Long-standing SDL ALSA bug

Long-standing SDL ALSA bug

[Sam Lantinga]

[-- Attachment #1: Type: text/plain, Size: 587 bytes --]

Hey guys, we're in the process of getting ready to release SDL 1.2.14,
and we're trying to fix a long standing issue with crackling and pops
in ALSA with some configurations:
http://bugzilla.libsdl.org/show_bug.cgi?id=650

What's really puzzling is that we're not able to reproduce it ourselves at all.

I'm attaching the latest versions of the SDL audio files, and I'd
really appreciate it if you take a look and sanity check our code.
You can get the full pre-release here:
http://www.libsdl.org/tmp/SDL-1.2.zip

Thanks!
-- 
	-Sam Lantinga, Founder and President, Galaxy Gameworks LLC

[-- Attachment #2: SDL_alsa_audio.c --]
[-- Type: application/octet-stream, Size: 17870 bytes --]

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2009 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 "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_set_rate_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
static int (*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 int (*SDL_NAME(snd_pcm_hw_params_get_channels))(const snd_pcm_hw_params_t *params, unsigned int *val);
static snd_pcm_sframes_t (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params);
static 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);
		alsa_handle = NULL;
		alsa_loaded = 0;
	}
}

static int LoadALSALibrary(void) {
	int i, retval = -1;

	alsa_handle = SDL_LoadObject(alsa_library);
	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 (!*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;
	snd_pcm_uframes_t frames_left;
	const Uint8 *sample_buf = (const Uint8 *) mixbuf;
	const int frame_size = ( ((int) this->spec.channels) *
	                         (((int) (this->spec.format & 0xFF)) / 8) );

	swizzle_alsa_channels(this);

	frames_left = ((snd_pcm_uframes_t) this->spec.samples) / this->spec.channels;

	while ( frames_left > 0 ) {
		status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, frames_left);
		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 * frame_size;
		frames_left -= 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;
	unsigned int         rate;
	unsigned int         periods;
	unsigned int 		 channels;
	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);
	channels = spec->channels;
	if ( status < 0 ) {
		status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels);
		if ( status < 0 ) {
			SDL_SetError("Couldn't set audio channels");
			ALSA_CloseAudio(this);
			return(-1);
		}
		spec->channels = channels;
	}

	/* Set the audio rate */
	rate = spec->freq;

	status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->freq = rate;

	/* Set the buffer size, in samples */
	frames = spec->samples;
	status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	spec->samples = frames;

	periods = 2;
	SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, 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);

	/* Get the parent process id (we're the parent of the audio thread) */
	parent = getpid();

	/* Switch to blocking mode for playback */
	SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);

	/* We're ready to rock and roll. :-) */
	return(0);
}

[-- Attachment #3: SDL_alsa_audio.h --]
[-- Type: application/octet-stream, Size: 1535 bytes --]

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2009 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"

#ifndef _ALSA_PCM_audio_h
#define _ALSA_PCM_audio_h

#include <alsa/asoundlib.h>

#include "../SDL_sysaudio.h"

/* Hidden "this" pointer for the video functions */
#define _THIS	SDL_AudioDevice *this

struct SDL_PrivateAudioData {
	/* The audio device handle */
	snd_pcm_t *pcm_handle;

	/* The parent process id, to detect when application quits */
	pid_t parent;

	/* Raw mixing buffer */
	Uint8 *mixbuf;
	int    mixlen;
};

/* Old variable names */
#define pcm_handle		(this->hidden->pcm_handle)
#define parent			(this->hidden->parent)
#define mixbuf			(this->hidden->mixbuf)
#define mixlen			(this->hidden->mixlen)

#endif /* _ALSA_PCM_audio_h */

[-- Attachment #4: Type: text/plain, Size: 160 bytes --]

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Re: Long-standing SDL ALSA bug

[The Source]

12.10.2009 22:14, Sam Lantinga ?????:
> Hey guys, we're in the process of getting ready to release SDL 1.2.14,
> and we're trying to fix a long standing issue with crackling and pops
> in ALSA with some configurations:
> http://bugzilla.libsdl.org/show_bug.cgi?id=650
>
> What's really puzzling is that we're not able to reproduce it ourselves at all.
>
> I'm attaching the latest versions of the SDL audio files, and I'd
> really appreciate it if you take a look and sanity check our code.
> You can get the full pre-release here:
> http://www.libsdl.org/tmp/SDL-1.2.zip
>
> Thanks!
>    
>
>
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
>    
I may be wrong but I have a feeling that the problem is in the 
pulseaudio actually not in alsa.
There's a thing called alsa-plugins-pulseaudio which creates virtual 
alsa device. This device sends all sound output to pulseaudio.
Many modern distributions like to use this device as default. And 
pulseaudio still has some crackling issues, yes.

