/**

 * $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 frame_bytes;

struct lxdream_config_entry alsa_config[] = {

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

    {NULL, CONFIG_TYPE_NONE}

};

gboolean audio_alsa_init(  )

{

    int err;

    snd_pcm_hw_params_t *hw_params;

    snd_pcm_sw_params_t *sw_params;

    snd_pcm_uframes_t frames;

    snd_pcm_uframes_t bufferSize;

    int rate = DEFAULT_SAMPLE_RATE;

    int format = DEFAULT_SAMPLE_FORMAT;

    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_shutdown(  )

{

    return TRUE;

}

struct audio_driver audio_alsa_driver = { 

    "alsa",

    N_("Linux ALSA system driver"),

    DEFAULT_SAMPLE_RATE,

    DEFAULT_SAMPLE_FORMAT,

    audio_alsa_init,

    NULL,

    audio_alsa_process_buffer,

    NULL,

    audio_alsa_shutdown

};