output/{alsa,oss}: convert to C++

1 files changed, 0 insertions, 819 deletions

diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
deleted file mode 100644
index d8b18427..00000000
--- a/src/output/alsa_output_plugin.c
+++ /dev/null
@@ -1,819 +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 "alsa_output_plugin.h"
-#include "output_api.h"
-#include "mixer_list.h"
-#include "pcm_export.h"
-
-#include <glib.h>
-#include <alsa/asoundlib.h>
-
-#undef G_LOG_DOMAIN
-#define G_LOG_DOMAIN "alsa"
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-static const char default_device[] = "default";
-
-enum {
-	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 audio_output base;
-
-	struct pcm_export_state export;
-
-	/** the configured name of the ALSA device; NULL for the
-	    default device */
-	char *device;
-
-	/** use memory mapped I/O? */
-	bool use_mmap;
-
-	/**
-	 * Enable DSD over USB according to the dCS suggested
-	 * standard?
-	 *
-	 * @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
-	 */
-	bool dsd_usb;
-
-	/** libasound's buffer_time setting (in microseconds) */
-	unsigned int buffer_time;
-
-	/** libasound's period_time setting (in microseconds) */
-	unsigned int period_time;
-
-	/** the mode flags passed to snd_pcm_open */
-	int mode;
-
-	/** the libasound PCM device handle */
-	snd_pcm_t *pcm;
-
-	/**
-	 * a pointer to the libasound writei() function, which is
-	 * snd_pcm_writei() or snd_pcm_mmap_writei(), depending on the
-	 * use_mmap configuration
-	 */
-	alsa_writei_t *writei;
-
-	/**
-	 * The size of one audio frame passed to method play().
-	 */
-	size_t in_frame_size;
-
-	/**
-	 * The size of one audio frame passed to libasound.
-	 */
-	size_t out_frame_size;
-
-	/**
-	 * The size of one period, in number of frames.
-	 */
-	snd_pcm_uframes_t period_frames;
-
-	/**
-	 * The number of frames written in the current period.
-	 */
-	snd_pcm_uframes_t period_position;
-};
-
-/**
- * The quark used for GError.domain.
- */
-static inline GQuark
-alsa_output_quark(void)
-{
-	return g_quark_from_static_string("alsa_output");
-}
-
-static const char *
-alsa_device(const struct alsa_data *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;
-}
-
-static void
-alsa_configure(struct alsa_data *ad, const struct config_param *param)
-{
-	ad->device = config_dup_block_string(param, "device", NULL);
-
-	ad->use_mmap = config_get_block_bool(param, "use_mmap", false);
-
-	ad->dsd_usb = config_get_block_bool(param, "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);
-
-#ifdef SND_PCM_NO_AUTO_RESAMPLE
-	if (!config_get_block_bool(param, "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))
-		ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_FORMAT
-	if (!config_get_block_bool(param, "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)
-{
-	struct alsa_data *ad = alsa_data_new();
-
-	if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
-		g_free(ad);
-		return NULL;
-	}
-
-	alsa_configure(ad, param);
-
-	return &ad->base;
-}
-
-static void
-alsa_finish(struct audio_output *ao)
-{
-	struct alsa_data *ad = (struct alsa_data *)ao;
-
-	ao_base_finish(&ad->base);
-
-	g_free(ad->device);
-	g_free(ad);
-
-	/* free libasound's config cache */
-	snd_config_update_free_global();
-}
-
-static bool
-alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
-{
-	struct alsa_data *ad = (struct alsa_data *)ao;
-
-	pcm_export_init(&ad->export);
-	return true;
-}
-
-static void
-alsa_output_disable(struct audio_output *ao)
-{
-	struct alsa_data *ad = (struct alsa_data *)ao;
-
-	pcm_export_deinit(&ad->export);
-}
-
-static bool
-alsa_test_default_device(void)
-{
-	snd_pcm_t *handle;
-
-	int ret = snd_pcm_open(&handle, default_device,
