mirror of
https://github.com/CTCaer/RetroArch.git
synced 2024-12-26 04:36:01 +00:00
270 lines
6.3 KiB
C
270 lines
6.3 KiB
C
/* RetroArch - A frontend for libretro.
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* Copyright (C) 2010-2013 - Hans-Kristian Arntzen
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*
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* RetroArch is free software: you can redistribute it and/or modify it under the terms
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* of the GNU General Public License as published by the Free Software Found-
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* ation, either version 3 of the License, or (at your option) any later version.
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*
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* RetroArch is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY;
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* without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR
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* PURPOSE. See the GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along with RetroArch.
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* If not, see <http://www.gnu.org/licenses/>.
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*/
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#include "../general.h"
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#include "../driver.h"
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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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#include <sys/asoundlib.h>
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typedef struct alsa
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{
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snd_pcm_t *pcm;
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size_t buffer_size;
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bool nonblock;
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bool has_float;
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bool can_pause;
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bool is_paused;
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} alsa_t;
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static void *alsa_qsa_init(const char *device, unsigned rate, unsigned latency)
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{
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(void)device;
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(void)rate;
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(void)latency;
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int err, card, dev;
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snd_pcm_channel_params_t params = {0};
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snd_pcm_channel_setup_t setup = {0};
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alsa_t *alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
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if (!alsa)
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return NULL;
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if ((err = snd_pcm_open_preferred(&alsa->pcm, &card, &dev,
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SND_PCM_OPEN_PLAYBACK)) < 0)
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{
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RARCH_ERR("[ALSA QSA]: Audio open error: %s\n", snd_strerror(err));
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goto error;
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}
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if((err = snd_pcm_nonblock_mode(alsa->pcm, 1)) < 0)
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{
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RARCH_ERR("[ALSA QSA]: Can't set blocking mode: %s\n", snd_strerror(err));
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goto error;
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}
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if ((err = snd_pcm_plugin_set_disable (alsa->pcm, PLUGIN_DISABLE_MMAP)) < 0)
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{
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RARCH_ERR("[ALSA QSA]: Can't disable MMAP plugin: %s\n", snd_strerror(err));
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goto error;
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}
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memset(¶ms, 0, sizeof(params));
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params.channel = SND_PCM_CHANNEL_PLAYBACK;
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params.mode = SND_PCM_MODE_BLOCK;
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params.format.interleave = 1;
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params.format.format = SND_PCM_SFMT_S16_LE;
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params.format.rate = rate;
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params.format.voices = 2;
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params.start_mode = SND_PCM_START_DATA;
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params.stop_mode = SND_PCM_STOP_STOP;
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params.buf.block.frag_size = 4096;
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params.buf.block.frags_min = 8;
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params.buf.block.frags_max = 16;
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if ((err = snd_pcm_plugin_params(alsa->pcm, ¶ms)) < 0)
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{
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RARCH_ERR("[ALSA QSA]: Channel Parameter Error: %s\n", snd_strerror(err));
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goto error;
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}
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setup.channel = SND_PCM_CHANNEL_PLAYBACK;
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if ((err = snd_pcm_plugin_setup(alsa->pcm, &setup)) < 0)
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{
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RARCH_ERR("[ALSA QSA]: Channel Parameter Read Back Error: %s\n", snd_strerror(err));
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goto error;
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}
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alsa->buffer_size = setup.buf.block.frag_size * (setup.buf.block.frags_max+1);
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RARCH_LOG("[ALSA QSA]: buffer size: %d bytes\n", alsa->buffer_size);
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if ((err = snd_pcm_plugin_prepare(alsa->pcm, SND_PCM_CHANNEL_PLAYBACK)) < 0)
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{
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RARCH_ERR("[ALSA QSA]: Channel Prepare Error: %s\n", snd_strerror(err));
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goto error;
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}
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alsa->has_float = false;
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#ifdef HAVE_BB10
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alsa->can_pause = true;
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#endif
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RARCH_LOG("[ALSA QSA]: Can pause: %s.\n", alsa->can_pause ? "yes" : "no");
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return alsa;
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error:
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return (void*)-1;
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}
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static ssize_t alsa_qsa_write(void *data, const void *buf, size_t size)
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{
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int written, status;
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alsa_t *alsa = (alsa_t*)data;
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snd_pcm_channel_status_t cstatus = {0};
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written = snd_pcm_plugin_write(alsa->pcm, buf, size);
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if (written != size)
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{
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/* Check if samples playback got stuck somewhere in hardware or in */
