示例#1
0
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 int Audio_Available(void)
{
	int available;
	int fd;

	available = 0;

	fd = SDL_OpenAudioPath(NULL, 0, OPEN_FLAGS, 0);
	if ( fd >= 0 ) {
		int caps;
		struct audio_buf_info info;

		if ( (ioctl(fd, SNDCTL_DSP_GETCAPS, &caps) == 0) &&
	             (caps & DSP_CAP_TRIGGER) && (caps & DSP_CAP_MMAP) &&
		     (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info) == 0) ) {
			available = 1;
		}
		close(fd);
	}
	return(available);
}
static int DMA_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
	char audiodev[1024];
	int format;
	int stereo;
	int value;
	Uint16 test_format;
	struct audio_buf_info info;

	/* 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);
	}
	dma_buf = NULL;
	ioctl(audio_fd, SNDCTL_DSP_RESET, 0);

	/* Get a list of supported hardware formats */
	if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) {
		SDL_SetError("Couldn't get audio format list");
		return(-1);
	}

	/* 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:
				if ( value & AFMT_U8 ) {
					format = AFMT_U8;
				}
				break;
			case AUDIO_S8:
				if ( value & AFMT_S8 ) {
					format = AFMT_S8;
				}
				break;
			case AUDIO_S16LSB:
				if ( value & AFMT_S16_LE ) {
					format = AFMT_S16_LE;
				}
				break;
			case AUDIO_S16MSB:
				if ( value & AFMT_S16_BE ) {
					format = AFMT_S16_BE;
				}
				break;
			case AUDIO_U16LSB:
				if ( value & AFMT_U16_LE ) {
					format = AFMT_U16_LE;
				}
				break;
			case AUDIO_U16MSB:
				if ( value & AFMT_U16_BE ) {
					format = AFMT_U16_BE;
				}
				break;
			default:
				format = 0;
				break;
		}
		if ( ! format ) {
			test_format = SDL_NextAudioFormat();
		}
	}
	if ( format == 0 ) {
		SDL_SetError("Couldn't find any hardware audio formats");
		return(-1);
	}
	spec->format = test_format;

	/* Set the audio format */
	value = format;
	if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
						(value != format) ) {
		SDL_SetError("Couldn't set audio format");
		return(-1);
	}

	/* Set mono or stereo audio (currently only two channels supported) */
	stereo = (spec->channels > 1);
	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
	if ( stereo ) {
		spec->channels = 2;
	} else {
		spec->channels = 1;
	}

	/* Because some drivers don't allow setting the buffer size
	   after setting the format, we must re-open the audio device
	   once we know what format and channels are supported
	 */
	if ( DMA_ReopenAudio(this, audiodev, format, stereo, spec) < 0 ) {
		/* Error is set by DMA_ReopenAudio() */
		return(-1);
	}

	/* Memory map the audio buffer */
	if ( ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info) < 0 ) {
		SDL_SetError("Couldn't get OSPACE parameters");
		return(-1);
	}
	spec->size = info.fragsize;
	spec->samples = spec->size / ((spec->format & 0xFF) / 8);
	spec->samples /= spec->channels;
	num_buffers = info.fragstotal;
	dma_len = num_buffers*spec->size;
	dma_buf = (Uint8 *)mmap(NULL, dma_len, PROT_WRITE, MAP_SHARED,
							audio_fd, 0);
	if ( dma_buf == MAP_FAILED ) {
		SDL_SetError("DMA memory map failed");
		dma_buf = NULL;
		return(-1);
	}
	SDL_memset(dma_buf, spec->silence, dma_len);

	/* 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;
		}
	}

	/* Trigger audio playback */
	value = 0;
	ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value);
	value = PCM_ENABLE_OUTPUT;
	if ( ioctl(audio_fd, SNDCTL_DSP_SETTRIGGER, &value) < 0 ) {
		SDL_SetError("Couldn't trigger audio output");
		return(-1);
	}

