/*

    SDL - Simple DirectMedia Layer

    Copyright (C) 1997-2004 Sam Lantinga

    This library is free software; you can redistribute it and/or

    modify it under the terms of the GNU Library General Public

    License as published by the Free Software Foundation; either

    version 2 of the License, or (at your option) any later version.

    This library 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

    Library General Public License for more details.

    You should have received a copy of the GNU Library General Public

    License along with this library; if not, write to the Free

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

    Sam Lantinga

    slouken@libsdl.org

*/

#include "SDL_config.h"

/* Allow access to a raw mixing buffer */

#include <sys/types.h>

#include <signal.h>	/* For kill() */

#include "SDL_timer.h"

#include "SDL_audio.h"

#include "../SDL_audiomem.h"

#include "../SDL_audio_c.h"

#include "SDL_alsa_audio.h"

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

#include <dlfcn.h>

#include "SDL_name.h"

#include "SDL_loadso.h"

#else

#define SDL_NAME(X)	X

#endif

/* The tag name used by ALSA audio */

#define DRIVER_NAME         "alsa"

/* The default ALSA audio driver */

#define DEFAULT_DEVICE	"default"

/* Audio driver functions */

static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec);

static void ALSA_WaitAudio(_THIS);

static void ALSA_PlayAudio(_THIS);

static Uint8 *ALSA_GetAudioBuf(_THIS);

static void ALSA_CloseAudio(_THIS);

#ifdef SDL_AUDIO_DRIVER_ALSA_DYNAMIC

static const char *alsa_library = SDL_AUDIO_DRIVER_ALSA_DYNAMIC;

static void *alsa_handle = NULL;

static int alsa_loaded = 0;

static int (*SDL_snd_pcm_open)(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);

static int (*SDL_NAME(snd_pcm_open))(snd_pcm_t **pcm, const char *name, snd_pcm_stream_t stream, int mode);

static int (*SDL_NAME(snd_pcm_close))(snd_pcm_t *pcm);

static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_writei))(snd_pcm_t *pcm, const void *buffer, snd_pcm_uframes_t size);

static int (*SDL_NAME(snd_pcm_resume))(snd_pcm_t *pcm);

static int (*SDL_NAME(snd_pcm_prepare))(snd_pcm_t *pcm);

static int (*SDL_NAME(snd_pcm_drain))(snd_pcm_t *pcm);

static const char *(*SDL_NAME(snd_strerror))(int errnum);

static size_t (*SDL_NAME(snd_pcm_hw_params_sizeof))(void);

static size_t (*SDL_NAME(snd_pcm_sw_params_sizeof))(void);

static int (*SDL_NAME(snd_pcm_hw_params_any))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);

static int (*SDL_NAME(snd_pcm_hw_params_set_access))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_access_t access);

static int (*SDL_NAME(snd_pcm_hw_params_set_format))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_format_t val);

static int (*SDL_NAME(snd_pcm_hw_params_set_channels))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val);

static int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params);

static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);

static snd_pcm_uframes_t (*SDL_NAME(snd_pcm_hw_params_set_period_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t val, int *dir);

static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params);

static unsigned int (*SDL_NAME(snd_pcm_hw_params_set_periods_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val, int *dir);

static int (*SDL_NAME(snd_pcm_hw_params_get_periods))(snd_pcm_hw_params_t *params);

static int (*SDL_NAME(snd_pcm_hw_params))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params);

/*

*/

static int (*SDL_NAME(snd_pcm_sw_params_current))(snd_pcm_t *pcm, snd_pcm_sw_params_t *swparams);

static int (*SDL_NAME(snd_pcm_sw_params_set_start_threshold))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);

static int (*SDL_NAME(snd_pcm_sw_params_set_avail_min))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val);

static int (*SDL_NAME(snd_pcm_sw_params))(snd_pcm_t *pcm, snd_pcm_sw_params_t *params);

static int (*SDL_NAME(snd_pcm_nonblock))(snd_pcm_t *pcm, int nonblock);

#define snd_pcm_hw_params_sizeof SDL_NAME(snd_pcm_hw_params_sizeof)

#define snd_pcm_sw_params_sizeof SDL_NAME(snd_pcm_sw_params_sizeof)

/* cast funcs to char* first, to please GCC's strict aliasing rules. */

static struct {

	const char *name;

	void **func;

} alsa_functions[] = {

	{ "snd_pcm_open",	(void**)(char*)&SDL_NAME(snd_pcm_open)		},

	{ "snd_pcm_close",	(void**)(char*)&SDL_NAME(snd_pcm_close)	},

	{ "snd_pcm_writei",	(void**)(char*)&SDL_NAME(snd_pcm_writei)	},

	{ "snd_pcm_resume",	(void**)(char*)&SDL_NAME(snd_pcm_resume)	},

	{ "snd_pcm_prepare",	(void**)(char*)&SDL_NAME(snd_pcm_prepare)	},

	{ "snd_pcm_drain",	(void**)(char*)&SDL_NAME(snd_pcm_drain)	},

	{ "snd_strerror",	(void**)(char*)&SDL_NAME(snd_strerror)		},

	{ "snd_pcm_hw_params_sizeof",		(void**)(char*)&SDL_NAME(snd_pcm_hw_params_sizeof)		},