-	                       SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-	if (ret) {
-		g_message("Error opening default ALSA device: %s\n",
-			  snd_strerror(-ret));
-		return false;
-	} else
-		snd_pcm_close(handle);
-
-	return true;
-}
-
-static snd_pcm_format_t
-get_bitformat(enum sample_format sample_format)
-{
-	switch (sample_format) {
-	case SAMPLE_FORMAT_UNDEFINED:
-	case SAMPLE_FORMAT_DSD:
-		return SND_PCM_FORMAT_UNKNOWN;
-
-	case SAMPLE_FORMAT_S8:
-		return SND_PCM_FORMAT_S8;
-
-	case SAMPLE_FORMAT_S16:
-		return SND_PCM_FORMAT_S16;
-
-	case SAMPLE_FORMAT_S24_P32:
-		return SND_PCM_FORMAT_S24;
-
-	case SAMPLE_FORMAT_S32:
-		return SND_PCM_FORMAT_S32;
-
-	case SAMPLE_FORMAT_FLOAT:
-		return SND_PCM_FORMAT_FLOAT;
-	}
-
-	assert(false);
-	return SND_PCM_FORMAT_UNKNOWN;
-}
-
-static snd_pcm_format_t
-byteswap_bitformat(snd_pcm_format_t fmt)
-{
-	switch(fmt) {
-	case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
-	case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
-	case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
-	case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
-	case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
-
-	case SND_PCM_FORMAT_S24_3BE:
-		return SND_PCM_FORMAT_S24_3LE;
-
-	case SND_PCM_FORMAT_S24_3LE:
-		return SND_PCM_FORMAT_S24_3BE;
-
-	case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
-	default: return SND_PCM_FORMAT_UNKNOWN;
-	}
-}
-
-static snd_pcm_format_t
-alsa_to_packed_format(snd_pcm_format_t fmt)
-{
-	switch (fmt) {
-	case SND_PCM_FORMAT_S24_LE:
-		return SND_PCM_FORMAT_S24_3LE;
-
-	case SND_PCM_FORMAT_S24_BE:
-		return SND_PCM_FORMAT_S24_3BE;
-
-	default:
-		return SND_PCM_FORMAT_UNKNOWN;
-	}
-}
-
-static int
-alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
-			  snd_pcm_format_t fmt, bool *packed_r)
-{
-	int err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
-	if (err == 0)
-		*packed_r = false;
-
-	if (err != -EINVAL)
-		return err;
-
-	fmt = alsa_to_packed_format(fmt);
-	if (fmt == SND_PCM_FORMAT_UNKNOWN)
-		return -EINVAL;
-
-	err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
-	if (err == 0)
-		*packed_r = true;
-
-	return err;
-}
-
-/**
- * Attempts to configure the specified sample format, and tries the
- * reversed host byte order if was not supported.
- */
-static int
-alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
-		       enum sample_format sample_format,
-		       bool *packed_r, bool *reverse_endian_r)
-{
-	snd_pcm_format_t alsa_format = get_bitformat(sample_format);
-	if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
-		return -EINVAL;
-
-	int err = alsa_try_format_or_packed(pcm, hwparams, alsa_format,
-					    packed_r);
-	if (err == 0)
-		*reverse_endian_r = false;
-
-	if (err != -EINVAL)
-		return err;
-
-	alsa_format = byteswap_bitformat(alsa_format);
-	if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
-		return -EINVAL;
-
-	err = alsa_try_format_or_packed(pcm, hwparams, alsa_format, packed_r);
-	if (err == 0)
-		*reverse_endian_r = true;
-
-	return err;
-}
-
-/**
- * Configure a sample format, and probe other formats if that fails.
- */
-static int
-alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
-			 struct audio_format *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,
-					 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,
-	};
-
-	for (unsigned i = 0;
-	     err == -EINVAL && probe_formats[i] != SAMPLE_FORMAT_UNDEFINED;
-	     ++i) {
-		const enum sample_format 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;
-	}
-
-	return err;
-}
-
-/**
- * Set up the snd_pcm_t object which was opened by the caller.  Set up
- * the configured settings and the audio format.
