SDL - Simple DirectMedia Layer

Copyright (C) 1997-2006 Sam Lantinga

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

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

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

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

Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA

Carsten Griwodz

griff@kom.tu-darmstadt.de

based on linux/SDL_dspaudio.c by Sam Lantinga

*/

/* Allow access to a raw mixing buffer */

#include <errno.h>

#include <unistd.h>

#include <fcntl.h>

#include <sys/time.h>

#include <sys/ioctl.h>

#include <sys/stat.h>

#include "SDL_stdlib.h"

#include "SDL_string.h"

#include "SDL_audio.h"

#include "SDL_error.h"

#include "SDL_audiomem.h"

#include "SDL_audio_c.h"

#include "SDL_timer.h"

#include "SDL_audiodev_c.h"

#include "SDL_paudio.h"

#define DEBUG_AUDIO 1

/* A conflict within AIX 4.3.3 <sys/> headers and probably others as well.

* I guess nobody ever uses audio... Shame over AIX header files. */

#include <sys/machine.h>

#undef BIG_ENDIAN

#include <sys/audio.h>

/* The tag name used by paud audio */

#define Paud_DRIVER_NAME "paud"

/* Open the audio device for playback, and don't block if busy */

/* #define OPEN_FLAGS (O_WRONLY|O_NONBLOCK) */

#define OPEN_FLAGS O_WRONLY

/* Audio driver functions */

static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec);

static void Paud_WaitAudio(_THIS);

static void Paud_PlayAudio(_THIS);

static Uint8 *Paud_GetAudioBuf(_THIS);

static void Paud_CloseAudio(_THIS);

/* Audio driver bootstrap functions */

static int Audio_Available(void)

{

int fd;

int available;

available = 0;

fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);

if ( fd >= 0 ) {

available = 1;

close(fd);

}

return(available);

}

static void Audio_DeleteDevice(SDL_AudioDevice *device)

{

SDL_free(device->hidden);

SDL_free(device);

}

static SDL_AudioDevice *Audio_CreateDevice(int devindex)

{

SDL_AudioDevice *this;

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

}

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

SDL_OutOfMemory();

if ( this ) {

}

return(0);

}

audio_fd = -1;

/* Set the function pointers */

this->OpenAudio = Paud_OpenAudio;

this->WaitAudio = Paud_WaitAudio;

this->PlayAudio = Paud_PlayAudio;

this->GetAudioBuf = Paud_GetAudioBuf;

this->CloseAudio = Paud_CloseAudio;

this->free = Audio_DeleteDevice;

return this;

}

AudioBootStrap Paud_bootstrap = {

Paud_DRIVER_NAME, "AIX Paudio",

Audio_Available, Audio_CreateDevice

};

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

static void Paud_WaitAudio(_THIS)

{

fd_set fdset;

/* See if we need to use timed audio synchronization */

if ( frame_ticks ) {

/* Use timer for general audio synchronization */

Sint32 ticks;

ticks = ((Sint32)(next_frame - SDL_GetTicks()))-FUDGE_TICKS;

if ( ticks > 0 ) {

SDL_Delay(ticks);

}

} else {

audio_buffer paud_bufinfo;

/* Use select() for audio synchronization */

struct timeval timeout;

FD_ZERO(&fdset);

FD_SET(audio_fd, &fdset);

if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {

#ifdef DEBUG_AUDIO

fprintf(stderr, "Couldn't get audio buffer information\n");

#endif

timeout.tv_sec = 10;

timeout.tv_usec = 0;

} else {

long ms_in_buf = paud_bufinfo.write_buf_time;

timeout.tv_sec = ms_in_buf/1000;

ms_in_buf = ms_in_buf - timeout.tv_sec*1000;

timeout.tv_usec = ms_in_buf*1000;

#ifdef DEBUG_AUDIO

fprintf( stderr,

"Waiting for write_buf_time=%ld,%ld\n",

timeout.tv_sec,

timeout.tv_usec );

#endif

}

#ifdef DEBUG_AUDIO

fprintf(stderr, "Waiting for audio to get ready\n");

#endif

if ( select(audio_fd+1, NULL, &fdset, NULL, &timeout) <= 0 ) {

const char *message = "Audio timeout - buggy audio driver? (disabled)";

/*

* In general we should never print to the screen,

* but in this case we have no other way of letting

* the user know what happened.