	{ "snd_pcm_sw_params_sizeof",		(void**)(char*)&SDL_NAME(snd_pcm_sw_params_sizeof)		},

	{ "snd_pcm_hw_params_any",		(void**)(char*)&SDL_NAME(snd_pcm_hw_params_any)		},

	{ "snd_pcm_hw_params_set_access",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_access)		},

	{ "snd_pcm_hw_params_set_format",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_format)		},

	{ "snd_pcm_hw_params_set_channels",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_channels)	},

	{ "snd_pcm_hw_params_get_channels",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_channels)	},

	{ "snd_pcm_hw_params_set_rate_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_rate_near)	},

	{ "snd_pcm_hw_params_set_period_size_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_period_size_near)	},

	{ "snd_pcm_hw_params_get_period_size",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_period_size)	},

	{ "snd_pcm_hw_params_set_periods_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_periods_near)	},

	{ "snd_pcm_hw_params_get_periods",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_periods)	},

	{ "snd_pcm_hw_params",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params)	},

	{ "snd_pcm_sw_params_current",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_current)	},

	{ "snd_pcm_sw_params_set_start_threshold",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_start_threshold)	},

	{ "snd_pcm_sw_params_set_avail_min",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params_set_avail_min)	},

	{ "snd_pcm_sw_params",	(void**)(char*)&SDL_NAME(snd_pcm_sw_params)	},

	{ "snd_pcm_nonblock",	(void**)(char*)&SDL_NAME(snd_pcm_nonblock)	},

};

static void UnloadALSALibrary(void) {

	if (alsa_loaded) {

/*		SDL_UnloadObject(alsa_handle);*/

		dlclose(alsa_handle);

		alsa_handle = NULL;

		alsa_loaded = 0;

	}

}

static int LoadALSALibrary(void) {

	int i, retval = -1;

/*	alsa_handle = SDL_LoadObject(alsa_library);*/

	alsa_handle = dlopen(alsa_library,RTLD_NOW);

	if (alsa_handle) {

		alsa_loaded = 1;

		retval = 0;

		for (i = 0; i < SDL_arraysize(alsa_functions); i++) {

/*			*alsa_functions[i].func = SDL_LoadFunction(alsa_handle,alsa_functions[i].name);*/

#if HAVE_DLVSYM

			*alsa_functions[i].func = dlvsym(alsa_handle,alsa_functions[i].name,"ALSA_0.9");

			if (!*alsa_functions[i].func)

#endif

				*alsa_functions[i].func = dlsym(alsa_handle,alsa_functions[i].name);

			if (!*alsa_functions[i].func) {

				retval = -1;

				UnloadALSALibrary();

				break;

			}

		}

	}

	return retval;

}

#else

static void UnloadALSALibrary(void) {

	return;

}

static int LoadALSALibrary(void) {

	return 0;

}

#endif /* SDL_AUDIO_DRIVER_ALSA_DYNAMIC */

static const char *get_audio_device(int channels)

{

	const char *device;

	device = SDL_getenv("AUDIODEV");	/* Is there a standard variable name? */

	if ( device == NULL ) {

		if (channels == 6) device = "surround51";

		else if (channels == 4) device = "surround40";

		else device = DEFAULT_DEVICE;

	}

	return device;

}

/* Audio driver bootstrap functions */

static int Audio_Available(void)

{

	int available;

	int status;

	snd_pcm_t *handle;

	available = 0;

	if (LoadALSALibrary() < 0) {

		return available;

	}

	status = SDL_NAME(snd_pcm_open)(&handle, get_audio_device(2), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

	if ( status >= 0 ) {

		available = 1;

        	SDL_NAME(snd_pcm_close)(handle);

	}

	UnloadALSALibrary();

	return(available);

}

static void Audio_DeleteDevice(SDL_AudioDevice *device)

{

	SDL_free(device->hidden);

	SDL_free(device);

	UnloadALSALibrary();

}

static SDL_AudioDevice *Audio_CreateDevice(int devindex)

{

	SDL_AudioDevice *this;

	/* Initialize all variables that we clean on shutdown */

	LoadALSALibrary();

	this = (SDL_AudioDevice *)SDL_malloc(sizeof(SDL_AudioDevice));

	if ( this ) {

		SDL_memset(this, 0, (sizeof *this));

		this->hidden = (struct SDL_PrivateAudioData *)

