diff options
Diffstat (limited to 'src/output/alsa_plugin.c')
| -rw-r--r-- | src/output/alsa_plugin.c | 683 |
1 files changed, 0 insertions, 683 deletions
diff --git a/src/output/alsa_plugin.c b/src/output/alsa_plugin.c deleted file mode 100644 index 0bbe231fd..000000000 --- a/src/output/alsa_plugin.c +++ /dev/null @@ -1,683 +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 "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(¶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(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, -}; |