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

	/* We're ready to rock and roll. :-) */
	return(0);
}
示例#4
0
static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
	char audiodev[1024];
	int format;
	int value;
	Uint16 test_format;

	/* 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);
	}
	mixbuf = NULL;

#ifdef USE_BLOCKING_WRITES
	/* Make the file descriptor use blocking writes with fcntl() */
	{ long flags;
		flags = fcntl(audio_fd, F_GETFL);
		flags &= ~O_NONBLOCK;
		if ( fcntl(audio_fd, F_SETFL, flags) < 0 ) {
			SDL_SetError("Couldn't set audio blocking mode");
			return(-1);
		}
	}
#endif

	/* Get a list of supported hardware formats */
	if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) {
		SDL_SetError("Couldn't get audio format list");
		return(-1);
	}

	/* 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:
				if ( value & AFMT_U8 ) {
					format = AFMT_U8;
				}
				break;
			case AUDIO_S8:
				if ( value & AFMT_S8 ) {
					format = AFMT_S8;
				}
				break;
			case AUDIO_S16LSB:
				if ( value & AFMT_S16_LE ) {
					format = AFMT_S16_LE;
				}
				break;
			case AUDIO_S16MSB:
				if ( value & AFMT_S16_BE ) {
					format = AFMT_S16_BE;
				}
				break;
			case AUDIO_U16LSB:
				if ( value & AFMT_U16_LE ) {
					format = AFMT_U16_LE;
				}
				break;
			case AUDIO_U16MSB:
				if ( value & AFMT_U16_BE ) {
					format = AFMT_U16_BE;
				}
				break;
			default:
				format = 0;
				break;
		}
		if ( ! format ) {
			test_format = SDL_NextAudioFormat();
		}
	}
	if ( format == 0 ) {
		SDL_SetError("Couldn't find any hardware audio formats");
		return(-1);
	}
	spec->format = test_format;

	/* Set the audio format */
	value = format;
	if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
						(value != format) ) {
		SDL_SetError("Couldn't set audio format");
		return(-1);
	}

	/* Set the number of channels of output */
	value = spec->channels;
#ifdef SNDCTL_DSP_CHANNELS
	if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) {
#endif
		value = (spec->channels > 1);
		ioctl(audio_fd, SNDCTL_DSP_STEREO, &value);
		value = (value ? 2 : 1);
#ifdef SNDCTL_DSP_CHANNELS
	}
#endif
	spec->channels = value;

	/* Because some drivers don't allow setting the buffer size
	   after setting the format, we must re-open the audio device
	   once we know what format and channels are supported
	 */
	if ( DSP_ReopenAudio(this, audiodev, format, spec) < 0 ) {
		/* Error is set by DSP_ReopenAudio() */
		return(-1);
	}

	/* Allocate mixing buffer */
	mixlen = spec->size;
	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
	if ( mixbuf == NULL ) {
		return(-1);
	}
	memset(mixbuf, spec->silence, spec->size);

#ifndef USE_BLOCKING_WRITES
	/* Check to see if we need to use select() workaround */
	{ char *workaround;
		workaround = getenv("SDL_DSP_NOSELECT");
		if ( workaround ) {
			frame_ticks = (float)(spec->samples*1000)/spec->freq;
			next_frame = SDL_GetTicks()+frame_ticks;
		}
	}
#endif /* !USE_BLOCKING_WRITES */

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

	/* We're ready to rock and roll. :-) */
	return(0);
}
示例#5
0
int
DSP_OpenAudio(_THIS, SDL_AudioSpec * spec)
{
    char audiodev[1024];
#ifdef AUDIO_SETINFO
    int enc;
#endif
    int desired_freq = spec->freq;

    /* Initialize our freeable variables, in case we fail */
    audio_fd = -1;
    mixbuf = NULL;
    ulaw_buf = NULL;

