自学教程：C++ AUDIO_INITINFO函数代码示例

static int pa_solaris_auto_format(int fd, int mode, pa_sample_spec *ss) {    audio_info_t info;    AUDIO_INITINFO(&info);    if (mode != O_RDONLY) {        info.play.sample_rate = ss->rate;        info.play.channels = ss->channels;        switch (ss->format) {        case PA_SAMPLE_U8:            info.play.precision = 8;            info.play.encoding = AUDIO_ENCODING_LINEAR;            break;        case PA_SAMPLE_ALAW:            info.play.precision = 8;            info.play.encoding = AUDIO_ENCODING_ALAW;            break;        case PA_SAMPLE_ULAW:            info.play.precision = 8;            info.play.encoding = AUDIO_ENCODING_ULAW;            break;        case PA_SAMPLE_S16NE:            info.play.precision = 16;            info.play.encoding = AUDIO_ENCODING_LINEAR;            break;        default:            return -1;        }    }    if (mode != O_WRONLY) {        info.record.sample_rate = ss->rate;        info.record.channels = ss->channels;        switch (ss->format) {        case PA_SAMPLE_U8:            info.record.precision = 8;            info.record.encoding = AUDIO_ENCODING_LINEAR;            break;        case PA_SAMPLE_ALAW:            info.record.precision = 8;            info.record.encoding = AUDIO_ENCODING_ALAW;            break;        case PA_SAMPLE_ULAW:            info.record.precision = 8;            info.record.encoding = AUDIO_ENCODING_ULAW;            break;        case PA_SAMPLE_S16NE:            info.record.precision = 16;            info.record.encoding = AUDIO_ENCODING_LINEAR;            break;        default:            return -1;        }    }    if (ioctl(fd, AUDIO_SETINFO, &info) < 0) {        if (errno == EINVAL)            pa_log("AUDIO_SETINFO: Unsupported sample format.");        else            pa_log("AUDIO_SETINFO: %s", pa_cstrerror(errno));        return -1;    }    return 0;}

/********************************************************************* * Try to open the named device. * If it opens, try to set various rates and formats and fill in * the device info structure. */PaError Pa_QueryDevice( const char *deviceName, internalPortAudioDevice *pad ){    int result = paHostError;    int tempDevHandle;    int numChannels, maxNumChannels;    int numSampleRates;    int sampleRate;    int numRatesToTry;    int ratesToTry[9] = {96000, 48000, 44100, 32000, 24000, 22050, 16000, 11025, 8000};    int i;    audio_info_t solaris_info;    audio_device_t device_info;    /* douglas:     we have to do this querying in a slightly different order. apparently     some sound cards will give you different info based on their settins.     e.g. a card might give you stereo at 22kHz but only mono at 44kHz.     the correct order for OSS is: format, channels, sample rate    */	/*	 to check a device for it's capabilities, it's probably better to use the	 equivalent "-ctl"-descriptor - MR	*/    char devname[strlen(deviceName) + 4];    snprintf(devname,(strlen(deviceName) + 4),"%sctl",deviceName);    if ((tempDevHandle = open(devname, O_WRONLY|O_NONBLOCK))  == -1 )    {        DBUG(("Pa_QueryDevice: could not open %s/n", deviceName ));        return paHostError;    }    /*  Ask OSS what formats are supported by the hardware. */    pad->pad_Info.nativeSampleFormats = 0;    AUDIO_INITINFO(&solaris_info);    /* SAM 12/31/01: Sparc native does mulaw, alaw and PCM.       