renamed src/audioOutputs/ to src/output/
Again, no CamelCase in the directory name.
This commit is contained in:
Max Kellermann
@@ -0,0 +1,444 @@
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/* the Music Player Daemon (MPD)
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* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
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* This project's homepage is: http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "../output_api.h"
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#ifdef HAVE_ALSA
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#define ALSA_PCM_NEW_HW_PARAMS_API
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#define ALSA_PCM_NEW_SW_PARAMS_API
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static const char default_device[] = "default";
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#define MPD_ALSA_RETRY_NR 5
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#include "../utils.h"
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#include "../log.h"
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#include <alsa/asoundlib.h>
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typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
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snd_pcm_uframes_t size);
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typedef struct _AlsaData {
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const char *device;
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/** the mode flags passed to snd_pcm_open */
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int mode;
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snd_pcm_t *pcmHandle;
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alsa_writei_t *writei;
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unsigned int buffer_time;
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unsigned int period_time;
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int sampleSize;
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int useMmap;
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} AlsaData;
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static AlsaData *newAlsaData(void)
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{
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AlsaData *ret = xmalloc(sizeof(AlsaData));
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ret->device = default_device;
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ret->mode = 0;
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ret->pcmHandle = NULL;
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ret->writei = snd_pcm_writei;
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ret->useMmap = 0;
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ret->buffer_time = 0;
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ret->period_time = 0;
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return ret;
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}
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static void freeAlsaData(AlsaData * ad)
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{
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if (ad->device && ad->device != default_device)
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xfree(ad->device);
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free(ad);
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}
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static void
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alsa_configure(AlsaData *ad, ConfigParam *param)
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{
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BlockParam *bp;
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if ((bp = getBlockParam(param, "device")))
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ad->device = xstrdup(bp->value);
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ad->useMmap = getBoolBlockParam(param, "use_mmap", 1);
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if (ad->useMmap == CONF_BOOL_UNSET)
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ad->useMmap = 0;
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if ((bp = getBlockParam(param, "buffer_time")))
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ad->buffer_time = atoi(bp->value);
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if ((bp = getBlockParam(param, "period_time")))
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ad->period_time = atoi(bp->value);
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#ifdef SND_PCM_NO_AUTO_RESAMPLE
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if (!getBoolBlockParam(param, "auto_resample", true))
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ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
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#endif
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#ifdef SND_PCM_NO_AUTO_CHANNELS
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if (!getBoolBlockParam(param, "auto_channels", true))
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ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
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#endif
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#ifdef SND_PCM_NO_AUTO_FORMAT
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if (!getBoolBlockParam(param, "auto_format", true))
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ad->mode |= SND_PCM_NO_AUTO_FORMAT;
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#endif
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}
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static void *alsa_initDriver(mpd_unused struct audio_output *ao,
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mpd_unused const struct audio_format *audio_format,
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ConfigParam * param)
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{
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/* no need for pthread_once thread-safety when reading config */
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static int free_global_registered;
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AlsaData *ad = newAlsaData();
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if (!free_global_registered) {
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atexit((void(*)(void))snd_config_update_free_global);
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free_global_registered = 1;
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}
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if (param)
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alsa_configure(ad, param);
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return ad;
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}
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static void alsa_finishDriver(void *data)
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{
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AlsaData *ad = data;
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freeAlsaData(ad);
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}
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static int alsa_testDefault(void)
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{
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snd_pcm_t *handle;
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int ret = snd_pcm_open(&handle, default_device,
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SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
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if (ret) {
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WARNING("Error opening default ALSA device: %s\n",
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snd_strerror(-ret));
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return -1;
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} else
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snd_pcm_close(handle);
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return 0;
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}
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static snd_pcm_format_t get_bitformat(const struct audio_format *af)
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{
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switch (af->bits) {
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case 8: return SND_PCM_FORMAT_S8;
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case 16: return SND_PCM_FORMAT_S16;
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case 24: return SND_PCM_FORMAT_S24;
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case 32: return SND_PCM_FORMAT_S32;
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}
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return SND_PCM_FORMAT_UNKNOWN;
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}
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static int alsa_openDevice(void *data, struct audio_format *audioFormat)
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{
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AlsaData *ad = data;
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snd_pcm_format_t bitformat;
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snd_pcm_hw_params_t *hwparams;
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snd_pcm_sw_params_t *swparams;
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unsigned int sample_rate = audioFormat->sample_rate;