- */
-static bool
-alsa_setup(struct alsa_data *ad, struct audio_format *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;
-	int err;
-	const char *cmd = NULL;
-	int retry = MPD_ALSA_RETRY_NR;
-	unsigned int period_time, period_time_ro;
-	unsigned int buffer_time;
-
-	period_time_ro = period_time = ad->period_time;
-configure_hw:
-	/* configure HW params */
-	snd_pcm_hw_params_alloca(&hwparams);
-	cmd = "snd_pcm_hw_params_any";
-	err = snd_pcm_hw_params_any(ad->pcm, hwparams);
-	if (err < 0)
-		goto error;
-
-	if (ad->use_mmap) {
-		err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
-						   SND_PCM_ACCESS_MMAP_INTERLEAVED);
-		if (err < 0) {
-			g_warning("Cannot set mmap'ed mode on ALSA device \"%s\":  %s\n",
-				  alsa_device(ad), snd_strerror(-err));
-			g_warning("Falling back to direct write mode\n");
-			ad->use_mmap = false;
-		} else
-			ad->writei = snd_pcm_mmap_writei;
-	}
-
-	if (!ad->use_mmap) {
-		cmd = "snd_pcm_hw_params_set_access";
-		err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
-						   SND_PCM_ACCESS_RW_INTERLEAVED);
-		if (err < 0)
-			goto error;
-		ad->writei = snd_pcm_writei;
-	}
-
-	err = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
-				       packed_r, reverse_endian_r);
-	if (err < 0) {
-		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),
-			    snd_strerror(-err));
-		return false;
-	}
-
-	snd_pcm_format_t format;
-	if (snd_pcm_hw_params_get_format(hwparams, &format) == 0)
-		g_debug("format=%s (%s)", snd_pcm_format_name(format),
-			snd_pcm_format_description(format));
-
-	err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
-						  &channels);
-	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,
-			    snd_strerror(-err));
-		return false;
-	}
-	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);
-		return false;
-	}
-	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);
-	snd_pcm_hw_params_get_buffer_size_max(hwparams, &buffer_size_max);
-	unsigned buffer_time_min, buffer_time_max;
-	snd_pcm_hw_params_get_buffer_time_min(hwparams, &buffer_time_min, 0);
-	snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time_max, 0);
-	g_debug("buffer: size=%u..%u time=%u..%u",
-		(unsigned)buffer_size_min, (unsigned)buffer_size_max,
-		buffer_time_min, buffer_time_max);
-
-	snd_pcm_uframes_t period_size_min, period_size_max;
-	snd_pcm_hw_params_get_period_size_min(hwparams, &period_size_min, 0);
-	snd_pcm_hw_params_get_period_size_max(hwparams, &period_size_max, 0);
-	unsigned period_time_min, period_time_max;
-	snd_pcm_hw_params_get_period_time_min(hwparams, &period_time_min, 0);
-	snd_pcm_hw_params_get_period_time_max(hwparams, &period_time_max, 0);
-	g_debug("period: size=%u..%u time=%u..%u",
-		(unsigned)period_size_min, (unsigned)period_size_max,
-		period_time_min, period_time_max);
-
-	if (ad->buffer_time > 0) {
-		buffer_time = ad->buffer_time;
-		cmd = "snd_pcm_hw_params_set_buffer_time_near";
-		err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm, hwparams,
-							     &buffer_time, NULL);
-		if (err < 0)
-			goto error;
-	} else {
-		err = snd_pcm_hw_params_get_buffer_time(hwparams, &buffer_time,
-							NULL);
-		if (err < 0)
-			buffer_time = 0;
-	}
-
-	if (period_time_ro == 0 && buffer_time >= 10000) {
-		period_time_ro = period_time = buffer_time / 4;
-
-		g_debug("default period_time = buffer_time/4 = %u/4 = %u",
-			buffer_time, period_time);
-	}
-
-	if (period_time_ro > 0) {
-		period_time = period_time_ro;
-		cmd = "snd_pcm_hw_params_set_period_time_near";
-		err = snd_pcm_hw_params_set_period_time_near(ad->pcm, hwparams,
-							     &period_time, NULL);
-		if (err < 0)
-			goto error;
-	}
-
-	cmd = "snd_pcm_hw_params";
-	err = snd_pcm_hw_params(ad->pcm, hwparams);
-	if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
-		period_time_ro = period_time_ro >> 1;
-		goto configure_hw;
-	} else if (err < 0)
-		goto error;
-	if (retry != MPD_ALSA_RETRY_NR)
-		g_debug("ALSA period_time set to %d\n", period_time);
-
-	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;
-
-	cmd = "snd_pcm_hw_params_get_period_size";
-	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
-						NULL);
-	if (err < 0)
-		goto error;
-
-	/* configure SW params */
-	snd_pcm_sw_params_alloca(&swparams);
-
-	cmd = "snd_pcm_sw_params_current";
-	err = snd_pcm_sw_params_current(ad->pcm, swparams);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_start_threshold";
-	err = snd_pcm_sw_params_set_start_threshold(ad->pcm, swparams,
-						    alsa_buffer_size -
-						    alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_avail_min";
-	err = snd_pcm_sw_params_set_avail_min(ad->pcm, swparams,
-					      alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params";
-	err = snd_pcm_sw_params(ad->pcm, swparams);
-	if (err < 0)
-		goto error;
-
-	g_debug("buffer_size=%u period_size=%u",
