugin,
+					   *ao.filter,
+					   mixer_listener,
+					   mixer_error);
+	if (ao.mixer == nullptr && mixer_error.IsDefined())
+		FormatError(mixer_error,
+			    "Failed to initialize hardware mixer for '%s'",
+			    ao.name);
+
+	/* use the hardware mixer for replay gain? */
+
+	if (strcmp(replay_gain_handler, "mixer") == 0) {
+		if (ao.mixer != nullptr)
+			replay_gain_filter_set_mixer(ao.replay_gain_filter,
+						     ao.mixer, 100);
+		else
+			FormatError(output_domain,
+				    "No such mixer for output '%s'", ao.name);
+	} else if (strcmp(replay_gain_handler, "software") != 0 &&
+		   ao.replay_gain_filter != nullptr) {
+		error.Set(config_domain,
+			  "Invalid \"replay_gain_handler\" value");
+		return false;
+	}
+
+	/* the "convert" filter must be the last one in the chain */
+
+	ao.convert_filter = filter_new(&convert_filter_plugin, config_param(),
+					IgnoreError());
+	assert(ao.convert_filter != nullptr);
+
+	filter_chain_append(*ao.filter, "convert", ao.convert_filter);
+
+	return true;
+}
+
+AudioOutput *
+audio_output_new(EventLoop &event_loop, const config_param &param,
+		 MixerListener &mixer_listener,
+		 PlayerControl &pc,
+		 Error &error)
+{
+	const AudioOutputPlugin *plugin;
+
+	if (!param.IsNull()) {
+		const char *p;
+
+		p = param.GetBlockValue(AUDIO_OUTPUT_TYPE);
+		if (p == nullptr) {
+			error.Set(config_domain,
+				  "Missing \"type\" configuration");
+			return nullptr;
+		}
+
+		plugin = AudioOutputPlugin_get(p);
+		if (plugin == nullptr) {
+			error.Format(config_domain,
+				     "No such audio output plugin: %s", p);
+			return nullptr;
+		}
+	} else {
+		LogWarning(output_domain,
+			   "No 'AudioOutput' defined in config file");
+
+		plugin = audio_output_detect(error);
+		if (plugin == nullptr)
+			return nullptr;
+
+		FormatDefault(output_domain,
+			      "Successfully detected a %s audio device",
+			      plugin->name);
+	}
+
+	AudioOutput *ao = ao_plugin_init(plugin, param, error);
+	if (ao == nullptr)
+		return nullptr;
+
+	if (!audio_output_setup(event_loop, *ao, mixer_listener,
+				param, error)) {
+		ao_plugin_finish(ao);
+		return nullptr;
+	}
+
+	ao->player_control = &pc;
+	return ao;
+}
diff --git a/src/output/Internal.hxx b/src/output/Internal.hxx
new file mode 100644
index 000000000..658ebd4e1
--- /dev/null
+++ b/src/output/Internal.hxx
@@ -0,0 +1,436 @@
+/*
+ * 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_OUTPUT_INTERNAL_HXX
+#define MPD_OUTPUT_INTERNAL_HXX
+
+#include "AudioFormat.hxx"
+#include "pcm/PcmBuffer.hxx"
+#include "pcm/PcmDither.hxx"
+#include "ReplayGainInfo.hxx"
+#include "thread/Mutex.hxx"
+#include "thread/Cond.hxx"
+#include "thread/Thread.hxx"
+#include "system/PeriodClock.hxx"
+
+class Error;
+class Filter;
+class MusicPipe;
+class EventLoop;
+class Mixer;
+class MixerListener;
+struct MusicChunk;
+struct config_param;
+struct PlayerControl;
+struct AudioOutputPlugin;
+
+enum audio_output_command {
+	AO_COMMAND_NONE = 0,
+	AO_COMMAND_ENABLE,
+	AO_COMMAND_DISABLE,
+	AO_COMMAND_OPEN,
+
+	/**
+	 * This command is invoked when the input audio format
+	 * changes.
+	 */
+	AO_COMMAND_REOPEN,
+
+	AO_COMMAND_CLOSE,
+	AO_COMMAND_PAUSE,
+
+	/**
+	 * Drains the internal (hardware) buffers of the device.  This
+	 * operation may take a while to complete.
+	 */
+	AO_COMMAND_DRAIN,
+
+	AO_COMMAND_CANCEL,
+	AO_COMMAND_KILL
+};
+
+struct AudioOutput {
+	/**
+	 * The device's configured display name.
+	 */
+	const char *name;
+
+	/**
+	 * The plugin which implements this output device.
+	 */
+	const AudioOutputPlugin &plugin;
+
+	/**
+	 * The #mixer object associated with this audio output device.
+	 * May be nullptr if none is available, or if software volume is
+	 * configured.
+	 */
+	Mixer *mixer;
+
+	/**
+	 * Will this output receive tags from the decoder?  The
+	 * default is true, but it may be configured to false to
+	 * suppress sending tags to the output.
+	 */
+	bool tags;
+
+	/**
+	 * Shall this output always play something (i.e. silence),
+	 * even when playback is stopped?
+	 */
+	bool always_on;
+
+	/**
+	 * Has the user enabled this device?
+	 */
+	bool enabled;
+
+	/**
+	 * Is this device actually enabled, i.e. the "enable" method
+	 * has succeeded?
+	 */
+	bool really_enabled;
+
+	/**
+	 * Is the device (already) open and functional?
+	 *
+	 * This attribute may only be modified by the output thread.
+	 * It is protected with #mutex: write accesses inside the
+	 * output thread and read accesses outside of it may only be
+	 * performed while the lock is held.
+	 */
+	bool open;
+
+	/**
+	 * Is the device paused?  i.e. the output thread is in the
+	 * ao_pause() loop.
+	 */
+	bool pause;
+
+	/**
+	 * When this flag is set, the output thread will not do any
+	 * playback.  It will wait until the flag is cleared.
+	 *
+	 * This is used to synchronize the "clear" operation on the
+	 * shared music pipe during the CANCEL command.
+	 */
+	bool allow_play;
+
+	/**
+	 * True while the OutputThread is inside ao_play().  This
+	 * means the PlayerThread does not need to wake up the
+	 * OutputThread when new chunks are added to the MusicPipe,
+	 * because the OutputThread is already watching that.
+	 */
+	bool in_playback_loop;
+
+	/**
+	 * Has the OutputThread been woken up to play more chunks?
+	 * This is set by audio_output_play() and reset by ao_play()
+	 * to reduce the number of duplicate wakeups.
+	 */
+	bool woken_for_play;
+
+	/**
+	 * If not nullptr, the device has failed, and this timer is used
+	 * to estimate how long it should stay disabled (unless
+	 * explicitly reopened with "play").
+	 */
+	PeriodClock fail_timer;
+
+	/**
+	 * The configured audio format.
+	 */
+	AudioFormat config_audio_format;
+
+	/**
+	 * The audio_format in which audio data is received from the
+	 * player thread (which in turn receives it from the decoder).
+	 */
+	AudioFormat in_audio_format;
+
+	/**
+	 * The audio_format which is really sent to the device.  This
+	 * is basically config_audio_format (if configured) or
+	 * in_audio_format, but may have been modified by
+	 * plugin->open().
+	 */
+	AudioFormat out_audio_format;
+
+	/**
+	 * The buffer used to allocate the cross-fading result.
+	 */
+	PcmBuffer cross_fade_buffer;
+
+	/**
+	 * The dithering state for cross-fading two streams.
+	 */
+	PcmDither cross_fade_dither;
+
+	/**
+	 * The filter object of this audio output.  This is an
+	 * instance of chain_filter_plugin.
+	 */
+	Filter *filter;
+
+	/**
+	 * The replay_gain_filter_plugin instance of this audio
+	 * output.
+	 */
+	Filter *replay_gain_filter;
+
+	/**
+	 * The serial number of the last replay gain info.  0 means no
+	 * replay gain info was available.
+	 */
+	unsigned replay_gain_serial;
+
+	/**
+	 * The replay_gain_filter_plugin instance of this audio
+	 * output, to be applied to the second chunk during
+	 * cross-fading.
+	 */
+	Filter *other_replay_gain_filter;
+
+	/**
+	 * The serial number of the last replay gain info by the
+	 * "other" chunk during cross-fading.
+	 */
+	unsigned other_replay_gain_serial;
+
+	/**
+	 * The convert_filter_plugin instance of this audio output.
+	 * It is the last item in the filter chain, and is responsible
+	 * for converting the input data into the appropriate format
+	 * for this audio output.
+	 */
+	Filter *convert_filter;
+
+	/**
+	 * The thread handle, or nullptr if the output thread isn't
+	 * running.
+	 */
+	Thread thread;
+
+	/**
+	 * The next command to be performed by the output thread.
+	 */
+	enum audio_output_command command;
+
+	/**
+	 * The music pipe which provides music chunks to be played.
+	 */
+	const MusicPipe *pipe;
+
+	/**
+	 * This mutex protects #open, #fail_timer, #current_chunk and
+	 * #current_chunk_finished.
+	 */
+	Mutex mutex;
+
+	/**
+	 * This condition object wakes up the output thread after
+	 * #command has been set.
+	 */
+	Cond cond;
+
+	/**
+	 * The PlayerControl object which "owns" this output.  This
+	 * object is needed to signal command completion.
+	 */
+	PlayerControl *player_control;
+
+	/**
+	 * The #MusicChunk which is currently being played.  All
+	 * chunks before this one may be returned to the
+	 * #music_buffer, because they are not going to be used by
+	 * this output anymore.
+	 */
+	const MusicChunk *current_chunk;
+
+	/**
+	 * Has the output finished playing #current_chunk?
+	 */
+	bool current_chunk_finished;
+
+	AudioOutput(const AudioOutputPlugin &_plugin);
+	~AudioOutput();
+
+	bool Configure(const config_param &param, Error &error);
+
+	void StartThread();
+	void StopThread();
+
+	void Finish();
+
+	bool IsOpen() const {
+		return open;
+	}
+
+	bool IsCommandFinished() const {
+		return command == AO_COMMAND_NONE;
+	}
+
+	/**
+	 * Waits for command completion.
+	 *
+	 * Caller must lock the mutex.
+	 */
+	void WaitForCommand();
+
+	/**
+	 * Sends a command, but does not wait for completion.
+	 *
+	 * Caller must lock the mutex.
+	 */
+	void CommandAsync(audio_output_command cmd);
+
+	/**
+	 * Sends a command to the #AudioOutput object and waits for
+	 * completion.
+	 *
+	 * Caller must lock the mutex.
+	 */
+	void CommandWait(audio_output_command cmd);
+
+	/**
+	 * Lock the #AudioOutput object and execute the command
+	 * synchronously.
+	 */
+	void LockCommandWait(audio_output_command cmd);
+
+	/**
+	 * Enables the device.
+	 */
+	void LockEnableWait();
+
+	/**
+	 * Disables the device.
+	 */
+	void LockDisableWait();
+
+	void LockPauseAsync();
+
+	/**
+	 * Same LockCloseWait(), but expects the lock to be
+	 * held by the caller.
+	 */
+	void CloseWait();
+	void LockCloseWait();
+
+	/**
+	 * Closes the audio output, but if the "always_on" flag is set, put it
+	 * into pause mode instead.
+	 */
+	void LockRelease();
+
+	void SetReplayGainMode(ReplayGainMode mode);
+
+	/**
+	 * Caller must lock the mutex.
+	 */
+	bool Open(const AudioFormat audio_format, const MusicPipe &mp);
+
+	/**
+	 * Opens or closes the device, depending on the "enabled"
+	 * flag.
+	 *
+	 * @return true if the device is open
+	 */
+	bool LockUpdate(const AudioFormat audio_format,
+			const MusicPipe &mp);
+
+	void LockPlay();
+
+	void LockDrainAsync();
+
+	/**
+	 * Clear the "allow_play" flag and send the "CANCEL" command
+	 * asynchronously.  To finish the operation, the caller has to
+	 * call LockAllowPlay().
+	 */
+	void LockCancelAsync();
+
+	/**
+	 * Set the "allow_play" and signal the thread.
+	 */
+	void LockAllowPlay();
+
+private:
+	void CommandFinished();
+
+	bool Enable();
+	void Disable();
+
+	void Open();
+	void Close(bool drain);
+	void Reopen();
+
+	AudioFormat OpenFilter(AudioFormat &format, Error &error_r);
+	void CloseFilter();
+	void ReopenFilter();
+
+	/**
+	 * Wait until the output's delay reaches zero.
+	 *
+	 * @return true if playback should be continued, false if a
+	 * command was issued
+	 */
+	bool WaitForDelay();
+
+	gcc_pure
+	const MusicChunk *GetNextChunk() const;
+
+	bool PlayChunk(const MusicChunk *chunk);
+
+	/**
+	 * Plays all remaining chunks, until the tail of the pipe has
+	 * been reached (and no more chunks are queued), or until a
+	 * command is received.
+	 *
+	 * @return true if at least one chunk has been available,
+	 * false if the tail of the pipe was already reached
+	 */
+	bool Play();
+
+	void Pause();
+
+	/**
+	 * The OutputThread.
+	 */
+	void Task();
+	static void Task(void *arg);
+};
+
+/**
+ * Notify object used by the thread's client, i.e. we will send a
+ * notify signal to this object, expecting the caller to wait on it.
+ */
+extern struct notify audio_output_client_notify;
+
+AudioOutput *
+audio_output_new(EventLoop &event_loop, const config_param &param,
+		 MixerListener &mixer_listener,
+		 PlayerControl &pc,
+		 Error &error);
+
+void
+audio_output_free(AudioOutput *ao);
+
+#endif
diff --git a/src/output/MultipleOutputs.cxx b/src/output/MultipleOutputs.cxx
new file mode 100644
index 000000000..33ab57894
--- /dev/null
+++ b/src/output/MultipleOutputs.cxx
@@ -0,0 +1,482 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "MultipleOutputs.hxx"
+#include "PlayerControl.hxx"
+#include "Internal.hxx"
+#include "Domain.hxx"
+#include "MusicBuffer.hxx"
+#include "MusicPipe.hxx"
+#include "MusicChunk.hxx"
+#include "system/FatalError.hxx"
+#include "util/Error.hxx"
+#include "config/ConfigData.hxx"
+#include "config/ConfigGlobal.hxx"
+#include "config/ConfigOption.hxx"
+#include "notify.hxx"
+
+#include <assert.h>
+#include <string.h>
+
+MultipleOutputs::MultipleOutputs(MixerListener &_mixer_listener)
+	:mixer_listener(_mixer_listener),
+	 input_audio_format(AudioFormat::Undefined()),
+	 buffer(nullptr), pipe(nullptr),
+	 elapsed_time(SignedSongTime::Negative())
+{
+}
+
+MultipleOutputs::~MultipleOutputs()
+{
+	for (auto i : outputs) {
+		i->LockDisableWait();
+		i->Finish();
+	}
+}
+
+static AudioOutput *
+LoadOutput(EventLoop &event_loop, MixerListener &mixer_listener,
+	   PlayerControl &pc, const config_param &param)
+{
+	Error error;
+	AudioOutput *output = audio_output_new(event_loop, param,
+					       mixer_listener,
+					       pc, error);
+	if (output == nullptr) {
+		if (param.line > 0)
+			FormatFatalError("line %i: %s",
+					 param.line,
+					 error.GetMessage());
+		else
+			FatalError(error);
+	}
+
+	return output;
+}
+
+void
+MultipleOutputs::Configure(EventLoop &event_loop, PlayerControl &pc)
+{
+	for (const config_param *param = config_get_param(CONF_AUDIO_OUTPUT);
+	     param != nullptr; param = param->next) {
+		auto output = LoadOutput(event_loop, mixer_listener,
+					 pc, *param);
+		if (FindByName(output->name) != nullptr)
+			FormatFatalError("output devices with identical "
+					 "names: %s", output->name);
+
+		outputs.push_back(output);
+	}
+
+	if (outputs.empty()) {
+		/* auto-detect device */
+		const config_param empty;
+		auto output = LoadOutput(event_loop, mixer_listener,
+					 pc, empty);
+		outputs.push_back(output);
+	}
+}
+
+AudioOutput *
+MultipleOutputs::FindByName(const char *name) const
+{
+	for (auto i : outputs)
+		if (strcmp(i->name, name) == 0)
+			return i;
+
+	return nullptr;
+}
+
+void
+MultipleOutputs::EnableDisable()
+{
+	for (auto ao : outputs) {
+		bool enabled;
+
+		ao->mutex.lock();
+		enabled = ao->really_enabled;
+		ao->mutex.unlock();
+
+		if (ao->enabled != enabled) {
+			if (ao->enabled)
+				ao->LockEnableWait();
+			else
+				ao->LockDisableWait();
+		}
+	}
+}
+
+bool
+MultipleOutputs::AllFinished() const
+{
+	for (auto ao : outputs) {
+		const ScopeLock protect(ao->mutex);
+		if (ao->IsOpen() && !ao->IsCommandFinished())
+			return false;
+	}
+
+	return true;
+}
+
+void
+MultipleOutputs::WaitAll()
+{
+	while (!AllFinished())
+		audio_output_client_notify.Wait();
+}
+
+void
+MultipleOutputs::AllowPlay()
+{
+	for (auto ao : outputs)
+		ao->LockAllowPlay();
+}
+
+static void
+audio_output_reset_reopen(AudioOutput *ao)
+{
+	const ScopeLock protect(ao->mutex);
+
+	ao->fail_timer.Reset();
+}
+
+void
+MultipleOutputs::ResetReopen()
+{
+	for (auto ao : outputs)
+		audio_output_reset_reopen(ao);
+}
+
+bool
+MultipleOutputs::Update()
+{
+	bool ret = false;
+
+	if (!input_audio_format.IsDefined())
+		return false;
+
+	for (auto ao : outputs)
+		ret = ao->LockUpdate(input_audio_format, *pipe)
+			|| ret;
+
+	return ret;
+}
+
+void
+MultipleOutputs::SetReplayGainMode(ReplayGainMode mode)
+{
+	for (auto ao : outputs)
+		ao->SetReplayGainMode(mode);
+}
+
+bool
+MultipleOutputs::Play(MusicChunk *chunk, Error &error)
+{
+	assert(buffer != nullptr);
+	assert(pipe != nullptr);
+	assert(chunk != nullptr);
+	assert(chunk->CheckFormat(input_audio_format));
+
+	if (!Update()) {
+		/* TODO: obtain real error */
+		error.Set(output_domain, "Failed to open audio output");
+		return false;
+	}
+
+	pipe->Push(chunk);
+
+	for (auto ao : outputs)
+		ao->LockPlay();
+
+	return true;
+}
+
+bool
+MultipleOutputs::Open(const AudioFormat audio_format,
+		      MusicBuffer &_buffer,
+		      Error &error)
+{
+	bool ret = false, enabled = false;
+
+	assert(buffer == nullptr || buffer == &_buffer);
+	assert((pipe == nullptr) == (buffer == nullptr));
+
+	buffer = &_buffer;
+
+	/* the audio format must be the same as existing chunks in the
+	   pipe */
+	assert(pipe == nullptr || pipe->CheckFormat(audio_format));
+
+	if (pipe == nullptr)
+		pipe = new MusicPipe();
+	else
+		/* if the pipe hasn't been cleared, the the audio
+		   format must not have changed */
+		assert(pipe->IsEmpty() || audio_format == input_audio_format);
+
+	input_audio_format = audio_format;
+
+	ResetReopen();
+	EnableDisable();
+	Update();
+
+	for (auto ao : outputs) {
+		if (ao->enabled)
+			enabled = true;
+
+		if (ao->open)
+			ret = true;
+	}
+
+	if (!enabled)
+		error.Set(output_domain, "All audio outputs are disabled");
+	else if (!ret)
+		/* TODO: obtain real error */
+		error.Set(output_domain, "Failed to open audio output");
+
+	if (!ret)
+		/* close all devices if there was an error */
+		Close();
+
+	return ret;
+}
+
+/**
+ * Has the specified audio output already consumed this chunk?
+ */
+gcc_pure
+static bool
+chunk_is_consumed_in(const AudioOutput *ao,
+		     gcc_unused const MusicPipe *pipe,
+		     const MusicChunk *chunk)
+{
+	if (!ao->open)
+		return true;
+
+	if (ao->current_chunk == nullptr)
+		return false;
+
+	assert(chunk == ao->current_chunk ||
+	       pipe->Contains(ao->current_chunk));
+
+	if (chunk != ao->current_chunk) {
+		assert(chunk->next != nullptr);
+		return true;
+	}
+
+	return ao->current_chunk_finished && chunk->next == nullptr;
+}
+
+bool
+MultipleOutputs::IsChunkConsumed(const MusicChunk *chunk) const
+{
+	for (auto ao : outputs) {
+		const ScopeLock protect(ao->mutex);
+		if (!chunk_is_consumed_in(ao, pipe, chunk))
+			return false;
+	}
+
+	return true;
+}
+
+inline void
+MultipleOutputs::ClearTailChunk(gcc_unused const MusicChunk *chunk,
+				bool *locked)
+{
+	assert(chunk->next == nullptr);
+	assert(pipe->Contains(chunk));
+
+	for (unsigned i = 0, n = outputs.size(); i != n; ++i) {
+		AudioOutput *ao = outputs[i];
+
+		/* this mutex will be unlocked by the caller when it's
+		   ready */
+		ao->mutex.lock();
+		locked[i] = ao->open;
+
+		if (!locked[i]) {
+			ao->mutex.unlock();
+			continue;
+		}
+
+		assert(ao->current_chunk == chunk);
+		assert(ao->current_chunk_finished);
+		ao->current_chunk = nullptr;
+	}
+}
+
+unsigned
+MultipleOutputs::Check()
+{
+	const MusicChunk *chunk;
+	bool is_tail;
+	MusicChunk *shifted;
+	bool locked[outputs.size()];
+
+	assert(buffer != nullptr);
+	assert(pipe != nullptr);
+
+	while ((chunk = pipe->Peek()) != nullptr) {
+		assert(!pipe->IsEmpty());
+
+		if (!IsChunkConsumed(chunk))
+			/* at least one output is not finished playing
+			   this chunk */
+			return pipe->GetSize();
+
+		if (chunk->length > 0 && !chunk->time.IsNegative())
+			/* only update elapsed_time if the chunk
+			   provides a defined value */
+			elapsed_time = chunk->time;
+
+		is_tail = chunk->next == nullptr;
+		if (is_tail)
+			/* this is the tail of the pipe - clear the
+			   chunk reference in all outputs */
+			ClearTailChunk(chunk, locked);
+
+		/* remove the chunk from the pipe */
+		shifted = pipe->Shift();
+		assert(shifted == chunk);
+
+		if (is_tail)
+			/* unlock all audio outputs which were locked
+			   by clear_tail_chunk() */
+			for (unsigned i = 0, n = outputs.size(); i != n; ++i)
+				if (locked[i])
+					outputs[i]->mutex.unlock();
+
+		/* return the chunk to the buffer */
+		buffer->Return(shifted);
+	}
+
+	return 0;
+}
+
+bool
+MultipleOutputs::Wait(PlayerControl &pc, unsigned threshold)
+{
+	pc.Lock();
+
+	if (Check() < threshold) {
+		pc.Unlock();
+		return true;
+	}
+
+	pc.Wait();
+	pc.Unlock();
+
+	return Check() < threshold;
+}
+
+void
+MultipleOutputs::Pause()
+{
+	Update();
+
+	for (auto ao : outputs)
+		ao->LockPauseAsync();
+
+	WaitAll();
+}
+
+void
+MultipleOutputs::Drain()
+{
+	for (auto ao : outputs)
+		ao->LockDrainAsync();
+
+	WaitAll();
+}
+
+void
+MultipleOutputs::Cancel()
+{
+	/* send the cancel() command to all audio outputs */
+
+	for (auto ao : outputs)
+		ao->LockCancelAsync();
+
+	WaitAll();
+
+	/* clear the music pipe and return all chunks to the buffer */
+
+	if (pipe != nullptr)
+		pipe->Clear(*buffer);
+
+	/* the audio outputs are now waiting for a signal, to
+	   synchronize the cleared music pipe */
+
+	AllowPlay();
+
+	/* invalidate elapsed_time */
+
+	elapsed_time = SignedSongTime::Negative();
+}
+
+void
+MultipleOutputs::Close()
+{
+	for (auto ao : outputs)
+		ao->LockCloseWait();
+
+	if (pipe != nullptr) {
+		assert(buffer != nullptr);
+
+		pipe->Clear(*buffer);
+		delete pipe;
+		pipe = nullptr;
+	}
+
+	buffer = nullptr;
+
+	input_audio_format.Clear();
+
+	elapsed_time = SignedSongTime::Negative();
+}
+
+void
+MultipleOutputs::Release()
+{
+	for (auto ao : outputs)
+		ao->LockRelease();
+
+	if (pipe != nullptr) {
+		assert(buffer != nullptr);
+
+		pipe->Clear(*buffer);
+		delete pipe;
+		pipe = nullptr;
+	}
+
+	buffer = nullptr;
+
+	input_audio_format.Clear();
+
+	elapsed_time = SignedSongTime::Negative();
+}
+
+void
+MultipleOutputs::SongBorder()
+{
+	/* clear the elapsed_time pointer at the beginning of a new
+	   song */
+	elapsed_time = SignedSongTime::zero();
+}
diff --git a/src/output/MultipleOutputs.hxx b/src/output/MultipleOutputs.hxx
new file mode 100644
index 000000000..2c6536e2a
--- /dev/null
+++ b/src/output/MultipleOutputs.hxx
@@ -0,0 +1,276 @@
+/*
+ * 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.
+ */
+
+/*
+ * Functions for dealing with all configured (enabled) audion outputs
+ * at once.
+ *
+ */
+
+#ifndef OUTPUT_ALL_H
+#define OUTPUT_ALL_H
+
+#include "AudioFormat.hxx"
+#include "ReplayGainInfo.hxx"
+#include "Chrono.hxx"
+#include "Compiler.h"
+
+#include <vector>
+
+#include <assert.h>
+
+struct AudioFormat;
+class MusicBuffer;
+class MusicPipe;
+class EventLoop;
+class MixerListener;
+struct MusicChunk;
+struct PlayerControl;
+struct AudioOutput;
+class Error;
+
+class MultipleOutputs {
+	MixerListener &mixer_listener;
+
+	std::vector<AudioOutput *> outputs;
+
+	AudioFormat input_audio_format;
+
+	/**
+	 * The #MusicBuffer object where consumed chunks are returned.
+	 */
+	MusicBuffer *buffer;
+
+	/**
+	 * The #MusicPipe object which feeds all audio outputs.  It is
+	 * filled by audio_output_all_play().
+	 */
+	MusicPipe *pipe;
+
+	/**
+	 * The "elapsed_time" stamp of the most recently finished
+	 * chunk.
+	 */
+	SignedSongTime elapsed_time;
+
+public:
+	/**
+	 * Load audio outputs from the configuration file and
+	 * initialize them.
+	 */
+	MultipleOutputs(MixerListener &_mixer_listener);
+	~MultipleOutputs();
+
+	void Configure(EventLoop &event_loop, PlayerControl &pc);
+
+	/**
+	 * Returns the total number of audio output devices, including
+	 * those which are disabled right now.
+	 */
+	gcc_pure
+	unsigned Size() const {
+		return outputs.size();
+	}
+
+	/**
+	 * Returns the "i"th audio output device.
+	 */
+	const AudioOutput &Get(unsigned i) const {
+		assert(i < Size());
+
+		return *outputs[i];
+	}
+
+	AudioOutput &Get(unsigned i) {
+		assert(i < Size());
+
+		return *outputs[i];
+	}
+
+	/**
+	 * Returns the audio output device with the specified name.
+	 * Returns nullptr if the name does not exist.
+	 */
+	gcc_pure
+	AudioOutput *FindByName(const char *name) const;
+
+	/**
+	 * Checks the "enabled" flag of all audio outputs, and if one has
+	 * changed, commit the change.
+	 */
+	void EnableDisable();
+
+	/**
+	 * Opens all audio outputs which are not disabled.
+	 *
+	 * @param audio_format the preferred audio format
+	 * @param buffer the #music_buffer where consumed #MusicChunk objects
+	 * should be returned
+	 * @return true on success, false on failure
+	 */
+	bool Open(const AudioFormat audio_format, MusicBuffer &_buffer,
+		  Error &error);
+
+	/**
+	 * Closes all audio outputs.
+	 */
+	void Close();
+
+	/**
+	 * Closes all audio outputs.  Outputs with the "always_on"
+	 * flag are put into pause mode.
+	 */
+	void Release();
+
+	void SetReplayGainMode(ReplayGainMode mode);
+
+	/**
+	 * Enqueue a #MusicChunk object for playing, i.e. pushes it to a
+	 * #MusicPipe.
+	 *
+	 * @param chunk the #MusicChunk object to be played
+	 * @return true on success, false if no audio output was able to play
+	 * (all closed then)
+	 */
+	bool Play(MusicChunk *chunk, Error &error);
+
+	/**
+	 * Checks if the output devices have drained their music pipe, and
+	 * returns the consumed music chunks to the #music_buffer.
+	 *
+	 * @return the number of chunks to play left in the #MusicPipe
+	 */
+	unsigned Check();
+
+	/**
+	 * Checks if the size of the #MusicPipe is below the #threshold.  If
+	 * not, it attempts to synchronize with all output threads, and waits
+	 * until another #MusicChunk is finished.
+	 *
+	 * @param threshold the maximum number of chunks in the pipe
+	 * @return true if there are less than #threshold chunks in the pipe
+	 */
+	bool Wait(PlayerControl &pc, unsigned threshold);
+
+	/**
+	 * Puts all audio outputs into pause mode.  Most implementations will
+	 * simply close it then.
+	 */
+	void Pause();
+
+	/**
+	 * Drain all audio outputs.
+	 */
+	void Drain();
+
+	/**
+	 * Try to cancel data which may still be in the device's buffers.
+	 */
+	void Cancel();
+
+	/**
+	 * Indicate that a new song will begin now.
+	 */
+	void SongBorder();
+
+	/**
+	 * Returns the "elapsed_time" stamp of the most recently finished
+	 * chunk.  A negative value is returned when no chunk has been
+	 * finished yet.
+	 */
+	gcc_pure
+	SignedSongTime GetElapsedTime() const {
+		return elapsed_time;
+	}
+
+	/**
+	 * Returns the average volume of all available mixers (range
+	 * 0..100).  Returns -1 if no mixer can be queried.
+	 */
+	gcc_pure
+	int GetVolume() const;
+
+	/**
+	 * Sets the volume on all available mixers.
+	 *
+	 * @param volume the volume (range 0..100)
+	 * @return true on success, false on failure
+	 */
+	bool SetVolume(unsigned volume);
+
+	/**
+	 * Similar to GetVolume(), but gets the volume only for
+	 * software mixers.  See #software_mixer_plugin.  This
+	 * function fails if no software mixer is configured.
+	 */
+	gcc_pure
+	int GetSoftwareVolume() const;
+
+	/**
+	 * Similar to SetVolume(), but sets the volume only for
+	 * software mixers.  See #software_mixer_plugin.  This
+	 * function cannot fail, because the underlying software
+	 * mixers cannot fail either.
+	 */
+	void SetSoftwareVolume(unsigned volume);
+
+private:
+	/**
+	 * Determine if all (active) outputs have finished the current
+	 * command.
+	 */
+	gcc_pure
+	bool AllFinished() const;
+
+	void WaitAll();
+
+	/**
+	 * Signals all audio outputs which are open.
+	 */
+	void AllowPlay();
+
+	/**
+	 * Resets the "reopen" flag on all audio devices.  MPD should
+	 * immediately retry to open the device instead of waiting for
+	 * the timeout when the user wants to start playback.
+	 */
+	void ResetReopen();
+
+	/**
+	 * Opens all output devices which are enabled, but closed.
+	 *
+	 * @return true if there is at least open output device which
+	 * is open
+	 */
+	bool Update();
+
+	/**
+	 * Has this chunk been consumed by all audio outputs?
+	 */
+	bool IsChunkConsumed(const MusicChunk *chunk) const;
+
+	/**
+	 * There's only one chunk left in the pipe (#pipe), and all
+	 * audio outputs have consumed it already.  Clear the
+	 * reference.
+	 */
+	void ClearTailChunk(const MusicChunk *chunk, bool *locked);
+};
+
+#endif
diff --git a/src/output/OutputAPI.hxx b/src/output/OutputAPI.hxx
new file mode 100644
index 000000000..e0fd6eec8
--- /dev/null
+++ b/src/output/OutputAPI.hxx
@@ -0,0 +1,33 @@
+/*
+ * 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_OUTPUT_API_HXX
+#define MPD_OUTPUT_API_HXX
+
+// IWYU pragma: begin_exports
+
+#include "OutputPlugin.hxx"
+#include "Internal.hxx"
+#include "AudioFormat.hxx"
+#include "tag/Tag.hxx"
+#include "config/ConfigData.hxx"
+
+// IWYU pragma: end_exports
+
+#endif
diff --git a/src/output/OutputCommand.cxx b/src/output/OutputCommand.cxx
new file mode 100644
index 000000000..6afb70cf1
--- /dev/null
+++ b/src/output/OutputCommand.cxx
@@ -0,0 +1,106 @@
+/*
+ * 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.
+ */
+
+/*
+ * Glue functions for controlling the audio outputs over the MPD
+ * protocol.  These functions perform extra validation on all
+ * parameters, because they might be from an untrusted source.
+ *
+ */
+
+#include "config.h"
+#include "OutputCommand.hxx"
+#include "MultipleOutputs.hxx"
+#include "Internal.hxx"
+#include "PlayerControl.hxx"
+#include "mixer/MixerControl.hxx"
+#include "Idle.hxx"
+
+extern unsigned audio_output_state_version;
+
+bool
+audio_output_enable_index(MultipleOutputs &outputs, unsigned idx)
+{
+	if (idx >= outputs.Size())
+		return false;
+
+	AudioOutput &ao = outputs.Get(idx);
+	if (ao.enabled)
+		return true;
+
+	ao.enabled = true;
+	idle_add(IDLE_OUTPUT);
+
+	ao.player_control->UpdateAudio();
+
+	++audio_output_state_version;
+
+	return true;
+}
+
+bool
+audio_output_disable_index(MultipleOutputs &outputs, unsigned idx)
+{
+	if (idx >= outputs.Size())
+		return false;
+
+	AudioOutput &ao = outputs.Get(idx);
+	if (!ao.enabled)
+		return true;
+
+	ao.enabled = false;
+	idle_add(IDLE_OUTPUT);
+
+	Mixer *mixer = ao.mixer;
+	if (mixer != nullptr) {
+		mixer_close(mixer);
+		idle_add(IDLE_MIXER);
+	}
+
+	ao.player_control->UpdateAudio();
+
+	++audio_output_state_version;
+
+	return true;
+}
+
+bool
+audio_output_toggle_index(MultipleOutputs &outputs, unsigned idx)
+{
+	if (idx >= outputs.Size())
+		return false;
+
+	AudioOutput &ao = outputs.Get(idx);
+	const bool enabled = ao.enabled = !ao.enabled;
+	idle_add(IDLE_OUTPUT);
+
+	if (!enabled) {
+		Mixer *mixer = ao.mixer;
+		if (mixer != nullptr) {
+			mixer_close(mixer);
+			idle_add(IDLE_MIXER);
+		}
+	}
+
+	ao.player_control->UpdateAudio();
+
+	++audio_output_state_version;
+
+	return true;
+}
diff --git a/src/output/OutputCommand.hxx b/src/output/OutputCommand.hxx
new file mode 100644
index 000000000..53fc5c95e
--- /dev/null
+++ b/src/output/OutputCommand.hxx
@@ -0,0 +1,53 @@
+/*
+ * 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.
+ */
+
+/*
+ * Glue functions for controlling the audio outputs over the MPD
+ * protocol.  These functions perform extra validation on all
+ * parameters, because they might be from an untrusted source.
+ *
+ */
+
+#ifndef MPD_OUTPUT_COMMAND_HXX
+#define MPD_OUTPUT_COMMAND_HXX
+
+class MultipleOutputs;
+
+/**
+ * Enables an audio output.  Returns false if the specified output
+ * does not exist.
+ */
+bool
+audio_output_enable_index(MultipleOutputs &outputs, unsigned idx);
+
+/**
+ * Disables an audio output.  Returns false if the specified output
+ * does not exist.
+ */
+bool
+audio_output_disable_index(MultipleOutputs &outputs, unsigned idx);
+
+/**
+ * Toggles an audio output.  Returns false if the specified output
+ * does not exist.
+ */
+bool
+audio_output_toggle_index(MultipleOutputs &outputs, unsigned idx);
+
+#endif
diff --git a/src/output/OutputControl.cxx b/src/output/OutputControl.cxx
new file mode 100644
index 000000000..89428fa87
--- /dev/null
+++ b/src/output/OutputControl.cxx
@@ -0,0 +1,295 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "Internal.hxx"
+#include "OutputPlugin.hxx"
+#include "Domain.hxx"
+#include "mixer/MixerControl.hxx"
+#include "notify.hxx"
+#include "filter/plugins/ReplayGainFilterPlugin.hxx"
+#include "util/Error.hxx"
+#include "Log.hxx"
+
+#include <assert.h>
+
+/** after a failure, wait this number of seconds before
+    automatically reopening the device */
+static constexpr unsigned REOPEN_AFTER = 10;
+
+struct notify audio_output_client_notify;
+
+void
+AudioOutput::WaitForCommand()
+{
+	while (!IsCommandFinished()) {
+		mutex.unlock();
+		audio_output_client_notify.Wait();
+		mutex.lock();
+	}
+}
+
+void
+AudioOutput::CommandAsync(audio_output_command cmd)
+{
+	assert(IsCommandFinished());
+
+	command = cmd;
+	cond.signal();
+}
+
+void
+AudioOutput::CommandWait(audio_output_command cmd)
+{
+	CommandAsync(cmd);
+	WaitForCommand();
+}
+
+void
+AudioOutput::LockCommandWait(audio_output_command cmd)
+{
+	const ScopeLock protect(mutex);
+	CommandWait(cmd);
+}
+
+void
+AudioOutput::SetReplayGainMode(ReplayGainMode mode)
+{
+	if (replay_gain_filter != nullptr)
+		replay_gain_filter_set_mode(replay_gain_filter, mode);
+	if (other_replay_gain_filter != nullptr)
+		replay_gain_filter_set_mode(other_replay_gain_filter, mode);
+}
+
+void
+AudioOutput::LockEnableWait()
+{
+	if (!thread.IsDefined()) {
+		if (plugin.enable == nullptr) {
+			/* don't bother to start the thread now if the
+			   device doesn't even have a enable() method;
+			   just assign the variable and we're done */
+			really_enabled = true;
+			return;
+		}
+
+		StartThread();
+	}
+
+	LockCommandWait(AO_COMMAND_ENABLE);
+}
+
+void
+AudioOutput::LockDisableWait()
