43 files changed, 2985 insertions, 3728 deletions

diff --git a/src/output/alsa_output_plugin.c b/src/output/AlsaOutputPlugin.cxx
index d8b184273..b26a3e1df 100644
--- a/src/output/alsa_output_plugin.c
+++ b/src/output/AlsaOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,14 +18,17 @@
  */
 
 #include "config.h"
-#include "alsa_output_plugin.h"
-#include "output_api.h"
-#include "mixer_list.h"
-#include "pcm_export.h"
+#include "AlsaOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "MixerList.hxx"
+#include "pcm/PcmExport.hxx"
+#include "util/Manual.hxx"
 
 #include <glib.h>
 #include <alsa/asoundlib.h>
 
+#include <string>
+
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "alsa"
 
@@ -34,23 +37,23 @@
 
 static const char default_device[] = "default";
 
-enum {
-	MPD_ALSA_BUFFER_TIME_US = 500000,
-};
+static constexpr unsigned MPD_ALSA_BUFFER_TIME_US = 500000;
 
 #define 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 alsa_data {
+struct AlsaOutput {
 	struct audio_output base;
 
-	struct pcm_export_state export;
+	Manual<PcmExport> pcm_export;
 
-	/** the configured name of the ALSA device; NULL for the
-	    default device */
-	char *device;
+	/**
+	 * The configured name of the ALSA device; empty for the
+	 * default device
+	 */
+	std::string device;
 
 	/** use memory mapped I/O? */
 	bool use_mmap;
@@ -101,6 +104,25 @@ struct alsa_data {
 	 * The number of frames written in the current period.
 	 */
 	snd_pcm_uframes_t period_position;
+
+	/**
+	 * This buffer gets allocated after opening the ALSA device.
+	 * It contains silence samples, enough to fill one period (see
+	 * #period_frames).
+	 */
+	void *silence;
+
+	AlsaOutput():mode(0), writei(snd_pcm_writei) {
+	}
+
+	bool Init(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &alsa_output_plugin,
+				    param, error_r);
+	}
+
+	void Deinit() {
+		ao_base_finish(&base);
+	}
 };
 
 /**
@@ -113,58 +135,47 @@ alsa_output_quark(void)
 }
 
 static const char *
-alsa_device(const struct alsa_data *ad)
+alsa_device(const AlsaOutput *ad)
 {
-	return ad->device != NULL ? ad->device : default_device;
-}
-
-static struct alsa_data *
-alsa_data_new(void)
-{
-	struct alsa_data *ret = g_new(struct alsa_data, 1);
-
-	ret->mode = 0;
-	ret->writei = snd_pcm_writei;
-
-	return ret;
+	return ad->device.empty() ? default_device : ad->device.c_str();
 }
 
 static void
-alsa_configure(struct alsa_data *ad, const struct config_param *param)
+alsa_configure(AlsaOutput *ad, const config_param &param)
 {
-	ad->device = config_dup_block_string(param, "device", NULL);
+	ad->device = param.GetBlockValue("device", "");
 
-	ad->use_mmap = config_get_block_bool(param, "use_mmap", false);
+	ad->use_mmap = param.GetBlockValue("use_mmap", false);
 
-	ad->dsd_usb = config_get_block_bool(param, "dsd_usb", false);
+	ad->dsd_usb = param.GetBlockValue("dsd_usb", false);
 
-	ad->buffer_time = config_get_block_unsigned(param, "buffer_time",
-			MPD_ALSA_BUFFER_TIME_US);
-	ad->period_time = config_get_block_unsigned(param, "period_time", 0);
+	ad->buffer_time = param.GetBlockValue("buffer_time",
+					      MPD_ALSA_BUFFER_TIME_US);
+	ad->period_time = param.GetBlockValue("period_time", 0u);
 
 #ifdef SND_PCM_NO_AUTO_RESAMPLE
-	if (!config_get_block_bool(param, "auto_resample", true))
+	if (!param.GetBlockValue("auto_resample", true))
 		ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
 #endif
 
 #ifdef SND_PCM_NO_AUTO_CHANNELS
-	if (!config_get_block_bool(param, "auto_channels", true))
+	if (!param.GetBlockValue("auto_channels", true))
 		ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
 #endif
 
 #ifdef SND_PCM_NO_AUTO_FORMAT
-	if (!config_get_block_bool(param, "auto_format", true))
+	if (!param.GetBlockValue("auto_format", true))
 		ad->mode |= SND_PCM_NO_AUTO_FORMAT;
 #endif
 }
 
 static struct audio_output *
-alsa_init(const struct config_param *param, GError **error_r)
+alsa_init(const config_param &param, GError **error_r)
 {
-	struct alsa_data *ad = alsa_data_new();
+	AlsaOutput *ad = new AlsaOutput();
 
-	if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
-		g_free(ad);
+	if (!ad->Init(param, error_r)) {
+		delete ad;
 		return NULL;
 	}
 
@@ -176,12 +187,10 @@ alsa_init(const struct config_param *param, GError **error_r)
 static void
 alsa_finish(struct audio_output *ao)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
-
-	ao_base_finish(&ad->base);
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
-	g_free(ad->device);
-	g_free(ad);
+	ad->Deinit();
+	delete ad;
 
 	/* free libasound's config cache */
 	snd_config_update_free_global();
@@ -190,18 +199,18 @@ alsa_finish(struct audio_output *ao)
 static bool
 alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
-	pcm_export_init(&ad->export);
+	ad->pcm_export.Construct();
 	return true;
 }
 
 static void
 alsa_output_disable(struct audio_output *ao)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
-	pcm_export_deinit(&ad->export);
+	ad->pcm_export.Destruct();
 }
 
 static bool
@@ -222,31 +231,31 @@ alsa_test_default_device(void)
 }
 
 static snd_pcm_format_t
-get_bitformat(enum sample_format sample_format)
+get_bitformat(SampleFormat sample_format)
 {
 	switch (sample_format) {
-	case SAMPLE_FORMAT_UNDEFINED:
-	case SAMPLE_FORMAT_DSD:
+	case SampleFormat::UNDEFINED:
+	case SampleFormat::DSD:
 		return SND_PCM_FORMAT_UNKNOWN;
 
-	case SAMPLE_FORMAT_S8:
+	case SampleFormat::S8:
 		return SND_PCM_FORMAT_S8;
 
-	case SAMPLE_FORMAT_S16:
+	case SampleFormat::S16:
 		return SND_PCM_FORMAT_S16;
 
-	case SAMPLE_FORMAT_S24_P32:
+	case SampleFormat::S24_P32:
 		return SND_PCM_FORMAT_S24;
 
-	case SAMPLE_FORMAT_S32:
+	case SampleFormat::S32:
 		return SND_PCM_FORMAT_S32;
 
-	case SAMPLE_FORMAT_FLOAT:
+	case SampleFormat::FLOAT:
 		return SND_PCM_FORMAT_FLOAT;
 	}
 
 	assert(false);
-	return SND_PCM_FORMAT_UNKNOWN;
+	gcc_unreachable();
 }
 
 static snd_pcm_format_t
@@ -313,7 +322,7 @@ alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
  */
 static int
 alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
-		       enum sample_format sample_format,
+		       SampleFormat sample_format,
 		       bool *packed_r, bool *reverse_endian_r)
 {
 	snd_pcm_format_t alsa_format = get_bitformat(sample_format);
@@ -344,35 +353,36 @@ alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
  */
 static int
 alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
-			 struct audio_format *audio_format,
+			 AudioFormat &audio_format,
 			 bool *packed_r, bool *reverse_endian_r)
 {
 	/* try the input format first */
 
-	int err = alsa_output_try_format(pcm, hwparams, audio_format->format,
+	int err = alsa_output_try_format(pcm, hwparams,
+					 audio_format.format,
 					 packed_r, reverse_endian_r);
 
 	/* if unsupported by the hardware, try other formats */
 
-	static const enum sample_format probe_formats[] = {
-		SAMPLE_FORMAT_S24_P32,
-		SAMPLE_FORMAT_S32,
-		SAMPLE_FORMAT_S16,
-		SAMPLE_FORMAT_S8,
-		SAMPLE_FORMAT_UNDEFINED,
+	static const SampleFormat probe_formats[] = {
+		SampleFormat::S24_P32,
+		SampleFormat::S32,
+		SampleFormat::S16,
+		SampleFormat::S8,
+		SampleFormat::UNDEFINED,
 	};
 
 	for (unsigned i = 0;
-	     err == -EINVAL && probe_formats[i] != SAMPLE_FORMAT_UNDEFINED;
+	     err == -EINVAL && probe_formats[i] != SampleFormat::UNDEFINED;
 	     ++i) {
-		const enum sample_format mpd_format = probe_formats[i];
-		if (mpd_format == audio_format->format)
+		const SampleFormat mpd_format = probe_formats[i];
+		if (mpd_format == audio_format.format)
 			continue;
 
 		err = alsa_output_try_format(pcm, hwparams, mpd_format,
 					     packed_r, reverse_endian_r);
 		if (err == 0)
-			audio_format->format = mpd_format;
+			audio_format.format = mpd_format;
 	}
 
 	return err;
@@ -383,15 +393,11 @@ alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
  * the configured settings and the audio format.
  */
 static bool
-alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
+alsa_setup(AlsaOutput *ad, AudioFormat &audio_format,
 	   bool *packed_r, bool *reverse_endian_r, GError **error)
 {
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_sw_params_t *swparams;
-	unsigned int sample_rate = audio_format->sample_rate;
-	unsigned int channels = audio_format->channels;
-	snd_pcm_uframes_t alsa_buffer_size;
-	snd_pcm_uframes_t alsa_period_size;
+	unsigned int sample_rate = audio_format.sample_rate;
+	unsigned int channels = audio_format.channels;
 	int err;
 	const char *cmd = NULL;
 	int retry = MPD_ALSA_RETRY_NR;
@@ -401,6 +407,7 @@ alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
 	period_time_ro = period_time = ad->period_time;
 configure_hw:
 	/* configure HW params */
+	snd_pcm_hw_params_t *hwparams;
 	snd_pcm_hw_params_alloca(&hwparams);
 	cmd = "snd_pcm_hw_params_any";
 	err = snd_pcm_hw_params_any(ad->pcm, hwparams);
@@ -434,7 +441,7 @@ configure_hw:
 		g_set_error(error, alsa_output_quark(), err,
 			    "ALSA device \"%s\" does not support format %s: %s",
 			    alsa_device(ad),
-			    sample_format_to_string(audio_format->format),
+			    sample_format_to_string(audio_format.format),
 			    snd_strerror(-err));
 		return false;
 	}
@@ -449,21 +456,21 @@ configure_hw:
 	if (err < 0) {
 		g_set_error(error, alsa_output_quark(), err,
 			    "ALSA device \"%s\" does not support %i channels: %s",
-			    alsa_device(ad), (int)audio_format->channels,
+			    alsa_device(ad), (int)audio_format.channels,
 			    snd_strerror(-err));
 		return false;
 	}
-	audio_format->channels = (int8_t)channels;
+	audio_format.channels = (int8_t)channels;
 
 	err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
 					      &sample_rate, NULL);
 	if (err < 0 || sample_rate == 0) {
 		g_set_error(error, alsa_output_quark(), err,
 			    "ALSA device \"%s\" does not support %u Hz audio",
-			    alsa_device(ad), audio_format->sample_rate);
+			    alsa_device(ad), audio_format.sample_rate);
 		return false;
 	}
-	audio_format->sample_rate = sample_rate;
+	audio_format.sample_rate = sample_rate;
 
 	snd_pcm_uframes_t buffer_size_min, buffer_size_max;
 	snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
@@ -525,11 +532,13 @@ configure_hw:
 	if (retry != MPD_ALSA_RETRY_NR)
 		g_debug("ALSA period_time set to %d\n", period_time);
 
+	snd_pcm_uframes_t alsa_buffer_size;
 	cmd = "snd_pcm_hw_params_get_buffer_size";
 	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
 	if (err < 0)
 		goto error;
 
+	snd_pcm_uframes_t alsa_period_size;
 	cmd = "snd_pcm_hw_params_get_period_size";
 	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
 						NULL);
@@ -537,6 +546,7 @@ configure_hw:
 		goto error;
 
 	/* configure SW params */
+	snd_pcm_sw_params_t *swparams;
 	snd_pcm_sw_params_alloca(&swparams);
 
 	cmd = "snd_pcm_sw_params_current";
@@ -576,6 +586,11 @@ 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));
+	snd_pcm_format_set_silence(format, ad->silence,
+				   alsa_period_size * channels);
+
 	return true;
 
 error:
@@ -586,22 +601,22 @@ error:
 }
 
 static bool
-alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
+alsa_setup_dsd(AlsaOutput *ad, const AudioFormat audio_format,
 	       bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
 	       GError **error_r)
 {
 	assert(ad->dsd_usb);
-	assert(audio_format->format == SAMPLE_FORMAT_DSD);
+	assert(audio_format.format == SampleFormat::DSD);
 
 	/* pass 24 bit to alsa_setup() */
 
-	struct audio_format usb_format = *audio_format;
-	usb_format.format = SAMPLE_FORMAT_S24_P32;
+	AudioFormat usb_format = audio_format;
+	usb_format.format = SampleFormat::S24_P32;
 	usb_format.sample_rate /= 2;
 
-	const struct audio_format check = usb_format;
+	const AudioFormat check = usb_format;
 
-	if (!alsa_setup(ad, &usb_format, packed_r, reverse_endian_r, error_r))
+	if (!alsa_setup(ad, usb_format, packed_r, reverse_endian_r, error_r))
 		return false;
 
 	/* if the device allows only 32 bit, shift all DSD-over-USB
@@ -609,16 +624,17 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
 	   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 == SAMPLE_FORMAT_S32;
-	if (usb_format.format == SAMPLE_FORMAT_S32)
-		usb_format.format = SAMPLE_FORMAT_S24_P32;
+	*shift8_r = usb_format.format == SampleFormat::S32;
+	if (usb_format.format == SampleFormat::S32)
+		usb_format.format = SampleFormat::S24_P32;
 
-	if (!audio_format_equals(&usb_format, &check)) {
+	if (usb_format != check) {
 		/* no bit-perfect playback, which is required
 		   for DSD over USB */
 		g_set_error(error_r, alsa_output_quark(), 0,
 			    "Failed to configure DSD-over-USB on ALSA device \"%s\"",
 			    alsa_device(ad));
+		g_free(ad->silence);
 		return false;
 	}
 
@@ -626,13 +642,13 @@ alsa_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
 }
 
 static bool
-alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
+alsa_setup_or_dsd(AlsaOutput *ad, AudioFormat &audio_format,
 		  GError **error_r)
 {
 	bool shift8 = false, packed, reverse_endian;
 
 	const bool dsd_usb = ad->dsd_usb &&
-		audio_format->format == SAMPLE_FORMAT_DSD;
+		audio_format.format == SampleFormat::DSD;
 	const bool success = dsd_usb
 		? alsa_setup_dsd(ad, audio_format,
 				 &shift8, &packed, &reverse_endian,
@@ -642,21 +658,19 @@ alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
 	if (!success)
 		return false;
 
-	pcm_export_open(&ad->export,
-			audio_format->format, audio_format->channels,
-			dsd_usb, shift8, packed, reverse_endian);
+	ad->pcm_export->Open(audio_format.format,
+			     audio_format.channels,
+			     dsd_usb, shift8, packed, reverse_endian);
 	return true;
 }
 
 static bool
-alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
+alsa_open(struct audio_output *ao, AudioFormat &audio_format, GError **error)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
-	int err;
-	bool success;
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
-	err = snd_pcm_open(&ad->pcm, alsa_device(ad),
-			   SND_PCM_STREAM_PLAYBACK, ad->mode);
+	int err = snd_pcm_open(&ad->pcm, alsa_device(ad),
+			       SND_PCM_STREAM_PLAYBACK, ad->mode);
 	if (err < 0) {
 		g_set_error(error, alsa_output_quark(), err,
 			    "Failed to open ALSA device \"%s\": %s",
@@ -667,20 +681,28 @@ alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
 	g_debug("opened %s type=%s", snd_pcm_name(ad->pcm),
 		snd_pcm_type_name(snd_pcm_type(ad->pcm)));
 
-	success = alsa_setup_or_dsd(ad, audio_format, error);
-	if (!success) {
+	if (!alsa_setup_or_dsd(ad, audio_format, error)) {
 		snd_pcm_close(ad->pcm);
 		return false;
 	}
 
-	ad->in_frame_size = audio_format_frame_size(audio_format);
-	ad->out_frame_size = pcm_export_frame_size(&ad->export, audio_format);
+	ad->in_frame_size = audio_format.GetFrameSize();
+	ad->out_frame_size = ad->pcm_export->GetFrameSize(audio_format);
 
 	return true;
 }
 
+/**
+ * Write silence to the ALSA device.
+ */
+static void
+alsa_write_silence(AlsaOutput *ad, snd_pcm_uframes_t nframes)
+{
+	ad->writei(ad->pcm, ad->silence, nframes);
+}
+
 static int
-alsa_recover(struct alsa_data *ad, int err)
+alsa_recover(AlsaOutput *ad, int err)
 {
 	if (err == -EPIPE) {
 		g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
@@ -701,6 +723,26 @@ alsa_recover(struct alsa_data *ad, int err)
 	case SND_PCM_STATE_XRUN:
 		ad->period_position = 0;
 		err = snd_pcm_prepare(ad->pcm);
+
+		if (err == 0) {
+			/* this works around a driver bug observed on
+			   the Raspberry Pi: after snd_pcm_drop(), the
+			   whole ring buffer must be invalidated, but
+			   the snd_pcm_prepare() call above makes the
+			   driver play random data that just happens
+			   to be still in the buffer; by adding and
+			   cancelling some silence, this bug does not
+			   occur */
+			alsa_write_silence(ad, ad->period_frames);
+
+			/* cancel the silence data right away to avoid
+			   increasing latency; even though this
+			   function call invalidates the portion of
+			   silence, the driver seems to avoid the
+			   bug */
+			snd_pcm_reset(ad->pcm);
+		}
+
 		break;
 	case SND_PCM_STATE_DISCONNECTED:
 		break;
@@ -719,7 +761,7 @@ alsa_recover(struct alsa_data *ad, int err)
 static void
 alsa_drain(struct audio_output *ao)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
 	if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
 		return;
@@ -729,20 +771,7 @@ alsa_drain(struct audio_output *ao)
 		   period */
 		snd_pcm_uframes_t nframes =
 			ad->period_frames - ad->period_position;
-		size_t nbytes = nframes * ad->out_frame_size;
-		void *buffer = g_malloc(nbytes);
-		snd_pcm_hw_params_t *params;
-		snd_pcm_format_t format;
-		unsigned channels;
-
-		snd_pcm_hw_params_alloca(&params);
-		snd_pcm_hw_params_current(ad->pcm, params);
-		snd_pcm_hw_params_get_format(params, &format);
-		snd_pcm_hw_params_get_channels(params, &channels);
-
-		snd_pcm_format_set_silence(format, buffer, nframes * channels);
-		ad->writei(ad->pcm, buffer, nframes);
-		g_free(buffer);
+		alsa_write_silence(ad, nframes);
 	}
 
 	snd_pcm_drain(ad->pcm);
@@ -753,7 +782,7 @@ alsa_drain(struct audio_output *ao)
 static void
 alsa_cancel(struct audio_output *ao)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
 	ad->period_position = 0;
 
@@ -763,20 +792,21 @@ alsa_cancel(struct audio_output *ao)
 static void
 alsa_close(struct audio_output *ao)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
 	snd_pcm_close(ad->pcm);
+	g_free(ad->silence);
 }
 
 static size_t
 alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 	  GError **error)
 {
-	struct alsa_data *ad = (struct alsa_data *)ao;
+	AlsaOutput *ad = (AlsaOutput *)ao;
 
 	assert(size % ad->in_frame_size == 0);
 
-	chunk = pcm_export(&ad->export, chunk, size, &size);
+	chunk = ad->pcm_export->Export(chunk, size, size);
 
 	assert(size % ad->out_frame_size == 0);
 
@@ -789,8 +819,7 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 				% ad->period_frames;
 
 			size_t bytes_written = ret * ad->out_frame_size;
-			return pcm_export_source_size(&ad->export,
-						      bytes_written);
+			return ad->pcm_export->CalcSourceSize(bytes_written);
 		}
 
 		if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
@@ -803,17 +832,20 @@ alsa_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 const struct audio_output_plugin alsa_output_plugin = {
-	.name = "alsa",
-	.test_default_device = alsa_test_default_device,
-	.init = alsa_init,
-	.finish = alsa_finish,
-	.enable = alsa_output_enable,
-	.disable = alsa_output_disable,
-	.open = alsa_open,
-	.play = alsa_play,
-	.drain = alsa_drain,
-	.cancel = alsa_cancel,
-	.close = alsa_close,
-
-	.mixer_plugin = &alsa_mixer_plugin,
+	"alsa",
+	alsa_test_default_device,
+	alsa_init,
+	alsa_finish,
+	alsa_output_enable,
+	alsa_output_disable,
+	alsa_open,
+	alsa_close,
+	nullptr,
+	nullptr,
+	alsa_play,
+	alsa_drain,
+	alsa_cancel,
+	nullptr,
+
+	&alsa_mixer_plugin,
 };
diff --git a/src/output/alsa_output_plugin.h b/src/output/AlsaOutputPlugin.hxx
index daa1f3615..dc7e639a8 100644
--- a/src/output/alsa_output_plugin.h
+++ b/src/output/AlsaOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_ALSA_OUTPUT_PLUGIN_H
-#define MPD_ALSA_OUTPUT_PLUGIN_H
+#ifndef MPD_ALSA_OUTPUT_PLUGIN_HXX
+#define MPD_ALSA_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin alsa_output_plugin;
 
diff --git a/src/output/ao_output_plugin.c b/src/output/AoOutputPlugin.cxx
index d7e577fa4..db7b9a360 100644
--- a/src/output/ao_output_plugin.c
+++ b/src/output/AoOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,28 +18,41 @@
  */
 
 #include "config.h"
-#include "ao_output_plugin.h"
-#include "output_api.h"
+#include "AoOutputPlugin.hxx"
+#include "OutputAPI.hxx"
 
 #include <ao/ao.h>
 #include <glib.h>
 
+#include <string.h>
+
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "ao"
 
 /* An ao_sample_format, with all fields set to zero: */
-static const ao_sample_format OUR_AO_FORMAT_INITIALIZER;
+static ao_sample_format OUR_AO_FORMAT_INITIALIZER;
 
 static unsigned ao_output_ref;
 
-struct ao_data {
+struct AoOutput {
 	struct audio_output base;
 
 	size_t write_size;
 	int driver;
 	ao_option *options;
 	ao_device *device;
-} AoData;
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &ao_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
+
+	bool Configure(const config_param &param, GError **error_r);
+};
 
 static inline GQuark
 ao_output_quark(void)
@@ -81,78 +94,83 @@ ao_output_error(GError **error_r)
 		    "%s", error);
 }
 
-static struct audio_output *
-ao_output_init(const struct config_param *param,
-	       GError **error)
+inline bool
+AoOutput::Configure(const config_param &param, GError **error_r)
 {
-	struct ao_data *ad = g_new(struct ao_data, 1);
-
-	if (!ao_base_init(&ad->base, &ao_output_plugin, param, error)) {
-		g_free(ad);
-		return NULL;
-	}
-
-	ao_info *ai;
 	const char *value;
 
-	ad->options = NULL;
+	options = nullptr;
 
-	ad->write_size = config_get_block_unsigned(param, "write_size", 1024);
+	write_size = param.GetBlockValue("write_size", 1024u);
 
 	if (ao_output_ref == 0) {
 		ao_initialize();
 	}
 	ao_output_ref++;
 
-	value = config_get_block_string(param, "driver", "default");
+	value = param.GetBlockValue("driver", "default");
 	if (0 == strcmp(value, "default"))
-		ad->driver = ao_default_driver_id();
+		driver = ao_default_driver_id();
 	else
-		ad->driver = ao_driver_id(value);
+		driver = ao_driver_id(value);
 
-	if (ad->driver < 0) {
-		g_set_error(error, ao_output_quark(), 0,
+	if (driver < 0) {
+		g_set_error(error_r, ao_output_quark(), 0,
 			    "\"%s\" is not a valid ao driver",
 			    value);
-		ao_base_finish(&ad->base);
-		g_free(ad);
-		return NULL;
+		return false;
 	}
 
-	if ((ai = ao_driver_info(ad->driver)) == NULL) {
-		g_set_error(error, ao_output_quark(), 0,
+	ao_info *ai = ao_driver_info(driver);
+	if (ai == nullptr) {
+		g_set_error(error_r, ao_output_quark(), 0,
 			    "problems getting driver info");
-		ao_base_finish(&ad->base);
-		g_free(ad);
-		return NULL;
+		return false;
 	}
 
 	g_debug("using ao driver \"%s\" for \"%s\"\n", ai->short_name,
-		config_get_block_string(param, "name", NULL));
+		param.GetBlockValue("name", nullptr));
 
-	value = config_get_block_string(param, "options", NULL);
-	if (value != NULL) {
-		gchar **options = g_strsplit(value, ";", 0);
+	value = param.GetBlockValue("options", nullptr);
+	if (value != nullptr) {
+		gchar **_options = g_strsplit(value, ";", 0);
 
-		for (unsigned i = 0; options[i] != NULL; ++i) {
-			gchar **key_value = g_strsplit(options[i], "=", 2);
+		for (unsigned i = 0; _options[i] != nullptr; ++i) {
+			gchar **key_value = g_strsplit(_options[i], "=", 2);
 
-			if (key_value[0] == NULL || key_value[1] == NULL) {
-				g_set_error(error, ao_output_quark(), 0,
+			if (key_value[0] == nullptr || key_value[1] == nullptr) {
+				g_set_error(error_r, ao_output_quark(), 0,
 					    "problems parsing options \"%s\"",
-					    options[i]);
-				ao_base_finish(&ad->base);
-				g_free(ad);
-				return NULL;
+					    _options[i]);
+				return false;
 			}
 
-			ao_append_option(&ad->options, key_value[0],
+			ao_append_option(&options, key_value[0],
 					 key_value[1]);
 
 			g_strfreev(key_value);
 		}
 
-		g_strfreev(options);
+		g_strfreev(_options);
+	}
+
+	return true;
+}
+
+static struct audio_output *
+ao_output_init(const config_param &param, GError **error_r)
+{
+	AoOutput *ad = new AoOutput();
+
+	if (!ad->Initialize(param, error_r)) {
+		delete ad;
+		return nullptr;
+	}
+
+	if (!ad->Configure(param, error_r)) {
+		ad->Deinitialize();
+		delete ad;
+		return nullptr;
 	}
 
 	return &ad->base;
@@ -161,11 +179,11 @@ ao_output_init(const struct config_param *param,
 static void
 ao_output_finish(struct audio_output *ao)
 {
-	struct ao_data *ad = (struct ao_data *)ao;
+	AoOutput *ad = (AoOutput *)ao;
 
 	ao_free_options(ad->options);
-	ao_base_finish(&ad->base);
-	g_free(ad);
+	ad->Deinitialize();
+	delete ad;
 
 	ao_output_ref--;
 
@@ -176,24 +194,24 @@ ao_output_finish(struct audio_output *ao)
 static void
 ao_output_close(struct audio_output *ao)
 {
-	struct ao_data *ad = (struct ao_data *)ao;
+	AoOutput *ad = (AoOutput *)ao;
 
 	ao_close(ad->device);
 }
 
 static bool
-ao_output_open(struct audio_output *ao, struct audio_format *audio_format,
+ao_output_open(struct audio_output *ao, AudioFormat &audio_format,
 	       GError **error)
 {
 	ao_sample_format format = OUR_AO_FORMAT_INITIALIZER;
-	struct ao_data *ad = (struct ao_data *)ao;
+	AoOutput *ad = (AoOutput *)ao;
 
-	switch (audio_format->format) {
-	case SAMPLE_FORMAT_S8:
+	switch (audio_format.format) {
+	case SampleFormat::S8:
 		format.bits = 8;
 		break;
 
-	case SAMPLE_FORMAT_S16:
+	case SampleFormat::S16:
 		format.bits = 16;
 		break;
 
@@ -201,18 +219,18 @@ ao_output_open(struct audio_output *ao, struct audio_format *audio_format,
 		/* support for 24 bit samples in libao is currently
 		   dubious, and until we have sorted that out,
 		   convert everything to 16 bit */
-		audio_format->format = SAMPLE_FORMAT_S16;
+		audio_format.format = SampleFormat::S16;
 		format.bits = 16;
 		break;
 	}
 
-	format.rate = audio_format->sample_rate;
+	format.rate = audio_format.sample_rate;
 	format.byte_format = AO_FMT_NATIVE;
-	format.channels = audio_format->channels;
+	format.channels = audio_format.channels;
 
 	ad->device = ao_open_live(ad->driver, &format, ad->options);
 
-	if (ad->device == NULL) {
+	if (ad->device == nullptr) {
 		ao_output_error(error);
 		return false;
 	}
@@ -230,7 +248,7 @@ static int ao_play_deconst(ao_device *device, const void *output_samples,
 {
 	union {
 		const void *in;
-		void *out;
+		char *out;
 	} u;
 
 	u.in = output_samples;
@@ -241,7 +259,7 @@ static size_t
 ao_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	       GError **error)
 {
-	struct ao_data *ad = (struct ao_data *)ao;
+	AoOutput *ad = (AoOutput *)ao;
 
 	if (size > ad->write_size)
 		size = ad->write_size;
@@ -255,10 +273,19 @@ ao_output_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 const struct audio_output_plugin ao_output_plugin = {
-	.name = "ao",
-	.init = ao_output_init,
-	.finish = ao_output_finish,
-	.open = ao_output_open,
-	.close = ao_output_close,
-	.play = ao_output_play,
+	"ao",
+	nullptr,
+	ao_output_init,
+	ao_output_finish,
+	nullptr,
+	nullptr,
+	ao_output_open,
+	ao_output_close,
+	nullptr,
+	nullptr,
+	ao_output_play,
+	nullptr,
+	nullptr,
+	nullptr,
+	nullptr,
 };
diff --git a/src/output/ao_output_plugin.h b/src/output/AoOutputPlugin.hxx
index 9a3a47c05..a44885e56 100644
--- a/src/output/ao_output_plugin.h
+++ b/src/output/AoOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_AO_OUTPUT_PLUGIN_H
-#define MPD_AO_OUTPUT_PLUGIN_H
+#ifndef MPD_AO_OUTPUT_PLUGIN_HXX
+#define MPD_AO_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin ao_output_plugin;
 
diff --git a/src/output/fifo_output_plugin.c b/src/output/FifoOutputPlugin.cxx
index 022be0b4a..50062988c 100644
--- a/src/output/fifo_output_plugin.c
+++ b/src/output/FifoOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,10 +18,9 @@
  */
 
 #include "config.h"
-#include "fifo_output_plugin.h"
-#include "output_api.h"
-#include "utils.h"
-#include "timer.h"
+#include "FifoOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "Timer.hxx"
 #include "fd_util.h"
 #include "open.h"
 
@@ -38,14 +37,37 @@
 
 #define FIFO_BUFFER_SIZE 65536 /* pipe capacity on Linux >= 2.6.11 */
 
-struct fifo_data {
+struct FifoOutput {
 	struct audio_output base;
 
 	char *path;
 	int input;
 	int output;
 	bool created;
-	struct timer *timer;
+	Timer *timer;
+
+	FifoOutput()
+		:path(nullptr), input(-1), output(-1), created(false) {}
+
+	~FifoOutput() {
+		g_free(path);
+	}
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &fifo_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
+
+	bool Create(GError **error_r);
+	bool Check(GError **error_r);
+	void Delete();
+
+	bool Open(GError **error_r);
+	void Close();
 };
 
 /**
@@ -57,156 +79,138 @@ fifo_output_quark(void)
 	return g_quark_from_static_string("fifo_output");
 }
 
-static struct fifo_data *fifo_data_new(void)
-{
-	struct fifo_data *ret;
-
-	ret = g_new(struct fifo_data, 1);
-
-	ret->path = NULL;
-	ret->input = -1;
-	ret->output = -1;
-	ret->created = false;
-
-	return ret;
-}
-
-static void fifo_data_free(struct fifo_data *fd)
+inline void
+FifoOutput::Delete()
 {
-	g_free(fd->path);
-	g_free(fd);
-}
-
-static void fifo_delete(struct fifo_data *fd)
-{
-	g_debug("Removing FIFO \"%s\"", fd->path);
+	g_debug("Removing FIFO \"%s\"", path);
 
-	if (unlink(fd->path) < 0) {
+	if (unlink(path) < 0) {
 		g_warning("Could not remove FIFO \"%s\": %s",
-			  fd->path, g_strerror(errno));
+			  path, g_strerror(errno));
 		return;
 	}
 
-	fd->created = false;
+	created = false;
 }
 
-static void
-fifo_close(struct fifo_data *fd)
+void
+FifoOutput::Close()
 {
 	struct stat st;
 
-	if (fd->input >= 0) {
-		close(fd->input);
-		fd->input = -1;
+	if (input >= 0) {
+		close(input);
+		input = -1;
 	}
 
-	if (fd->output >= 0) {
-		close(fd->output);
-		fd->output = -1;
+	if (output >= 0) {
+		close(output);
+		output = -1;
 	}
 
-	if (fd->created && (stat(fd->path, &st) == 0))
-		fifo_delete(fd);
+	if (created && (stat(path, &st) == 0))
+		Delete();
 }
 
-static bool
-fifo_make(struct fifo_data *fd, GError **error)
+inline bool
+FifoOutput::Create(GError **error_r)
 {
-	if (mkfifo(fd->path, 0666) < 0) {
-		g_set_error(error, fifo_output_quark(), errno,
+	if (mkfifo(path, 0666) < 0) {
+		g_set_error(error_r, fifo_output_quark(), errno,
 			    "Couldn't create FIFO \"%s\": %s",
-			    fd->path, g_strerror(errno));
+			    path, g_strerror(errno));
 		return false;
 	}
 
-	fd->created = true;
-
+	created = true;
 	return true;
 }
 
-static bool
-fifo_check(struct fifo_data *fd, GError **error)
+inline bool
+FifoOutput::Check(GError **error_r)
 {
 	struct stat st;
-
-	if (stat(fd->path, &st) < 0) {
+	if (stat(path, &st) < 0) {
 		if (errno == ENOENT) {
 			/* Path doesn't exist */
-			return fifo_make(fd, error);
+			return Create(error_r);
 		}
 
-		g_set_error(error, fifo_output_quark(), errno,
+		g_set_error(error_r, fifo_output_quark(), errno,
 			    "Failed to stat FIFO \"%s\": %s",
-			    fd->path, g_strerror(errno));
+			    path, g_strerror(errno));
 		return false;
 	}
 
 	if (!S_ISFIFO(st.st_mode)) {
-		g_set_error(error, fifo_output_quark(), 0,
+		g_set_error(error_r, fifo_output_quark(), 0,
 			    "\"%s\" already exists, but is not a FIFO",
-			    fd->path);
+			    path);
 		return false;
 	}
 
 	return true;
 }
 
-static bool
-fifo_open(struct fifo_data *fd, GError **error)
+inline bool
+FifoOutput::Open(GError **error_r)
 {
-	if (!fifo_check(fd, error))
+	if (!Check(error_r))
 		return false;
 
-	fd->input = open_cloexec(fd->path, O_RDONLY|O_NONBLOCK|O_BINARY, 0);
-	if (fd->input < 0) {
-		g_set_error(error, fifo_output_quark(), errno,
+	input = open_cloexec(path, O_RDONLY|O_NONBLOCK|O_BINARY, 0);
+	if (input < 0) {
+		g_set_error(error_r, fifo_output_quark(), errno,
 			    "Could not open FIFO \"%s\" for reading: %s",
-			    fd->path, g_strerror(errno));
-		fifo_close(fd);
+			    path, g_strerror(errno));
+		Close();
 		return false;
 	}
 
-	fd->output = open_cloexec(fd->path, O_WRONLY|O_NONBLOCK|O_BINARY, 0);
-	if (fd->output < 0) {
-		g_set_error(error, fifo_output_quark(), errno,
+	output = open_cloexec(path, O_WRONLY|O_NONBLOCK|O_BINARY, 0);
+	if (output < 0) {
+		g_set_error(error_r, fifo_output_quark(), errno,
 			    "Could not open FIFO \"%s\" for writing: %s",
-			    fd->path, g_strerror(errno));
-		fifo_close(fd);
+			    path, g_strerror(errno));
+		Close();
 		return false;
 	}
 
 	return true;
 }
 
-static struct audio_output *
-fifo_output_init(const struct config_param *param,
-		 GError **error_r)
+static bool
+fifo_open(FifoOutput *fd, GError **error_r)
 {
-	struct fifo_data *fd;
+	return fd->Open(error_r);
+}
 
-	GError *error = NULL;
-	char *path = config_dup_block_path(param, "path", &error);
+static struct audio_output *
+fifo_output_init(const config_param &param, GError **error_r)
+{
+	GError *error = nullptr;
+	char *path = param.DupBlockPath("path", &error);
 	if (!path) {
-		if (error != NULL)
+		if (error != nullptr)
 			g_propagate_error(error_r, error);
 		else
 			g_set_error(error_r, fifo_output_quark(), 0,
 				    "No \"path\" parameter specified");
-		return NULL;
+		return nullptr;
 	}
 