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unsigned int channels = audioFormat->channels;
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snd_pcm_uframes_t alsa_buffer_size;
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snd_pcm_uframes_t alsa_period_size;
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int err;
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const char *cmd = NULL;
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int retry = MPD_ALSA_RETRY_NR;
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unsigned int period_time, period_time_ro;
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unsigned int buffer_time;
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if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
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ERROR("ALSA device \"%s\" doesn't support %u bit audio\n",
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ad->device, audioFormat->bits);
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err = snd_pcm_open(&ad->pcmHandle, ad->device,
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SND_PCM_STREAM_PLAYBACK, ad->mode);
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if (err < 0) {
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ad->pcmHandle = NULL;
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goto error;
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}
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period_time_ro = period_time = ad->period_time;
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configure_hw:
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/* configure HW params */
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snd_pcm_hw_params_alloca(&hwparams);
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cmd = "snd_pcm_hw_params_any";
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err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
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if (err < 0)
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goto error;
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if (ad->useMmap) {
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err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
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SND_PCM_ACCESS_MMAP_INTERLEAVED);
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if (err < 0) {
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ERROR("Cannot set mmap'ed mode on ALSA device \"%s\": "
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" %s\n", ad->device, snd_strerror(-err));
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ERROR("Falling back to direct write mode\n");
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ad->useMmap = 0;
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} else
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ad->writei = snd_pcm_mmap_writei;
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}
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if (!ad->useMmap) {
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cmd = "snd_pcm_hw_params_set_access";
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err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
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SND_PCM_ACCESS_RW_INTERLEAVED);
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if (err < 0)
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goto error;
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ad->writei = snd_pcm_writei;
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}
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err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
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if (err == -EINVAL && audioFormat->bits != 16) {
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/* fall back to 16 bit, let pcm_utils.c do the conversion */
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err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams,
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SND_PCM_FORMAT_S16);
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if (err == 0) {
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DEBUG("ALSA device \"%s\": converting %u bit to 16 bit\n",
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ad->device, audioFormat->bits);
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audioFormat->bits = 16;
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}
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}
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if (err < 0) {
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ERROR("ALSA device \"%s\" does not support %u bit audio: "
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"%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
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goto fail;
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}
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err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
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&channels);
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if (err < 0) {
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ERROR("ALSA device \"%s\" does not support %i channels: "
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"%s\n", ad->device, (int)audioFormat->channels,
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snd_strerror(-err));
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goto fail;
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}
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audioFormat->channels = (int8_t)channels;
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err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
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&sample_rate, NULL);
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if (err < 0 || sample_rate == 0) {
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ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
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ad->device, audioFormat->sample_rate);
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goto fail;
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}
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audioFormat->sample_rate = sample_rate;
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if (ad->buffer_time > 0) {
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buffer_time = ad->buffer_time;
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cmd = "snd_pcm_hw_params_set_buffer_time_near";
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err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
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&buffer_time, NULL);
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if (err < 0)
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goto error;
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}
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if (period_time_ro > 0) {
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period_time = period_time_ro;
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cmd = "snd_pcm_hw_params_set_period_time_near";
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err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
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&period_time, NULL);
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if (err < 0)
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goto error;
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}
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cmd = "snd_pcm_hw_params";
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err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
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if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
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period_time_ro = period_time_ro >> 1;
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goto configure_hw;
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} else if (err < 0)
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goto error;
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if (retry != MPD_ALSA_RETRY_NR)
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DEBUG("ALSA period_time set to %d\n", period_time);
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cmd = "snd_pcm_hw_params_get_buffer_size";
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err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
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if (err < 0)
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goto error;
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cmd = "snd_pcm_hw_params_get_period_size";
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err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
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NULL);
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if (err < 0)
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goto error;
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/* configure SW params */
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snd_pcm_sw_params_alloca(&swparams);
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cmd = "snd_pcm_sw_params_current";
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err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