				SDL_malloc((sizeof *this->hidden));

	}

	if ( (this == NULL) || (this->hidden == NULL) ) {

		SDL_OutOfMemory();

		if ( this ) {

			SDL_free(this);

		}

		return(0);

	}

	SDL_memset(this->hidden, 0, (sizeof *this->hidden));

	/* Set the function pointers */

	this->OpenAudio = ALSA_OpenAudio;

	this->WaitAudio = ALSA_WaitAudio;

	this->PlayAudio = ALSA_PlayAudio;

	this->GetAudioBuf = ALSA_GetAudioBuf;

	this->CloseAudio = ALSA_CloseAudio;

	this->free = Audio_DeleteDevice;

	return this;

}

AudioBootStrap ALSA_bootstrap = {

	DRIVER_NAME, "ALSA 0.9 PCM audio",

	Audio_Available, Audio_CreateDevice

};

/* This function waits until it is possible to write a full sound buffer */

static void ALSA_WaitAudio(_THIS)

{

	/* Check to see if the thread-parent process is still alive */

	{ static int cnt = 0;

		/* Note that this only works with thread implementations 

		   that use a different process id for each thread.

		*/

		if (parent && (((++cnt)%10) == 0)) { /* Check every 10 loops */

			if ( kill(parent, 0) < 0 ) {

				this->enabled = 0;

			}

		}

	}

}

/*

 * http://bugzilla.libsdl.org/show_bug.cgi?id=110

 * "For Linux ALSA, this is FL-FR-RL-RR-C-LFE

 *  and for Windows DirectX [and CoreAudio], this is FL-FR-C-LFE-RL-RR"

 */

#define SWIZ6(T) \

    T *ptr = (T *) mixbuf; \

    const Uint32 count = (this->spec.samples / 6); \

    Uint32 i; \

    for (i = 0; i < count; i++, ptr += 6) { \

        T tmp; \

        tmp = ptr[2]; ptr[2] = ptr[4]; ptr[4] = tmp; \

        tmp = ptr[3]; ptr[3] = ptr[5]; ptr[5] = tmp; \

    }

static __inline__ void swizzle_alsa_channels_6_64bit(_THIS) { SWIZ6(Uint64); }

static __inline__ void swizzle_alsa_channels_6_32bit(_THIS) { SWIZ6(Uint32); }

static __inline__ void swizzle_alsa_channels_6_16bit(_THIS) { SWIZ6(Uint16); }

static __inline__ void swizzle_alsa_channels_6_8bit(_THIS) { SWIZ6(Uint8); }

#undef SWIZ6

/*

 * Called right before feeding this->mixbuf to the hardware. Swizzle channels

 *  from Windows/Mac order to the format alsalib will want.

 */

static __inline__ void swizzle_alsa_channels(_THIS)

{

    if (this->spec.channels == 6) {

        const Uint16 fmtsize = (this->spec.format & 0xFF); /* bits/channel. */

        if (fmtsize == 16)

            swizzle_alsa_channels_6_16bit(this);

        else if (fmtsize == 8)

            swizzle_alsa_channels_6_8bit(this);

        else if (fmtsize == 32)

            swizzle_alsa_channels_6_32bit(this);

        else if (fmtsize == 64)

            swizzle_alsa_channels_6_64bit(this);

    }

    /* !!! FIXME: update this for 7.1 if needed, later. */

}

static void ALSA_PlayAudio(_THIS)

{

	int           status;

	int           sample_len;

	signed short *sample_buf;

	swizzle_alsa_channels(this);

	sample_len = this->spec.samples;

	sample_buf = (signed short *)mixbuf;

	while ( sample_len > 0 ) {

		status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, sample_len);

		if ( status < 0 ) {

			if ( status == -EAGAIN ) {

				SDL_Delay(1);

				continue;

			}

			if ( status == -ESTRPIPE ) {

				do {

					SDL_Delay(1);

					status = SDL_NAME(snd_pcm_resume)(pcm_handle);

				} while ( status == -EAGAIN );

			}

			if ( status < 0 ) {

				status = SDL_NAME(snd_pcm_prepare)(pcm_handle);

			}

			if ( status < 0 ) {

				/* Hmm, not much we can do - abort */

				this->enabled = 0;

				return;

			}

			continue;

		}

		sample_buf += status * this->spec.channels;

		sample_len -= status;

	}

}

static Uint8 *ALSA_GetAudioBuf(_THIS)

{

	return(mixbuf);

}

static void ALSA_CloseAudio(_THIS)

{

	if ( mixbuf != NULL ) {

		SDL_FreeAudioMem(mixbuf);

		mixbuf = NULL;

	}

	if ( pcm_handle ) {

		SDL_NAME(snd_pcm_drain)(pcm_handle);

		SDL_NAME(snd_pcm_close)(pcm_handle);

		pcm_handle = NULL;

