1 files changed, 694 insertions, 0 deletions

diff --git a/src/output/alsa_output_plugin.c b/src/output/alsa_output_plugin.c
new file mode 100644
index 000000000..af6d1b184
--- /dev/null
+++ b/src/output/alsa_output_plugin.c
@@ -0,0 +1,694 @@
+/*
+ * 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 <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;
+
+	/** the configured name of the ALSA device; NULL for the
+	    default device */
+	char *device;
+
+	/** use memory mapped I/O? */
+	bool use_mmap;
+
+	/** 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 */
+	size_t 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_data_free(struct alsa_data *ad)
+{
+	g_free(ad->device);
+	g_free(ad);
+}
+
+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->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);
+	alsa_data_free(ad);
+
+	/* free libasound's config cache */
+	snd_config_update_free_global();
+}
+
+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:
+		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_S24:
+		return G_BYTE_ORDER == G_BIG_ENDIAN
+			? SND_PCM_FORMAT_S24_3BE
+			: SND_PCM_FORMAT_S24_3LE;
+
+	case SAMPLE_FORMAT_S32:
+		return SND_PCM_FORMAT_S32;
+	}
+
+	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;
+	}
+}
+
+/**
+ * Attempts to configure the specified sample format.
+ */
+static int
+alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+		       struct audio_format *audio_format,
+		       enum sample_format sample_format)
+{
+	snd_pcm_format_t alsa_format = get_bitformat(sample_format);
+	if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
+		return -EINVAL;
+
+	int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format);
+	if (err == 0)
+		audio_format->format = sample_format;
+
+	return err;
+}
+
+/**
+ * Attempts to configure the specified sample format with reversed
+ * host byte order.
+ */
+static int
+alsa_output_try_reverse(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+		       struct audio_format *audio_format,
+		       enum sample_format sample_format)
+{
+	snd_pcm_format_t alsa_format =
+		byteswap_bitformat(get_bitformat(sample_format));
+	if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
+		return -EINVAL;
+
+	int err = snd_pcm_hw_params_set_format(pcm, hwparams, alsa_format);
+	if (err == 0) {
+		audio_format->format = sample_format;
+		audio_format->reverse_endian = 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_both(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
+			    struct audio_format *audio_format,
+			    enum sample_format sample_format)
+{
+	int err = alsa_output_try_format(pcm, hwparams, audio_format,
+					 sample_format);
+	if (err == -EINVAL)
+		err = alsa_output_try_reverse(pcm, hwparams, audio_format,
+					      sample_format);
+
+	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)
+{
+	/* try the input format first */
+
+	int err = alsa_output_try_format_both(pcm, hwparams, audio_format,
+					      audio_format->format);
+	if (err != -EINVAL)
+		return err;
+
+	/* if unsupported by the hardware, try other formats */
+
+	static const enum sample_format probe_formats[] = {
+		SAMPLE_FORMAT_S24_P32,
+		SAMPLE_FORMAT_S32,
+		SAMPLE_FORMAT_S24,
+		SAMPLE_FORMAT_S16,
+		SAMPLE_FORMAT_S8,
+		SAMPLE_FORMAT_UNDEFINED,
+	};
+
+	for (unsigned i = 0; probe_formats[i] != SAMPLE_FORMAT_UNDEFINED; ++i) {
+		if (probe_formats[i] == audio_format->format)
+			continue;
+
+		err = alsa_output_try_format_both(pcm, hwparams, audio_format,
+						  probe_formats[i]);
+		if (err != -EINVAL)
+			return err;
+	}
+
+	return -EINVAL;
+}
+
+/**
+ * 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,
+	   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);
+	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;
+	}
+
+	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_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;
+	}
+
+	success = alsa_setup(ad, audio_format, error);
+	if (!success) {
+		snd_pcm_close(ad->pcm);
+		return false;
+	}
+
+	ad->frame_size = audio_format_frame_size(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->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;
+
+	size /= ad->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;
+			return ret * ad->frame_size;
+		}
+
+		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,
+	.open = alsa_open,
+	.play = alsa_play,
+	.drain = alsa_drain,
+	.cancel = alsa_cancel,
+	.close = alsa_close,
+
+	.mixer_plugin = &alsa_mixer_plugin,
+};