+{
+	if (!thread.IsDefined()) {
+		if (plugin.disable == nullptr)
+			really_enabled = false;
+		else
+			/* if there's no thread yet, the device cannot
+			   be enabled */
+			assert(!really_enabled);
+
+		return;
+	}
+
+	LockCommandWait(AO_COMMAND_DISABLE);
+}
+
+inline bool
+AudioOutput::Open(const AudioFormat audio_format, const MusicPipe &mp)
+{
+	assert(allow_play);
+	assert(audio_format.IsValid());
+
+	fail_timer.Reset();
+
+	if (open && audio_format == in_audio_format) {
+		assert(pipe == &mp || (always_on && pause));
+
+		if (pause) {
+			current_chunk = nullptr;
+			pipe = &mp;
+
+			/* unpause with the CANCEL command; this is a
+			   hack, but suits well for forcing the thread
+			   to leave the ao_pause() thread, and we need
+			   to flush the device buffer anyway */
+
+			/* we're not using audio_output_cancel() here,
+			   because that function is asynchronous */
+			CommandWait(AO_COMMAND_CANCEL);
+		}
+
+		return true;
+	}
+
+	in_audio_format = audio_format;
+	current_chunk = nullptr;
+
+	pipe = &mp;
+
+	if (!thread.IsDefined())
+		StartThread();
+
+	CommandWait(open ? AO_COMMAND_REOPEN : AO_COMMAND_OPEN);
+	const bool open2 = open;
+
+	if (open2 && mixer != nullptr) {
+		Error error;
+		if (!mixer_open(mixer, error))
+			FormatWarning(output_domain,
+				      "Failed to open mixer for '%s'", name);
+	}
+
+	return open2;
+}
+
+void
+AudioOutput::CloseWait()
+{
+	assert(allow_play);
+
+	if (mixer != nullptr)
+		mixer_auto_close(mixer);
+
+	assert(!open || !fail_timer.IsDefined());
+
+	if (open)
+		CommandWait(AO_COMMAND_CLOSE);
+	else
+		fail_timer.Reset();
+}
+
+bool
+AudioOutput::LockUpdate(const AudioFormat audio_format,
+			const MusicPipe &mp)
+{
+	const ScopeLock protect(mutex);
+
+	if (enabled && really_enabled) {
+		if (fail_timer.Check(REOPEN_AFTER * 1000)) {
+			return Open(audio_format, mp);
+		}
+	} else if (IsOpen())
+		CloseWait();
+
+	return false;
+}
+
+void
+AudioOutput::LockPlay()
+{
+	const ScopeLock protect(mutex);
+
+	assert(allow_play);
+
+	if (IsOpen() && !in_playback_loop && !woken_for_play) {
+		woken_for_play = true;
+		cond.signal();
+	}
+}
+
+void
+AudioOutput::LockPauseAsync()
+{
+	if (mixer != nullptr && plugin.pause == nullptr)
+		/* the device has no pause mode: close the mixer,
+		   unless its "global" flag is set (checked by
+		   mixer_auto_close()) */
+		mixer_auto_close(mixer);
+
+	const ScopeLock protect(mutex);
+
+	assert(allow_play);
+	if (IsOpen())
+		CommandAsync(AO_COMMAND_PAUSE);
+}
+
+void
+AudioOutput::LockDrainAsync()
+{
+	const ScopeLock protect(mutex);
+
+	assert(allow_play);
+	if (IsOpen())
+		CommandAsync(AO_COMMAND_DRAIN);
+}
+
+void
+AudioOutput::LockCancelAsync()
+{
+	const ScopeLock protect(mutex);
+
+	if (IsOpen()) {
+		allow_play = false;
+		CommandAsync(AO_COMMAND_CANCEL);
+	}
+}
+
+void
+AudioOutput::LockAllowPlay()
+{
+	const ScopeLock protect(mutex);
+
+	allow_play = true;
+	if (IsOpen())
+		cond.signal();
+}
+
+void
+AudioOutput::LockRelease()
+{
+	if (always_on)
+		LockPauseAsync();
+	else
+		LockCloseWait();
+}
+
+void
+AudioOutput::LockCloseWait()
+{
+	assert(!open || !fail_timer.IsDefined());
+
+	const ScopeLock protect(mutex);
+	CloseWait();
+}
+
+void
+AudioOutput::StopThread()
+{
+	assert(thread.IsDefined());
+	assert(allow_play);
+
+	LockCommandWait(AO_COMMAND_KILL);
+	thread.Join();
+}
+
+void
+AudioOutput::Finish()
+{
+	LockCloseWait();
+
+	assert(!fail_timer.IsDefined());
+
+	if (thread.IsDefined())
+		StopThread();
+
+	audio_output_free(this);
+}
diff --git a/src/output/OutputControl.hxx b/src/output/OutputControl.hxx
new file mode 100644
index 000000000..fff3fe406
--- /dev/null
+++ b/src/output/OutputControl.hxx
@@ -0,0 +1,25 @@
+/*
+ * 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_OUTPUT_CONTROL_HXX
+#define MPD_OUTPUT_CONTROL_HXX
+
+struct AudioOutput;
+
+#endif
diff --git a/src/output/OutputPlugin.cxx b/src/output/OutputPlugin.cxx
new file mode 100644
index 000000000..33bb854d4
--- /dev/null
+++ b/src/output/OutputPlugin.cxx
@@ -0,0 +1,109 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "OutputPlugin.hxx"
+#include "Internal.hxx"
+
+AudioOutput *
+ao_plugin_init(const AudioOutputPlugin *plugin,
+	       const config_param &param,
+	       Error &error)
+{
+	assert(plugin != nullptr);
+	assert(plugin->init != nullptr);
+
+	return plugin->init(param, error);
+}
+
+void
+ao_plugin_finish(AudioOutput *ao)
+{
+	ao->plugin.finish(ao);
+}
+
+bool
+ao_plugin_enable(AudioOutput *ao, Error &error_r)
+{
+	return ao->plugin.enable != nullptr
+		? ao->plugin.enable(ao, error_r)
+		: true;
+}
+
+void
+ao_plugin_disable(AudioOutput *ao)
+{
+	if (ao->plugin.disable != nullptr)
+		ao->plugin.disable(ao);
+}
+
+bool
+ao_plugin_open(AudioOutput *ao, AudioFormat &audio_format,
+	       Error &error)
+{
+	return ao->plugin.open(ao, audio_format, error);
+}
+
+void
+ao_plugin_close(AudioOutput *ao)
+{
+	ao->plugin.close(ao);
+}
+
+unsigned
+ao_plugin_delay(AudioOutput *ao)
+{
+	return ao->plugin.delay != nullptr
+		? ao->plugin.delay(ao)
+		: 0;
+}
+
+void
+ao_plugin_send_tag(AudioOutput *ao, const Tag *tag)
+{
+	if (ao->plugin.send_tag != nullptr)
+		ao->plugin.send_tag(ao, tag);
+}
+
+size_t
+ao_plugin_play(AudioOutput *ao, const void *chunk, size_t size,
+	       Error &error)
+{
+	return ao->plugin.play(ao, chunk, size, error);
+}
+
+void
+ao_plugin_drain(AudioOutput *ao)
+{
+	if (ao->plugin.drain != nullptr)
+		ao->plugin.drain(ao);
+}
+
+void
+ao_plugin_cancel(AudioOutput *ao)
+{
+	if (ao->plugin.cancel != nullptr)
+		ao->plugin.cancel(ao);
+}
+
+bool
+ao_plugin_pause(AudioOutput *ao)
+{
+	return ao->plugin.pause != nullptr && ao->plugin.pause(ao);
+}
diff --git a/src/output/OutputPlugin.hxx b/src/output/OutputPlugin.hxx
new file mode 100644
index 000000000..00fa36bc0
--- /dev/null
+++ b/src/output/OutputPlugin.hxx
@@ -0,0 +1,204 @@
+/*
+ * 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_OUTPUT_PLUGIN_HXX
+#define MPD_OUTPUT_PLUGIN_HXX
+
+#include "Compiler.h"
+
+#include <stddef.h>
+
+struct config_param;
+struct AudioFormat;
+struct Tag;
+struct AudioOutput;
+struct MixerPlugin;
+class Error;
+
+/**
+ * A plugin which controls an audio output device.
+ */
+struct AudioOutputPlugin {
+	/**
+	 * the plugin's name
+	 */
+	const char *name;
+
+	/**
+	 * Test if this plugin can provide a default output, in case
+	 * none has been configured.  This method is optional.
+	 */
+	bool (*test_default_device)(void);
+
+	/**
+	 * Configure and initialize the device, but do not open it
+	 * yet.
+	 *
+	 * @param param the configuration section, or nullptr if there is
+	 * no configuration
+	 * @return nullptr on error, or an opaque pointer to the plugin's
+	 * data
+	 */
+	AudioOutput *(*init)(const config_param &param,
+				     Error &error);
+
+	/**
+	 * Free resources allocated by this device.
+	 */
+	void (*finish)(AudioOutput *data);
+
+	/**
+	 * Enable the device.  This may allocate resources, preparing
+	 * for the device to be opened.  Enabling a device cannot
+	 * fail: if an error occurs during that, it should be reported
+	 * by the open() method.
+	 *
+	 * @return true on success, false on error
+	 */
+	bool (*enable)(AudioOutput *data, Error &error);
+
+	/**
+	 * Disables the device.  It is closed before this method is
+	 * called.
+	 */
+	void (*disable)(AudioOutput *data);
+
+	/**
+	 * Really open the device.
+	 *
+	 * @param audio_format the audio format in which data is going
+	 * to be delivered; may be modified by the plugin
+	 */
+	bool (*open)(AudioOutput *data, AudioFormat &audio_format,
+		     Error &error);
+
+	/**
+	 * Close the device.
+	 */
+	void (*close)(AudioOutput *data);
+
+	/**
+	 * Returns a positive number if the output thread shall delay
+	 * the next call to play() or pause().  This should be
+	 * implemented instead of doing a sleep inside the plugin,
+	 * because this allows MPD to listen to commands meanwhile.
+	 *
+	 * @return the number of milliseconds to wait
+	 */
+	unsigned (*delay)(AudioOutput *data);
+
+	/**
+	 * Display metadata for the next chunk.  Optional method,
+	 * because not all devices can display metadata.
+	 */
+	void (*send_tag)(AudioOutput *data, const Tag *tag);
+
+	/**
+	 * Play a chunk of audio data.
+	 *
+	 * @return the number of bytes played, or 0 on error
+	 */
+	size_t (*play)(AudioOutput *data,
+		       const void *chunk, size_t size,
+		       Error &error);
+
+	/**
+	 * Wait until the device has finished playing.
+	 */
+	void (*drain)(AudioOutput *data);
+
+	/**
+	 * Try to cancel data which may still be in the device's
+	 * buffers.
+	 */
+	void (*cancel)(AudioOutput *data);
+
+	/**
+	 * Pause the device.  If supported, it may perform a special
+	 * action, which keeps the device open, but does not play
+	 * anything.  Output plugins like "shout" might want to play
+	 * silence during pause, so their clients won't be
+	 * disconnected.  Plugins which do not support pausing will
+	 * simply be closed, and have to be reopened when unpaused.
+	 *
+	 * @return false on error (output will be closed then), true
+	 * for continue to pause
+	 */
+	bool (*pause)(AudioOutput *data);
+
+	/**
+	 * The mixer plugin associated with this output plugin.  This
+	 * may be nullptr if no mixer plugin is implemented.  When
+	 * created, this mixer plugin gets the same #config_param as
+	 * this audio output device.
+	 */
+	const MixerPlugin *mixer_plugin;
+};
+
+static inline bool
+ao_plugin_test_default_device(const AudioOutputPlugin *plugin)
+{
+	return plugin->test_default_device != nullptr
+		? plugin->test_default_device()
+		: false;
+}
+
+gcc_malloc
+AudioOutput *
+ao_plugin_init(const AudioOutputPlugin *plugin,
+	       const config_param &param,
+	       Error &error);
+
+void
+ao_plugin_finish(AudioOutput *ao);
+
+bool
+ao_plugin_enable(AudioOutput *ao, Error &error);
+
+void
+ao_plugin_disable(AudioOutput *ao);
+
+bool
+ao_plugin_open(AudioOutput *ao, AudioFormat &audio_format,
+	       Error &error);
+
+void
+ao_plugin_close(AudioOutput *ao);
+
+gcc_pure
+unsigned
+ao_plugin_delay(AudioOutput *ao);
+
+void
+ao_plugin_send_tag(AudioOutput *ao, const Tag *tag);
+
+size_t
+ao_plugin_play(AudioOutput *ao, const void *chunk, size_t size,
+	       Error &error);
+
+void
+ao_plugin_drain(AudioOutput *ao);
+
+void
+ao_plugin_cancel(AudioOutput *ao);
+
+bool
+ao_plugin_pause(AudioOutput *ao);
+
+#endif
diff --git a/src/output/OutputPrint.cxx b/src/output/OutputPrint.cxx
new file mode 100644
index 000000000..414a86e32
--- /dev/null
+++ b/src/output/OutputPrint.cxx
@@ -0,0 +1,43 @@
+/*
+ * 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.
+ */
+
+/*
+ * Protocol specific code for the audio output library.
+ *
+ */
+
+#include "config.h"
+#include "OutputPrint.hxx"
+#include "MultipleOutputs.hxx"
+#include "Internal.hxx"
+#include "client/Client.hxx"
+
+void
+printAudioDevices(Client &client, const MultipleOutputs &outputs)
+{
+	for (unsigned i = 0, n = outputs.Size(); i != n; ++i) {
+		const AudioOutput &ao = outputs.Get(i);
+
+		client_printf(client,
+			      "outputid: %i\n"
+			      "outputname: %s\n"
+			      "outputenabled: %i\n",
+			      i, ao.name, ao.enabled);
+	}
+}
diff --git a/src/output/OutputPrint.hxx b/src/output/OutputPrint.hxx
new file mode 100644
index 000000000..29aa2b11c
--- /dev/null
+++ b/src/output/OutputPrint.hxx
@@ -0,0 +1,34 @@
+/*
+ * 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.
+ */
+
+/*
+ * Protocol specific code for the audio output library.
+ *
+ */
+
+#ifndef MPD_OUTPUT_PRINT_HXX
+#define MPD_OUTPUT_PRINT_HXX
+
+class Client;
+class MultipleOutputs;
+
+void
+printAudioDevices(Client &client, const MultipleOutputs &outputs);
+
+#endif
diff --git a/src/output/OutputState.cxx b/src/output/OutputState.cxx
new file mode 100644
index 000000000..fb01b1c65
--- /dev/null
+++ b/src/output/OutputState.cxx
@@ -0,0 +1,88 @@
+/*
+ * 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.
+ */
+
+/*
+ * Saving and loading the audio output states to/from the state file.
+ *
+ */
+
+#include "config.h"
+#include "OutputState.hxx"
+#include "MultipleOutputs.hxx"
+#include "Internal.hxx"
+#include "Domain.hxx"
+#include "Log.hxx"
+#include "fs/io/BufferedOutputStream.hxx"
+#include "util/StringUtil.hxx"
+
+#include <assert.h>
+#include <stdlib.h>
+
+#define AUDIO_DEVICE_STATE "audio_device_state:"
+
+unsigned audio_output_state_version;
+
+void
+audio_output_state_save(BufferedOutputStream &os,
+			const MultipleOutputs &outputs)
+{
+	for (unsigned i = 0, n = outputs.Size(); i != n; ++i) {
+		const AudioOutput &ao = outputs.Get(i);
+
+		os.Format(AUDIO_DEVICE_STATE "%d:%s\n", ao.enabled, ao.name);
+	}
+}
+
+bool
+audio_output_state_read(const char *line, MultipleOutputs &outputs)
+{
+	long value;
+	char *endptr;
+	const char *name;
+
+	if (!StringStartsWith(line, AUDIO_DEVICE_STATE))
+		return false;
+
+	line += sizeof(AUDIO_DEVICE_STATE) - 1;
+
+	value = strtol(line, &endptr, 10);
+	if (*endptr != ':' || (value != 0 && value != 1))
+		return false;
+
+	if (value != 0)
+		/* state is "enabled": no-op */
+		return true;
+
+	name = endptr + 1;
+	AudioOutput *ao = outputs.FindByName(name);
+	if (ao == NULL) {
+		FormatDebug(output_domain,
+			    "Ignoring device state for '%s'", name);
+		return true;
+	}
+
+	ao->enabled = false;
+	return true;
+}
+
+unsigned
+audio_output_state_get_version(void)
+{
+	return audio_output_state_version;
+}
diff --git a/src/output/OutputState.hxx b/src/output/OutputState.hxx
new file mode 100644
index 000000000..47f8429d5
--- /dev/null
+++ b/src/output/OutputState.hxx
@@ -0,0 +1,46 @@
+/*
+ * 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.
+ */
+
+/*
+ * Saving and loading the audio output states to/from the state file.
+ *
+ */
+
+#ifndef MPD_OUTPUT_STATE_HXX
+#define MPD_OUTPUT_STATE_HXX
+
+class MultipleOutputs;
+class BufferedOutputStream;
+
+bool
+audio_output_state_read(const char *line, MultipleOutputs &outputs);
+
+void
+audio_output_state_save(BufferedOutputStream &os,
+			const MultipleOutputs &outputs);
+
+/**
+ * Generates a version number for the current state of the audio
+ * outputs.  This is used by timer_save_state_file() to determine
+ * whether the state has changed and the state file should be saved.
+ */
+unsigned
+audio_output_state_get_version(void);
+
+#endif
diff --git a/src/output/OutputThread.cxx b/src/output/OutputThread.cxx
new file mode 100644
index 000000000..e46425097
--- /dev/null
+++ b/src/output/OutputThread.cxx
@@ -0,0 +1,664 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "Internal.hxx"
+#include "OutputAPI.hxx"
+#include "Domain.hxx"
+#include "pcm/PcmMix.hxx"
+#include "notify.hxx"
+#include "filter/FilterInternal.hxx"
+#include "filter/plugins/ConvertFilterPlugin.hxx"
+#include "filter/plugins/ReplayGainFilterPlugin.hxx"
+#include "PlayerControl.hxx"
+#include "MusicPipe.hxx"
+#include "MusicChunk.hxx"
+#include "thread/Util.hxx"
+#include "thread/Slack.hxx"
+#include "thread/Name.hxx"
+#include "system/FatalError.hxx"
+#include "util/Error.hxx"
+#include "util/ConstBuffer.hxx"
+#include "Log.hxx"
+#include "Compiler.h"
+
+#include <assert.h>
+#include <string.h>
+
+void
+AudioOutput::CommandFinished()
+{
+	assert(command != AO_COMMAND_NONE);
+	command = AO_COMMAND_NONE;
+
+	mutex.unlock();
+	audio_output_client_notify.Signal();
+	mutex.lock();
+}
+
+inline bool
+AudioOutput::Enable()
+{
+	if (really_enabled)
+		return true;
+
+	mutex.unlock();
+	Error error;
+	bool success = ao_plugin_enable(this, error);
+	mutex.lock();
+	if (!success) {
+		FormatError(error,
+			    "Failed to enable \"%s\" [%s]",
+			    name, plugin.name);
+		return false;
+	}
+
+	really_enabled = true;
+	return true;
+}
+
+inline void
+AudioOutput::Disable()
+{
+	if (open)
+		Close(false);
+
+	if (really_enabled) {
+		really_enabled = false;
+
+		mutex.unlock();
+		ao_plugin_disable(this);
+		mutex.lock();
+	}
+}
+
+inline AudioFormat
+AudioOutput::OpenFilter(AudioFormat &format, Error &error_r)
+{
+	assert(format.IsValid());
+
+	/* the replay_gain filter cannot fail here */
+	if (replay_gain_filter != nullptr &&
+	    !replay_gain_filter->Open(format, error_r).IsDefined())
+		return AudioFormat::Undefined();
+
+	if (other_replay_gain_filter != nullptr &&
+	    !other_replay_gain_filter->Open(format, error_r).IsDefined()) {
+		if (replay_gain_filter != nullptr)
+			replay_gain_filter->Close();
+		return AudioFormat::Undefined();
+	}
+
+	const AudioFormat af = filter->Open(format, error_r);
+	if (!af.IsDefined()) {
+		if (replay_gain_filter != nullptr)
+			replay_gain_filter->Close();
+		if (other_replay_gain_filter != nullptr)
+			other_replay_gain_filter->Close();
+	}
+
+	return af;
+}
+
+void
+AudioOutput::CloseFilter()
+{
+	if (replay_gain_filter != nullptr)
+		replay_gain_filter->Close();
+	if (other_replay_gain_filter != nullptr)
+		other_replay_gain_filter->Close();
+
+	filter->Close();
+}
+
+inline void
+AudioOutput::Open()
+{
+	bool success;
+	Error error;
+	struct audio_format_string af_string;
+
+	assert(!open);
+	assert(pipe != nullptr);
+	assert(current_chunk == nullptr);
+	assert(in_audio_format.IsValid());
+
+	fail_timer.Reset();
+
+	/* enable the device (just in case the last enable has failed) */
+
+	if (!Enable())
+		/* still no luck */
+		return;
+
+	/* open the filter */
+
+	const AudioFormat filter_audio_format =
+		OpenFilter(in_audio_format, error);
+	if (!filter_audio_format.IsDefined()) {
+		FormatError(error, "Failed to open filter for \"%s\" [%s]",
+			    name, plugin.name);
+
+		fail_timer.Update();
+		return;
+	}
+
+	assert(filter_audio_format.IsValid());
+
+	out_audio_format = filter_audio_format;
+	out_audio_format.ApplyMask(config_audio_format);
+
+	mutex.unlock();
+	success = ao_plugin_open(this, out_audio_format, error);
+	mutex.lock();
+
+	assert(!open);
+
+	if (!success) {
+		FormatError(error, "Failed to open \"%s\" [%s]",
+			    name, plugin.name);
+
+		CloseFilter();
+		fail_timer.Update();
+		return;
+	}
+
+	if (!convert_filter_set(convert_filter, out_audio_format,
+				error)) {
+		FormatError(error, "Failed to convert for \"%s\" [%s]",
+			    name, plugin.name);
+
+		CloseFilter();
+		fail_timer.Update();
+		return;
+	}
+
+	open = true;
+
+	FormatDebug(output_domain,
+		    "opened plugin=%s name=\"%s\" audio_format=%s",
+		    plugin.name, name,
+		    audio_format_to_string(out_audio_format, &af_string));
+
+	if (in_audio_format != out_audio_format)
+		FormatDebug(output_domain, "converting from %s",
+			    audio_format_to_string(in_audio_format,
+						   &af_string));
+}
+
+void
+AudioOutput::Close(bool drain)
+{
+	assert(open);
+
+	pipe = nullptr;
+
+	current_chunk = nullptr;
+	open = false;
+
+	mutex.unlock();
+
+	if (drain)
+		ao_plugin_drain(this);
+	else
+		ao_plugin_cancel(this);
+
+	ao_plugin_close(this);
+	CloseFilter();
+
+	mutex.lock();
+
+	FormatDebug(output_domain, "closed plugin=%s name=\"%s\"",
+		    plugin.name, name);
+}
+
+void
+AudioOutput::ReopenFilter()
+{
+	Error error;
+
+	CloseFilter();
+	const AudioFormat filter_audio_format =
+		OpenFilter(in_audio_format, error);
+	if (!filter_audio_format.IsDefined() ||
+	    !convert_filter_set(convert_filter, out_audio_format,
+				error)) {
+		FormatError(error,
+			    "Failed to open filter for \"%s\" [%s]",
+			    name, plugin.name);
+
+		/* this is a little code duplication from Close(),
+		   but we cannot call this function because we must
+		   not call filter_close(filter) again */
+
+		pipe = nullptr;
+
+		current_chunk = nullptr;
+		open = false;
+		fail_timer.Update();
+
+		mutex.unlock();
+		ao_plugin_close(this);
+		mutex.lock();
+
+		return;
+	}
+}
+
+void
+AudioOutput::Reopen()
+{
+	if (!config_audio_format.IsFullyDefined()) {
+		if (open) {
+			const MusicPipe *mp = pipe;
+			Close(true);
+			pipe = mp;
+		}
+
+		/* no audio format is configured: copy in->out, let
+		   the output's open() method determine the effective
+		   out_audio_format */
+		out_audio_format = in_audio_format;
+		out_audio_format.ApplyMask(config_audio_format);
+	}
+
+	if (open)
+		/* the audio format has changed, and all filters have
+		   to be reconfigured */
+		ReopenFilter();
+	else
+		Open();
+}
+
+/**
+ * Wait until the output's delay reaches zero.
+ *
+ * @return true if playback should be continued, false if a command
+ * was issued
+ */
+inline bool
+AudioOutput::WaitForDelay()
+{
+	while (true) {
+		unsigned delay = ao_plugin_delay(this);
+		if (delay == 0)
+			return true;
+
+		(void)cond.timed_wait(mutex, delay);
+
+		if (command != AO_COMMAND_NONE)
+			return false;
+	}
+}
+
+static ConstBuffer<void>
+ao_chunk_data(AudioOutput *ao, const MusicChunk *chunk,
+	      Filter *replay_gain_filter,
+	      unsigned *replay_gain_serial_p)
+{
+	assert(chunk != nullptr);
+	assert(!chunk->IsEmpty());
+	assert(chunk->CheckFormat(ao->in_audio_format));
+
+	ConstBuffer<void> data(chunk->data, chunk->length);
+
+	(void)ao;
+
+	assert(data.size % ao->in_audio_format.GetFrameSize() == 0);
+
+	if (!data.IsEmpty() && replay_gain_filter != nullptr) {
+		if (chunk->replay_gain_serial != *replay_gain_serial_p) {
+			replay_gain_filter_set_info(replay_gain_filter,
+						    chunk->replay_gain_serial != 0
+						    ? &chunk->replay_gain_info
+						    : nullptr);
+			*replay_gain_serial_p = chunk->replay_gain_serial;
+		}
+
+		Error error;
+		data = replay_gain_filter->FilterPCM(data, error);
+		if (data.IsNull())
+			FormatError(error, "\"%s\" [%s] failed to filter",
+				    ao->name, ao->plugin.name);
+	}
+
+	return data;
+}
+
+static ConstBuffer<void>
+ao_filter_chunk(AudioOutput *ao, const MusicChunk *chunk)
+{
+	ConstBuffer<void> data =
+		ao_chunk_data(ao, chunk, ao->replay_gain_filter,
+			      &ao->replay_gain_serial);
+	if (data.IsEmpty())
+		return data;
+
+	/* cross-fade */
+
+	if (chunk->other != nullptr) {
+		ConstBuffer<void> other_data =
+			ao_chunk_data(ao, chunk->other,
+				      ao->other_replay_gain_filter,
+				      &ao->other_replay_gain_serial);
+		if (other_data.IsNull())
+			return nullptr;
+
+		if (other_data.IsEmpty())
+			return data;
+
+		/* if the "other" chunk is longer, then that trailer
+		   is used as-is, without mixing; it is part of the
+		   "next" song being faded in, and if there's a rest,
+		   it means cross-fading ends here */
+
+		if (data.size > other_data.size)
+			data.size = other_data.size;
+
+		float mix_ratio = chunk->mix_ratio;
+		if (mix_ratio >= 0)
+			/* reverse the mix ratio (because the
+			   arguments to pcm_mix() are reversed), but
+			   only if the mix ratio is non-negative; a
+			   negative mix ratio is a MixRamp special
+			   case */
+			mix_ratio = 1.0 - mix_ratio;
+
+		void *dest = ao->cross_fade_buffer.Get(other_data.size);
+		memcpy(dest, other_data.data, other_data.size);
+		if (!pcm_mix(ao->cross_fade_dither, dest, data.data, data.size,
+			     ao->in_audio_format.format,
+			     mix_ratio)) {
+			FormatError(output_domain,
+				    "Cannot cross-fade format %s",
+				    sample_format_to_string(ao->in_audio_format.format));
+			return nullptr;
+		}
+
+		data.data = dest;
+		data.size = other_data.size;
+	}
+
+	/* apply filter chain */
+
+	Error error;
+	data = ao->filter->FilterPCM(data, error);
+	if (data.IsNull()) {
+		FormatError(error, "\"%s\" [%s] failed to filter",
+			    ao->name, ao->plugin.name);
+		return nullptr;
+	}
+
+	return data;
+}
+
+inline bool
+AudioOutput::PlayChunk(const MusicChunk *chunk)
+{
+	assert(filter != nullptr);
+
+	if (tags && gcc_unlikely(chunk->tag != nullptr)) {
+		mutex.unlock();
+		ao_plugin_send_tag(this, chunk->tag);
+		mutex.lock();
+	}
+
+	auto data = ConstBuffer<char>::FromVoid(ao_filter_chunk(this, chunk));
+	if (data.IsNull()) {
+		Close(false);
+
+		/* don't automatically reopen this device for 10
+		   seconds */
+		fail_timer.Update();
+		return false;
+	}
+
+	Error error;
+
+	while (!data.IsEmpty() && command == AO_COMMAND_NONE) {
+		if (!WaitForDelay())
+			break;
+
+		mutex.unlock();
+		size_t nbytes = ao_plugin_play(this, data.data, data.size,
+					       error);
+		mutex.lock();
+		if (nbytes == 0) {
+			/* play()==0 means failure */
+			FormatError(error, "\"%s\" [%s] failed to play",
+				    name, plugin.name);
+
+			Close(false);
+
+			/* don't automatically reopen this device for
+			   10 seconds */
+			assert(!fail_timer.IsDefined());
+			fail_timer.Update();
+
+			return false;
+		}
+
+		assert(nbytes <= data.size);
+		assert(nbytes % out_audio_format.GetFrameSize() == 0);
+
+		data.data += nbytes;
+		data.size -= nbytes;
+	}
+
+	return true;
+}
+
+inline const MusicChunk *
+AudioOutput::GetNextChunk() const
+{
+	return current_chunk != nullptr
+		/* continue the previous play() call */
+		? current_chunk->next
+		/* get the first chunk from the pipe */
+		: pipe->Peek();
+}
+
+inline bool
+AudioOutput::Play()
+{
+	assert(pipe != nullptr);
+
+	const MusicChunk *chunk = GetNextChunk();
+	if (chunk == nullptr)
+		/* no chunk available */
+		return false;
+
+	current_chunk_finished = false;
+
+	assert(!in_playback_loop);
+	in_playback_loop = true;
+
+	while (chunk != nullptr && command == AO_COMMAND_NONE) {
+		assert(!current_chunk_finished);
+
+		current_chunk = chunk;
+
+		if (!PlayChunk(chunk)) {
+			assert(current_chunk == nullptr);
+			break;
+		}
+
+		assert(current_chunk == chunk);
+		chunk = chunk->next;
+	}
+
+	assert(in_playback_loop);
+	in_playback_loop = false;
+
+	current_chunk_finished = true;
+
+	mutex.unlock();
+	player_control->LockSignal();
+	mutex.lock();
+
+	return true;
+}
+
+inline void
+AudioOutput::Pause()
+{
+	mutex.unlock();
+	ao_plugin_cancel(this);
+	mutex.lock();
+
+	pause = true;
+	CommandFinished();
+
+	do {
+		if (!WaitForDelay())
+			break;
+
+		mutex.unlock();
+		bool success = ao_plugin_pause(this);
+		mutex.lock();
+
+		if (!success) {
+			Close(false);
+			break;
+		}
+	} while (command == AO_COMMAND_NONE);
+
+	pause = false;
+}
+
+inline void
+AudioOutput::Task()
+{
+	FormatThreadName("output:%s", name);
+
+	SetThreadRealtime();
+	SetThreadTimerSlackUS(100);
+
+	mutex.lock();
+
+	while (1) {
+		switch (command) {
+		case AO_COMMAND_NONE:
+			break;
+
+		case AO_COMMAND_ENABLE:
+			Enable();
+			CommandFinished();
+			break;
+
+		case AO_COMMAND_DISABLE:
+			Disable();
+			CommandFinished();
+			break;
+
+		case AO_COMMAND_OPEN:
+			Open();
+			CommandFinished();
+			break;
+
+		case AO_COMMAND_REOPEN:
+			Reopen();
+			CommandFinished();
+			break;
+
+		case AO_COMMAND_CLOSE:
+			assert(open);
+			assert(pipe != nullptr);
+
+			Close(false);
+			CommandFinished();
+			break;
+
+		case AO_COMMAND_PAUSE:
+			if (!open) {
+				/* the output has failed after
+				   audio_output_all_pause() has
+				   submitted the PAUSE command; bail
+				   out */
+				CommandFinished();
+				break;
+			}
+
+			Pause();
+			/* don't "break" here: this might cause
+			   Play() to be called when command==CLOSE
+			   ends the paused state - "continue" checks
+			   the new command first */
+			continue;
+
+		case AO_COMMAND_DRAIN:
+			if (open) {
+				assert(current_chunk == nullptr);
+				assert(pipe->Peek() == nullptr);
+
+				mutex.unlock();
+				ao_plugin_drain(this);
+				mutex.lock();
+			}
+
+			CommandFinished();
+			continue;
+
+		case AO_COMMAND_CANCEL:
+			current_chunk = nullptr;
+
+			if (open) {
+				mutex.unlock();
+				ao_plugin_cancel(this);
+				mutex.lock();
+			}
+
+			CommandFinished();
+			continue;
+
+		case AO_COMMAND_KILL:
+			current_chunk = nullptr;
+			CommandFinished();
+			mutex.unlock();
+			return;
+		}
+
+		if (open && allow_play && Play())
+			/* don't wait for an event if there are more
+			   chunks in the pipe */
+			continue;
+
+		if (command == AO_COMMAND_NONE) {
+			woken_for_play = false;
+			cond.wait(mutex);
+		}
+	}
+}
+
+void
+AudioOutput::Task(void *arg)
+{
+	AudioOutput *ao = (AudioOutput *)arg;
+	ao->Task();
+}
+
+void
+AudioOutput::StartThread()
+{
+	assert(command == AO_COMMAND_NONE);
+
+	Error error;
+	if (!thread.Start(Task, this, error))
+		FatalError(error);
+}
diff --git a/src/output/Registry.cxx b/src/output/Registry.cxx
new file mode 100644
index 000000000..566f6b6a8
--- /dev/null
+++ b/src/output/Registry.cxx
@@ -0,0 +1,104 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "Registry.hxx"
+#include "OutputAPI.hxx"
+#include "plugins/AlsaOutputPlugin.hxx"
+#include "plugins/AoOutputPlugin.hxx"
+#include "plugins/FifoOutputPlugin.hxx"
+#include "plugins/httpd/HttpdOutputPlugin.hxx"
+#include "plugins/JackOutputPlugin.hxx"
+#include "plugins/NullOutputPlugin.hxx"
+#include "plugins/OpenALOutputPlugin.hxx"
+#include "plugins/OssOutputPlugin.hxx"
+#include "plugins/OSXOutputPlugin.hxx"
+#include "plugins/PipeOutputPlugin.hxx"
+#include "plugins/PulseOutputPlugin.hxx"
+#include "plugins/RecorderOutputPlugin.hxx"
+#include "plugins/RoarOutputPlugin.hxx"
+#include "plugins/ShoutOutputPlugin.hxx"
+#include "plugins/sles/SlesOutputPlugin.hxx"
+#include "plugins/SolarisOutputPlugin.hxx"
+#include "plugins/WinmmOutputPlugin.hxx"
+
+#include <string.h>
+
+const AudioOutputPlugin *const audio_output_plugins[] = {
+#ifdef HAVE_SHOUT
+	&shout_output_plugin,
+#endif
+	&null_output_plugin,
+#ifdef ANDROID
+	&sles_output_plugin,
+#endif
+#ifdef HAVE_FIFO
+	&fifo_output_plugin,
+#endif
+#ifdef ENABLE_PIPE_OUTPUT
+	&pipe_output_plugin,
+#endif
+#ifdef HAVE_ALSA
+	&alsa_output_plugin,
+#endif
+#ifdef HAVE_ROAR
+	&roar_output_plugin,
+#endif
+#ifdef HAVE_AO
+	&ao_output_plugin,
+#endif
+#ifdef HAVE_OSS
+	&oss_output_plugin,
+#endif
+#ifdef HAVE_OPENAL
+	&openal_output_plugin,
+#endif
+#ifdef HAVE_OSX
+	&osx_output_plugin,
+#endif
+#ifdef ENABLE_SOLARIS_OUTPUT
+	&solaris_output_plugin,
+#endif
+#ifdef HAVE_PULSE
+	&pulse_output_plugin,
+#endif
+#ifdef HAVE_JACK
+	&jack_output_plugin,
+#endif
+#ifdef ENABLE_HTTPD_OUTPUT
+	&httpd_output_plugin,
+#endif
+#ifdef ENABLE_RECORDER_OUTPUT
+	&recorder_output_plugin,
+#endif
+#ifdef ENABLE_WINMM_OUTPUT
+	&winmm_output_plugin,
+#endif
+	nullptr
+};
+
+const AudioOutputPlugin *
+AudioOutputPlugin_get(const char *name)
+{
+	audio_output_plugins_for_each(plugin)
+		if (strcmp(plugin->name, name) == 0)
+			return plugin;
+
+	return nullptr;
+}
diff --git a/src/output/Registry.hxx b/src/output/Registry.hxx
new file mode 100644
index 000000000..bc9c1ae2b
--- /dev/null
+++ b/src/output/Registry.hxx
@@ -0,0 +1,35 @@
+/*
+ * 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_OUTPUT_LIST_HXX
+#define MPD_OUTPUT_LIST_HXX
+
+struct AudioOutputPlugin;
+
+extern const AudioOutputPlugin *const audio_output_plugins[];
+
+const AudioOutputPlugin *
+AudioOutputPlugin_get(const char *name);
+
+#define audio_output_plugins_for_each(plugin) \
+	for (const AudioOutputPlugin *plugin, \
+		*const*output_plugin_iterator = &audio_output_plugins[0]; \
+		(plugin = *output_plugin_iterator) != nullptr; ++output_plugin_iterator)
+
+#endif
diff --git a/src/output/Timer.cxx b/src/output/Timer.cxx
new file mode 100644
index 000000000..d3dcc714d
--- /dev/null
+++ b/src/output/Timer.cxx
@@ -0,0 +1,67 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "Timer.hxx"
+#include "AudioFormat.hxx"
+#include "system/Clock.hxx"
+
+#include <limits>
+
+#include <assert.h>
+
+Timer::Timer(const AudioFormat af)
+	: time(0),
+	  started(false),
+	 rate(af.sample_rate * af.GetFrameSize())
+{
+}
+
+void Timer::Start()
+{
+	time = MonotonicClockUS();
+	started = true;
+}
+
+void Timer::Reset()
+{
+	time = 0;
+	started = false;
+}
+
+void Timer::Add(int size)
+{
+	assert(started);
+
+	// (size samples) / (rate samples per second) = duration seconds
+	// duration seconds * 1000000 = duration us
+	time += ((uint64_t)size * 1000000) / rate;
+}
+
+unsigned Timer::GetDelay() const
+{
+	int64_t delay = (int64_t)(time - MonotonicClockUS()) / 1000;
+	if (delay < 0)
+		return 0;
+
+	if (delay > std::numeric_limits<int>::max())
+		delay = std::numeric_limits<int>::max();
+
+	return delay;
+}
diff --git a/src/output/Timer.hxx b/src/output/Timer.hxx
new file mode 100644
index 000000000..3c935cfac
--- /dev/null
+++ b/src/output/Timer.hxx
@@ -0,0 +1,47 @@
+/*
+ * 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_TIMER_HXX
+#define MPD_TIMER_HXX
+
+#include <stdint.h>
+
+struct AudioFormat;
+
+class Timer {
+	uint64_t time;
+	bool started;
+	const int rate;
+public:
+	explicit Timer(AudioFormat af);
+
+	bool IsStarted() const { return started; }
+
+	void Start();
+	void Reset();
+
+	void Add(int size);
+
+	/**
+	 * Returns the number of milliseconds to sleep to get back to sync.
+	 */
+	unsigned GetDelay() const;
+};
+
+#endif
diff --git a/src/output/AlsaOutputPlugin.cxx b/src/output/plugins/AlsaOutputPlugin.cxx
index f8aae13a1..33a090274 100644
--- a/src/output/AlsaOutputPlugin.cxx
+++ b/src/output/plugins/AlsaOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,33 +19,36 @@
 
