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authorMax Kellermann <max@duempel.org>2008-10-26 11:29:44 +0100
committerMax Kellermann <max@duempel.org>2008-10-26 11:29:44 +0100
commitece8c1347caae044db0fc4565ed3db6028d7b90e (patch)
tree27ec1bee18cd10b6997a9a44a4043dd4a4449153 /src/audioOutputs/audioOutput_alsa.c
parente11355f47d545fe523b019481415b1347aecd4bd (diff)
downloadmpd-ece8c1347caae044db0fc4565ed3db6028d7b90e.tar.gz
mpd-ece8c1347caae044db0fc4565ed3db6028d7b90e.tar.xz
mpd-ece8c1347caae044db0fc4565ed3db6028d7b90e.zip
renamed src/audioOutputs/ to src/output/
Again, no CamelCase in the directory name.
Diffstat (limited to 'src/audioOutputs/audioOutput_alsa.c')
-rw-r--r--src/audioOutputs/audioOutput_alsa.c444
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 */