diff options
Diffstat (limited to 'src/audioOutputs/audioOutput_alsa.c')
| -rw-r--r-- | src/audioOutputs/audioOutput_alsa.c | 444 |
1 files changed, 0 insertions, 444 deletions
diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c deleted file mode 100644 index 1845f1b76..000000000 --- a/src/audioOutputs/audioOutput_alsa.c +++ /dev/null @@ -1,444 +0,0 @@ -/* the Music Player Daemon (MPD) - * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com) - * This project's homepage is: 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -#include "../output_api.h" - -#ifdef HAVE_ALSA - -#define ALSA_PCM_NEW_HW_PARAMS_API -#define ALSA_PCM_NEW_SW_PARAMS_API - -static const char default_device[] = "default"; - -#define MPD_ALSA_RETRY_NR 5 - -#include "../utils.h" -#include "../log.h" - -#include <alsa/asoundlib.h> - -typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer, - snd_pcm_uframes_t size); - -typedef struct _AlsaData { - const char *device; - - /** the mode flags passed to snd_pcm_open */ - int mode; - - snd_pcm_t *pcmHandle; - alsa_writei_t *writei; - unsigned int buffer_time; - unsigned int period_time; - int sampleSize; - int useMmap; -} AlsaData; - -static AlsaData *newAlsaData(void) -{ - AlsaData *ret = xmalloc(sizeof(AlsaData)); - - ret->device = default_device; - ret->mode = 0; - ret->pcmHandle = NULL; - ret->writei = snd_pcm_writei; - ret->useMmap = 0; - ret->buffer_time = 0; - ret->period_time = 0; - - return ret; -} - -static void freeAlsaData(AlsaData * ad) -{ - if (ad->device && ad->device != default_device) - xfree(ad->device); - free(ad); -} - -static void -alsa_configure(AlsaData *ad, ConfigParam *param) -{ - BlockParam *bp; - - if ((bp = getBlockParam(param, "device"))) - ad->device = xstrdup(bp->value); - ad->useMmap = getBoolBlockParam(param, "use_mmap", 1); - if (ad->useMmap == CONF_BOOL_UNSET) - ad->useMmap = 0; - if ((bp = getBlockParam(param, "buffer_time"))) - ad->buffer_time = atoi(bp->value); - if ((bp = getBlockParam(param, "period_time"))) - ad->period_time = atoi(bp->value); - -#ifdef SND_PCM_NO_AUTO_RESAMPLE - if (!getBoolBlockParam(param, "auto_resample", true)) - ad->mode |= SND_PCM_NO_AUTO_RESAMPLE; -#endif - -#ifdef SND_PCM_NO_AUTO_CHANNELS - if (!getBoolBlockParam(param, "auto_channels", true)) - ad->mode |= SND_PCM_NO_AUTO_CHANNELS; -#endif - -#ifdef SND_PCM_NO_AUTO_FORMAT - if (!getBoolBlockParam(param, "auto_format", true)) - ad->mode |= SND_PCM_NO_AUTO_FORMAT; -#endif -} - -static void *alsa_initDriver(mpd_unused struct audio_output *ao, - mpd_unused const struct audio_format *audio_format, - ConfigParam * param) -{ - /* no need for pthread_once thread-safety when reading config */ - static int free_global_registered; - AlsaData *ad = newAlsaData(); - - if (!free_global_registered) { - atexit((void(*)(void))snd_config_update_free_global); - free_global_registered = 1; - } - - if (param) - alsa_configure(ad, param); - - return ad; -} - -static void alsa_finishDriver(void *data) -{ - AlsaData *ad = data; - - freeAlsaData(ad); -} - -static int alsa_testDefault(void) -{ - snd_pcm_t *handle; - - int ret = snd_pcm_open(&handle, default_device, - SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK); - if (ret) { - WARNING("Error opening default ALSA device: %s\n", - snd_strerror(-ret)); - return -1; - } else - snd_pcm_close(handle); - - return 0; -} - -static snd_pcm_format_t get_bitformat(const struct audio_format *af) -{ - switch (af->bits) { - case 8: return SND_PCM_FORMAT_S8; - case 16: return SND_PCM_FORMAT_S16; - case 24: return SND_PCM_FORMAT_S24; - case 32: return SND_PCM_FORMAT_S32; - } - return SND_PCM_FORMAT_UNKNOWN; -} - -static int alsa_openDevice(void *data, struct audio_format *audioFormat) -{ - AlsaData *ad = data; - snd_pcm_format_t bitformat; - snd_pcm_hw_params_t *hwparams; - snd_pcm_sw_params_t *swparams; - unsigned int sample_rate = audioFormat->sample_rate; - unsigned int channels = audioFormat->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; - - if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN) - ERROR("ALSA device \"%s\" doesn't support %u bit audio\n", - ad->device, audioFormat->bits); - - err = snd_pcm_open(&ad->pcmHandle, ad->device, - SND_PCM_STREAM_PLAYBACK, ad->mode); - if (err < 0) { - ad->pcmHandle = NULL; - goto error; - } - - 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->pcmHandle, hwparams); - if (err < 0) - goto error; - - if (ad->useMmap) { - err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, - SND_PCM_ACCESS_MMAP_INTERLEAVED); - if (err < 0) { - ERROR("Cannot set mmap'ed mode on ALSA device \"%s\": " - " %s\n", ad->device, snd_strerror(-err)); - ERROR("Falling back to direct write mode\n"); - ad->useMmap = 0; - } else - ad->writei = snd_pcm_mmap_writei; - } - - if (!ad->useMmap) { - cmd = "snd_pcm_hw_params_set_access"; - err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams, - SND_PCM_ACCESS_RW_INTERLEAVED); - if (err < 0) - goto error; - ad->writei = snd_pcm_writei; - } - - err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat); - if (err == -EINVAL && audioFormat->bits != 16) { - /* fall back to 16 bit, let pcm_utils.c do the conversion */ - err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, - SND_PCM_FORMAT_S16); - if (err == 0) { - DEBUG("ALSA device \"%s\": converting %u bit to 16 bit\n", - ad->device, audioFormat->bits); - audioFormat->bits = 16; - } - } - - if (err < 0) { - ERROR("ALSA device \"%s\" does not support %u bit audio: " - "%s\n", ad->device, audioFormat->bits, snd_strerror(-err)); - goto fail; - } - - err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams, - &channels); - if (err < 0) { - ERROR("ALSA device \"%s\" does not support %i channels: " - "%s\n", ad->device, (int)audioFormat->channels, - snd_strerror(-err)); - goto fail; - } - audioFormat->channels = (int8_t)channels; - - err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams, - &sample_rate, NULL); - if (err < 0 || sample_rate == 0) { - ERROR("ALSA device \"%s\" does not support %u Hz audio\n", - ad->device, audioFormat->sample_rate); - goto fail; - } - audioFormat->sample_rate = sample_rate; - - 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->pcmHandle, hwparams, - &buffer_time, NULL); - if (err < 0) - goto error; - } - - 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->pcmHandle, hwparams, - &period_time, NULL); - if (err < 0) - goto error; - } - - cmd = "snd_pcm_hw_params"; - err = snd_pcm_hw_params(ad->pcmHandle, 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) - 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->pcmHandle, swparams); - if (err < 0) - goto error; - - cmd = "snd_pcm_sw_params_set_start_threshold"; - err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, 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->pcmHandle, swparams, - alsa_period_size); - if (err < 0) - goto error; - - cmd = "snd_pcm_sw_params"; - err = snd_pcm_sw_params(ad->pcmHandle, swparams); - if (err < 0) - goto error; - - ad->sampleSize = audio_format_frame_size(audioFormat); - - DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at " - "%u Hz\n", ad->device, audioFormat->bits, - channels, sample_rate); - - return 0; - -error: - if (cmd) { - ERROR("Error opening ALSA device \"%s\" (%s): %s\n", - ad->device, cmd, snd_strerror(-err)); - } else { - ERROR("Error opening ALSA device \"%s\": %s\n", ad->device, - snd_strerror(-err)); - } -fail: - if (ad->pcmHandle) - snd_pcm_close(ad->pcmHandle); - ad->pcmHandle = NULL; - return -1; -} - -static int alsa_errorRecovery(AlsaData * ad, int err) -{ - if (err == -EPIPE) { - DEBUG("Underrun on ALSA device \"%s\"\n", ad->device); - } else if (err == -ESTRPIPE) { - DEBUG("ALSA device \"%s\" was suspended\n", ad->device); - } - - switch (snd_pcm_state(ad->pcmHandle)) { - case SND_PCM_STATE_PAUSED: - err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0); - break; - case SND_PCM_STATE_SUSPENDED: - err = snd_pcm_resume(ad->pcmHandle); - if (err == -EAGAIN) - return 0; - /* fall-through to snd_pcm_prepare: */ - case SND_PCM_STATE_SETUP: - case SND_PCM_STATE_XRUN: - err = snd_pcm_prepare(ad->pcmHandle); - break; - case SND_PCM_STATE_DISCONNECTED: - /* so alsa_closeDevice won't try to drain: */ - snd_pcm_close(ad->pcmHandle); - ad->pcmHandle = NULL; - 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_dropBufferedAudio(void *data) -{ - AlsaData *ad = data; - - alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle)); -} - -static void alsa_closeDevice(void *data) -{ - AlsaData *ad = data; - - if (ad->pcmHandle) { - if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) { - snd_pcm_drain(ad->pcmHandle); - } - snd_pcm_close(ad->pcmHandle); - ad->pcmHandle = NULL; - } -} - -static int alsa_playAudio(void *data, const char *playChunk, size_t size) -{ - AlsaData *ad = data; - int ret; - - size /= ad->sampleSize; - - while (size > 0) { - ret = ad->writei(ad->pcmHandle, playChunk, size); - - if (ret == -EAGAIN || ret == -EINTR) - continue; - - if (ret < 0) { - if (alsa_errorRecovery(ad, ret) < 0) { - ERROR("closing ALSA device \"%s\" due to write " - "error: %s\n", ad->device, - snd_strerror(-errno)); - alsa_closeDevice(ad); - return -1; - } - continue; - } - - playChunk += ret * ad->sampleSize; - size -= ret; - } - - return 0; -} - -const struct audio_output_plugin alsaPlugin = { - .name = "alsa", - .test_default_device = alsa_testDefault, - .init = alsa_initDriver, - .finish = alsa_finishDriver, - .open = alsa_openDevice, - .play = alsa_playAudio, - .cancel = alsa_dropBufferedAudio, - .close = alsa_closeDevice, -}; - -#else /* HAVE ALSA */ - -DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin) -#endif /* HAVE_ALSA */ |