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/* the audio device driver */
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if ((errno == EAGAIN) && (written == 0))
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return 0;
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if ((errno == EINVAL) || (errno == EIO))
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{
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cstatus.channel = SND_PCM_CHANNEL_PLAYBACK;
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status = snd_pcm_plugin_status(alsa->pcm, &cstatus);
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if (status > 0)
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return 0;
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if ((cstatus.status == SND_PCM_STATUS_UNDERRUN) ||
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(cstatus.status == SND_PCM_STATUS_READY))
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{
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status = snd_pcm_plugin_prepare(alsa->pcm, SND_PCM_CHANNEL_PLAYBACK);
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if (status < 0)
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return 0;
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}
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}
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}
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return written;
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}
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static bool alsa_qsa_stop(void *data)
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{
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alsa_t *alsa = (alsa_t*)data;
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if (alsa->can_pause && !alsa->is_paused)
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{
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#ifdef HAVE_BB10
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if (snd_pcm_playback_pause(alsa->pcm) == 0)
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#else
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if (snd_pcm_plugin_flush(alsa->pcm, SND_PCM_CHANNEL_PLAYBACK) == 0)
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#endif
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{
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alsa->is_paused = true;
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return true;
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}
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else
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return false;
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}
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else
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return true;
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}
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static void alsa_qsa_set_nonblock_state(void *data, bool state)
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{
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alsa_t *alsa = (alsa_t*)data;
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int err;
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if((err = snd_pcm_nonblock_mode(alsa->pcm, state)) < 0)
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{
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RARCH_ERR("Can't set blocking mode to %d: %s\n", state,
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snd_strerror(err));
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return;
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}
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alsa->nonblock = state;
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}
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static bool alsa_qsa_start(void *data)
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{
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alsa_t *alsa = (alsa_t*)data;
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if (alsa->can_pause && alsa->is_paused)
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{
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#ifdef HAVE_BB10
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int ret = snd_pcm_playback_resume(alsa->pcm);
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#else
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int ret = 0;
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#endif
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if (ret < 0)
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{
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RARCH_ERR("[ALSA QSA]: Failed to unpause: %s.\n", snd_strerror(ret));
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return false;
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}
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else
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{
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alsa->is_paused = false;
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return true;
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}
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}
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else
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return true;
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}
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static bool alsa_qsa_use_float(void *data)
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{
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alsa_t *alsa = (alsa_t*)data;
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return alsa->has_float;
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}
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static void alsa_qsa_free(void *data)
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{
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alsa_t *alsa = (alsa_t*)data;
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if (alsa)
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{
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if (alsa->pcm)
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{
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snd_pcm_close(alsa->pcm);
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alsa->pcm = NULL;
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}
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free(alsa);
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}
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}
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static size_t alsa_qsa_write_avail(void *data)
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{
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alsa_t *alsa = (alsa_t*)data;
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snd_pcm_channel_status_t status = {0};
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status.channel = SND_PCM_CHANNEL_PLAYBACK;
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snd_pcm_plugin_status(alsa->pcm, &status);
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return status.free;
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}
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static size_t alsa_qsa_buffer_size(void *data)
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{
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alsa_t *alsa = (alsa_t*)data;
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return alsa->buffer_size;
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}
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const audio_driver_t audio_alsa = {
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alsa_qsa_init,
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alsa_qsa_write,
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alsa_qsa_stop,
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alsa_qsa_start,
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alsa_qsa_set_nonblock_state,
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alsa_qsa_free,
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alsa_qsa_use_float,
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"alsa",
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alsa_qsa_write_avail,
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alsa_qsa_buffer_size,
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};
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