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-rw-r--r--src/output/alsa_output_plugin.c683
1 files changed, 683 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..0bbe231fd
--- /dev/null
+++ b/src/output/alsa_output_plugin.c
@@ -0,0 +1,683 @@
+/*
+ * 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 "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 {
+ /** 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 void *
+alsa_init(G_GNUC_UNUSED const struct audio_format *audio_format,
+ const struct config_param *param,
+ G_GNUC_UNUSED GError **error)
+{
+ struct alsa_data *ad = alsa_data_new();
+
+ alsa_configure(ad, param);
+
+ return ad;
+}
+
+static void
+alsa_finish(void *data)
+{
+ struct alsa_data *ad = data;
+
+ 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_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;
+
+ default:
+ 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(void *data, struct audio_format *audio_format, GError **error)
+{
+ struct alsa_data *ad = data;
+ 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(void *data)
+{
+ struct alsa_data *ad = data;
+
+ 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(void *data)
+{
+ struct alsa_data *ad = data;
+
+ ad->period_position = 0;
+
+ snd_pcm_drop(ad->pcm);
+}
+
+static void
+alsa_close(void *data)
+{
+ struct alsa_data *ad = data;
+
+ snd_pcm_close(ad->pcm);
+}
+
+static size_t
+alsa_play(void *data, const void *chunk, size_t size, GError **error)
+{
+ struct alsa_data *ad = data;
+
+ 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 alsaPlugin = {
+ .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,
+};
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