*/

fprintf(stderr, "SDL: %s - %s\n", strerror(errno), message);

this->enabled = 0;

/* Don't try to close - may hang */

audio_fd = -1;

#ifdef DEBUG_AUDIO

fprintf(stderr, "Done disabling audio\n");

#endif

}

#ifdef DEBUG_AUDIO

fprintf(stderr, "Ready!\n");

#endif

}

}

static void Paud_PlayAudio(_THIS)

{

int written;

/* Write the audio data, checking for EAGAIN on broken audio drivers */

do {

written = write(audio_fd, mixbuf, mixlen);

if ( (written < 0) && ((errno == 0) || (errno == EAGAIN)) ) {

SDL_Delay(1); /* Let a little CPU time go by */

}

} while ( (written < 0) &&

((errno == 0) || (errno == EAGAIN) || (errno == EINTR)) );

/* If timer synchronization is enabled, set the next write frame */

if ( frame_ticks ) {

next_frame += frame_ticks;

}

/* If we couldn't write, assume fatal error for now */

if ( written < 0 ) {

this->enabled = 0;

}

#ifdef DEBUG_AUDIO

fprintf(stderr, "Wrote %d bytes of audio data\n", written);

#endif

}

static Uint8 *Paud_GetAudioBuf(_THIS)

{

return mixbuf;

}

static void Paud_CloseAudio(_THIS)

{

if ( mixbuf != NULL ) {

SDL_FreeAudioMem(mixbuf);

mixbuf = NULL;

}

if ( audio_fd >= 0 ) {

close(audio_fd);

audio_fd = -1;

}

}

static int Paud_OpenAudio(_THIS, SDL_AudioSpec *spec)

{

char audiodev[1024];

int format;

int bytes_per_sample;

Uint16 test_format;

audio_init paud_init;

audio_buffer paud_bufinfo;

audio_status paud_status;

audio_control paud_control;

audio_change paud_change;

/* Reset the timer synchronization flag */

frame_ticks = 0.0;

/* Open the audio device */

audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 0);

if ( audio_fd < 0 ) {

SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));

return -1;

}

/*

* We can't set the buffer size - just ask the device for the maximum

* that we can have.

*/

if ( ioctl(audio_fd, AUDIO_BUFFER, &paud_bufinfo) < 0 ) {

SDL_SetError("Couldn't get audio buffer information");

return -1;

}

mixbuf = NULL;

if ( spec->channels > 1 )

spec->channels = 2;

else

spec->channels = 1;

/*

* Fields in the audio_init structure:

*

* Ignored by us:

*

* paud.loadpath[LOAD_PATH]; * DSP code to load, MWave chip only?

* paud.slot_number; * slot number of the adapter

* paud.device_id; * adapter identification number

*

* Input:

*

* paud.srate; * the sampling rate in Hz

* paud.bits_per_sample; * 8, 16, 32, ...

* paud.bsize; * block size for this rate

* paud.mode; * ADPCM, PCM, MU_LAW, A_LAW, SOURCE_MIX

* paud.channels; * 1=mono, 2=stereo

* paud.flags; * FIXED - fixed length data

* * LEFT_ALIGNED, RIGHT_ALIGNED (var len only)

* * TWOS_COMPLEMENT - 2's complement data

* * SIGNED - signed? comment seems wrong in sys/audio.h

* * BIG_ENDIAN

* paud.operation; * PLAY, RECORD

*

* Output:

*

* paud.flags; * PITCH - pitch is supported

* * INPUT - input is supported

* * OUTPUT - output is supported

* * MONITOR - monitor is supported

* * VOLUME - volume is supported

* * VOLUME_DELAY - volume delay is supported

* * BALANCE - balance is supported

* * BALANCE_DELAY - balance delay is supported

* * TREBLE - treble control is supported

* * BASS - bass control is supported

* * BESTFIT_PROVIDED - best fit returned

* * LOAD_CODE - DSP load needed

* paud.rc; * NO_PLAY - DSP code can't do play requests

* * NO_RECORD - DSP code can't do record requests

* * INVALID_REQUEST - request was invalid

* * CONFLICT - conflict with open's flags

* * OVERLOADED - out of DSP MIPS or memory

* paud.position_resolution; * smallest increment for position

*/

paud_init.srate = spec->freq;

paud_init.mode = PCM;

paud_init.operation = PLAY;

paud_init.channels = spec->channels;

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

format = 0;

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

! format && test_format; ) {

#ifdef DEBUG_AUDIO

fprintf(stderr, "Trying format 0x%4.4x\n", test_format);

#endif

switch ( test_format ) {

case AUDIO_U8:

bytes_per_sample = 1;

paud_init.bits_per_sample = 8;

paud_init.flags = TWOS_COMPLEMENT | FIXED;

format = 1;

break;

case AUDIO_S8:

bytes_per_sample = 1;

paud_init.bits_per_sample = 8;

paud_init.flags = SIGNED |

TWOS_COMPLEMENT | FIXED;

format = 1;

break;

case AUDIO_S16LSB:

bytes_per_sample = 2;

paud_init.bits_per_sample = 16;

paud_init.flags = SIGNED |

TWOS_COMPLEMENT | FIXED;

format = 1;

break;

case AUDIO_S16MSB:

bytes_per_sample = 2;

paud_init.bits_per_sample = 16;

paud_init.flags = BIG_ENDIAN |

SIGNED |

TWOS_COMPLEMENT | FIXED;

format = 1;

break;

case AUDIO_U16LSB:

bytes_per_sample = 2;

paud_init.bits_per_sample = 16;

paud_init.flags = TWOS_COMPLEMENT | FIXED;

format = 1;

break;

case AUDIO_U16MSB:

bytes_per_sample = 2;

paud_init.bits_per_sample = 16;

paud_init.flags = BIG_ENDIAN |

TWOS_COMPLEMENT | FIXED;

format = 1;

break;

default:

break;