	}

}

static int ALSA_OpenAudio(_THIS, SDL_AudioSpec *spec)

{

	int                  status;

	snd_pcm_hw_params_t *hwparams;

	snd_pcm_sw_params_t *swparams;

	snd_pcm_format_t     format;

	snd_pcm_uframes_t    frames;

	Uint16               test_format;

	/* Open the audio device */

	/* Name of device should depend on # channels in spec */

	status = SDL_NAME(snd_pcm_open)(&pcm_handle, get_audio_device(spec->channels), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);

	if ( status < 0 ) {

		SDL_SetError("Couldn't open audio device: %s", SDL_NAME(snd_strerror)(status));

		return(-1);

	}

	/* Figure out what the hardware is capable of */

	snd_pcm_hw_params_alloca(&hwparams);

	status = SDL_NAME(snd_pcm_hw_params_any)(pcm_handle, hwparams);

	if ( status < 0 ) {

		SDL_SetError("Couldn't get hardware config: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

	/* SDL only uses interleaved sample output */

	status = SDL_NAME(snd_pcm_hw_params_set_access)(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED);

	if ( status < 0 ) {

		SDL_SetError("Couldn't set interleaved access: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

	/* Try for a closest match on audio format */

	status = -1;

	for ( test_format = SDL_FirstAudioFormat(spec->format);

	      test_format && (status < 0); ) {

		switch ( test_format ) {

			case AUDIO_U8:

				format = SND_PCM_FORMAT_U8;

				break;

			case AUDIO_S8:

				format = SND_PCM_FORMAT_S8;

				break;

			case AUDIO_S16LSB:

				format = SND_PCM_FORMAT_S16_LE;

				break;

			case AUDIO_S16MSB:

				format = SND_PCM_FORMAT_S16_BE;

				break;

			case AUDIO_U16LSB:

				format = SND_PCM_FORMAT_U16_LE;

				break;

			case AUDIO_U16MSB:

				format = SND_PCM_FORMAT_U16_BE;

				break;

			default:

				format = 0;

				break;

		}

		if ( format != 0 ) {

			status = SDL_NAME(snd_pcm_hw_params_set_format)(pcm_handle, hwparams, format);

		}

		if ( status < 0 ) {

			test_format = SDL_NextAudioFormat();

		}

	}

	if ( status < 0 ) {

		SDL_SetError("Couldn't find any hardware audio formats");

		ALSA_CloseAudio(this);

		return(-1);

	}

	spec->format = test_format;

	/* Set the number of channels */

	status = SDL_NAME(snd_pcm_hw_params_set_channels)(pcm_handle, hwparams, spec->channels);

	if ( status < 0 ) {

		status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams);

		if ( (status <= 0) || (status > 2) ) {

			SDL_SetError("Couldn't set audio channels");

			ALSA_CloseAudio(this);

			return(-1);

		}

		spec->channels = status;

	}

	/* Set the audio rate */

	status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, spec->freq, NULL);

	if ( status < 0 ) {

		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

	spec->freq = status;

	/* Set the buffer size, in samples */

	frames = spec->samples;

	frames = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, frames, NULL);

	spec->samples = frames;

	SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, 2, NULL);

	/* "set" the hardware with the desired parameters */

	status = SDL_NAME(snd_pcm_hw_params)(pcm_handle, hwparams);

	if ( status < 0 ) {

		SDL_SetError("Couldn't set hardware audio parameters: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

/* This is useful for debugging... */

/*

{ snd_pcm_sframes_t bufsize; int fragments;

   bufsize = SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams);

   fragments = SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams);

   fprintf(stderr, "ALSA: bufsize = %ld, fragments = %d\n", bufsize, fragments);

}

*/

	/* Set the software parameters */

	snd_pcm_sw_params_alloca(&swparams);

	status = SDL_NAME(snd_pcm_sw_params_current)(pcm_handle, swparams);

	if ( status < 0 ) {

		SDL_SetError("Couldn't get software config: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

	status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0);

	if ( status < 0 ) {

		SDL_SetError("Couldn't set start threshold: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

	status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames);

	if ( status < 0 ) {

		SDL_SetError("Couldn't set avail min: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

	status = SDL_NAME(snd_pcm_sw_params)(pcm_handle, swparams);

	if ( status < 0 ) {

		SDL_SetError("Couldn't set software audio parameters: %s", SDL_NAME(snd_strerror)(status));

		ALSA_CloseAudio(this);

		return(-1);

	}

	/* Calculate the final parameters for this audio specification */

	SDL_CalculateAudioSpec(spec);

	/* Allocate mixing buffer */

	mixlen = spec->size;

	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);

	if ( mixbuf == NULL ) {

		ALSA_CloseAudio(this);

		return(-1);

	}

	SDL_memset(mixbuf, spec->silence, spec->size);