-	fd = fifo_data_new();
+	FifoOutput *fd = new FifoOutput();
 	fd->path = path;
 
-	if (!ao_base_init(&fd->base, &fifo_output_plugin, param, error_r)) {
-		fifo_data_free(fd);
-		return NULL;
+	if (!fd->Initialize(param, error_r)) {
+		delete fd;
+		return nullptr;
 	}
 
 	if (!fifo_open(fd, error_r)) {
-		ao_base_finish(&fd->base);
-		fifo_data_free(fd);
-		return NULL;
+		fd->Deinitialize();
+		delete fd;
+		return nullptr;
 	}
 
 	return &fd->base;
@@ -215,20 +219,20 @@ fifo_output_init(const struct config_param *param,
 static void
 fifo_output_finish(struct audio_output *ao)
 {
-	struct fifo_data *fd = (struct fifo_data *)ao;
+	FifoOutput *fd = (FifoOutput *)ao;
 
-	fifo_close(fd);
-	ao_base_finish(&fd->base);
-	fifo_data_free(fd);
+	fd->Close();
+	fd->Deinitialize();
+	delete fd;
 }
 
 static bool
-fifo_output_open(struct audio_output *ao, struct audio_format *audio_format,
+fifo_output_open(struct audio_output *ao, AudioFormat &audio_format,
 		 G_GNUC_UNUSED GError **error)
 {
-	struct fifo_data *fd = (struct fifo_data *)ao;
+	FifoOutput *fd = (FifoOutput *)ao;
 
-	fd->timer = timer_new(audio_format);
+	fd->timer = new Timer(audio_format);
 
 	return true;
 }
@@ -236,19 +240,19 @@ fifo_output_open(struct audio_output *ao, struct audio_format *audio_format,
 static void
 fifo_output_close(struct audio_output *ao)
 {
-	struct fifo_data *fd = (struct fifo_data *)ao;
+	FifoOutput *fd = (FifoOutput *)ao;
 
-	timer_free(fd->timer);
+	delete fd->timer;
 }
 
 static void
 fifo_output_cancel(struct audio_output *ao)
 {
-	struct fifo_data *fd = (struct fifo_data *)ao;
+	FifoOutput *fd = (FifoOutput *)ao;
 	char buf[FIFO_BUFFER_SIZE];
 	int bytes = 1;
 
-	timer_reset(fd->timer);
+	fd->timer->Reset();
 
 	while (bytes > 0 && errno != EINTR)
 		bytes = read(fd->input, buf, FIFO_BUFFER_SIZE);
@@ -262,10 +266,10 @@ fifo_output_cancel(struct audio_output *ao)
 static unsigned
 fifo_output_delay(struct audio_output *ao)
 {
-	struct fifo_data *fd = (struct fifo_data *)ao;
+	FifoOutput *fd = (FifoOutput *)ao;
 
-	return fd->timer->started
-		? timer_delay(fd->timer)
+	return fd->timer->IsStarted()
+		? fd->timer->GetDelay()
 		: 0;
 }
 
@@ -273,12 +277,12 @@ static size_t
 fifo_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		 GError **error)
 {
-	struct fifo_data *fd = (struct fifo_data *)ao;
+	FifoOutput *fd = (FifoOutput *)ao;
 	ssize_t bytes;
 
-	if (!fd->timer->started)
-		timer_start(fd->timer);
-	timer_add(fd->timer, size);
+	if (!fd->timer->IsStarted())
+		fd->timer->Start();
+	fd->timer->Add(size);
 
 	while (true) {
 		bytes = write(fd->output, chunk, size);
@@ -304,12 +308,19 @@ fifo_output_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 const struct audio_output_plugin fifo_output_plugin = {
-	.name = "fifo",
-	.init = fifo_output_init,
-	.finish = fifo_output_finish,
-	.open = fifo_output_open,
-	.close = fifo_output_close,
-	.delay = fifo_output_delay,
-	.play = fifo_output_play,
-	.cancel = fifo_output_cancel,
+	"fifo",
+	nullptr,
+	fifo_output_init,
+	fifo_output_finish,
+	nullptr,
+	nullptr,
+	fifo_output_open,
+	fifo_output_close,
+	fifo_output_delay,
+	nullptr,
+	fifo_output_play,
+	nullptr,
+	fifo_output_cancel,
+	nullptr,
+	nullptr,
 };
diff --git a/src/output/fifo_output_plugin.h b/src/output/FifoOutputPlugin.hxx
index 85f7985e1..dca2886d8 100644
--- a/src/output/fifo_output_plugin.h
+++ b/src/output/FifoOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_FIFO_OUTPUT_PLUGIN_H
-#define MPD_FIFO_OUTPUT_PLUGIN_H
+#ifndef MPD_FIFO_OUTPUT_PLUGIN_HXX
+#define MPD_FIFO_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin fifo_output_plugin;
 
diff --git a/src/output/HttpdClient.cxx b/src/output/HttpdClient.cxx
new file mode 100644
index 000000000..0a00ee2f9
--- /dev/null
+++ b/src/output/HttpdClient.cxx
@@ -0,0 +1,444 @@
+/*
+ * Copyright (C) 2003-2011 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 "HttpdClient.hxx"
+#include "HttpdInternal.hxx"
+#include "util/fifo_buffer.h"
+#include "Page.hxx"
+#include "IcyMetaDataServer.hxx"
+#include "SocketError.hxx"
+
+#include <assert.h>
+#include <string.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "httpd_output"
+
+HttpdClient::~HttpdClient()
+{
+	if (state == RESPONSE) {
+		if (current_page != nullptr)
+			current_page->Unref();
+
+		for (auto page : pages)
+			page->Unref();
+	}
+
+	if (metadata)
+		metadata->Unref();
+}
+
+void
+HttpdClient::Close()
+{
+	httpd->RemoveClient(*this);
+}
+
+void
+HttpdClient::LockClose()
+{
+	const ScopeLock protect(httpd->mutex);
+	Close();
+}
+
+void
+HttpdClient::BeginResponse()
+{
+	assert(state != RESPONSE);
+
+	state = RESPONSE;
+	current_page = nullptr;
+
+	httpd->SendHeader(*this);
+}
+
+/**
+ * Handle a line of the HTTP request.
+ */
+bool
+HttpdClient::HandleLine(const char *line)
+{
+	assert(state != RESPONSE);
+
+	if (state == REQUEST) {
+		if (strncmp(line, "GET /", 5) != 0) {
+			/* only GET is supported */
+			g_warning("malformed request line from client");
+			return false;
+		}
+
+		line = strchr(line + 5, ' ');
+		if (line == nullptr || strncmp(line + 1, "HTTP/", 5) != 0) {
+			/* HTTP/0.9 without request headers */
+			BeginResponse();
+			return true;
+		}
+
+		/* after the request line, request headers follow */
+		state = HEADERS;
+		return true;
+	} else {
+		if (*line == 0) {
+			/* empty line: request is finished */
+			BeginResponse();
+			return true;
+		}
+
+		if (g_ascii_strncasecmp(line, "Icy-MetaData: 1", 15) == 0) {
+			/* Send icy metadata */
+			metadata_requested = metadata_supported;
+			return true;
+		}
+
+		if (g_ascii_strncasecmp(line, "transferMode.dlna.org: Streaming", 32) == 0) {
+			/* Send as dlna */
+			dlna_streaming_requested = true;
+			/* metadata is not supported by dlna streaming, so disable it */
+			metadata_supported = false;
+			metadata_requested = false;
+			return true;
+		}
+
+		/* expect more request headers */
+		return true;
+	}
+}
+
+/**
+ * Sends the status line and response headers to the client.
+ */
+bool
+HttpdClient::SendResponse()
+{
+	char buffer[1024];
+	assert(state == RESPONSE);
+
+	if (dlna_streaming_requested) {
+		g_snprintf(buffer, sizeof(buffer),
+			   "HTTP/1.1 206 OK\r\n"
+			   "Content-Type: %s\r\n"
+			   "Content-Length: 10000\r\n"
+			   "Content-RangeX: 0-1000000/1000000\r\n"
+			   "transferMode.dlna.org: Streaming\r\n"
+			   "Accept-Ranges: bytes\r\n"
+			   "Connection: close\r\n"
+			   "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);
+
+	} else if (metadata_requested) {
+		gchar *metadata_header;
+
+		metadata_header =
+			icy_server_metadata_header(httpd->name, httpd->genre,
+						   httpd->website,
+						   httpd->content_type,
+						   metaint);
+
+		g_strlcpy(buffer, metadata_header, sizeof(buffer));
+
+		g_free(metadata_header);
+
+       } else { /* revert to a normal HTTP request */
+		g_snprintf(buffer, sizeof(buffer),
+			   "HTTP/1.1 200 OK\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",
+			   httpd->content_type);
+	}
+
+	ssize_t nbytes = SocketMonitor::Write(buffer, strlen(buffer));
+	if (gcc_unlikely(nbytes < 0)) {
+		const SocketErrorMessage msg;
+		g_warning("failed to write to client: %s", (const char *)msg);
+		Close();
+		return false;
+	}
+
+	return true;
+}
+
+HttpdClient::HttpdClient(HttpdOutput *_httpd, int _fd, EventLoop &_loop,
+			 bool _metadata_supported)
+	:BufferedSocket(_fd, _loop),
+	 httpd(_httpd),
+	 state(REQUEST),
+	 dlna_streaming_requested(false),
+	 metadata_supported(_metadata_supported),
+	 metadata_requested(false), metadata_sent(true),
+	 metaint(8192), /*TODO: just a std value */
+	 metadata(nullptr),
+	 metadata_current_position(0), metadata_fill(0)
+{
+}
+
+size_t
+HttpdClient::GetQueueSize() const
+{
+	if (state != RESPONSE)
+		return 0;
+
+	size_t size = 0;
+	for (auto page : pages)
+		size += page->size;
+	return size;
+}
+
+void
+HttpdClient::CancelQueue()
+{
+	if (state != RESPONSE)
+		return;
+
+	for (auto page : pages)
+		page->Unref();
+	pages.clear();
+
+	if (current_page == nullptr)
+		CancelWrite();
+}
+
+ssize_t
+HttpdClient::TryWritePage(const Page &page, size_t position)
+{
+	assert(position < page.size);
+
+	return Write(page.data + position, page.size - position);
+}
+
+ssize_t
+HttpdClient::TryWritePageN(const Page &page, size_t position, ssize_t n)
+{
+	return n >= 0
+		? Write(page.data + position, n)
+		: TryWritePage(page, position);
+}
+
+ssize_t
+HttpdClient::GetBytesTillMetaData() const
+{
+	if (metadata_requested &&
+	    current_page->size - current_position > metaint - metadata_fill)
+		return metaint - metadata_fill;
+
+	return -1;
+}
+
+inline bool
+HttpdClient::TryWrite()
+{
+	const ScopeLock protect(httpd->mutex);
+
+	assert(state == RESPONSE);
+
+	if (current_page == nullptr) {
+		if (pages.empty()) {
+			/* another thread has removed the event source
+			   while this thread was waiting for
+			   httpd->mutex */
+			CancelWrite();
+			return true;
+		}
+
+		current_page = pages.front();
+		pages.pop_front();
+		current_position = 0;
+	}
+
+	const ssize_t bytes_to_write = GetBytesTillMetaData();
+	if (bytes_to_write == 0) {
+		if (!metadata_sent) {
+			ssize_t nbytes = TryWritePage(*metadata,
+						      metadata_current_position);
+			if (nbytes < 0) {
+				auto e = GetSocketError();
+				if (IsSocketErrorAgain(e))
+					return true;
+
+				if (!IsSocketErrorClosed(e)) {
+					SocketErrorMessage msg(e);
+					g_warning("failed to write to client: %s",
+						  (const char *)msg);
+				}
+
+				Close();
+				return false;
+			}
+
+			metadata_current_position += nbytes;
+
+			if (metadata->size - metadata_current_position == 0) {
+				metadata_fill = 0;
+				metadata_current_position = 0;
+				metadata_sent = true;
+			}
+		} else {
+			guchar empty_data = 0;
+
+			ssize_t nbytes = Write(&empty_data, 1);
+			if (nbytes < 0) {
+				auto e = GetSocketError();
+				if (IsSocketErrorAgain(e))
+					return true;
+
+				if (!IsSocketErrorClosed(e)) {
+					SocketErrorMessage msg(e);
+					g_warning("failed to write to client: %s",
+						  (const char *)msg);
+				}
+
+				Close();
+				return false;
+			}
+
+			metadata_fill = 0;
+			metadata_current_position = 0;
+		}
+	} else {
+		ssize_t nbytes =
+			TryWritePageN(*current_page, current_position,
+				      bytes_to_write);
+		if (nbytes < 0) {
+			auto e = GetSocketError();
+			if (IsSocketErrorAgain(e))
+				return true;
+
+			if (!IsSocketErrorClosed(e)) {
+				SocketErrorMessage msg(e);
+				g_warning("failed to write to client: %s",
+					  (const char *)msg);
+			}
+
+			Close();
+			return false;
+		}
+
+		current_position += nbytes;
+		assert(current_position <= current_page->size);
+
+		if (metadata_requested)
+			metadata_fill += nbytes;
+
+		if (current_position >= current_page->size) {
+			current_page->Unref();
+			current_page = nullptr;
+
+			if (pages.empty())
+				/* all pages are sent: remove the
+				   event source */
+				CancelWrite();
+		}
+	}
+
+	return true;
+}
+
+void
+HttpdClient::PushPage(Page *page)
+{
+	if (state != RESPONSE)
+		/* the client is still writing the HTTP request */
+		return;
+
+	page->Ref();
+	pages.push_back(page);
+
+	ScheduleWrite();
+}
+
+void
+HttpdClient::PushMetaData(Page *page)
+{
+	if (metadata) {
+		metadata->Unref();
+		metadata = nullptr;
+	}
+
+	g_return_if_fail (page);
+
+	page->Ref();
+	metadata = page;
+	metadata_sent = false;
+}
+
+bool
+HttpdClient::OnSocketReady(unsigned flags)
+{
+	if (!BufferedSocket::OnSocketReady(flags))
+		return false;
+
+	if (flags & WRITE)
+		if (!TryWrite())
+			return false;
+
+	return true;
+}
+
+BufferedSocket::InputResult
+HttpdClient::OnSocketInput(const void *data, size_t length)
+{
+	if (state == RESPONSE) {
+		g_warning("unexpected input from client");
+		LockClose();
+		return InputResult::CLOSED;
+	}
+
+	const char *line = (const char *)data;
+	const char *newline = (const char *)memchr(line, '\n', length);
+	if (newline == nullptr)
+		return InputResult::MORE;
+
+	ConsumeInput(newline + 1 - line);
+
+	if (newline > line && newline[-1] == '\r')
+		--newline;
+
+	/* terminate the string at the end of the line; the const_cast
+	   is a dirty hack */
+	*const_cast<char *>(newline) = 0;
+
+	if (!HandleLine(line)) {
+		assert(state == RESPONSE);
+		LockClose();
+		return InputResult::CLOSED;
+	}
+
+	if (state == RESPONSE && !SendResponse())
+		return InputResult::CLOSED;
+
+	return InputResult::AGAIN;
+}
+
+void
+HttpdClient::OnSocketError(GError *error)
+{
+	g_warning("error on HTTP client: %s", error->message);
+	g_error_free(error);
+}
+
+void
+HttpdClient::OnSocketClosed()
+{
+	LockClose();
+}
diff --git a/src/output/HttpdClient.hxx b/src/output/HttpdClient.hxx
new file mode 100644
index 000000000..46196d2e3
--- /dev/null
+++ b/src/output/HttpdClient.hxx
@@ -0,0 +1,186 @@
+/*
+ * Copyright (C) 2003-2013 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_HTTPD_CLIENT_HXX
+#define MPD_OUTPUT_HTTPD_CLIENT_HXX
+
+#include "event/BufferedSocket.hxx"
+#include "gcc.h"
+
+#include <list>
+
+#include <stddef.h>
+
+struct HttpdOutput;
+class Page;
+
+class HttpdClient final : public BufferedSocket {
+	/**
+	 * The httpd output object this client is connected to.
+	 */
+	HttpdOutput *const httpd;
+
+	/**
+	 * The current state of the client.
+	 */
+	enum {
+		/** reading the request line */
+		REQUEST,
+
+		/** reading the request headers */
+		HEADERS,
+
+		/** sending the HTTP response */
+		RESPONSE,
+	} state;
+
+	/**
+	 * A queue of #Page objects to be sent to the client.
+	 */
+	std::list<Page *> pages;
+
+	/**
+	 * The #page which is currently being sent to the client.
+	 */
+	Page *current_page;
+
+	/**
+	 * The amount of bytes which were already sent from
+	 * #current_page.
+	 */
+	size_t current_position;
+
+	/**
+         * If DLNA streaming was an option.
+         */
+	bool dlna_streaming_requested;
+
+	/* ICY */
+
+	/**
+	 * Do we support sending Icy-Metadata to the client?  This is
+	 * disabled if the httpd audio output uses encoder tags.
+	 */
+	bool metadata_supported;
+
+	/**
+	 * If we should sent icy metadata.
+	 */
+	bool metadata_requested;
+
+	/**
+	 * If the current metadata was already sent to the client.
+	 */
+	bool metadata_sent;
+
+	/**
+	 * The amount of streaming data between each metadata block
+	 */
+	guint metaint;
+
+	/**
+	 * The metadata as #Page which is currently being sent to the client.
+	 */
+	Page *metadata;
+
+	/*
+	 * The amount of bytes which were already sent from the metadata.
+	 */
+	size_t metadata_current_position;
+
+	/**
+	 * The amount of streaming data sent to the client
+	 * since the last icy information was sent.
+	 */
+	guint metadata_fill;
+
+public:
+	/**
+	 * @param httpd the HTTP output device
+	 * @param fd the socket file descriptor
+	 */
+	HttpdClient(HttpdOutput *httpd, int _fd, EventLoop &_loop,
+		    bool _metadata_supported);
+
+	/**
+	 * Note: this does not remove the client from the
+	 * #HttpdOutput object.
+	 */
+	~HttpdClient();
+
+	/**
+	 * Frees the client and removes it from the server's client list.
+	 */
+	void Close();
+
+	void LockClose();
+
+	/**
+	 * Returns the total size of this client's page queue.
+	 */
+	gcc_pure
+	size_t GetQueueSize() const;
+
+	/**
+	 * Clears the page queue.
+	 */
+	void CancelQueue();
+
+	/**
+	 * Handle a line of the HTTP request.
+	 */
+	bool HandleLine(const char *line);
+
+	/**
+	 * Switch the client to the "RESPONSE" state.
+	 */
+	void BeginResponse();
+
+	/**
+	 * Sends the status line and response headers to the client.
+	 */
+	bool SendResponse();
+
+	gcc_pure
+	ssize_t GetBytesTillMetaData() const;
+
+	ssize_t TryWritePage(const Page &page, size_t position);
+	ssize_t TryWritePageN(const Page &page, size_t position, ssize_t n);
+
+	bool TryWrite();
+
+	/**
+	 * Appends a page to the client's queue.
+	 */
+	void PushPage(Page *page);
+
+	/**
+	 * Sends the passed metadata.
+	 */
+	void PushMetaData(Page *page);
+
+protected:
+	virtual bool OnSocketReady(unsigned flags) override;
+	virtual InputResult OnSocketInput(const void *data,
+					  size_t length) override;
+	virtual void OnSocketError(GError *error) override;
+	virtual void OnSocketClosed() override;
+};
+
+#endif
diff --git a/src/output/httpd_internal.h b/src/output/HttpdInternal.hxx
index 5dcb8ab9b..e8a37e033 100644
--- a/src/output/httpd_internal.h
+++ b/src/output/HttpdInternal.hxx
@@ -25,16 +25,22 @@
 #ifndef MPD_OUTPUT_HTTPD_INTERNAL_H
 #define MPD_OUTPUT_HTTPD_INTERNAL_H
 
-#include "output_internal.h"
-#include "timer.h"
-
-#include <glib.h>
-
-#include <stdbool.h>
-
-struct httpd_client;
-
-struct httpd_output {
+#include "OutputInternal.hxx"
+#include "Timer.hxx"
+#include "thread/Mutex.hxx"
+#include "event/ServerSocket.hxx"
+
+#include <forward_list>
+
+struct config_param;
+class EventLoop;
+class ServerSocket;
+class HttpdClient;
+class Page;
+struct Encoder;
+struct Tag;
+
+struct HttpdOutput final : private ServerSocket {
 	struct audio_output base;
 
 	/**
@@ -46,7 +52,7 @@ struct httpd_output {
 	/**
 	 * The configured encoder plugin.
 	 */
-	struct encoder *encoder;
+	Encoder *encoder;
 
 	/**
 	 * Number of bytes which were fed into the encoder, without
@@ -65,28 +71,23 @@ struct httpd_output {
 	 * This mutex protects the listener socket and the client
 	 * list.
 	 */
-	GMutex *mutex;
+	mutable Mutex mutex;
 
 	/**
-	 * A #timer object to synchronize this output with the
+	 * A #Timer object to synchronize this output with the
 	 * wallclock.
 	 */
-	struct timer *timer;
-
-	/**
-	 * The listener socket.
-	 */
-	struct server_socket *server_socket;
+	Timer *timer;
 
 	/**
 	 * The header page, which is sent to every client on connect.
 	 */
-	struct page *header;
+	Page *header;
 
 	/**
 	 * The metadata, which is sent to every client.
 	 */
-	struct page *metadata;
+	Page *metadata;
 
 	/**
 	 * The configured name.
@@ -105,7 +106,7 @@ struct httpd_output {
 	 * A linked list containing all clients which are currently
 	 * connected.
 	 */
-	GList *clients;
+	std::forward_list<HttpdClient> clients;
 
 	/**
 	 * A temporary buffer for the httpd_output_read_page()
@@ -118,21 +119,88 @@ struct httpd_output {
 	 * at the same time.
 	 */
 	guint clients_max, clients_cnt;
-};
 
-/**
- * Removes a client from the httpd_output.clients linked list.
- */
-void
-httpd_output_remove_client(struct httpd_output *httpd,
-			   struct httpd_client *client);
+	HttpdOutput(EventLoop &_loop);
+	~HttpdOutput();
 
-/**
- * Sends the encoder header to the client.  This is called right after
- * the response headers have been sent.
- */
-void
-httpd_output_send_header(struct httpd_output *httpd,
-			 struct httpd_client *client);
+	bool Configure(const config_param &param, GError **error_r);
+
+	bool Bind(GError **error_r);
+	void Unbind();
+
+	/**
+	 * Caller must lock the mutex.
+	 */
+	bool OpenEncoder(AudioFormat &audio_format,
+			 GError **error_r);
+
+	/**
+	 * Caller must lock the mutex.
+	 */
+	bool Open(AudioFormat &audio_format, GError **error_r);
+
+	/**
+	 * Caller must lock the mutex.
+	 */
+	void Close();
+
+	/**
+	 * Check whether there is at least one client.
+	 *
+	 * Caller must lock the mutex.
+	 */
+	gcc_pure
+	bool HasClients() const {
+		return !clients.empty();
+	}
+
+	/**
+	 * Check whether there is at least one client.
+	 */
+	gcc_pure
+	bool LockHasClients() const {
+		const ScopeLock protect(mutex);
+		return HasClients();
+	}
+
+	void AddClient(int fd);
+
+	/**
+	 * Removes a client from the httpd_output.clients linked list.
+	 */
+	void RemoveClient(HttpdClient &client);
+
+	/**
+	 * Sends the encoder header to the client.  This is called
+	 * right after the response headers have been sent.
+	 */
+	void SendHeader(HttpdClient &client) const;
+
+	/**
+	 * Reads data from the encoder (as much as available) and
+	 * returns it as a new #page object.
+	 */
+	Page *ReadPage();
+
+	/**
+	 * Broadcasts a page struct to all clients.
+	 *
+	 * Mutext must not be locked.
+	 */
+	void BroadcastPage(Page *page);
+
+	/**
+	 * Broadcasts data from the encoder to all clients.
+	 */
+	void BroadcastFromEncoder();
+
+	bool EncodeAndPlay(const void *chunk, size_t size, GError **error_r);
+
+	void SendTag(const Tag *tag);
+
+private:
+	virtual void OnAccept(int fd, const sockaddr &address,
+			      size_t address_length, int uid) override;
+};
 
 #endif
diff --git a/src/output/HttpdOutputPlugin.cxx b/src/output/HttpdOutputPlugin.cxx
new file mode 100644
index 000000000..4169aabfa
--- /dev/null
+++ b/src/output/HttpdOutputPlugin.cxx
@@ -0,0 +1,568 @@
+/*
+ * Copyright (C) 2003-2013 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 "HttpdOutputPlugin.hxx"
+#include "HttpdInternal.hxx"
+#include "HttpdClient.hxx"
+#include "OutputAPI.hxx"
+#include "EncoderPlugin.hxx"
+#include "EncoderList.hxx"
+#include "resolver.h"
+#include "Page.hxx"
+#include "IcyMetaDataServer.hxx"
+#include "fd_util.h"
+#include "Main.hxx"
+
+#include <assert.h>
+
+#include <sys/types.h>
+#include <unistd.h>
+#include <string.h>
+#include <errno.h>
+
+#ifdef HAVE_LIBWRAP
+#include <sys/socket.h> /* needed for AF_UNIX */
+#include <tcpd.h>
+#endif
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "httpd_output"
+
+/**
+ * The quark used for GError.domain.
+ */
+static inline GQuark
+httpd_output_quark(void)
+{
+	return g_quark_from_static_string("httpd_output");
+}
+
+inline
+HttpdOutput::HttpdOutput(EventLoop &_loop)
+	:ServerSocket(_loop),
+	 encoder(nullptr), unflushed_input(0),
+	 metadata(nullptr)
+{
+}
+
+HttpdOutput::~HttpdOutput()
+{
+	if (metadata != nullptr)
+		metadata->Unref();
+
+	if (encoder != nullptr)
+		encoder_finish(encoder);
+
+}
+
+inline bool
+HttpdOutput::Bind(GError **error_r)
+{
+	open = false;
+
+	const ScopeLock protect(mutex);
+	return ServerSocket::Open(error_r);
+}
+
+inline void
+HttpdOutput::Unbind()
+{
+	assert(!open);
+
+	const ScopeLock protect(mutex);
+	ServerSocket::Close();
+}
+
+inline bool
+HttpdOutput::Configure(const config_param &param, GError **error_r)
+{
+	/* read configuration */
+	name = param.GetBlockValue("name", "Set name in config");
+	genre = param.GetBlockValue("genre", "Set genre in config");
+	website = param.GetBlockValue("website", "Set website in config");
+
+	guint port = param.GetBlockValue("port", 8000u);
+
+	const char *encoder_name =
+		param.GetBlockValue("encoder", "vorbis");
+	const auto encoder_plugin = encoder_plugin_get(encoder_name);
+	if (encoder_plugin == NULL) {
+		g_set_error(error_r, httpd_output_quark(), 0,
+			    "No such encoder: %s", encoder_name);
+		return false;
+	}
+
+	clients_max = param.GetBlockValue("max_clients", 0u);
+
+	/* set up bind_to_address */
+
+	const char *bind_to_address = param.GetBlockValue("bind_to_address");
+	bool success = bind_to_address != NULL &&
+		strcmp(bind_to_address, "any") != 0
+		? AddHost(bind_to_address, port, error_r)
+		: AddPort(port, error_r);
+	if (!success)
+		return false;
+
+	/* initialize encoder */
+
+	encoder = encoder_init(*encoder_plugin, param, error_r);
+	if (encoder == nullptr)
+		return false;
+
+	/* determine content type */
+	content_type = encoder_get_mime_type(encoder);
+	if (content_type == nullptr)
+		content_type = "application/octet-stream";
+
+	return true;
+}
+
+static struct audio_output *
+httpd_output_init(const struct config_param &param,
+		  GError **error_r)
+{
+	HttpdOutput *httpd = new HttpdOutput(*main_loop);
+
+	if (!ao_base_init(&httpd->base, &httpd_output_plugin, param,
+			  error_r)) {
+		delete httpd;
+		return nullptr;
+	}
+
+	if (!httpd->Configure(param, error_r)) {
+		ao_base_finish(&httpd->base);
+		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));
+}
+
+#if GCC_CHECK_VERSION(4,6) || defined(__clang__)
+#pragma GCC diagnostic pop
+#endif
+
+static void
+httpd_output_finish(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	ao_base_finish(&httpd->base);
+	delete httpd;
+}
+
+/**
+ * Creates a new #HttpdClient object and adds it into the
+ * HttpdOutput.clients linked list.
+ */
+inline void
+HttpdOutput::AddClient(int fd)
+{
+	clients.emplace_front(this, fd, GetEventLoop(),
+			      encoder->plugin.tag == nullptr);
+	++clients_cnt;
+
+	/* pass metadata to client */
+	if (metadata != nullptr)
+		clients.front().PushMetaData(metadata);
+}
+
+void
+HttpdOutput::OnAccept(int fd, const sockaddr &address,
+		      size_t address_length, gcc_unused int uid)
+{
+	/* the listener socket has become readable - a client has
+	   connected */
+
+#ifdef HAVE_LIBWRAP
+	if (address.sa_family != AF_UNIX) {
+		char *hostaddr = sockaddr_to_string(&address, address_length, NULL);
+		const char *progname = g_get_prgname();
+
+		struct request_info req;
+		request_init(&req, RQ_FILE, fd, RQ_DAEMON, progname, 0);
+
+		fromhost(&req);
+
+		if (!hosts_access(&req)) {
+			/* tcp wrappers says no */
+			g_warning("libwrap refused connection (libwrap=%s) from %s",
+			      progname, hostaddr);
+			g_free(hostaddr);
+			close_socket(fd);
+			return;
+		}
+
+		g_free(hostaddr);
+	}
+#else
+	(void)address;
+	(void)address_length;
+#endif	/* HAVE_WRAP */
+
+	const ScopeLock protect(mutex);
+
+	if (fd >= 0) {
+		/* can we allow additional client */
+		if (open && (clients_max == 0 ||  clients_cnt < clients_max))
+			AddClient(fd);
+		else
+			close_socket(fd);
+	} else if (fd < 0 && errno != EINTR) {
+		g_warning("accept() failed: %s", g_strerror(errno));
+	}
+}
+
+Page *
+HttpdOutput::ReadPage()
+{
+	if (unflushed_input >= 65536) {
+		/* we have fed a lot of input into the encoder, but it
+		   didn't give anything back yet - flush now to avoid
+		   buffer underruns */
+		encoder_flush(encoder, NULL);
+		unflushed_input = 0;
+	}
+
+	size_t size = 0;
+	do {
+		size_t nbytes = encoder_read(encoder,
+					     buffer + size,
+					     sizeof(buffer) - size);
+		if (nbytes == 0)
+			break;
+
+		unflushed_input = 0;
+
+		size += nbytes;
+	} while (size < sizeof(buffer));
+
+	if (size == 0)
+		return NULL;
+
+	return Page::Copy(buffer, size);
+}
+
+static bool
+httpd_output_enable(struct audio_output *ao, GError **error_r)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	return httpd->Bind(error_r);
+}
+
+static void
+httpd_output_disable(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	httpd->Unbind();
+}
+
+inline bool
+HttpdOutput::OpenEncoder(AudioFormat &audio_format, GError **error)
+{
+	if (!encoder_open(encoder, audio_format, error))
+		return false;
+
+	/* we have to remember the encoder header, i.e. the first
+	   bytes of encoder output after opening it, because it has to
+	   be sent to every new client */
+	header = ReadPage();
+
+	unflushed_input = 0;
+
+	return true;
+}
+
+inline bool
+HttpdOutput::Open(AudioFormat &audio_format, GError **error_r)
+{
+	assert(!open);
+	assert(clients.empty());
+
+	/* open the encoder */
+
+	if (!OpenEncoder(audio_format, error_r))
+		return false;
+
+	/* initialize other attributes */
+
+	clients_cnt = 0;
+	timer = new Timer(audio_format);
+
+	open = true;
+
+	return true;
+}
+
+static bool
+httpd_output_open(struct audio_output *ao, AudioFormat &audio_format,
+		  GError **error)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	assert(httpd->clients.empty());
+
+	const ScopeLock protect(httpd->mutex);
+	return httpd->Open(audio_format, error);
+}
+
+inline void
+HttpdOutput::Close()
+{
+	assert(open);
+
+	open = false;
+
+	delete timer;
+
+	clients.clear();
+
+	if (header != NULL)
+		header->Unref();
+
+	encoder_close(encoder);
+}
+
+static void
+httpd_output_close(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	const ScopeLock protect(httpd->mutex);
+	httpd->Close();
+}
+
+void
+HttpdOutput::RemoveClient(HttpdClient &client)
+{
+	assert(clients_cnt > 0);
+
+	for (auto prev = clients.before_begin(), i = std::next(prev);;
+	     prev = i, i = std::next(prev)) {
+		assert(i != clients.end());
+		if (&*i == &client) {
+			clients.erase_after(prev);
+			clients_cnt--;
+			break;
+		}
+	}
+}
+
+void
+HttpdOutput::SendHeader(HttpdClient &client) const
+{
+	if (header != NULL)
+		client.PushPage(header);
+}
+
+static unsigned
+httpd_output_delay(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	if (!httpd->LockHasClients() && httpd->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();
+
+		/* some arbitrary delay that is long enough to avoid
+		   consuming too much CPU, and short enough to notice
+		   new clients quickly enough */
+		return 1000;
+	}
+
+	return httpd->timer->IsStarted()
+		? httpd->timer->GetDelay()
+		: 0;
+}
+
+void
+HttpdOutput::BroadcastPage(Page *page)
+{
+	assert(page != NULL);
+
+	const ScopeLock protect(mutex);
+	for (auto &client : clients)
+		client.PushPage(page);
+}
+
+void
+HttpdOutput::BroadcastFromEncoder()
+{
+	mutex.lock();
+	for (auto &client : clients) {
+		if (client.GetQueueSize() > 256 * 1024) {
+			g_debug("client is too slow, flushing its queue");
+			client.CancelQueue();
+		}
+	}
+	mutex.unlock();
+
+	Page *page;
+	while ((page = ReadPage()) != nullptr) {
+		BroadcastPage(page);
+		page->Unref();
+	}
+}
+
+inline bool
+HttpdOutput::EncodeAndPlay(const void *chunk, size_t size, GError **error_r)
+{
+	if (!encoder_write(encoder, chunk, size, error_r))
+		return false;
+
+	unflushed_input += size;
+
+	BroadcastFromEncoder();
+	return true;
+}
+
+static size_t
+httpd_output_play(struct audio_output *ao, const void *chunk, size_t size,
+		  GError **error_r)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	if (httpd->LockHasClients()) {
+		if (!httpd->EncodeAndPlay(chunk, size, error_r))
+			return 0;
+	}
+
+	if (!httpd->timer->IsStarted())
+		httpd->timer->Start();
+	httpd->timer->Add(size);
+
+	return size;
+}
+
+static bool
+httpd_output_pause(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	if (httpd->LockHasClients()) {
+		static const char silence[1020] = { 0 };
+		return httpd_output_play(ao, silence, sizeof(silence),
+					 NULL) > 0;
+	} else {
+		return true;
+	}
+}
+
+inline void
+HttpdOutput::SendTag(const Tag *tag)
+{
+	assert(tag != NULL);
+
+	if (encoder->plugin.tag != nullptr) {
+		/* embed encoder tags */
+
+		/* flush the current stream, and end it */
+
+		encoder_pre_tag(encoder, NULL);
+		BroadcastFromEncoder();
+
+		/* send the tag to the encoder - which starts a new
+		   stream now */
+
+		encoder_tag(encoder, tag, NULL);
+
+		/* the first page generated by the encoder will now be
+		   used as the new "header" page, which is sent to all
+		   new clients */
+
+		Page *page = ReadPage();
+		if (page != NULL) {
+			if (header != NULL)
+				header->Unref();
+			header = page;
+			BroadcastPage(page);
+		}
+	} else {
+		/* use Icy-Metadata */
+
+		if (metadata != NULL)
+			metadata->Unref();
+
+		static constexpr tag_type types[] = {
+			TAG_ALBUM, TAG_ARTIST, TAG_TITLE,
+			TAG_NUM_OF_ITEM_TYPES
+		};
+
+		metadata = icy_server_metadata_page(*tag, &types[0]);
+		if (metadata != NULL) {
+			const ScopeLock protect(mutex);
+			for (auto &client : clients)
+				client.PushMetaData(metadata);
+		}
+	}
+}
+
+static void
+httpd_output_tag(struct audio_output *ao, const Tag *tag)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	httpd->SendTag(tag);
+}
+
+static void
+httpd_output_cancel(struct audio_output *ao)
+{
+	HttpdOutput *httpd = Cast(ao);
+
+	const ScopeLock protect(httpd->mutex);
+	for (auto &client : httpd->clients)
+		client.CancelQueue();
+}
+
+const struct audio_output_plugin httpd_output_plugin = {
+	"httpd",
+	nullptr,
+	httpd_output_init,
+	httpd_output_finish,
+	httpd_output_enable,
+	httpd_output_disable,
+	httpd_output_open,
+	httpd_output_close,
+	httpd_output_delay,
+	httpd_output_tag,
+	httpd_output_play,
+	nullptr,
+	httpd_output_cancel,
+	httpd_output_pause,
+	nullptr,
+};
diff --git a/src/output/httpd_output_plugin.h b/src/output/HttpdOutputPlugin.hxx
index d0eb1533f..c74d2bd4a 100644
--- a/src/output/httpd_output_plugin.h
+++ b/src/output/HttpdOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_HTTPD_OUTPUT_PLUGIN_H
-#define MPD_HTTPD_OUTPUT_PLUGIN_H
+#ifndef MPD_HTTPD_OUTPUT_PLUGIN_HXX
+#define MPD_HTTPD_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin httpd_output_plugin;
 
diff --git a/src/output/jack_output_plugin.c b/src/output/JackOutputPlugin.cxx
index d5c8ca412..241857d82 100644
--- a/src/output/jack_output_plugin.c
+++ b/src/output/JackOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,8 +18,8 @@
  */
 