    /* Determine the audio parameters from the AudioSpec */
    switch (SDL_AUDIO_BITSIZE(spec->format)) {

    case 8:
        {                       /* Unsigned 8 bit audio data */
            spec->format = AUDIO_U8;
#ifdef AUDIO_SETINFO
            enc = AUDIO_ENCODING_LINEAR8;
#endif
        }
        break;

    case 16:
        {                       /* Signed 16 bit audio data */
            spec->format = AUDIO_S16SYS;
#ifdef AUDIO_SETINFO
            enc = AUDIO_ENCODING_LINEAR;
#endif
        }
        break;

    default:
        {
            /* !!! FIXME: fallback to conversion on unsupported types! */
            SDL_SetError("Unsupported audio format");
            return (-1);
        }
    }
    audio_fmt = spec->format;

    /* Open the audio device */
    audio_fd = SDL_OpenAudioPath(audiodev, sizeof(audiodev), OPEN_FLAGS, 1);
    if (audio_fd < 0) {
        SDL_SetError("Couldn't open %s: %s", audiodev, strerror(errno));
        return (-1);
    }

    ulaw_only = 0;              /* modern Suns do support linear audio */
#ifdef AUDIO_SETINFO
    for (;;) {
        audio_info_t info;
        AUDIO_INITINFO(&info);  /* init all fields to "no change" */

        /* Try to set the requested settings */
        info.play.sample_rate = spec->freq;
        info.play.channels = spec->channels;
        info.play.precision = (enc == AUDIO_ENCODING_ULAW)
            ? 8 : spec->format & 0xff;
        info.play.encoding = enc;
        if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0) {

            /* Check to be sure we got what we wanted */
            if (ioctl(audio_fd, AUDIO_GETINFO, &info) < 0) {
                SDL_SetError("Error getting audio parameters: %s",
                             strerror(errno));
                return -1;
            }
            if (info.play.encoding == enc
                && info.play.precision == (spec->format & 0xff)
                && info.play.channels == spec->channels) {
                /* Yow! All seems to be well! */
                spec->freq = info.play.sample_rate;
                break;
            }
        }

        switch (enc) {
        case AUDIO_ENCODING_LINEAR8:
            /* unsigned 8bit apparently not supported here */
            enc = AUDIO_ENCODING_LINEAR;
            spec->format = AUDIO_S16SYS;
            break;              /* try again */

        case AUDIO_ENCODING_LINEAR:
            /* linear 16bit didn't work either, resort to µ-law */
            enc = AUDIO_ENCODING_ULAW;
            spec->channels = 1;
            spec->freq = 8000;
            spec->format = AUDIO_U8;
            ulaw_only = 1;
            break;

        default:
            /* oh well... */
            SDL_SetError("Error setting audio parameters: %s",
                         strerror(errno));
            return -1;
        }
    }
#endif /* AUDIO_SETINFO */
    written = 0;

    /* We can actually convert on-the-fly to U-Law */
    if (ulaw_only) {
        spec->freq = desired_freq;
        fragsize = (spec->samples * 1000) / (spec->freq / 8);
        frequency = 8;
        ulaw_buf = (Uint8 *) SDL_malloc(fragsize);
        if (ulaw_buf == NULL) {
            SDL_OutOfMemory();
            return (-1);
        }
        spec->channels = 1;
    } else {
        fragsize = spec->samples;
        frequency = spec->freq / 1000;
    }
#ifdef DEBUG_AUDIO
    fprintf(stderr, "Audio device %s U-Law only\n",
            ulaw_only ? "is" : "is not");
    fprintf(stderr, "format=0x%x chan=%d freq=%d\n",
            spec->format, spec->channels, spec->freq);
#endif

    /* Update the fragment size as size in bytes */
    SDL_CalculateAudioSpec(spec);