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if (err < 0)
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goto error;
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cmd = "snd_pcm_sw_params_set_start_threshold";
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err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
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alsa_buffer_size -
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alsa_period_size);
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if (err < 0)
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goto error;
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cmd = "snd_pcm_sw_params_set_avail_min";
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err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
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alsa_period_size);
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if (err < 0)
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goto error;
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cmd = "snd_pcm_sw_params";
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err = snd_pcm_sw_params(ad->pcmHandle, swparams);
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if (err < 0)
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goto error;
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ad->sampleSize = audio_format_frame_size(audioFormat);
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DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
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"%u Hz\n", ad->device, audioFormat->bits,
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channels, sample_rate);
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return 0;
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error:
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if (cmd) {
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ERROR("Error opening ALSA device \"%s\" (%s): %s\n",
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ad->device, cmd, snd_strerror(-err));
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} else {
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ERROR("Error opening ALSA device \"%s\": %s\n", ad->device,
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snd_strerror(-err));
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}
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fail:
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if (ad->pcmHandle)
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snd_pcm_close(ad->pcmHandle);
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ad->pcmHandle = NULL;
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return -1;
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}
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static int alsa_errorRecovery(AlsaData * ad, int err)
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{
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if (err == -EPIPE) {
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DEBUG("Underrun on ALSA device \"%s\"\n", ad->device);
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} else if (err == -ESTRPIPE) {
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DEBUG("ALSA device \"%s\" was suspended\n", ad->device);
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}
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switch (snd_pcm_state(ad->pcmHandle)) {
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case SND_PCM_STATE_PAUSED:
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err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
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break;
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case SND_PCM_STATE_SUSPENDED:
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err = snd_pcm_resume(ad->pcmHandle);
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if (err == -EAGAIN)
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return 0;
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/* fall-through to snd_pcm_prepare: */
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case SND_PCM_STATE_SETUP:
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case SND_PCM_STATE_XRUN:
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err = snd_pcm_prepare(ad->pcmHandle);
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break;
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case SND_PCM_STATE_DISCONNECTED:
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/* so alsa_closeDevice won't try to drain: */
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snd_pcm_close(ad->pcmHandle);
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ad->pcmHandle = NULL;
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break;
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/* this is no error, so just keep running */
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case SND_PCM_STATE_RUNNING:
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err = 0;
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break;
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default:
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/* unknown state, do nothing */
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break;
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}
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return err;
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}
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static void alsa_dropBufferedAudio(void *data)
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{
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AlsaData *ad = data;
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alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
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}
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static void alsa_closeDevice(void *data)
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{
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AlsaData *ad = data;
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if (ad->pcmHandle) {
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if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
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snd_pcm_drain(ad->pcmHandle);
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}
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snd_pcm_close(ad->pcmHandle);
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ad->pcmHandle = NULL;
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}
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}
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static int alsa_playAudio(void *data, const char *playChunk, size_t size)
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{
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AlsaData *ad = data;
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int ret;
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size /= ad->sampleSize;
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while (size > 0) {
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ret = ad->writei(ad->pcmHandle, playChunk, size);
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if (ret == -EAGAIN || ret == -EINTR)
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continue;
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if (ret < 0) {
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if (alsa_errorRecovery(ad, ret) < 0) {
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ERROR("closing ALSA device \"%s\" due to write "
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"error: %s\n", ad->device,
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snd_strerror(-errno));
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alsa_closeDevice(ad);
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return -1;
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}
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continue;
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}
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playChunk += ret * ad->sampleSize;
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size -= ret;
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}
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return 0;
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}
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const struct audio_output_plugin alsaPlugin = {
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.name = "alsa",
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.test_default_device = alsa_testDefault,
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.init = alsa_initDriver,
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.finish = alsa_finishDriver,
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.open = alsa_openDevice,
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.play = alsa_playAudio,
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.cancel = alsa_dropBufferedAudio,
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.close = alsa_closeDevice,
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};
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#else /* HAVE ALSA */
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DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
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#endif /* HAVE_ALSA */
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