 #include "config.h"
-#include "jack_output_plugin.h"
-#include "output_api.h"
+#include "JackOutputPlugin.hxx"
+#include "OutputAPI.hxx"
 
 #include <assert.h>
 
@@ -29,6 +29,7 @@
 #include <jack/ringbuffer.h>
 
 #include <stdlib.h>
+#include <string.h>
 #include <stdio.h>
 #include <sys/types.h>
 #include <unistd.h>
@@ -43,7 +44,7 @@ enum {
 
 static const size_t jack_sample_size = sizeof(jack_default_audio_sample_t);
 
-struct jack_data {
+struct JackOutput {
 	struct audio_output base;
 
 	/**
@@ -66,7 +67,7 @@ struct jack_data {
 	size_t ringbuffer_size;
 
 	/* the current audio format */
-	struct audio_format audio_format;
+	AudioFormat audio_format;
 
 	/* jack library stuff */
 	jack_port_t *ports[MAX_PORTS];
@@ -80,6 +81,15 @@ struct jack_data {
 	 * silence.
 	 */
 	bool pause;
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &jack_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
 };
 
 /**
@@ -96,7 +106,7 @@ jack_output_quark(void)
  * channels.
  */
 static jack_nframes_t
-mpd_jack_available(const struct jack_data *jd)
+mpd_jack_available(const JackOutput *jd)
 {
 	size_t min = jack_ringbuffer_read_space(jd->ringbuffer[0]);
 
@@ -114,8 +124,7 @@ mpd_jack_available(const struct jack_data *jd)
 static int
 mpd_jack_process(jack_nframes_t nframes, void *arg)
 {
-	struct jack_data *jd = (struct jack_data *) arg;
-	jack_default_audio_sample_t *out;
+	JackOutput *jd = (JackOutput *) arg;
 
 	if (nframes <= 0)
 		return 0;
@@ -131,7 +140,9 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 		/* generate silence while MPD is paused */
 
 		for (unsigned i = 0; i < jd->audio_format.channels; ++i) {
-			out = jack_port_get_buffer(jd->ports[i], nframes);
+			jack_default_audio_sample_t *out =
+				(jack_default_audio_sample_t *)
+				jack_port_get_buffer(jd->ports[i], nframes);
 
 			for (jack_nframes_t f = 0; f < nframes; ++f)
 				out[f] = 0.0;
@@ -145,8 +156,10 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 		available = nframes;
 
 	for (unsigned i = 0; i < jd->audio_format.channels; ++i) {
-		out = jack_port_get_buffer(jd->ports[i], nframes);
-		if (out == NULL)
+		jack_default_audio_sample_t *out =
+			(jack_default_audio_sample_t *)
+			jack_port_get_buffer(jd->ports[i], nframes);
+		if (out == nullptr)
 			/* workaround for libjack1 bug: if the server
 			   connection fails, the process callback is
 			   invoked anyway, but unable to get a
@@ -165,8 +178,10 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 
 	for (unsigned i = jd->audio_format.channels;
 	     i < jd->num_source_ports; ++i) {
-		out = jack_port_get_buffer(jd->ports[i], nframes);
-		if (out == NULL)
+		jack_default_audio_sample_t *out =
+			(jack_default_audio_sample_t *)
+			jack_port_get_buffer(jd->ports[i], nframes);
+		if (out == nullptr)
 			/* workaround for libjack1 bug: if the server
 			   connection fails, the process callback is
 			   invoked anyway, but unable to get a
@@ -183,23 +198,23 @@ mpd_jack_process(jack_nframes_t nframes, void *arg)
 static void
 mpd_jack_shutdown(void *arg)
 {
-	struct jack_data *jd = (struct jack_data *) arg;
+	JackOutput *jd = (JackOutput *) arg;
 	jd->shutdown = true;
 }
 
 static void
-set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
+set_audioformat(JackOutput *jd, AudioFormat &audio_format)
 {
-	audio_format->sample_rate = jack_get_sample_rate(jd->client);
+	audio_format.sample_rate = jack_get_sample_rate(jd->client);
 
 	if (jd->num_source_ports == 1)
-		audio_format->channels = 1;
-	else if (audio_format->channels > jd->num_source_ports)
-		audio_format->channels = 2;
+		audio_format.channels = 1;
+	else if (audio_format.channels > jd->num_source_ports)
+		audio_format.channels = 2;
 
-	if (audio_format->format != SAMPLE_FORMAT_S16 &&
-	    audio_format->format != SAMPLE_FORMAT_S24_P32)
-		audio_format->format = SAMPLE_FORMAT_S24_P32;
+	if (audio_format.format != SampleFormat::S16 &&
+	    audio_format.format != SampleFormat::S24_P32)
+		audio_format.format = SampleFormat::S24_P32;
 }
 
 static void
@@ -220,14 +235,14 @@ mpd_jack_info(const char *msg)
  * Disconnect the JACK client.
  */
 static void
-mpd_jack_disconnect(struct jack_data *jd)
+mpd_jack_disconnect(JackOutput *jd)
 {
-	assert(jd != NULL);
-	assert(jd->client != NULL);
+	assert(jd != nullptr);
+	assert(jd->client != nullptr);
 
 	jack_deactivate(jd->client);
 	jack_client_close(jd->client);
-	jd->client = NULL;
+	jd->client = nullptr;
 }
 
 /**
@@ -235,17 +250,17 @@ mpd_jack_disconnect(struct jack_data *jd)
  * (e.g. register callbacks).
  */
 static bool
-mpd_jack_connect(struct jack_data *jd, GError **error_r)
+mpd_jack_connect(JackOutput *jd, GError **error_r)
 {
 	jack_status_t status;
 
-	assert(jd != NULL);
+	assert(jd != nullptr);
 
 	jd->shutdown = false;
 
 	jd->client = jack_client_open(jd->name, jd->options, &status,
 				      jd->server_name);
-	if (jd->client == NULL) {
+	if (jd->client == nullptr) {
 		g_set_error(error_r, jack_output_quark(), 0,
 			    "Failed to connect to JACK server, status=%d",
 			    status);
@@ -260,7 +275,7 @@ mpd_jack_connect(struct jack_data *jd, GError **error_r)
 						  jd->source_ports[i],
 						  JACK_DEFAULT_AUDIO_TYPE,
 						  JackPortIsOutput, 0);
-		if (jd->ports[i] == NULL) {
+		if (jd->ports[i] == nullptr) {
 			g_set_error(error_r, jack_output_quark(), 0,
 				    "Cannot register output port \"%s\"",
 				    jd->source_ports[i]);
@@ -284,7 +299,7 @@ parse_port_list(int line, const char *source, char **dest, GError **error_r)
 	char **list = g_strsplit(source, ",", 0);
 	unsigned n = 0;
 
-	for (n = 0; list[n] != NULL; ++n) {
+	for (n = 0; list[n] != nullptr; ++n) {
 		if (n >= MAX_PORTS) {
 			g_set_error(error_r, jack_output_quark(), 0,
 				    "too many port names in line %d",
@@ -308,59 +323,59 @@ parse_port_list(int line, const char *source, char **dest, GError **error_r)
 }
 
 static struct audio_output *
-mpd_jack_init(const struct config_param *param, GError **error_r)
+mpd_jack_init(const config_param &param, GError **error_r)
 {
-	struct jack_data *jd = g_new(struct jack_data, 1);
+	JackOutput *jd = new JackOutput();
 
-	if (!ao_base_init(&jd->base, &jack_output_plugin, param, error_r)) {
-		g_free(jd);
-		return NULL;
+	if (!jd->Initialize(param, error_r)) {
+		delete jd;
+		return nullptr;
 	}
 
 	const char *value;
 
 	jd->options = JackNullOption;
 
-	jd->name = config_get_block_string(param, "client_name", NULL);
-	if (jd->name != NULL)
-		jd->options |= JackUseExactName;
+	jd->name = param.GetBlockValue("client_name", nullptr);
+	if (jd->name != nullptr)
+		jd->options = jack_options_t(jd->options | JackUseExactName);
 	else
 		/* if there's a no configured client name, we don't
 		   care about the JackUseExactName option */
 		jd->name = "Music Player Daemon";
 
-	jd->server_name = config_get_block_string(param, "server_name", NULL);
-	if (jd->server_name != NULL)
-		jd->options |= JackServerName;
+	jd->server_name = param.GetBlockValue("server_name", nullptr);
+	if (jd->server_name != nullptr)
+		jd->options = jack_options_t(jd->options | JackServerName);
 
-	if (!config_get_block_bool(param, "autostart", false))
-		jd->options |= JackNoStartServer;
+	if (!param.GetBlockValue("autostart", false))
+		jd->options = jack_options_t(jd->options | JackNoStartServer);
 
 	/* configure the source ports */
 
-	value = config_get_block_string(param, "source_ports", "left,right");
-	jd->num_source_ports = parse_port_list(param->line, value,
+	value = param.GetBlockValue("source_ports", "left,right");
+	jd->num_source_ports = parse_port_list(param.line, value,
 					       jd->source_ports, error_r);
 	if (jd->num_source_ports == 0)
-		return NULL;
+		return nullptr;
 
 	/* configure the destination ports */
 
-	value = config_get_block_string(param, "destination_ports", NULL);
-	if (value == NULL) {
+	value = param.GetBlockValue("destination_ports", nullptr);
+	if (value == nullptr) {
 		/* compatibility with MPD < 0.16 */
-		value = config_get_block_string(param, "ports", NULL);
-		if (value != NULL)
+		value = param.GetBlockValue("ports", nullptr);
+		if (value != nullptr)
 			g_warning("deprecated option 'ports' in line %d",
-				  param->line);
+				  param.line);
 	}
 
-	if (value != NULL) {
+	if (value != nullptr) {
 		jd->num_destination_ports =
-			parse_port_list(param->line, value,
+			parse_port_list(param.line, value,
 					jd->destination_ports, error_r);
 		if (jd->num_destination_ports == 0)
-			return NULL;
+			return nullptr;
 	} else {
 		jd->num_destination_ports = 0;
 	}
@@ -370,10 +385,9 @@ mpd_jack_init(const struct config_param *param, GError **error_r)
 		g_warning("number of source ports (%u) mismatches the "
 			  "number of destination ports (%u) in line %d",
 			  jd->num_source_ports, jd->num_destination_ports,
-			  param->line);
+			  param.line);
 
-	jd->ringbuffer_size =
-		config_get_block_unsigned(param, "ringbuffer_size", 32768);
+	jd->ringbuffer_size = param.GetBlockValue("ringbuffer_size", 32768u);
 
 	jack_set_error_function(mpd_jack_error);
 
@@ -387,7 +401,7 @@ mpd_jack_init(const struct config_param *param, GError **error_r)
 static void
 mpd_jack_finish(struct audio_output *ao)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
+	JackOutput *jd = (JackOutput *)ao;
 
 	for (unsigned i = 0; i < jd->num_source_ports; ++i)
 		g_free(jd->source_ports[i]);
@@ -395,17 +409,17 @@ mpd_jack_finish(struct audio_output *ao)
 	for (unsigned i = 0; i < jd->num_destination_ports; ++i)
 		g_free(jd->destination_ports[i]);
 
-	ao_base_finish(&jd->base);
-	g_free(jd);
+	jd->Deinitialize();
+	delete jd;
 }
 
 static bool
 mpd_jack_enable(struct audio_output *ao, GError **error_r)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
+	JackOutput *jd = (JackOutput *)ao;
 
 	for (unsigned i = 0; i < jd->num_source_ports; ++i)
-		jd->ringbuffer[i] = NULL;
+		jd->ringbuffer[i] = nullptr;
 
 	return mpd_jack_connect(jd, error_r);
 }
@@ -413,15 +427,15 @@ mpd_jack_enable(struct audio_output *ao, GError **error_r)
 static void
 mpd_jack_disable(struct audio_output *ao)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
+	JackOutput *jd = (JackOutput *)ao;
 
-	if (jd->client != NULL)
+	if (jd->client != nullptr)
 		mpd_jack_disconnect(jd);
 
 	for (unsigned i = 0; i < jd->num_source_ports; ++i) {
-		if (jd->ringbuffer[i] != NULL) {
+		if (jd->ringbuffer[i] != nullptr) {
 			jack_ringbuffer_free(jd->ringbuffer[i]);
-			jd->ringbuffer[i] = NULL;
+			jd->ringbuffer[i] = nullptr;
 		}
 	}
 }
@@ -430,11 +444,11 @@ mpd_jack_disable(struct audio_output *ao)
  * Stops the playback on the JACK connection.
  */
 static void
-mpd_jack_stop(struct jack_data *jd)
+mpd_jack_stop(JackOutput *jd)
 {
-	assert(jd != NULL);
+	assert(jd != nullptr);
 
-	if (jd->client == NULL)
+	if (jd->client == nullptr)
 		return;
 
 	if (jd->shutdown)
@@ -446,13 +460,13 @@ mpd_jack_stop(struct jack_data *jd)
 }
 
 static bool
-mpd_jack_start(struct jack_data *jd, GError **error_r)
+mpd_jack_start(JackOutput *jd, GError **error_r)
 {
 	const char *destination_ports[MAX_PORTS], **jports;
-	const char *duplicate_port = NULL;
+	const char *duplicate_port = nullptr;
 	unsigned num_destination_ports;
 
-	assert(jd->client != NULL);
+	assert(jd->client != nullptr);
 	assert(jd->audio_format.channels <= jd->num_source_ports);
 
 	/* allocate the ring buffers on the first open(); these
@@ -460,7 +474,7 @@ mpd_jack_start(struct jack_data *jd, GError **error_r)
 	   because we can never know when mpd_jack_process() gets
 	   called */
 	for (unsigned i = 0; i < jd->num_source_ports; ++i) {
-		if (jd->ringbuffer[i] == NULL)
+		if (jd->ringbuffer[i] == nullptr)
 			jd->ringbuffer[i] =
 				jack_ringbuffer_create(jd->ringbuffer_size);
 
@@ -479,20 +493,20 @@ mpd_jack_start(struct jack_data *jd, GError **error_r)
 	if (jd->num_destination_ports == 0) {
 		/* no output ports were configured - ask libjack for
 		   defaults */
-		jports = jack_get_ports(jd->client, NULL, NULL,
+		jports = jack_get_ports(jd->client, nullptr, nullptr,
 					JackPortIsPhysical | JackPortIsInput);
-		if (jports == NULL) {
+		if (jports == nullptr) {
 			g_set_error(error_r, jack_output_quark(), 0,
 				    "no ports found");
 			mpd_jack_stop(jd);
 			return false;
 		}
 
-		assert(*jports != NULL);
+		assert(*jports != nullptr);
 
 		for (num_destination_ports = 0;
 		     num_destination_ports < MAX_PORTS &&
-			     jports[num_destination_ports] != NULL;
+			     jports[num_destination_ports] != nullptr;
 		     ++num_destination_ports) {
 			g_debug("destination_port[%u] = '%s'\n",
 				num_destination_ports,
@@ -507,7 +521,7 @@ mpd_jack_start(struct jack_data *jd, GError **error_r)
 		memcpy(destination_ports, jd->destination_ports,
 		       num_destination_ports * sizeof(*destination_ports));
 
-		jports = NULL;
+		jports = nullptr;
 	}
 
 	assert(num_destination_ports > 0);
@@ -541,7 +555,7 @@ mpd_jack_start(struct jack_data *jd, GError **error_r)
 				    "Not a valid JACK port: %s",
 				    destination_ports[i]);
 
-			if (jports != NULL)
+			if (jports != nullptr)
 				free(jports);
 
 			mpd_jack_stop(jd);
@@ -549,7 +563,7 @@ mpd_jack_start(struct jack_data *jd, GError **error_r)
 		}
 	}
 
-	if (duplicate_port != NULL) {
+	if (duplicate_port != nullptr) {
 		/* mono input file: connect the one source channel to
 		   the both destination channels */
 		int ret;
@@ -561,7 +575,7 @@ mpd_jack_start(struct jack_data *jd, GError **error_r)
 				    "Not a valid JACK port: %s",
 				    duplicate_port);
 
-			if (jports != NULL)
+			if (jports != nullptr)
 				free(jports);
 
 			mpd_jack_stop(jd);
@@ -569,30 +583,30 @@ mpd_jack_start(struct jack_data *jd, GError **error_r)
 		}
 	}
 
-	if (jports != NULL)
+	if (jports != nullptr)
 		free(jports);
 
 	return true;
 }
 
 static bool
-mpd_jack_open(struct audio_output *ao, struct audio_format *audio_format,
+mpd_jack_open(struct audio_output *ao, AudioFormat &audio_format,
 	      GError **error_r)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
+	JackOutput *jd = (JackOutput *)ao;
 
-	assert(jd != NULL);
+	assert(jd != nullptr);
 
 	jd->pause = false;
 
-	if (jd->client != NULL && jd->shutdown)
+	if (jd->client != nullptr && jd->shutdown)
 		mpd_jack_disconnect(jd);
 
-	if (jd->client == NULL && !mpd_jack_connect(jd, error_r))
+	if (jd->client == nullptr && !mpd_jack_connect(jd, error_r))
 		return false;
 
 	set_audioformat(jd, audio_format);
-	jd->audio_format = *audio_format;
+	jd->audio_format = audio_format;
 
 	if (!mpd_jack_start(jd, error_r))
 		return false;
@@ -603,7 +617,7 @@ mpd_jack_open(struct audio_output *ao, struct audio_format *audio_format,
 static void
 mpd_jack_close(G_GNUC_UNUSED struct audio_output *ao)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
+	JackOutput *jd = (JackOutput *)ao;
 
 	mpd_jack_stop(jd);
 }
@@ -611,7 +625,7 @@ mpd_jack_close(G_GNUC_UNUSED struct audio_output *ao)
 static unsigned
 mpd_jack_delay(struct audio_output *ao)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
+	JackOutput *jd = (JackOutput *)ao;
 
 	return jd->base.pause && jd->pause && !jd->shutdown
 		? 1000
@@ -625,7 +639,7 @@ sample_16_to_jack(int16_t sample)
 }
 
 static void
-mpd_jack_write_samples_16(struct jack_data *jd, const int16_t *src,
+mpd_jack_write_samples_16(JackOutput *jd, const int16_t *src,
 			  unsigned num_samples)
 {
 	jack_default_audio_sample_t sample;
@@ -634,7 +648,8 @@ mpd_jack_write_samples_16(struct jack_data *jd, const int16_t *src,
 	while (num_samples-- > 0) {
 		for (i = 0; i < jd->audio_format.channels; ++i) {
 			sample = sample_16_to_jack(*src++);
-			jack_ringbuffer_write(jd->ringbuffer[i], (void*)&sample,
+			jack_ringbuffer_write(jd->ringbuffer[i],
+					      (const char *)&sample,
 					      sizeof(sample));
 		}
 	}
@@ -647,7 +662,7 @@ sample_24_to_jack(int32_t sample)
 }
 
 static void
-mpd_jack_write_samples_24(struct jack_data *jd, const int32_t *src,
+mpd_jack_write_samples_24(JackOutput *jd, const int32_t *src,
 			  unsigned num_samples)
 {
 	jack_default_audio_sample_t sample;
@@ -656,29 +671,31 @@ mpd_jack_write_samples_24(struct jack_data *jd, const int32_t *src,
 	while (num_samples-- > 0) {
 		for (i = 0; i < jd->audio_format.channels; ++i) {
 			sample = sample_24_to_jack(*src++);
-			jack_ringbuffer_write(jd->ringbuffer[i], (void*)&sample,
+			jack_ringbuffer_write(jd->ringbuffer[i],
+					      (const char *)&sample,
 					      sizeof(sample));
 		}
 	}
 }
 
 static void
-mpd_jack_write_samples(struct jack_data *jd, const void *src,
+mpd_jack_write_samples(JackOutput *jd, const void *src,
 		       unsigned num_samples)
 {
 	switch (jd->audio_format.format) {
-	case SAMPLE_FORMAT_S16:
+	case SampleFormat::S16:
 		mpd_jack_write_samples_16(jd, (const int16_t*)src,
 					  num_samples);
 		break;
 
-	case SAMPLE_FORMAT_S24_P32:
+	case SampleFormat::S24_P32:
 		mpd_jack_write_samples_24(jd, (const int32_t*)src,
 					  num_samples);
 		break;
 
 	default:
 		assert(false);
+		gcc_unreachable();
 	}
 }
 
@@ -686,8 +703,8 @@ static size_t
 mpd_jack_play(struct audio_output *ao, const void *chunk, size_t size,
 	      GError **error_r)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
-	const size_t frame_size = audio_format_frame_size(&jd->audio_format);
+	JackOutput *jd = (JackOutput *)ao;
+	const size_t frame_size = jd->audio_format.GetFrameSize();
 	size_t space = 0, space1;
 
 	jd->pause = false;
@@ -730,7 +747,7 @@ mpd_jack_play(struct audio_output *ao, const void *chunk, size_t size,
 static bool
 mpd_jack_pause(struct audio_output *ao)
 {
-	struct jack_data *jd = (struct jack_data *)ao;
+	JackOutput *jd = (JackOutput *)ao;
 
 	if (jd->shutdown)
 		return false;
@@ -741,15 +758,19 @@ mpd_jack_pause(struct audio_output *ao)
 }
 
 const struct audio_output_plugin jack_output_plugin = {
-	.name = "jack",
-	.test_default_device = mpd_jack_test_default_device,
-	.init = mpd_jack_init,
-	.finish = mpd_jack_finish,
-	.enable = mpd_jack_enable,
-	.disable = mpd_jack_disable,
-	.open = mpd_jack_open,
-	.delay = mpd_jack_delay,
-	.play = mpd_jack_play,
-	.pause = mpd_jack_pause,
-	.close = mpd_jack_close,
+	"jack",
+	mpd_jack_test_default_device,
+	mpd_jack_init,
+	mpd_jack_finish,
+	mpd_jack_enable,
+	mpd_jack_disable,
+	mpd_jack_open,
+	mpd_jack_close,
+	mpd_jack_delay,
+	nullptr,
+	mpd_jack_play,
+	nullptr,
+	nullptr,
+	mpd_jack_pause,
+	nullptr,
 };
diff --git a/src/output/jack_output_plugin.h b/src/output/JackOutputPlugin.hxx
index 2f94ae7dc..908105ad2 100644
--- a/src/output/jack_output_plugin.h
+++ b/src/output/JackOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_JACK_OUTPUT_PLUGIN_H
-#define MPD_JACK_OUTPUT_PLUGIN_H
+#ifndef MPD_JACK_OUTPUT_PLUGIN_HXX
+#define MPD_JACK_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin jack_output_plugin;
 
diff --git a/src/output/NullOutputPlugin.cxx b/src/output/NullOutputPlugin.cxx
new file mode 100644
index 000000000..0ce32fbda
--- /dev/null
+++ b/src/output/NullOutputPlugin.cxx
@@ -0,0 +1,143 @@
+/*
+ * Copyright (C) 2003-2013 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 "NullOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "Timer.hxx"
+
+#include <assert.h>
+
+struct NullOutput {
+	struct audio_output base;
+
+	bool sync;
+
+	Timer *timer;
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &null_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
+};
+
+static struct audio_output *
+null_init(const config_param &param, GError **error_r)
+{
+	NullOutput *nd = new NullOutput();
+
+	if (!nd->Initialize(param, error_r)) {
+		delete nd;
+		return nullptr;
+	}
+
+	nd->sync = param.GetBlockValue("sync", true);
+
+	return &nd->base;
+}
+
+static void
+null_finish(struct audio_output *ao)
+{
+	NullOutput *nd = (NullOutput *)ao;
+
+	nd->Deinitialize();
+	delete nd;
+}
+
+static bool
+null_open(struct audio_output *ao, AudioFormat &audio_format,
+	  gcc_unused GError **error)
+{
+	NullOutput *nd = (NullOutput *)ao;
+
+	if (nd->sync)
+		nd->timer = new Timer(audio_format);
+
+	return true;
+}
+
+static void
+null_close(struct audio_output *ao)
+{
+	NullOutput *nd = (NullOutput *)ao;
+
+	if (nd->sync)
+		delete nd->timer;
+}
+
+static unsigned
+null_delay(struct audio_output *ao)
+{
+	NullOutput *nd = (NullOutput *)ao;
+
+	return nd->sync && nd->timer->IsStarted()
+		? nd->timer->GetDelay()
+		: 0;
+}
+
+static size_t
+null_play(struct audio_output *ao, gcc_unused const void *chunk, size_t size,
+	  gcc_unused GError **error)
+{
+	NullOutput *nd = (NullOutput *)ao;
+	Timer *timer = nd->timer;
+
+	if (!nd->sync)
+		return size;
+
+	if (!timer->IsStarted())
+		timer->Start();
+	timer->Add(size);
+
+	return size;
+}
+
+static void
+null_cancel(struct audio_output *ao)
+{
+	NullOutput *nd = (NullOutput *)ao;
+
+	if (!nd->sync)
+		return;
+
+	nd->timer->Reset();
+}
+
+const struct audio_output_plugin null_output_plugin = {
+	"null",
+	nullptr,
+	null_init,
+	null_finish,
+	nullptr,
+	nullptr,
+	null_open,
+	null_close,
+	null_delay,
+	nullptr,
+	null_play,
+	nullptr,
+	null_cancel,
+	nullptr,
+	nullptr,
+};
diff --git a/src/output/null_output_plugin.h b/src/output/NullOutputPlugin.hxx
index 392bf0aa3..a58f1cb13 100644
--- a/src/output/null_output_plugin.h
+++ b/src/output/NullOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_NULL_OUTPUT_PLUGIN_H
-#define MPD_NULL_OUTPUT_PLUGIN_H
+#ifndef MPD_NULL_OUTPUT_PLUGIN_HXX
+#define MPD_NULL_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin null_output_plugin;
 
diff --git a/src/output/osx_output_plugin.c b/src/output/OSXOutputPlugin.cxx
index cd51fca20..6e42b2518 100644
--- a/src/output/osx_output_plugin.c
+++ b/src/output/OSXOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2012 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,9 +18,11 @@
  */
 
 #include "config.h"
-#include "osx_output_plugin.h"
-#include "output_api.h"
-#include "fifo_buffer.h"
+#include "OSXOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "util/fifo_buffer.h"
+#include "thread/Mutex.hxx"
+#include "thread/Cond.hxx"
 
 #include <glib.h>
 #include <CoreAudio/AudioHardware.h>
@@ -30,7 +32,7 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "osx"
 
-struct osx_output {
+struct OSXOutput {
 	struct audio_output base;
 
 	/* configuration settings */
@@ -39,8 +41,8 @@ struct osx_output {
 	const char *device_name;
 
 	AudioUnit au;
-	GMutex *mutex;
-	GCond *condition;
+	Mutex mutex;
+	Cond condition;
 
 	struct fifo_buffer *buffer;
 };
@@ -63,9 +65,9 @@ osx_output_test_default_device(void)
 }
 
 static void
-osx_output_configure(struct osx_output *oo, const struct config_param *param)
+osx_output_configure(OSXOutput *oo, const config_param &param)
 {
-	const char *device = config_get_block_string(param, "device", NULL);
+	const char *device = param.GetBlockValue("device");
 
 	if (device == NULL || 0 == strcmp(device, "default")) {
 		oo->component_subtype = kAudioUnitSubType_DefaultOutput;
@@ -83,17 +85,15 @@ osx_output_configure(struct osx_output *oo, const struct config_param *param)
 }
 
 static struct audio_output *
-osx_output_init(const struct config_param *param, GError **error_r)
+osx_output_init(const config_param &param, GError **error_r)
 {
-	struct osx_output *oo = g_new(struct osx_output, 1);
+	OSXOutput *oo = new OSXOutput();
 	if (!ao_base_init(&oo->base, &osx_output_plugin, param, error_r)) {
-		g_free(oo);
+		delete oo;
 		return NULL;
 	}
 
 	osx_output_configure(oo, param);
-	oo->mutex = g_mutex_new();
-	oo->condition = g_cond_new();
 
 	return &oo->base;
 }
@@ -101,15 +101,13 @@ osx_output_init(const struct config_param *param, GError **error_r)
 static void
 osx_output_finish(struct audio_output *ao)
 {
-	struct osx_output *od = (struct osx_output *)ao;
+	OSXOutput *oo = (OSXOutput *)ao;
 
-	g_mutex_free(od->mutex);
-	g_cond_free(od->condition);
-	g_free(od);
+	delete oo;
 }
 
 static bool
-osx_output_set_device(struct osx_output *oo, GError **error)
+osx_output_set_device(OSXOutput *oo, GError **error)
 {
 	bool ret = true;
 	OSStatus status;
@@ -135,7 +133,7 @@ osx_output_set_device(struct osx_output *oo, GError **error)
 
 	/* what are the available audio device IDs? */
 	numdevices = size / sizeof(AudioDeviceID);
-	deviceids = g_malloc(size);
+	deviceids = new AudioDeviceID[numdevices];
 	status = AudioHardwareGetProperty(kAudioHardwarePropertyDevices,
 					  &size,
 					  deviceids);
@@ -192,8 +190,7 @@ osx_output_set_device(struct osx_output *oo, GError **error)
 		(unsigned int) deviceids[i], name);
 
 done:
-	if (deviceids != NULL)
-		g_free(deviceids);
+	delete[] deviceids;
 	return ret;
 }
 
@@ -205,13 +202,13 @@ osx_render(void *vdata,
 	   G_GNUC_UNUSED UInt32 in_number_frames,
 	   AudioBufferList *buffer_list)
 {
-	struct osx_output *od = (struct osx_output *) vdata;
+	OSXOutput *od = (OSXOutput *) vdata;
 	AudioBuffer *buffer = &buffer_list->mBuffers[0];
 	size_t buffer_size = buffer->mDataByteSize;
 
 	assert(od->buffer != NULL);
 
-	g_mutex_lock(od->mutex);
+	od->mutex.lock();
 
 	size_t nbytes;
 	const void *src = fifo_buffer_read(od->buffer, &nbytes);
@@ -225,8 +222,8 @@ osx_render(void *vdata,
 	} else
 		nbytes = 0;
 
-	g_cond_signal(od->condition);
-	g_mutex_unlock(od->mutex);
+	od->condition.signal();
+	od->mutex.unlock();
 
 	buffer->mDataByteSize = nbytes;
 
@@ -242,7 +239,7 @@ osx_render(void *vdata,
 static bool
 osx_output_enable(struct audio_output *ao, GError **error_r)
 {
-	struct osx_output *oo = (struct osx_output *)ao;
+	OSXOutput *oo = (OSXOutput *)ao;
 
 	ComponentDescription desc;
 	desc.componentType = kAudioUnitType_Output;
@@ -293,7 +290,7 @@ osx_output_enable(struct audio_output *ao, GError **error_r)
 static void
 osx_output_disable(struct audio_output *ao)
 {
-	struct osx_output *oo = (struct osx_output *)ao;
+	OSXOutput *oo = (OSXOutput *)ao;
 
 	CloseComponent(oo->au);
 }
@@ -301,17 +298,16 @@ osx_output_disable(struct audio_output *ao)
 static void
 osx_output_cancel(struct audio_output *ao)
 {
-	struct osx_output *od = (struct osx_output *)ao;
+	OSXOutput *od = (OSXOutput *)ao;
 
-	g_mutex_lock(od->mutex);
+	const ScopeLock protect(od->mutex);
 	fifo_buffer_clear(od->buffer);
-	g_mutex_unlock(od->mutex);
 }
 
 static void
 osx_output_close(struct audio_output *ao)
 {
-	struct osx_output *od = (struct osx_output *)ao;
+	OSXOutput *od = (OSXOutput *)ao;
 
 	AudioOutputUnitStop(od->au);
 	AudioUnitUninitialize(od->au);
@@ -320,30 +316,30 @@ osx_output_close(struct audio_output *ao)
 }
 
 static bool
-osx_output_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
+osx_output_open(struct audio_output *ao, AudioFormat &audio_format, GError **error)
 {
-	struct osx_output *od = (struct osx_output *)ao;
+	OSXOutput *od = (OSXOutput *)ao;
 
 	AudioStreamBasicDescription stream_description;
-	stream_description.mSampleRate = audio_format->sample_rate;
+	stream_description.mSampleRate = audio_format.sample_rate;
 	stream_description.mFormatID = kAudioFormatLinearPCM;
 	stream_description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
 
-	switch (audio_format->format) {
-	case SAMPLE_FORMAT_S8:
+	switch (audio_format.format) {
+	case SampleFormat::S8:
 		stream_description.mBitsPerChannel = 8;
 		break;
 
-	case SAMPLE_FORMAT_S16:
+	case SampleFormat::S16:
 		stream_description.mBitsPerChannel = 16;
 		break;
 
-	case SAMPLE_FORMAT_S32:
+	case SampleFormat::S32:
 		stream_description.mBitsPerChannel = 32;
 		break;
 
 	default:
-		audio_format->format = SAMPLE_FORMAT_S32;
+		audio_format.format = SampleFormat::S32;
 		stream_description.mBitsPerChannel = 32;
 		break;
 	}
@@ -352,11 +348,10 @@ osx_output_open(struct audio_output *ao, struct audio_format *audio_format, GErr
 	stream_description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
 
-	stream_description.mBytesPerPacket =
-		audio_format_frame_size(audio_format);
+	stream_description.mBytesPerPacket = audio_format.GetFrameSize();
 	stream_description.mFramesPerPacket = 1;
 	stream_description.mBytesPerFrame = stream_description.mBytesPerPacket;
-	stream_description.mChannelsPerFrame = audio_format->channels;
+	stream_description.mChannelsPerFrame = audio_format.channels;
 
 	ComponentResult result =
 		AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
@@ -378,8 +373,8 @@ osx_output_open(struct audio_output *ao, struct audio_format *audio_format, GErr
 	}
 
 	/* create a buffer of 1s */
-	od->buffer = fifo_buffer_new(audio_format->sample_rate *
-				     audio_format_frame_size(audio_format));
+	od->buffer = fifo_buffer_new(audio_format.sample_rate *
+				     audio_format.GetFrameSize());
 
 	status = AudioOutputUnitStart(od->au);
 	if (status != 0) {
@@ -397,9 +392,9 @@ static size_t
 osx_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		G_GNUC_UNUSED GError **error)
 {
-	struct osx_output *od = (struct osx_output *)ao;
+	OSXOutput *od = (OSXOutput *)ao;
 
-	g_mutex_lock(od->mutex);
+	const ScopeLock protect(od->mutex);
 
 	void *dest;
 	size_t max_length;
@@ -410,7 +405,7 @@ osx_output_play(struct audio_output *ao, const void *chunk, size_t size,
 			break;
 
 		/* wait for some free space in the buffer */
-		g_cond_wait(od->condition, od->mutex);
+		od->condition.wait(od->mutex);
 	}
 
 	if (size > max_length)
@@ -419,20 +414,23 @@ osx_output_play(struct audio_output *ao, const void *chunk, size_t size,
 	memcpy(dest, chunk, size);
 	fifo_buffer_append(od->buffer, size);
 
-	g_mutex_unlock(od->mutex);
-
 	return size;
 }
 
 const struct audio_output_plugin osx_output_plugin = {
-	.name = "osx",
-	.test_default_device = osx_output_test_default_device,
-	.init = osx_output_init,
-	.finish = osx_output_finish,
-	.enable = osx_output_enable,
-	.disable = osx_output_disable,
-	.open = osx_output_open,
-	.close = osx_output_close,
-	.play = osx_output_play,
-	.cancel = osx_output_cancel,
+	"osx",
+	osx_output_test_default_device,
+	osx_output_init,
+	osx_output_finish,
+	osx_output_enable,
+	osx_output_disable,
+	osx_output_open,
+	osx_output_close,
+	nullptr,
+	nullptr,
+	osx_output_play,
+	nullptr,
+	osx_output_cancel,
+	nullptr,
+	nullptr,
 };
diff --git a/src/output/osx_output_plugin.h b/src/output/OSXOutputPlugin.hxx
index 814702d4f..2a4172880 100644
--- a/src/output/osx_output_plugin.h
+++ b/src/output/OSXOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_OSX_OUTPUT_PLUGIN_H
-#define MPD_OSX_OUTPUT_PLUGIN_H
+#ifndef MPD_OSX_OUTPUT_PLUGIN_HXX
+#define MPD_OSX_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin osx_output_plugin;
 
diff --git a/src/output/openal_output_plugin.c b/src/output/OpenALOutputPlugin.cxx
index ebd35ef12..1864052fa 100644
--- a/src/output/openal_output_plugin.c
+++ b/src/output/OpenALOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,8 +18,8 @@
  */
 