}

if ( ! format ) {

test_format = SDL_NextAudioFormat();

}

}

if ( format == 0 ) {

#ifdef DEBUG_AUDIO

fprintf(stderr, "Couldn't find any hardware audio formats\n");

#endif

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

return -1;

}

spec->format = test_format;

/*

* We know the buffer size and the max number of subsequent writes

* that can be pending. If more than one can pend, allow the application

* to do something like double buffering between our write buffer and

* the device's own buffer that we are filling with write() anyway.

*

* We calculate spec->samples like this because SDL_CalculateAudioSpec()

* will give put paud_bufinfo.write_buf_cap (or paud_bufinfo.write_buf_cap/2)

* into spec->size in return.

*/

if ( paud_bufinfo.request_buf_cap == 1 )

{

spec->samples = paud_bufinfo.write_buf_cap

/ bytes_per_sample

/ spec->channels;

}

else

{

spec->samples = paud_bufinfo.write_buf_cap

/ bytes_per_sample

/ spec->channels

/ 2;

}

paud_init.bsize = bytes_per_sample * spec->channels;

SDL_CalculateAudioSpec(spec);

/*

* The AIX paud device init can't modify the values of the audio_init

* structure that we pass to it. So we don't need any recalculation

* of this stuff and no reinit call as in linux dsp and dma code.

*

* /dev/paud supports all of the encoding formats, so we don't need

* to do anything like reopening the device, either.

*/

if ( ioctl(audio_fd, AUDIO_INIT, &paud_init) < 0 ) {

switch ( paud_init.rc )

{

case 1 :

SDL_SetError("Couldn't set audio format: DSP can't do play requests");

return -1;

break;

case 2 :

SDL_SetError("Couldn't set audio format: DSP can't do record requests");

return -1;

break;

case 4 :

SDL_SetError("Couldn't set audio format: request was invalid");

return -1;

break;

case 5 :

SDL_SetError("Couldn't set audio format: conflict with open's flags");

return -1;

break;

case 6 :

SDL_SetError("Couldn't set audio format: out of DSP MIPS or memory");

return -1;

break;

default :

SDL_SetError("Couldn't set audio format: not documented in sys/audio.h");

return -1;

break;

}

}

/* Allocate mixing buffer */

mixlen = spec->size;

mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);

if ( mixbuf == NULL ) {

return -1;

}

/*

* Set some paramters: full volume, first speaker that we can find.

* Ignore the other settings for now.

*/

paud_change.input = AUDIO_IGNORE; /* the new input source */

paud_change.output = OUTPUT_1; /* EXTERNAL_SPEAKER,INTERNAL_SPEAKER,OUTPUT_1 */

paud_change.monitor = AUDIO_IGNORE; /* the new monitor state */

paud_change.volume = 0x7fffffff; /* volume level [0-0x7fffffff] */

paud_change.volume_delay = AUDIO_IGNORE; /* the new volume delay */

paud_change.balance = 0x3fffffff; /* the new balance */

paud_change.balance_delay = AUDIO_IGNORE; /* the new balance delay */

paud_change.treble = AUDIO_IGNORE; /* the new treble state */

paud_change.bass = AUDIO_IGNORE; /* the new bass state */

paud_change.pitch = AUDIO_IGNORE; /* the new pitch state */

paud_control.ioctl_request = AUDIO_CHANGE;

paud_control.request_info = (char*)&paud_change;

if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {

#ifdef DEBUG_AUDIO

fprintf(stderr, "Can't change audio display settings\n" );

#endif

}

/*

* Tell the device to expect data. Actual start will wait for

* the first write() call.

*/

paud_control.ioctl_request = AUDIO_START;

paud_control.position = 0;

if ( ioctl(audio_fd, AUDIO_CONTROL, &paud_control) < 0 ) {

#ifdef DEBUG_AUDIO

fprintf(stderr, "Can't start audio play\n" );

#endif

SDL_SetError("Can't start audio play");

return -1;

}

/* Check to see if we need to use select() workaround */

{ char *workaround;

if ( workaround ) {

frame_ticks = (float)(spec->samples*1000)/spec->freq;

next_frame = SDL_GetTicks()+frame_ticks;

}

}

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

parent = getpid();

/* We're ready to rock and roll. :-) */

return 0;

}