예제 #1
0
파일: sound.c 프로젝트: hhirsch/netrek
/*
 * Open and initialize an audio port.
 */
static ALport sfxInitAudioPort(int init) {
	ALport ap;
	ALconfig audioPortConfig;
	long pvbuf[6];

	if (init) {
		pvbuf[0] = AL_LEFT_SPEAKER_GAIN;
		pvbuf[2] = AL_RIGHT_SPEAKER_GAIN;
		pvbuf[4] = AL_OUTPUT_RATE;
		ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 6L);
		currLeftGain = origLeftGain = pvbuf[1];
		currRightGain = origRightGain = pvbuf[3];
		origOutputRate = pvbuf[5];
		sfxSetGainIndex((origLeftGain+origRightGain)/2);
	}

	/*
	 * Configure and open audio port.
	 */
	audioPortConfig = ALnewconfig();
	ALsetwidth(audioPortConfig, AL_SAMPLE_16);
	ALsetchannels(audioPortConfig, AL_STEREO);
	ALsetqueuesize(audioPortConfig, 16000);
	ap = ALopenport("spacetrek", "w", audioPortConfig);
	ALfreeconfig(audioPortConfig);

	return ap;
}
예제 #2
0
int sound_init()
{
  long pvbuf[2];
  pvbuf[0] = AL_OUTPUT_COUNT;
  pvbuf[2] = AL_MONITOR_CTL;
  if (ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 4) < 0) 
  {    
    fprintf(stderr,"sound driver : No audio hardware\n");
    return 0;
  }
  pvbuf[0] = AL_OUTPUT_RATE;
  pvbuf[1] = 11025;
  if (ALsetparams(AL_DEFAULT_DEVICE,pvbuf,4) < 0)
  {    
    fprintf(stderr,"sound driver : Could not set sample rate\n");
    return 0;
  }

  ALseterrorhandler(0);
  audioconfig = ALnewconfig();
  if (!audioconfig)
  {
    fprintf(stderr,"failed to create audio config\n");
    return 0;
  }
  else if (ALsetchannels(audioconfig,AL_MONO))
  { fprintf(stderr,"sound driver : could not set audio channels\n");
    return 0;
  }
  else if (ALsetqueuesize(audioconfig,BUF_SIZE))
  { 
    fprintf(stderr,"sound driver : could not set audio que size\n");
    ALfreeconfig(audioconfig);
    return 0;
  } else if (ALsetwidth (audioconfig, AL_SAMPLE_8))
  {
    fprintf(stderr,"sound driver :could not set 8 bit samples\n");
    ALfreeconfig(audioconfig);
    return 0;
  }

  audioport=ALopenport("Abuse sound driver","w",audioconfig);
  if (!audioport)
  {
    fprintf(stderr,"sound driver : could not open audio port\n");
    ALfreeconfig(audioconfig);
    return 0;
  }

  int i=0,j;
  uchar *vd=volume_table;
  for (;i<32;i++) 
  {
    for (j=0;j<256;j++,vd++)
      *vd=(j-128)*i/31+128;
  }
  return 1;
}
예제 #3
0
int audriv_get_play_volume(void)
/* 演奏音量を 0 〜 255 内で得ます.0 は無音,255 は最大音量.
 * 失敗すると -1 を返し,そうでない場合は 0 〜 255 内の音量を返します.
 */
{
#ifndef SGI_OLDAL
    ALfixed lrgain[2];
    ALpv pv;
    double gain, l, r, min, max;
    int volume;
    int resource;

    min = alFixedToDouble(out_ginfo.min.ll);
    max = alFixedToDouble(out_ginfo.max.ll);
    pv.param = AL_GAIN;
    pv.value.ptr = lrgain;
    pv.sizeIn = 2;
    if(out == NULL)
	resource = AL_DEFAULT_OUTPUT;
    else
	resource = alGetResource(out);

    if(alGetParams(resource, &pv, 1) < 0)
    {
	audriv_err(ALERROR);
	return -1;
    }
    l = alFixedToDouble(lrgain[0]);
    r = alFixedToDouble(lrgain[1]);
    if(l < min) l = min; else if(l > max) l = max;
    if(r < min) r = min; else if(r > max) r = max;
    gain = (l + r) / 2;
    volume = (gain - min) * 256 / (max - min);
    if(volume < 0)
	volume = 0;
    else if(volume > 255)
	volume = 255;
    return volume;
#else
    long gain[4];
    int volume;

    gain[0] = AL_LEFT_SPEAKER_GAIN;
    gain[2] = AL_RIGHT_SPEAKER_GAIN;
    if(ALgetparams(AL_DEFAULT_DEVICE, gain, 4) < 0)
    {
	audriv_err(ALERROR);
	return -1;
    }

    volume = (gain[1] + gain[3]) / 2;
    if(volume < 0)
	volume = 0;
    else if(volume > 255)
	volume = 255;
    return volume;
#endif /* SGI_OLDAL */
}
예제 #4
0
static void get_current_volumes(AudioDevice device)
{
	long params[4];
	params[0] = AL_LEFT_SPEAKER_GAIN;
	params[2] = AL_RIGHT_SPEAKER_GAIN;
	ALgetparams(device->device, params, 4);
	device->left_speaker_gain = params[1];
	device->right_speaker_gain = params[3];
}
예제 #5
0
int
sgi_audio_get_ogain(audio_desc_t ad)
{
	long	cmd[2];

        UNUSED(ad); assert(audio_fd > 0);

	cmd[0] = AL_LEFT_SPEAKER_GAIN;
	ALgetparams(AL_DEFAULT_DEVICE, cmd, 2L);
	return (SGI_DEVICE_TO_RAT(cmd[1]));
}
예제 #6
0
int
sgi_audio_get_igain(audio_desc_t ad)
{
	long	cmd[2];

