diff options
Diffstat (limited to 'src/output/alsa_output_plugin.c')
| -rw-r--r-- | src/output/alsa_output_plugin.c | 819 |
1 files changed, 819 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..d8b184273 --- /dev/null +++ b/src/output/alsa_output_plugin.c @@ -0,0 +1,819 @@ +/* + * 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(¶ms); + 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, +}; |