Re: Long-standing SDL ALSA bug

[Peter Lawler]

On 13/10/09 14:36, The Source wrote:
 > 12.10.2009 22:14, Sam Lantinga ?????:
 >> http://bugzilla.libsdl.org/show_bug.cgi?id=650

 > I may be wrong but I have a feeling that the problem is in the
 > pulseaudio actually not in alsa.

I'd take a look, but libsdl.org seems down atm.

Re: Long-standing SDL ALSA bug

[Clemens Ladisch]

Sam Lantinga wrote:
> I'm attaching the latest versions of the SDL audio files, and I'd
> really appreciate it if you take a look and sanity check our code.

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

ALSA's devices look at certain environment variables; you are supposed
to use names like "default" or "surround51" to get that default
configuration.

> 	if (channels == 6) device = "surround51";
> 	else if (channels == 4) device = "surround40";

The devices do not have automatic sample format/rate conversion; you
might want to use "plug:surround.." instead.

> /* This function waits until it is possible to write a full sound buffer */
> static void ALSA_WaitAudio(_THIS)

This function doesn't actually wait ...

> 	if ( status == -EAGAIN ) {
> 		SDL_Delay(1);
> 		continue;
> 	}

Not necessary in blocking mode.

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

The returned rate could be wildly different, but I guess SDL correctly
handles the new value in spec->freq.

> 	/* Set the buffer size, in samples */
> 	frames = spec->samples;
> 	status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);

This does _not_ set the buffer size.

> 	periods = 2;
> 	SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL);

A bigger number of periods would make the writing of audio data less
bursty.

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

Zero doesn't really make sense; the default value 1 would be OK.

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

The default value is the period size anyway, so you can remove this.
(If you change the buffer size code above to use _set_buffer_size_near,
this call would be wrong.)


Best regards,
Clemens

Re: Long-standing SDL ALSA bug

[Sam Lantinga]

[-- Attachment #1: Type: text/plain, Size: 2411 bytes --]

Thanks for the feedback!  I think that improved our ALSA driver
significantly (attached!)

On Tue, Oct 13, 2009 at 12:03 AM, Clemens Ladisch <clemens@ladisch.de> wrote:
> Sam Lantinga wrote:
>> I'm attaching the latest versions of the SDL audio files, and I'd
>> really appreciate it if you take a look and sanity check our code.
>
>>       device = SDL_getenv("AUDIODEV");        /* Is there a standard variable name? */
>
> ALSA's devices look at certain environment variables; you are supposed
> to use names like "default" or "surround51" to get that default
> configuration.
>
>>       if (channels == 6) device = "surround51";
>>       else if (channels == 4) device = "surround40";
>
> The devices do not have automatic sample format/rate conversion; you
> might want to use "plug:surround.." instead.
>
>> /* This function waits until it is possible to write a full sound buffer */
>> static void ALSA_WaitAudio(_THIS)
>
> This function doesn't actually wait ...
>
>>       if ( status == -EAGAIN ) {
>>               SDL_Delay(1);
>>               continue;
>>       }
>
> Not necessary in blocking mode.
>
>>       rate = spec->freq;
>>       status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL);
>>       spec->freq = rate;
>
> The returned rate could be wildly different, but I guess SDL correctly
> handles the new value in spec->freq.
>
>>       /* Set the buffer size, in samples */
>>       frames = spec->samples;
>>       status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);
>
> This does _not_ set the buffer size.
>
>>       periods = 2;
>>       SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL);
>
> A bigger number of periods would make the writing of audio data less
> bursty.
>
>>       status = SDL_NAME(snd_pcm_sw_params_set_start_threshold)(pcm_handle, swparams, 0);
>
> Zero doesn't really make sense; the default value 1 would be OK.
>
>>       status = SDL_NAME(snd_pcm_sw_params_set_avail_min)(pcm_handle, swparams, frames);
>
> The default value is the period size anyway, so you can remove this.
> (If you change the buffer size code above to use _set_buffer_size_near,
> this call would be wrong.)
>
>
> Best regards,
> Clemens
>



-- 
	-Sam Lantinga, Founder and President, Galaxy Gameworks LLC

[-- Attachment #2: SDL_alsa_audio.c --]
[-- Type: application/octet-stream, Size: 18375 bytes --]