I think PCM is signed. */    for (i = 8; i <= 32; i += 8) {      solaris_info.play.precision = i;      solaris_info.play.encoding = AUDIO_ENCODING_LINEAR;      /* If there are no errors, add the format. */      if (ioctl(tempDevHandle, AUDIO_SETINFO, &solaris_info) > -1) {	switch (i) {	case 8:	  pad->pad_Info.nativeSampleFormats |= paInt8;	  break;	case 16:	  pad->pad_Info.nativeSampleFormats |= paInt16;	  break;	case 24:	  pad->pad_Info.nativeSampleFormats |= paInt24;	  break;	case 32:	  pad->pad_Info.nativeSampleFormats |= paInt32;	  break;	}      }    }    maxNumChannels = 0;    for( numChannels = 1; numChannels <= 16; numChannels++ )      {	int temp = numChannels;	DBUG(("Pa_QueryDevice: use SNDCTL_DSP_CHANNELS, numChannels = %d/n", numChannels ))	  AUDIO_INITINFO(&solaris_info);	solaris_info.play.channels = temp;	if (ioctl(tempDevHandle, AUDIO_SETINFO, &solaris_info) < 0)	  {	    /* ioctl() failed so bail out if we already have stereo */	    if( numChannels > 2 ) break;	  }	else	  {	    /* ioctl() worked but bail out if it does not support numChannels.	     * We don't want to leave gaps in the numChannels supported.	     */	    if( (numChannels > 2) && (temp != numChannels) ) break;	    DBUG(("Pa_QueryDevice: temp = %d/n", temp ))	      if( temp > maxNumChannels ) maxNumChannels = temp; /* Save maximum. */	  }      }    pad->pad_Info.maxOutputChannels = maxNumChannels;    DBUG(("Pa_QueryDevice: maxNumChannels = %d/n", maxNumChannels))    /* FIXME - for now, assume maxInputChannels = maxOutputChannels.     *    Eventually do separate queries for O_WRONLY and O_RDONLY    */    pad->pad_Info.maxInputChannels = pad->pad_Info.maxOutputChannels;    DBUG(("Pa_QueryDevice: maxInputChannels = %d/n",          pad->pad_Info.maxInputChannels))//.........这里部分代码省略.........

static void input_sound(unsigned int sample_rate, unsigned int overlap,                        const char *ifname){    audio_info_t audioinfo;    audio_info_t audioinfo2;    audio_device_t audiodev;    int fd;    short buffer[8192];    float fbuf[16384];    unsigned int fbuf_cnt = 0;    int i;    short *sp;        if ((fd = open(ifname ? ifname : "/dev/audio", O_RDONLY)) < 0) {        perror("open");        exit (10);    }    if (ioctl(fd, AUDIO_GETDEV, &audiodev) == -1) {        perror("ioctl: AUDIO_GETDEV");        exit (10);    }    AUDIO_INITINFO(&audioinfo);    audioinfo.record.sample_rate = sample_rate;    audioinfo.record.channels = 1;    audioinfo.record.precision = 16;    audioinfo.record.encoding = AUDIO_ENCODING_LINEAR;    /*audioinfo.record.gain = 0x20;      audioinfo.record.port = AUDIO_LINE_IN;      audioinfo.monitor_gain = 0;*/    if (ioctl(fd, AUDIO_SETINFO, &audioinfo) == -1) {        perror("ioctl: AUDIO_SETINFO");        exit (10);    }    if (ioctl(fd, I_FLUSH, FLUSHR) == -1) {        perror("ioctl: I_FLUSH");        exit (10);    }    if (ioctl(fd, AUDIO_GETINFO, &audioinfo2) == -1) {        perror("ioctl: AUDIO_GETINFO");        exit (10);    }    fprintf(stdout, "Audio device: name %s, ver %s, config %s, "            "sampling rate %d/n", audiodev.name, audiodev.version,            audiodev.config, audioinfo.record.sample_rate);    for (;;) {        i = read(fd, sp = buffer, sizeof(buffer));        if (i < 0 && errno != EAGAIN) {            perror("read");            exit(4);        }        if (!i)            break;        if (i > 0) {            if(integer_only)        {                fbuf_cnt = i/sizeof(buffer[0]);        }            else            {                for (; i >= sizeof(buffer[0]); i -= sizeof(buffer[0]), sp++)                    fbuf[fbuf_cnt++] = (*sp) * (1.0/32768.0);                if (i)                    fprintf(stderr, "warning: noninteger number of samples read/n");            }            if (fbuf_cnt > overlap) {                process_buffer(fbuf, buffer, fbuf_cnt-overlap);                memmove(fbuf, fbuf+fbuf_cnt-overlap, overlap*sizeof(fbuf[0]));                fbuf_cnt = overlap;            }        }    }    close(fd);}

