/*
	snd_alsa.c

	Sound code taken from icculus quake2 and modified to
	work with Warsow
	
	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:

		Free Software Foundation, Inc.
		59 Temple Place - Suite 330
		Boston, MA  02111-1307, USA

	Copyright 2005 Brendan Burns
	Copyright 2005 Andreas Schneider
*/

#define BUFFER_SIZE 4096

#include <alsa/asoundlib.h>

#include "../../client/client.h"
#include "../../client/snd_loc.h"

snd_pcm_sframes_t snd_buf = BUFFER_SIZE;

static int  snd_inited;
static short *buffer;

/* Handle for the PCM device */ 
static snd_pcm_t *playback_handle;

/* This structure contains information about    */
/* the hardware and can be used to specify the  */      
/* configuration to be used for the PCM stream. */ 
static snd_pcm_hw_params_t *hw_params;

cvar_t *sndbits;
cvar_t *sndspeed;
cvar_t *sndchannels;
cvar_t *snddevice;

static int tryrates[] = { 44100, 22051, 11025, 48000, 8000 };

qboolean SNDDMA_Init (void)
{
  int i;
  int err;
  snd_pcm_uframes_t buffersize;
  int framesize;
  int format;

  if (snd_inited) { return 1; }
  
  sndbits = Cvar_Get("sndbits", "16", CVAR_ARCHIVE);
  sndspeed = Cvar_Get("sndspeed", "0", CVAR_ARCHIVE);
  sndchannels = Cvar_Get("sndchannels", "2", CVAR_ARCHIVE);
  snddevice = Cvar_Get("snddevice", "default", CVAR_ARCHIVE);

  /* Allocate the snd_pcm_hw_params_t structure on the stack. */
  snd_pcm_hw_params_alloca(&hw_params);
 
  /* Open PCM. The last parameter of this function is the mode. */
  /* If this is set to 0, the standard mode is used. Possible   */
  /* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC.       */ 
  /* If SND_PCM_NONBLOCK is used, read / write access to the    */
  /* PCM device will return immediately. If SND_PCM_ASYNC is    */
  /* specified, SIGIO will be emitted whenever a period has     */
  /* been completely processed by the soundcard.                */
  err = snd_pcm_open(&playback_handle, snddevice->string, 
		     SND_PCM_STREAM_PLAYBACK, 0);
  if (err < 0) {
    Com_Printf("ALSA snd error, cannot open device %s (%s)\n", 
	       snddevice->string,
	       snd_strerror(err));
    return 0;
  }
  

  err = snd_pcm_hw_params_malloc(&hw_params);
  if (err < 0) {
    Com_Printf("ALSA snd error, cannot allocate hw params (%s)\n",
	       snd_strerror(err));
    return 0;
  }
  
  
  err = snd_pcm_hw_params_any(playback_handle, hw_params);
  if (err < 0) {
    Com_Printf("ALSA snd error, cannot init hw params (%s)\n",
	       snd_strerror(err));
    snd_pcm_hw_params_free(hw_params);
    return 0;
  }
  
  /* Set access type. This can be either    */
  /* SND_PCM_ACCESS_RW_INTERLEAVED or       */
  /* SND_PCM_ACCESS_RW_NONINTERLEAVED.      */
  /* There are also access types for MMAPed */
  err = snd_pcm_hw_params_set_access(playback_handle, hw_params, 
				     SND_PCM_ACCESS_RW_INTERLEAVED);
  if (err < 0) {
    Com_Printf("ALSA snd error, cannot set access (%s)\n",
	       snd_strerror(err));
    snd_pcm_hw_params_free(hw_params);
    return 0;
  }
  