 #include "config.h"
 #include "AlsaOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "MixerList.hxx"
+#include "../OutputAPI.hxx"
+#include "mixer/MixerList.hxx"
 #include "pcm/PcmExport.hxx"
+#include "config/ConfigError.hxx"
 #include "util/Manual.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
+#include "util/ConstBuffer.hxx"
 #include "Log.hxx"
 
-#include <glib.h>
 #include <alsa/asoundlib.h>
 
 #include <string>
 
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
+#if SND_LIB_VERSION >= 0x1001c
+/* alsa-lib supports DSD since version 1.0.27.1 */
+#define HAVE_ALSA_DSD
+#endif
 
 static const char default_device[] = "default";
 
 static constexpr unsigned MPD_ALSA_BUFFER_TIME_US = 500000;
 
-#define MPD_ALSA_RETRY_NR 5
+static constexpr unsigned MPD_ALSA_RETRY_NR = 5;
 
 typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
 					snd_pcm_uframes_t size);
 
 struct AlsaOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	Manual<PcmExport> pcm_export;
 
@@ -59,12 +62,11 @@ struct AlsaOutput {
 	bool use_mmap;
 
 	/**
-	 * Enable DSD over USB according to the dCS suggested
-	 * standard?
+	 * Enable DSD over PCM according to the DoP standard standard?
 	 *
-	 * @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
+	 * @see http://dsd-guide.com/dop-open-standard
 	 */
-	bool dsd_usb;
+	bool dop;
 
 	/** libasound's buffer_time setting (in microseconds) */
 	unsigned int buffer_time;
@@ -122,19 +124,14 @@ struct AlsaOutput {
 	 * It contains silence samples, enough to fill one period (see
 	 * #period_frames).
 	 */
-	void *silence;
-
-	AlsaOutput():mode(0), writei(snd_pcm_writei) {
-	}
+	uint8_t *silence;
 
-	bool Init(const config_param &param, Error &error) {
-		return ao_base_init(&base, &alsa_output_plugin,
-				    param, error);
+	AlsaOutput()
+		:base(alsa_output_plugin),
+		 mode(0), writei(snd_pcm_writei) {
 	}
 
-	void Deinit() {
-		ao_base_finish(&base);
-	}
+	bool Configure(const config_param &param, Error &error);
 };
 
 static constexpr Domain alsa_output_domain("alsa_output");
@@ -145,56 +142,60 @@ alsa_device(const AlsaOutput *ad)
 	return ad->device.empty() ? default_device : ad->device.c_str();
 }
 
-static void
-alsa_configure(AlsaOutput *ad, const config_param &param)
+inline bool
+AlsaOutput::Configure(const config_param &param, Error &error)
 {
-	ad->device = param.GetBlockValue("device", "");
+	if (!base.Configure(param, error))
+		return false;
 
-	ad->use_mmap = param.GetBlockValue("use_mmap", false);
+	device = param.GetBlockValue("device", "");
 
-	ad->dsd_usb = param.GetBlockValue("dsd_usb", false);
+	use_mmap = param.GetBlockValue("use_mmap", false);
 
-	ad->buffer_time = param.GetBlockValue("buffer_time",
+	dop = param.GetBlockValue("dop", false) ||
+		/* legacy name from MPD 0.18 and older: */
+		param.GetBlockValue("dsd_usb", false);
+
+	buffer_time = param.GetBlockValue("buffer_time",
 					      MPD_ALSA_BUFFER_TIME_US);
-	ad->period_time = param.GetBlockValue("period_time", 0u);
+	period_time = param.GetBlockValue("period_time", 0u);
 
 #ifdef SND_PCM_NO_AUTO_RESAMPLE
 	if (!param.GetBlockValue("auto_resample", true))
-		ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
+		mode |= SND_PCM_NO_AUTO_RESAMPLE;
 #endif
 
 #ifdef SND_PCM_NO_AUTO_CHANNELS
 	if (!param.GetBlockValue("auto_channels", true))
-		ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
+		mode |= SND_PCM_NO_AUTO_CHANNELS;
 #endif
 
 #ifdef SND_PCM_NO_AUTO_FORMAT
 	if (!param.GetBlockValue("auto_format", true))
-		ad->mode |= SND_PCM_NO_AUTO_FORMAT;
+		mode |= SND_PCM_NO_AUTO_FORMAT;
 #endif
+
+	return true;
 }
 
-static struct audio_output *
+static AudioOutput *
 alsa_init(const config_param &param, Error &error)
 {
 	AlsaOutput *ad = new AlsaOutput();
 
-	if (!ad->Init(param, error)) {
+	if (!ad->Configure(param, error)) {
 		delete ad;
 		return nullptr;
 	}
 
