/**

  * $Id: audio_esd.c 602 2008-01-15 20:50:23Z nkeynes $

  * 

  * The asla  audio driver

  *

  * Copyright (c) 2008 Jonathan Muller

  *

  * 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.

  */

 #include <stdio.h>

 #include <unistd.h>

 /* Use the newer ALSA API */

 #define ALSA_PCM_NEW_HW_PARAMS_API

 #include <alsa/asoundlib.h>

 #include "config.h"

 #include "aica/audio.h"

 #include "dream.h"

 static snd_pcm_t *_soundDevice = NULL;

 static int frames;

 static int frame_bytes;

 struct lxdream_config_entry alsa_config[] = {

     {"device", CONFIG_TYPE_FILE, "default"},

     {NULL, CONFIG_TYPE_NONE}

 };

 gboolean audio_alsa_init(  )

 {

     int err;

     return TRUE;

 }

 gboolean audio_alsa_set_format( uint32_t rate, uint32_t format )

 {

     int i;

     int err;

     snd_pcm_hw_params_t *hw_params;

     snd_pcm_sw_params_t *sw_params;

     snd_pcm_uframes_t frames;

     unsigned int resample = 1;

     unsigned int actualRate = rate;

     snd_pcm_uframes_t bufferSize;

     int dir;

     // Open the device we were told to open.

     err = snd_pcm_open( &_soundDevice, alsa_config[0].value,

                         SND_PCM_STREAM_PLAYBACK, 0 );

     // Check for error on open.

     if ( err < 0 ) {

         ERROR( "Init: cannot open audio device %s (%s)\n",

                alsa_config[0].value, snd_strerror( err ) );

         return FALSE;

     } else {

         DEBUG( "Audio device opened successfully." );

     }

     frame_bytes = ( 2 * ( snd_pcm_format_width( SND_PCM_FORMAT_S16_LE ) / 8 ) );

     //snd_pcm_hw_params_alloca (&hw_params);

     // Allocate the hardware parameter structure.

     if ( ( err = snd_pcm_hw_params_malloc( &hw_params ) ) < 0 ) {

         ERROR( "Init: cannot allocate hardware parameter structure (%s)\n",

                snd_strerror( err ) );

         return FALSE;

     }

     if ( ( err = snd_pcm_hw_params_any( _soundDevice, hw_params ) ) < 0 ) {

         ERROR( "Init: cannot allocate hardware parameter structure (%s)\n",

                snd_strerror( err ) );

         return FALSE;

     }

     // Set access to RW interleaved.

     if ( ( err = snd_pcm_hw_params_set_access( _soundDevice, hw_params,

                                                SND_PCM_ACCESS_RW_INTERLEAVED ) )

          < 0 ) {

         ERROR( " Init: cannot set access type (%s)\n", snd_strerror( err ) );

         return FALSE;

     }

     if ( ( err = snd_pcm_hw_params_set_format( _soundDevice, hw_params,

                                                SND_PCM_FORMAT_S16_LE ) ) <

          0 ) {

         ERROR( "Init: cannot set sample format (%s)\n", snd_strerror( err ) );

         return FALSE;

     }

     err = snd_pcm_hw_params_set_rate_near( _soundDevice, hw_params, &rate, 0 );

     if ( err < 0 ) {

         ERROR( "Init: Resampling setup failed for playback: %s\n",

                snd_strerror( err ) );

         return err;

     }

     // Set channels to stereo (2).

     err = snd_pcm_hw_params_set_channels( _soundDevice, hw_params, 2 );

     if ( err < 0 ) {

         ERROR( "Init: cannot set channel count (%s)\n", snd_strerror( err ) );

         return FALSE;

     }

     // frames = 4410;

     // snd_pcm_hw_params_set_period_size_near( _soundDevice, hw_params, &frames,

     //                                     &dir );

     // Apply the hardware parameters that we've set.

     err = snd_pcm_hw_params( _soundDevice, hw_params );

     if ( err < 0 ) {

         DEBUG( "Init: cannot set parameters (%s)\n", snd_strerror( err ) );

         return FALSE;

     } else {

         DEBUG( "Audio device parameters have been set successfully." );

     }

     snd_pcm_hw_params_get_period_size( hw_params, &frames, &dir );

     DEBUG( "period size = %d\n", frames );

     // Get the buffer size.

     snd_pcm_hw_params_get_buffer_size( hw_params, &bufferSize );

     DEBUG("Buffer Size = %d\n", bufferSize);

     // If we were going to do more with our sound device we would want to store

     // the buffer size so we know how much data we will need to fill it with.

     //cout << "Init: Buffer size = " << bufferSize << " frames." << endl;

     // Display the bit size of samples.

     //cout << "Init: Significant bits for linear samples = " << snd_pcm_hw_params_get_sbits(hw_params) << endl;

     // Free the hardware parameters now that we're done with them.

     snd_pcm_hw_params_free( hw_params );

     // Set the start threshold to reduce inter-buffer gaps

     snd_pcm_sw_params_alloca( &sw_params );

     snd_pcm_sw_params_current( _soundDevice, sw_params );

     snd_pcm_sw_params_set_start_threshold( _soundDevice, sw_params, bufferSize/2 );

     err = snd_pcm_sw_params( _soundDevice, sw_params );

     if( err < 0 ) {

 	ERROR("Unable to set sw params for alsa driver: %s\n", snd_strerror(err));

 	return FALSE;

     } 

     err = snd_pcm_prepare( _soundDevice );

     if ( err < 0 ) {

         ERROR( "Init: cannot prepare audio interface for use (%s)\n",

                snd_strerror( err ) );

         return FALSE;

     }

     return TRUE;

 }

 gboolean audio_alsa_process_buffer( audio_buffer_t buffer )

 {

     int err;

     int length;

     length = buffer->length / frame_bytes;

     err = snd_pcm_writei( _soundDevice, buffer->data, length );

     if( err == -EPIPE ) {

 	snd_pcm_prepare( _soundDevice );

     } else if( err == -ESTRPIPE ) {

 	snd_pcm_resume( _soundDevice );

     }

     return TRUE;

 }

 gboolean audio_alsa_close(  )

 {

     int err;

     return TRUE;

 }

 struct audio_driver audio_alsa_driver = { 

     "alsa",

     audio_alsa_init,

     audio_alsa_set_format,

     audio_alsa_process_buffer,

     audio_alsa_close

 };