/*
    SDL - Simple DirectMedia Layer
    Copyright (C) 1997-2009 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 "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"

/* Whether we should set the buffer size or the period size */
/*#define SET_PERIOD_SIZE*/
/*#define DEBUG_PERIOD_SIZE*/

/* 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_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, unsigned int *val);
static 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 int (*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 int (*SDL_NAME(snd_pcm_hw_params_get_period_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *frames, int *dir);
static 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))(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
static int (*SDL_NAME(snd_pcm_hw_params_set_buffer_size_near))(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
static int (*SDL_NAME(snd_pcm_hw_params_get_buffer_size))(const snd_pcm_hw_params_t *params, snd_pcm_uframes_t *val);
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))(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_set_buffer_size_near",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_set_buffer_size_near) },
	{ "snd_pcm_hw_params_get_buffer_size",	(void**)(char*)&SDL_NAME(snd_pcm_hw_params_get_buffer_size) },
	{ "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",	(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);
		alsa_handle = NULL;
		alsa_loaded = 0;
	}
}

static int LoadALSALibrary(void) {
	int i, retval = -1;

	alsa_handle = SDL_LoadObject(alsa_library);
	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 (!*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 ) {
		switch (channels) {
		case 6:
			device = "plug:surround51";
			break;
		case 4:
			device = "plug:surround40";
			break;
		default:
			device = "default";
			break;
		}
	}
	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;
	snd_pcm_uframes_t frames_left;
	const Uint8 *sample_buf = (const Uint8 *) mixbuf;
	const int sample_size = ((int) (this->spec.format & 0xFF)) / 8;

	swizzle_alsa_channels(this);

	frames_left = ((snd_pcm_uframes_t) this->spec.samples);

	while ( frames_left > 0 ) {
		status = SDL_NAME(snd_pcm_writei)(pcm_handle, sample_buf, frames_left);
		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 * sample_size;
		frames_left -= 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;
	unsigned int         rate;
#ifdef SET_PERIOD_SIZE
	unsigned int         periods;
#endif
	unsigned int 	     channels;
	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);
	channels = spec->channels;
	if ( status < 0 ) {
		status = SDL_NAME(snd_pcm_hw_params_get_channels)(hwparams, &channels);
		if ( status < 0 ) {
			SDL_SetError("Couldn't set audio channels");
			ALSA_CloseAudio(this);
			return(-1);
		}
		spec->channels = channels;
	}

	/* Set the audio rate */
	rate = spec->freq;

	status = SDL_NAME(snd_pcm_hw_params_set_rate_near)(pcm_handle, hwparams, &rate, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set audio frequency: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
	spec->freq = rate;

	/* Set the buffer size, in samples */
#ifdef SET_PERIOD_SIZE
	frames = spec->samples;
	status = SDL_NAME(snd_pcm_hw_params_set_period_size_near)(pcm_handle, hwparams, &frames, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set period size: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}

	spec->samples = frames;

	periods = 2;
	status = SDL_NAME(snd_pcm_hw_params_set_periods_near)(pcm_handle, hwparams, &periods, NULL);
	if ( status < 0 ) {
		SDL_SetError("Couldn't set period count: %s", SDL_NAME(snd_strerror)(status));
		ALSA_CloseAudio(this);
		return(-1);
	}
#else
	frames = spec->samples * 2;
	status = SDL_NAME(snd_pcm_hw_params_set_buffer_size_near)(pcm_handle, hwparams, &frames);
#endif

	/* "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... */
#ifdef DEBUG_PERIOD_SIZE
{ snd_pcm_uframes_t bufsize; snd_pcm_sframes_t persize; unsigned int periods; int dir;
   SDL_NAME(snd_pcm_hw_params_get_buffer_size)(hwparams, &bufsize);
   SDL_NAME(snd_pcm_hw_params_get_period_size)(hwparams, &persize, &dir);
   SDL_NAME(snd_pcm_hw_params_get_periods)(hwparams, &periods, &dir);

   fprintf(stderr, "ALSA: period size = %ld, periods = %u, buffer size = %lu\n", persize, periods, bufsize);
}
#endif

	/* 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, 1);
	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)(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);

	/* Get the parent process id (we're the parent of the audio thread) */
	parent = getpid();

	/* Switch to blocking mode for playback */
	SDL_NAME(snd_pcm_nonblock)(pcm_handle, 0);

	/* We're ready to rock and roll. :-) */
	return(0);
}

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