-	alsa_configure(ad, param);
-
 	return &ad->base;
 }
 
 static void
-alsa_finish(struct audio_output *ao)
+alsa_finish(AudioOutput *ao)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
-	ad->Deinit();
 	delete ad;
 
 	/* free libasound's config cache */
@@ -202,7 +203,7 @@ alsa_finish(struct audio_output *ao)
 }
 
 static bool
-alsa_output_enable(struct audio_output *ao, gcc_unused Error &error)
+alsa_output_enable(AudioOutput *ao, gcc_unused Error &error)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
@@ -211,7 +212,7 @@ alsa_output_enable(struct audio_output *ao, gcc_unused Error &error)
 }
 
 static void
-alsa_output_disable(struct audio_output *ao)
+alsa_output_disable(AudioOutput *ao)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
@@ -219,12 +220,12 @@ alsa_output_disable(struct audio_output *ao)
 }
 
 static bool
-alsa_test_default_device(void)
+alsa_test_default_device()
 {
 	snd_pcm_t *handle;
 
 	int ret = snd_pcm_open(&handle, default_device,
-	                       SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
+			       SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
 	if (ret) {
 		FormatError(alsa_output_domain,
 			    "Error opening default ALSA device: %s",
@@ -236,13 +237,24 @@ alsa_test_default_device(void)
 	return true;
 }
 
+/**
+ * Convert MPD's #SampleFormat enum to libasound's snd_pcm_format_t
+ * enum.  Returns SND_PCM_FORMAT_UNKNOWN if there is no according ALSA
+ * PCM format.
+ */
 static snd_pcm_format_t
 get_bitformat(SampleFormat sample_format)
 {
 	switch (sample_format) {
 	case SampleFormat::UNDEFINED:
+		return SND_PCM_FORMAT_UNKNOWN;
+
 	case SampleFormat::DSD:
+#ifdef HAVE_ALSA_DSD
+		return SND_PCM_FORMAT_DSD_U8;
+#else
 		return SND_PCM_FORMAT_UNKNOWN;
+#endif
 
 	case SampleFormat::S8:
 		return SND_PCM_FORMAT_S8;
@@ -264,10 +276,14 @@ get_bitformat(SampleFormat sample_format)
 	gcc_unreachable();
 }
 
+/**
+ * Determine the byte-swapped PCM format.  Returns
+ * SND_PCM_FORMAT_UNKNOWN if the format cannot be byte-swapped.
+ */
 static snd_pcm_format_t
 byteswap_bitformat(snd_pcm_format_t fmt)
 {
-	switch(fmt) {
+	switch (fmt) {
 	case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
 	case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
 	case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
@@ -285,6 +301,10 @@ byteswap_bitformat(snd_pcm_format_t fmt)
 	}
 }
 
+/**
+ * Check if there is a "packed" version of the give PCM format.
+ * Returns SND_PCM_FORMAT_UNKNOWN if not.
+ */
 static snd_pcm_format_t
 alsa_to_packed_format(snd_pcm_format_t fmt)
 {
@@ -300,6 +320,10 @@ alsa_to_packed_format(snd_pcm_format_t fmt)
 	}
 }
 
+/**
+ * Attempts to configure the specified sample format.  On failure,
+ * fall back to the packed version.
+ */
 static int
 alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 			  snd_pcm_format_t fmt, bool *packed_r)
@@ -370,7 +394,7 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
 
 	/* if unsupported by the hardware, try other formats */
 
-	static const SampleFormat probe_formats[] = {
+	static constexpr SampleFormat probe_formats[] = {
 		SampleFormat::S24_P32,
 		SampleFormat::S32,
 		SampleFormat::S16,
@@ -406,7 +430,7 @@ alsa_setup(AlsaOutput *ad, AudioFormat &audio_format,
 	unsigned int channels = audio_format.channels;
 	int err;
 	const char *cmd = nullptr;
-	int retry = MPD_ALSA_RETRY_NR;
+	unsigned retry = MPD_ALSA_RETRY_NR;
 	unsigned int period_time, period_time_ro;
 	unsigned int buffer_time;
 
@@ -597,8 +621,8 @@ configure_hw:
 	ad->period_frames = alsa_period_size;
 	ad->period_position = 0;
 
-	ad->silence = g_malloc(snd_pcm_frames_to_bytes(ad->pcm,
-						       alsa_period_size));
+	ad->silence = new uint8_t[snd_pcm_frames_to_bytes(ad->pcm,
+							  alsa_period_size)];
 	snd_pcm_format_set_silence(format, ad->silence,
 				   alsa_period_size * channels);
 
@@ -616,36 +640,36 @@ alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format,
 	       bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
 	       Error &error)
 {
-	assert(ad->dsd_usb);
+	assert(ad->dop);
 	assert(audio_format.format == SampleFormat::DSD);
 
 	/* pass 24 bit to alsa_setup() */
 
-	AudioFormat usb_format = audio_format;
-	usb_format.format = SampleFormat::S24_P32;
-	usb_format.sample_rate /= 2;
+	AudioFormat dop_format = audio_format;
+	dop_format.format = SampleFormat::S24_P32;
+	dop_format.sample_rate /= 2;
 
-	const AudioFormat check = usb_format;
+	const AudioFormat check = dop_format;
 
-	if (!alsa_setup(ad, usb_format, packed_r, reverse_endian_r, error))
+	if (!alsa_setup(ad, dop_format, packed_r, reverse_endian_r, error))
 		return false;
 
-	/* if the device allows only 32 bit, shift all DSD-over-USB
+	/* if the device allows only 32 bit, shift all DoP
 	   samples left by 8 bit and leave the lower 8 bit cleared;
 	   the DSD-over-USB documentation does not specify whether
 	   this is legal, but there is anecdotical evidence that this
 	   is possible (and the only option for some devices) */
-	*shift8_r = usb_format.format == SampleFormat::S32;
-	if (usb_format.format == SampleFormat::S32)
-		usb_format.format = SampleFormat::S24_P32;
+	*shift8_r = dop_format.format == SampleFormat::S32;
+	if (dop_format.format == SampleFormat::S32)
+		dop_format.format = SampleFormat::S24_P32;
 
-	if (usb_format != check) {
+	if (dop_format != check) {
 		/* no bit-perfect playback, which is required
 		   for DSD over USB */
 		error.Format(alsa_output_domain,
-			     "Failed to configure DSD-over-USB on ALSA device \"%s\"",
+			     "Failed to configure DSD-over-PCM on ALSA device \"%s\"",
 			     alsa_device(ad));
-		g_free(ad->silence);
+		delete[] ad->silence;
 		return false;
 	}
 
@@ -658,9 +682,9 @@ alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
 {
 	bool shift8 = false, packed, reverse_endian;
 
-	const bool dsd_usb = ad->dsd_usb &&
+	const bool dop = ad->dop &&
 		audio_format.format == SampleFormat::DSD;
-	const bool success = dsd_usb
+	const bool success = dop
 		? alsa_setup_dsd(ad, audio_format,
 				 &shift8, &packed, &reverse_endian,
 				 error)
@@ -671,12 +695,12 @@ alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
 
 	ad->pcm_export->Open(audio_format.format,
 			     audio_format.channels,
-			     dsd_usb, shift8, packed, reverse_endian);
+			     dop, shift8, packed, reverse_endian);
 	return true;
 }
 
 static bool
-alsa_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
+alsa_open(AudioOutput *ao, AudioFormat &audio_format, Error &error)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
@@ -756,7 +780,7 @@ alsa_recover(AlsaOutput *ad, int err)
 }
 
 static void
-alsa_drain(struct audio_output *ao)
+alsa_drain(AudioOutput *ao)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
@@ -777,7 +801,7 @@ alsa_drain(struct audio_output *ao)
 }
 
 static void
-alsa_cancel(struct audio_output *ao)
+alsa_cancel(AudioOutput *ao)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
@@ -788,16 +812,16 @@ alsa_cancel(struct audio_output *ao)
 }
 
 static void
-alsa_close(struct audio_output *ao)
+alsa_close(AudioOutput *ao)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
 
 	snd_pcm_close(ad->pcm);
-	g_free(ad->silence);
+	delete[] ad->silence;
 }
 
 static size_t
-alsa_play(struct audio_output *ao, const void *chunk, size_t size,
+alsa_play(AudioOutput *ao, const void *chunk, size_t size,
 	  Error &error)
 {
 	AlsaOutput *ad = (AlsaOutput *)ao;
@@ -815,16 +839,18 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 		}
 	}
 
-	const size_t original_size = size;
-	chunk = ad->pcm_export->Export(chunk, size, size);
-	if (size == 0)
+	const auto e = ad->pcm_export->Export({chunk, size});
+	if (e.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;
+		return size;
+
+	chunk = e.data;
+	size = e.size;
 
 	assert(size % ad->out_frame_size == 0);
 
@@ -849,7 +875,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 	}
 }
 
-const struct audio_output_plugin alsa_output_plugin = {
+const struct AudioOutputPlugin alsa_output_plugin = {
 	"alsa",
 	alsa_test_default_device,
 	alsa_init,
diff --git a/src/output/AlsaOutputPlugin.hxx b/src/output/plugins/AlsaOutputPlugin.hxx
index dc7e639a8..f72116f91 100644
--- a/src/output/AlsaOutputPlugin.hxx
+++ b/src/output/plugins/AlsaOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_ALSA_OUTPUT_PLUGIN_HXX
 #define MPD_ALSA_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin alsa_output_plugin;
+extern const struct AudioOutputPlugin alsa_output_plugin;
 
 #endif
diff --git a/src/output/AoOutputPlugin.cxx b/src/output/plugins/AoOutputPlugin.cxx
index e66969e20..af8c88fa1 100644
--- a/src/output/AoOutputPlugin.cxx
+++ b/src/output/plugins/AoOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,7 +19,7 @@
 
 #include "config.h"
 #include "AoOutputPlugin.hxx"
-#include "OutputAPI.hxx"
+#include "../OutputAPI.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "Log.hxx"
@@ -35,20 +35,18 @@ static ao_sample_format OUR_AO_FORMAT_INITIALIZER;
 static unsigned ao_output_ref;
 
 struct AoOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	size_t write_size;
 	int driver;
 	ao_option *options;
 	ao_device *device;
 
-	bool Initialize(const config_param &param, Error &error) {
-		return ao_base_init(&base, &ao_output_plugin, param,
-				    error);
-	}
+	AoOutput()
+		:base(ao_output_plugin) {}
 
-	void Deinitialize() {
-		ao_base_finish(&base);
+	bool Initialize(const config_param &param, Error &error) {
+		return base.Configure(param, error);
 	}
 
 	bool Configure(const config_param &param, Error &error);
@@ -152,7 +150,7 @@ AoOutput::Configure(const config_param &param, Error &error)
 	return true;
 }
 
-static struct audio_output *
+static AudioOutput *
 ao_output_init(const config_param &param, Error &error)
 {
 	AoOutput *ad = new AoOutput();
@@ -163,7 +161,6 @@ ao_output_init(const config_param &param, Error &error)
 	}
 
 	if (!ad->Configure(param, error)) {
-		ad->Deinitialize();
 		delete ad;
 		return nullptr;
 	}
@@ -172,12 +169,11 @@ ao_output_init(const config_param &param, Error &error)
 }
 
 static void
-ao_output_finish(struct audio_output *ao)
+ao_output_finish(AudioOutput *ao)
 {
 	AoOutput *ad = (AoOutput *)ao;
 
 	ao_free_options(ad->options);
-	ad->Deinitialize();
 	delete ad;
 
 	ao_output_ref--;
@@ -187,7 +183,7 @@ ao_output_finish(struct audio_output *ao)
 }
 
 static void
-ao_output_close(struct audio_output *ao)
+ao_output_close(AudioOutput *ao)
 {
 	AoOutput *ad = (AoOutput *)ao;
 
@@ -195,7 +191,7 @@ ao_output_close(struct audio_output *ao)
 }
 
 static bool
-ao_output_open(struct audio_output *ao, AudioFormat &audio_format,
+ao_output_open(AudioOutput *ao, AudioFormat &audio_format,
 	       Error &error)
 {
 	ao_sample_format format = OUR_AO_FORMAT_INITIALIZER;
@@ -251,7 +247,7 @@ static int ao_play_deconst(ao_device *device, const void *output_samples,
 }
 
 static size_t
-ao_output_play(struct audio_output *ao, const void *chunk, size_t size,
+ao_output_play(AudioOutput *ao, const void *chunk, size_t size,
 	       Error &error)
 {
 	AoOutput *ad = (AoOutput *)ao;
@@ -267,7 +263,7 @@ ao_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	return size;
 }
 
-const struct audio_output_plugin ao_output_plugin = {
+const struct AudioOutputPlugin ao_output_plugin = {
 	"ao",
 	nullptr,
 	ao_output_init,
diff --git a/src/output/AoOutputPlugin.hxx b/src/output/plugins/AoOutputPlugin.hxx
index a44885e56..07c2ba16b 100644
--- a/src/output/AoOutputPlugin.hxx
+++ b/src/output/plugins/AoOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_AO_OUTPUT_PLUGIN_HXX
 #define MPD_AO_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin ao_output_plugin;
+extern const struct AudioOutputPlugin ao_output_plugin;
 
 #endif
diff --git a/src/output/FifoOutputPlugin.cxx b/src/output/plugins/FifoOutputPlugin.cxx
index aeb9a6a87..9df5a74dd 100644
--- a/src/output/FifoOutputPlugin.cxx
+++ b/src/output/plugins/FifoOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,10 +19,9 @@
 
 #include "config.h"
 #include "FifoOutputPlugin.hxx"
-#include "ConfigError.hxx"
-#include "OutputAPI.hxx"
-#include "Timer.hxx"
-#include "system/fd_util.h"
+#include "config/ConfigError.hxx"
+#include "../OutputAPI.hxx"
+#include "../Timer.hxx"
 #include "fs/AllocatedPath.hxx"
 #include "fs/FileSystem.hxx"
 #include "util/Error.hxx"
@@ -30,16 +29,14 @@
 #include "Log.hxx"
 #include "open.h"
 
-#include <sys/types.h>
 #include <sys/stat.h>
 #include <errno.h>
-#include <string.h>
 #include <unistd.h>
 
 #define FIFO_BUFFER_SIZE 65536 /* pipe capacity on Linux >= 2.6.11 */
 
 struct FifoOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	AllocatedPath path;
 	std::string path_utf8;
@@ -50,16 +47,12 @@ struct FifoOutput {
 	Timer *timer;
 
 	FifoOutput()
-		:path(AllocatedPath::Null()), input(-1), output(-1),
+		:base(fifo_output_plugin),
+		 path(AllocatedPath::Null()), input(-1), output(-1),
 		 created(false) {}
 
 	bool Initialize(const config_param &param, Error &error) {
-		return ao_base_init(&base, &fifo_output_plugin, param,
-				    error);
-	}
-
-	void Deinitialize() {
-		ao_base_finish(&base);
+		return base.Configure(param, error);
 	}
 
 	bool Create(Error &error);
@@ -175,7 +168,7 @@ fifo_open(FifoOutput *fd, Error &error)
 	return fd->Open(error);
 }
 
-static struct audio_output *
+static AudioOutput *
 fifo_output_init(const config_param &param, Error &error)
 {
 	FifoOutput *fd = new FifoOutput();
@@ -198,7 +191,6 @@ fifo_output_init(const config_param &param, Error &error)
 	}
 
 	if (!fifo_open(fd, error)) {
-		fd->Deinitialize();
 		delete fd;
 		return nullptr;
 	}
@@ -207,17 +199,16 @@ fifo_output_init(const config_param &param, Error &error)
 }
 
 static void
-fifo_output_finish(struct audio_output *ao)
+fifo_output_finish(AudioOutput *ao)
 {
 	FifoOutput *fd = (FifoOutput *)ao;
 
 	fd->Close();
-	fd->Deinitialize();
 	delete fd;
 }
 
 static bool
-fifo_output_open(struct audio_output *ao, AudioFormat &audio_format,
+fifo_output_open(AudioOutput *ao, AudioFormat &audio_format,
 		 gcc_unused Error &error)
 {
 	FifoOutput *fd = (FifoOutput *)ao;
@@ -228,7 +219,7 @@ fifo_output_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-fifo_output_close(struct audio_output *ao)
+fifo_output_close(AudioOutput *ao)
 {
 	FifoOutput *fd = (FifoOutput *)ao;
 
@@ -236,7 +227,7 @@ fifo_output_close(struct audio_output *ao)
 }
 
 static void
-fifo_output_cancel(struct audio_output *ao)
+fifo_output_cancel(AudioOutput *ao)
 {
 	FifoOutput *fd = (FifoOutput *)ao;
 	char buf[FIFO_BUFFER_SIZE];
@@ -255,7 +246,7 @@ fifo_output_cancel(struct audio_output *ao)
 }
 
 static unsigned
-fifo_output_delay(struct audio_output *ao)
+fifo_output_delay(AudioOutput *ao)
 {
 	FifoOutput *fd = (FifoOutput *)ao;
 
@@ -265,7 +256,7 @@ fifo_output_delay(struct audio_output *ao)
 }
 
 static size_t
-fifo_output_play(struct audio_output *ao, const void *chunk, size_t size,
+fifo_output_play(AudioOutput *ao, const void *chunk, size_t size,
 		 Error &error)
 {
 	FifoOutput *fd = (FifoOutput *)ao;
@@ -297,7 +288,7 @@ fifo_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	}
 }
 
-const struct audio_output_plugin fifo_output_plugin = {
+const struct AudioOutputPlugin fifo_output_plugin = {
 	"fifo",
 	nullptr,
 	fifo_output_init,
diff --git a/src/output/FifoOutputPlugin.hxx b/src/output/plugins/FifoOutputPlugin.hxx
index dca2886d8..f41ceded6 100644
--- a/src/output/FifoOutputPlugin.hxx
+++ b/src/output/plugins/FifoOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_FIFO_OUTPUT_PLUGIN_HXX
 #define MPD_FIFO_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin fifo_output_plugin;
+extern const struct AudioOutputPlugin fifo_output_plugin;
 
 #endif
diff --git a/src/output/JackOutputPlugin.cxx b/src/output/plugins/JackOutputPlugin.cxx
index 7ed672f95..e1dad7893 100644
--- a/src/output/JackOutputPlugin.cxx
+++ b/src/output/plugins/JackOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,8 +19,8 @@
 
 #include "config.h"
 #include "JackOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "ConfigError.hxx"
+#include "../OutputAPI.hxx"
+#include "config/ConfigError.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "Log.hxx"
@@ -34,10 +34,6 @@
 
 #include <stdlib.h>
 #include <string.h>
-#include <stdio.h>
-#include <sys/types.h>
-#include <unistd.h>
-#include <errno.h>
 
 enum {
 	MAX_PORTS = 16,
@@ -46,7 +42,7 @@ enum {
 static const size_t jack_sample_size = sizeof(jack_default_audio_sample_t);
 
 struct JackOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	/**
 	 * libjack options passed to jack_client_open().
@@ -83,13 +79,11 @@ struct JackOutput {
 	 */
 	bool pause;
 
-	bool Initialize(const config_param &param, Error &error_r) {
-		return ao_base_init(&base, &jack_output_plugin, param,
-				    error_r);
-	}
+	JackOutput()
+		:base(jack_output_plugin) {}
 
-	void Deinitialize() {
-		ao_base_finish(&base);
+	bool Initialize(const config_param &param, Error &error_r) {
+		return base.Configure(param, error_r);
 	}
 };
 
@@ -314,7 +308,7 @@ parse_port_list(const char *source, char **dest, Error &error)
 	return n;
 }
 
-static struct audio_output *
+static AudioOutput *
 mpd_jack_init(const config_param &param, Error &error)
 {
 	JackOutput *jd = new JackOutput();
@@ -393,7 +387,7 @@ mpd_jack_init(const config_param &param, Error &error)
 }
 
 static void
-mpd_jack_finish(struct audio_output *ao)
+mpd_jack_finish(AudioOutput *ao)
 {
 	JackOutput *jd = (JackOutput *)ao;
 
@@ -403,12 +397,11 @@ mpd_jack_finish(struct audio_output *ao)
 	for (unsigned i = 0; i < jd->num_destination_ports; ++i)
 		g_free(jd->destination_ports[i]);
 
-	jd->Deinitialize();
 	delete jd;
 }
 
 static bool
-mpd_jack_enable(struct audio_output *ao, Error &error)
+mpd_jack_enable(AudioOutput *ao, Error &error)
 {
 	JackOutput *jd = (JackOutput *)ao;
 
@@ -419,7 +412,7 @@ mpd_jack_enable(struct audio_output *ao, Error &error)
 }
 
 static void
-mpd_jack_disable(struct audio_output *ao)
+mpd_jack_disable(AudioOutput *ao)
 {
 	JackOutput *jd = (JackOutput *)ao;
 
@@ -583,7 +576,7 @@ mpd_jack_start(JackOutput *jd, Error &error)
 }
 
 static bool
-mpd_jack_open(struct audio_output *ao, AudioFormat &audio_format,
+mpd_jack_open(AudioOutput *ao, AudioFormat &audio_format,
 	      Error &error)
 {
 	JackOutput *jd = (JackOutput *)ao;
@@ -608,7 +601,7 @@ mpd_jack_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-mpd_jack_close(gcc_unused struct audio_output *ao)
+mpd_jack_close(gcc_unused AudioOutput *ao)
 {
 	JackOutput *jd = (JackOutput *)ao;
 
@@ -616,7 +609,7 @@ mpd_jack_close(gcc_unused struct audio_output *ao)
 }
 
 static unsigned
-mpd_jack_delay(struct audio_output *ao)
+mpd_jack_delay(AudioOutput *ao)
 {
 	JackOutput *jd = (JackOutput *)ao;
 
@@ -693,7 +686,7 @@ mpd_jack_write_samples(JackOutput *jd, const void *src,
 }
 
 static size_t
-mpd_jack_play(struct audio_output *ao, const void *chunk, size_t size,
+mpd_jack_play(AudioOutput *ao, const void *chunk, size_t size,
 	      Error &error)
 {
 	JackOutput *jd = (JackOutput *)ao;
@@ -738,7 +731,7 @@ mpd_jack_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 static bool
-mpd_jack_pause(struct audio_output *ao)
+mpd_jack_pause(AudioOutput *ao)
 {
 	JackOutput *jd = (JackOutput *)ao;
 
@@ -750,7 +743,7 @@ mpd_jack_pause(struct audio_output *ao)
 	return true;
 }
 
-const struct audio_output_plugin jack_output_plugin = {
+const struct AudioOutputPlugin jack_output_plugin = {
 	"jack",
 	mpd_jack_test_default_device,
 	mpd_jack_init,
diff --git a/src/output/JackOutputPlugin.hxx b/src/output/plugins/JackOutputPlugin.hxx
index 908105ad2..6f1f7ecb9 100644
--- a/src/output/JackOutputPlugin.hxx
+++ b/src/output/plugins/JackOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_JACK_OUTPUT_PLUGIN_HXX
 #define MPD_JACK_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin jack_output_plugin;
+extern const struct AudioOutputPlugin jack_output_plugin;
 
 #endif
diff --git a/src/output/NullOutputPlugin.cxx b/src/output/plugins/NullOutputPlugin.cxx
index e2eec9dbc..098f58926 100644
--- a/src/output/NullOutputPlugin.cxx
+++ b/src/output/plugins/NullOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,29 +19,25 @@
 
 #include "config.h"
 #include "NullOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "Timer.hxx"
-
-#include <assert.h>
+#include "../OutputAPI.hxx"
+#include "../Timer.hxx"
 
 struct NullOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	bool sync;
 
 	Timer *timer;
 
-	bool Initialize(const config_param &param, Error &error) {
-		return ao_base_init(&base, &null_output_plugin, param,
-				    error);
-	}
+	NullOutput()
+		:base(null_output_plugin) {}
 
-	void Deinitialize() {
-		ao_base_finish(&base);
+	bool Initialize(const config_param &param, Error &error) {
+		return base.Configure(param, error);
 	}
 };
 
-static struct audio_output *
+static AudioOutput *
 null_init(const config_param &param, Error &error)
 {
 	NullOutput *nd = new NullOutput();
@@ -57,16 +53,15 @@ null_init(const config_param &param, Error &error)
 }
 
 static void
-null_finish(struct audio_output *ao)
+null_finish(AudioOutput *ao)
 {
 	NullOutput *nd = (NullOutput *)ao;
 
-	nd->Deinitialize();
 	delete nd;
 }
 
 static bool
-null_open(struct audio_output *ao, AudioFormat &audio_format,
+null_open(AudioOutput *ao, AudioFormat &audio_format,
 	  gcc_unused Error &error)
 {
 	NullOutput *nd = (NullOutput *)ao;
@@ -78,7 +73,7 @@ null_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-null_close(struct audio_output *ao)
+null_close(AudioOutput *ao)
 {
 	NullOutput *nd = (NullOutput *)ao;
 
@@ -87,7 +82,7 @@ null_close(struct audio_output *ao)
 }
 
 static unsigned
-null_delay(struct audio_output *ao)
+null_delay(AudioOutput *ao)
 {
 	NullOutput *nd = (NullOutput *)ao;
 
@@ -97,7 +92,7 @@ null_delay(struct audio_output *ao)
 }
 
 static size_t
-null_play(struct audio_output *ao, gcc_unused const void *chunk, size_t size,
+null_play(AudioOutput *ao, gcc_unused const void *chunk, size_t size,
 	  gcc_unused Error &error)
 {
 	NullOutput *nd = (NullOutput *)ao;
@@ -114,7 +109,7 @@ null_play(struct audio_output *ao, gcc_unused const void *chunk, size_t size,
 }
 
 static void
-null_cancel(struct audio_output *ao)
+null_cancel(AudioOutput *ao)
 {
 	NullOutput *nd = (NullOutput *)ao;
 
@@ -124,7 +119,7 @@ null_cancel(struct audio_output *ao)
 	nd->timer->Reset();
 }
 
-const struct audio_output_plugin null_output_plugin = {
+const struct AudioOutputPlugin null_output_plugin = {
 	"null",
 	nullptr,
 	null_init,
diff --git a/src/output/NullOutputPlugin.hxx b/src/output/plugins/NullOutputPlugin.hxx
index a58f1cb13..f25f5b9f3 100644
--- a/src/output/NullOutputPlugin.hxx
+++ b/src/output/plugins/NullOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_NULL_OUTPUT_PLUGIN_HXX
 #define MPD_NULL_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin null_output_plugin;
+extern const struct AudioOutputPlugin null_output_plugin;
 
 #endif
diff --git a/src/output/OSXOutputPlugin.cxx b/src/output/plugins/OSXOutputPlugin.cxx
index 97ebae056..13ac7b35e 100644
--- a/src/output/OSXOutputPlugin.cxx
+++ b/src/output/plugins/OSXOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,8 +19,8 @@
 
 #include "config.h"
 #include "OSXOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "util/fifo_buffer.h"
+#include "../OutputAPI.hxx"
+#include "util/DynamicFifoBuffer.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "thread/Mutex.hxx"
@@ -33,7 +33,7 @@
 #include <CoreServices/CoreServices.h>
 
 struct OSXOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	/* configuration settings */
 	OSType component_subtype;
@@ -44,7 +44,10 @@ struct OSXOutput {
 	Mutex mutex;
 	Cond condition;
 
-	struct fifo_buffer *buffer;
+	DynamicFifoBuffer<uint8_t> *buffer;
+
+	OSXOutput()
+		:base(osx_output_plugin) {}
 };
 
 static constexpr Domain osx_output_domain("osx_output");
@@ -72,16 +75,16 @@ osx_output_configure(OSXOutput *oo, const config_param &param)
 	}
 	else {
 		oo->component_subtype = kAudioUnitSubType_HALOutput;
-		/* XXX am I supposed to g_strdup() this? */
+		/* XXX am I supposed to strdup() this? */
 		oo->device_name = device;
 	}
 }
 