 #include "config.h"
-#include "openal_output_plugin.h"
-#include "output_api.h"
+#include "OpenALOutputPlugin.hxx"
+#include "OutputAPI.hxx"
 
 #include <glib.h>
 
@@ -37,7 +37,7 @@
 /* should be enough for buffer size = 2048 */
 #define NUM_BUFFERS 16
 
-struct openal_data {
+struct OpenALOutput {
 	struct audio_output base;
 
 	const char *device_name;
@@ -48,6 +48,15 @@ struct openal_data {
 	ALuint source;
 	ALenum format;
 	ALuint frequency;
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &openal_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
 };
 
 static inline GQuark
@@ -57,33 +66,33 @@ openal_output_quark(void)
 }
 
 static ALenum
-openal_audio_format(struct audio_format *audio_format)
+openal_audio_format(AudioFormat &audio_format)
 {
-	/* note: cannot map SAMPLE_FORMAT_S8 to AL_FORMAT_STEREO8 or
+	/* note: cannot map SampleFormat::S8 to AL_FORMAT_STEREO8 or
 	   AL_FORMAT_MONO8 since OpenAL expects unsigned 8 bit
 	   samples, while MPD uses signed samples */
 
-	switch (audio_format->format) {
-	case SAMPLE_FORMAT_S16:
-		if (audio_format->channels == 2)
+	switch (audio_format.format) {
+	case SampleFormat::S16:
+		if (audio_format.channels == 2)
 			return AL_FORMAT_STEREO16;
-		if (audio_format->channels == 1)
+		if (audio_format.channels == 1)
 			return AL_FORMAT_MONO16;
 
 		/* fall back to mono */
-		audio_format->channels = 1;
+		audio_format.channels = 1;
 		return openal_audio_format(audio_format);
 
 	default:
 		/* fall back to 16 bit */
-		audio_format->format = SAMPLE_FORMAT_S16;
+		audio_format.format = SampleFormat::S16;
 		return openal_audio_format(audio_format);
 	}
 }
 
 G_GNUC_PURE
 static inline ALint
-openal_get_source_i(const struct openal_data *od, ALenum param)
+openal_get_source_i(const OpenALOutput *od, ALenum param)
 {
 	ALint value;
 	alGetSourcei(od->source, param, &value);
@@ -92,34 +101,34 @@ openal_get_source_i(const struct openal_data *od, ALenum param)
 
 G_GNUC_PURE
 static inline bool
-openal_has_processed(const struct openal_data *od)
+openal_has_processed(const OpenALOutput *od)
 {
 	return openal_get_source_i(od, AL_BUFFERS_PROCESSED) > 0;
 }
 
 G_GNUC_PURE
 static inline ALint
-openal_is_playing(const struct openal_data *od)
+openal_is_playing(const OpenALOutput *od)
 {
 	return openal_get_source_i(od, AL_SOURCE_STATE) == AL_PLAYING;
 }
 
 static bool
-openal_setup_context(struct openal_data *od,
+openal_setup_context(OpenALOutput *od,
 		     GError **error)
 {
 	od->device = alcOpenDevice(od->device_name);
 
-	if (od->device == NULL) {
+	if (od->device == nullptr) {
 		g_set_error(error, openal_output_quark(), 0,
 			    "Error opening OpenAL device \"%s\"\n",
 			    od->device_name);
 		return false;
 	}
 
-	od->context = alcCreateContext(od->device, NULL);
+	od->context = alcCreateContext(od->device, nullptr);
 
-	if (od->context == NULL) {
+	if (od->context == nullptr) {
 		g_set_error(error, openal_output_quark(), 0,
 			    "Error creating context for \"%s\"\n",
 			    od->device_name);
@@ -131,19 +140,17 @@ openal_setup_context(struct openal_data *od,
 }
 
 static struct audio_output *
-openal_init(const struct config_param *param, GError **error_r)
+openal_init(const config_param &param, GError **error_r)
 {
-	const char *device_name = config_get_block_string(param, "device", NULL);
-	struct openal_data *od;
-
-	if (device_name == NULL) {
-		device_name = alcGetString(NULL, ALC_DEFAULT_DEVICE_SPECIFIER);
+	const char *device_name = param.GetBlockValue("device");
+	if (device_name == nullptr) {
+		device_name = alcGetString(nullptr, ALC_DEFAULT_DEVICE_SPECIFIER);
 	}
 
-	od = g_new(struct openal_data, 1);
-	if (!ao_base_init(&od->base, &openal_output_plugin, param, error_r)) {
-		g_free(od);
-		return NULL;
+	OpenALOutput *od = new OpenALOutput();
+	if (!od->Initialize(param, error_r)) {
+		delete od;
+		return nullptr;
 	}
 
 	od->device_name = device_name;
@@ -154,17 +161,17 @@ openal_init(const struct config_param *param, GError **error_r)
 static void
 openal_finish(struct audio_output *ao)
 {
-	struct openal_data *od = (struct openal_data *)ao;
+	OpenALOutput *od = (OpenALOutput *)ao;
 
-	ao_base_finish(&od->base);
-	g_free(od);
+	od->Deinitialize();
+	delete od;
 }
 
 static bool
-openal_open(struct audio_output *ao, struct audio_format *audio_format,
+openal_open(struct audio_output *ao, AudioFormat &audio_format,
 	    GError **error)
 {
-	struct openal_data *od = (struct openal_data *)ao;
+	OpenALOutput *od = (OpenALOutput *)ao;
 
 	od->format = openal_audio_format(audio_format);
 
@@ -191,7 +198,7 @@ openal_open(struct audio_output *ao, struct audio_format *audio_format,
 	}
 
 	od->filled = 0;
-	od->frequency = audio_format->sample_rate;
+	od->frequency = audio_format.sample_rate;
 
 	return true;
 }
@@ -199,7 +206,7 @@ openal_open(struct audio_output *ao, struct audio_format *audio_format,
 static void
 openal_close(struct audio_output *ao)
 {
-	struct openal_data *od = (struct openal_data *)ao;
+	OpenALOutput *od = (OpenALOutput *)ao;
 
 	alcMakeContextCurrent(od->context);
 	alDeleteSources(1, &od->source);
@@ -211,7 +218,7 @@ openal_close(struct audio_output *ao)
 static unsigned
 openal_delay(struct audio_output *ao)
 {
-	struct openal_data *od = (struct openal_data *)ao;
+	OpenALOutput *od = (OpenALOutput *)ao;
 
 	return od->filled < NUM_BUFFERS || openal_has_processed(od)
 		? 0
@@ -225,7 +232,7 @@ static size_t
 openal_play(struct audio_output *ao, const void *chunk, size_t size,
 	    G_GNUC_UNUSED GError **error)
 {
-	struct openal_data *od = (struct openal_data *)ao;
+	OpenALOutput *od = (OpenALOutput *)ao;
 	ALuint buffer;
 
 	if (alcGetCurrentContext() != od->context) {
@@ -256,7 +263,7 @@ openal_play(struct audio_output *ao, const void *chunk, size_t size,
 static void
 openal_cancel(struct audio_output *ao)
 {
-	struct openal_data *od = (struct openal_data *)ao;
+	OpenALOutput *od = (OpenALOutput *)ao;
 
 	od->filled = 0;
 	alcMakeContextCurrent(od->context);
@@ -268,12 +275,19 @@ openal_cancel(struct audio_output *ao)
 }
 
 const struct audio_output_plugin openal_output_plugin = {
-	.name = "openal",
-	.init = openal_init,
-	.finish = openal_finish,
-	.open = openal_open,
-	.close = openal_close,
-	.delay = openal_delay,
-	.play = openal_play,
-	.cancel = openal_cancel,
+	"openal",
+	nullptr,
+	openal_init,
+	openal_finish,
+	nullptr,
+	nullptr,
+	openal_open,
+	openal_close,
+	openal_delay,
+	nullptr,
+	openal_play,
+	nullptr,
+	openal_cancel,
+	nullptr,
+	nullptr,
 };
diff --git a/src/output/openal_output_plugin.h b/src/output/OpenALOutputPlugin.hxx
index 25f6ccf46..e1ebf3d4f 100644
--- a/src/output/openal_output_plugin.h
+++ b/src/output/OpenALOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_OPENAL_OUTPUT_PLUGIN_H
-#define MPD_OPENAL_OUTPUT_PLUGIN_H
+#ifndef MPD_OPENAL_OUTPUT_PLUGIN_HXX
+#define MPD_OPENAL_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin openal_output_plugin;
 
diff --git a/src/output/oss_output_plugin.c b/src/output/OssOutputPlugin.cxx
index e366a4537..2ef0edd67 100644
--- a/src/output/oss_output_plugin.c
+++ b/src/output/OssOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,11 +18,10 @@
  */
 
 #include "config.h"
-#include "oss_output_plugin.h"
-#include "output_api.h"
-#include "mixer_list.h"
+#include "OssOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "MixerList.hxx"
 #include "fd_util.h"
-#include "glib_compat.h"
 
 #include <glib.h>
 
@@ -53,14 +52,15 @@
 #endif
 
 #ifdef AFMT_S24_PACKED
-#include "pcm_export.h"
+#include "pcm/PcmExport.hxx"
+#include "util/Manual.hxx"
 #endif
 
-struct oss_data {
+struct OssOutput {
 	struct audio_output base;
 
 #ifdef AFMT_S24_PACKED
-	struct pcm_export_state export;
+	Manual<PcmExport> pcm_export;
 #endif
 
 	int fd;
@@ -70,13 +70,24 @@ struct oss_data {
 	 * The current input audio format.  This is needed to reopen
 	 * the device after cancel().
 	 */
-	struct audio_format audio_format;
+	AudioFormat audio_format;
 
 	/**
 	 * The current OSS audio format.  This is needed to reopen the
 	 * device after cancel().
 	 */
 	int oss_format;
+
+	OssOutput():fd(-1), device(nullptr) {}
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &oss_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
 };
 
 /**
@@ -88,23 +99,6 @@ oss_output_quark(void)
 	return g_quark_from_static_string("oss_output");
 }
 
-static struct oss_data *
-oss_data_new(void)
-{
-	struct oss_data *ret = g_new(struct oss_data, 1);
-
-	ret->device = NULL;
-	ret->fd = -1;
-
-	return ret;
-}
-
-static void
-oss_data_free(struct oss_data *od)
-{
-	g_free(od);
-}
-
 enum oss_stat {
 	OSS_STAT_NO_ERROR = 0,
 	OSS_STAT_NOT_CHAR_DEV = -1,
@@ -163,17 +157,16 @@ oss_output_test_default_device(void)
 static struct audio_output *
 oss_open_default(GError **error)
 {
-	int i;
 	int err[G_N_ELEMENTS(default_devices)];
 	enum oss_stat ret[G_N_ELEMENTS(default_devices)];
 
-	for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
+	const config_param empty;
+	for (int i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
 		ret[i] = oss_stat_device(default_devices[i], &err[i]);
 		if (ret[i] == OSS_STAT_NO_ERROR) {
-			struct oss_data *od = oss_data_new();
-			if (!ao_base_init(&od->base, &oss_output_plugin, NULL,
-					  error)) {
-				g_free(od);
+			OssOutput *od = new OssOutput();
+			if (!od->Initialize(empty, error)) {
+				delete od;
 				return NULL;
 			}
 
@@ -182,7 +175,7 @@ oss_open_default(GError **error)
 		}
 	}
 
-	for (i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
+	for (int i = G_N_ELEMENTS(default_devices); --i >= 0; ) {
 		const char *dev = default_devices[i];
 		switch(ret[i]) {
 		case OSS_STAT_NO_ERROR:
@@ -209,14 +202,13 @@ oss_open_default(GError **error)
 }
 
 static struct audio_output *
-oss_output_init(const struct config_param *param, GError **error)
+oss_output_init(const config_param &param, GError **error_r)
 {
-	const char *device = config_get_block_string(param, "device", NULL);
+	const char *device = param.GetBlockValue("device");
 	if (device != NULL) {
-		struct oss_data *od = oss_data_new();
-		if (!ao_base_init(&od->base, &oss_output_plugin, param,
-				  error)) {
-			g_free(od);
+		OssOutput *od = new OssOutput();
+		if (!od->Initialize(param, error_r)) {
+			delete od;
 			return NULL;
 		}
 
@@ -224,16 +216,16 @@ oss_output_init(const struct config_param *param, GError **error)
 		return &od->base;
 	}
 
-	return oss_open_default(error);
+	return oss_open_default(error_r);
 }
 
 static void
 oss_output_finish(struct audio_output *ao)
 {
-	struct oss_data *od = (struct oss_data *)ao;
+	OssOutput *od = (OssOutput *)ao;
 
 	ao_base_finish(&od->base);
-	oss_data_free(od);
+	delete od;
 }
 
 #ifdef AFMT_S24_PACKED
@@ -241,24 +233,24 @@ oss_output_finish(struct audio_output *ao)
 static bool
 oss_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
 {
-	struct oss_data *od = (struct oss_data *)ao;
+	OssOutput *od = (OssOutput *)ao;
 
-	pcm_export_init(&od->export);
+	od->pcm_export.Construct();
 	return true;
 }
 
 static void
 oss_output_disable(struct audio_output *ao)
 {
-	struct oss_data *od = (struct oss_data *)ao;
+	OssOutput *od = (OssOutput *)ao;
 
-	pcm_export_deinit(&od->export);
+	od->pcm_export.Destruct();
 }
 
 #endif
 
 static void
-oss_close(struct oss_data *od)
+oss_close(OssOutput *od)
 {
 	if (od->fd >= 0)
 		close(od->fd);
@@ -317,10 +309,10 @@ oss_try_ioctl(int fd, unsigned long request, int value,
  * specified number is not supported.
  */
 static bool
-oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r)
+oss_setup_channels(int fd, AudioFormat &audio_format, GError **error_r)
 {
 	const char *const msg = "Failed to set channel count";
-	int channels = audio_format->channels;
+	int channels = audio_format.channels;
 	enum oss_setup_result result =
 		oss_try_ioctl_r(fd, SNDCTL_DSP_CHANNELS, &channels, msg, error_r);
 	switch (result) {
@@ -328,7 +320,7 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r)
 		if (!audio_valid_channel_count(channels))
 		    break;
 
-		audio_format->channels = channels;
+		audio_format.channels = channels;
 		return true;
 
 	case ERROR:
@@ -339,7 +331,7 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r)
 	}
 
 	for (unsigned i = 1; i < 2; ++i) {
-		if (i == audio_format->channels)
+		if (i == audio_format.channels)
 			/* don't try that again */
 			continue;
 
@@ -351,7 +343,7 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r)
 			if (!audio_valid_channel_count(channels))
 			    break;
 
-			audio_format->channels = channels;
+			audio_format.channels = channels;
 			return true;
 
 		case ERROR:
@@ -371,11 +363,11 @@ oss_setup_channels(int fd, struct audio_format *audio_format, GError **error_r)
  * specified sample rate is not supported.
  */
 static bool
-oss_setup_sample_rate(int fd, struct audio_format *audio_format,
+oss_setup_sample_rate(int fd, AudioFormat &audio_format,
 		      GError **error_r)
 {
 	const char *const msg = "Failed to set sample rate";
-	int sample_rate = audio_format->sample_rate;
+	int sample_rate = audio_format.sample_rate;
 	enum oss_setup_result result =
 		oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate,
 				msg, error_r);
@@ -384,7 +376,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
 		if (!audio_valid_sample_rate(sample_rate))
 			break;
 
-		audio_format->sample_rate = sample_rate;
+		audio_format.sample_rate = sample_rate;
 		return true;
 
 	case ERROR:
@@ -397,7 +389,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
 	static const int sample_rates[] = { 48000, 44100, 0 };
 	for (unsigned i = 0; sample_rates[i] != 0; ++i) {
 		sample_rate = sample_rates[i];
-		if (sample_rate == (int)audio_format->sample_rate)
+		if (sample_rate == (int)audio_format.sample_rate)
 			continue;
 
 		result = oss_try_ioctl_r(fd, SNDCTL_DSP_SPEED, &sample_rate,
@@ -407,7 +399,7 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
 			if (!audio_valid_sample_rate(sample_rate))
 				break;
 
-			audio_format->sample_rate = sample_rate;
+			audio_format.sample_rate = sample_rate;
 			return true;
 
 		case ERROR:
@@ -427,28 +419,28 @@ oss_setup_sample_rate(int fd, struct audio_format *audio_format,
  * AFMT_QUERY if there is no direct counterpart.
  */
 static int
-sample_format_to_oss(enum sample_format format)
+sample_format_to_oss(SampleFormat format)
 {
 	switch (format) {
-	case SAMPLE_FORMAT_UNDEFINED:
-	case SAMPLE_FORMAT_FLOAT:
-	case SAMPLE_FORMAT_DSD:
+	case SampleFormat::UNDEFINED:
+	case SampleFormat::FLOAT:
+	case SampleFormat::DSD:
 		return AFMT_QUERY;
 
-	case SAMPLE_FORMAT_S8:
+	case SampleFormat::S8:
 		return AFMT_S8;
 
-	case SAMPLE_FORMAT_S16:
+	case SampleFormat::S16:
 		return AFMT_S16_NE;
 
-	case SAMPLE_FORMAT_S24_P32:
+	case SampleFormat::S24_P32:
 #ifdef AFMT_S24_NE
 		return AFMT_S24_NE;
 #else
 		return AFMT_QUERY;
 #endif
 
-	case SAMPLE_FORMAT_S32:
+	case SampleFormat::S32:
 #ifdef AFMT_S32_NE
 		return AFMT_S32_NE;
 #else
@@ -461,50 +453,50 @@ sample_format_to_oss(enum sample_format format)
 
 /**
  * Convert an OSS sample format to its MPD counterpart.  Returns
- * SAMPLE_FORMAT_UNDEFINED if there is no direct counterpart.
+ * SampleFormat::UNDEFINED if there is no direct counterpart.
  */
-static enum sample_format
+static SampleFormat
 sample_format_from_oss(int format)
 {
 	switch (format) {
 	case AFMT_S8:
-		return SAMPLE_FORMAT_S8;
+		return SampleFormat::S8;
 
 	case AFMT_S16_NE:
-		return SAMPLE_FORMAT_S16;
+		return SampleFormat::S16;
 
 #ifdef AFMT_S24_PACKED
 	case AFMT_S24_PACKED:
-		return SAMPLE_FORMAT_S24_P32;
+		return SampleFormat::S24_P32;
 #endif
 
 #ifdef AFMT_S24_NE
 	case AFMT_S24_NE:
-		return SAMPLE_FORMAT_S24_P32;
+		return SampleFormat::S24_P32;
 #endif
 
 #ifdef AFMT_S32_NE
 	case AFMT_S32_NE:
-		return SAMPLE_FORMAT_S32;
+		return SampleFormat::S32;
 #endif
 
 	default:
-		return SAMPLE_FORMAT_UNDEFINED;
+		return SampleFormat::UNDEFINED;
 	}
 }
 
 /**
  * Probe one sample format.
  *
- * @return the selected sample format or SAMPLE_FORMAT_UNDEFINED on
+ * @return the selected sample format or SampleFormat::UNDEFINED on
  * error
  */
 static enum oss_setup_result
-oss_probe_sample_format(int fd, enum sample_format sample_format,
-			enum sample_format *sample_format_r,
+oss_probe_sample_format(int fd, SampleFormat sample_format,
+			SampleFormat *sample_format_r,
 			int *oss_format_r,
 #ifdef AFMT_S24_PACKED
-			struct pcm_export_state *export,
+			PcmExport &pcm_export,
 #endif
 			GError **error_r)
 {
@@ -518,7 +510,7 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
 				"Failed to set sample format", error_r);
 
 #ifdef AFMT_S24_PACKED
-	if (result == UNSUPPORTED && sample_format == SAMPLE_FORMAT_S24_P32) {
+	if (result == UNSUPPORTED && sample_format == SampleFormat::S24_P32) {
 		/* if the driver doesn't support padded 24 bit, try
 		   packed 24 bit */
 		oss_format = AFMT_S24_PACKED;
@@ -532,14 +524,14 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
 		return result;
 
 	sample_format = sample_format_from_oss(oss_format);
-	if (sample_format == SAMPLE_FORMAT_UNDEFINED)
+	if (sample_format == SampleFormat::UNDEFINED)
 		return UNSUPPORTED;
 
 	*sample_format_r = sample_format;
 	*oss_format_r = oss_format;
 
 #ifdef AFMT_S24_PACKED
-	pcm_export_open(export, sample_format, 0, false, false,
+	pcm_export.Open(sample_format, 0, false, false,
 			oss_format == AFMT_S24_PACKED,
 			oss_format == AFMT_S24_PACKED &&
 			G_BYTE_ORDER != G_LITTLE_ENDIAN);
@@ -553,24 +545,24 @@ oss_probe_sample_format(int fd, enum sample_format sample_format,
  * specified format is not supported.
  */
 static bool
-oss_setup_sample_format(int fd, struct audio_format *audio_format,
+oss_setup_sample_format(int fd, AudioFormat &audio_format,
 			int *oss_format_r,
 #ifdef AFMT_S24_PACKED
-			struct pcm_export_state *export,
+			PcmExport &pcm_export,
 #endif
 			GError **error_r)
 {
-	enum sample_format mpd_format;
+	SampleFormat mpd_format;
 	enum oss_setup_result result =
-		oss_probe_sample_format(fd, audio_format->format,
+		oss_probe_sample_format(fd, audio_format.format,
 					&mpd_format, oss_format_r,
 #ifdef AFMT_S24_PACKED
-					export,
+					pcm_export,
 #endif
 					error_r);
 	switch (result) {
 	case SUCCESS:
-		audio_format->format = mpd_format;
+		audio_format.format = mpd_format;
 		return true;
 
 	case ERROR:
@@ -586,29 +578,29 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
 	/* the requested sample format is not available - probe for
 	   other formats supported by MPD */
 
-	static const enum sample_format sample_formats[] = {
-		SAMPLE_FORMAT_S24_P32,
-		SAMPLE_FORMAT_S32,
-		SAMPLE_FORMAT_S16,
-		SAMPLE_FORMAT_S8,
-		SAMPLE_FORMAT_UNDEFINED /* sentinel */
+	static const SampleFormat sample_formats[] = {
+		SampleFormat::S24_P32,
+		SampleFormat::S32,
+		SampleFormat::S16,
+		SampleFormat::S8,
+		SampleFormat::UNDEFINED /* sentinel */
 	};
 
-	for (unsigned i = 0; sample_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) {
+	for (unsigned i = 0; sample_formats[i] != SampleFormat::UNDEFINED; ++i) {
 		mpd_format = sample_formats[i];
-		if (mpd_format == audio_format->format)
+		if (mpd_format == audio_format.format)
 			/* don't try that again */
 			continue;
 
 		result = oss_probe_sample_format(fd, mpd_format,
 						 &mpd_format, oss_format_r,
 #ifdef AFMT_S24_PACKED
-						 export,
+						 pcm_export,
 #endif
 						 error_r);
 		switch (result) {
 		case SUCCESS:
-			audio_format->format = mpd_format;
+			audio_format.format = mpd_format;
 			return true;
 
 		case ERROR:
@@ -628,14 +620,14 @@ oss_setup_sample_format(int fd, struct audio_format *audio_format,
  * Sets up the OSS device which was opened before.
  */
 static bool
-oss_setup(struct oss_data *od, struct audio_format *audio_format,
+oss_setup(OssOutput *od, AudioFormat &audio_format,
 	  GError **error_r)
 {
 	return oss_setup_channels(od->fd, audio_format, error_r) &&
 		oss_setup_sample_rate(od->fd, audio_format, error_r) &&
 		oss_setup_sample_format(od->fd, audio_format, &od->oss_format,
 #ifdef AFMT_S24_PACKED
-					&od->export,
+					od->pcm_export,
 #endif
 					error_r);
 }
@@ -644,7 +636,7 @@ oss_setup(struct oss_data *od, struct audio_format *audio_format,
  * Reopen the device with the saved audio_format, without any probing.
  */
 static bool
-oss_reopen(struct oss_data *od, GError **error_r)
+oss_reopen(OssOutput *od, GError **error_r)
 {
 	assert(od->fd < 0);
 
@@ -696,10 +688,10 @@ oss_reopen(struct oss_data *od, GError **error_r)
 }
 
 static bool
-oss_output_open(struct audio_output *ao, struct audio_format *audio_format,
+oss_output_open(struct audio_output *ao, AudioFormat &audio_format,
 		GError **error)
 {
-	struct oss_data *od = (struct oss_data *)ao;
+	OssOutput *od = (OssOutput *)ao;
 
 	od->fd = open_cloexec(od->device, O_WRONLY, 0);
 	if (od->fd < 0) {
@@ -714,14 +706,14 @@ oss_output_open(struct audio_output *ao, struct audio_format *audio_format,
 		return false;
 	}
 
-	od->audio_format = *audio_format;
+	od->audio_format = audio_format;
 	return true;
 }
 
 static void
 oss_output_close(struct audio_output *ao)
 {
-	struct oss_data *od = (struct oss_data *)ao;
+	OssOutput *od = (OssOutput *)ao;
 
 	oss_close(od);
 }
@@ -729,7 +721,7 @@ oss_output_close(struct audio_output *ao)
 static void
 oss_output_cancel(struct audio_output *ao)
 {
-	struct oss_data *od = (struct oss_data *)ao;
+	OssOutput *od = (OssOutput *)ao;
 
 	if (od->fd >= 0) {
 		ioctl(od->fd, SNDCTL_DSP_RESET, 0);
@@ -741,7 +733,7 @@ static size_t
 oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		GError **error)
 {
-	struct oss_data *od = (struct oss_data *)ao;
+	OssOutput *od = (OssOutput *)ao;
 	ssize_t ret;
 
 	/* reopen the device since it was closed by dropBufferedAudio */
@@ -749,14 +741,14 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		return 0;
 
 #ifdef AFMT_S24_PACKED
-	chunk = pcm_export(&od->export, chunk, size, &size);
+	chunk = od->pcm_export->Export(chunk, size, size);
 #endif
 
 	while (true) {
 		ret = write(od->fd, chunk, size);
 		if (ret > 0) {
 #ifdef AFMT_S24_PACKED
-			ret = pcm_export_source_size(&od->export, ret);
+			ret = od->pcm_export->CalcSourceSize(ret);
 #endif
 			return ret;
 		}
@@ -771,18 +763,25 @@ oss_output_play(struct audio_output *ao, const void *chunk, size_t size,
 }
 
 const struct audio_output_plugin oss_output_plugin = {
-	.name = "oss",
-	.test_default_device = oss_output_test_default_device,
-	.init = oss_output_init,
-	.finish = oss_output_finish,
+	"oss",
+	oss_output_test_default_device,
+	oss_output_init,
+	oss_output_finish,
 #ifdef AFMT_S24_PACKED
-	.enable = oss_output_enable,
-	.disable = oss_output_disable,
+	oss_output_enable,
+	oss_output_disable,
+#else
+	nullptr,
+	nullptr,
 #endif
-	.open = oss_output_open,
-	.close = oss_output_close,
-	.play = oss_output_play,
-	.cancel = oss_output_cancel,
-
-	.mixer_plugin = &oss_mixer_plugin,
+	oss_output_open,
+	oss_output_close,
+	nullptr,
+	nullptr,
+	oss_output_play,
+	nullptr,
+	oss_output_cancel,
+	nullptr,
+
+	&oss_mixer_plugin,
 };
diff --git a/src/output/oss_output_plugin.h b/src/output/OssOutputPlugin.hxx
index 2aecc2b3a..6c5c9530b 100644
--- a/src/output/oss_output_plugin.h
+++ b/src/output/OssOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_OSS_OUTPUT_PLUGIN_H
-#define MPD_OSS_OUTPUT_PLUGIN_H
+#ifndef MPD_OSS_OUTPUT_PLUGIN_HXX
+#define MPD_OSS_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin oss_output_plugin;
 
diff --git a/src/output/pipe_output_plugin.c b/src/output/PipeOutputPlugin.cxx
index 90c5a5331..f485f1554 100644
--- a/src/output/pipe_output_plugin.c
+++ b/src/output/PipeOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,17 +18,28 @@
  */
 
 #include "config.h"
-#include "pipe_output_plugin.h"
-#include "output_api.h"
+#include "PipeOutputPlugin.hxx"
+#include "OutputAPI.hxx"
 
 #include <stdio.h>
 #include <errno.h>
 
-struct pipe_output {
+struct PipeOutput {
 	struct audio_output base;
 
 	char *cmd;
 	FILE *fh;
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &pipe_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
+
+	bool Configure(const config_param &param, GError **error_r);
 };
 
 /**
@@ -40,22 +51,33 @@ pipe_output_quark(void)
 	return g_quark_from_static_string("pipe_output");
 }
 
+inline bool
+PipeOutput::Configure(const config_param &param, GError **error_r)
+{
+	cmd = param.DupBlockString("command");
+	if (cmd == nullptr) {
+		g_set_error(error_r, pipe_output_quark(), 0,
+			    "No \"command\" parameter specified");
+		return false;
+	}
+
+	return true;
+}
+
 static struct audio_output *
-pipe_output_init(const struct config_param *param,
-		 GError **error)
+pipe_output_init(const config_param &param, GError **error_r)
 {
-	struct pipe_output *pd = g_new(struct pipe_output, 1);
+	PipeOutput *pd = new PipeOutput();
 
-	if (!ao_base_init(&pd->base, &pipe_output_plugin, param, error)) {
-		g_free(pd);
-		return NULL;
+	if (!pd->Initialize(param, error_r)) {
+		delete pd;
+		return nullptr;
 	}
 
-	pd->cmd = config_dup_block_string(param, "command", NULL);
-	if (pd->cmd == NULL) {
-		g_set_error(error, pipe_output_quark(), 0,
-			    "No \"command\" parameter specified");
-		return NULL;
+	if (!pd->Configure(param, error_r)) {
+		pd->Deinitialize();
+		delete pd;
+		return nullptr;
 	}
 
 	return &pd->base;
@@ -64,22 +86,22 @@ pipe_output_init(const struct config_param *param,
 static void
 pipe_output_finish(struct audio_output *ao)
 {
-	struct pipe_output *pd = (struct pipe_output *)ao;
+	PipeOutput *pd = (PipeOutput *)ao;
 
 	g_free(pd->cmd);
-	ao_base_finish(&pd->base);
-	g_free(pd);
+	pd->Deinitialize();
+	delete pd;
 }
 
 static bool
 pipe_output_open(struct audio_output *ao,
-		 G_GNUC_UNUSED struct audio_format *audio_format,
+		 G_GNUC_UNUSED AudioFormat &audio_format,
 		 G_GNUC_UNUSED GError **error)
 {
-	struct pipe_output *pd = (struct pipe_output *)ao;
+	PipeOutput *pd = (PipeOutput *)ao;
 
 	pd->fh = popen(pd->cmd, "w");
-	if (pd->fh == NULL) {
+	if (pd->fh == nullptr) {
 		g_set_error(error, pipe_output_quark(), errno,
 			    "Error opening pipe \"%s\": %s",
 			    pd->cmd, g_strerror(errno));
@@ -92,7 +114,7 @@ pipe_output_open(struct audio_output *ao,
 static void
 pipe_output_close(struct audio_output *ao)
 {
-	struct pipe_output *pd = (struct pipe_output *)ao;
+	PipeOutput *pd = (PipeOutput *)ao;
 
 	pclose(pd->fh);
 }
@@ -100,7 +122,7 @@ pipe_output_close(struct audio_output *ao)
 static size_t
 pipe_output_play(struct audio_output *ao, const void *chunk, size_t size, GError **error)
 {
-	struct pipe_output *pd = (struct pipe_output *)ao;
+	PipeOutput *pd = (PipeOutput *)ao;
 	size_t ret;
 
 	ret = fwrite(chunk, 1, size, pd->fh);
@@ -112,10 +134,19 @@ pipe_output_play(struct audio_output *ao, const void *chunk, size_t size, GError
 }
 
 const struct audio_output_plugin pipe_output_plugin = {
-	.name = "pipe",
-	.init = pipe_output_init,
-	.finish = pipe_output_finish,
-	.open = pipe_output_open,
-	.close = pipe_output_close,
-	.play = pipe_output_play,
+	"pipe",
+	nullptr,
+	pipe_output_init,
+	pipe_output_finish,
+	nullptr,
+	nullptr,
+	pipe_output_open,
+	pipe_output_close,
+	nullptr,
+	nullptr,
+	pipe_output_play,
+	nullptr,
+	nullptr,
+	nullptr,
+	nullptr,
 };
diff --git a/src/output/pipe_output_plugin.h b/src/output/PipeOutputPlugin.hxx
index 9f014f829..f0c29706b 100644
--- a/src/output/pipe_output_plugin.h
+++ b/src/output/PipeOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_PIPE_OUTPUT_PLUGIN_H
-#define MPD_PIPE_OUTPUT_PLUGIN_H
+#ifndef MPD_PIPE_OUTPUT_PLUGIN_HXX
+#define MPD_PIPE_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin pipe_output_plugin;
 
diff --git a/src/output/pulse_output_plugin.c b/src/output/PulseOutputPlugin.cxx
index e267427df..f59c8d76e 100644
--- a/src/output/pulse_output_plugin.c
+++ b/src/output/PulseOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,10 +18,10 @@
  */
 
 #include "config.h"
-#include "pulse_output_plugin.h"
-#include "output_api.h"
-#include "mixer_list.h"
-#include "mixer/pulse_mixer_plugin.h"
+#include "PulseOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "MixerList.hxx"
+#include "mixer/PulseMixerPlugin.hxx"
 
 #include <glib.h>
 
@@ -45,14 +45,14 @@
 #define PA_CHECK_VERSION(a,b,c) false
 #endif
 
-struct pulse_output {
+struct PulseOutput {
 	struct audio_output base;
 
 	const char *name;
 	const char *server;
 	const char *sink;
 
-	struct pulse_mixer *mixer;
+	PulseMixer *mixer;
 
 	struct pa_threaded_mainloop *mainloop;
 	struct pa_context *context;
@@ -79,36 +79,36 @@ pulse_output_quark(void)
 }
 
 void
-pulse_output_lock(struct pulse_output *po)
+pulse_output_lock(PulseOutput *po)
 {
 	pa_threaded_mainloop_lock(po->mainloop);
 }
 
 void
-pulse_output_unlock(struct pulse_output *po)
+pulse_output_unlock(PulseOutput *po)
 {
 	pa_threaded_mainloop_unlock(po->mainloop);
 }
 
 void
-pulse_output_set_mixer(struct pulse_output *po, struct pulse_mixer *pm)
+pulse_output_set_mixer(PulseOutput *po, PulseMixer *pm)
 {
-	assert(po != NULL);
-	assert(po->mixer == NULL);
-	assert(pm != NULL);
+	assert(po != nullptr);
+	assert(po->mixer == nullptr);
+	assert(pm != nullptr);
 
 	po->mixer = pm;
 
-	if (po->mainloop == NULL)
+	if (po->mainloop == nullptr)
 		return;
 
 	pa_threaded_mainloop_lock(po->mainloop);
 
-	if (po->context != NULL &&
+	if (po->context != nullptr &&
 	    pa_context_get_state(po->context) == PA_CONTEXT_READY) {
 		pulse_mixer_on_connect(pm, po->context);
 
-		if (po->stream != NULL &&
+		if (po->stream != nullptr &&
 		    pa_stream_get_state(po->stream) == PA_STREAM_READY)
 			pulse_mixer_on_change(pm, po->context, po->stream);
 	}
@@ -117,23 +117,22 @@ pulse_output_set_mixer(struct pulse_output *po, struct pulse_mixer *pm)
 }
 
 void
-pulse_output_clear_mixer(struct pulse_output *po,
-			 G_GNUC_UNUSED struct pulse_mixer *pm)
+pulse_output_clear_mixer(PulseOutput *po, gcc_unused PulseMixer *pm)
 {
-	assert(po != NULL);
-	assert(pm != NULL);
+	assert(po != nullptr);
+	assert(pm != nullptr);
 	assert(po->mixer == pm);
 
-	po->mixer = NULL;
+	po->mixer = nullptr;
 }
 
 bool
-pulse_output_set_volume(struct pulse_output *po,
+pulse_output_set_volume(PulseOutput *po,
 			const struct pa_cvolume *volume, GError **error_r)
 {
 	pa_operation *o;
 