-		(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
-
-	if (alsa_period_size == 0)
-		/* this works around a SIGFPE bug that occurred when
-		   an ALSA driver indicated period_size==0; this
-		   caused a division by zero in alsa_play().  By using
-		   the fallback "1", we make sure that this won't
-		   happen again. */
-		alsa_period_size = 1;
-
-	ad->period_frames = alsa_period_size;
-	ad->period_position = 0;
-
-	return true;
-
-error:
-	g_set_error(error, alsa_output_quark(), err,
-		    "Error opening ALSA device \"%s\" (%s): %s",
-		    alsa_device(ad), cmd, snd_strerror(-err));
-	return false;
-}
-
-static bool
-alsa_setup_dsd(struct alsa_data *ad, struct audio_format *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);
-
-	/* pass 24 bit to alsa_setup() */
-
-	struct audio_format usb_format = *audio_format;
-	usb_format.format = SAMPLE_FORMAT_S24_P32;
-	usb_format.sample_rate /= 2;
-
-	const struct audio_format check = usb_format;
-
-	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
-	   samples left by 8 bit and leave the lower 8 bit cleared;
-	   the DSD-over-USB documentation does not specify whether
-	   this is legal, but there is anecdotical evidence that this
-	   is possible (and the only option for some devices) */
-	*shift8_r = usb_format.format == SAMPLE_FORMAT_S32;
-	if (usb_format.format == SAMPLE_FORMAT_S32)
-		usb_format.format = SAMPLE_FORMAT_S24_P32;
-
-	if (!audio_format_equals(&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));
-		return false;
-	}
-
-	return true;
-}
-
-static bool
-alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
-		  GError **error_r)
-{
-	bool shift8 = false, packed, reverse_endian;
-
-	const bool dsd_usb = ad->dsd_usb &&
-		audio_format->format == SAMPLE_FORMAT_DSD;
-	const bool success = dsd_usb
-		? alsa_setup_dsd(ad, audio_format,
-				 &shift8, &packed, &reverse_endian,
-				 error_r)
-		: alsa_setup(ad, audio_format, &packed, &reverse_endian,
-			     error_r);
-	if (!success)
-		return false;
-
-	pcm_export_open(&ad->export,
-			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)
-{
-	struct alsa_data *ad = (struct alsa_data *)ao;
-	int err;
-	bool success;
-
-	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",
-			    alsa_device(ad), snd_strerror(err));
-		return false;
-	}
-
-	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) {
-		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);
-
-	return true;
-}
-
-static int
-alsa_recover(struct alsa_data *ad, int err)
-{
-	if (err == -EPIPE) {
-		g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
-	} else if (err == -ESTRPIPE) {
-		g_debug("ALSA device \"%s\" was suspended\n", alsa_device(ad));
-	}
-
-	switch (snd_pcm_state(ad->pcm)) {
-	case SND_PCM_STATE_PAUSED:
-		err = snd_pcm_pause(ad->pcm, /* disable */ 0);
-		break;
-	case SND_PCM_STATE_SUSPENDED:
-		err = snd_pcm_resume(ad->pcm);
-		if (err == -EAGAIN)
-			return 0;
-		/* fall-through to snd_pcm_prepare: */
-	case SND_PCM_STATE_SETUP:
-	case SND_PCM_STATE_XRUN:
-		ad->period_position = 0;
-		err = snd_pcm_prepare(ad->pcm);
-		break;
-	case SND_PCM_STATE_DISCONNECTED:
-		break;
-	/* this is no error, so just keep running */
-	case SND_PCM_STATE_RUNNING:
-		err = 0;
-		break;
-	default:
-		/* unknown state, do nothing */
-		break;
-	}
-
-	return err;
-}
-
-static void
-alsa_drain(struct audio_output *ao)
-{
-	struct alsa_data *ad = (struct alsa_data *)ao;
-
-	if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
-		return;
-
-	if (ad->period_position > 0) {
-		/* generate some silence to finish the partial
-		   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);
-	}
-
-	snd_pcm_drain(ad->pcm);
-
-	ad->period_position = 0;
-}
-
-static void
-alsa_cancel(struct audio_output *ao)
-{
-	struct alsa_data *ad = (struct alsa_data *)ao;
-
-	ad->period_position = 0;
-
-	snd_pcm_drop(ad->pcm);
-}
-
-static void
-alsa_close(struct audio_output *ao)
-{
-	struct alsa_data *ad = (struct alsa_data *)ao;
-
-	snd_pcm_close(ad->pcm);
-}
-
-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;
-
-	assert(size % ad->in_frame_size == 0);
-
-	chunk = pcm_export(&ad->export, chunk, size, &size);
-
-	assert(size % ad->out_frame_size == 0);
-
-	size /= ad->out_frame_size;
-
-	while (true) {
-		snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
-		if (ret > 0) {
-			ad->period_position = (ad->period_position + ret)
-				% ad->period_frames;
-
-			size_t bytes_written = ret * ad->out_frame_size;
-			return pcm_export_source_size(&ad->export,
-						      bytes_written);
-		}
-
-		if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
-		    alsa_recover(ad, ret) < 0) {
-			g_set_error(error, alsa_output_quark(), errno,
-				    "%s", snd_strerror(-errno));
-			return 0;
-		}
-	}
-}
-
-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,
-};