-static struct audio_output *
+static AudioOutput *
 osx_output_init(const config_param &param, Error &error)
 {
 	OSXOutput *oo = new OSXOutput();
-	if (!ao_base_init(&oo->base, &osx_output_plugin, param, error)) {
+	if (!oo->base.Configure(param, error)) {
 		delete oo;
 		return NULL;
 	}
@@ -92,7 +95,7 @@ osx_output_init(const config_param &param, Error &error)
 }
 
 static void
-osx_output_finish(struct audio_output *ao)
+osx_output_finish(AudioOutput *ao)
 {
 	OSXOutput *oo = (OSXOutput *)ao;
 
@@ -207,22 +210,19 @@ osx_render(void *vdata,
 
 	od->mutex.lock();
 
-	size_t nbytes;
-	const void *src = fifo_buffer_read(od->buffer, &nbytes);
-
-	if (src != NULL) {
-		if (nbytes > buffer_size)
-			nbytes = buffer_size;
+	auto src = od->buffer->Read();
+	if (!src.IsEmpty()) {
+		if (src.size > buffer_size)
+			src.size = buffer_size;
 
-		memcpy(buffer->mData, src, nbytes);
-		fifo_buffer_consume(od->buffer, nbytes);
-	} else
-		nbytes = 0;
+		memcpy(buffer->mData, src.data, src.size);
+		od->buffer->Consume(src.size);
+	}
 
 	od->condition.signal();
 	od->mutex.unlock();
 
-	buffer->mDataByteSize = nbytes;
+	buffer->mDataByteSize = src.size;
 
 	unsigned i;
 	for (i = 1; i < buffer_list->mNumberBuffers; ++i) {
@@ -234,7 +234,7 @@ osx_render(void *vdata,
 }
 
 static bool
-osx_output_enable(struct audio_output *ao, Error &error)
+osx_output_enable(AudioOutput *ao, Error &error)
 {
 	OSXOutput *oo = (OSXOutput *)ao;
 
@@ -285,7 +285,7 @@ osx_output_enable(struct audio_output *ao, Error &error)
 }
 
 static void
-osx_output_disable(struct audio_output *ao)
+osx_output_disable(AudioOutput *ao)
 {
 	OSXOutput *oo = (OSXOutput *)ao;
 
@@ -293,27 +293,27 @@ osx_output_disable(struct audio_output *ao)
 }
 
 static void
-osx_output_cancel(struct audio_output *ao)
+osx_output_cancel(AudioOutput *ao)
 {
 	OSXOutput *od = (OSXOutput *)ao;
 
 	const ScopeLock protect(od->mutex);
-	fifo_buffer_clear(od->buffer);
+	od->buffer->Clear();
 }
 
 static void
-osx_output_close(struct audio_output *ao)
+osx_output_close(AudioOutput *ao)
 {
 	OSXOutput *od = (OSXOutput *)ao;
 
 	AudioOutputUnitStop(od->au);
 	AudioUnitUninitialize(od->au);
 
-	fifo_buffer_free(od->buffer);
+	delete od->buffer;
 }
 
 static bool
-osx_output_open(struct audio_output *ao, AudioFormat &audio_format,
+osx_output_open(AudioOutput *ao, AudioFormat &audio_format,
 		Error &error)
 {
 	OSXOutput *od = (OSXOutput *)ao;
@@ -370,8 +370,8 @@ osx_output_open(struct audio_output *ao, AudioFormat &audio_format,
 	}
 
 	/* create a buffer of 1s */
-	od->buffer = fifo_buffer_new(audio_format.sample_rate *
-				     audio_format.GetFrameSize());
+	od->buffer = new DynamicFifoBuffer<uint8_t>(audio_format.sample_rate *
+						    audio_format.GetFrameSize());
 
 	status = AudioOutputUnitStart(od->au);
 	if (status != 0) {
@@ -386,35 +386,33 @@ osx_output_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static size_t
-osx_output_play(struct audio_output *ao, const void *chunk, size_t size,
+osx_output_play(AudioOutput *ao, const void *chunk, size_t size,
 		gcc_unused Error &error)
 {
 	OSXOutput *od = (OSXOutput *)ao;
 
 	const ScopeLock protect(od->mutex);
 
-	void *dest;
-	size_t max_length;
-
+	DynamicFifoBuffer<uint8_t>::Range dest;
 	while (true) {
-		dest = fifo_buffer_write(od->buffer, &max_length);
-		if (dest != NULL)
+		dest = od->buffer->Write();
+		if (!dest.IsEmpty())
 			break;
 
 		/* wait for some free space in the buffer */
 		od->condition.wait(od->mutex);
 	}
 
-	if (size > max_length)
-		size = max_length;
+	if (size > dest.size)
+		size = dest.size;
 
-	memcpy(dest, chunk, size);
-	fifo_buffer_append(od->buffer, size);
+	memcpy(dest.data, chunk, size);
+	od->buffer->Append(size);
 
 	return size;
 }
 
-const struct audio_output_plugin osx_output_plugin = {
+const struct AudioOutputPlugin osx_output_plugin = {
 	"osx",
 	osx_output_test_default_device,
 	osx_output_init,
diff --git a/src/output/OSXOutputPlugin.hxx b/src/output/plugins/OSXOutputPlugin.hxx
index 2a4172880..d7aed40b6 100644
--- a/src/output/OSXOutputPlugin.hxx
+++ b/src/output/plugins/OSXOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_OSX_OUTPUT_PLUGIN_HXX
 #define MPD_OSX_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin osx_output_plugin;
+extern const struct AudioOutputPlugin osx_output_plugin;
 
 #endif
diff --git a/src/output/OpenALOutputPlugin.cxx b/src/output/plugins/OpenALOutputPlugin.cxx
index 268cf17cc..2f095c0a4 100644
--- a/src/output/OpenALOutputPlugin.cxx
+++ b/src/output/plugins/OpenALOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,11 +19,11 @@
 
 #include "config.h"
 #include "OpenALOutputPlugin.hxx"
-#include "OutputAPI.hxx"
+#include "../OutputAPI.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 
-#include <glib.h>
+#include <unistd.h>
 
 #ifndef __APPLE__
 #include <AL/al.h>
@@ -37,7 +37,7 @@
 #define NUM_BUFFERS 16
 
 struct OpenALOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	const char *device_name;
 	ALCdevice *device;
@@ -48,13 +48,11 @@ struct OpenALOutput {
 	ALenum format;
 	ALuint frequency;
 
-	bool Initialize(const config_param &param, Error &error_r) {
-		return ao_base_init(&base, &openal_output_plugin, param,
-				    error_r);
-	}
+	OpenALOutput()
+		:base(openal_output_plugin) {}
 
-	void Deinitialize() {
-		ao_base_finish(&base);
+	bool Initialize(const config_param &param, Error &error_r) {
+		return base.Configure(param, error_r);
 	}
 };
 
@@ -133,7 +131,7 @@ openal_setup_context(OpenALOutput *od, Error &error)
 	return true;
 }
 
-static struct audio_output *
+static AudioOutput *
 openal_init(const config_param &param, Error &error)
 {
 	const char *device_name = param.GetBlockValue("device");
@@ -153,16 +151,15 @@ openal_init(const config_param &param, Error &error)
 }
 
 static void
-openal_finish(struct audio_output *ao)
+openal_finish(AudioOutput *ao)
 {
 	OpenALOutput *od = (OpenALOutput *)ao;
 
-	od->Deinitialize();
 	delete od;
 }
 
 static bool
-openal_open(struct audio_output *ao, AudioFormat &audio_format,
+openal_open(AudioOutput *ao, AudioFormat &audio_format,
 	    Error &error)
 {
 	OpenALOutput *od = (OpenALOutput *)ao;
@@ -196,7 +193,7 @@ openal_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-openal_close(struct audio_output *ao)
+openal_close(AudioOutput *ao)
 {
 	OpenALOutput *od = (OpenALOutput *)ao;
 
@@ -208,7 +205,7 @@ openal_close(struct audio_output *ao)
 }
 
 static unsigned
-openal_delay(struct audio_output *ao)
+openal_delay(AudioOutput *ao)
 {
 	OpenALOutput *od = (OpenALOutput *)ao;
 
@@ -221,7 +218,7 @@ openal_delay(struct audio_output *ao)
 }
 
 static size_t
-openal_play(struct audio_output *ao, const void *chunk, size_t size,
+openal_play(AudioOutput *ao, const void *chunk, size_t size,
 	    gcc_unused Error &error)
 {
 	OpenALOutput *od = (OpenALOutput *)ao;
@@ -238,7 +235,7 @@ openal_play(struct audio_output *ao, const void *chunk, size_t size,
 	} else {
 		/* wait for processed buffer */
 		while (!openal_has_processed(od))
-			g_usleep(10);
+			usleep(10);
 
 		alSourceUnqueueBuffers(od->source, 1, &buffer);
 	}
@@ -253,7 +250,7 @@ openal_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 static void
-openal_cancel(struct audio_output *ao)
+openal_cancel(AudioOutput *ao)
 {
 	OpenALOutput *od = (OpenALOutput *)ao;
 
@@ -266,7 +263,7 @@ openal_cancel(struct audio_output *ao)
 	od->filled = 0;
 }
 
-const struct audio_output_plugin openal_output_plugin = {
+const struct AudioOutputPlugin openal_output_plugin = {
 	"openal",
 	nullptr,
 	openal_init,
diff --git a/src/output/OpenALOutputPlugin.hxx b/src/output/plugins/OpenALOutputPlugin.hxx
index e1ebf3d4f..a27e6b53c 100644
--- a/src/output/OpenALOutputPlugin.hxx
+++ b/src/output/plugins/OpenALOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_OPENAL_OUTPUT_PLUGIN_HXX
 #define MPD_OPENAL_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin openal_output_plugin;
+extern const struct AudioOutputPlugin openal_output_plugin;
 
 #endif
diff --git a/src/output/OssOutputPlugin.cxx b/src/output/plugins/OssOutputPlugin.cxx
index cdde6d562..39d87fc35 100644
--- a/src/output/OssOutputPlugin.cxx
+++ b/src/output/plugins/OssOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,9 +19,10 @@
 
 #include "config.h"
 #include "OssOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "MixerList.hxx"
+#include "../OutputAPI.hxx"
+#include "mixer/MixerList.hxx"
 #include "system/fd_util.h"
+#include "util/ConstBuffer.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "util/Macros.hxx"
@@ -57,7 +58,7 @@
 #endif
 
 struct OssOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 #ifdef AFMT_S24_PACKED
 	Manual<PcmExport> pcm_export;
@@ -78,15 +79,12 @@ struct OssOutput {
 	 */
 	int oss_format;
 
-	OssOutput():fd(-1), device(nullptr) {}
+	OssOutput()
+		:base(oss_output_plugin),
+		 fd(-1), device(nullptr) {}
 
 	bool Initialize(const config_param &param, Error &error_r) {
-		return ao_base_init(&base, &oss_output_plugin, param,
-				    error_r);
-	}
-
-	void Deinitialize() {
-		ao_base_finish(&base);
+		return base.Configure(param, error_r);
 	}
 };
 
@@ -149,7 +147,7 @@ oss_output_test_default_device(void)
 	return false;
 }
 
-static struct audio_output *
+static AudioOutput *
 oss_open_default(Error &error)
 {
 	int err[ARRAY_SIZE(default_devices)];
@@ -199,7 +197,7 @@ oss_open_default(Error &error)
 	return NULL;
 }
 
-static struct audio_output *
+static AudioOutput *
 oss_output_init(const config_param &param, Error &error)
 {
 	const char *device = param.GetBlockValue("device");
@@ -218,18 +216,17 @@ oss_output_init(const config_param &param, Error &error)
 }
 
 static void
-oss_output_finish(struct audio_output *ao)
+oss_output_finish(AudioOutput *ao)
 {
 	OssOutput *od = (OssOutput *)ao;
 
-	ao_base_finish(&od->base);
 	delete od;
 }
 
 #ifdef AFMT_S24_PACKED
 
 static bool
-oss_output_enable(struct audio_output *ao, gcc_unused Error &error)
+oss_output_enable(AudioOutput *ao, gcc_unused Error &error)
 {
 	OssOutput *od = (OssOutput *)ao;
 
@@ -238,7 +235,7 @@ oss_output_enable(struct audio_output *ao, gcc_unused Error &error)
 }
 
 static void
-oss_output_disable(struct audio_output *ao)
+oss_output_disable(AudioOutput *ao)
 {
 	OssOutput *od = (OssOutput *)ao;
 
@@ -680,7 +677,7 @@ oss_reopen(OssOutput *od, Error &error)
 }
 
 static bool
-oss_output_open(struct audio_output *ao, AudioFormat &audio_format,
+oss_output_open(AudioOutput *ao, AudioFormat &audio_format,
 		Error &error)
 {
 	OssOutput *od = (OssOutput *)ao;
@@ -702,7 +699,7 @@ oss_output_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-oss_output_close(struct audio_output *ao)
+oss_output_close(AudioOutput *ao)
 {
 	OssOutput *od = (OssOutput *)ao;
 
@@ -710,7 +707,7 @@ oss_output_close(struct audio_output *ao)
 }
 
 static void
-oss_output_cancel(struct audio_output *ao)
+oss_output_cancel(AudioOutput *ao)
 {
 	OssOutput *od = (OssOutput *)ao;
 
@@ -721,7 +718,7 @@ oss_output_cancel(struct audio_output *ao)
 }
 
 static size_t
-oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
+oss_output_play(AudioOutput *ao, const void *chunk, size_t size,
 		Error &error)
 {
 	OssOutput *od = (OssOutput *)ao;
@@ -734,7 +731,9 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		return 0;
 
 #ifdef AFMT_S24_PACKED
-	chunk = od->pcm_export->Export(chunk, size, size);
+	const auto e = od->pcm_export->Export({chunk, size});
+	chunk = e.data;
+	size = e.size;
 #endif
 
 	assert(size > 0);
@@ -755,7 +754,7 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	}
 }
 
-const struct audio_output_plugin oss_output_plugin = {
+const struct AudioOutputPlugin oss_output_plugin = {
 	"oss",
 	oss_output_test_default_device,
 	oss_output_init,
diff --git a/src/output/OssOutputPlugin.hxx b/src/output/plugins/OssOutputPlugin.hxx
index 6c5c9530b..f9970c8f0 100644
--- a/src/output/OssOutputPlugin.hxx
+++ b/src/output/plugins/OssOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_OSS_OUTPUT_PLUGIN_HXX
 #define MPD_OSS_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin oss_output_plugin;
+extern const struct AudioOutputPlugin oss_output_plugin;
 
 #endif
diff --git a/src/output/PipeOutputPlugin.cxx b/src/output/plugins/PipeOutputPlugin.cxx
index 34d615284..7a1f32258 100644
--- a/src/output/PipeOutputPlugin.cxx
+++ b/src/output/plugins/PipeOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,8 +19,8 @@
 
 #include "config.h"
 #include "PipeOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "ConfigError.hxx"
+#include "../OutputAPI.hxx"
+#include "config/ConfigError.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 
@@ -29,18 +29,16 @@
 #include <stdio.h>
 
 struct PipeOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	std::string cmd;
 	FILE *fh;
 
-	bool Initialize(const config_param &param, Error &error) {
-		return ao_base_init(&base, &pipe_output_plugin, param,
-				    error);
-	}
+	PipeOutput()
+		:base(pipe_output_plugin) {}
 
-	void Deinitialize() {
-		ao_base_finish(&base);
+	bool Initialize(const config_param &param, Error &error) {
+		return base.Configure(param, error);
 	}
 
 	bool Configure(const config_param &param, Error &error);
@@ -61,7 +59,7 @@ PipeOutput::Configure(const config_param &param, Error &error)
 	return true;
 }
 
-static struct audio_output *
+static AudioOutput *
 pipe_output_init(const config_param &param, Error &error)
 {
 	PipeOutput *pd = new PipeOutput();
@@ -72,7 +70,6 @@ pipe_output_init(const config_param &param, Error &error)
 	}
 
 	if (!pd->Configure(param, error)) {
-		pd->Deinitialize();
 		delete pd;
 		return nullptr;
 	}
@@ -81,16 +78,15 @@ pipe_output_init(const config_param &param, Error &error)
 }
 
 static void
-pipe_output_finish(struct audio_output *ao)
+pipe_output_finish(AudioOutput *ao)
 {
 	PipeOutput *pd = (PipeOutput *)ao;
 
-	pd->Deinitialize();
 	delete pd;
 }
 
 static bool
-pipe_output_open(struct audio_output *ao,
+pipe_output_open(AudioOutput *ao,
 		 gcc_unused AudioFormat &audio_format,
 		 Error &error)
 {
@@ -107,7 +103,7 @@ pipe_output_open(struct audio_output *ao,
 }
 
 static void
-pipe_output_close(struct audio_output *ao)
+pipe_output_close(AudioOutput *ao)
 {
 	PipeOutput *pd = (PipeOutput *)ao;
 
@@ -115,7 +111,7 @@ pipe_output_close(struct audio_output *ao)
 }
 
 static size_t
-pipe_output_play(struct audio_output *ao, const void *chunk, size_t size,
+pipe_output_play(AudioOutput *ao, const void *chunk, size_t size,
 		 Error &error)
 {
 	PipeOutput *pd = (PipeOutput *)ao;
@@ -128,7 +124,7 @@ pipe_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	return ret;
 }
 
-const struct audio_output_plugin pipe_output_plugin = {
+const struct AudioOutputPlugin pipe_output_plugin = {
 	"pipe",
 	nullptr,
 	pipe_output_init,
diff --git a/src/output/PipeOutputPlugin.hxx b/src/output/plugins/PipeOutputPlugin.hxx
index f0c29706b..bdaf2ecfd 100644
--- a/src/output/PipeOutputPlugin.hxx
+++ b/src/output/plugins/PipeOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_PIPE_OUTPUT_PLUGIN_HXX
 #define MPD_PIPE_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin pipe_output_plugin;
+extern const struct AudioOutputPlugin pipe_output_plugin;
 
 #endif
diff --git a/src/output/PulseOutputPlugin.cxx b/src/output/plugins/PulseOutputPlugin.cxx
index 1eece448a..120bad090 100644
--- a/src/output/PulseOutputPlugin.cxx
+++ b/src/output/plugins/PulseOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,15 +19,13 @@
 
 #include "config.h"
 #include "PulseOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "MixerList.hxx"
-#include "mixer/PulseMixerPlugin.hxx"
+#include "../OutputAPI.hxx"
+#include "mixer/MixerList.hxx"
+#include "mixer/plugins/PulseMixerPlugin.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "Log.hxx"
 
-#include <glib.h>
-
 #include <pulse/thread-mainloop.h>
 #include <pulse/context.h>
 #include <pulse/stream.h>
@@ -38,11 +36,12 @@
 
 #include <assert.h>
 #include <stddef.h>
+#include <stdlib.h>
 
 #define MPD_PULSE_NAME "Music Player Daemon"
 
 struct PulseOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	const char *name;
 	const char *server;
@@ -55,6 +54,9 @@ struct PulseOutput {
 	struct pa_stream *stream;
 
 	size_t writable;
+
+	PulseOutput()
+		:base(pulse_output_plugin) {}
 };
 
 static constexpr Domain pulse_output_domain("pulse_output");
@@ -67,70 +69,66 @@ SetError(Error &error, pa_context *context, const char *msg)
 }
 
 void
-pulse_output_lock(PulseOutput *po)
+pulse_output_lock(PulseOutput &po)
 {
-	pa_threaded_mainloop_lock(po->mainloop);
+	pa_threaded_mainloop_lock(po.mainloop);
 }
 
 void
-pulse_output_unlock(PulseOutput *po)
+pulse_output_unlock(PulseOutput &po)
 {
-	pa_threaded_mainloop_unlock(po->mainloop);
+	pa_threaded_mainloop_unlock(po.mainloop);
 }
 
 void
-pulse_output_set_mixer(PulseOutput *po, PulseMixer *pm)
+pulse_output_set_mixer(PulseOutput &po, PulseMixer &pm)
 {
-	assert(po != nullptr);
-	assert(po->mixer == nullptr);
-	assert(pm != nullptr);
+	assert(po.mixer == nullptr);
 
-	po->mixer = pm;
+	po.mixer = &pm;
 
-	if (po->mainloop == nullptr)
+	if (po.mainloop == nullptr)
 		return;
 
-	pa_threaded_mainloop_lock(po->mainloop);
+	pa_threaded_mainloop_lock(po.mainloop);
 
-	if (po->context != nullptr &&
-	    pa_context_get_state(po->context) == PA_CONTEXT_READY) {
-		pulse_mixer_on_connect(pm, po->context);
+	if (po.context != nullptr &&
+	    pa_context_get_state(po.context) == PA_CONTEXT_READY) {
+		pulse_mixer_on_connect(pm, po.context);
 
-		if (po->stream != nullptr &&
-		    pa_stream_get_state(po->stream) == PA_STREAM_READY)
-			pulse_mixer_on_change(pm, po->context, po->stream);
+		if (po.stream != nullptr &&
+		    pa_stream_get_state(po.stream) == PA_STREAM_READY)
+			pulse_mixer_on_change(pm, po.context, po.stream);
 	}
 
-	pa_threaded_mainloop_unlock(po->mainloop);
+	pa_threaded_mainloop_unlock(po.mainloop);
 }
 
 void
-pulse_output_clear_mixer(PulseOutput *po, gcc_unused PulseMixer *pm)
+pulse_output_clear_mixer(PulseOutput &po, gcc_unused PulseMixer &pm)
 {
-	assert(po != nullptr);
-	assert(pm != nullptr);
-	assert(po->mixer == pm);
+	assert(po.mixer == &pm);
 
-	po->mixer = nullptr;
+	po.mixer = nullptr;
 }
 
 bool
-pulse_output_set_volume(PulseOutput *po, const pa_cvolume *volume,
+pulse_output_set_volume(PulseOutput &po, const pa_cvolume *volume,
 			Error &error)
 {
 	pa_operation *o;
 
-	if (po->context == nullptr || po->stream == nullptr ||
-	    pa_stream_get_state(po->stream) != PA_STREAM_READY) {
+	if (po.context == nullptr || po.stream == nullptr ||
+	    pa_stream_get_state(po.stream) != PA_STREAM_READY) {
 		error.Set(pulse_output_domain, "disconnected");
 		return false;
 	}
 
-	o = pa_context_set_sink_input_volume(po->context,
-					     pa_stream_get_index(po->stream),
+	o = pa_context_set_sink_input_volume(po.context,
+					     pa_stream_get_index(po.stream),
 					     volume, nullptr, nullptr);
 	if (o == nullptr) {
-		SetError(error, po->context,
+		SetError(error, po.context,
 			 "failed to set PulseAudio volume");
 		return false;
 	}
@@ -150,16 +148,13 @@ static bool
 pulse_wait_for_operation(struct pa_threaded_mainloop *mainloop,
 			 struct pa_operation *operation)
 {
-	pa_operation_state_t state;
-
 	assert(mainloop != nullptr);
 	assert(operation != nullptr);
 
-	state = pa_operation_get_state(operation);
-	while (state == PA_OPERATION_RUNNING) {
+	pa_operation_state_t state;
+	while ((state = pa_operation_get_state(operation))
+	       == PA_OPERATION_RUNNING)
 		pa_threaded_mainloop_wait(mainloop);
-		state = pa_operation_get_state(operation);
-	}
 
 	pa_operation_unref(operation);
 
@@ -187,7 +182,7 @@ pulse_output_context_state_cb(struct pa_context *context, void *userdata)
 	switch (pa_context_get_state(context)) {
 	case PA_CONTEXT_READY:
 		if (po->mixer != nullptr)
-			pulse_mixer_on_connect(po->mixer, context);
+			pulse_mixer_on_connect(*po->mixer, context);
 
 		pa_threaded_mainloop_signal(po->mainloop, 0);
 		break;
@@ -195,7 +190,7 @@ pulse_output_context_state_cb(struct pa_context *context, void *userdata)
 	case PA_CONTEXT_TERMINATED:
 	case PA_CONTEXT_FAILED:
 		if (po->mixer != nullptr)
-			pulse_mixer_on_disconnect(po->mixer);
+			pulse_mixer_on_disconnect(*po->mixer);
 
 		/* the caller thread might be waiting for these
 		   states */
@@ -228,7 +223,7 @@ pulse_output_subscribe_cb(pa_context *context,
 	    idx == pa_stream_get_index(po->stream) &&
 	    (type == PA_SUBSCRIPTION_EVENT_NEW ||
 	     type == PA_SUBSCRIPTION_EVENT_CHANGE))
-		pulse_mixer_on_change(po->mixer, context, po->stream);
+		pulse_mixer_on_change(*po->mixer, context, po->stream);
 }
 
 /**
@@ -323,15 +318,16 @@ pulse_output_setup_context(PulseOutput *po, Error &error)
 	return true;
 }
 
-static struct audio_output *
+static AudioOutput *
 pulse_output_init(const config_param &param, Error &error)
 {
 	PulseOutput *po;
 
-	g_setenv("PULSE_PROP_media.role", "music", true);
+	setenv("PULSE_PROP_media.role", "music", true);
+	setenv("PULSE_PROP_application.icon_name", "mpd", true);
 
 	po = new PulseOutput();
-	if (!ao_base_init(&po->base, &pulse_output_plugin, param, error)) {
+	if (!po->base.Configure(param, error)) {
 		delete po;
 		return nullptr;
 	}
@@ -349,16 +345,15 @@ pulse_output_init(const config_param &param, Error &error)
 }
 
 static void
-pulse_output_finish(struct audio_output *ao)
+pulse_output_finish(AudioOutput *ao)
 {
 	PulseOutput *po = (PulseOutput *)ao;
 
-	ao_base_finish(&po->base);
 	delete po;
 }
 
 static bool
-pulse_output_enable(struct audio_output *ao, Error &error)
+pulse_output_enable(AudioOutput *ao, Error &error)
 {
 	PulseOutput *po = (PulseOutput *)ao;
 
@@ -402,7 +397,7 @@ pulse_output_enable(struct audio_output *ao, Error &error)
 }
 
 static void
-pulse_output_disable(struct audio_output *ao)
+pulse_output_disable(AudioOutput *ao)
 {
 	PulseOutput *po = (PulseOutput *)ao;
 
@@ -483,7 +478,7 @@ pulse_output_stream_state_cb(pa_stream *stream, void *userdata)
 	switch (pa_stream_get_state(stream)) {
 	case PA_STREAM_READY:
 		if (po->mixer != nullptr)
-			pulse_mixer_on_change(po->mixer, po->context, stream);
+			pulse_mixer_on_change(*po->mixer, po->context, stream);
 
 		pa_threaded_mainloop_signal(po->mainloop, 0);
 		break;
@@ -491,7 +486,7 @@ pulse_output_stream_state_cb(pa_stream *stream, void *userdata)
 	case PA_STREAM_FAILED:
 	case PA_STREAM_TERMINATED:
 		if (po->mixer != nullptr)
-			pulse_mixer_on_disconnect(po->mixer);
+			pulse_mixer_on_disconnect(*po->mixer);
 
 		pa_threaded_mainloop_signal(po->mainloop, 0);
 		break;
@@ -546,7 +541,7 @@ pulse_output_setup_stream(PulseOutput *po, const pa_sample_spec *ss,
 }
 
 static bool
-pulse_output_open(struct audio_output *ao, AudioFormat &audio_format,
+pulse_output_open(AudioOutput *ao, AudioFormat &audio_format,
 		  Error &error)
 {
 	PulseOutput *po = (PulseOutput *)ao;
@@ -614,7 +609,7 @@ pulse_output_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-pulse_output_close(struct audio_output *ao)
+pulse_output_close(AudioOutput *ao)
 {
 	PulseOutput *po = (PulseOutput *)ao;
 	pa_operation *o;
@@ -700,7 +695,7 @@ pulse_output_stream_pause(PulseOutput *po, bool pause,
 }
 
 static unsigned
-pulse_output_delay(struct audio_output *ao)
+pulse_output_delay(AudioOutput *ao)
 {
 	PulseOutput *po = (PulseOutput *)ao;
 	unsigned result = 0;
@@ -718,7 +713,7 @@ pulse_output_delay(struct audio_output *ao)
 }
 
 static size_t
-pulse_output_play(struct audio_output *ao, const void *chunk, size_t size,
+pulse_output_play(AudioOutput *ao, const void *chunk, size_t size,
 		  Error &error)
 {
 	PulseOutput *po = (PulseOutput *)ao;
@@ -783,7 +778,7 @@ pulse_output_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 static void
-pulse_output_cancel(struct audio_output *ao)
+pulse_output_cancel(AudioOutput *ao)
 {
 	PulseOutput *po = (PulseOutput *)ao;
 	pa_operation *o;
@@ -816,7 +811,7 @@ pulse_output_cancel(struct audio_output *ao)
 }
 
 static bool
-pulse_output_pause(struct audio_output *ao)
+pulse_output_pause(AudioOutput *ao)
 {
 	PulseOutput *po = (PulseOutput *)ao;
 