-	if (po->context == NULL || po->stream == NULL ||
+	if (po->context == nullptr || po->stream == nullptr ||
 	    pa_stream_get_state(po->stream) != PA_STREAM_READY) {
 		g_set_error(error_r, pulse_output_quark(), 0, "disconnected");
 		return false;
@@ -141,8 +140,8 @@ pulse_output_set_volume(struct pulse_output *po,
 
 	o = pa_context_set_sink_input_volume(po->context,
 					     pa_stream_get_index(po->stream),
-					     volume, NULL, NULL);
-	if (o == NULL) {
+					     volume, nullptr, nullptr);
+	if (o == nullptr) {
 		g_set_error(error_r, pulse_output_quark(), 0,
 			    "failed to set PulseAudio volume: %s",
 			    pa_strerror(pa_context_errno(po->context)));
@@ -166,8 +165,8 @@ pulse_wait_for_operation(struct pa_threaded_mainloop *mainloop,
 {
 	pa_operation_state_t state;
 
-	assert(mainloop != NULL);
-	assert(operation != NULL);
+	assert(mainloop != nullptr);
+	assert(operation != nullptr);
 
 	state = pa_operation_get_state(operation);
 	while (state == PA_OPERATION_RUNNING) {
@@ -188,7 +187,7 @@ static void
 pulse_output_stream_success_cb(G_GNUC_UNUSED pa_stream *s,
 			       G_GNUC_UNUSED int success, void *userdata)
 {
-	struct pulse_output *po = userdata;
+	PulseOutput *po = (PulseOutput *)userdata;
 
 	pa_threaded_mainloop_signal(po->mainloop, 0);
 }
@@ -196,11 +195,11 @@ pulse_output_stream_success_cb(G_GNUC_UNUSED pa_stream *s,
 static void
 pulse_output_context_state_cb(struct pa_context *context, void *userdata)
 {
-	struct pulse_output *po = userdata;
+	PulseOutput *po = (PulseOutput *)userdata;
 
 	switch (pa_context_get_state(context)) {
 	case PA_CONTEXT_READY:
-		if (po->mixer != NULL)
+		if (po->mixer != nullptr)
 			pulse_mixer_on_connect(po->mixer, context);
 
 		pa_threaded_mainloop_signal(po->mainloop, 0);
@@ -208,7 +207,7 @@ pulse_output_context_state_cb(struct pa_context *context, void *userdata)
 
 	case PA_CONTEXT_TERMINATED:
 	case PA_CONTEXT_FAILED:
-		if (po->mixer != NULL)
+		if (po->mixer != nullptr)
 			pulse_mixer_on_disconnect(po->mixer);
 
 		/* the caller thread might be waiting for these
@@ -229,15 +228,15 @@ pulse_output_subscribe_cb(pa_context *context,
 			  pa_subscription_event_type_t t,
 			  uint32_t idx, void *userdata)
 {
-	struct pulse_output *po = userdata;
-	pa_subscription_event_type_t facility
-		= t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK;
-	pa_subscription_event_type_t type
-		= t & PA_SUBSCRIPTION_EVENT_TYPE_MASK;
+	PulseOutput *po = (PulseOutput *)userdata;
+	pa_subscription_event_type_t facility =
+		pa_subscription_event_type_t(t & PA_SUBSCRIPTION_EVENT_FACILITY_MASK);
+	pa_subscription_event_type_t type =
+		pa_subscription_event_type_t(t & PA_SUBSCRIPTION_EVENT_TYPE_MASK);
 
-	if (po->mixer != NULL &&
+	if (po->mixer != nullptr &&
 	    facility == PA_SUBSCRIPTION_EVENT_SINK_INPUT &&
-	    po->stream != NULL &&
+	    po->stream != nullptr &&
 	    pa_stream_get_state(po->stream) == PA_STREAM_READY &&
 	    idx == pa_stream_get_index(po->stream) &&
 	    (type == PA_SUBSCRIPTION_EVENT_NEW ||
@@ -251,15 +250,15 @@ pulse_output_subscribe_cb(pa_context *context,
  * @return true on success, false on error
  */
 static bool
-pulse_output_connect(struct pulse_output *po, GError **error_r)
+pulse_output_connect(PulseOutput *po, GError **error_r)
 {
-	assert(po != NULL);
-	assert(po->context != NULL);
+	assert(po != nullptr);
+	assert(po->context != nullptr);
 
 	int error;
 
 	error = pa_context_connect(po->context, po->server,
-				   (pa_context_flags_t)0, NULL);
+				   (pa_context_flags_t)0, nullptr);
 	if (error < 0) {
 		g_set_error(error_r, pulse_output_quark(), 0,
 			    "pa_context_connect() has failed: %s",
@@ -274,21 +273,21 @@ pulse_output_connect(struct pulse_output *po, GError **error_r)
  * Frees and clears the stream.
  */
 static void
-pulse_output_delete_stream(struct pulse_output *po)
+pulse_output_delete_stream(PulseOutput *po)
 {
-	assert(po != NULL);
-	assert(po->stream != NULL);
+	assert(po != nullptr);
+	assert(po->stream != nullptr);
 
 #if PA_CHECK_VERSION(0,9,8)
-	pa_stream_set_suspended_callback(po->stream, NULL, NULL);
+	pa_stream_set_suspended_callback(po->stream, nullptr, nullptr);
 #endif
 
-	pa_stream_set_state_callback(po->stream, NULL, NULL);
-	pa_stream_set_write_callback(po->stream, NULL, NULL);
+	pa_stream_set_state_callback(po->stream, nullptr, nullptr);
+	pa_stream_set_write_callback(po->stream, nullptr, nullptr);
 
 	pa_stream_disconnect(po->stream);
 	pa_stream_unref(po->stream);
-	po->stream = NULL;
+	po->stream = nullptr;
 }
 
 /**
@@ -297,17 +296,17 @@ pulse_output_delete_stream(struct pulse_output *po)
  * Caller must lock the main loop.
  */
 static void
-pulse_output_delete_context(struct pulse_output *po)
+pulse_output_delete_context(PulseOutput *po)
 {
-	assert(po != NULL);
-	assert(po->context != NULL);
+	assert(po != nullptr);
+	assert(po->context != nullptr);
 
-	pa_context_set_state_callback(po->context, NULL, NULL);
-	pa_context_set_subscribe_callback(po->context, NULL, NULL);
+	pa_context_set_state_callback(po->context, nullptr, nullptr);
+	pa_context_set_subscribe_callback(po->context, nullptr, nullptr);
 
 	pa_context_disconnect(po->context);
 	pa_context_unref(po->context);
-	po->context = NULL;
+	po->context = nullptr;
 }
 
 /**
@@ -318,14 +317,14 @@ pulse_output_delete_context(struct pulse_output *po)
  * @return true on success, false on error
  */
 static bool
-pulse_output_setup_context(struct pulse_output *po, GError **error_r)
+pulse_output_setup_context(PulseOutput *po, GError **error_r)
 {
-	assert(po != NULL);
-	assert(po->mainloop != NULL);
+	assert(po != nullptr);
+	assert(po->mainloop != nullptr);
 
 	po->context = pa_context_new(pa_threaded_mainloop_get_api(po->mainloop),
 				     MPD_PULSE_NAME);
-	if (po->context == NULL) {
+	if (po->context == nullptr) {
 		g_set_error(error_r, pulse_output_quark(), 0,
 			    "pa_context_new() has failed");
 		return false;
@@ -345,26 +344,26 @@ pulse_output_setup_context(struct pulse_output *po, GError **error_r)
 }
 
 static struct audio_output *
-pulse_output_init(const struct config_param *param, GError **error_r)
+pulse_output_init(const config_param &param, GError **error_r)
 {
-	struct pulse_output *po;
+	PulseOutput *po;
 
 	g_setenv("PULSE_PROP_media.role", "music", true);
 
-	po = g_new(struct pulse_output, 1);
+	po = new PulseOutput();
 	if (!ao_base_init(&po->base, &pulse_output_plugin, param, error_r)) {
-		g_free(po);
-		return NULL;
+		delete po;
+		return nullptr;
 	}
 
-	po->name = config_get_block_string(param, "name", "mpd_pulse");
-	po->server = config_get_block_string(param, "server", NULL);
-	po->sink = config_get_block_string(param, "sink", NULL);
+	po->name = param.GetBlockValue("name", "mpd_pulse");
+	po->server = param.GetBlockValue("server");
+	po->sink = param.GetBlockValue("sink");
 
-	po->mixer = NULL;
-	po->mainloop = NULL;
-	po->context = NULL;
-	po->stream = NULL;
+	po->mixer = nullptr;
+	po->mainloop = nullptr;
+	po->context = nullptr;
+	po->stream = nullptr;
 
 	return &po->base;
 }
@@ -372,24 +371,24 @@ pulse_output_init(const struct config_param *param, GError **error_r)
 static void
 pulse_output_finish(struct audio_output *ao)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 
 	ao_base_finish(&po->base);
-	g_free(po);
+	delete po;
 }
 
 static bool
 pulse_output_enable(struct audio_output *ao, GError **error_r)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 
-	assert(po->mainloop == NULL);
-	assert(po->context == NULL);
+	assert(po->mainloop == nullptr);
+	assert(po->context == nullptr);
 
 	/* create the libpulse mainloop and start the thread */
 
 	po->mainloop = pa_threaded_mainloop_new();
-	if (po->mainloop == NULL) {
+	if (po->mainloop == nullptr) {
 		g_free(po);
 
 		g_set_error(error_r, pulse_output_quark(), 0,
@@ -402,7 +401,7 @@ pulse_output_enable(struct audio_output *ao, GError **error_r)
 	if (pa_threaded_mainloop_start(po->mainloop) < 0) {
 		pa_threaded_mainloop_unlock(po->mainloop);
 		pa_threaded_mainloop_free(po->mainloop);
-		po->mainloop = NULL;
+		po->mainloop = nullptr;
 
 		g_set_error(error_r, pulse_output_quark(), 0,
 			    "pa_threaded_mainloop_start() has failed");
@@ -415,7 +414,7 @@ pulse_output_enable(struct audio_output *ao, GError **error_r)
 		pa_threaded_mainloop_unlock(po->mainloop);
 		pa_threaded_mainloop_stop(po->mainloop);
 		pa_threaded_mainloop_free(po->mainloop);
-		po->mainloop = NULL;
+		po->mainloop = nullptr;
 		return false;
 	}
 
@@ -427,15 +426,15 @@ pulse_output_enable(struct audio_output *ao, GError **error_r)
 static void
 pulse_output_disable(struct audio_output *ao)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 
-	assert(po->mainloop != NULL);
+	assert(po->mainloop != nullptr);
 
 	pa_threaded_mainloop_stop(po->mainloop);
-	if (po->context != NULL)
+	if (po->context != nullptr)
 		pulse_output_delete_context(po);
 	pa_threaded_mainloop_free(po->mainloop);
-	po->mainloop = NULL;
+	po->mainloop = nullptr;
 }
 
 /**
@@ -447,13 +446,13 @@ pulse_output_disable(struct audio_output *ao)
  * @return true on success, false on error
  */
 static bool
-pulse_output_wait_connection(struct pulse_output *po, GError **error_r)
+pulse_output_wait_connection(PulseOutput *po, GError **error_r)
 {
-	assert(po->mainloop != NULL);
+	assert(po->mainloop != nullptr);
 
 	pa_context_state_t state;
 
-	if (po->context == NULL && !pulse_output_setup_context(po, error_r))
+	if (po->context == nullptr && !pulse_output_setup_context(po, error_r))
 		return false;
 
 	while (true) {
@@ -488,10 +487,10 @@ pulse_output_wait_connection(struct pulse_output *po, GError **error_r)
 static void
 pulse_output_stream_suspended_cb(G_GNUC_UNUSED pa_stream *stream, void *userdata)
 {
-	struct pulse_output *po = userdata;
+	PulseOutput *po = (PulseOutput *)userdata;
 
-	assert(stream == po->stream || po->stream == NULL);
-	assert(po->mainloop != NULL);
+	assert(stream == po->stream || po->stream == nullptr);
+	assert(po->mainloop != nullptr);
 
 	/* wake up the main loop to break out of the loop in
 	   pulse_output_play() */
@@ -503,15 +502,15 @@ pulse_output_stream_suspended_cb(G_GNUC_UNUSED pa_stream *stream, void *userdata
 static void
 pulse_output_stream_state_cb(pa_stream *stream, void *userdata)
 {
-	struct pulse_output *po = userdata;
+	PulseOutput *po = (PulseOutput *)userdata;
 
-	assert(stream == po->stream || po->stream == NULL);
-	assert(po->mainloop != NULL);
-	assert(po->context != NULL);
+	assert(stream == po->stream || po->stream == nullptr);
+	assert(po->mainloop != nullptr);
+	assert(po->context != nullptr);
 
 	switch (pa_stream_get_state(stream)) {
 	case PA_STREAM_READY:
-		if (po->mixer != NULL)
+		if (po->mixer != nullptr)
 			pulse_mixer_on_change(po->mixer, po->context, stream);
 
 		pa_threaded_mainloop_signal(po->mainloop, 0);
@@ -519,7 +518,7 @@ pulse_output_stream_state_cb(pa_stream *stream, void *userdata)
 
 	case PA_STREAM_FAILED:
 	case PA_STREAM_TERMINATED:
-		if (po->mixer != NULL)
+		if (po->mixer != nullptr)
 			pulse_mixer_on_disconnect(po->mixer);
 
 		pa_threaded_mainloop_signal(po->mainloop, 0);
@@ -535,9 +534,9 @@ static void
 pulse_output_stream_write_cb(G_GNUC_UNUSED pa_stream *stream, size_t nbytes,
 			     void *userdata)
 {
-	struct pulse_output *po = userdata;
+	PulseOutput *po = (PulseOutput *)userdata;
 
-	assert(po->mainloop != NULL);
+	assert(po->mainloop != nullptr);
 
 	po->writable = nbytes;
 	pa_threaded_mainloop_signal(po->mainloop, 0);
@@ -551,14 +550,14 @@ pulse_output_stream_write_cb(G_GNUC_UNUSED pa_stream *stream, size_t nbytes,
  * @return true on success, false on error
  */
 static bool
-pulse_output_setup_stream(struct pulse_output *po, const pa_sample_spec *ss,
+pulse_output_setup_stream(PulseOutput *po, const pa_sample_spec *ss,
 			  GError **error_r)
 {
-	assert(po != NULL);
-	assert(po->context != NULL);
+	assert(po != nullptr);
+	assert(po->context != nullptr);
 
-	po->stream = pa_stream_new(po->context, po->name, ss, NULL);
-	if (po->stream == NULL) {
+	po->stream = pa_stream_new(po->context, po->name, ss, nullptr);
+	if (po->stream == nullptr) {
 		g_set_error(error_r, pulse_output_quark(), 0,
 			    "pa_stream_new() has failed: %s",
 			    pa_strerror(pa_context_errno(po->context)));
@@ -579,18 +578,18 @@ pulse_output_setup_stream(struct pulse_output *po, const pa_sample_spec *ss,
 }
 
 static bool
-pulse_output_open(struct audio_output *ao, struct audio_format *audio_format,
+pulse_output_open(struct audio_output *ao, AudioFormat &audio_format,
 		  GError **error_r)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 	pa_sample_spec ss;
 	int error;
 
-	assert(po->mainloop != NULL);
+	assert(po->mainloop != nullptr);
 
 	pa_threaded_mainloop_lock(po->mainloop);
 
-	if (po->context != NULL) {
+	if (po->context != nullptr) {
 		switch (pa_context_get_state(po->context)) {
 		case PA_CONTEXT_UNCONNECTED:
 		case PA_CONTEXT_TERMINATED:
@@ -616,11 +615,11 @@ pulse_output_open(struct audio_output *ao, struct audio_format *audio_format,
 
 	/* MPD doesn't support the other pulseaudio sample formats, so
 	   we just force MPD to send us everything as 16 bit */
-	audio_format->format = SAMPLE_FORMAT_S16;
+	audio_format.format = SampleFormat::S16;
 
 	ss.format = PA_SAMPLE_S16NE;
-	ss.rate = audio_format->sample_rate;
-	ss.channels = audio_format->channels;
+	ss.rate = audio_format.sample_rate;
+	ss.channels = audio_format.channels;
 
 	/* create a stream .. */
 
@@ -632,7 +631,8 @@ pulse_output_open(struct audio_output *ao, struct audio_format *audio_format,
 	/* .. and connect it (asynchronously) */
 
 	error = pa_stream_connect_playback(po->stream, po->sink,
-					   NULL, 0, NULL, NULL);
+					   nullptr, pa_stream_flags_t(0),
+					   nullptr, nullptr);
 	if (error < 0) {
 		pulse_output_delete_stream(po);
 
@@ -655,17 +655,17 @@ pulse_output_open(struct audio_output *ao, struct audio_format *audio_format,
 static void
 pulse_output_close(struct audio_output *ao)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 	pa_operation *o;
 
-	assert(po->mainloop != NULL);
+	assert(po->mainloop != nullptr);
 
 	pa_threaded_mainloop_lock(po->mainloop);
 
 	if (pa_stream_get_state(po->stream) == PA_STREAM_READY) {
 		o = pa_stream_drain(po->stream,
 				    pulse_output_stream_success_cb, po);
-		if (o == NULL) {
+		if (o == nullptr) {
 			g_warning("pa_stream_drain() has failed: %s",
 				  pa_strerror(pa_context_errno(po->context)));
 		} else
@@ -674,7 +674,7 @@ pulse_output_close(struct audio_output *ao)
 
 	pulse_output_delete_stream(po);
 
-	if (po->context != NULL &&
+	if (po->context != nullptr &&
 	    pa_context_get_state(po->context) != PA_CONTEXT_READY)
 		pulse_output_delete_context(po);
 
@@ -688,7 +688,7 @@ pulse_output_close(struct audio_output *ao)
  * @return true on success, false on error
  */
 static bool
-pulse_output_wait_stream(struct pulse_output *po, GError **error_r)
+pulse_output_wait_stream(PulseOutput *po, GError **error_r)
 {
 	while (true) {
 		switch (pa_stream_get_state(po->stream)) {
@@ -716,9 +716,9 @@ pulse_output_wait_stream(struct pulse_output *po, GError **error_r)
  * 0.9.11, it uses a custom pause flag.
  */
 static bool
-pulse_output_stream_is_paused(struct pulse_output *po)
+pulse_output_stream_is_paused(PulseOutput *po)
 {
-	assert(po->stream != NULL);
+	assert(po->stream != nullptr);
 
 #if !defined(PA_CHECK_VERSION) || !PA_CHECK_VERSION(0,9,11)
 	return po->pause;
@@ -731,18 +731,18 @@ pulse_output_stream_is_paused(struct pulse_output *po)
  * Sets cork mode on the stream.
  */
 static bool
-pulse_output_stream_pause(struct pulse_output *po, bool pause,
+pulse_output_stream_pause(PulseOutput *po, bool pause,
 			  GError **error_r)
 {
 	pa_operation *o;
 
-	assert(po->mainloop != NULL);
-	assert(po->context != NULL);
-	assert(po->stream != NULL);
+	assert(po->mainloop != nullptr);
+	assert(po->context != nullptr);
+	assert(po->stream != nullptr);
 
 	o = pa_stream_cork(po->stream, pause,
 			   pulse_output_stream_success_cb, po);
-	if (o == NULL) {
+	if (o == nullptr) {
 		g_set_error(error_r, pulse_output_quark(), 0,
 			    "pa_stream_cork() has failed: %s",
 			    pa_strerror(pa_context_errno(po->context)));
@@ -765,7 +765,7 @@ pulse_output_stream_pause(struct pulse_output *po, bool pause,
 static unsigned
 pulse_output_delay(struct audio_output *ao)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 	unsigned result = 0;
 
 	pa_threaded_mainloop_lock(po->mainloop);
@@ -784,11 +784,11 @@ static size_t
 pulse_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		  GError **error_r)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 	int error;
 
-	assert(po->mainloop != NULL);
-	assert(po->stream != NULL);
+	assert(po->mainloop != nullptr);
+	assert(po->stream != nullptr);
 
 	pa_threaded_mainloop_lock(po->mainloop);
 
@@ -799,7 +799,7 @@ pulse_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		return 0;
 	}
 
-	assert(po->context != NULL);
+	assert(po->context != nullptr);
 
 	/* unpause if previously paused */
 
@@ -839,7 +839,7 @@ pulse_output_play(struct audio_output *ao, const void *chunk, size_t size,
 
 	po->writable -= size;
 
-	error = pa_stream_write(po->stream, chunk, size, NULL,
+	error = pa_stream_write(po->stream, chunk, size, nullptr,
 				0, PA_SEEK_RELATIVE);
 	pa_threaded_mainloop_unlock(po->mainloop);
 	if (error < 0) {
@@ -854,11 +854,11 @@ pulse_output_play(struct audio_output *ao, const void *chunk, size_t size,
 static void
 pulse_output_cancel(struct audio_output *ao)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
+	PulseOutput *po = (PulseOutput *)ao;
 	pa_operation *o;
 
-	assert(po->mainloop != NULL);
-	assert(po->stream != NULL);
+	assert(po->mainloop != nullptr);
+	assert(po->stream != nullptr);
 
 	pa_threaded_mainloop_lock(po->mainloop);
 
@@ -869,10 +869,10 @@ pulse_output_cancel(struct audio_output *ao)
 		return;
 	}
 
-	assert(po->context != NULL);
+	assert(po->context != nullptr);
 
 	o = pa_stream_flush(po->stream, pulse_output_stream_success_cb, po);
-	if (o == NULL) {
+	if (o == nullptr) {
 		g_warning("pa_stream_flush() has failed: %s",
 			  pa_strerror(pa_context_errno(po->context)));
 		pa_threaded_mainloop_unlock(po->mainloop);
@@ -886,11 +886,11 @@ pulse_output_cancel(struct audio_output *ao)
 static bool
 pulse_output_pause(struct audio_output *ao)
 {
-	struct pulse_output *po = (struct pulse_output *)ao;
-	GError *error = NULL;
+	PulseOutput *po = (PulseOutput *)ao;
+	GError *error = nullptr;
 
-	assert(po->mainloop != NULL);
-	assert(po->stream != NULL);
+	assert(po->mainloop != nullptr);
+	assert(po->stream != nullptr);
 
 	pa_threaded_mainloop_lock(po->mainloop);
 
@@ -903,7 +903,7 @@ pulse_output_pause(struct audio_output *ao)
 		return false;
 	}
 
-	assert(po->context != NULL);
+	assert(po->context != nullptr);
 
 	/* cork the stream */
 
@@ -923,33 +923,34 @@ pulse_output_pause(struct audio_output *ao)
 static bool
 pulse_output_test_default_device(void)
 {
-	struct pulse_output *po;
 	bool success;
 
-	po = (struct pulse_output *)pulse_output_init(NULL, NULL);
-	if (po == NULL)
+	const config_param empty;
+	PulseOutput *po = (PulseOutput *)pulse_output_init(empty, nullptr);
+	if (po == nullptr)
 		return false;
 
-	success = pulse_output_wait_connection(po, NULL);
+	success = pulse_output_wait_connection(po, nullptr);
 	pulse_output_finish(&po->base);
 
 	return success;
 }
 
 const struct audio_output_plugin pulse_output_plugin = {
-	.name = "pulse",
-
-	.test_default_device = pulse_output_test_default_device,
-	.init = pulse_output_init,
-	.finish = pulse_output_finish,
-	.enable = pulse_output_enable,
-	.disable = pulse_output_disable,
-	.open = pulse_output_open,
-	.delay = pulse_output_delay,
-	.play = pulse_output_play,
-	.cancel = pulse_output_cancel,
-	.pause = pulse_output_pause,
-	.close = pulse_output_close,
-
-	.mixer_plugin = &pulse_mixer_plugin,
+	"pulse",
+	pulse_output_test_default_device,
+	pulse_output_init,
+	pulse_output_finish,
+	pulse_output_enable,
+	pulse_output_disable,
+	pulse_output_open,
+	pulse_output_close,
+	pulse_output_delay,
+	nullptr,
+	pulse_output_play,
+	nullptr,
+	pulse_output_cancel,
+	pulse_output_pause,
+
+	&pulse_mixer_plugin,
 };
diff --git a/src/output/pulse_output_plugin.h b/src/output/PulseOutputPlugin.hxx
index 02a51f27b..58dc6703c 100644
--- a/src/output/pulse_output_plugin.h
+++ b/src/output/PulseOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,33 +17,39 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_PULSE_OUTPUT_PLUGIN_H
-#define MPD_PULSE_OUTPUT_PLUGIN_H
+#ifndef MPD_PULSE_OUTPUT_PLUGIN_HXX
+#define MPD_PULSE_OUTPUT_PLUGIN_HXX
 
-#include <stdbool.h>
+#include "gerror.h"
 
-#include <glib.h>
-
-struct pulse_output;
-struct pulse_mixer;
+struct PulseOutput;
+struct PulseMixer;
 struct pa_cvolume;
 
 extern const struct audio_output_plugin pulse_output_plugin;
 
+#ifdef __cplusplus
+extern "C" {
+#endif
+
 void
-pulse_output_lock(struct pulse_output *po);
+pulse_output_lock(PulseOutput *po);
 
 void
-pulse_output_unlock(struct pulse_output *po);
+pulse_output_unlock(PulseOutput *po);
 
 void
-pulse_output_set_mixer(struct pulse_output *po, struct pulse_mixer *pm);
+pulse_output_set_mixer(PulseOutput *po, PulseMixer *pm);
 
 void
-pulse_output_clear_mixer(struct pulse_output *po, struct pulse_mixer *pm);
+pulse_output_clear_mixer(PulseOutput *po, PulseMixer *pm);
 
 bool
-pulse_output_set_volume(struct pulse_output *po,
+pulse_output_set_volume(PulseOutput *po,
 			const struct pa_cvolume *volume, GError **error_r);
 
+#ifdef __cplusplus
+}
+#endif
+
 #endif
diff --git a/src/output/recorder_output_plugin.c b/src/output/RecorderOutputPlugin.cxx
index b84cb244c..afae17e84 100644
--- a/src/output/recorder_output_plugin.c
+++ b/src/output/RecorderOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,10 +18,10 @@
  */
 
 #include "config.h"
-#include "recorder_output_plugin.h"
-#include "output_api.h"
-#include "encoder_plugin.h"
-#include "encoder_list.h"
+#include "RecorderOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "EncoderPlugin.hxx"
+#include "EncoderList.hxx"
 #include "fd_util.h"
 #include "open.h"
 
@@ -34,13 +34,13 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "recorder"
 
-struct recorder_output {
+struct RecorderOutput {
 	struct audio_output base;
 
 	/**
 	 * The configured encoder plugin.
 	 */
-	struct encoder *encoder;
+	Encoder *encoder;
 
 	/**
 	 * The destination file name.
@@ -56,6 +56,24 @@ struct recorder_output {
 	 * The buffer for encoder_read().
 	 */
 	char buffer[32768];
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &recorder_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
+
+	bool Configure(const config_param &param, GError **error_r);
+
+	bool WriteToFile(const void *data, size_t length, GError **error_r);
+
+	/**
+	 * Writes pending data from the encoder to the output file.
+	 */
+	bool EncoderToFile(GError **error_r);
 };
 
 /**
@@ -67,68 +85,70 @@ recorder_output_quark(void)
 	return g_quark_from_static_string("recorder_output");
 }
 
-static struct audio_output *
-recorder_output_init(const struct config_param *param, GError **error_r)
+inline bool
+RecorderOutput::Configure(const config_param &param, GError **error_r)
 {
-	struct recorder_output *recorder = g_new(struct recorder_output, 1);
-	if (!ao_base_init(&recorder->base, &recorder_output_plugin, param,
-			  error_r)) {
-		g_free(recorder);
-		return NULL;
-	}
-
 	/* read configuration */
 
 	const char *encoder_name =
-		config_get_block_string(param, "encoder", "vorbis");
-	const struct encoder_plugin *encoder_plugin =
-		encoder_plugin_get(encoder_name);
-	if (encoder_plugin == NULL) {
+		param.GetBlockValue("encoder", "vorbis");
+	const auto encoder_plugin = encoder_plugin_get(encoder_name);
+	if (encoder_plugin == nullptr) {
 		g_set_error(error_r, recorder_output_quark(), 0,
 			    "No such encoder: %s", encoder_name);
-		goto failure;
+		return false;
 	}
 
-	recorder->path = config_get_block_string(param, "path", NULL);
-	if (recorder->path == NULL) {
+	path = param.GetBlockValue("path");
+	if (path == nullptr) {
 		g_set_error(error_r, recorder_output_quark(), 0,
 			    "'path' not configured");
-		goto failure;
+		return false;
 	}
 
 	/* initialize encoder */
 
-	recorder->encoder = encoder_init(encoder_plugin, param, error_r);
-	if (recorder->encoder == NULL)
-		goto failure;
+	encoder = encoder_init(*encoder_plugin, param, error_r);
+	if (encoder == nullptr)
+		return false;
 
-	return &recorder->base;
+	return true;
+}
+
+static audio_output *
+recorder_output_init(const config_param &param, GError **error_r)
+{
+	RecorderOutput *recorder = new RecorderOutput();
+
+	if (!recorder->Initialize(param, error_r)) {
+		delete recorder;
+		return nullptr;
+	}
+
+	if (!recorder->Configure(param, error_r)) {
+		recorder->Deinitialize();
+		delete recorder;
+		return nullptr;
+	}
 
-failure:
-	ao_base_finish(&recorder->base);
-	g_free(recorder);
-	return NULL;
+	return &recorder->base;
 }
 
 static void
 recorder_output_finish(struct audio_output *ao)
 {
-	struct recorder_output *recorder = (struct recorder_output *)ao;
+	RecorderOutput *recorder = (RecorderOutput *)ao;
 
 	encoder_finish(recorder->encoder);
-	ao_base_finish(&recorder->base);
-	g_free(recorder);
+	recorder->Deinitialize();
+	delete recorder;
 }
 
-static bool
-recorder_write_to_file(struct recorder_output *recorder,
-		       const void *_data, size_t length,
-		       GError **error_r)
+inline bool
+RecorderOutput::WriteToFile(const void *_data, size_t length, GError **error_r)
 {
 	assert(length > 0);
 
-	const int fd = recorder->fd;
-
 	const uint8_t *data = (const uint8_t *)_data, *end = data + length;
 
 	while (true) {
@@ -145,43 +165,37 @@ recorder_write_to_file(struct recorder_output *recorder,
 		} else if (errno != EINTR) {
 			g_set_error(error_r, recorder_output_quark(), 0,
 				    "Failed to write to '%s': %s",
-				    recorder->path, g_strerror(errno));
+				    path, g_strerror(errno));
 			return false;
 		}
 	}
 }
 
-/**
- * Writes pending data from the encoder to the output file.
- */
-static bool
-recorder_output_encoder_to_file(struct recorder_output *recorder,
-				GError **error_r)
+inline bool
+RecorderOutput::EncoderToFile(GError **error_r)
 {
-	assert(recorder->fd >= 0);
+	assert(fd >= 0);
 
 	while (true) {
 		/* read from the encoder */
 
-		size_t size = encoder_read(recorder->encoder, recorder->buffer,
-					   sizeof(recorder->buffer));
+		size_t size = encoder_read(encoder, buffer, sizeof(buffer));
 		if (size == 0)
 			return true;
 
 		/* write everything into the file */
 
-		if (!recorder_write_to_file(recorder, recorder->buffer, size,
-					    error_r))
+		if (!WriteToFile(buffer, size, error_r))
 			return false;
 	}
 }
 
 static bool
 recorder_output_open(struct audio_output *ao,
-		     struct audio_format *audio_format,
+		     AudioFormat &audio_format,
 		     GError **error_r)
 {
-	struct recorder_output *recorder = (struct recorder_output *)ao;
+	RecorderOutput *recorder = (RecorderOutput *)ao;
 
 	/* create the output file */
 
@@ -203,7 +217,7 @@ recorder_output_open(struct audio_output *ao,
 		return false;
 	}
 
-	if (!recorder_output_encoder_to_file(recorder, error_r)) {
+	if (!recorder->EncoderToFile(error_r)) {
 		encoder_close(recorder->encoder);
 		close(recorder->fd);
 		unlink(recorder->path);
@@ -216,12 +230,12 @@ recorder_output_open(struct audio_output *ao,
 static void
 recorder_output_close(struct audio_output *ao)
 {
-	struct recorder_output *recorder = (struct recorder_output *)ao;
+	RecorderOutput *recorder = (RecorderOutput *)ao;
 
 	/* flush the encoder and write the rest to the file */
 
-	if (encoder_end(recorder->encoder, NULL))
-		recorder_output_encoder_to_file(recorder, NULL);
+	if (encoder_end(recorder->encoder, nullptr))
+		recorder->EncoderToFile(nullptr);
 
 	/* now really close everything */
 
@@ -234,18 +248,27 @@ static size_t
 recorder_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		     GError **error_r)
 {
-	struct recorder_output *recorder = (struct recorder_output *)ao;
+	RecorderOutput *recorder = (RecorderOutput *)ao;
 
 	return encoder_write(recorder->encoder, chunk, size, error_r) &&
-		recorder_output_encoder_to_file(recorder, error_r)
+		recorder->EncoderToFile(error_r)
 		? size : 0;
 }
 
 const struct audio_output_plugin recorder_output_plugin = {
-	.name = "recorder",
-	.init = recorder_output_init,
-	.finish = recorder_output_finish,
-	.open = recorder_output_open,
-	.close = recorder_output_close,
-	.play = recorder_output_play,
+	"recorder",
+	nullptr,
+	recorder_output_init,
+	recorder_output_finish,
+	nullptr,
+	nullptr,
+	recorder_output_open,
+	recorder_output_close,
+	nullptr,
+	nullptr,
+	recorder_output_play,
+	nullptr,
+	nullptr,
+	nullptr,
+	nullptr,
 };
diff --git a/src/output/recorder_output_plugin.h b/src/output/RecorderOutputPlugin.hxx
index a9bf755bd..a27f51e23 100644
--- a/src/output/recorder_output_plugin.h
+++ b/src/output/RecorderOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_RECORDER_OUTPUT_PLUGIN_H
-#define MPD_RECORDER_OUTPUT_PLUGIN_H
+#ifndef MPD_RECORDER_OUTPUT_PLUGIN_HXX
+#define MPD_RECORDER_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin recorder_output_plugin;
 
diff --git a/src/output/roar_output_plugin.c b/src/output/RoarOutputPlugin.cxx
index 1c2c48321..36f7c395b 100644
--- a/src/output/roar_output_plugin.c
+++ b/src/output/RoarOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2010 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * Copyright (C) 2010-2011 Philipp 'ph3-der-loewe' Schafft
  * Copyright (C) 2010-2011 Hans-Kristian 'maister' Arntzen
  *
@@ -19,25 +19,23 @@
  */
 
 #include "config.h"
-#include "roar_output_plugin.h"
-#include "output_api.h"
-#include "mixer_list.h"
-#include "roar_output_plugin.h"
+#include "RoarOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "MixerList.hxx"
+#include "thread/Mutex.hxx"
 
 #include <glib.h>
-#include <stdint.h>
-#include <unistd.h>
-#include <stdlib.h>
-#include <string.h>
-#include <stdint.h>
 
+/* libroar/services.h declares roar_service_stream::new - work around
+   this C++ problem */
+#define new _new
 #include <roaraudio.h>
+#undef new
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "roaraudio"
 
-typedef struct roar
-{
+struct RoarOutput {
 	struct audio_output base;
 
 	roar_vs_t * vss;
@@ -47,9 +45,18 @@ typedef struct roar
 	int role;
 	struct roar_connection con;
 	struct roar_audio_info info;
-	GMutex *lock;
+	Mutex mutex;
 	volatile bool alive;
-} roar_t;
+
+	RoarOutput()
+		:err(ROAR_ERROR_NONE),
+		 host(nullptr), name(nullptr) {}
+
+	~RoarOutput() {
+		g_free(host);
+		g_free(name);
+	}
+};
 
 static inline GQuark
 roar_output_quark(void)
@@ -58,9 +65,9 @@ roar_output_quark(void)
 }
 
 static int
-roar_output_get_volume_locked(struct roar *roar)
+roar_output_get_volume_locked(RoarOutput *roar)
 {
-	if (roar->vss == NULL || !roar->alive)
+	if (roar->vss == nullptr || !roar->alive)
 		return -1;
 
 	float l, r;
@@ -72,20 +79,18 @@ roar_output_get_volume_locked(struct roar *roar)
 }
 
 int
-roar_output_get_volume(struct roar *roar)
+roar_output_get_volume(RoarOutput *roar)
 {
-	g_mutex_lock(roar->lock);
-	int volume = roar_output_get_volume_locked(roar);
-	g_mutex_unlock(roar->lock);
-	return volume;
+	const ScopeLock protect(roar->mutex);
+	return roar_output_get_volume_locked(roar);
 }
 
 static bool
-roar_output_set_volume_locked(struct roar *roar, unsigned volume)
+roar_output_set_volume_locked(RoarOutput *roar, unsigned volume)
 {
 	assert(volume <= 100);
 