  /* Set sample format */
  dma.samplebits = sndbits->integer;
  if (dma.samplebits == 16 || dma.samplebits != 8) {
    err = snd_pcm_hw_params_set_format(playback_handle, hw_params,
				       SND_PCM_FORMAT_S16_LE);
    if (err < 0) {
      Com_Printf("ALSA snd error, 16 bit sound not supported, trying 8\n");
      dma.samplebits = 8;
    }
    else {
      format = SND_PCM_FORMAT_S16_LE;
    }
  }
  if (dma.samplebits == 8) {
    err = snd_pcm_hw_params_set_format(playback_handle, hw_params,
				       SND_PCM_FORMAT_U8);
  }
  if (err < 0) {
    Com_Printf("ALSA snd error, cannot set sample format (%s)\n",
	       snd_strerror(err));
    snd_pcm_hw_params_free(hw_params);
    return 0;
  }
  else {
    format = SND_PCM_FORMAT_U8;
  }
 
  /* Set sample rate */
  dma.speed = sndspeed->integer;
  if (!dma.speed) {
    for (i = 0 ; i < sizeof(tryrates); i++) {
      int test = tryrates[i];
      err = snd_pcm_hw_params_set_rate_near(playback_handle, hw_params,
					    &test, 0);
      if (err < 0) {
	Com_Printf("ALSA snd error, cannot set sample rate %d (%s)\n",
		   tryrates[i], snd_strerror(err));
      }
      else {
	dma.speed = test;
	break;
      }
    }
  }
  if (!dma.speed) {
    Com_Printf("ALSA snd error couldn't set rate.\n");
    snd_pcm_hw_params_free(hw_params);
    return 0;
  }

  /* Set number of channels */
  dma.channels = sndchannels->integer;
  if (dma.channels < 1 || dma.channels > 2)
  {
    dma.channels = 2;
  }
    
  err = snd_pcm_hw_params_set_channels(playback_handle, hw_params, dma.channels);
  if (err < 0) {
    Com_Printf("ALSA snd error couldn't set channels %d (%s).\n",
	       dma.channels, snd_strerror(err));
    snd_pcm_hw_params_free(hw_params);
    return 0;
  }
  
  /* Apply HW parameter settings to */
  /* PCM device and prepare device  */
  err = snd_pcm_hw_params(playback_handle, hw_params);
  if (err < 0) {
    Com_Printf("ALSA snd error couldn't set params (%s).\n",snd_strerror(err));
    snd_pcm_hw_params_free(hw_params);
    return 0;
  }


  snd_pcm_uframes_t p_size; 
  buffersize = snd_pcm_hw_params_get_buffer_size(hw_params, &p_size);
  framesize = (snd_pcm_format_physical_width(format) * dma.channels) / 8;
    
  snd_pcm_hw_params_free(hw_params);
  hw_params = NULL;
    
  if ((err = snd_pcm_prepare(playback_handle)) < 0) {
    Com_Printf("ALSA snd error preparing audio (%s)\n", snd_strerror(err));
    return 0;
  }
    
  snd_buf = buffersize * framesize;
  

  buffer=malloc(snd_buf);
  memset(buffer, 0, snd_buf);

  dma.samplepos = 0;
  dma.samples = snd_buf / (dma.samplebits / 8);
  dma.submission_chunk = 1;
  dma.buffer = (char *)buffer;

  snd_inited = 1;
  return 1;
}

int SNDDMA_GetDMAPos (void)
{
  if(snd_inited)
    return dma.samplepos;
  else
    Com_Printf ("Sound not inizialized\n");
  return 0;
}

void SNDDMA_Shutdown (void)
{
  if (snd_inited) {
    /* Stop PCM device and drop pending frames */
    snd_pcm_drop(playback_handle);
    
    /* Close PCM device */
    snd_pcm_close(playback_handle);
    snd_inited = 0;
  }
  free(dma.buffer);
  dma.buffer = NULL;
}

/*
** SNDDMA_Submit
** Send sound to device if buffer isn't really the dma buffer
*/
void SNDDMA_Submit (void)
{
  snd_pcm_sframes_t written;
  
  /* Write snd_buf frames from buffer data to       */ 
  /* the PCM device pointed to by pcm_handle.       */
  /* Returns the number of frames actually written. */
  if ((written = snd_pcm_writei(playback_handle, dma.buffer, snd_buf)) < 0) {
    Com_Printf("alsa: buffer underrun\n");
  }
}


void SNDDMA_BeginPainting(void)
{    
}