@@ -867,7 +862,7 @@ pulse_output_test_default_device(void)
 	return success;
 }
 
-const struct audio_output_plugin pulse_output_plugin = {
+const struct AudioOutputPlugin pulse_output_plugin = {
 	"pulse",
 	pulse_output_test_default_device,
 	pulse_output_init,
diff --git a/src/output/PulseOutputPlugin.hxx b/src/output/plugins/PulseOutputPlugin.hxx
index 0ed8404bc..9219780a5 100644
--- a/src/output/PulseOutputPlugin.hxx
+++ b/src/output/plugins/PulseOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -21,26 +21,26 @@
 #define MPD_PULSE_OUTPUT_PLUGIN_HXX
 
 struct PulseOutput;
-struct PulseMixer;
+class PulseMixer;
 struct pa_cvolume;
 class Error;
 
-extern const struct audio_output_plugin pulse_output_plugin;
+extern const struct AudioOutputPlugin pulse_output_plugin;
 
 void
-pulse_output_lock(PulseOutput *po);
+pulse_output_lock(PulseOutput &po);
 
 void
-pulse_output_unlock(PulseOutput *po);
+pulse_output_unlock(PulseOutput &po);
 
 void
-pulse_output_set_mixer(PulseOutput *po, PulseMixer *pm);
+pulse_output_set_mixer(PulseOutput &po, PulseMixer &pm);
 
 void
-pulse_output_clear_mixer(PulseOutput *po, PulseMixer *pm);
+pulse_output_clear_mixer(PulseOutput &po, PulseMixer &pm);
 
 bool
-pulse_output_set_volume(PulseOutput *po,
-			const struct pa_cvolume *volume, Error &error);
+pulse_output_set_volume(PulseOutput &po,
+			const pa_cvolume *volume, Error &error);
 
 #endif
diff --git a/src/output/RecorderOutputPlugin.cxx b/src/output/plugins/RecorderOutputPlugin.cxx
index 9a7eba01f..87e23f55a 100644
--- a/src/output/RecorderOutputPlugin.cxx
+++ b/src/output/plugins/RecorderOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,10 +19,10 @@
 
 #include "config.h"
 #include "RecorderOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "EncoderPlugin.hxx"
-#include "EncoderList.hxx"
-#include "ConfigError.hxx"
+#include "../OutputAPI.hxx"
+#include "encoder/EncoderPlugin.hxx"
+#include "encoder/EncoderList.hxx"
+#include "config/ConfigError.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "system/fd_util.h"
@@ -35,7 +35,7 @@
 #include <errno.h>
 
 struct RecorderOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	/**
 	 * The configured encoder plugin.
@@ -57,13 +57,11 @@ struct RecorderOutput {
 	 */
 	char buffer[32768];
 
-	bool Initialize(const config_param &param, Error &error_r) {
-		return ao_base_init(&base, &recorder_output_plugin, param,
-				    error_r);
-	}
+	RecorderOutput()
+		:base(recorder_output_plugin) {}
 
-	void Deinitialize() {
-		ao_base_finish(&base);
+	bool Initialize(const config_param &param, Error &error_r) {
+		return base.Configure(param, error_r);
 	}
 
 	bool Configure(const config_param &param, Error &error);
@@ -107,7 +105,7 @@ RecorderOutput::Configure(const config_param &param, Error &error)
 	return true;
 }
 
-static audio_output *
+static AudioOutput *
 recorder_output_init(const config_param &param, Error &error)
 {
 	RecorderOutput *recorder = new RecorderOutput();
@@ -118,7 +116,6 @@ recorder_output_init(const config_param &param, Error &error)
 	}
 
 	if (!recorder->Configure(param, error)) {
-		recorder->Deinitialize();
 		delete recorder;
 		return nullptr;
 	}
@@ -127,12 +124,11 @@ recorder_output_init(const config_param &param, Error &error)
 }
 
 static void
-recorder_output_finish(struct audio_output *ao)
+recorder_output_finish(AudioOutput *ao)
 {
 	RecorderOutput *recorder = (RecorderOutput *)ao;
 
 	encoder_finish(recorder->encoder);
-	recorder->Deinitialize();
 	delete recorder;
 }
 
@@ -181,7 +177,7 @@ RecorderOutput::EncoderToFile(Error &error)
 }
 
 static bool
-recorder_output_open(struct audio_output *ao,
+recorder_output_open(AudioOutput *ao,
 		     AudioFormat &audio_format,
 		     Error &error)
 {
@@ -216,7 +212,7 @@ recorder_output_open(struct audio_output *ao,
 }
 
 static void
-recorder_output_close(struct audio_output *ao)
+recorder_output_close(AudioOutput *ao)
 {
 	RecorderOutput *recorder = (RecorderOutput *)ao;
 
@@ -233,7 +229,7 @@ recorder_output_close(struct audio_output *ao)
 }
 
 static size_t
-recorder_output_play(struct audio_output *ao, const void *chunk, size_t size,
+recorder_output_play(AudioOutput *ao, const void *chunk, size_t size,
 		     Error &error)
 {
 	RecorderOutput *recorder = (RecorderOutput *)ao;
@@ -243,7 +239,7 @@ recorder_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		? size : 0;
 }
 
-const struct audio_output_plugin recorder_output_plugin = {
+const struct AudioOutputPlugin recorder_output_plugin = {
 	"recorder",
 	nullptr,
 	recorder_output_init,
diff --git a/src/output/RecorderOutputPlugin.hxx b/src/output/plugins/RecorderOutputPlugin.hxx
index a27f51e23..ea1044e0f 100644
--- a/src/output/RecorderOutputPlugin.hxx
+++ b/src/output/plugins/RecorderOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_RECORDER_OUTPUT_PLUGIN_HXX
 #define MPD_RECORDER_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin recorder_output_plugin;
+extern const struct AudioOutputPlugin recorder_output_plugin;
 
 #endif
diff --git a/src/output/RoarOutputPlugin.cxx b/src/output/plugins/RoarOutputPlugin.cxx
index 895a165d1..04836cc46 100644
--- a/src/output/RoarOutputPlugin.cxx
+++ b/src/output/plugins/RoarOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * Copyright (C) 2003-2014 The Music Player Daemon Project
  * Copyright (C) 2010-2011 Philipp 'ph3-der-loewe' Schafft
  * Copyright (C) 2010-2011 Hans-Kristian 'maister' Arntzen
  *
@@ -20,8 +20,8 @@
 
 #include "config.h"
 #include "RoarOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "MixerList.hxx"
+#include "../OutputAPI.hxx"
+#include "mixer/MixerList.hxx"
 #include "thread/Mutex.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
@@ -36,7 +36,7 @@
 #undef new
 
 class RoarOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	std::string host, name;
 
@@ -50,19 +50,15 @@ class RoarOutput {
 
 public:
 	RoarOutput()
-		:err(ROAR_ERROR_NONE) {}
+		:base(roar_output_plugin),
+		 err(ROAR_ERROR_NONE) {}
 
-	operator audio_output *() {
+	operator AudioOutput *() {
 		return &base;
 	}
 
 	bool Initialize(const config_param &param, Error &error) {
-		return ao_base_init(&base, &roar_output_plugin, param,
-				    error);
-	}
-
-	void Deinitialize() {
-		ao_base_finish(&base);
+		return base.Configure(param, error);
 	}
 
 	void Configure(const config_param &param);
@@ -97,9 +93,9 @@ RoarOutput::GetVolume() const
 }
 
 int
-roar_output_get_volume(RoarOutput *roar)
+roar_output_get_volume(RoarOutput &roar)
 {
-	return roar->GetVolume();
+	return roar.GetVolume();
 }
 
 bool
@@ -119,9 +115,9 @@ RoarOutput::SetVolume(unsigned volume)
 }
 
 bool
-roar_output_set_volume(RoarOutput *roar, unsigned volume)
+roar_output_set_volume(RoarOutput &roar, unsigned volume)
 {
-	return roar->SetVolume(volume);
+	return roar.SetVolume(volume);
 }
 
 inline void
@@ -136,7 +132,7 @@ RoarOutput::Configure(const config_param &param)
 		: ROAR_ROLE_MUSIC;
 }
 
-static struct audio_output *
+static AudioOutput *
 roar_init(const config_param &param, Error &error)
 {
 	RoarOutput *self = new RoarOutput();
@@ -151,11 +147,10 @@ roar_init(const config_param &param, Error &error)
 }
 
 static void
-roar_finish(struct audio_output *ao)
+roar_finish(AudioOutput *ao)
 {
 	RoarOutput *self = (RoarOutput *)ao;
 
-	self->Deinitialize();
 	delete self;
 }
 
@@ -227,7 +222,7 @@ RoarOutput::Open(AudioFormat &audio_format, Error &error)
 }
 
 static bool
-roar_open(struct audio_output *ao, AudioFormat &audio_format, Error &error)
+roar_open(AudioOutput *ao, AudioFormat &audio_format, Error &error)
 {
 	RoarOutput *self = (RoarOutput *)ao;
 
@@ -248,7 +243,7 @@ RoarOutput::Close()
 }
 
 static void
-roar_close(struct audio_output *ao)
+roar_close(AudioOutput *ao)
 {
 	RoarOutput *self = (RoarOutput *)ao;
 	self->Close();
@@ -283,7 +278,7 @@ RoarOutput::Cancel()
 }
 
 static void
-roar_cancel(struct audio_output *ao)
+roar_cancel(AudioOutput *ao)
 {
 	RoarOutput *self = (RoarOutput *)ao;
 
@@ -308,7 +303,7 @@ RoarOutput::Play(const void *chunk, size_t size, Error &error)
 }
 
 static size_t
-roar_play(struct audio_output *ao, const void *chunk, size_t size,
+roar_play(AudioOutput *ao, const void *chunk, size_t size,
 	  Error &error)
 {
 	RoarOutput *self = (RoarOutput *)ao;
@@ -368,31 +363,37 @@ RoarOutput::SendTag(const Tag &tag)
 
 	const ScopeLock protect(mutex);
 
-	size_t cnt = 1;
+	size_t cnt = 0;
 	struct roar_keyval vals[32];
 	char uuid_buf[32][64];
 
 	char timebuf[16];
-	snprintf(timebuf, sizeof(timebuf), "%02d:%02d:%02d",
-		 tag.time / 3600, (tag.time % 3600) / 60, tag.time % 60);
+	if (!tag.duration.IsNegative()) {
+		const unsigned seconds = tag.duration.ToS();
+		snprintf(timebuf, sizeof(timebuf), "%02d:%02d:%02d",
+			 seconds / 3600, (seconds % 3600) / 60, seconds % 60);
+
+		vals[cnt].key = const_cast<char *>("LENGTH");
+		vals[cnt].value = timebuf;
+		++cnt;
+	}
 
-	vals[0].key = const_cast<char *>("LENGTH");
-	vals[0].value = timebuf;
+	for (const auto &item : tag) {
+		if (cnt >= 32)
+			break;
 
-	for (unsigned i = 0; i < tag.num_items && cnt < 32; i++)
-	{
 		bool is_uuid = false;
-		const char *key = roar_tag_convert(tag.items[i]->type,
+		const char *key = roar_tag_convert(item.type,
 						   &is_uuid);
 		if (key != nullptr) {
 			vals[cnt].key = const_cast<char *>(key);
 
 			if (is_uuid) {
 				snprintf(uuid_buf[cnt], sizeof(uuid_buf[0]), "{UUID}%s",
-					 tag.items[i]->value);
+					 item.value);
 				vals[cnt].value = uuid_buf[cnt];
 			} else {
-				vals[cnt].value = tag.items[i]->value;
+				vals[cnt].value = const_cast<char *>(item.value);
 			}
 
 			cnt++;
@@ -403,13 +404,13 @@ RoarOutput::SendTag(const Tag &tag)
 }
 
 static void
-roar_send_tag(struct audio_output *ao, const Tag *meta)
+roar_send_tag(AudioOutput *ao, const Tag *meta)
 {
 	RoarOutput *self = (RoarOutput *)ao;
 	self->SendTag(*meta);
 }
 
-const struct audio_output_plugin roar_output_plugin = {
+const struct AudioOutputPlugin roar_output_plugin = {
 	"roar",
 	nullptr,
 	roar_init,
diff --git a/src/output/RoarOutputPlugin.hxx b/src/output/plugins/RoarOutputPlugin.hxx
index 04949e421..5f5a9246e 100644
--- a/src/output/RoarOutputPlugin.hxx
+++ b/src/output/plugins/RoarOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -22,12 +22,12 @@
 
 class RoarOutput;
 
-extern const struct audio_output_plugin roar_output_plugin;
+extern const struct AudioOutputPlugin roar_output_plugin;
 
 int
-roar_output_get_volume(RoarOutput *roar);
+roar_output_get_volume(RoarOutput &roar);
 
 bool
-roar_output_set_volume(RoarOutput *roar, unsigned volume);
+roar_output_set_volume(RoarOutput &roar, unsigned volume);
 
 #endif
diff --git a/src/output/ShoutOutputPlugin.cxx b/src/output/plugins/ShoutOutputPlugin.cxx
index 19f2b61cd..0341e1cf7 100644
--- a/src/output/ShoutOutputPlugin.cxx
+++ b/src/output/plugins/ShoutOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,17 +19,16 @@
 
 #include "config.h"
 #include "ShoutOutputPlugin.hxx"
-#include "OutputAPI.hxx"
-#include "EncoderPlugin.hxx"
-#include "EncoderList.hxx"
-#include "ConfigError.hxx"
+#include "../OutputAPI.hxx"
+#include "encoder/EncoderPlugin.hxx"
+#include "encoder/EncoderList.hxx"
+#include "config/ConfigError.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "system/FatalError.hxx"
 #include "Log.hxx"
 
 #include <shout/shout.h>
-#include <glib.h>
 
 #include <assert.h>
 #include <stdlib.h>
@@ -39,7 +38,7 @@
 static constexpr unsigned DEFAULT_CONN_TIMEOUT = 2;
 
 struct ShoutOutput final {
-	struct audio_output base;
+	AudioOutput base;
 
 	shout_t *shout_conn;
 	shout_metadata_t *shout_meta;
@@ -54,7 +53,8 @@ struct ShoutOutput final {
 	uint8_t buffer[32768];
 
 	ShoutOutput()
-		:shout_conn(shout_new()),
+		:base(shout_output_plugin),
+		 shout_conn(shout_new()),
 		shout_meta(shout_metadata_new()),
 		quality(-2.0),
 		bitrate(-1),
@@ -68,12 +68,7 @@ struct ShoutOutput final {
 	}
 
 	bool Initialize(const config_param &param, Error &error) {
-		return ao_base_init(&base, &shout_output_plugin, param,
-				    error);
-	}
-
-	void Deinitialize() {
-		ao_base_finish(&base);
+		return base.Configure(param, error);
 	}
 
 	bool Configure(const config_param &param, Error &error);
@@ -275,7 +270,7 @@ ShoutOutput::Configure(const config_param &param, Error &error)
 	return true;
 }
 
-static struct audio_output *
+static AudioOutput *
 my_shout_init_driver(const config_param &param, Error &error)
 {
 	ShoutOutput *sd = new ShoutOutput();
@@ -285,7 +280,6 @@ my_shout_init_driver(const config_param &param, Error &error)
 	}
 
 	if (!sd->Configure(param, error)) {
-		sd->Deinitialize();
 		delete sd;
 		return nullptr;
 	}
@@ -363,13 +357,12 @@ static void close_shout_conn(ShoutOutput * sd)
 }
 
 static void
-my_shout_finish_driver(struct audio_output *ao)
+my_shout_finish_driver(AudioOutput *ao)
 {
 	ShoutOutput *sd = (ShoutOutput *)ao;
 
 	encoder_finish(sd->encoder);
 
-	sd->Deinitialize();
 	delete sd;
 
 	shout_init_count--;
@@ -379,7 +372,7 @@ my_shout_finish_driver(struct audio_output *ao)
 }
 
 static void
-my_shout_drop_buffered_audio(struct audio_output *ao)
+my_shout_drop_buffered_audio(AudioOutput *ao)
 {
 	gcc_unused
 	ShoutOutput *sd = (ShoutOutput *)ao;
@@ -388,7 +381,7 @@ my_shout_drop_buffered_audio(struct audio_output *ao)
 }
 
 static void
-my_shout_close_device(struct audio_output *ao)
+my_shout_close_device(AudioOutput *ao)
 {
 	ShoutOutput *sd = (ShoutOutput *)ao;
 
@@ -414,7 +407,7 @@ shout_connect(ShoutOutput *sd, Error &error)
 }
 
 static bool
-my_shout_open_device(struct audio_output *ao, AudioFormat &audio_format,
+my_shout_open_device(AudioOutput *ao, AudioFormat &audio_format,
 		     Error &error)
 {
 	ShoutOutput *sd = (ShoutOutput *)ao;
@@ -437,7 +430,7 @@ my_shout_open_device(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static unsigned
-my_shout_delay(struct audio_output *ao)
+my_shout_delay(AudioOutput *ao)
 {
 	ShoutOutput *sd = (ShoutOutput *)ao;
 
@@ -449,7 +442,7 @@ my_shout_delay(struct audio_output *ao)
 }
 
 static size_t
-my_shout_play(struct audio_output *ao, const void *chunk, size_t size,
+my_shout_play(AudioOutput *ao, const void *chunk, size_t size,
 	      Error &error)
 {
 	ShoutOutput *sd = (ShoutOutput *)ao;
@@ -461,7 +454,7 @@ my_shout_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 static bool
-my_shout_pause(struct audio_output *ao)
+my_shout_pause(AudioOutput *ao)
 {
 	static char silence[1020];
 
@@ -477,13 +470,13 @@ shout_tag_to_metadata(const Tag *tag, char *dest, size_t size)
 	artist[0] = 0;
 	title[0] = 0;
 
-	for (unsigned i = 0; i < tag->num_items; i++) {
-		switch (tag->items[i]->type) {
+	for (const auto &item : *tag) {
+		switch (item.type) {
 		case TAG_ARTIST:
-			strncpy(artist, tag->items[i]->value, size);
+			strncpy(artist, item.value, size);
 			break;
 		case TAG_TITLE:
-			strncpy(title, tag->items[i]->value, size);
+			strncpy(title, item.value, size);
 			break;
 
 		default:
@@ -494,7 +487,7 @@ shout_tag_to_metadata(const Tag *tag, char *dest, size_t size)
 	snprintf(dest, size, "%s - %s", artist, title);
 }
 
-static void my_shout_set_tag(struct audio_output *ao,
+static void my_shout_set_tag(AudioOutput *ao,
 			     const Tag *tag)
 {
 	ShoutOutput *sd = (ShoutOutput *)ao;
@@ -525,7 +518,7 @@ static void my_shout_set_tag(struct audio_output *ao,
 	write_page(sd, IgnoreError());
 }
 
-const struct audio_output_plugin shout_output_plugin = {
+const struct AudioOutputPlugin shout_output_plugin = {
 	"shout",
 	nullptr,
 	my_shout_init_driver,
diff --git a/src/output/ShoutOutputPlugin.hxx b/src/output/plugins/ShoutOutputPlugin.hxx
index 496b77975..9f706fc3b 100644
--- a/src/output/ShoutOutputPlugin.hxx
+++ b/src/output/plugins/ShoutOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_SHOUT_OUTPUT_PLUGIN_HXX
 #define MPD_SHOUT_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin shout_output_plugin;
+extern const struct AudioOutputPlugin shout_output_plugin;
 
 #endif
diff --git a/src/output/SolarisOutputPlugin.cxx b/src/output/plugins/SolarisOutputPlugin.cxx
index 0836dc2e2..30745f97c 100644
--- a/src/output/SolarisOutputPlugin.cxx
+++ b/src/output/plugins/SolarisOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,7 +19,7 @@
 
 #include "config.h"
 #include "SolarisOutputPlugin.hxx"
-#include "OutputAPI.hxx"
+#include "../OutputAPI.hxx"
 #include "system/fd_util.h"
 #include "util/Error.hxx"
 
@@ -50,20 +50,18 @@ struct audio_info {
 #endif
 
 struct SolarisOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	/* configuration */
 	const char *device;
 
 	int fd;
 
-	bool Initialize(const config_param &param, Error &error_r) {
-		return ao_base_init(&base, &solaris_output_plugin, param,
-				    error_r);
-	}
+	SolarisOutput()
+		:base(solaris_output_plugin) {}
 
-	void Deinitialize() {
-		ao_base_finish(&base);
+	bool Initialize(const config_param &param, Error &error_r) {
+		return base.Configure(param, error_r);
 	}
 };
 
@@ -76,7 +74,7 @@ solaris_output_test_default_device(void)
 		access("/dev/audio", W_OK) == 0;
 }
 
-static struct audio_output *
+static AudioOutput *
 solaris_output_init(const config_param &param, Error &error_r)
 {
 	SolarisOutput *so = new SolarisOutput();
@@ -91,16 +89,15 @@ solaris_output_init(const config_param &param, Error &error_r)
 }
 
 static void
-solaris_output_finish(struct audio_output *ao)
+solaris_output_finish(AudioOutput *ao)
 {
 	SolarisOutput *so = (SolarisOutput *)ao;
 
-	so->Deinitialize();
 	delete so;
 }
 
 static bool
-solaris_output_open(struct audio_output *ao, AudioFormat &audio_format,
+solaris_output_open(AudioOutput *ao, AudioFormat &audio_format,
 		    Error &error)
 {
 	SolarisOutput *so = (SolarisOutput *)ao;
@@ -151,7 +148,7 @@ solaris_output_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-solaris_output_close(struct audio_output *ao)
+solaris_output_close(AudioOutput *ao)
 {
 	SolarisOutput *so = (SolarisOutput *)ao;
 
@@ -159,7 +156,7 @@ solaris_output_close(struct audio_output *ao)
 }
 
 static size_t
-solaris_output_play(struct audio_output *ao, const void *chunk, size_t size,
+solaris_output_play(AudioOutput *ao, const void *chunk, size_t size,
 		    Error &error)
 {
 	SolarisOutput *so = (SolarisOutput *)ao;
@@ -175,14 +172,14 @@ solaris_output_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 static void
-solaris_output_cancel(struct audio_output *ao)
+solaris_output_cancel(AudioOutput *ao)
 {
 	SolarisOutput *so = (SolarisOutput *)ao;
 
 	ioctl(so->fd, I_FLUSH);
 }
 
-const struct audio_output_plugin solaris_output_plugin = {
+const struct AudioOutputPlugin solaris_output_plugin = {
 	"solaris",
 	solaris_output_test_default_device,
 	solaris_output_init,
diff --git a/src/output/SolarisOutputPlugin.hxx b/src/output/plugins/SolarisOutputPlugin.hxx
index d0fbd32c8..3f9ede7a6 100644
--- a/src/output/SolarisOutputPlugin.hxx
+++ b/src/output/plugins/SolarisOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_SOLARIS_OUTPUT_PLUGIN_HXX
 #define MPD_SOLARIS_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin solaris_output_plugin;
+extern const struct AudioOutputPlugin solaris_output_plugin;
 
 #endif
diff --git a/src/output/WinmmOutputPlugin.cxx b/src/output/plugins/WinmmOutputPlugin.cxx
index d2508ee2a..e5c5a6f0c 100644
--- a/src/output/WinmmOutputPlugin.cxx
+++ b/src/output/plugins/WinmmOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -19,15 +19,13 @@
 
 #include "config.h"
 #include "WinmmOutputPlugin.hxx"
-#include "OutputAPI.hxx"
+#include "../OutputAPI.hxx"
 #include "pcm/PcmBuffer.hxx"
-#include "MixerList.hxx"
+#include "mixer/MixerList.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "util/Macros.hxx"
 
-#include <glib.h>
-
 #include <stdlib.h>
 #include <string.h>
 
@@ -38,7 +36,7 @@ struct WinmmBuffer {
 };
 
 struct WinmmOutput {
-	struct audio_output base;
+	AudioOutput base;
 
 	UINT device_id;
 	HWAVEOUT handle;
@@ -51,14 +49,17 @@ struct WinmmOutput {
 
 	WinmmBuffer buffers[8];
 	unsigned next_buffer;
+
+	WinmmOutput()
+		:base(winmm_output_plugin) {}
 };
 
 static constexpr Domain winmm_output_domain("winmm_output");
 
 HWAVEOUT
-winmm_output_get_handle(WinmmOutput *output)
+winmm_output_get_handle(WinmmOutput &output)
 {
-	return output->handle;
+	return output.handle;
 }
 
 static bool
@@ -108,18 +109,17 @@ fail:
 	return false;
 }
 
-static struct audio_output *
+static AudioOutput *
 winmm_output_init(const config_param &param, Error &error)
 {
 	WinmmOutput *wo = new WinmmOutput();
-	if (!ao_base_init(&wo->base, &winmm_output_plugin, param, error)) {
+	if (!wo->base.Configure(param, error)) {
 		delete wo;
 		return nullptr;
 	}
 
 	const char *device = param.GetBlockValue("device");
 	if (!get_device_id(device, &wo->device_id, error)) {
-		ao_base_finish(&wo->base);
 		delete wo;
 		return nullptr;
 	}
@@ -128,16 +128,15 @@ winmm_output_init(const config_param &param, Error &error)
 }
 
 static void
-winmm_output_finish(struct audio_output *ao)
+winmm_output_finish(AudioOutput *ao)
 {
 	WinmmOutput *wo = (WinmmOutput *)ao;
 
-	ao_base_finish(&wo->base);
 	delete wo;
 }
 
 static bool
-winmm_output_open(struct audio_output *ao, AudioFormat &audio_format,
+winmm_output_open(AudioOutput *ao, AudioFormat &audio_format,
 		  Error &error)
 {
 	WinmmOutput *wo = (WinmmOutput *)ao;
@@ -194,7 +193,7 @@ winmm_output_open(struct audio_output *ao, AudioFormat &audio_format,
 }
 
 static void
-winmm_output_close(struct audio_output *ao)
+winmm_output_close(AudioOutput *ao)
 {
 	WinmmOutput *wo = (WinmmOutput *)ao;
 
@@ -263,7 +262,7 @@ winmm_drain_buffer(WinmmOutput *wo, WinmmBuffer *buffer,
 }
 
 static size_t
-winmm_output_play(struct audio_output *ao, const void *chunk, size_t size, Error &error)
+winmm_output_play(AudioOutput *ao, const void *chunk, size_t size, Error &error)
 {
 	WinmmOutput *wo = (WinmmOutput *)ao;
 
@@ -318,7 +317,7 @@ winmm_stop(WinmmOutput *wo)
 }
 
 static void
-winmm_output_drain(struct audio_output *ao)
+winmm_output_drain(AudioOutput *ao)
 {
 	WinmmOutput *wo = (WinmmOutput *)ao;
 
@@ -327,14 +326,14 @@ winmm_output_drain(struct audio_output *ao)
 }
 
 static void
-winmm_output_cancel(struct audio_output *ao)
+winmm_output_cancel(AudioOutput *ao)
 {
 	WinmmOutput *wo = (WinmmOutput *)ao;
 
 	winmm_stop(wo);
 }
 
-const struct audio_output_plugin winmm_output_plugin = {
+const struct AudioOutputPlugin winmm_output_plugin = {
 	"winmm",
 	winmm_output_test_default_device,
 	winmm_output_init,
diff --git a/src/output/WinmmOutputPlugin.hxx b/src/output/plugins/WinmmOutputPlugin.hxx
index a6b7733ec..50fae4f2f 100644
--- a/src/output/WinmmOutputPlugin.hxx
+++ b/src/output/plugins/WinmmOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -31,11 +31,11 @@
 
 struct WinmmOutput;
 