-	if (roar->vss == NULL || !roar->alive)
+	if (roar->vss == nullptr || !roar->alive)
 		return false;
 
 	int error;
@@ -96,38 +101,34 @@ roar_output_set_volume_locked(struct roar *roar, unsigned volume)
 }
 
 bool
-roar_output_set_volume(struct roar *roar, unsigned volume)
+roar_output_set_volume(RoarOutput *roar, unsigned volume)
 {
-	g_mutex_lock(roar->lock);
-	bool success = roar_output_set_volume_locked(roar, volume);
-	g_mutex_unlock(roar->lock);
-	return success;
+	const ScopeLock protect(roar->mutex);
+	return roar_output_set_volume_locked(roar, volume);
 }
 
 static void
-roar_configure(struct roar * self, const struct config_param *param)
+roar_configure(RoarOutput *self, const config_param &param)
 {
-	self->host = config_dup_block_string(param, "server", NULL);
-	self->name = config_dup_block_string(param, "name", "MPD");
+	self->host = param.DupBlockString("server", nullptr);
+	self->name = param.DupBlockString("name", "MPD");
 
-	const char *role = config_get_block_string(param, "role", "music");
-	self->role = role != NULL
+	const char *role = param.GetBlockValue("role", "music");
+	self->role = role != nullptr
 		? roar_str2role(role)
 		: ROAR_ROLE_MUSIC;
 }
 
 static struct audio_output *
-roar_init(const struct config_param *param, GError **error_r)
+roar_init(const config_param &param, GError **error_r)
 {
-	struct roar *self = g_new0(struct roar, 1);
+	RoarOutput *self = new RoarOutput();
 
 	if (!ao_base_init(&self->base, &roar_output_plugin, param, error_r)) {
-		g_free(self);
-		return NULL;
+		delete self;
+		return nullptr;
 	}
 
-	self->lock = g_mutex_new();
-	self->err = ROAR_ERROR_NONE;
 	roar_configure(self, param);
 	return &self->base;
 }
@@ -135,70 +136,66 @@ roar_init(const struct config_param *param, GError **error_r)
 static void
 roar_finish(struct audio_output *ao)
 {
-	struct roar *self = (struct roar *)ao;
-
-	g_free(self->host);
-	g_free(self->name);
-	g_mutex_free(self->lock);
+	RoarOutput *self = (RoarOutput *)ao;
 
 	ao_base_finish(&self->base);
-	g_free(self);
+	delete self;
 }
 
 static void
 roar_use_audio_format(struct roar_audio_info *info,
-		      struct audio_format *audio_format)
+		      AudioFormat &audio_format)
 {
-	info->rate = audio_format->sample_rate;
-	info->channels = audio_format->channels;
+	info->rate = audio_format.sample_rate;
+	info->channels = audio_format.channels;
 	info->codec = ROAR_CODEC_PCM_S;
 
-	switch (audio_format->format) {
-	case SAMPLE_FORMAT_UNDEFINED:
+	switch (audio_format.format) {
+	case SampleFormat::UNDEFINED:
+	case SampleFormat::FLOAT:
+	case SampleFormat::DSD:
 		info->bits = 16;
-		audio_format->format = SAMPLE_FORMAT_S16;
+		audio_format.format = SampleFormat::S16;
 		break;
 
-	case SAMPLE_FORMAT_S8:
+	case SampleFormat::S8:
 		info->bits = 8;
 		break;
 
-	case SAMPLE_FORMAT_S16:
+	case SampleFormat::S16:
 		info->bits = 16;
 		break;
 
-	case SAMPLE_FORMAT_S24_P32:
+	case SampleFormat::S24_P32:
 		info->bits = 32;
-		audio_format->format = SAMPLE_FORMAT_S32;
+		audio_format.format = SampleFormat::S32;
 		break;
 
-	case SAMPLE_FORMAT_S32:
+	case SampleFormat::S32:
 		info->bits = 32;
 		break;
 	}
 }
 
 static bool
-roar_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
+roar_open(struct audio_output *ao, AudioFormat &audio_format, GError **error)
 {
-	struct roar *self = (struct roar *)ao;
-	g_mutex_lock(self->lock);
+	RoarOutput *self = (RoarOutput *)ao;
+	const ScopeLock protect(self->mutex);
 
 	if (roar_simple_connect(&(self->con), self->host, self->name) < 0)
 	{
 		g_set_error(error, roar_output_quark(), 0,
 				"Failed to connect to Roar server");
-		g_mutex_unlock(self->lock);
 		return false;
 	}
 
 	self->vss = roar_vs_new_from_con(&(self->con), &(self->err));
 
-	if (self->vss == NULL || self->err != ROAR_ERROR_NONE)
+	if (self->vss == nullptr || self->err != ROAR_ERROR_NONE)
 	{
 		g_set_error(error, roar_output_quark(), 0,
 				"Failed to connect to server");
-		g_mutex_unlock(self->lock);
 		return false;
 	}
 
@@ -208,43 +205,41 @@ roar_open(struct audio_output *ao, struct audio_format *audio_format, GError **e
 				&(self->err)) < 0)
 	{
 		g_set_error(error, roar_output_quark(), 0, "Failed to start stream");
-		g_mutex_unlock(self->lock);
 		return false;
 	}
 	roar_vs_role(self->vss, self->role, &(self->err));
 	self->alive = true;
 
-	g_mutex_unlock(self->lock);
 	return true;
 }
 
 static void
 roar_close(struct audio_output *ao)
 {
-	struct roar *self = (struct roar *)ao;
-	g_mutex_lock(self->lock);
+	RoarOutput *self = (RoarOutput *)ao;
+	const ScopeLock protect(self->mutex);
+
 	self->alive = false;
 
-	if (self->vss != NULL)
+	if (self->vss != nullptr)
 		roar_vs_close(self->vss, ROAR_VS_TRUE, &(self->err));
-	self->vss = NULL;
+	self->vss = nullptr;
 	roar_disconnect(&(self->con));
-	g_mutex_unlock(self->lock);
 }
 
 static void
-roar_cancel_locked(struct roar *self)
+roar_cancel_locked(RoarOutput *self)
 {
-	if (self->vss == NULL)
+	if (self->vss == nullptr)
 		return;
 
 	roar_vs_t *vss = self->vss;
-	self->vss = NULL;
+	self->vss = nullptr;
 	roar_vs_close(vss, ROAR_VS_TRUE, &(self->err));
 	self->alive = false;
 
 	vss = roar_vs_new_from_con(&(self->con), &(self->err));
-	if (vss == NULL)
+	if (vss == nullptr)
 		return;
 
 	if (roar_vs_stream(vss, &(self->info), ROAR_DIR_PLAY,
@@ -262,20 +257,19 @@ roar_cancel_locked(struct roar *self)
 static void
 roar_cancel(struct audio_output *ao)
 {
-	struct roar *self = (struct roar *)ao;
+	RoarOutput *self = (RoarOutput *)ao;
 
-	g_mutex_lock(self->lock);
+	const ScopeLock protect(self->mutex);
 	roar_cancel_locked(self);
-	g_mutex_unlock(self->lock);
 }
 
 static size_t
 roar_play(struct audio_output *ao, const void *chunk, size_t size, GError **error)
 {
-	struct roar *self = (struct roar *)ao;
+	RoarOutput *self = (RoarOutput *)ao;
 	ssize_t rc;
 
-	if (self->vss == NULL)
+	if (self->vss == nullptr)
 	{
 		g_set_error(error, roar_output_quark(), 0, "Connection is invalid");
 		return 0;
@@ -332,19 +326,20 @@ roar_tag_convert(enum tag_type type, bool *is_uuid)
 			return "HASH";
 
 		default:
-			return NULL;
+			return nullptr;
 	}
 }
 
 static void
-roar_send_tag(struct audio_output *ao, const struct tag *meta)
+roar_send_tag(struct audio_output *ao, const Tag *meta)
 {
-	struct roar *self = (struct roar *)ao;
+	RoarOutput *self = (RoarOutput *)ao;
 
-	if (self->vss == NULL)
+	if (self->vss == nullptr)
 		return;
 
-	g_mutex_lock(self->lock);
+	const ScopeLock protect(self->mutex);
+
 	size_t cnt = 1;
 	struct roar_keyval vals[32];
 	memset(vals, 0, sizeof(vals));
@@ -361,7 +356,7 @@ roar_send_tag(struct audio_output *ao, const struct tag *meta)
 	{
 		bool is_uuid = false;
 		const char *key = roar_tag_convert(meta->items[i]->type, &is_uuid);
-		if (key != NULL)
+		if (key != nullptr)
 		{
 			if (is_uuid)
 			{
@@ -383,19 +378,22 @@ roar_send_tag(struct audio_output *ao, const struct tag *meta)
 
 	for (unsigned i = 0; i < 32; i++)
 		g_free(vals[i].key);
-
-	g_mutex_unlock(self->lock);
 }
 
 const struct audio_output_plugin roar_output_plugin = {
-	.name = "roar",
-	.init = roar_init,
-	.finish = roar_finish,
-	.open = roar_open,
-	.play = roar_play,
-	.cancel = roar_cancel,
-	.close = roar_close,
-	.send_tag = roar_send_tag,
-
-	.mixer_plugin = &roar_mixer_plugin
+	"roar",
+	nullptr,
+	roar_init,
+	roar_finish,
+	nullptr,
+	nullptr,
+	roar_open,
+	roar_close,
+	nullptr,
+	roar_send_tag,
+	roar_play,
+	nullptr,
+	roar_cancel,
+	nullptr,
+	&roar_mixer_plugin,
 };
diff --git a/src/output/roar_output_plugin.h b/src/output/RoarOutputPlugin.hxx
index 78b628cc4..faa4b4d5c 100644
--- a/src/output/roar_output_plugin.h
+++ b/src/output/RoarOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -20,16 +20,14 @@
 #ifndef MPD_ROAR_OUTPUT_PLUGIN_H
 #define MPD_ROAR_OUTPUT_PLUGIN_H
 
-#include <stdbool.h>
-
-struct roar;
+struct RoarOutput;
 
 extern const struct audio_output_plugin roar_output_plugin;
 
 int
-roar_output_get_volume(struct roar *roar);
+roar_output_get_volume(RoarOutput *roar);
 
 bool
-roar_output_set_volume(struct roar *roar, unsigned volume);
+roar_output_set_volume(RoarOutput *roar, unsigned volume);
 
 #endif
diff --git a/src/output/shout_output_plugin.c b/src/output/ShoutOutputPlugin.cxx
index 56456a0ea..2d2c0afd0 100644
--- a/src/output/shout_output_plugin.c
+++ b/src/output/ShoutOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,10 +18,10 @@
  */
 
 #include "config.h"
-#include "shout_output_plugin.h"
-#include "output_api.h"
-#include "encoder_plugin.h"
-#include "encoder_list.h"
+#include "ShoutOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "EncoderPlugin.hxx"
+#include "EncoderList.hxx"
 #include "mpd_error.h"
 
 #include <shout/shout.h>
@@ -29,20 +29,21 @@
 
 #include <assert.h>
 #include <stdlib.h>
+#include <string.h>
 #include <stdio.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "shout"
 
-#define DEFAULT_CONN_TIMEOUT  2
+static constexpr unsigned DEFAULT_CONN_TIMEOUT = 2;
 
-struct shout_data {
+struct ShoutOutput final {
 	struct audio_output base;
 
 	shout_t *shout_conn;
 	shout_metadata_t *shout_meta;
 
-	struct encoder *encoder;
+	Encoder *encoder;
 
 	float quality;
 	int bitrate;
@@ -50,6 +51,31 @@ struct shout_data {
 	int timeout;
 
 	uint8_t buffer[32768];
+
+	ShoutOutput()
+		:shout_conn(shout_new()),
+		shout_meta(shout_metadata_new()),
+		quality(-2.0),
+		bitrate(-1),
+		timeout(DEFAULT_CONN_TIMEOUT) {}
+
+	~ShoutOutput() {
+		if (shout_meta != nullptr)
+			shout_metadata_free(shout_meta);
+		if (shout_conn != nullptr)
+			shout_free(shout_conn);
+	}
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &shout_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
+
+	bool Configure(const config_param &param, GError **error_r);
 };
 
 static int shout_init_count;
@@ -63,7 +89,7 @@ shout_output_quark(void)
 	return g_quark_from_static_string("shout_output");
 }
 
-static const struct encoder_plugin *
+static const EncoderPlugin *
 shout_encoder_plugin_get(const char *name)
 {
 	if (strcmp(name, "ogg") == 0)
@@ -74,137 +100,94 @@ shout_encoder_plugin_get(const char *name)
 	return encoder_plugin_get(name);
 }
 
-static struct shout_data *new_shout_data(void)
+gcc_pure
+static const char *
+require_block_string(const config_param &param, const char *name)
 {
-	struct shout_data *ret = g_new(struct shout_data, 1);
-
-	ret->shout_conn = shout_new();
-	ret->shout_meta = shout_metadata_new();
-	ret->bitrate = -1;
-	ret->quality = -2.0;
-	ret->timeout = DEFAULT_CONN_TIMEOUT;
+	const char *value = param.GetBlockValue(name);
+	if (value == nullptr)
+		MPD_ERROR("no \"%s\" defined for shout device defined at line " \
+			  "%i\n", name, param.line);
 
-	return ret;
+	return value;
 }
 
-static void free_shout_data(struct shout_data *sd)
-{
-	if (sd->shout_meta)
-		shout_metadata_free(sd->shout_meta);
-	if (sd->shout_conn)
-		shout_free(sd->shout_conn);
-
-	g_free(sd);
-}
-
-#define check_block_param(name) {		  \
-		block_param = config_get_block_param(param, name);	\
-		if (!block_param) {					\
-			MPD_ERROR("no \"%s\" defined for shout device defined at line " \
-				  "%i\n", name, param->line);		\
-		}							\
-	}
-
-static struct audio_output *
-my_shout_init_driver(const struct config_param *param,
-		     GError **error)
+inline bool
+ShoutOutput::Configure(const config_param &param, GError **error_r)
 {
-	struct shout_data *sd = new_shout_data();
-	if (!ao_base_init(&sd->base, &shout_output_plugin, param, error)) {
-		free_shout_data(sd);
-		return NULL;
-	}
 
-	const struct audio_format *audio_format =
-		&sd->base.config_audio_format;
-	if (!audio_format_fully_defined(audio_format)) {
-		g_set_error(error, shout_output_quark(), 0,
+	const AudioFormat audio_format = base.config_audio_format;
+	if (!audio_format.IsFullyDefined()) {
+		g_set_error(error_r, shout_output_quark(), 0,
 			    "Need full audio format specification");
-		ao_base_finish(&sd->base);
-		free_shout_data(sd);
-		return NULL;
+		return nullptr;
 	}
 
-	if (shout_init_count == 0)
-		shout_init();
-
-	shout_init_count++;
-
-	const struct block_param *block_param;
-	check_block_param("host");
-	char *host = block_param->value;
-
-	check_block_param("mount");
-	char *mount = block_param->value;
-
-	unsigned port = config_get_block_unsigned(param, "port", 0);
+	const char *host = require_block_string(param, "host");
+	const char *mount = require_block_string(param, "mount");
+	unsigned port = param.GetBlockValue("port", 0u);
 	if (port == 0) {
-		g_set_error(error, shout_output_quark(), 0,
+		g_set_error(error_r, shout_output_quark(), 0,
 			    "shout port must be configured");
-		goto failure;
+		return false;
 	}
 
-	check_block_param("password");
-	const char *passwd = block_param->value;
+	const char *passwd = require_block_string(param, "password");
+	const char *name = require_block_string(param, "name");
 
-	check_block_param("name");
-	const char *name = block_param->value;
+	bool is_public = param.GetBlockValue("public", false);
 
-	bool public = config_get_block_bool(param, "public", false);
+	const char *user = param.GetBlockValue("user", "source");
 
-	const char *user = config_get_block_string(param, "user", "source");
-
-	const char *value = config_get_block_string(param, "quality", NULL);
-	if (value != NULL) {
+	const char *value = param.GetBlockValue("quality");
+	if (value != nullptr) {
 		char *test;
-		sd->quality = strtod(value, &test);
+		quality = strtod(value, &test);
 
-		if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
-			g_set_error(error, shout_output_quark(), 0,
+		if (*test != '\0' || quality < -1.0 || quality > 10.0) {
+			g_set_error(error_r, shout_output_quark(), 0,
 				    "shout quality \"%s\" is not a number in the "
 				    "range -1 to 10, line %i",
-				    value, param->line);
-			goto failure;
+				    value, param.line);
+			return false;
 		}
 
-		if (config_get_block_string(param, "bitrate", NULL) != NULL) {
-			g_set_error(error, shout_output_quark(), 0,
+		if (param.GetBlockValue("bitrate") != nullptr) {
+			g_set_error(error_r, shout_output_quark(), 0,
 				    "quality and bitrate are "
 				    "both defined");
-			goto failure;
+			return false;
 		}
 	} else {
-		value = config_get_block_string(param, "bitrate", NULL);
-		if (value == NULL) {
-			g_set_error(error, shout_output_quark(), 0,
+		value = param.GetBlockValue("bitrate");
+		if (value == nullptr) {
+			g_set_error(error_r, shout_output_quark(), 0,
 				    "neither bitrate nor quality defined");
-			goto failure;
+			return false;
 		}
 
 		char *test;
-		sd->bitrate = strtol(value, &test, 10);
+		bitrate = strtol(value, &test, 10);
 
-		if (*test != '\0' || sd->bitrate <= 0) {
-			g_set_error(error, shout_output_quark(), 0,
+		if (*test != '\0' || bitrate <= 0) {
+			g_set_error(error_r, shout_output_quark(), 0,
 				    "bitrate must be a positive integer");
-			goto failure;
+			return false;
 		}
 	}
 
-	const char *encoding = config_get_block_string(param, "encoding",
-						       "ogg");
-	const struct encoder_plugin *encoder_plugin =
-		shout_encoder_plugin_get(encoding);
-	if (encoder_plugin == NULL) {
-		g_set_error(error, shout_output_quark(), 0,
+	const char *encoding = param.GetBlockValue("encoding", "ogg");
+	const auto encoder_plugin = shout_encoder_plugin_get(encoding);
+	if (encoder_plugin == nullptr) {
+		g_set_error(error_r, shout_output_quark(), 0,
 			    "couldn't find shout encoder plugin \"%s\"",
 			    encoding);
-		goto failure;
+		return false;
 	}
 
-	sd->encoder = encoder_init(encoder_plugin, param, error);
-	if (sd->encoder == NULL)
-		goto failure;
+	encoder = encoder_init(*encoder_plugin, param, error_r);
+	if (encoder == nullptr)
+		return false;
 
 	unsigned shout_format;
 	if (strcmp(encoding, "mp3") == 0 || strcmp(encoding, "lame") == 0)
@@ -213,14 +196,14 @@ my_shout_init_driver(const struct config_param *param,
 		shout_format = SHOUT_FORMAT_OGG;
 
 	unsigned protocol;
-	value = config_get_block_string(param, "protocol", NULL);
-	if (value != NULL) {
+	value = param.GetBlockValue("protocol");
+	if (value != nullptr) {
 		if (0 == strcmp(value, "shoutcast") &&
 		    0 != strcmp(encoding, "mp3")) {
-			g_set_error(error, shout_output_quark(), 0,
+			g_set_error(error_r, shout_output_quark(), 0,
 				    "you cannot stream \"%s\" to shoutcast, use mp3",
 				    encoding);
-			goto failure;
+			return false;
 		} else if (0 == strcmp(value, "shoutcast"))
 			protocol = SHOUT_PROTOCOL_ICY;
 		else if (0 == strcmp(value, "icecast1"))
@@ -228,89 +211,106 @@ my_shout_init_driver(const struct config_param *param,
 		else if (0 == strcmp(value, "icecast2"))
 			protocol = SHOUT_PROTOCOL_HTTP;
 		else {
-			g_set_error(error, shout_output_quark(), 0,
+			g_set_error(error_r, shout_output_quark(), 0,
 				    "shout protocol \"%s\" is not \"shoutcast\" or "
 				    "\"icecast1\"or \"icecast2\"",
 				    value);
-			goto failure;
+			return false;
 		}
 	} else {
 		protocol = SHOUT_PROTOCOL_HTTP;
 	}
 
-	if (shout_set_host(sd->shout_conn, host) != SHOUTERR_SUCCESS ||
-	    shout_set_port(sd->shout_conn, port) != SHOUTERR_SUCCESS ||
-	    shout_set_password(sd->shout_conn, passwd) != SHOUTERR_SUCCESS ||
-	    shout_set_mount(sd->shout_conn, mount) != SHOUTERR_SUCCESS ||
-	    shout_set_name(sd->shout_conn, name) != SHOUTERR_SUCCESS ||
-	    shout_set_user(sd->shout_conn, user) != SHOUTERR_SUCCESS ||
-	    shout_set_public(sd->shout_conn, public) != SHOUTERR_SUCCESS ||
-	    shout_set_format(sd->shout_conn, shout_format)
+	if (shout_set_host(shout_conn, host) != SHOUTERR_SUCCESS ||
+	    shout_set_port(shout_conn, port) != SHOUTERR_SUCCESS ||
+	    shout_set_password(shout_conn, passwd) != SHOUTERR_SUCCESS ||
+	    shout_set_mount(shout_conn, mount) != SHOUTERR_SUCCESS ||
+	    shout_set_name(shout_conn, name) != SHOUTERR_SUCCESS ||
+	    shout_set_user(shout_conn, user) != SHOUTERR_SUCCESS ||
+	    shout_set_public(shout_conn, is_public) != SHOUTERR_SUCCESS ||
+	    shout_set_format(shout_conn, shout_format)
 	    != SHOUTERR_SUCCESS ||
-	    shout_set_protocol(sd->shout_conn, protocol) != SHOUTERR_SUCCESS ||
-	    shout_set_agent(sd->shout_conn, "MPD") != SHOUTERR_SUCCESS) {
-		g_set_error(error, shout_output_quark(), 0,
-			    "%s", shout_get_error(sd->shout_conn));
-		goto failure;
+	    shout_set_protocol(shout_conn, protocol) != SHOUTERR_SUCCESS ||
+	    shout_set_agent(shout_conn, "MPD") != SHOUTERR_SUCCESS) {
+		g_set_error(error_r, shout_output_quark(), 0,
+			    "%s", shout_get_error(shout_conn));
+		return false;
 	}
 
 	/* optional paramters */
-	sd->timeout = config_get_block_unsigned(param, "timeout",
-						DEFAULT_CONN_TIMEOUT);
+	timeout = param.GetBlockValue("timeout", DEFAULT_CONN_TIMEOUT);
 
-	value = config_get_block_string(param, "genre", NULL);
-	if (value != NULL && shout_set_genre(sd->shout_conn, value)) {
-		g_set_error(error, shout_output_quark(), 0,
-			    "%s", shout_get_error(sd->shout_conn));
-		goto failure;
+	value = param.GetBlockValue("genre");
+	if (value != nullptr && shout_set_genre(shout_conn, value)) {
+		g_set_error(error_r, shout_output_quark(), 0,
+			    "%s", shout_get_error(shout_conn));
+		return false;
 	}
 
-	value = config_get_block_string(param, "description", NULL);
-	if (value != NULL && shout_set_description(sd->shout_conn, value)) {
-		g_set_error(error, shout_output_quark(), 0,
-			    "%s", shout_get_error(sd->shout_conn));
-		goto failure;
+	value = param.GetBlockValue("description");
+	if (value != nullptr && shout_set_description(shout_conn, value)) {
+		g_set_error(error_r, shout_output_quark(), 0,
+			    "%s", shout_get_error(shout_conn));
+		return false;
 	}
 
-	value = config_get_block_string(param, "url", NULL);
-	if (value != NULL && shout_set_url(sd->shout_conn, value)) {
-		g_set_error(error, shout_output_quark(), 0,
-			    "%s", shout_get_error(sd->shout_conn));
-		goto failure;
+	value = param.GetBlockValue("url");
+	if (value != nullptr && shout_set_url(shout_conn, value)) {
+		g_set_error(error_r, shout_output_quark(), 0,
+			    "%s", shout_get_error(shout_conn));
+		return false;
 	}
 
 	{
 		char temp[11];
 		memset(temp, 0, sizeof(temp));
 
-		snprintf(temp, sizeof(temp), "%u", audio_format->channels);
-		shout_set_audio_info(sd->shout_conn, SHOUT_AI_CHANNELS, temp);
+		snprintf(temp, sizeof(temp), "%u", audio_format.channels);
+		shout_set_audio_info(shout_conn, SHOUT_AI_CHANNELS, temp);
 
-		snprintf(temp, sizeof(temp), "%u", audio_format->sample_rate);
+		snprintf(temp, sizeof(temp), "%u", audio_format.sample_rate);
 
-		shout_set_audio_info(sd->shout_conn, SHOUT_AI_SAMPLERATE, temp);
+		shout_set_audio_info(shout_conn, SHOUT_AI_SAMPLERATE, temp);
 
-		if (sd->quality >= -1.0) {
-			snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
-			shout_set_audio_info(sd->shout_conn, SHOUT_AI_QUALITY,
+		if (quality >= -1.0) {
+			snprintf(temp, sizeof(temp), "%2.2f", quality);
+			shout_set_audio_info(shout_conn, SHOUT_AI_QUALITY,
 					     temp);
 		} else {
-			snprintf(temp, sizeof(temp), "%d", sd->bitrate);
-			shout_set_audio_info(sd->shout_conn, SHOUT_AI_BITRATE,
+			snprintf(temp, sizeof(temp), "%d", bitrate);
+			shout_set_audio_info(shout_conn, SHOUT_AI_BITRATE,
 					     temp);
 		}
 	}
 
-	return &sd->base;
+	return true;
+}
+
+static struct audio_output *
+my_shout_init_driver(const config_param &param, GError **error_r)
+{
+	ShoutOutput *sd = new ShoutOutput();
+	if (!sd->Initialize(param, error_r)) {
+		delete sd;
+		return nullptr;
+	}
 
-failure:
-	ao_base_finish(&sd->base);
-	free_shout_data(sd);
-	return NULL;
+	if (!sd->Configure(param, error_r)) {
+		sd->Deinitialize();
+		delete sd;
+		return nullptr;
+	}
+
+	if (shout_init_count == 0)
+		shout_init();
+
+	shout_init_count++;
+
+	return &sd->base;
 }
 
 static bool
-handle_shout_error(struct shout_data *sd, int err, GError **error)
+handle_shout_error(ShoutOutput *sd, int err, GError **error)
 {
 	switch (err) {
 	case SHOUTERR_SUCCESS:
@@ -338,9 +338,9 @@ handle_shout_error(struct shout_data *sd, int err, GError **error)
 }
 
 static bool
-write_page(struct shout_data *sd, GError **error)
+write_page(ShoutOutput *sd, GError **error)
 {
-	assert(sd->encoder != NULL);
+	assert(sd->encoder != nullptr);
 
 	while (true) {
 		size_t nbytes = encoder_read(sd->encoder,
@@ -356,11 +356,11 @@ write_page(struct shout_data *sd, GError **error)
 	return true;
 }
 
-static void close_shout_conn(struct shout_data * sd)
+static void close_shout_conn(ShoutOutput * sd)
 {
-	if (sd->encoder != NULL) {
-		if (encoder_end(sd->encoder, NULL))
-			write_page(sd, NULL);
+	if (sd->encoder != nullptr) {
+		if (encoder_end(sd->encoder, nullptr))
+			write_page(sd, nullptr);
 
 		encoder_close(sd->encoder);
 	}
@@ -375,12 +375,12 @@ static void close_shout_conn(struct shout_data * sd)
 static void
 my_shout_finish_driver(struct audio_output *ao)
 {
-	struct shout_data *sd = (struct shout_data *)ao;
+	ShoutOutput *sd = (ShoutOutput *)ao;
 
 	encoder_finish(sd->encoder);
 
-	ao_base_finish(&sd->base);
-	free_shout_data(sd);
+	sd->Deinitialize();
+	delete sd;
 
 	shout_init_count--;
 
@@ -392,7 +392,7 @@ static void
 my_shout_drop_buffered_audio(struct audio_output *ao)
 {
 	G_GNUC_UNUSED
-	struct shout_data *sd = (struct shout_data *)ao;
+	ShoutOutput *sd = (ShoutOutput *)ao;
 
 	/* needs to be implemented for shout */
 }
@@ -400,13 +400,13 @@ my_shout_drop_buffered_audio(struct audio_output *ao)
 static void
 my_shout_close_device(struct audio_output *ao)
 {
-	struct shout_data *sd = (struct shout_data *)ao;
+	ShoutOutput *sd = (ShoutOutput *)ao;
 
 	close_shout_conn(sd);
 }
 
 static bool
-shout_connect(struct shout_data *sd, GError **error)
+shout_connect(ShoutOutput *sd, GError **error)
 {
 	switch (shout_open(sd->shout_conn)) {
 	case SHOUTERR_SUCCESS:
@@ -424,10 +424,10 @@ shout_connect(struct shout_data *sd, GError **error)
 }
 
 static bool
-my_shout_open_device(struct audio_output *ao, struct audio_format *audio_format,
+my_shout_open_device(struct audio_output *ao, AudioFormat &audio_format,
 		     GError **error)
 {
-	struct shout_data *sd = (struct shout_data *)ao;
+	ShoutOutput *sd = (ShoutOutput *)ao;
 
 	if (!shout_connect(sd, error))
 		return false;
@@ -449,7 +449,7 @@ my_shout_open_device(struct audio_output *ao, struct audio_format *audio_format,
 static unsigned
 my_shout_delay(struct audio_output *ao)
 {
-	struct shout_data *sd = (struct shout_data *)ao;
+	ShoutOutput *sd = (ShoutOutput *)ao;
 
 	int delay = shout_delay(sd->shout_conn);
 	if (delay < 0)
@@ -462,7 +462,7 @@ static size_t
 my_shout_play(struct audio_output *ao, const void *chunk, size_t size,
 	      GError **error)
 {
-	struct shout_data *sd = (struct shout_data *)ao;
+	ShoutOutput *sd = (ShoutOutput *)ao;
 
 	return encoder_write(sd->encoder, chunk, size, error) &&
 		write_page(sd, error)
@@ -473,13 +473,13 @@ my_shout_play(struct audio_output *ao, const void *chunk, size_t size,
 static bool
 my_shout_pause(struct audio_output *ao)
 {
-	static const char silence[1020];
+	static char silence[1020];
 
-	return my_shout_play(ao, silence, sizeof(silence), NULL);
+	return my_shout_play(ao, silence, sizeof(silence), nullptr);
 }
 
 static void
-shout_tag_to_metadata(const struct tag *tag, char *dest, size_t size)
+shout_tag_to_metadata(const Tag *tag, char *dest, size_t size)
 {
 	char artist[size];
 	char title[size];
@@ -505,12 +505,12 @@ shout_tag_to_metadata(const struct tag *tag, char *dest, size_t size)
 }
 
 static void my_shout_set_tag(struct audio_output *ao,
-			     const struct tag *tag)
+			     const Tag *tag)
 {
-	struct shout_data *sd = (struct shout_data *)ao;
-	GError *error = NULL;
+	ShoutOutput *sd = (ShoutOutput *)ao;
+	GError *error = nullptr;
 
-	if (sd->encoder->plugin->tag != NULL) {
+	if (sd->encoder->plugin.tag != nullptr) {
 		/* encoder plugin supports stream tags */
 
 		if (!encoder_pre_tag(sd->encoder, &error)) {
@@ -519,7 +519,7 @@ static void my_shout_set_tag(struct audio_output *ao,
 			return;
 		}
 
-		if (!write_page(sd, NULL))
+		if (!write_page(sd, nullptr))
 			return;
 
 		if (!encoder_tag(sd->encoder, tag, &error)) {
@@ -538,18 +538,23 @@ static void my_shout_set_tag(struct audio_output *ao,
 		}
 	}
 
-	write_page(sd, NULL);
+	write_page(sd, nullptr);
 }
 
 const struct audio_output_plugin shout_output_plugin = {
-	.name = "shout",
-	.init = my_shout_init_driver,
-	.finish = my_shout_finish_driver,
-	.open = my_shout_open_device,
-	.delay = my_shout_delay,
-	.play = my_shout_play,
-	.pause = my_shout_pause,
-	.cancel = my_shout_drop_buffered_audio,
-	.close = my_shout_close_device,
-	.send_tag = my_shout_set_tag,
+	"shout",
+	nullptr,
+	my_shout_init_driver,
+	my_shout_finish_driver,
+	nullptr,
+	nullptr,
+	my_shout_open_device,
+	my_shout_close_device,
+	my_shout_delay,
+	my_shout_set_tag,
+	my_shout_play,
+	nullptr,
+	my_shout_drop_buffered_audio,
+	my_shout_pause,
+	nullptr,
 };
diff --git a/src/output/shout_output_plugin.h b/src/output/ShoutOutputPlugin.hxx
index 9a7378803..496b77975 100644
--- a/src/output/shout_output_plugin.h
+++ b/src/output/ShoutOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_SHOUT_OUTPUT_PLUGIN_H
-#define MPD_SHOUT_OUTPUT_PLUGIN_H
+#ifndef MPD_SHOUT_OUTPUT_PLUGIN_HXX
+#define MPD_SHOUT_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin shout_output_plugin;
 
diff --git a/src/output/solaris_output_plugin.c b/src/output/SolarisOutputPlugin.cxx
index ce726009a..074eae728 100644
--- a/src/output/solaris_output_plugin.c
+++ b/src/output/SolarisOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,8 +18,8 @@
  */
 
 #include "config.h"
-#include "solaris_output_plugin.h"
-#include "output_api.h"
+#include "SolarisOutputPlugin.hxx"
+#include "OutputAPI.hxx"
 #include "fd_util.h"
 
 #include <glib.h>
@@ -53,13 +53,22 @@ struct audio_info {
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "solaris_output"
 
-struct solaris_output {
+struct SolarisOutput {
 	struct audio_output base;
 
 	/* configuration */
 	const char *device;
 
 	int fd;
+
+	bool Initialize(const config_param &param, GError **error_r) {
+		return ao_base_init(&base, &solaris_output_plugin, param,
+				    error_r);
+	}
+
+	void Deinitialize() {
+		ao_base_finish(&base);
+	}
 };
 
 /**
@@ -81,16 +90,15 @@ solaris_output_test_default_device(void)
 }
 
 static struct audio_output *
-solaris_output_init(const struct config_param *param, GError **error_r)
+solaris_output_init(const config_param &param, GError **error_r)
 {
-	struct solaris_output *so = g_new(struct solaris_output, 1);
-
-	if (!ao_base_init(&so->base, &solaris_output_plugin, param, error_r)) {
-		g_free(so);
-		return NULL;
+	SolarisOutput *so = new SolarisOutput();
+	if (!so->Initialize(param, error_r)) {
+		delete so;
+		return nullptr;
 	}
 
-	so->device = config_get_block_string(param, "device", "/dev/audio");
+	so->device = param.GetBlockValue("device", "/dev/audio");
 
 	return &so->base;
 }
@@ -98,23 +106,23 @@ solaris_output_init(const struct config_param *param, GError **error_r)
 static void
 solaris_output_finish(struct audio_output *ao)
 {
-	struct solaris_output *so = (struct solaris_output *)ao;
+	SolarisOutput *so = (SolarisOutput *)ao;
 
-	ao_base_finish(&so->base);
-	g_free(so);
+	so->Deinitialize();
+	delete so;
 }
 
 static bool
-solaris_output_open(struct audio_output *ao, struct audio_format *audio_format,
+solaris_output_open(struct audio_output *ao, AudioFormat &audio_format,
 		    GError **error)
 {
-	struct solaris_output *so = (struct solaris_output *)ao;
+	SolarisOutput *so = (SolarisOutput *)ao;
 	struct audio_info info;
 	int ret, flags;
 
 	/* support only 16 bit mono/stereo for now; nothing else has
 	   been tested */
-	audio_format->format = SAMPLE_FORMAT_S16;
+	audio_format.format = SampleFormat::S16;
 
 	/* open the device in non-blocking mode */
 
@@ -142,8 +150,8 @@ solaris_output_open(struct audio_output *ao, struct audio_format *audio_format,
 		return false;
 	}
 
-	info.play.sample_rate = audio_format->sample_rate;
-	info.play.channels = audio_format->channels;
+	info.play.sample_rate = audio_format.sample_rate;
+	info.play.channels = audio_format.channels;
 	info.play.precision = 16;
 	info.play.encoding = AUDIO_ENCODING_LINEAR;
 