static intSUNAUDIO_OpenDevice(_THIS, void *handle, const char *devname, int iscapture){    const int flags = ((iscapture) ? OPEN_FLAGS_INPUT : OPEN_FLAGS_OUTPUT);    SDL_AudioFormat format = 0;    audio_info_t info;    /* We don't care what the devname is...we'll try to open anything. */    /*  ...but default to first name in the list... */    if (devname == NULL) {        devname = SDL_GetAudioDeviceName(0, iscapture);        if (devname == NULL) {            return SDL_SetError("No such audio device");        }    }    /* Initialize all variables that we clean on shutdown */    this->hidden = (struct SDL_PrivateAudioData *)        SDL_malloc((sizeof *this->hidden));    if (this->hidden == NULL) {        return SDL_OutOfMemory();    }    SDL_memset(this->hidden, 0, (sizeof *this->hidden));    /* Open the audio device */    this->hidden->audio_fd = open(devname, flags, 0);    if (this->hidden->audio_fd < 0) {        return SDL_SetError("Couldn't open %s: %s", devname, strerror(errno));    }#ifdef AUDIO_SETINFO    int enc;#endif    int desired_freq = this->spec.freq;    /* Determine the audio parameters from the AudioSpec */    switch (SDL_AUDIO_BITSIZE(this->spec.format)) {    case 8:        {                       /* Unsigned 8 bit audio data */            this->spec.format = AUDIO_U8;#ifdef AUDIO_SETINFO            enc = AUDIO_ENCODING_LINEAR8;#endif        }        break;    case 16:        {                       /* Signed 16 bit audio data */            this->spec.format = AUDIO_S16SYS;#ifdef AUDIO_SETINFO            enc = AUDIO_ENCODING_LINEAR;#endif        }        break;    default:        {            /* !!! FIXME: fallback to conversion on unsupported types! */            return SDL_SetError("Unsupported audio format");        }    }    this->hidden->audio_fmt = this->spec.format;    this->hidden->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 = this->spec.freq;        info.play.channels = this->spec.channels;        info.play.precision = (enc == AUDIO_ENCODING_ULAW)            ? 8 : this->spec.format & 0xff;        info.play.encoding = enc;        if (ioctl(this->hidden->audio_fd, AUDIO_SETINFO, &info) == 0) {            /* Check to be sure we got what we wanted */            if (ioctl(this->hidden->audio_fd, AUDIO_GETINFO, &info) < 0) {                return SDL_SetError("Error getting audio parameters: %s",                                    strerror(errno));            }            if (info.play.encoding == enc                && info.play.precision == (this->spec.format & 0xff)                && info.play.channels == this->spec.channels) {                /* Yow! All seems to be well! */                this->spec.freq = info.play.sample_rate;                break;            }        }        switch (enc) {        case AUDIO_ENCODING_LINEAR8:            /* unsigned 8bit apparently not supported here */            enc = AUDIO_ENCODING_LINEAR;            this->spec.format = AUDIO_S16SYS;            break;              /* try again */        case AUDIO_ENCODING_LINEAR://.........这里部分代码省略.........