-extern const struct audio_output_plugin winmm_output_plugin;
+extern const struct AudioOutputPlugin winmm_output_plugin;
 
 gcc_pure
 HWAVEOUT
-winmm_output_get_handle(WinmmOutput *);
+winmm_output_get_handle(WinmmOutput &output);
 
 #endif
 
diff --git a/src/output/HttpdClient.cxx b/src/output/plugins/httpd/HttpdClient.cxx
index dc337053d..3797c3d26 100644
--- a/src/output/HttpdClient.cxx
+++ b/src/output/plugins/httpd/HttpdClient.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -26,8 +26,6 @@
 #include "system/SocketError.hxx"
 #include "Log.hxx"
 
-#include <glib.h>
-
 #include <assert.h>
 #include <string.h>
 #include <stdio.h>
@@ -38,24 +36,26 @@ HttpdClient::~HttpdClient()
 		if (current_page != nullptr)
 			current_page->Unref();
 
-		for (auto page : pages)
-			page->Unref();
+		ClearQueue();
 	}
 
 	if (metadata)
 		metadata->Unref();
+
+	if (IsDefined())
+		BufferedSocket::Close();
 }
 
 void
 HttpdClient::Close()
 {
-	httpd->RemoveClient(*this);
+	httpd.RemoveClient(*this);
 }
 
 void
 HttpdClient::LockClose()
 {
-	const ScopeLock protect(httpd->mutex);
+	const ScopeLock protect(httpd.mutex);
 	Close();
 }
 
@@ -68,7 +68,7 @@ HttpdClient::BeginResponse()
 	current_page = nullptr;
 
 	if (!head_method)
-		httpd->SendHeader(*this);
+		httpd.SendHeader(*this);
 }
 
 /**
@@ -141,7 +141,9 @@ HttpdClient::HandleLine(const char *line)
 bool
 HttpdClient::SendResponse()
 {
-	char buffer[1024];
+	char buffer[1024], *allocated = nullptr;
+	const char *response;
+
 	assert(state == RESPONSE);
 
 	if (dlna_streaming_requested) {
@@ -156,19 +158,15 @@ HttpdClient::SendResponse()
 			 "realTimeInfo.dlna.org: DLNA.ORG_TLAG=*\r\n"
 			 "contentFeatures.dlna.org: DLNA.ORG_OP=01;DLNA.ORG_CI=0\r\n"
 			 "\r\n",
-			 httpd->content_type);
+			 httpd.content_type);
+		response = buffer;
 
 	} else if (metadata_requested) {
-		char *metadata_header =
-			icy_server_metadata_header(httpd->name, httpd->genre,
-						   httpd->website,
-						   httpd->content_type,
+		response = allocated =
+			icy_server_metadata_header(httpd.name, httpd.genre,
+						   httpd.website,
+						   httpd.content_type,
 						   metaint);
-
-		g_strlcpy(buffer, metadata_header, sizeof(buffer));
-
-		delete[] metadata_header;
-
        } else { /* revert to a normal HTTP request */
 		snprintf(buffer, sizeof(buffer),
 			 "HTTP/1.1 200 OK\r\n"
@@ -177,10 +175,12 @@ HttpdClient::SendResponse()
 			 "Pragma: no-cache\r\n"
 			 "Cache-Control: no-cache, no-store\r\n"
 			 "\r\n",
-			 httpd->content_type);
+			 httpd.content_type);
+		response = buffer;
 	}
 
-	ssize_t nbytes = SocketMonitor::Write(buffer, strlen(buffer));
+	ssize_t nbytes = SocketMonitor::Write(response, strlen(response));
+	delete[] allocated;
 	if (gcc_unlikely(nbytes < 0)) {
 		const SocketErrorMessage msg;
 		FormatWarning(httpd_output_domain,
@@ -193,11 +193,12 @@ HttpdClient::SendResponse()
 	return true;
 }
 
-HttpdClient::HttpdClient(HttpdOutput *_httpd, int _fd, EventLoop &_loop,
+HttpdClient::HttpdClient(HttpdOutput &_httpd, int _fd, EventLoop &_loop,
 			 bool _metadata_supported)
 	:BufferedSocket(_fd, _loop),
 	 httpd(_httpd),
 	 state(REQUEST),
+	 queue_size(0),
 	 head_method(false),
 	 dlna_streaming_requested(false),
 	 metadata_supported(_metadata_supported),
@@ -208,16 +209,24 @@ HttpdClient::HttpdClient(HttpdOutput *_httpd, int _fd, EventLoop &_loop,
 {
 }
 
-size_t
-HttpdClient::GetQueueSize() const
+void
+HttpdClient::ClearQueue()
 {
-	if (state != RESPONSE)
-		return 0;
+	assert(state == RESPONSE);
+
+	while (!pages.empty()) {
+		Page *page = pages.front();
+		pages.pop();
 
-	size_t size = 0;
-	for (auto page : pages)
-		size += page->size;
-	return size;
+#ifndef NDEBUG
+		assert(queue_size >= page->size);
+		queue_size -= page->size;
+#endif
+
+		page->Unref();
+	}
+
+	assert(queue_size == 0);
 }
 
 void
@@ -226,9 +235,7 @@ HttpdClient::CancelQueue()
 	if (state != RESPONSE)
 		return;
 
-	for (auto page : pages)
-		page->Unref();
-	pages.clear();
+	ClearQueue();
 
 	if (current_page == nullptr)
 		CancelWrite();
@@ -263,7 +270,7 @@ HttpdClient::GetBytesTillMetaData() const
 inline bool
 HttpdClient::TryWrite()
 {
-	const ScopeLock protect(httpd->mutex);
+	const ScopeLock protect(httpd.mutex);
 
 	assert(state == RESPONSE);
 
@@ -271,14 +278,17 @@ HttpdClient::TryWrite()
 		if (pages.empty()) {
 			/* another thread has removed the event source
 			   while this thread was waiting for
-			   httpd->mutex */
+			   httpd.mutex */
 			CancelWrite();
 			return true;
 		}
 
 		current_page = pages.front();
-		pages.pop_front();
+		pages.pop();
 		current_position = 0;
+
+		assert(queue_size >= current_page->size);
+		queue_size -= current_page->size;
 	}
 
 	const ssize_t bytes_to_write = GetBytesTillMetaData();
@@ -310,7 +320,7 @@ HttpdClient::TryWrite()
 				metadata_sent = true;
 			}
 		} else {
-			guchar empty_data = 0;
+			char empty_data = 0;
 
 			ssize_t nbytes = Write(&empty_data, 1);
 			if (nbytes < 0) {
@@ -379,8 +389,15 @@ HttpdClient::PushPage(Page *page)
 		/* the client is still writing the HTTP request */
 		return;
 
+	if (queue_size > 256 * 1024) {
+		FormatDebug(httpd_output_domain,
+			    "client is too slow, flushing its queue");
+		ClearQueue();
+	}
+
 	page->Ref();
-	pages.push_back(page);
+	pages.push(page);
+	queue_size += page->size;
 
 	ScheduleWrite();
 }
@@ -388,13 +405,13 @@ HttpdClient::PushPage(Page *page)
 void
 HttpdClient::PushMetaData(Page *page)
 {
+	assert(page != nullptr);
+
 	if (metadata) {
 		metadata->Unref();
 		metadata = nullptr;
 	}
 
-	g_return_if_fail (page);
-
 	page->Ref();
 	metadata = page;
 	metadata_sent = false;
diff --git a/src/output/HttpdClient.hxx b/src/output/plugins/httpd/HttpdClient.hxx
index 66a819232..f94f05769 100644
--- a/src/output/HttpdClient.hxx
+++ b/src/output/plugins/httpd/HttpdClient.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -23,18 +23,19 @@
 #include "event/BufferedSocket.hxx"
 #include "Compiler.h"
 
+#include <queue>
 #include <list>
 
 #include <stddef.h>
 
-struct HttpdOutput;
+class HttpdOutput;
 class Page;
 
-class HttpdClient final : public BufferedSocket {
+class HttpdClient final : BufferedSocket {
 	/**
 	 * The httpd output object this client is connected to.
 	 */
-	HttpdOutput *const httpd;
+	HttpdOutput &httpd;
 
 	/**
 	 * The current state of the client.
@@ -53,7 +54,12 @@ class HttpdClient final : public BufferedSocket {
 	/**
 	 * A queue of #Page objects to be sent to the client.
 	 */
-	std::list<Page *> pages;
+	std::queue<Page *, std::list<Page *>> pages;
+
+	/**
+	 * The sum of all page sizes in #pages.
+	 */
+	size_t queue_size;
 
 	/**
 	 * The #page which is currently being sent to the client.
@@ -120,7 +126,7 @@ public:
 	 * @param httpd the HTTP output device
 	 * @param fd the socket file descriptor
 	 */
-	HttpdClient(HttpdOutput *httpd, int _fd, EventLoop &_loop,
+	HttpdClient(HttpdOutput &httpd, int _fd, EventLoop &_loop,
 		    bool _metadata_supported);
 
 	/**
@@ -137,12 +143,6 @@ public:
 	void LockClose();
 
 	/**
-	 * Returns the total size of this client's page queue.
-	 */
-	gcc_pure
-	size_t GetQueueSize() const;
-
-	/**
 	 * Clears the page queue.
 	 */
 	void CancelQueue();
@@ -180,6 +180,9 @@ public:
 	 */
 	void PushMetaData(Page *page);
 
+private:
+	void ClearQueue();
+
 protected:
 	virtual bool OnSocketReady(unsigned flags) override;
 	virtual InputResult OnSocketInput(void *data, size_t length) override;
diff --git a/src/output/HttpdInternal.hxx b/src/output/plugins/httpd/HttpdInternal.hxx
index b76493a44..20ff15e42 100644
--- a/src/output/HttpdInternal.hxx
+++ b/src/output/plugins/httpd/HttpdInternal.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -25,10 +25,13 @@
 #ifndef MPD_OUTPUT_HTTPD_INTERNAL_H
 #define MPD_OUTPUT_HTTPD_INTERNAL_H
 
-#include "OutputInternal.hxx"
-#include "Timer.hxx"
+#include "output/Internal.hxx"
+#include "output/Timer.hxx"
 #include "thread/Mutex.hxx"
 #include "event/ServerSocket.hxx"
+#include "event/DeferredMonitor.hxx"
+#include "util/Cast.hxx"
+#include "Compiler.h"
 
 #ifdef _LIBCPP_VERSION
 /* can't use incomplete template arguments with libc++ */
@@ -36,6 +39,8 @@
 #endif
 
 #include <forward_list>
+#include <queue>
+#include <list>
 
 struct config_param;
 class Error;
@@ -46,8 +51,8 @@ class Page;
 struct Encoder;
 struct Tag;
 
-struct HttpdOutput final : private ServerSocket {
-	struct audio_output base;
+class HttpdOutput final : ServerSocket, DeferredMonitor {
+	AudioOutput base;
 
 	/**
 	 * True if the audio output is open and accepts client
@@ -68,6 +73,7 @@ struct HttpdOutput final : private ServerSocket {
 	 */
 	size_t unflushed_input;
 
+public:
 	/**
 	 * The MIME type produced by the #encoder.
 	 */
@@ -80,6 +86,13 @@ struct HttpdOutput final : private ServerSocket {
 	mutable Mutex mutex;
 
 	/**
+	 * This condition gets signalled when an item is removed from
+	 * #pages.
+	 */
+	Cond cond;
+
+private:
+	/**
 	 * A #Timer object to synchronize this output with the
 	 * wallclock.
 	 */
@@ -96,6 +109,15 @@ struct HttpdOutput final : private ServerSocket {
 	Page *metadata;
 
 	/**
+	 * The page queue, i.e. pages from the encoder to be
+	 * broadcasted to all clients.  This container is necessary to
+	 * pass pages from the OutputThread to the IOThread.  It is
+	 * protected by #mutex, and removing signals #cond.
+	 */
+	std::queue<Page *, std::list<Page *>> pages;
+
+ public:
+	/**
 	 * The configured name.
 	 */
 	char const *name;
@@ -108,6 +130,7 @@ struct HttpdOutput final : private ServerSocket {
 	 */
 	char const *website;
 
+private:
 	/**
 	 * A linked list containing all clients which are currently
 	 * connected.
@@ -126,11 +149,34 @@ struct HttpdOutput final : private ServerSocket {
 	 */
 	unsigned clients_max, clients_cnt;
 
+public:
 	HttpdOutput(EventLoop &_loop);
 	~HttpdOutput();
 
+#if defined(__clang__) || GCC_CHECK_VERSION(4,7)
+	constexpr
+#endif
+	static HttpdOutput *Cast(AudioOutput *ao) {
+		return &ContainerCast(*ao, &HttpdOutput::base);
+	}
+
+	using DeferredMonitor::GetEventLoop;
+
+	bool Init(const config_param &param, Error &error);
+
 	bool Configure(const config_param &param, Error &error);
 
+	AudioOutput *InitAndConfigure(const config_param &param,
+				       Error &error) {
+		if (!Init(param, error))
+			return nullptr;
+
+		if (!Configure(param, error))
+			return nullptr;
+
+		return &base;
+	}
+
 	bool Bind(Error &error);
 	void Unbind();
 
@@ -181,6 +227,9 @@ struct HttpdOutput final : private ServerSocket {
 	 */
 	void SendHeader(HttpdClient &client) const;
 
+	gcc_pure
+	unsigned Delay() const;
+
 	/**
 	 * Reads data from the encoder (as much as available) and
 	 * returns it as a new #page object.
@@ -203,7 +252,13 @@ struct HttpdOutput final : private ServerSocket {
 
 	void SendTag(const Tag *tag);
 
+	size_t Play(const void *chunk, size_t size, Error &error);
+
+	void CancelAllClients();
+
 private:
+	virtual void RunDeferred() override;
+
 	virtual void OnAccept(int fd, const sockaddr &address,
 			      size_t address_length, int uid) override;
 };
diff --git a/src/output/HttpdOutputPlugin.cxx b/src/output/plugins/httpd/HttpdOutputPlugin.cxx
index 369c06937..e3ba7727d 100644
--- a/src/output/HttpdOutputPlugin.cxx
+++ b/src/output/plugins/httpd/HttpdOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -21,20 +21,19 @@
 #include "HttpdOutputPlugin.hxx"
 #include "HttpdInternal.hxx"
 #include "HttpdClient.hxx"
-#include "OutputAPI.hxx"
-#include "EncoderPlugin.hxx"
-#include "EncoderList.hxx"
+#include "output/OutputAPI.hxx"
+#include "encoder/EncoderPlugin.hxx"
+#include "encoder/EncoderList.hxx"
 #include "system/Resolver.hxx"
 #include "Page.hxx"
 #include "IcyMetaDataServer.hxx"
 #include "system/fd_util.h"
-#include "Main.hxx"
+#include "IOThread.hxx"
+#include "event/Call.hxx"
 #include "util/Error.hxx"
 #include "util/Domain.hxx"
 #include "Log.hxx"
 
-#include <glib.h>
-
 #include <assert.h>
 
 #include <sys/types.h>
@@ -51,7 +50,8 @@ const Domain httpd_output_domain("httpd_output");
 
 inline
 HttpdOutput::HttpdOutput(EventLoop &_loop)
-	:ServerSocket(_loop),
+	:ServerSocket(_loop), DeferredMonitor(_loop),
+	 base(httpd_output_plugin),
 	 encoder(nullptr), unflushed_input(0),
 	 metadata(nullptr)
 {
@@ -72,8 +72,11 @@ HttpdOutput::Bind(Error &error)
 {
 	open = false;
 
-	const ScopeLock protect(mutex);
-	return ServerSocket::Open(error);
+	bool result = false;
+	BlockingCall(GetEventLoop(), [this, &error, &result](){
+			result = ServerSocket::Open(error);
+		});
+	return result;
 }
 
 inline void
@@ -81,8 +84,9 @@ HttpdOutput::Unbind()
 {
 	assert(!open);
 
-	const ScopeLock protect(mutex);
-	ServerSocket::Close();
+	BlockingCall(GetEventLoop(), [this](){
+			ServerSocket::Close();
+		});
 }
 
 inline bool
@@ -130,47 +134,29 @@ HttpdOutput::Configure(const config_param &param, Error &error)
 	return true;
 }
 
-static struct audio_output *
-httpd_output_init(const config_param &param, Error &error)
+inline bool
+HttpdOutput::Init(const config_param &param, Error &error)
 {
-	HttpdOutput *httpd = new HttpdOutput(*main_loop);
+	return base.Configure(param, error);
+}
 
-	if (!ao_base_init(&httpd->base, &httpd_output_plugin, param,
-			  error)) {
-		delete httpd;
-		return nullptr;
-	}
+static AudioOutput *
+httpd_output_init(const config_param &param, Error &error)
+{
+	HttpdOutput *httpd = new HttpdOutput(io_thread_get());
 
-	if (!httpd->Configure(param, error)) {
-		ao_base_finish(&httpd->base);
+	AudioOutput *result = httpd->InitAndConfigure(param, error);
+	if (result == nullptr)
 		delete httpd;
-		return nullptr;
-	}
-
-	return &httpd->base;
-}
 
-#if GCC_CHECK_VERSION(4,6) || defined(__clang__)
-#pragma GCC diagnostic push
-#pragma GCC diagnostic ignored "-Winvalid-offsetof"
-#endif
-
-static inline constexpr HttpdOutput *
-Cast(audio_output *ao)
-{
-	return (HttpdOutput *)((char *)ao - offsetof(HttpdOutput, base));
+	return result;
 }
 
-#if GCC_CHECK_VERSION(4,6) || defined(__clang__)
-#pragma GCC diagnostic pop
-#endif
-
 static void
-httpd_output_finish(struct audio_output *ao)
+httpd_output_finish(AudioOutput *ao)
 {
-	HttpdOutput *httpd = Cast(ao);
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
 
-	ao_base_finish(&httpd->base);
 	delete httpd;
 }
 
@@ -181,7 +167,7 @@ httpd_output_finish(struct audio_output *ao)
 inline void
 HttpdOutput::AddClient(int fd)
 {
-	clients.emplace_front(this, fd, GetEventLoop(),
+	clients.emplace_front(*this, fd, GetEventLoop(),
 			      encoder->plugin.tag == nullptr);
 	++clients_cnt;
 
@@ -191,6 +177,29 @@ HttpdOutput::AddClient(int fd)
 }
 
 void
+HttpdOutput::RunDeferred()
+{
+	/* this method runs in the IOThread; it broadcasts pages from
+	   our own queue to all clients */
+
+	const ScopeLock protect(mutex);
+
+	while (!pages.empty()) {
+		Page *page = pages.front();
+		pages.pop();
+
+		for (auto &client : clients)
+			client.PushPage(page);
+
+		page->Unref();
+	}
+
+	/* wake up the client that may be waiting for the queue to be
+	   flushed */
+	cond.broadcast();
+}
+
+void
 HttpdOutput::OnAccept(int fd, const sockaddr &address,
 		      size_t address_length, gcc_unused int uid)
 {
@@ -199,9 +208,10 @@ HttpdOutput::OnAccept(int fd, const sockaddr &address,
 
 #ifdef HAVE_LIBWRAP
 	if (address.sa_family != AF_UNIX) {
-		char *hostaddr = sockaddr_to_string(&address, address_length,
-						    IgnoreError());
-		const char *progname = g_get_prgname();
+		const auto hostaddr = sockaddr_to_string(&address,
+							 address_length);
+		// TODO: shall we obtain the program name from argv[0]?
+		const char *progname = "mpd";
 
 		struct request_info req;
 		request_init(&req, RQ_FILE, fd, RQ_DAEMON, progname, 0);
@@ -212,13 +222,10 @@ HttpdOutput::OnAccept(int fd, const sockaddr &address,
 			/* tcp wrappers says no */
 			FormatWarning(httpd_output_domain,
 				      "libwrap refused connection (libwrap=%s) from %s",
-				      progname, hostaddr);
-			g_free(hostaddr);
+				      progname, hostaddr.c_str());
 			close_socket(fd);
 			return;
 		}
-
-		g_free(hostaddr);
 	}
 #else
 	(void)address;
@@ -269,17 +276,17 @@ HttpdOutput::ReadPage()
 }
 
 static bool
-httpd_output_enable(struct audio_output *ao, Error &error)
+httpd_output_enable(AudioOutput *ao, Error &error)
 {
-	HttpdOutput *httpd = Cast(ao);
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
 
 	return httpd->Bind(error);
 }
 
 static void
-httpd_output_disable(struct audio_output *ao)
+httpd_output_disable(AudioOutput *ao)
 {
-	HttpdOutput *httpd = Cast(ao);
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
 
 	httpd->Unbind();
 }
@@ -322,12 +329,10 @@ HttpdOutput::Open(AudioFormat &audio_format, Error &error)
 }
 
 static bool
-httpd_output_open(struct audio_output *ao, AudioFormat &audio_format,
+httpd_output_open(AudioOutput *ao, AudioFormat &audio_format,
 		  Error &error)
 {
-	HttpdOutput *httpd = Cast(ao);
-
-	assert(httpd->clients.empty());
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
 
 	const ScopeLock protect(httpd->mutex);
 	return httpd->Open(audio_format, error);
@@ -342,7 +347,9 @@ HttpdOutput::Close()
 
 	delete timer;
 
-	clients.clear();
+	BlockingCall(GetEventLoop(), [this](){
+			clients.clear();
+		});
 
 	if (header != nullptr)
 		header->Unref();
@@ -351,9 +358,9 @@ HttpdOutput::Close()
 }
 
 static void
-httpd_output_close(struct audio_output *ao)
+httpd_output_close(AudioOutput *ao)
 {
-	HttpdOutput *httpd = Cast(ao);
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
 
 	const ScopeLock protect(httpd->mutex);
 	httpd->Close();
@@ -382,17 +389,15 @@ HttpdOutput::SendHeader(HttpdClient &client) const
 		client.PushPage(header);
 }
 
-static unsigned
-httpd_output_delay(struct audio_output *ao)
+inline unsigned
+HttpdOutput::Delay() const
 {
-	HttpdOutput *httpd = Cast(ao);
-
-	if (!httpd->LockHasClients() && httpd->base.pause) {
+	if (!LockHasClients() && base.pause) {
 		/* if there's no client and this output is paused,
 		   then httpd_output_pause() will not do anything, it
 		   will not fill the buffer and it will not update the
 		   timer; therefore, we reset the timer here */
-		httpd->timer->Reset();
+		timer->Reset();
 
 		/* some arbitrary delay that is long enough to avoid
 		   consuming too much CPU, and short enough to notice
@@ -400,39 +405,47 @@ httpd_output_delay(struct audio_output *ao)
 		return 1000;
 	}
 
-	return httpd->timer->IsStarted()
-		? httpd->timer->GetDelay()
+	return timer->IsStarted()
+		? timer->GetDelay()
 		: 0;
 }
 
+static unsigned
+httpd_output_delay(AudioOutput *ao)
+{
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
+
+	return httpd->Delay();
+}
+
 void
 HttpdOutput::BroadcastPage(Page *page)
 {
 	assert(page != nullptr);
 
-	const ScopeLock protect(mutex);
-	for (auto &client : clients)
-		client.PushPage(page);
+	mutex.lock();
+	pages.push(page);
+	page->Ref();
+	mutex.unlock();
+
+	DeferredMonitor::Schedule();
 }
 
 void
 HttpdOutput::BroadcastFromEncoder()
 {
+	/* synchronize with the IOThread */
 	mutex.lock();
-	for (auto &client : clients) {
-		if (client.GetQueueSize() > 256 * 1024) {
-			FormatDebug(httpd_output_domain,
-				    "client is too slow, flushing its queue");
-			client.CancelQueue();
-		}
-	}
-	mutex.unlock();
+	while (!pages.empty())
+		cond.wait(mutex);
 
 	Page *page;
-	while ((page = ReadPage()) != nullptr) {
-		BroadcastPage(page);
-		page->Unref();
-	}
+	while ((page = ReadPage()) != nullptr)
+		pages.push(page);
+
+	mutex.unlock();
+
+	DeferredMonitor::Schedule();
 }
 
 inline bool
@@ -447,28 +460,34 @@ HttpdOutput::EncodeAndPlay(const void *chunk, size_t size, Error &error)
 	return true;
 }
 
-static size_t
-httpd_output_play(struct audio_output *ao, const void *chunk, size_t size,
-		  Error &error)
+inline size_t
+HttpdOutput::Play(const void *chunk, size_t size, Error &error)
 {
-	HttpdOutput *httpd = Cast(ao);
-
-	if (httpd->LockHasClients()) {
-		if (!httpd->EncodeAndPlay(chunk, size, error))
+	if (LockHasClients()) {
+		if (!EncodeAndPlay(chunk, size, error))
 			return 0;
 	}
 
-	if (!httpd->timer->IsStarted())
-		httpd->timer->Start();
-	httpd->timer->Add(size);
+	if (!timer->IsStarted())
+		timer->Start();
+	timer->Add(size);
 
 	return size;
 }
 
+static size_t
+httpd_output_play(AudioOutput *ao, const void *chunk, size_t size,
+		  Error &error)
+{
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
+
+	return httpd->Play(chunk, size, error);
+}
+
 static bool
-httpd_output_pause(struct audio_output *ao)
+httpd_output_pause(AudioOutput *ao)
 {
-	HttpdOutput *httpd = Cast(ao);
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
 
 	if (httpd->LockHasClients()) {
 		static const char silence[1020] = { 0 };
@@ -529,24 +548,41 @@ HttpdOutput::SendTag(const Tag *tag)
 }
 
 static void
-httpd_output_tag(struct audio_output *ao, const Tag *tag)
+httpd_output_tag(AudioOutput *ao, const Tag *tag)
 {
-	HttpdOutput *httpd = Cast(ao);
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
 
 	httpd->SendTag(tag);
 }
 
-static void
-httpd_output_cancel(struct audio_output *ao)
+inline void
+HttpdOutput::CancelAllClients()
 {
-	HttpdOutput *httpd = Cast(ao);
+	const ScopeLock protect(mutex);
 
-	const ScopeLock protect(httpd->mutex);
-	for (auto &client : httpd->clients)
+	while (!pages.empty()) {
+		Page *page = pages.front();
+		pages.pop();
+		page->Unref();
+	}
+
+	for (auto &client : clients)
 		client.CancelQueue();
+
+	cond.broadcast();
+}
+
+static void
+httpd_output_cancel(AudioOutput *ao)
+{
+	HttpdOutput *httpd = HttpdOutput::Cast(ao);
+
+	BlockingCall(io_thread_get(), [httpd](){
+			httpd->CancelAllClients();
+		});
 }
 
-const struct audio_output_plugin httpd_output_plugin = {
+const struct AudioOutputPlugin httpd_output_plugin = {
 	"httpd",
 	nullptr,
 	httpd_output_init,
diff --git a/src/output/HttpdOutputPlugin.hxx b/src/output/plugins/httpd/HttpdOutputPlugin.hxx
index c74d2bd4a..df99e2b43 100644
--- a/src/output/HttpdOutputPlugin.hxx
+++ b/src/output/plugins/httpd/HttpdOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * 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
@@ -20,6 +20,6 @@
 #ifndef MPD_HTTPD_OUTPUT_PLUGIN_HXX
 #define MPD_HTTPD_OUTPUT_PLUGIN_HXX
 
-extern const struct audio_output_plugin httpd_output_plugin;
+extern const struct AudioOutputPlugin httpd_output_plugin;
 
