view src/audio/paudio/SDL_paudio.c @ 4168:69bcba65c388 SDL-1.2

Fixed bug #526 Comment #1 From Simon Howard 2009-03-20 16:50:56 Hi, I'm the author of Chocolate Doom, one of the other source ports that James mentioned. This is a patch against the current SVN version of SDL 1.2 that fixes the bug. It has been tested and hopefully should be obviously correct from examining the changes. I'll give a brief explanation. When the palette is set with SDL_SetPalette, the IDirectDrawPalette_SetEntries DirectX function is invoked. However, when this happens, a WM_PALETTECHANGED message is sent to the window. A WM_PALETTECHANGED message can also be received if the palette is changed for some other reason, like if the system palette is changed. Therefore, the palette change handler (DX5_PaletteChanged) has code to deal with this case. It distinguishes "expected" palette changes (set with SDL_SetPalette) from "unexpected" palette changes using the colorchange_expected variable, which is set before calling IDirectDrawPalette_SetEntries. However, the code to set this variable is missing in the fullscreen code path. By setting this variable, the palette change is handled properly and the freezes go away.
author Sam Lantinga <slouken@libsdl.org>
date Mon, 13 Apr 2009 00:53:12 +0000
parents a1b03ba2fcd0
children
line wrap: on
line source

/*
    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 Lesser General Public
    License as published by the Free Software Foundation; either
    version 2.1 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
    Lesser General Public License for more details.

    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
*/
#include "SDL_config.h"

/* 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_timer.h"
#include "SDL_audio.h"
#include "../SDL_audiomem.h"
#include "../SDL_audio_c.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 */
	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));
	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;
	}
	SDL_memset(mixbuf, spec->silence, spec->size);

	/*
	 * 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;
                workaround = SDL_getenv("SDL_DSP_NOSELECT");
                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;
}