@@ -161,7 +169,7 @@ solaris_output_open(struct audio_output *ao, struct audio_format *audio_format,
 static void
 solaris_output_close(struct audio_output *ao)
 {
-	struct solaris_output *so = (struct solaris_output *)ao;
+	SolarisOutput *so = (SolarisOutput *)ao;
 
 	close(so->fd);
 }
@@ -170,7 +178,7 @@ static size_t
 solaris_output_play(struct audio_output *ao, const void *chunk, size_t size,
 		    GError **error)
 {
-	struct solaris_output *so = (struct solaris_output *)ao;
+	SolarisOutput *so = (SolarisOutput *)ao;
 	ssize_t nbytes;
 
 	nbytes = write(so->fd, chunk, size);
@@ -186,18 +194,25 @@ solaris_output_play(struct audio_output *ao, const void *chunk, size_t size,
 static void
 solaris_output_cancel(struct audio_output *ao)
 {
-	struct solaris_output *so = (struct solaris_output *)ao;
+	SolarisOutput *so = (SolarisOutput *)ao;
 
 	ioctl(so->fd, I_FLUSH);
 }
 
 const struct audio_output_plugin solaris_output_plugin = {
-	.name = "solaris",
-	.test_default_device = solaris_output_test_default_device,
-	.init = solaris_output_init,
-	.finish = solaris_output_finish,
-	.open = solaris_output_open,
-	.close = solaris_output_close,
-	.play = solaris_output_play,
-	.cancel = solaris_output_cancel,
+	"solaris",
+	solaris_output_test_default_device,
+	solaris_output_init,
+	solaris_output_finish,
+	nullptr,
+	nullptr,
+	solaris_output_open,
+	solaris_output_close,
+	nullptr,
+	nullptr,
+	solaris_output_play,
+	nullptr,
+	solaris_output_cancel,
+	nullptr,
+	nullptr,
 };
diff --git a/src/output/solaris_output_plugin.h b/src/output/SolarisOutputPlugin.hxx
index 600aea8c2..d0fbd32c8 100644
--- a/src/output/solaris_output_plugin.h
+++ b/src/output/SolarisOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,8 +17,8 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_SOLARIS_OUTPUT_PLUGIN_H
-#define MPD_SOLARIS_OUTPUT_PLUGIN_H
+#ifndef MPD_SOLARIS_OUTPUT_PLUGIN_HXX
+#define MPD_SOLARIS_OUTPUT_PLUGIN_HXX
 
 extern const struct audio_output_plugin solaris_output_plugin;
 
diff --git a/src/output/winmm_output_plugin.c b/src/output/WinmmOutputPlugin.cxx
index 4d95834b9..d02b52c58 100644
--- a/src/output/winmm_output_plugin.c
+++ b/src/output/WinmmOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -18,26 +18,24 @@
  */
 
 #include "config.h"
-#include "winmm_output_plugin.h"
-#include "output_api.h"
-#include "pcm_buffer.h"
-#include "mixer_list.h"
-#include "winmm_output_plugin.h"
+#include "WinmmOutputPlugin.hxx"
+#include "OutputAPI.hxx"
+#include "pcm/PcmBuffer.hxx"
+#include "MixerList.hxx"
 
 #include <stdlib.h>
 #include <string.h>
-#include <windows.h>
 
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "winmm_output"
 
-struct winmm_buffer {
-	struct pcm_buffer buffer;
+struct WinmmBuffer {
+	PcmBuffer buffer;
 
 	WAVEHDR hdr;
 };
 
-struct winmm_output {
+struct WinmmOutput {
 	struct audio_output base;
 
 	UINT device_id;
@@ -49,7 +47,7 @@ struct winmm_output {
 	 */
 	HANDLE event;
 
-	struct winmm_buffer buffers[8];
+	WinmmBuffer buffers[8];
 	unsigned next_buffer;
 };
 
@@ -63,7 +61,7 @@ winmm_output_quark(void)
 }
 
 HWAVEOUT
-winmm_output_get_handle(struct winmm_output* output)
+winmm_output_get_handle(WinmmOutput *output)
 {
 	return output->handle;
 }
@@ -78,7 +76,7 @@ static bool
 get_device_id(const char *device_name, UINT *device_id, GError **error_r)
 {
 	/* if device is not specified use wave mapper */
-	if (device_name == NULL) {
+	if (device_name == nullptr) {
 		*device_id = WAVE_MAPPER;
 		return true;
 	}
@@ -116,19 +114,19 @@ fail:
 }
 
 static struct audio_output *
-winmm_output_init(const struct config_param *param, GError **error_r)
+winmm_output_init(const config_param &param, GError **error_r)
 {
-	struct winmm_output *wo = g_new(struct winmm_output, 1);
+	WinmmOutput *wo = new WinmmOutput();
 	if (!ao_base_init(&wo->base, &winmm_output_plugin, param, error_r)) {
 		g_free(wo);
-		return NULL;
+		return nullptr;
 	}
 
-	const char *device = config_get_block_string(param, "device", NULL);
+	const char *device = param.GetBlockValue("device");
 	if (!get_device_id(device, &wo->device_id, error_r)) {
 		ao_base_finish(&wo->base);
 		g_free(wo);
-		return NULL;
+		return nullptr;
 	}
 
 	return &wo->base;
@@ -137,49 +135,51 @@ winmm_output_init(const struct config_param *param, GError **error_r)
 static void
 winmm_output_finish(struct audio_output *ao)
 {
-	struct winmm_output *wo = (struct winmm_output *)ao;
+	WinmmOutput *wo = (WinmmOutput *)ao;
 
 	ao_base_finish(&wo->base);
-	g_free(wo);
+	delete wo;
 }
 
 static bool
-winmm_output_open(struct audio_output *ao, struct audio_format *audio_format,
+winmm_output_open(struct audio_output *ao, AudioFormat &audio_format,
 		  GError **error_r)
 {
-	struct winmm_output *wo = (struct winmm_output *)ao;
+	WinmmOutput *wo = (WinmmOutput *)ao;
 
-	wo->event = CreateEvent(NULL, false, false, NULL);
-	if (wo->event == NULL) {
+	wo->event = CreateEvent(nullptr, false, false, nullptr);
+	if (wo->event == nullptr) {
 		g_set_error(error_r, winmm_output_quark(), 0,
 			    "CreateEvent() failed");
 		return false;
 	}
 
-	switch (audio_format->format) {
-	case SAMPLE_FORMAT_S8:
-	case SAMPLE_FORMAT_S16:
+	switch (audio_format.format) {
+	case SampleFormat::S8:
+	case SampleFormat::S16:
 		break;
 
-	case SAMPLE_FORMAT_S24_P32:
-	case SAMPLE_FORMAT_S32:
-	case SAMPLE_FORMAT_UNDEFINED:
+	case SampleFormat::S24_P32:
+	case SampleFormat::S32:
+	case SampleFormat::FLOAT:
+	case SampleFormat::DSD:
+	case SampleFormat::UNDEFINED:
 		/* we havn't tested formats other than S16 */
-		audio_format->format = SAMPLE_FORMAT_S16;
+		audio_format.format = SampleFormat::S16;
 		break;
 	}
 
-	if (audio_format->channels > 2)
+	if (audio_format.channels > 2)
 		/* same here: more than stereo was not tested */
-		audio_format->channels = 2;
+		audio_format.channels = 2;
 
 	WAVEFORMATEX format;
 	format.wFormatTag = WAVE_FORMAT_PCM;
-	format.nChannels = audio_format->channels;
-	format.nSamplesPerSec = audio_format->sample_rate;
-	format.nBlockAlign = audio_format_frame_size(audio_format);
+	format.nChannels = audio_format.channels;
+	format.nSamplesPerSec = audio_format.sample_rate;
+	format.nBlockAlign = audio_format.GetFrameSize();
 	format.nAvgBytesPerSec = format.nSamplesPerSec * format.nBlockAlign;
-	format.wBitsPerSample = audio_format_sample_size(audio_format) * 8;
+	format.wBitsPerSample = audio_format.GetSampleSize() * 8;
 	format.cbSize = 0;
 
 	MMRESULT result = waveOutOpen(&wo->handle, wo->device_id, &format,
@@ -192,7 +192,6 @@ winmm_output_open(struct audio_output *ao, struct audio_format *audio_format,
 	}
 
 	for (unsigned i = 0; i < G_N_ELEMENTS(wo->buffers); ++i) {
-		pcm_buffer_init(&wo->buffers[i].buffer);
 		memset(&wo->buffers[i].hdr, 0, sizeof(wo->buffers[i].hdr));
 	}
 
@@ -204,10 +203,10 @@ winmm_output_open(struct audio_output *ao, struct audio_format *audio_format,
 static void
 winmm_output_close(struct audio_output *ao)
 {
-	struct winmm_output *wo = (struct winmm_output *)ao;
+	WinmmOutput *wo = (WinmmOutput *)ao;
 
 	for (unsigned i = 0; i < G_N_ELEMENTS(wo->buffers); ++i)
-		pcm_buffer_deinit(&wo->buffers[i].buffer);
+		wo->buffers[i].buffer.Clear();
 
 	waveOutClose(wo->handle);
 
@@ -218,17 +217,17 @@ winmm_output_close(struct audio_output *ao)
  * Copy data into a buffer, and prepare the wave header.
  */
 static bool
-winmm_set_buffer(struct winmm_output *wo, struct winmm_buffer *buffer,
+winmm_set_buffer(WinmmOutput *wo, WinmmBuffer *buffer,
 		 const void *data, size_t size,
 		 GError **error_r)
 {
-	void *dest = pcm_buffer_get(&buffer->buffer, size);
-	assert(dest != NULL);
+	void *dest = buffer->buffer.Get(size);
+	assert(dest != nullptr);
 
 	memcpy(dest, data, size);
 
 	memset(&buffer->hdr, 0, sizeof(buffer->hdr));
-	buffer->hdr.lpData = dest;
+	buffer->hdr.lpData = (LPSTR)dest;
 	buffer->hdr.dwBufferLength = size;
 
 	MMRESULT result = waveOutPrepareHeader(wo->handle, &buffer->hdr,
@@ -246,7 +245,7 @@ winmm_set_buffer(struct winmm_output *wo, struct winmm_buffer *buffer,
  * Wait until the buffer is finished.
  */
 static bool
-winmm_drain_buffer(struct winmm_output *wo, struct winmm_buffer *buffer,
+winmm_drain_buffer(WinmmOutput *wo, WinmmBuffer *buffer,
 		   GError **error_r)
 {
 	if ((buffer->hdr.dwFlags & WHDR_DONE) == WHDR_DONE)
@@ -273,10 +272,10 @@ winmm_drain_buffer(struct winmm_output *wo, struct winmm_buffer *buffer,
 static size_t
 winmm_output_play(struct audio_output *ao, const void *chunk, size_t size, GError **error_r)
 {
-	struct winmm_output *wo = (struct winmm_output *)ao;
+	WinmmOutput *wo = (WinmmOutput *)ao;
 
 	/* get the next buffer from the ring and prepare it */
-	struct winmm_buffer *buffer = &wo->buffers[wo->next_buffer];
+	WinmmBuffer *buffer = &wo->buffers[wo->next_buffer];
 	if (!winmm_drain_buffer(wo, buffer, error_r) ||
 	    !winmm_set_buffer(wo, buffer, chunk, size, error_r))
 		return 0;
@@ -300,7 +299,7 @@ winmm_output_play(struct audio_output *ao, const void *chunk, size_t size, GErro
 }
 
 static bool
-winmm_drain_all_buffers(struct winmm_output *wo, GError **error_r)
+winmm_drain_all_buffers(WinmmOutput *wo, GError **error_r)
 {
 	for (unsigned i = wo->next_buffer; i < G_N_ELEMENTS(wo->buffers); ++i)
 		if (!winmm_drain_buffer(wo, &wo->buffers[i], error_r))
@@ -314,12 +313,12 @@ winmm_drain_all_buffers(struct winmm_output *wo, GError **error_r)
 }
 
 static void
-winmm_stop(struct winmm_output *wo)
+winmm_stop(WinmmOutput *wo)
 {
 	waveOutReset(wo->handle);
 
 	for (unsigned i = 0; i < G_N_ELEMENTS(wo->buffers); ++i) {
-		struct winmm_buffer *buffer = &wo->buffers[i];
+		WinmmBuffer *buffer = &wo->buffers[i];
 		waveOutUnprepareHeader(wo->handle, &buffer->hdr,
 				       sizeof(buffer->hdr));
 	}
@@ -328,29 +327,34 @@ winmm_stop(struct winmm_output *wo)
 static void
 winmm_output_drain(struct audio_output *ao)
 {
-	struct winmm_output *wo = (struct winmm_output *)ao;
+	WinmmOutput *wo = (WinmmOutput *)ao;
 
-	if (!winmm_drain_all_buffers(wo, NULL))
+	if (!winmm_drain_all_buffers(wo, nullptr))
 		winmm_stop(wo);
 }
 
 static void
 winmm_output_cancel(struct audio_output *ao)
 {
-	struct winmm_output *wo = (struct winmm_output *)ao;
+	WinmmOutput *wo = (WinmmOutput *)ao;
 
 	winmm_stop(wo);
 }
 
 const struct audio_output_plugin winmm_output_plugin = {
-	.name = "winmm",
-	.test_default_device = winmm_output_test_default_device,
-	.init = winmm_output_init,
-	.finish = winmm_output_finish,
-	.open = winmm_output_open,
-	.close = winmm_output_close,
-	.play = winmm_output_play,
-	.drain = winmm_output_drain,
-	.cancel = winmm_output_cancel,
-	.mixer_plugin = &winmm_mixer_plugin,
+	"winmm",
+	winmm_output_test_default_device,
+	winmm_output_init,
+	winmm_output_finish,
+	nullptr,
+	nullptr,
+	winmm_output_open,
+	winmm_output_close,
+	nullptr,
+	nullptr,
+	winmm_output_play,
+	winmm_output_drain,
+	winmm_output_cancel,
+	nullptr,
+	&winmm_mixer_plugin,
 };
diff --git a/src/output/winmm_output_plugin.h b/src/output/WinmmOutputPlugin.hxx
index 0605530e1..e8688782e 100644
--- a/src/output/winmm_output_plugin.h
+++ b/src/output/WinmmOutputPlugin.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2011 The Music Player Daemon Project
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -17,20 +17,25 @@
  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  */
 
-#ifndef MPD_WINMM_OUTPUT_PLUGIN_H
-#define MPD_WINMM_OUTPUT_PLUGIN_H
+#ifndef MPD_WINMM_OUTPUT_PLUGIN_HXX
+#define MPD_WINMM_OUTPUT_PLUGIN_HXX
 
 #include "check.h"
 
 #ifdef ENABLE_WINMM_OUTPUT
 
+#include "gcc.h"
+
 #include <windows.h>
+#include <mmsystem.h>
 
-struct winmm_output;
+struct WinmmOutput;
 
 extern const struct audio_output_plugin winmm_output_plugin;
 
-HWAVEOUT winmm_output_get_handle(struct winmm_output*);
+gcc_pure
+HWAVEOUT
+winmm_output_get_handle(WinmmOutput *);
 