 #endif
diff --git a/src/output/plugins/httpd/IcyMetaDataServer.cxx b/src/output/plugins/httpd/IcyMetaDataServer.cxx
new file mode 100644
index 000000000..146df23d1
--- /dev/null
+++ b/src/output/plugins/httpd/IcyMetaDataServer.cxx
@@ -0,0 +1,134 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "IcyMetaDataServer.hxx"
+#include "Page.hxx"
+#include "tag/Tag.hxx"
+#include "util/FormatString.hxx"
+
+#include <glib.h>
+
+#include <string.h>
+
+char*
+icy_server_metadata_header(const char *name,
+			   const char *genre, const char *url,
+			   const char *content_type, int metaint)
+{
+	return FormatNew("ICY 200 OK\r\n"
+			 "icy-notice1:<BR>This stream requires an audio player!<BR>\r\n" /* TODO */
+			 "icy-notice2:MPD - The music player daemon<BR>\r\n"
+			 "icy-name: %s\r\n"             /* TODO */
+			 "icy-genre: %s\r\n"            /* TODO */
+			 "icy-url: %s\r\n"              /* TODO */
+			 "icy-pub:1\r\n"
+			 "icy-metaint:%d\r\n"
+			 /* TODO "icy-br:%d\r\n" */
+			 "Content-Type: %s\r\n"
+			 "Connection: close\r\n"
+			 "Pragma: no-cache\r\n"
+			 "Cache-Control: no-cache, no-store\r\n"
+			 "\r\n",
+			 name,
+			 genre,
+			 url,
+			 metaint,
+			 /* bitrate, */
+			 content_type);
+}
+
+static char *
+icy_server_metadata_string(const char *stream_title, const char* stream_url)
+{
+	gchar *icy_metadata;
+	guint meta_length;
+
+	// The leading n is a placeholder for the length information
+	icy_metadata = FormatNew("nStreamTitle='%s';"
+				 "StreamUrl='%s';",
+				 stream_title,
+				 stream_url);
+
+	meta_length = strlen(icy_metadata);
+
+	meta_length--; // subtract placeholder
+
+	meta_length = ((int)meta_length / 16) + 1;
+
+	icy_metadata[0] = meta_length;
+
+	if (meta_length > 255) {
+		delete[] icy_metadata;
+		return nullptr;
+	}
+
+	return icy_metadata;
+}
+
+Page *
+icy_server_metadata_page(const Tag &tag, const TagType *types)
+{
+	const gchar *tag_items[TAG_NUM_OF_ITEM_TYPES];
+	gint last_item, item;
+	guint position;
+	gchar *icy_string;
+	gchar stream_title[(1 + 255 - 28) * 16]; // Length + Metadata -
+						 // "StreamTitle='';StreamUrl='';"
+						 // = 4081 - 28
+	stream_title[0] =  '\0';
+
+	last_item = -1;
+
+	while (*types != TAG_NUM_OF_ITEM_TYPES) {
+		const gchar *tag_item = tag.GetValue(*types++);
+		if (tag_item)
+			tag_items[++last_item] = tag_item;
+	}
+
+	position = item = 0;
+	while (position < sizeof(stream_title) && item <= last_item) {
+		gint length = 0;
+
+		length = g_strlcpy(stream_title + position,
+				   tag_items[item++],
+				   sizeof(stream_title) - position);
+
+		position += length;
+
+		if (item <= last_item) {
+			length = g_strlcpy(stream_title + position,
+					   " - ",
+					   sizeof(stream_title) - position);
+
+			position += length;
+		}
+	}
+
+	icy_string = icy_server_metadata_string(stream_title, "");
+
+	if (icy_string == nullptr)
+		return nullptr;
+
+	Page *icy_metadata = Page::Copy(icy_string, (icy_string[0] * 16) + 1);
+
+	delete[] icy_string;
+
+	return icy_metadata;
+}
diff --git a/src/output/plugins/httpd/IcyMetaDataServer.hxx b/src/output/plugins/httpd/IcyMetaDataServer.hxx
new file mode 100644
index 000000000..773b46641
--- /dev/null
+++ b/src/output/plugins/httpd/IcyMetaDataServer.hxx
@@ -0,0 +1,39 @@
+/*
+ * 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_ICY_META_DATA_SERVER_HXX
+#define MPD_ICY_META_DATA_SERVER_HXX
+
+#include "tag/TagType.h"
+
+struct Tag;
+class Page;
+
+/**
+ * Free the return value with delete[].
+ */
+char*
+icy_server_metadata_header(const char *name,
+			   const char *genre, const char *url,
+			   const char *content_type, int metaint);
+
+Page *
+icy_server_metadata_page(const Tag &tag, const TagType *types);
+
+#endif
diff --git a/src/output/plugins/httpd/Page.cxx b/src/output/plugins/httpd/Page.cxx
new file mode 100644
index 000000000..e22134bbc
--- /dev/null
+++ b/src/output/plugins/httpd/Page.cxx
@@ -0,0 +1,70 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "Page.hxx"
+#include "util/Alloc.hxx"
+
+#include <new>
+
+#include <assert.h>
+#include <string.h>
+#include <stdlib.h>
+
+Page *
+Page::Create(size_t size)
+{
+	void *p = xalloc(sizeof(Page) + size -
+			 sizeof(Page::data));
+	return ::new(p) Page(size);
+}
+
+Page *
+Page::Copy(const void *data, size_t size)
+{
+	assert(data != nullptr);
+
+	Page *page = Create(size);
+	memcpy(page->data, data, size);
+	return page;
+}
+
+Page *
+Page::Concat(const Page &a, const Page &b)
+{
+	Page *page = Create(a.size + b.size);
+
+	memcpy(page->data, a.data, a.size);
+	memcpy(page->data + a.size, b.data, b.size);
+
+	return page;
+}
+
+bool
+Page::Unref()
+{
+	bool unused = ref.Decrement();
+
+	if (unused) {
+		this->Page::~Page();
+		free(this);
+	}
+
+	return unused;
+}
diff --git a/src/output/plugins/httpd/Page.hxx b/src/output/plugins/httpd/Page.hxx
new file mode 100644
index 000000000..95f35d06a
--- /dev/null
+++ b/src/output/plugins/httpd/Page.hxx
@@ -0,0 +1,102 @@
+/*
+ * 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.
+ */
+
+/** \file
+ *
+ * This is a library which manages reference counted buffers.
+ */
+
+#ifndef MPD_PAGE_HXX
+#define MPD_PAGE_HXX
+
+#include "util/RefCount.hxx"
+
+#include <stddef.h>
+
+/**
+ * A dynamically allocated buffer which keeps track of its reference
+ * count.  This is useful for passing buffers around, when several
+ * instances hold references to one buffer.
+ */
+class Page {
+	/**
+	 * The number of references to this buffer.  This library uses
+	 * atomic functions to access it, i.e. no locks are required.
+	 * As soon as this attribute reaches zero, the buffer is
+	 * freed.
+	 */
+	RefCount ref;
+
+public:
+	/**
+	 * The size of this buffer in bytes.
+	 */
+	const size_t size;
+
+	/**
+	 * Dynamic array containing the buffer data.
+	 */
+	unsigned char data[sizeof(long)];
+
+protected:
+	Page(size_t _size):size(_size) {}
+	~Page() = default;
+
+	/**
+	 * Allocates a new #Page object, without filling the data
+	 * element.
+	 */
+	static Page *Create(size_t size);
+
+public:
+	/**
+	 * Creates a new #page object, and copies data from the
+	 * specified buffer.  It is initialized with a reference count
+	 * of 1.
+	 *
+	 * @param data the source buffer
+	 * @param size the size of the source buffer
+	 */
+	static Page *Copy(const void *data, size_t size);
+
+	/**
+	 * Concatenates two pages to a new page.
+	 *
+	 * @param a the first page
+	 * @param b the second page, which is appended
+	 */
+	static Page *Concat(const Page &a, const Page &b);
+
+	/**
+	 * Increases the reference counter.
+	 */
+	void Ref() {
+		ref.Increment();
+	}
+
+	/**
+	 * Decreases the reference counter.  If it reaches zero, the #page is
+	 * freed.
+	 *
+	 * @return true if the #page has been freed
+	 */
+	bool Unref();
+};
+
+#endif
diff --git a/src/output/plugins/sles/AndroidSimpleBufferQueue.hxx b/src/output/plugins/sles/AndroidSimpleBufferQueue.hxx
new file mode 100644
index 000000000..c7dd4ccca
--- /dev/null
+++ b/src/output/plugins/sles/AndroidSimpleBufferQueue.hxx
@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the
+ * distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ * FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE
+ * FOUNDATION OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+ * STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
+ * OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef SLES_ANDROID_SIMPLE_BUFFER_QUEUE_HPP
+#define SLES_ANDROID_SIMPLE_BUFFER_QUEUE_HPP
+
+#include <SLES/OpenSLES_Android.h>
+
+namespace SLES {
+	/**
+	 * OO wrapper for an OpenSL/ES SLAndroidSimpleBufferQueueItf
+	 * variable.
+	 */
+	class AndroidSimpleBufferQueue {
+		SLAndroidSimpleBufferQueueItf queue;
+
+	public:
+		AndroidSimpleBufferQueue() = default;
+		explicit AndroidSimpleBufferQueue(SLAndroidSimpleBufferQueueItf _queue)
+			:queue(_queue) {}
+
+		SLresult Enqueue(const void *pBuffer, SLuint32 size) {
+			return (*queue)->Enqueue(queue, pBuffer, size);
+		}
+
+		SLresult Clear() {
+			return (*queue)->Clear(queue);
+		}
+
+		SLresult GetState(SLAndroidSimpleBufferQueueState *pState) {
+			return (*queue)->GetState(queue, pState);
+		}
+
+		SLresult RegisterCallback(slAndroidSimpleBufferQueueCallback callback,
+					  void *pContext) {
+			return (*queue)->RegisterCallback(queue, callback, pContext);
+		}
+	};
+}
+
+#endif
diff --git a/src/output/plugins/sles/Engine.hxx b/src/output/plugins/sles/Engine.hxx
new file mode 100644
index 000000000..7c6e3cf50
--- /dev/null
+++ b/src/output/plugins/sles/Engine.hxx
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the
+ * distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ * FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE
+ * FOUNDATION OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+ * STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
+ * OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef SLES_ENGINE_HPP
+#define SLES_ENGINE_HPP
+
+#include <SLES/OpenSLES.h>
+
+namespace SLES {
+	/**
+	 * OO wrapper for an OpenSL/ES SLEngineItf variable.
+	 */
+	class Engine {
+		SLEngineItf engine;
+
+	public:
+		Engine() = default;
+		explicit Engine(SLEngineItf _engine):engine(_engine) {}
+
+		SLresult CreateAudioPlayer(SLObjectItf *pPlayer,
+					   SLDataSource *pAudioSrc, SLDataSink *pAudioSnk,
+					   SLuint32 numInterfaces,
+					   const SLInterfaceID *pInterfaceIds,
+					   const SLboolean *pInterfaceRequired) {
+			return (*engine)->CreateAudioPlayer(engine, pPlayer,
+							    pAudioSrc, pAudioSnk,
+							    numInterfaces, pInterfaceIds,
+							    pInterfaceRequired);
+		}
+
+		SLresult CreateOutputMix(SLObjectItf *pMix,
+					 SLuint32 numInterfaces,
+					 const SLInterfaceID *pInterfaceIds,
+					 const SLboolean *pInterfaceRequired) {
+			return (*engine)->CreateOutputMix(engine, pMix,
+							  numInterfaces, pInterfaceIds,
+							  pInterfaceRequired);
+		}
+	};
+}
+
+#endif
diff --git a/src/output/plugins/sles/Object.hxx b/src/output/plugins/sles/Object.hxx
new file mode 100644
index 000000000..852d62d0d
--- /dev/null
+++ b/src/output/plugins/sles/Object.hxx
@@ -0,0 +1,64 @@
+/*
+ * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the
+ * distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ * FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE
+ * FOUNDATION OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+ * STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
+ * OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef SLES_OBJECT_HPP
+#define SLES_OBJECT_HPP
+
+#include <SLES/OpenSLES.h>
+
+namespace SLES {
+	/**
+	 * OO wrapper for an OpenSL/ES SLObjectItf variable.
+	 */
+	class Object {
+		SLObjectItf object;
+
+	public:
+		Object() = default;
+		explicit Object(SLObjectItf _object):object(_object) {}
+
+		operator SLObjectItf() {
+			return object;
+		}
+
+		SLresult Realize(bool async) {
+			return (*object)->Realize(object, async);
+		}
+
+		void Destroy() {
+			(*object)->Destroy(object);
+		}
+
+		SLresult GetInterface(const SLInterfaceID iid, void *pInterface) {
+			return (*object)->GetInterface(object, iid, pInterface);
+		}
+	};
+}
+
+#endif
diff --git a/src/output/plugins/sles/Play.hxx b/src/output/plugins/sles/Play.hxx
new file mode 100644
index 000000000..c760151ef
--- /dev/null
+++ b/src/output/plugins/sles/Play.hxx
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2011-2012 Max Kellermann <max@duempel.org>
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the
+ * distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
+ * FOR A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE
+ * FOUNDATION OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT,
+ * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
+ * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT,
+ * STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED
+ * OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef SLES_PLAY_HPP
+#define SLES_PLAY_HPP
+
+#include <SLES/OpenSLES.h>
+
+namespace SLES {
+	/**
+	 * OO wrapper for an OpenSL/ES SLPlayItf variable.
+	 */
+	class Play {
+		SLPlayItf play;
+
+	public:
+		Play() = default;
+		explicit Play(SLPlayItf _play):play(_play) {}
+
+		SLresult SetPlayState(SLuint32 state) {
+			return (*play)->SetPlayState(play, state);
+		}
+	};
+}
+
+#endif
diff --git a/src/output/plugins/sles/SlesOutputPlugin.cxx b/src/output/plugins/sles/SlesOutputPlugin.cxx
new file mode 100644
index 000000000..85fd9f2f2
--- /dev/null
+++ b/src/output/plugins/sles/SlesOutputPlugin.cxx
@@ -0,0 +1,539 @@
+/*
+ * 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.
+ */
+
+#include "config.h"
+#include "SlesOutputPlugin.hxx"
+#include "Object.hxx"
+#include "Engine.hxx"
+#include "Play.hxx"
+#include "AndroidSimpleBufferQueue.hxx"
+#include "../../OutputAPI.hxx"
+#include "util/Macros.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "system/ByteOrder.hxx"
+#include "Log.hxx"
+
+#include <SLES/OpenSLES.h>
+#include <SLES/OpenSLES_Android.h>
+
+class SlesOutput {
+	static constexpr unsigned N_BUFFERS = 3;
+	static constexpr size_t BUFFER_SIZE = 65536;
+
+	AudioOutput base;
+
+	SLES::Object engine_object, mix_object, play_object;
+	SLES::Play play;
+	SLES::AndroidSimpleBufferQueue queue;
+
+	/**
+	 * This mutex protects the attributes "next" and "filled".  It
+	 * is only needed while playback is launched, when the initial
+	 * buffers are being enqueued in the caller thread, while
+	 * another thread may invoke the registered callback.
+	 */
+	Mutex mutex;
+
+	Cond cond;
+
+	bool pause, cancel;
+
+	/**
+	 * The number of buffers queued to OpenSLES.
+	 */
+	unsigned n_queued;
+
+	/**
+	 * The index of the next buffer to be enqueued.
+	 */
+	unsigned next;
+
+	/**
+	 * Does the "next" buffer already contain synthesised samples?
+	 * This can happen when PCMSynthesiser::Synthesise() has been
+	 * called, but the OpenSL/ES buffer queue was full.  The
+	 * buffer will then be postponed.
+	 */
+	unsigned filled;
+
+	/**
+	 * An array of buffers.  It's one more than being managed by
+	 * OpenSL/ES, and the one not enqueued (see attribute #next)
+	 * will be written to.
+	 */
+	uint8_t buffers[N_BUFFERS][BUFFER_SIZE];
+
+public:
+	SlesOutput()
+		:base(sles_output_plugin) {}
+
+	operator AudioOutput *() {
+		return &base;
+	}
+
+	bool Initialize(const config_param &param, Error &error) {
+		return base.Configure(param, error);
+	}
+
+	bool Configure(const config_param &param, Error &error);
+
+	bool Open(AudioFormat &audio_format, Error &error);
+	void Close();
+
+	unsigned Delay() {
+		return pause && !cancel ? 100 : 0;
+	}
+
+	size_t Play(const void *chunk, size_t size, Error &error);
+
+	void Drain();
+	void Cancel();
+	bool Pause();
+
+private:
+	void PlayedCallback();
+
+	/**
+	 * OpenSL/ES callback which gets invoked when a buffer has
+	 * been consumed.  It synthesises and enqueues the next
+	 * buffer.
+	 */
+	static void PlayedCallback(gcc_unused SLAndroidSimpleBufferQueueItf caller,
+				   void *pContext)
+	{
+		SlesOutput &sles = *(SlesOutput *)pContext;
+		sles.PlayedCallback();
+	}
+};
+
+static constexpr Domain sles_domain("sles");
+
+inline bool
+SlesOutput::Configure(const config_param &, Error &)
+{
+	return true;
+}
+
+inline bool
+SlesOutput::Open(AudioFormat &audio_format, Error &error)
+{
+	SLresult result;
+	SLObjectItf _object;
+
+	result = slCreateEngine(&_object, 0, nullptr, 0,
+				nullptr, nullptr);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result), "slCreateEngine() failed");
+		return false;
+	}
+
+	engine_object = SLES::Object(_object);
+
+	result = engine_object.Realize(false);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result), "Engine.Realize() failed");
+		engine_object.Destroy();
+		return false;
+	}
+
+	SLEngineItf _engine;
+	result = engine_object.GetInterface(SL_IID_ENGINE, &_engine);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Engine.GetInterface(IID_ENGINE) failed");
+		engine_object.Destroy();
+		return false;
+	}
+
+	SLES::Engine engine(_engine);
+
+	result = engine.CreateOutputMix(&_object, 0, nullptr, nullptr);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Engine.CreateOutputMix() failed");
+		engine_object.Destroy();
+		return false;
+	}
+
+	mix_object = SLES::Object(_object);
+
+	result = mix_object.Realize(false);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Mix.Realize() failed");
+		mix_object.Destroy();
+		engine_object.Destroy();
+		return false;
+	}
+
+	SLDataLocator_AndroidSimpleBufferQueue loc_bufq = {
+		SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
+		N_BUFFERS,
+	};
+
+	if (audio_format.channels > 2)
+		audio_format.channels = 1;
+
+	SLDataFormat_PCM format_pcm;
+	format_pcm.formatType = SL_DATAFORMAT_PCM;
+	format_pcm.numChannels = audio_format.channels;
+	/* from the Android NDK docs: "Note that the field samplesPerSec is
+	   actually in units of milliHz, despite the misleading name." */
+	format_pcm.samplesPerSec = audio_format.sample_rate * 1000u;
+	format_pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
+	format_pcm.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
+	format_pcm.channelMask = audio_format.channels == 1
+		? SL_SPEAKER_FRONT_CENTER
+		: SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
+	format_pcm.endianness = IsLittleEndian()
+		? SL_BYTEORDER_LITTLEENDIAN
+		: SL_BYTEORDER_BIGENDIAN;
+
+	SLDataSource audioSrc = { &loc_bufq, &format_pcm };
+
+	SLDataLocator_OutputMix loc_outmix = {
+		SL_DATALOCATOR_OUTPUTMIX,
+		mix_object,
+	};
+
+	SLDataSink audioSnk = {
+		&loc_outmix,
+		nullptr,
+	};
+
+	const SLInterfaceID ids2[] = {
+		SL_IID_PLAY,
+		SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+		SL_IID_ANDROIDCONFIGURATION,
+	};
+
+	static constexpr SLboolean req2[] = {
+		SL_BOOLEAN_TRUE,
+		SL_BOOLEAN_TRUE,
+		SL_BOOLEAN_TRUE,
+	};
+
+	result = engine.CreateAudioPlayer(&_object, &audioSrc, &audioSnk,
+					  ARRAY_SIZE(ids2), ids2, req2);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Engine.CreateAudioPlayer() failed");
+		mix_object.Destroy();
+		engine_object.Destroy();
+		return false;
+	}
+
+	play_object = SLES::Object(_object);
+
+	SLAndroidConfigurationItf android_config;
+	if (play_object.GetInterface(SL_IID_ANDROIDCONFIGURATION,
+				     &android_config) == SL_RESULT_SUCCESS) {
+		SLint32 stream_type = SL_ANDROID_STREAM_MEDIA;
+		(*android_config)->SetConfiguration(android_config,
+						    SL_ANDROID_KEY_STREAM_TYPE,
+						    &stream_type,
+						    sizeof(stream_type));
+	}
+
+	result = play_object.Realize(false);
+
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Play.Realize() failed");
+		play_object.Destroy();
+		mix_object.Destroy();
+		engine_object.Destroy();
+		return false;
+	}
+
+	SLPlayItf _play;
+	result = play_object.GetInterface(SL_IID_PLAY, &_play);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Play.GetInterface(IID_PLAY) failed");
+		play_object.Destroy();
+		mix_object.Destroy();
+		engine_object.Destroy();
+		return false;
+	}
+
+	play = SLES::Play(_play);
+
+	SLAndroidSimpleBufferQueueItf _queue;
+	result = play_object.GetInterface(SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
+					  &_queue);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Play.GetInterface(IID_ANDROIDSIMPLEBUFFERQUEUE) failed");
+		play_object.Destroy();
+		mix_object.Destroy();
+		engine_object.Destroy();
+		return false;
+	}
+
+	queue = SLES::AndroidSimpleBufferQueue(_queue);
+	result = queue.RegisterCallback(PlayedCallback, (void *)this);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Play.RegisterCallback() failed");
+		play_object.Destroy();
+		mix_object.Destroy();
+		engine_object.Destroy();
+		return false;
+	}
+
+	result = play.SetPlayState(SL_PLAYSTATE_PLAYING);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "Play.SetPlayState(PLAYING) failed");
+		play_object.Destroy();
+		mix_object.Destroy();
+		engine_object.Destroy();
+		return false;
+	}
+
+	pause = cancel = false;
+	n_queued = 0;
+	next = 0;
+	filled = 0;
+
+	// TODO: support other sample formats
+	audio_format.format = SampleFormat::S16;
+
+	return true;
+}
+
+inline void
+SlesOutput::Close()
+{
+	play.SetPlayState(SL_PLAYSTATE_STOPPED);
+	play_object.Destroy();
+	mix_object.Destroy();
+	engine_object.Destroy();
+}
+
+inline size_t
+SlesOutput::Play(const void *chunk, size_t size, Error &error)
+{
+	cancel = false;
+
+	if (pause) {
+		SLresult result = play.SetPlayState(SL_PLAYSTATE_PLAYING);
+		if (result != SL_RESULT_SUCCESS) {
+			error.Set(sles_domain, int(result),
+				  "Play.SetPlayState(PLAYING) failed");
+			return false;
+		}
+
+		pause = false;
+	}
+
+	const ScopeLock protect(mutex);
+
+	assert(filled < BUFFER_SIZE);
+
+	while (n_queued == N_BUFFERS) {
+		assert(filled == 0);
+		cond.wait(mutex);
+	}
+
+	size_t nbytes = std::min(BUFFER_SIZE - filled, size);
+	memcpy(buffers[next] + filled, chunk, nbytes);
+	filled += nbytes;
+	if (filled < BUFFER_SIZE)
+		return nbytes;
+
+	SLresult result = queue.Enqueue(buffers[next], BUFFER_SIZE);
+	if (result != SL_RESULT_SUCCESS) {
+		error.Set(sles_domain, int(result),
+			  "AndroidSimpleBufferQueue.Enqueue() failed");
+		return 0;
+	}
+
+	++n_queued;
+	next = (next + 1) % N_BUFFERS;
+	filled = 0;
+
+	return nbytes;
+}
+
+inline void
+SlesOutput::Drain()
+{
+	const ScopeLock protect(mutex);
+
+	assert(filled < BUFFER_SIZE);
+
+	while (n_queued > 0)
+		cond.wait(mutex);
+}
+
+inline void
+SlesOutput::Cancel()
+{
+	pause = true;
+	cancel = true;
+
+	SLresult result = play.SetPlayState(SL_PLAYSTATE_PAUSED);
+	if (result != SL_RESULT_SUCCESS)
+		FormatError(sles_domain,  "Play.SetPlayState(PAUSED) failed");
+
+	result = queue.Clear();
+	if (result != SL_RESULT_SUCCESS)
+		FormatWarning(sles_domain,
+			      "AndroidSimpleBufferQueue.Clear() failed");
+
+	const ScopeLock protect(mutex);
+	n_queued = 0;
+	filled = 0;
+}
+
+inline bool
+SlesOutput::Pause()
+{
+	cancel = false;
+
+	if (pause)
+		return true;
+
+	pause = true;
+
+	SLresult result = play.SetPlayState(SL_PLAYSTATE_PAUSED);
+	if (result != SL_RESULT_SUCCESS) {
+		FormatError(sles_domain, "Play.SetPlayState(PAUSED) failed");
+		return false;
+	}
+
+	return true;
+}
+
+inline void
+SlesOutput::PlayedCallback()
+{
+	const ScopeLock protect(mutex);
+	assert(n_queued > 0);
+	--n_queued;
+	cond.signal();
+}
+
+static bool
+sles_test_default_device()
+{
+	/* this is the default output plugin on Android, and it should
+	   be available in any case */
+	return true;
+}
+
+static AudioOutput *
+sles_output_init(const config_param &param, Error &error)
+{
+	SlesOutput *sles = new SlesOutput();
+
+	if (!sles->Initialize(param, error) ||
+	    !sles->Configure(param, error)) {
+		delete sles;
+		return nullptr;
+	}
+
+	return *sles;
+}
+
+static void
+sles_output_finish(AudioOutput *ao)
+{
+	SlesOutput *sles = (SlesOutput *)ao;
+
+	delete sles;
+}
+
+static bool
+sles_output_open(AudioOutput *ao, AudioFormat &audio_format, Error &error)
+{
+	SlesOutput &sles = *(SlesOutput *)ao;
+
+	return sles.Open(audio_format, error);
+}
+
+static void
+sles_output_close(AudioOutput *ao)
+{
+	SlesOutput &sles = *(SlesOutput *)ao;
+
+	sles.Close();
+}
+
+static unsigned
+sles_output_delay(AudioOutput *ao)
+{
+	SlesOutput &sles = *(SlesOutput *)ao;
+
+	return sles.Delay();
+}
+
+static size_t
+sles_output_play(AudioOutput *ao, const void *chunk, size_t size,
+		 Error &error)
+{
+	SlesOutput &sles = *(SlesOutput *)ao;
+
+	return sles.Play(chunk, size, error);
+}
+
+static void
+sles_output_drain(AudioOutput *ao)
+{
+	SlesOutput &sles = *(SlesOutput *)ao;
+
+	sles.Drain();
+}
+
+static void
+sles_output_cancel(AudioOutput *ao)
+{
+	SlesOutput &sles = *(SlesOutput *)ao;
+
+	sles.Cancel();
+}
+
+static bool
+sles_output_pause(AudioOutput *ao)
+{
+	SlesOutput &sles = *(SlesOutput *)ao;
+
+	return sles.Pause();
+}
+
+const struct AudioOutputPlugin sles_output_plugin = {
+	"sles",
+	sles_test_default_device,
+	sles_output_init,
+	sles_output_finish,
+	nullptr,
+	nullptr,
+	sles_output_open,
+	sles_output_close,
+	sles_output_delay,
+	nullptr,
+	sles_output_play,
+	sles_output_drain,
+	sles_output_cancel,
+	sles_output_pause,
+	nullptr,
+};
diff --git a/src/output/plugins/sles/SlesOutputPlugin.hxx b/src/output/plugins/sles/SlesOutputPlugin.hxx
new file mode 100644
index 000000000..5424dec2e
--- /dev/null
+++ b/src/output/plugins/sles/SlesOutputPlugin.hxx
@@ -0,0 +1,25 @@
+/*
+ * 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_SLES_OUTPUT_PLUGIN_HXX
+#define MPD_SLES_OUTPUT_PLUGIN_HXX
+
+extern const struct AudioOutputPlugin sles_output_plugin;
+
+#endif