    /* Allocate mixing buffer */
    mixbuf = (Uint8 *) SDL_AllocAudioMem(spec->size);
    if (mixbuf == NULL) {
        SDL_OutOfMemory();
        return (-1);
    }
    SDL_memset(mixbuf, spec->silence, spec->size);

    /* We're ready to rock and roll. :-) */
    return (0);
}
示例#6
0
static int DSP_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
	char audiodev[1024];
	int format;
	int value;
	int frag_spec;
	Uint16 test_format;

	/* 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);
	}
	mixbuf = NULL;

	/* Make the file descriptor use blocking writes with fcntl() */
	{ long flags;
		flags = fcntl(audio_fd, F_GETFL);
		flags &= ~O_NONBLOCK;
		if ( fcntl(audio_fd, F_SETFL, flags) < 0 ) {
			SDL_SetError("Couldn't set audio blocking mode");
			return(-1);
		}
	}

	/* Get a list of supported hardware formats */
	if ( ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &value) < 0 ) {
		perror("SNDCTL_DSP_GETFMTS");
		SDL_SetError("Couldn't get audio format list");
		return(-1);
	}

	/* 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:
				if ( value & AFMT_U8 ) {
					format = AFMT_U8;
				}
				break;
			case AUDIO_S16LSB:
				if ( value & AFMT_S16_LE ) {
					format = AFMT_S16_LE;
				}
				break;
			case AUDIO_S16MSB:
				if ( value & AFMT_S16_BE ) {
					format = AFMT_S16_BE;
				}
				break;
#if 0
/*
 * These formats are not used by any real life systems so they are not 
 * needed here.
 */
			case AUDIO_S8:
				if ( value & AFMT_S8 ) {
					format = AFMT_S8;
				}
				break;
			case AUDIO_U16LSB:
				if ( value & AFMT_U16_LE ) {
					format = AFMT_U16_LE;
				}
				break;
			case AUDIO_U16MSB:
				if ( value & AFMT_U16_BE ) {
					format = AFMT_U16_BE;
				}
				break;
#endif
			default:
				format = 0;
				break;
		}
		if ( ! format ) {
			test_format = SDL_NextAudioFormat();
		}
	}
	if ( format == 0 ) {
		SDL_SetError("Couldn't find any hardware audio formats");
		return(-1);
	}
	spec->format = test_format;

	/* Set the audio format */
	value = format;
	if ( (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &value) < 0) ||
						(value != format) ) {
		perror("SNDCTL_DSP_SETFMT");
		SDL_SetError("Couldn't set audio format");
		return(-1);
	}

	/* Set the number of channels of output */
	value = spec->channels;
	if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &value) < 0 ) {
		perror("SNDCTL_DSP_CHANNELS");
		SDL_SetError("Cannot set the number of channels");
		return(-1);
	}
	spec->channels = value;

	/* Set the DSP frequency */
	value = spec->freq;
	if ( ioctl(audio_fd, SNDCTL_DSP_SPEED, &value) < 0 ) {
		perror("SNDCTL_DSP_SPEED");
		SDL_SetError("Couldn't set audio frequency");
		return(-1);
	}
	spec->freq = value;

	/* Calculate the final parameters for this audio specification */
	SDL_CalculateAudioSpec(spec);

	/* Determine the power of two of the fragment size */
	for ( frag_spec = 0; (0x01<<frag_spec) < spec->size; ++frag_spec );
	if ( (0x01<<frag_spec) != spec->size ) {
		SDL_SetError("Fragment size must be a power of two");
		return(-1);
	}
	frag_spec |= 0x00020000;	/* two fragments, for low latency */