FILE *out_file_open(char *outFile, int rate, int *channels){   FILE *fout=NULL;   /*Open output file*/   if (strlen(outFile)==0)   {#if defined HAVE_SYS_SOUNDCARD_H      int audio_fd, format, stereo;      audio_fd=open("/dev/dsp", O_WRONLY);      if (audio_fd<0)      {         perror("Cannot open /dev/dsp");         exit(1);               }     format=AFMT_S16_NE;//       format=AFMT_S16_LE;      if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format)==-1)      {         perror("SNDCTL_DSP_SETFMT");         close(audio_fd);         exit(1);      }      stereo=0;      if (*channels==2)         stereo=1;      if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo)==-1)      {         perror("SNDCTL_DSP_STEREO");         close(audio_fd);         exit(1);      }      if (stereo!=0)      {         if (*channels==1)            fprintf (stderr, "Cannot set mono mode, will decode in stereo/n");         *channels=2;      }      if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate)==-1)      {         perror("SNDCTL_DSP_SPEED");         close(audio_fd);         exit(1);      }      fout = fdopen(audio_fd, "w");	  	  printf ("/dev/dsp opened");	  #elif defined HAVE_SYS_AUDIOIO_H      audio_info_t info;      int audio_fd;            audio_fd = open("/dev/audio", O_WRONLY);      if (audio_fd<0)      {         perror("Cannot open /dev/audio");         exit(1);      }      AUDIO_INITINFO(&info);#ifdef AUMODE_PLAY    /* NetBSD/OpenBSD */      info.mode = AUMODE_PLAY;#endif      info.play.encoding = AUDIO_ENCODING_SLINEAR;      info.play.precision = 16;      info.play.sample_rate = rate;      info.play.channels = *channels;            if (ioctl(audio_fd, AUDIO_SETINFO, &info) < 0)      {         perror ("AUDIO_SETINFO");         exit(1);      }      fout = fdopen(audio_fd, "w");#elif defined WIN32 || defined _WIN32      {         unsigned int speex_channels = *channels;         if (Set_WIN_Params (INVALID_FILEDESC, rate, SAMPLE_SIZE, speex_channels))         {            fprintf (stderr, "Can't access %s/n", "WAVE OUT");            exit(1);         }      }#else      fprintf (stderr, "No soundcard support/n");      exit(1);#endif   }       return fout;}

static intsio_sun_setpar(struct sio_hdl *sh, struct sio_par *par){#define NRETRIES 8	struct sio_sun_hdl *hdl = (struct sio_sun_hdl *)sh;	struct audio_info aui;	unsigned int i, infr, ibpf, onfr, obpf;	unsigned int bufsz, round;	unsigned int rate, req_rate, prec, enc;	/*	 * try to set parameters until the device accepts	 * a common encoding and rate for play and record	 */	rate = par->rate;	prec = par->bits;	sio_sun_enctoinfo(hdl, &enc, par);	for (i = 0;; i++) {		if (i == NRETRIES) {			DPRINTF("sio_sun_setpar: couldn't set parameters/n");			hdl->sio.eof = 1;			return 0;		}		AUDIO_INITINFO(&aui);		if (hdl->sio.mode & SIO_PLAY) {			aui.play.sample_rate = rate;			aui.play.precision = prec;			aui.play.encoding = enc;			aui.play.channels = par->pchan;		}		if (hdl->sio.mode & SIO_REC) {			aui.record.sample_rate = rate;			aui.record.precision = prec;			aui.record.encoding = enc;			aui.record.channels = par->rchan;		}		DPRINTFN(2, "sio_sun_setpar: %i: trying pars = %u/%u/%u/n",		    i, rate, prec, enc);		if (ioctl(hdl->fd, AUDIO_SETINFO, &aui) < 0 && errno != EINVAL) {			DPERROR("sio_sun_setpar: setinfo(pars)");			hdl->sio.eof = 1;			return 0;		}		if (ioctl(hdl->fd, AUDIO_GETINFO, &aui) < 0) {			DPERROR("sio_sun_setpar: getinfo(pars)");			hdl->sio.eof = 1;			return 0;		}		enc = (hdl->sio.mode & SIO_REC) ?		    