RetroArch/blackberry-qnx/alsa_qsa.c

/*  RetroArch - A frontend for libretro.
 *  Copyright (C) 2010-2013 - Hans-Kristian Arntzen
 * 
 *  RetroArch 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 Found-
 *  ation, either version 3 of the License, or (at your option) any later version.
 *
 *  RetroArch 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 RetroArch.
 *  If not, see <http://www.gnu.org/licenses/>.
 */

#include "../general.h"
#include "../driver.h"

#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#include <sys/asoundlib.h>

typedef struct alsa
{
   snd_pcm_t *pcm;
   size_t buffer_size;
   snd_pcm_channel_status_t status;
   bool nonblock;
   bool has_float;
   bool can_pause;
   bool is_paused;
} alsa_t;

static void *alsa_qsa_init(const char *device, unsigned rate, unsigned latency)
{
   (void)device;
   (void)rate;
   (void)latency;

   int err, card, dev;
   snd_pcm_channel_params_t params;
   snd_pcm_channel_setup_t setup;

   alsa_t *alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
   if (!alsa)
      return NULL;

   if ((err = snd_pcm_open_preferred(&alsa->pcm, &card, &dev,
               SND_PCM_OPEN_PLAYBACK)) < 0)
   {
      RARCH_ERR("Audio open error: %s\n", snd_strerror(err));
      goto error;
   }

   if((err = snd_pcm_nonblock_mode(alsa->pcm, 1)) < 0)
   {
      RARCH_ERR("Can't set blocking mode: %s\n", snd_strerror(err));
      goto error;
   }

   if ((err = snd_pcm_plugin_set_disable (alsa->pcm, PLUGIN_DISABLE_MMAP)) < 0)
   {
      RARCH_ERR("Can't disable MMAP plugin: %s\n", snd_strerror(err));
      goto error;
   }

   params.channel = SND_PCM_CHANNEL_PLAYBACK;
   params.mode = SND_PCM_MODE_BLOCK;

   params.format.interleave = 1;
   params.format.format = SND_PCM_SFMT_S16_LE;
   params.format.rate = rate; 
   params.format.voices = 2;

   params.start_mode = SND_PCM_START_DATA;
   params.stop_mode = SND_PCM_STOP_STOP;

   params.buf.block.frag_size = 4096;
   params.buf.block.frags_min = 1;
   params.buf.block.frags_max = 19;

   if ((err = snd_pcm_plugin_params(alsa->pcm, &params)) < 0)
   {
      RARCH_ERR("Channel Parameter Error: %s\n", snd_strerror(err));
      goto error;
   }

   setup.channel = SND_PCM_CHANNEL_PLAYBACK;

   if ((err = snd_pcm_plugin_setup(alsa->pcm, &setup)) < 0)
   {
      RARCH_ERR("Channel Parameter Read Back Error: %s\n", snd_strerror(err));
      goto error;
   }

   alsa->buffer_size = setup.buf.block.frag_size * 19;
   RARCH_LOG("ALSA buffer size: %d\n", alsa->buffer_size);

   if ((err = snd_pcm_plugin_prepare(alsa->pcm, SND_PCM_CHANNEL_PLAYBACK)) < 0)
   {
      RARCH_ERR("Channel Prepare Error: %s\n", snd_strerror(err));
      goto error;
   }

   alsa->has_float = false;

   return alsa;

error:
   return (void*)-1;
}

static void alsa_qsa_free(void *data)
{
   alsa_t *alsa = (alsa_t*)data;

   if (alsa->pcm != NULL)
   {
      snd_pcm_close(alsa->pcm);
      alsa->pcm = NULL;
   }
}

static ssize_t alsa_qsa_write(void *data, const void *buf, size_t size)
{
   alsa_t *alsa = (alsa_t*)data;

   int err = snd_pcm_plugin_write(alsa->pcm, buf, size);

   return size;
}

static bool alsa_qsa_stop(void *data)
{
   alsa_t *alsa = (alsa_t*)data;

   if (alsa->can_pause && !alsa->is_paused)
   {
      /* TODO */
      return false;
   }
   else return true;
}

static bool alsa_qsa_start(void *data)
{
   (void)data;
   return true;
}

static void alsa_qsa_set_nonblock_state(void *data, bool state)
{
   alsa_t *alsa = (alsa_t*)data;

   int err;

   if((err = snd_pcm_nonblock_mode(alsa->pcm, state)) < 0)
   {
      RARCH_ERR("Can't set blocking mode to %d: %s\n", state,
            snd_strerror(err));
      return;
   }

   alsa->nonblock = state;
}

static bool alsa_qsa_use_float(void *data)
{
   alsa_t *alsa = (alsa_t*)data;
   return alsa->has_float;
}

const audio_driver_t audio_alsa = {
   alsa_qsa_init,
   alsa_qsa_write,
   alsa_qsa_stop,
   alsa_qsa_start,
   alsa_qsa_set_nonblock_state,
   alsa_qsa_free,
   alsa_qsa_use_float,
   "alsa",
};