	/* Set the audio buffering parameters */
#ifdef DEBUG_AUDIO
	fprintf(stderr, "Requesting %d fragments of size %d\n",
		(frag_spec >> 16), 1<<(frag_spec&0xFFFF));
#endif
	if ( ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag_spec) < 0 ) {
		perror("SNDCTL_DSP_SETFRAGMENT");
		fprintf(stderr, "Warning: Couldn't set audio fragment size\n");
	}
#ifdef DEBUG_AUDIO
	{ audio_buf_info info;
	  ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &info);
	  fprintf(stderr, "fragments = %d\n", info.fragments);
	  fprintf(stderr, "fragstotal = %d\n", info.fragstotal);
	  fprintf(stderr, "fragsize = %d\n", info.fragsize);
	  fprintf(stderr, "bytes = %d\n", info.bytes);
	}
#endif

	/* Allocate mixing buffer */
	mixlen = spec->size;
	mixbuf = (Uint8 *)SDL_AllocAudioMem(mixlen);
	if ( mixbuf == NULL ) {
		return(-1);
	}
	memset(mixbuf, spec->silence, spec->size);

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

	/* We're ready to rock and roll. :-) */
	return(0);
}
示例#7
0
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;
}
示例#8
0
static int
OBSD_OpenAudio(_THIS, SDL_AudioSpec *spec)
{
    char audiodev[64];
    Uint16 format;
    audio_info_t info;

    AUDIO_INITINFO(&info);
    
    
    SDL_CalculateAudioSpec(spec);

#ifdef USE_TIMER_SYNC
    frame_ticks = 0.0;
#endif

    
    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);
    }
    
    
    info.mode = AUMODE_PLAY;
    if(ioctl(audio_fd, AUDIO_SETINFO, &info) < 0) {
	SDL_SetError("Couldn't put device into play mode");
	return(-1);
    }
    
    mixbuf = NULL;
    AUDIO_INITINFO(&info);
    for (format = SDL_FirstAudioFormat(spec->format); 
    	format; format = SDL_NextAudioFormat())
    {
	switch(format) {
	case AUDIO_U8:
	    info.play.encoding = AUDIO_ENCODING_ULINEAR;
	    info.play.precision = 8;
	    break;
	case AUDIO_S8:
	    info.play.encoding = AUDIO_ENCODING_SLINEAR;
	    info.play.precision = 8;
	    break;
	case AUDIO_S16LSB:
	    info.play.encoding = AUDIO_ENCODING_SLINEAR_LE;
	    info.play.precision = 16;
	    break;
	case AUDIO_S16MSB:
	    info.play.encoding = AUDIO_ENCODING_SLINEAR_BE;
	    info.play.precision = 16;
	    break;
	case AUDIO_U16LSB:
	    info.play.encoding = AUDIO_ENCODING_ULINEAR_LE;
	    info.play.precision = 16;
	    break;
	case AUDIO_U16MSB:
	    info.play.encoding = AUDIO_ENCODING_ULINEAR_BE;
	    info.play.precision = 16;
	    break;
	default:
	    continue;
	}
	if (ioctl(audio_fd, AUDIO_SETINFO, &info) == 0)
	    break;
    }

    if(!format) {
	SDL_SetError("No supported encoding for 0x%x", spec->format);
	return(-1);
    }

    spec->format = format;

    AUDIO_INITINFO(&info);
    info.play.channels = spec->channels;
    if (ioctl(audio_fd, AUDIO_SETINFO, &info) == -1)
    	spec->channels = 1;
    AUDIO_INITINFO(&info);
    info.play.sample_rate = spec->freq;
    info.blocksize = spec->size;
    info.hiwat = 5;
    info.lowat = 3;
    (void)ioctl(audio_fd, AUDIO_SETINFO, &info);
    (void)ioctl(audio_fd, AUDIO_GETINFO, &info);
    spec->freq  = info.play.sample_rate;
    
    mixlen = spec->size;
    mixbuf = (Uint8*)SDL_AllocAudioMem(mixlen);
    if(mixbuf == NULL) {
	return(-1);
    }
    SDL_memset(mixbuf, spec->silence, spec->size);
    
    
    parent = getpid();

#ifdef DEBUG_AUDIO
    OBSD_Status(this);
#endif

    
    return(0);
}