aui.record.encoding : aui.play.encoding;		switch (enc) {		case AUDIO_ENCODING_SLINEAR_LE:		case AUDIO_ENCODING_SLINEAR_BE:		case AUDIO_ENCODING_ULINEAR_LE:		case AUDIO_ENCODING_ULINEAR_BE:		case AUDIO_ENCODING_SLINEAR:		case AUDIO_ENCODING_ULINEAR:			break;		default:			DPRINTF("sio_sun_setpar: couldn't set linear encoding/n");			hdl->sio.eof = 1;			return 0;		}		if (hdl->sio.mode != (SIO_REC | SIO_PLAY))			break;		if (aui.play.sample_rate == aui.record.sample_rate &&		    aui.play.precision == aui.record.precision &&		    aui.play.encoding == aui.record.encoding)			break;		if (i < NRETRIES / 2) {			rate = aui.play.sample_rate;			prec = aui.play.precision;			enc = aui.play.encoding;		} else {			rate = aui.record.sample_rate;			prec = aui.record.precision;			enc = aui.record.encoding;		}	}	/*	 * If the rate that the hardware is using is different than	 * the requested rate, scale buffer sizes so they will be the	 * same time duration as what was requested.  This just gets	 * the rates to use for scaling, that actual scaling is done	 * later.	 */	rate = (hdl->sio.mode & SIO_REC) ? aui.record.sample_rate :	    aui.play.sample_rate;	req_rate = rate;	if (par->rate && par->rate != ~0U)		req_rate = par->rate;	/*	 * if block size and buffer size are not both set then	 * set the blocksize to half the buffer size	 */	bufsz = par->appbufsz;	round = par->round;	if (bufsz != ~0U) {//.........这里部分代码省略.........

//PORTING: This function contains a ton of OS specific stuff. Hack and//         slash at will.Error SoundCardPMO::Init(OutputInfo * info){   m_properlyInitialized = false;   if (!info)   {      info = myInfo;   }   else   {      // got info, so this is the beginning...      if ((audio_fd = open("/dev/audio", O_WRONLY, 0)) < 0)      {         if (errno == EBUSY)         {            ReportError("Audio device is busy. Please make sure that "                        "another program is not using the device.");            return (Error) pmoError_DeviceOpenFailed;         }         else         {            ReportError("Cannot open audio device. Please make sure that "                        "the audio device is properly configured.");            return (Error) pmoError_DeviceOpenFailed;         }      }      m_iDataSize = info->max_buffer_size;   }   int       fd = audio_fd;   struct audio_info ainfo;   if (ioctl(audio_fd, AUDIO_GETINFO, &ainfo) < 0)   {      ReportError("Cannot get the flags on the audio device.");      return (Error) pmoError_IOCTL_F_GETFL;   }   audio_fd = fd;   channels = info->number_of_channels;   for (unsigned int i = 0; i < info->number_of_channels; ++i)      bufferp[i] = buffer + i;   // configure the device:   int       play_precision = 16;//   int       play_stereo = channels - 1;   int       play_sample_rate = info->samples_per_second;   if (ioctl(audio_fd, I_FLUSH, FLUSHRW) == -1)   {      ReportError("Cannot reset the soundcard.");      return (Error) pmoError_IOCTL_SNDCTL_DSP_RESET;   }      AUDIO_INITINFO(&ainfo);   ainfo.play.precision = play_precision;   ainfo.play.channels = channels;   ainfo.play.sample_rate = play_sample_rate;   ainfo.play.encoding = AUDIO_ENCODING_LINEAR;   if (ioctl(audio_fd, AUDIO_SETINFO, &ainfo) == -1)   {      ReportError("Cannot set the soundcard's sampling speed.");      return (Error) pmoError_IOCTL_SNDCTL_DSP_SPEED;   }   myInfo->bits_per_sample = info->bits_per_sample;   myInfo->number_of_channels = info->number_of_channels;   myInfo->samples_per_second = info->samples_per_second;   myInfo->max_buffer_size = info->max_buffer_size;   m_properlyInitialized = true;   // PORTING: The GETOSPACE ioctl determines how much space the kernel's   // output buffer has. Your OS may not have this.   m_iTotalFragments = 2048; /* An arbitrary value of 2048. */   m_iOutputBufferSize = play_precision * m_iTotalFragments;   m_iBytesPerSample = info->number_of_channels * (info->bits_per_sample / 8);   return kError_NoErr;}

//.........这里部分代码省略.........	rplay_audio_table = dbri_table;		/* use the dbri table */    }    else    {	report(REPORT_ERROR, "`%s' unknown audio device detected/n", d.name);	return -1;    }    /* Verify the precision and format. */    switch (rplay_audio_precision)    {    case 8:	if (rplay_audio_format != RPLAY_FORMAT_ULAW	    && rplay_audio_format != RPLAY_FORMAT_LINEAR_8)	{	    report(REPORT_ERROR, "rplay_audio_init: can't use %d bits with format=%d/n",		   rplay_audio_precision, rplay_audio_format);	    return -1;	}	break;    case 16:	if (rplay_audio_format != RPLAY_FORMAT_LINEAR_16)	{	    report(REPORT_ERROR, "rplay_audio_init: can't use %d bits with format=%d/n",		   rplay_audio_precision, rplay_audio_format);	    return -1;	}	break;    default:	report(REPORT_ERROR, "rplay_audio_init: `%d' unsupported audio precision/n",	       rplay_audio_precision);	return -1;    }    AUDIO_INITINFO(&a);    switch (rplay_audio_format)    {    case RPLAY_FORMAT_ULAW:	a.play.encoding = AUDIO_ENCODING_ULAW;	break;    case RPLAY_FORMAT_LINEAR_8:    case RPLAY_FORMAT_LINEAR_16:	a.play.encoding = AUDIO_ENCODING_LINEAR;	break;    default:	report(REPORT_ERROR, "rplay_audio_init: unsupported audio format `%d'/n",	       rplay_audio_format);	return -1;    }    /* Audio port. */    if (rplay_audio_port == RPLAY_AUDIO_PORT_NONE)    {	a.play.port = ~0;	/* see AUDIO_INITINFO in /usr/include/sys/audioio.h. */    }    else    {	a.play.port = 0;	if (BIT(rplay_audio_port, RPLAY_AUDIO_PORT_LINEOUT))	{#ifdef AUDIO_LINE_OUT	    SET_BIT(a.play.port, AUDIO_LINE_OUT);#else	    CLR_BIT(rplay_audio_port, RPLAY_AUDIO_PORT_LINEOUT);#endif	}	if (BIT(rplay_audio_port, RPLAY_AUDIO_PORT_HEADPHONE))	{#ifdef AUDIO_HEADPHONE	    SET_BIT(a.play.port, AUDIO_HEADPHONE);#else	    CLR_BIT(rplay_audio_port, RPLAY_AUDIO_PORT_HEADPHONE);#endif	}	if (BIT(rplay_audio_port, RPLAY_AUDIO_PORT_SPEAKER))	{#ifdef AUDIO_SPEAKER	    SET_BIT(a.play.port, AUDIO_SPEAKER);#endif	    /* Assume speaker is okay. */	}    }    a.play.sample_rate = rplay_audio_sample_rate;    a.play.precision = rplay_audio_precision;    a.play.channels = rplay_audio_channels;    if (ioctl(rplay_audio_fd, AUDIO_SETINFO, &a) < 0)    {	report(REPORT_ERROR, "rplay_audio_init: AUDIO_SETINFO: %s/n", sys_err_str(errno));	return -1;    }    return 0;}

intDSP_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 
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