 #endif
 
diff --git a/src/output/ffado_output_plugin.c b/src/output/ffado_output_plugin.c
deleted file mode 100644
index ba239a4ad..000000000
--- a/src/output/ffado_output_plugin.c
+++ /dev/null
@@ -1,359 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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.
- */
-
-/*
- * Warning: this plugin was not tested successfully.  I just couldn't
- * keep libffado2 from crashing.  Use at your own risk.
- *
- * For details, see my Debian bug reports:
- *
- *   http://bugs.debian.org/601657
- *   http://bugs.debian.org/601659
- *   http://bugs.debian.org/601663
- *
- */
-
-#include "config.h"
-#include "ffado_output_plugin.h"
-#include "output_api.h"
-#include "timer.h"
-
-#include <glib.h>
-#include <assert.h>
-
-#include <libffado/ffado.h>
-
-#undef G_LOG_DOMAIN
-#define G_LOG_DOMAIN "ffado"
-
-enum {
-	MAX_STREAMS = 8,
-};
-
-struct mpd_ffado_stream {
-	/** libffado's stream number */
-	int number;
-
-	float *buffer;
-};
-
-struct mpd_ffado_device {
-	struct audio_output base;
-
-	char *device_name;
-	int verbose;
-	unsigned period_size, nb_buffers;
-
-	ffado_device_t *dev;
-
-	/**
-	 * The current sample position inside the stream buffers.  New
-	 * samples get appended at this position on all streams at the
-	 * same time.  When the buffers are full
-	 * (buffer_position==period_size),
-	 * ffado_streaming_transfer_playback_buffers() gets called to
-	 * hand them over to libffado.
-	 */
-	unsigned buffer_position;
-
-	/**
-	 * The number of streams which are really used by MPD.
-	 */
-	int num_streams;
-	struct mpd_ffado_stream streams[MAX_STREAMS];
-};
-
-static inline GQuark
-ffado_output_quark(void)
-{
-	return g_quark_from_static_string("ffado_output");
-}
-
-static struct audio_output *
-ffado_init(const struct config_param *param,
-	   GError **error_r)
-{
-	g_debug("using libffado version %s, API=%d",
-		ffado_get_version(), ffado_get_api_version());
-
-	struct mpd_ffado_device *fd = g_new(struct mpd_ffado_device, 1);
-	if (!ao_base_init(&fd->base, &ffado_output_plugin, param, error_r)) {
-		g_free(fd);
-		return NULL;
-	}
-
-	fd->device_name = config_dup_block_string(param, "device", NULL);
-	fd->verbose = config_get_block_unsigned(param, "verbose", 0);
-
-	fd->period_size = config_get_block_unsigned(param, "period_size",
-						    1024);
-	if (fd->period_size == 0 || fd->period_size > 1024 * 1024) {
-		ao_base_finish(&fd->base);
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "invalid period_size setting");
-		return false;
-	}
-
-	fd->nb_buffers = config_get_block_unsigned(param, "nb_buffers", 3);
-	if (fd->nb_buffers == 0 || fd->nb_buffers > 1024) {
-		ao_base_finish(&fd->base);
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "invalid nb_buffers setting");
-		return false;
-	}
-
-	return &fd->base;
-}
-
-static void
-ffado_finish(struct audio_output *ao)
-{
-	struct mpd_ffado_device *fd = (struct mpd_ffado_device *)ao;
-
-	g_free(fd->device_name);
-	ao_base_finish(&fd->base);
-	g_free(fd);
-}
-
-static bool
-ffado_configure_stream(ffado_device_t *dev, struct mpd_ffado_stream *stream,
-		       GError **error_r)
-{
-	char *buffer = (char *)stream->buffer;
-	if (ffado_streaming_set_playback_stream_buffer(dev, stream->number,
-						       buffer) != 0) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "failed to configure stream buffer");
-		return false;
-	}
-
-	if (ffado_streaming_playback_stream_onoff(dev, stream->number,
-						  1) != 0) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "failed to disable stream");
-		return false;
-	}
-
-	return true;
-}
-
-static bool
-ffado_configure(struct mpd_ffado_device *fd, struct audio_format *audio_format,
-		GError **error_r)
-{
-	assert(fd != NULL);
-	assert(fd->dev != NULL);
-	assert(audio_format->channels <= MAX_STREAMS);
-
-	if (ffado_streaming_set_audio_datatype(fd->dev,
-					       ffado_audio_datatype_float) != 0) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "ffado_streaming_set_audio_datatype() failed");
-		return false;
-	}
-
-	int num_streams = ffado_streaming_get_nb_playback_streams(fd->dev);
-	if (num_streams < 0) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "ffado_streaming_get_nb_playback_streams() failed");
-		return false;
-	}
-
-	g_debug("there are %d playback streams", num_streams);
-
-	fd->num_streams = 0;
-	for (int i = 0; i < num_streams; ++i) {
-		char name[256];
-		ffado_streaming_get_playback_stream_name(fd->dev, i, name,
-							 sizeof(name) - 1);
-
-		ffado_streaming_stream_type type =
-			ffado_streaming_get_playback_stream_type(fd->dev, i);
-		if (type != ffado_stream_type_audio) {
-			g_debug("stream %d name='%s': not an audio stream",
-				i, name);
-			continue;
-		}
-
-		if (fd->num_streams >= audio_format->channels) {
-			g_debug("stream %d name='%s': ignoring",
-				i, name);
-			continue;
-		}
-
-		g_debug("stream %d name='%s'", i, name);
-
-		struct mpd_ffado_stream *stream =
-			&fd->streams[fd->num_streams++];
-
-		stream->number = i;
-
-		/* allocated buffer is zeroed = silence */
-		stream->buffer = g_new0(float, fd->period_size);
-
-		if (!ffado_configure_stream(fd->dev, stream, error_r))
-			return false;
-	}
-
-	if (!audio_valid_channel_count(fd->num_streams)) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "invalid channel count from libffado: %u",
-			    audio_format->channels);
-		return false;
-	}
-
-	g_debug("configured %d audio streams", fd->num_streams);
-
-	if (ffado_streaming_prepare(fd->dev) != 0) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "ffado_streaming_prepare() failed");
-		return false;
-	}
-
-	if (ffado_streaming_start(fd->dev) != 0) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "ffado_streaming_start() failed");
-		return false;
-	}
-
-	audio_format->channels = fd->num_streams;
-	return true;
-}
-
-static bool
-ffado_open(struct audio_output *ao, struct audio_format *audio_format,
-	   GError **error_r)
-{
-	struct mpd_ffado_device *fd = (struct mpd_ffado_device *)ao;
-
-	/* will be converted to floating point, choose best input
-	   format */
-	audio_format->format = SAMPLE_FORMAT_S24_P32;
-
-	ffado_device_info_t device_info;
-	memset(&device_info, 0, sizeof(device_info));
-	if (fd->device_name != NULL) {
-		device_info.nb_device_spec_strings = 1;
-		device_info.device_spec_strings = &fd->device_name;
-	}
-
-	ffado_options_t options;
-	memset(&options, 0, sizeof(options));
-	options.sample_rate = audio_format->sample_rate;
-	options.period_size = fd->period_size;
-	options.nb_buffers = fd->nb_buffers;
-	options.verbose = fd->verbose;
-
-	fd->dev = ffado_streaming_init(device_info, options);
-	if (fd->dev == NULL) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "ffado_streaming_init() failed");
-		return false;
-	}
-
-	if (!ffado_configure(fd, audio_format, error_r)) {
-		ffado_streaming_finish(fd->dev);
-
-		for (int i = 0; i < fd->num_streams; ++i) {
-			struct mpd_ffado_stream *stream = &fd->streams[i];
-			g_free(stream->buffer);
-		}
-
-		return false;
-	}
-
-	fd->buffer_position = 0;
-
-	return true;
-}
-
-static void
-ffado_close(struct audio_output *ao)
-{
-	struct mpd_ffado_device *fd = (struct mpd_ffado_device *)ao;
-
-	ffado_streaming_stop(fd->dev);
-	ffado_streaming_finish(fd->dev);
-
-	for (int i = 0; i < fd->num_streams; ++i) {
-		struct mpd_ffado_stream *stream = &fd->streams[i];
-		g_free(stream->buffer);
-	}
-}
-
-static size_t
-ffado_play(struct audio_output *ao, const void *chunk, size_t size,
-	   GError **error_r)
-{
-	struct mpd_ffado_device *fd = (struct mpd_ffado_device *)ao;
-
-	/* wait for prefious buffer to finish (if it was full) */
-
-	if (fd->buffer_position >= fd->period_size) {
-		switch (ffado_streaming_wait(fd->dev)) {
-		case ffado_wait_ok:
-		case ffado_wait_xrun:
-			break;
-
-		default:
-			g_set_error(error_r, ffado_output_quark(), 0,
-				    "ffado_streaming_wait() failed");
-			return 0;
-		}
-
-		fd->buffer_position = 0;
-	}
-
-	/* copy samples to stream buffers, non-interleaved */
-
-	const int32_t *p = chunk;
-	unsigned num_frames = size / sizeof(*p) / fd->num_streams;
-	if (num_frames > fd->period_size - fd->buffer_position)
-		num_frames = fd->period_size - fd->buffer_position;
-
-	for (unsigned i = num_frames; i > 0; --i) {
-		for (int stream = 0; stream < fd->num_streams; ++stream)
-			fd->streams[stream].buffer[fd->buffer_position] =
-				*p++ / (float)(1 << 23);
-		++fd->buffer_position;
-	}
-
-	/* if buffer full, transfer to device */
-
-	if (fd->buffer_position >= fd->period_size &&
-	    /* libffado documentation says this function returns -1 on
-	       error, but that is a lie - it returns a boolean value,
-	       and "false" means error */
-	    !ffado_streaming_transfer_playback_buffers(fd->dev)) {
-		g_set_error(error_r, ffado_output_quark(), 0,
-			    "ffado_streaming_transfer_playback_buffers() failed");
-		return 0;
-	}
-
-	return num_frames * sizeof(*p) * fd->num_streams;
-}
-
-const struct audio_output_plugin ffado_output_plugin = {
-	.name = "ffado",
-	.init = ffado_init,
-	.finish = ffado_finish,
-	.open = ffado_open,
-	.close = ffado_close,
-	.play = ffado_play,
-};
diff --git a/src/output/ffado_output_plugin.h b/src/output/ffado_output_plugin.h
deleted file mode 100644
index 4dde01859..000000000
--- a/src/output/ffado_output_plugin.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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_FFADO_OUTPUT_PLUGIN_H
-#define MPD_FFADO_OUTPUT_PLUGIN_H
-
-extern const struct audio_output_plugin ffado_output_plugin;
-
-#endif
diff --git a/src/output/httpd_client.c b/src/output/httpd_client.c
deleted file mode 100644
index 72de90457..000000000
--- a/src/output/httpd_client.c
+++ /dev/null
@@ -1,764 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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 "httpd_client.h"
-#include "httpd_internal.h"
-#include "fifo_buffer.h"
-#include "page.h"
-#include "icy_server.h"
-#include "glib_socket.h"
-
-#include <stdbool.h>
-#include <assert.h>
-#include <string.h>
-
-#undef G_LOG_DOMAIN
-#define G_LOG_DOMAIN "httpd_output"
-
-struct httpd_client {
-	/**
-	 * The httpd output object this client is connected to.
-	 */
-	struct httpd_output *httpd;
-
-	/**
-	 * The TCP socket.
-	 */
-	GIOChannel *channel;
-
-	/**
-	 * The GLib main loop source id for reading from the socket,
-	 * and to detect errors.
-	 */
-	guint read_source_id;
-
-	/**
-	 * The GLib main loop source id for writing to the socket.  If
-	 * 0, then there is no event source currently (because there
-	 * are no queued pages).
-	 */
-	guint write_source_id;
-
-	/**
-	 * For buffered reading.  This pointer is only valid while the
-	 * HTTP request is read.
-	 */
-	struct fifo_buffer *input;
-
-	/**
-	 * The current state of the client.
-	 */
-	enum {
-		/** reading the request line */
-		REQUEST,
-
-		/** reading the request headers */
-		HEADERS,
-
-		/** sending the HTTP response */
-		RESPONSE,
-	} state;
-
-	/**
-	 * A queue of #page objects to be sent to the client.
-	 */
-	GQueue *pages;
-
-	/**
-	 * The #page which is currently being sent to the client.
-	 */
-	struct page *current_page;
-
-	/**
-	 * The amount of bytes which were already sent from
-	 * #current_page.
-	 */
-	size_t current_position;
-
-        /**
-         * If DLNA streaming was an option.
-         */
-        bool dlna_streaming_requested;
-
-	/* ICY */
-
-	/**
-	 * Do we support sending Icy-Metadata to the client?  This is
-	 * disabled if the httpd audio output uses encoder tags.
-	 */
-	bool metadata_supported;
-
-	/**
-	 * If we should sent icy metadata.
-	 */
-	bool metadata_requested;
-
-	/**
-	 * If the current metadata was already sent to the client.
-	 */
-	bool metadata_sent;
-
-	/**
-	 * The amount of streaming data between each metadata block
-	 */
-	guint metaint;
-
-	/**
-	 * The metadata as #page which is currently being sent to the client.
-	 */
-	struct page *metadata;
-
-	/*
-	 * The amount of bytes which were already sent from the metadata.
-	 */
-	size_t metadata_current_position;
-
-	/**
-	 * The amount of streaming data sent to the client
-	 * since the last icy information was sent.
-	 */
-	guint metadata_fill;
-};
-
-static void
-httpd_client_unref_page(gpointer data, G_GNUC_UNUSED gpointer user_data)
-{
-	struct page *page = data;
-
-	page_unref(page);
-}
-
-void
-httpd_client_free(struct httpd_client *client)
-{
-	assert(client != NULL);
-
-	if (client->state == RESPONSE) {
-		if (client->write_source_id != 0)
-			g_source_remove(client->write_source_id);
-
-		if (client->current_page != NULL)
-			page_unref(client->current_page);
-
-		g_queue_foreach(client->pages, httpd_client_unref_page, NULL);
-		g_queue_free(client->pages);
-	} else
-		fifo_buffer_free(client->input);
-
-	if (client->metadata)
-		page_unref (client->metadata);
-
-	g_source_remove(client->read_source_id);
-	g_io_channel_unref(client->channel);
-	g_free(client);
-}
-
-/**
- * Frees the client and removes it from the server's client list.
- */
-static void
-httpd_client_close(struct httpd_client *client)
-{
-	assert(client != NULL);
-
-	httpd_output_remove_client(client->httpd, client);
-	httpd_client_free(client);
-}
-
-/**
- * Switch the client to the "RESPONSE" state.
- */
-static void
-httpd_client_begin_response(struct httpd_client *client)
-{
-	assert(client != NULL);
-	assert(client->state != RESPONSE);
-
-	client->state = RESPONSE;
-	client->write_source_id = 0;
-	client->pages = g_queue_new();
-	client->current_page = NULL;
-
-	httpd_output_send_header(client->httpd, client);
-}
-
-/**
- * Handle a line of the HTTP request.
- */
-static bool
-httpd_client_handle_line(struct httpd_client *client, const char *line)
-{
-	assert(client->state != RESPONSE);
-
-	if (client->state == REQUEST) {
-		if (strncmp(line, "GET /", 5) != 0) {
-			/* only GET is supported */
-			g_warning("malformed request line from client");
-			return false;
-		}
-
-		line = strchr(line + 5, ' ');
-		if (line == NULL || strncmp(line + 1, "HTTP/", 5) != 0) {
-			/* HTTP/0.9 without request headers */
-			httpd_client_begin_response(client);
-			return true;
-		}
-
-		/* after the request line, request headers follow */
-		client->state = HEADERS;
-		return true;
-	} else {
-		if (*line == 0) {
-			/* empty line: request is finished */
-			httpd_client_begin_response(client);
-			return true;
-		}
-
-		if (g_ascii_strncasecmp(line, "Icy-MetaData: 1", 15) == 0) {
-			/* Send icy metadata */
-			client->metadata_requested =
-				client->metadata_supported;
-			return true;
-		}
-
-		if (g_ascii_strncasecmp(line, "transferMode.dlna.org: Streaming", 32) == 0) {
-			/* Send as dlna */
-			client->dlna_streaming_requested = true;
-			/* metadata is not supported by dlna streaming, so disable it */
-			client->metadata_supported = false;
-			client->metadata_requested = false;
-			return true;
-		}
-
-		/* expect more request headers */
-		return true;
-	}
-}
-
-/**
- * Check if a complete line of input is present in the input buffer,
- * and duplicates it.  It is removed from the input buffer.  The
- * return value has to be freed with g_free().
- */
-static char *
-httpd_client_read_line(struct httpd_client *client)
-{
-	assert(client != NULL);
-	assert(client->state != RESPONSE);
-
-	const char *p, *newline;
-	size_t length;
-	char *line;
-
-	p = fifo_buffer_read(client->input, &length);
-	if (p == NULL)
-		/* empty input buffer */
-		return NULL;
-
-	newline = memchr(p, '\n', length);
-	if (newline == NULL)
-		/* incomplete line */
-		return NULL;
-
-	line = g_strndup(p, newline - p);
-	fifo_buffer_consume(client->input, newline - p + 1);
-
-	/* remove trailing whitespace (e.g. '\r') */
-	return g_strchomp(line);
-}
-
-/**
- * Sends the status line and response headers to the client.
- */
-static bool
-httpd_client_send_response(struct httpd_client *client)
-{
-	char buffer[1024];
-	GError *error = NULL;
-	GIOStatus status;
-	gsize bytes_written;
-
-	assert(client != NULL);
-	assert(client->state == RESPONSE);
-
-	if (client->dlna_streaming_requested) {
-		g_snprintf(buffer, sizeof(buffer),
-			   "HTTP/1.1 206 OK\r\n"
-			   "Content-Type: %s\r\n"
-			   "Content-Length: 10000\r\n"
-			   "Content-RangeX: 0-1000000/1000000\r\n"
-			   "transferMode.dlna.org: Streaming\r\n"
-			   "Accept-Ranges: bytes\r\n"
-			   "Connection: close\r\n"
-			   "realTimeInfo.dlna.org: DLNA.ORG_TLAG=*\r\n"
-			   "contentFeatures.dlna.org: DLNA.ORG_OP=01;DLNA.ORG_CI=0\r\n"
-			   "\r\n",
-			   client->httpd->content_type);
-
-	} else if (client->metadata_requested) {
-		gchar *metadata_header;
-
-		metadata_header = icy_server_metadata_header(
-			client->httpd->name,
-			client->httpd->genre,
-			client->httpd->website,
-			client->httpd->content_type,
-			client->metaint);
-
-		g_strlcpy(buffer, metadata_header, sizeof(buffer));
-
-		g_free(metadata_header);
-
-       } else { /* revert to a normal HTTP request */
-		g_snprintf(buffer, sizeof(buffer),
-			   "HTTP/1.1 200 OK\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",
-			   client->httpd->content_type);
-	}
-
-	status = g_io_channel_write_chars(client->channel,
-					  buffer, strlen(buffer),
-					  &bytes_written, &error);
-
-	switch (status) {
-	case G_IO_STATUS_NORMAL:
-	case G_IO_STATUS_AGAIN:
-		return true;
-
-	case G_IO_STATUS_EOF:
-		/* client has disconnected */
-
-		httpd_client_close(client);
-		return false;
-
-	case G_IO_STATUS_ERROR:
-		/* I/O error */
-
-		g_warning("failed to write to client: %s", error->message);
-		g_error_free(error);
-
-		httpd_client_close(client);
-		return false;
-	}
-
-	/* unreachable */
-	httpd_client_close(client);
-	return false;
-}
-
-/**
- * Data has been received from the client and it is appended to the
- * input buffer.
- */
-static bool
-httpd_client_received(struct httpd_client *client)
-{
-	assert(client != NULL);
-	assert(client->state != RESPONSE);
-
-	char *line;
-	bool success;
-
-	while ((line = httpd_client_read_line(client)) != NULL) {
-		success = httpd_client_handle_line(client, line);
-		g_free(line);
-		if (!success) {
-			assert(client->state != RESPONSE);
-			return false;
-		}
-
-		if (client->state == RESPONSE) {
-			if (!fifo_buffer_is_empty(client->input)) {
-				g_warning("unexpected input from client");
-				return false;
-			}
-
-			fifo_buffer_free(client->input);
-
-			return httpd_client_send_response(client);
-		}
-	}
-
-	return true;
-}
-
-static bool
-httpd_client_read(struct httpd_client *client)
-{
-	char *p;
-	size_t max_length;
-	GError *error = NULL;
-	GIOStatus status;
-	gsize bytes_read;
-
-	if (client->state == RESPONSE) {
-		/* the client has already sent the request, and he
-		   must not send more */
-		char buffer[1];
-
-		status = g_io_channel_read_chars(client->channel, buffer,
-						 sizeof(buffer), &bytes_read,
-						 NULL);
-		if (status == G_IO_STATUS_NORMAL)
-			g_warning("unexpected input from client");
-
-		return false;
-	}
-
-	p = fifo_buffer_write(client->input, &max_length);
-	if (p == NULL) {
-		g_warning("buffer overflow");
-		return false;
-	}
-
-	status = g_io_channel_read_chars(client->channel, p, max_length,
-					 &bytes_read, &error);
-	switch (status) {
-	case G_IO_STATUS_NORMAL:
-		fifo_buffer_append(client->input, bytes_read);
-		return httpd_client_received(client);
-
-	case G_IO_STATUS_AGAIN:
-		/* try again later, after select() */
-		return true;
-
-	case G_IO_STATUS_EOF:
-		/* peer disconnected */
-		return false;
-
-	case G_IO_STATUS_ERROR:
-		/* I/O error */
-		g_warning("failed to read from client: %s",
-			  error->message);
-		g_error_free(error);
-		return false;
-	}
-
-	/* unreachable */
-	return false;
-}
-
-static gboolean
-httpd_client_in_event(G_GNUC_UNUSED GIOChannel *source, GIOCondition condition,
-		      gpointer data)
-{
-	struct httpd_client *client = data;
-	struct httpd_output *httpd = client->httpd;
-	bool ret;
-
-	g_mutex_lock(httpd->mutex);
-
-	if (condition == G_IO_IN && httpd_client_read(client)) {
-		ret = true;
-	} else {
-		httpd_client_close(client);
-		ret = false;
-	}
-
-	g_mutex_unlock(httpd->mutex);
-
-	return ret;
-}
-
-struct httpd_client *
-httpd_client_new(struct httpd_output *httpd, int fd, bool metadata_supported)
-{
-	struct httpd_client *client = g_new(struct httpd_client, 1);
-
-	client->httpd = httpd;
-
-	client->channel = g_io_channel_new_socket(fd);
-
-	/* GLib is responsible for closing the file descriptor */
-	g_io_channel_set_close_on_unref(client->channel, true);
-	/* NULL encoding means the stream is binary safe */
-	g_io_channel_set_encoding(client->channel, NULL, NULL);
-	/* we prefer to do buffering */
-	g_io_channel_set_buffered(client->channel, false);
-
-	client->read_source_id = g_io_add_watch(client->channel,
-						G_IO_IN|G_IO_ERR|G_IO_HUP,
-						httpd_client_in_event, client);
-
-	client->input = fifo_buffer_new(4096);
-	client->state = REQUEST;
-
-	client->dlna_streaming_requested = false;
-	client->metadata_supported = metadata_supported;
-	client->metadata_requested = false;
-	client->metadata_sent = true;
-	client->metaint = 8192; /*TODO: just a std value */
-	client->metadata = NULL;
-	client->metadata_current_position = 0;
-	client->metadata_fill = 0;
-
-	return client;
-}
-
-static void
-httpd_client_add_page_size(gpointer data, gpointer user_data)
-{
-	struct page *page = data;
-	size_t *size = user_data;
-
-	*size += page->size;
-}
-
-size_t
-httpd_client_queue_size(const struct httpd_client *client)
-{
-	size_t size = 0;
-
-	if (client->state != RESPONSE)
-		return 0;
-
-	g_queue_foreach(client->pages, httpd_client_add_page_size, &size);
-	return size;
-}
-
-void
-httpd_client_cancel(struct httpd_client *client)
-{
-	if (client->state != RESPONSE)
-		return;
-
-	g_queue_foreach(client->pages, httpd_client_unref_page, NULL);
-	g_queue_clear(client->pages);
-
-	if (client->write_source_id != 0 && client->current_page == NULL) {
-		g_source_remove(client->write_source_id);
-		client->write_source_id = 0;
-	}
-}
-
-static GIOStatus
-write_page_to_channel(GIOChannel *channel,
-		      const struct page *page, size_t position,
-		      gsize *bytes_written_r, GError **error)
-{
-	assert(channel != NULL);
-	assert(page != NULL);
-	assert(position < page->size);
-
-	return g_io_channel_write_chars(channel,
-					(const gchar*)page->data + position,
-					page->size - position,
-					bytes_written_r, error);
-}
-
-static GIOStatus
-write_n_bytes_to_channel(GIOChannel *channel, const struct page *page,
-			 size_t position, gint n,
-			 gsize *bytes_written_r, GError **error)
-{
-	GIOStatus status;
-
-	assert(channel != NULL);
-	assert(page != NULL);
-	assert(position < page->size);
-
-	if (n == -1) {
-		status =  write_page_to_channel (channel, page, position,
-						 bytes_written_r, error);
-	} else {
-		status = g_io_channel_write_chars(channel,
-						  (const gchar*)page->data + position,
-						  n, bytes_written_r, error);
-	}
-
-	return status;
-}
-
-static gint
-bytes_left_till_metadata (struct httpd_client *client)
-{
-	assert(client != NULL);
-
-	if (client->metadata_requested &&
-	    client->current_page->size - client->current_position
-	    > client->metaint - client->metadata_fill)
-		return client->metaint - client->metadata_fill;
-
-	return -1;
-}
-
-static gboolean
-httpd_client_out_event(GIOChannel *source,
-		       G_GNUC_UNUSED GIOCondition condition, gpointer data)
-{
-	struct httpd_client *client = data;
-	struct httpd_output *httpd = client->httpd;
-	GError *error = NULL;
-	GIOStatus status;
-	gsize bytes_written;
-	gint bytes_to_write;
-
-	g_mutex_lock(httpd->mutex);
-
-	assert(condition == G_IO_OUT);
-	assert(client->state == RESPONSE);
-
-	if (client->write_source_id == 0) {
-		/* another thread has removed the event source while
-		   this thread was waiting for httpd->mutex */
-		g_mutex_unlock(httpd->mutex);
-		return false;
-	}
-
-	if (client->current_page == NULL) {
-		client->current_page = g_queue_pop_head(client->pages);
-		client->current_position = 0;
-	}
-
-	bytes_to_write = bytes_left_till_metadata(client);
-
-	if (bytes_to_write == 0) {
-		gint metadata_to_write;
-
-		metadata_to_write = client->metadata_current_position;
-
-		if (!client->metadata_sent) {
-			status = write_page_to_channel(source,
-						       client->metadata,
-						       metadata_to_write,
-						       &bytes_written, &error);
-
-			client->metadata_current_position += bytes_written;
-
-			if (client->metadata->size
-			    - client->metadata_current_position == 0) {
-				client->metadata_fill = 0;
-				client->metadata_current_position = 0;
-				client->metadata_sent = true;
-			}
-		} else {
-			struct page *empty_meta;
-			guchar empty_data = 0;
-
-			empty_meta = page_new_copy(&empty_data, 1);
-
-			status = write_page_to_channel(source,
-						       empty_meta,
-						       metadata_to_write,
-						       &bytes_written, &error);
-
-			client->metadata_current_position += bytes_written;
-
-			if (empty_meta->size
-			    - client->metadata_current_position == 0) {
-				client->metadata_fill = 0;
-				client->metadata_current_position = 0;
-			}
-		}
-
-		bytes_written = 0;
-	} else {
-		status = write_n_bytes_to_channel(source, client->current_page,
-						  client->current_position, bytes_to_write,
-						  &bytes_written, &error);
-	}
-
-	switch (status) {
-	case G_IO_STATUS_NORMAL:
-		client->current_position += bytes_written;
-		assert(client->current_position <= client->current_page->size);
-
-		if (client->metadata_requested)
-			client->metadata_fill += bytes_written;
-
-		if (client->current_position >= client->current_page->size) {
-			page_unref(client->current_page);
-			client->current_page = NULL;
-
-			if (g_queue_is_empty(client->pages)) {
-				/* all pages are sent: remove the
-				   event source */
-				client->write_source_id = 0;
-
-				g_mutex_unlock(httpd->mutex);
-				return false;
-			}
-		}
-
-		g_mutex_unlock(httpd->mutex);
-		return true;
-
-	case G_IO_STATUS_AGAIN:
-		g_mutex_unlock(httpd->mutex);
-		return true;
-
-	case G_IO_STATUS_EOF:
-		/* client has disconnected */
-
-		httpd_client_close(client);
-		g_mutex_unlock(httpd->mutex);
-		return false;
-
-	case G_IO_STATUS_ERROR:
-		/* I/O error */
-
-		g_warning("failed to write to client: %s", error->message);
-		g_error_free(error);
-
-		httpd_client_close(client);
-		g_mutex_unlock(httpd->mutex);
-		return false;
-	}
-
-	/* unreachable */
-	httpd_client_close(client);
-	g_mutex_unlock(httpd->mutex);
-	return false;
-}
-
-void
-httpd_client_send(struct httpd_client *client, struct page *page)
-{
-	if (client->state != RESPONSE)
-		/* the client is still writing the HTTP request */
-		return;
-
-	page_ref(page);
-	g_queue_push_tail(client->pages, page);
-
-	if (client->write_source_id == 0)
-		client->write_source_id =
-			g_io_add_watch(client->channel, G_IO_OUT,
-				       httpd_client_out_event, client);
-}
-
-void
-httpd_client_send_metadata(struct httpd_client *client, struct page *page)
-{
-	if (client->metadata) {
-		page_unref(client->metadata);
-		client->metadata = NULL;
-	}
-
-	g_return_if_fail (page);
-
-	page_ref(page);
-	client->metadata = page;
-	client->metadata_sent = false;
-}
diff --git a/src/output/httpd_client.h b/src/output/httpd_client.h
deleted file mode 100644
index 739163f42..000000000
--- a/src/output/httpd_client.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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_HTTPD_CLIENT_H
-#define MPD_OUTPUT_HTTPD_CLIENT_H
-
-#include <glib.h>
-
-#include <stdbool.h>
-
-struct httpd_client;
-struct httpd_output;
-struct page;
-
-/**
- * Creates a new #httpd_client object
- *
- * @param httpd the HTTP output device
- * @param fd the socket file descriptor
- */
-struct httpd_client *
-httpd_client_new(struct httpd_output *httpd, int fd, bool metadata_supported);
-
-/**
- * Frees memory and resources allocated by the #httpd_client object.
- * This does not remove it from the #httpd_output object.
- */
-void
-httpd_client_free(struct httpd_client *client);
-
-/**
- * Returns the total size of this client's page queue.
- */
-size_t
-httpd_client_queue_size(const struct httpd_client *client);
-
-/**
- * Clears the page queue.
- */
-void
-httpd_client_cancel(struct httpd_client *client);
-
-/**
- * Appends a page to the client's queue.
- */
-void
-httpd_client_send(struct httpd_client *client, struct page *page);
-
-/**
- * Sends the passed metadata.
- */
-void
-httpd_client_send_metadata(struct httpd_client *client, struct page *page);
-
-#endif
diff --git a/src/output/httpd_output_plugin.c b/src/output/httpd_output_plugin.c
deleted file mode 100644
index 1d730df7f..000000000
--- a/src/output/httpd_output_plugin.c
+++ /dev/null
@@ -1,623 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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 "httpd_output_plugin.h"
-#include "httpd_internal.h"
-#include "httpd_client.h"
-#include "output_api.h"
-#include "encoder_plugin.h"
-#include "encoder_list.h"
-#include "resolver.h"
-#include "page.h"
-#include "icy_server.h"
-#include "fd_util.h"
-#include "server_socket.h"
-
-#include <assert.h>
-
-#include <sys/types.h>
-#include <unistd.h>
-#include <errno.h>
-
-#ifdef HAVE_LIBWRAP
-#include <sys/socket.h> /* needed for AF_UNIX */
-#include <tcpd.h>
-#endif
-
-#undef G_LOG_DOMAIN
-#define G_LOG_DOMAIN "httpd_output"
-
-/**
- * The quark used for GError.domain.
- */
-static inline GQuark
-httpd_output_quark(void)
-{
-	return g_quark_from_static_string("httpd_output");
-}
-
-/**
- * Check whether there is at least one client.
- *
- * Caller must lock the mutex.
- */
-G_GNUC_PURE
-static bool
-httpd_output_has_clients(const struct httpd_output *httpd)
-{
-	return httpd->clients != NULL;
-}
-
-/**
- * Check whether there is at least one client.
- */
-G_GNUC_PURE
-static bool
-httpd_output_lock_has_clients(const struct httpd_output *httpd)
-{
-	g_mutex_lock(httpd->mutex);
-	bool result = httpd_output_has_clients(httpd);
-	g_mutex_unlock(httpd->mutex);
-	return result;
-}
-
-static void
-httpd_listen_in_event(int fd, const struct sockaddr *address,
-		      size_t address_length, int uid, void *ctx);
-
-static bool
-httpd_output_bind(struct httpd_output *httpd, GError **error_r)
-{
-	httpd->open = false;
-
-	g_mutex_lock(httpd->mutex);
-	bool success = server_socket_open(httpd->server_socket, error_r);
-	g_mutex_unlock(httpd->mutex);
-
-	return success;
-}
-
-static void
-httpd_output_unbind(struct httpd_output *httpd)
-{
-	assert(!httpd->open);
-
-	g_mutex_lock(httpd->mutex);
-	server_socket_close(httpd->server_socket);
-	g_mutex_unlock(httpd->mutex);
-}
-
-static struct audio_output *
-httpd_output_init(const struct config_param *param,
-		  GError **error)
-{
-	struct httpd_output *httpd = g_new(struct httpd_output, 1);
-	if (!ao_base_init(&httpd->base, &httpd_output_plugin, param, error)) {
-		g_free(httpd);
-		return NULL;
-	}
-
-	/* read configuration */
-	httpd->name =
-		config_get_block_string(param, "name", "Set name in config");
-	httpd->genre =
-		config_get_block_string(param, "genre", "Set genre in config");
-	httpd->website =
-		config_get_block_string(param, "website", "Set website in config");
-
-	guint port = config_get_block_unsigned(param, "port", 8000);
-
-	const char *encoder_name =
-		config_get_block_string(param, "encoder", "vorbis");
-	const struct encoder_plugin *encoder_plugin =
-		encoder_plugin_get(encoder_name);
-	if (encoder_plugin == NULL) {
-		g_set_error(error, httpd_output_quark(), 0,
-			    "No such encoder: %s", encoder_name);
-		ao_base_finish(&httpd->base);
-		g_free(httpd);
-		return NULL;
-	}
-
-	httpd->clients_max = config_get_block_unsigned(param,"max_clients", 0);
-
-	/* set up bind_to_address */
-
-	httpd->server_socket = server_socket_new(httpd_listen_in_event, httpd);
-
-	const char *bind_to_address =
-		config_get_block_string(param, "bind_to_address", NULL);
-	bool success = bind_to_address != NULL &&
-		strcmp(bind_to_address, "any") != 0
-		? server_socket_add_host(httpd->server_socket, bind_to_address,
-					 port, error)
-		: server_socket_add_port(httpd->server_socket, port, error);
-	if (!success) {
-		ao_base_finish(&httpd->base);
-		g_free(httpd);
-		return NULL;
-	}
-
-	/* initialize metadata */
-	httpd->metadata = NULL;
-	httpd->unflushed_input = 0;
-
-	/* initialize encoder */
-
-	httpd->encoder = encoder_init(encoder_plugin, param, error);
-	if (httpd->encoder == NULL) {
-		ao_base_finish(&httpd->base);
-		g_free(httpd);
-		return NULL;
-	}
-
-	/* determine content type */
-	httpd->content_type = encoder_get_mime_type(httpd->encoder);
-	if (httpd->content_type == NULL) {
-		httpd->content_type = "application/octet-stream";
-	}
-
-	httpd->mutex = g_mutex_new();
-
-	return &httpd->base;
-}
-
-static void
-httpd_output_finish(struct audio_output *ao)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	if (httpd->metadata)
-		page_unref(httpd->metadata);
-
-	encoder_finish(httpd->encoder);
-	server_socket_free(httpd->server_socket);
-	g_mutex_free(httpd->mutex);
-	ao_base_finish(&httpd->base);
-	g_free(httpd);
-}
-
-/**
- * Creates a new #httpd_client object and adds it into the
- * httpd_output.clients linked list.
- */
-static void
-httpd_client_add(struct httpd_output *httpd, int fd)
-{
-	struct httpd_client *client =
-		httpd_client_new(httpd, fd,
-				 httpd->encoder->plugin->tag == NULL);
-
-	httpd->clients = g_list_prepend(httpd->clients, client);
-	httpd->clients_cnt++;
-
-	/* pass metadata to client */
-	if (httpd->metadata)
-		httpd_client_send_metadata(client, httpd->metadata);
-}
-
-static void
-httpd_listen_in_event(int fd, const struct sockaddr *address,
-		      size_t address_length, G_GNUC_UNUSED int uid, void *ctx)
-{
-	struct httpd_output *httpd = ctx;
-
-	/* the listener socket has become readable - a client has
-	   connected */
-
-#ifdef HAVE_LIBWRAP
-	if (address->sa_family != AF_UNIX) {
-		char *hostaddr = sockaddr_to_string(address, address_length, NULL);
-		const char *progname = g_get_prgname();
-
-		struct request_info req;
-		request_init(&req, RQ_FILE, fd, RQ_DAEMON, progname, 0);
-
-		fromhost(&req);
-
-		if (!hosts_access(&req)) {
-			/* tcp wrappers says no */
-			g_warning("libwrap refused connection (libwrap=%s) from %s",
-			      progname, hostaddr);
-			g_free(hostaddr);
-			close_socket(fd);
-			g_mutex_unlock(httpd->mutex);
-			return;
-		}
-
-		g_free(hostaddr);
-	}
-#else
-	(void)address;
-	(void)address_length;
-#endif	/* HAVE_WRAP */
-
-	g_mutex_lock(httpd->mutex);
-
-	if (fd >= 0) {
-		/* can we allow additional client */
-		if (httpd->open &&
-		    (httpd->clients_max == 0 ||
-		     httpd->clients_cnt < httpd->clients_max))
-			httpd_client_add(httpd, fd);
-		else
-			close_socket(fd);
-	} else if (fd < 0 && errno != EINTR) {
-		g_warning("accept() failed: %s", g_strerror(errno));
-	}
-
-	g_mutex_unlock(httpd->mutex);
-}
-
-/**
- * Reads data from the encoder (as much as available) and returns it
- * as a new #page object.
- */
-static struct page *
-httpd_output_read_page(struct httpd_output *httpd)
-{
-	if (httpd->unflushed_input >= 65536) {
-		/* we have fed a lot of input into the encoder, but it
-		   didn't give anything back yet - flush now to avoid
-		   buffer underruns */
-		encoder_flush(httpd->encoder, NULL);
-		httpd->unflushed_input = 0;
-	}
-
-	size_t size = 0;
-	do {
-		size_t nbytes = encoder_read(httpd->encoder,
-					     httpd->buffer + size,
-					     sizeof(httpd->buffer) - size);
-		if (nbytes == 0)
-			break;
-
-		httpd->unflushed_input = 0;
-
-		size += nbytes;
-	} while (size < sizeof(httpd->buffer));
-
-	if (size == 0)
-		return NULL;
-
-	return page_new_copy(httpd->buffer, size);
-}
-
-static bool
-httpd_output_encoder_open(struct httpd_output *httpd,
-			  struct audio_format *audio_format,
-			  GError **error)
-{
-	if (!encoder_open(httpd->encoder, audio_format, error))
-		return false;
-
-	/* we have to remember the encoder header, i.e. the first
-	   bytes of encoder output after opening it, because it has to
-	   be sent to every new client */
-	httpd->header = httpd_output_read_page(httpd);
-
-	httpd->unflushed_input = 0;
-
-	return true;
-}
-
-static bool
-httpd_output_enable(struct audio_output *ao, GError **error_r)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	return httpd_output_bind(httpd, error_r);
-}
-
-static void
-httpd_output_disable(struct audio_output *ao)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	httpd_output_unbind(httpd);
-}
-
-static bool
-httpd_output_open(struct audio_output *ao, struct audio_format *audio_format,
-		  GError **error)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	g_mutex_lock(httpd->mutex);
-
-	/* open the encoder */
-
-	if (!httpd_output_encoder_open(httpd, audio_format, error)) {
-		g_mutex_unlock(httpd->mutex);
-		return false;
-	}
-
-	/* initialize other attributes */
-
-	httpd->clients = NULL;
-	httpd->clients_cnt = 0;
-	httpd->timer = timer_new(audio_format);
-
-	httpd->open = true;
-
-	g_mutex_unlock(httpd->mutex);
-	return true;
-}
-
-static void
-httpd_client_delete(gpointer data, G_GNUC_UNUSED gpointer user_data)
-{
-	struct httpd_client *client = data;
-
-	httpd_client_free(client);
-}
-
-static void
-httpd_output_close(struct audio_output *ao)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	g_mutex_lock(httpd->mutex);
-
-	httpd->open = false;
-
-	timer_free(httpd->timer);
-
-	g_list_foreach(httpd->clients, httpd_client_delete, NULL);
-	g_list_free(httpd->clients);
-
-	if (httpd->header != NULL)
-		page_unref(httpd->header);
-
-	encoder_close(httpd->encoder);
-
-	g_mutex_unlock(httpd->mutex);
-}
-
-void
-httpd_output_remove_client(struct httpd_output *httpd,
-			   struct httpd_client *client)
-{
-	assert(httpd != NULL);
-	assert(client != NULL);
-
-	httpd->clients = g_list_remove(httpd->clients, client);
-	httpd->clients_cnt--;
-}
-
-void
-httpd_output_send_header(struct httpd_output *httpd,
-			 struct httpd_client *client)
-{
-	if (httpd->header != NULL)
-		httpd_client_send(client, httpd->header);
-}
-
-static unsigned
-httpd_output_delay(struct audio_output *ao)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	if (!httpd_output_lock_has_clients(httpd) && httpd->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 */
-		timer_reset(httpd->timer);
-
-		/* some arbitrary delay that is long enough to avoid
-		   consuming too much CPU, and short enough to notice
-		   new clients quickly enough */
-		return 1000;
-	}
-
-	return httpd->timer->started
-		? timer_delay(httpd->timer)
-		: 0;
-}
-
-static void
-httpd_client_check_queue(gpointer data, G_GNUC_UNUSED gpointer user_data)
-{
-	struct httpd_client *client = data;
-
-	if (httpd_client_queue_size(client) > 256 * 1024) {
-		g_debug("client is too slow, flushing its queue");
-		httpd_client_cancel(client);
-	}
-}
-
-static void
-httpd_client_send_page(gpointer data, gpointer user_data)
-{
-	struct httpd_client *client = data;
-	struct page *page = user_data;
-
-	httpd_client_send(client, page);
-}
-
-/**
- * Broadcasts a page struct to all clients.
- */
-static void
-httpd_output_broadcast_page(struct httpd_output *httpd, struct page *page)
-{
-	assert(page != NULL);
-
-	g_mutex_lock(httpd->mutex);
-	g_list_foreach(httpd->clients, httpd_client_send_page, page);
-	g_mutex_unlock(httpd->mutex);
-}
-
-/**
- * Broadcasts data from the encoder to all clients.
- */
-static void
-httpd_output_encoder_to_clients(struct httpd_output *httpd)
-{
-	struct page *page;
-
-	g_mutex_lock(httpd->mutex);
-	g_list_foreach(httpd->clients, httpd_client_check_queue, NULL);
-	g_mutex_unlock(httpd->mutex);
-
-	while ((page = httpd_output_read_page(httpd)) != NULL) {
-		httpd_output_broadcast_page(httpd, page);
-		page_unref(page);
-	}
-}
-
-static bool
-httpd_output_encode_and_play(struct httpd_output *httpd,
-			     const void *chunk, size_t size, GError **error)
-{
-	if (!encoder_write(httpd->encoder, chunk, size, error))
-		return false;
-
-	httpd->unflushed_input += size;
-
-	httpd_output_encoder_to_clients(httpd);
-
-	return true;
-}
-
-static size_t
-httpd_output_play(struct audio_output *ao, const void *chunk, size_t size,
-		  GError **error_r)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	if (httpd_output_lock_has_clients(httpd)) {
-		if (!httpd_output_encode_and_play(httpd, chunk, size, error_r))
-			return 0;
-	}
-
-	if (!httpd->timer->started)
-		timer_start(httpd->timer);
-	timer_add(httpd->timer, size);
-
-	return size;
-}
-
-static bool
-httpd_output_pause(struct audio_output *ao)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	if (httpd_output_lock_has_clients(httpd)) {
-		static const char silence[1020];
-		return httpd_output_play(ao, silence, sizeof(silence),
-					 NULL) > 0;
-	} else {
-		return true;
-	}
-}
-
-static void
-httpd_send_metadata(gpointer data, gpointer user_data)
-{
-	struct httpd_client *client = data;
-	struct page *icy_metadata = user_data;
-
-	httpd_client_send_metadata(client, icy_metadata);
-}
-
-static void
-httpd_output_tag(struct audio_output *ao, const struct tag *tag)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	assert(tag != NULL);
-
-	if (httpd->encoder->plugin->tag != NULL) {
-		/* embed encoder tags */
-
-		/* flush the current stream, and end it */
-
-		encoder_pre_tag(httpd->encoder, NULL);
-		httpd_output_encoder_to_clients(httpd);
-
-		/* send the tag to the encoder - which starts a new
-		   stream now */
-
-		encoder_tag(httpd->encoder, tag, NULL);
-
-		/* the first page generated by the encoder will now be
-		   used as the new "header" page, which is sent to all
-		   new clients */
-
-		struct page *page = httpd_output_read_page(httpd);
-		if (page != NULL) {
-			if (httpd->header != NULL)
-				page_unref(httpd->header);
-			httpd->header = page;
-			httpd_output_broadcast_page(httpd, page);
-		}
-	} else {
-		/* use Icy-Metadata */
-
-		if (httpd->metadata != NULL)
-			page_unref (httpd->metadata);
-
-		httpd->metadata =
-			icy_server_metadata_page(tag, TAG_ALBUM,
-						 TAG_ARTIST, TAG_TITLE,
-						 TAG_NUM_OF_ITEM_TYPES);
-		if (httpd->metadata != NULL) {
-			g_mutex_lock(httpd->mutex);
-			g_list_foreach(httpd->clients,
-				       httpd_send_metadata, httpd->metadata);
-			g_mutex_unlock(httpd->mutex);
-		}
-	}
-}
-
-static void
-httpd_client_cancel_callback(gpointer data, G_GNUC_UNUSED gpointer user_data)
-{
-	struct httpd_client *client = data;
-
-	httpd_client_cancel(client);
-}
-
-static void
-httpd_output_cancel(struct audio_output *ao)
-{
-	struct httpd_output *httpd = (struct httpd_output *)ao;
-
-	g_mutex_lock(httpd->mutex);
-	g_list_foreach(httpd->clients, httpd_client_cancel_callback, NULL);
-	g_mutex_unlock(httpd->mutex);
-}
-
-const struct audio_output_plugin httpd_output_plugin = {
-	.name = "httpd",
-	.init = httpd_output_init,
-	.finish = httpd_output_finish,
-	.enable = httpd_output_enable,
-	.disable = httpd_output_disable,
-	.open = httpd_output_open,
-	.close = httpd_output_close,
-	.delay = httpd_output_delay,
-	.send_tag = httpd_output_tag,
-	.play = httpd_output_play,
-	.pause = httpd_output_pause,
-	.cancel = httpd_output_cancel,
-};
diff --git a/src/output/mvp_output_plugin.c b/src/output/mvp_output_plugin.c
deleted file mode 100644
index 37e0f7c93..000000000
--- a/src/output/mvp_output_plugin.c
+++ /dev/null
@@ -1,344 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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.
- */
-
-/*
- * Media MVP audio output based on code from MVPMC project:
- * http://mvpmc.sourceforge.net/
- */
-
-#include "config.h"
-#include "mvp_output_plugin.h"
-#include "output_api.h"
-#include "fd_util.h"
-
-#include <glib.h>
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-#include <errno.h>
-#include <unistd.h>
-#include <stdlib.h>
-
-#undef G_LOG_DOMAIN
-#define G_LOG_DOMAIN "mvp"
-
-typedef struct {
-	unsigned long dsp_status;
-	unsigned long stream_decode_type;
-	unsigned long sample_rate;
-	unsigned long bit_rate;
-	unsigned long raw[64 / sizeof(unsigned long)];
-} aud_status_t;
-
-#define MVP_SET_AUD_STOP		_IOW('a',1,int)
-#define MVP_SET_AUD_PLAY		_IOW('a',2,int)
-#define MVP_SET_AUD_PAUSE	_IOW('a',3,int)
-#define MVP_SET_AUD_UNPAUSE	_IOW('a',4,int)
-#define MVP_SET_AUD_SRC		_IOW('a',5,int)
-#define MVP_SET_AUD_MUTE		_IOW('a',6,int)
-#define MVP_SET_AUD_BYPASS	_IOW('a',8,int)
-#define MVP_SET_AUD_CHANNEL	_IOW('a',9,int)
-#define MVP_GET_AUD_STATUS	_IOR('a',10,aud_status_t)
-#define MVP_SET_AUD_VOLUME	_IOW('a',13,int)
-#define MVP_GET_AUD_VOLUME	_IOR('a',14,int)
-#define MVP_SET_AUD_STREAMTYPE	_IOW('a',15,int)
-#define MVP_SET_AUD_FORMAT	_IOW('a',16,int)
-#define MVP_GET_AUD_SYNC		_IOR('a',21,pts_sync_data_t*)
-#define MVP_SET_AUD_STC		_IOW('a',22,long long int *)
-#define MVP_SET_AUD_SYNC		_IOW('a',23,int)
-#define MVP_SET_AUD_END_STREAM	_IOW('a',25,int)
-#define MVP_SET_AUD_RESET	_IOW('a',26,int)
-#define MVP_SET_AUD_DAC_CLK	_IOW('a',27,int)
-#define MVP_GET_AUD_REGS		_IOW('a',28,aud_ctl_regs_t*)
-
-struct mvp_data {
-	struct audio_output base;
-
-	struct audio_format audio_format;
-	int fd;
-};
-
-static const unsigned mvp_sample_rates[][3] = {
-	{9, 8000, 32000},
-	{10, 11025, 44100},
-	{11, 12000, 48000},
-	{1, 16000, 32000},
-	{2, 22050, 44100},
-	{3, 24000, 48000},
-	{5, 32000, 32000},
-	{0, 44100, 44100},
-	{7, 48000, 48000},
-	{13, 64000, 32000},
-	{14, 88200, 44100},
-	{15, 96000, 48000}
-};
-
-/**
- * The quark used for GError.domain.
- */
-static inline GQuark
-mvp_output_quark(void)
-{
-	return g_quark_from_static_string("mvp_output");
-}
-
-/**
- * Translate a sample rate to a MVP sample rate.
- *
- * @param sample_rate the sample rate in Hz
- */
-static unsigned
-mvp_find_sample_rate(unsigned sample_rate)
-{
-	for (unsigned i = 0; i < G_N_ELEMENTS(mvp_sample_rates); ++i)
-		if (mvp_sample_rates[i][1] == sample_rate)
-			return mvp_sample_rates[i][0];
-
-	return (unsigned)-1;
-}
-
-static bool
-mvp_output_test_default_device(void)
-{
-	int fd;
-
-	fd = open_cloexec("/dev/adec_pcm", O_WRONLY, 0);
-
-	if (fd >= 0) {
-		close(fd);
-		return true;
-	}
-
-	g_warning("Error opening PCM device \"/dev/adec_pcm\": %s\n",
-		  g_strerror(errno));
-
-	return false;
-}
-
-static struct audio_output *
-mvp_output_init(G_GNUC_UNUSED const struct config_param *param, GError **error)
-{
-	struct mvp_data *md = g_new(struct mvp_data, 1);
-
-	if (!ao_base_init(&md->base, &mvp_output_plugin, param, error)) {
-		g_free(md);
-		return NULL;
-	}
-
-	md->fd = -1;
-
-	return &md->base;
-}
-
-static void
-mvp_output_finish(struct audio_output *ao)
-{
-	struct mvp_data *md = (struct mvp_data *)ao;
-	ao_base_finish(&md->base);
-	g_free(md);
-}
-
-static bool
-mvp_set_pcm_params(struct mvp_data *md, struct audio_format *audio_format,
-		   GError **error)
-{
-	unsigned mix[5];
-
-	switch (audio_format->channels) {
-	case 1:
-		mix[0] = 1;
-		break;
-
-	case 2:
-		mix[0] = 0;
-		break;
-
-	default:
-		g_debug("unsupported channel count %u - falling back to stereo",
-			audio_format->channels);
-		audio_format->channels = 2;
-		mix[0] = 0;
-		break;
-	}
-
-	/* 0,1=24bit(24) , 2,3=16bit */
-	switch (audio_format->format) {
-	case SAMPLE_FORMAT_S16:
-		mix[1] = 2;
-		break;
-
-	case SAMPLE_FORMAT_S24_P32:
-		mix[1] = 0;
-		break;
-
-	default:
-		g_debug("unsupported sample format %s - falling back to 16 bit",
-			sample_format_to_string(audio_format->format));
-		audio_format->format = SAMPLE_FORMAT_S16;
-		mix[1] = 2;
-		break;
-	}
-
-	mix[3] = 0;	/* stream type? */
-	mix[4] = G_BYTE_ORDER == G_LITTLE_ENDIAN;
-
-	/*
-	 * if there is an exact match for the frequency, use it.
-	 */
-	mix[2] = mvp_find_sample_rate(audio_format->sample_rate);
-	if (mix[2] == (unsigned)-1) {
-		g_set_error(error, mvp_output_quark(), 0,
-			    "Can not find suitable output frequency for %u",
-			    audio_format->sample_rate);
-		return false;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Can not set audio format");
-		return false;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Can not set audio sync");
-		return false;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Can not set audio play mode");
-		return false;
-	}
-
-	return true;
-}
-
-static bool
-mvp_output_open(struct audio_output *ao, struct audio_format *audio_format,
-		GError **error)
-{
-	struct mvp_data *md = (struct mvp_data *)ao;
-	long long int stc = 0;
-	int mix[5] = { 0, 2, 7, 1, 0 };
-	bool success;
-
-	md->fd = open_cloexec("/dev/adec_pcm", O_RDWR | O_NONBLOCK, 0);
-	if (md->fd < 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Error opening /dev/adec_pcm: %s",
-			    g_strerror(errno));
-		return false;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Error setting audio source: %s",
-			    g_strerror(errno));
-		return false;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Error setting audio streamtype: %s",
-			    g_strerror(errno));
-		return false;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Error setting audio format: %s",
-			    g_strerror(errno));
-		return false;
-	}
-	ioctl(md->fd, MVP_SET_AUD_STC, &stc);
-	if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
-		g_set_error(error, mvp_output_quark(), errno,
-			    "Error setting audio streamtype: %s",
-			    g_strerror(errno));
-		return false;
-	}
-
-	success = mvp_set_pcm_params(md, audio_format, error);
-	if (!success)
-		return false;
-
-	md->audio_format = *audio_format;
-	return true;
-}
-
-static void mvp_output_close(struct audio_output *ao)
-{
-	struct mvp_data *md = (struct mvp_data *)ao;
-	if (md->fd >= 0)
-		close(md->fd);
-	md->fd = -1;
-}
-
-static void mvp_output_cancel(struct audio_output *ao)
-{
-	struct mvp_data *md = (struct mvp_data *)ao;
-	if (md->fd >= 0) {
-		ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
-		close(md->fd);
-		md->fd = -1;
-	}
-}
-
-static size_t
-mvp_output_play(struct audio_output *ao, const void *chunk, size_t size,
-		GError **error)
-{
-	struct mvp_data *md = (struct mvp_data *)ao;
-	ssize_t ret;
-
-	/* reopen the device since it was closed by dropBufferedAudio */
-	if (md->fd < 0) {
-		bool success;
-
-		success = mvp_output_open(ao, &md->audio_format, error);
-		if (!success)
-			return 0;
-	}
-
-	while (true) {
-		ret = write(md->fd, chunk, size);
-		if (ret > 0)
-			return (size_t)ret;
-
-		if (ret < 0) {
-			if (errno == EINTR)
-				continue;
-
-			g_set_error(error, mvp_output_quark(), errno,
-				    "Failed to write: %s", g_strerror(errno));
-			return 0;
-		}
-	}
-}
-
-const struct audio_output_plugin mvp_output_plugin = {
-	.name = "mvp",
-	.test_default_device = mvp_output_test_default_device,
-	.init = mvp_output_init,
-	.finish = mvp_output_finish,
-	.open = mvp_output_open,
-	.close = mvp_output_close,
-	.play = mvp_output_play,
-	.cancel = mvp_output_cancel,
-};
diff --git a/src/output/mvp_output_plugin.h b/src/output/mvp_output_plugin.h
deleted file mode 100644
index e403de2b7..000000000
--- a/src/output/mvp_output_plugin.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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_MVP_OUTPUT_PLUGIN_H
-#define MPD_MVP_OUTPUT_PLUGIN_H
-
-extern const struct audio_output_plugin mvp_output_plugin;
-
-#endif
diff --git a/src/output/null_output_plugin.c b/src/output/null_output_plugin.c
deleted file mode 100644
index 9d7588fff..000000000
--- a/src/output/null_output_plugin.c
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- * Copyright (C) 2003-2011 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 "null_output_plugin.h"
-#include "output_api.h"
-#include "timer.h"
-
-#include <glib.h>
-
-#include <assert.h>
-
-struct null_data {
-	struct audio_output base;
-
-	bool sync;
-
-	struct timer *timer;
-};
-
-static struct audio_output *
-null_init(const struct config_param *param, GError **error_r)
-{
-	struct null_data *nd = g_new(struct null_data, 1);
-
-	if (!ao_base_init(&nd->base, &null_output_plugin, param, error_r)) {
-		g_free(nd);
-		return NULL;
-	}
-
-	nd->sync = config_get_block_bool(param, "sync", true);
-
-	return &nd->base;
-}
-
-static void
-null_finish(struct audio_output *ao)
-{
-	struct null_data *nd = (struct null_data *)ao;
-
-	ao_base_finish(&nd->base);
-	g_free(nd);
-}
-
-static bool
-null_open(struct audio_output *ao, struct audio_format *audio_format,
-	  G_GNUC_UNUSED GError **error)
-{
-	struct null_data *nd = (struct null_data *)ao;
-
-	if (nd->sync)
-		nd->timer = timer_new(audio_format);
-
-	return true;
-}
-
-static void
-null_close(struct audio_output *ao)
-{
-	struct null_data *nd = (struct null_data *)ao;
-
-	if (nd->sync)
-		timer_free(nd->timer);
-}
-
-static unsigned
-null_delay(struct audio_output *ao)
-{
-	struct null_data *nd = (struct null_data *)ao;
-
-	return nd->sync && nd->timer->started
-		? timer_delay(nd->timer)
-		: 0;
-}
-
-static size_t
-null_play(struct audio_output *ao, G_GNUC_UNUSED const void *chunk, size_t size,
-	  G_GNUC_UNUSED GError **error)
-{
-	struct null_data *nd = (struct null_data *)ao;
-	struct timer *timer = nd->timer;
-
-	if (!nd->sync)
-		return size;
-
-	if (!timer->started)
-		timer_start(timer);
-	timer_add(timer, size);
-
-	return size;
-}
-
-static void
-null_cancel(struct audio_output *ao)
-{
-	struct null_data *nd = (struct null_data *)ao;
-
-	if (!nd->sync)
-		return;
-
-	timer_reset(nd->timer);
-}
-
-const struct audio_output_plugin null_output_plugin = {
-	.name = "null",
-	.init = null_init,
-	.finish = null_finish,
-	.open = null_open,
-	.close = null_close,
-	.delay = null_delay,
-	.play = null_play,
-	.cancel = null_cancel,
-};