        UNUSED(ad); assert(audio_fd > 0);

	cmd[0] = AL_LEFT_INPUT_ATTEN;
	ALgetparams(AL_DEFAULT_DEVICE, cmd, 2L);
	return (MAX_AMP - SGI_DEVICE_TO_RAT(cmd[1]));
}
예제 #7
0
static int sgi_init(const char *param, int *speed, int *fragsize, int *fragnr, int *channels)
{
    long chpars[] = { AL_OUTPUT_RATE, 0 };
    int st;

    /* No stereo capability. */
    *channels = 1;

    ALseterrorhandler(sgi_errorhandler);
    chpars[1] = *speed;
    st = ALsetparams(AL_DEFAULT_DEVICE, chpars, 2);
    if (st < 0) {
        return 1;
    }
    st = ALgetparams(AL_DEFAULT_DEVICE, chpars, 2);
    if (st < 0) {
        return 1;
    }
    *speed = chpars[1];

    sgi_audioconfig = ALnewconfig();
    if (!sgi_audioconfig) {
        return 1;
    }
    st = ALsetchannels(sgi_audioconfig, AL_MONO);
    if (st < 0) {
        goto fail;
    }
    st = ALsetwidth(sgi_audioconfig, AL_SAMPLE_16);
    if (st < 0) {
        goto fail;
    }
    st = ALsetqueuesize(sgi_audioconfig, *fragsize * *fragnr);
    if (st < 0) {
        goto fail;
    }
    sgi_audioport = ALopenport("outport", "w", sgi_audioconfig);
    if (!sgi_audioport) {
        goto fail;
    }
    return 0;
fail:
    ALfreeconfig(sgi_audioconfig);
    sgi_audioconfig = NULL;
    return 1;
}
예제 #8
0
파일: sound.c 프로젝트: hhirsch/netrek
/*
 * Check for external volume changes.
 */
static void sfxCheckVolume(void) {
	long pvbuf[6];

	/*
	 * Check to see if the volume was changed externally.
	 */
	pvbuf[0] = AL_LEFT_SPEAKER_GAIN;
	pvbuf[2] = AL_RIGHT_SPEAKER_GAIN;

	ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 4L);
	if (pvbuf[1] != currLeftGain || pvbuf[3] != currRightGain) {
		origLeftGain = currLeftGain = pvbuf[1];
		origRightGain = currRightGain = pvbuf[3];
		sfxSetGainIndex((pvbuf[1] + pvbuf[3]) / 2);
	}
	return;
}
예제 #9
0
static AudioContext
audio_initialize (unsigned char *data, int length, int volume)
{
  Lisp_Object audio_port_state[3];
  static AudioContextRec desc;
  AudioContext ac;

  desc.ac_right_speaker_gain
    = desc.ac_left_speaker_gain
      = volume * 256 / 100;
  desc.ac_device = AL_DEFAULT_DEVICE;

#if HAVE_SND_FILES
  if (LOOKING_AT_SND_HEADER_P (data))
    {
      if (parse_snd_header (data, length, & desc)==-1)
	report_file_error ("decoding .snd header", Qnil);
    }
  else
#endif
      {
	desc.ac_data = data;
	desc.ac_size = length;
	desc.ac_output_rate = DEFAULT_SAMPLING_RATE;
	desc.ac_nchan = DEFAULT_CHANNEL_COUNT;
	desc.ac_format = DEFAULT_FORMAT;
	desc.ac_write_chunk_function = write_mulaw_8_chunk;
      }

  /* Make sure that the audio port is reset to
     its initial characteristics after exit */
  ALgetparams (desc.ac_device, saved_device_state,
	       sizeof (saved_device_state) / sizeof (long));
  audio_port_state[0] = make_int (saved_device_state[1]);
  audio_port_state[1] = make_int (saved_device_state[3]);
  audio_port_state[2] = make_int (saved_device_state[5]);
  record_unwind_protect (restore_audio_port,
			 Fvector (3, &audio_port_state[0]));

  ac = initialize_audio_port (& desc);
  desc = * ac;
  return ac;
}
예제 #10
0
int esd_audio_open()
{
    ALconfig audioconfig;
    audioconfig = ALnewconfig();
  
	rate_params[1] = esd_audio_rate;
	rate_params[3] = esd_audio_rate;

    if (!audioconfig) {
	printf( "Couldn't initialize new audio config\n" );
	esd_audio_fd = -1;
	return esd_audio_fd;
    } else {
	long pvbuf[] = { AL_OUTPUT_COUNT, 0, 
			 AL_MONITOR_CTL, 0, 
			 AL_OUTPUT_RATE, 0 };
    
	if (ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 6) < 0)
	    if (oserror() == AL_BAD_DEVICE_ACCESS) {
		esd_audio_fd = -1;
		return esd_audio_fd;
	    }
    
	if (pvbuf[1] == 0 && pvbuf[3] == AL_MONITOR_OFF) {
	    ALsetparams(AL_DEFAULT_DEVICE, rate_params, 2);
	} else
	    if (pvbuf[5] != esd_audio_rate) {
		printf("audio device is already in use with wrong sample output rate\n");
		esd_audio_fd = -1;
		return esd_audio_fd;
	
	    }
    
	/* ALsetsampfmt(audioconfig, AL_SAMPFMT_TWOSCOMP); this is the default */
	/* ALsetwidth(audioconfig, AL_SAMPLE_16); this is the default */
    
	if ( (esd_audio_format & ESD_MASK_CHAN) == ESD_MONO)
	    ALsetchannels(audioconfig, AL_MONO);
	/* else ALsetchannels(audioconfig, AL_STEREO); this is the default */

	ALsetqueuesize(audioconfig, ESD_BUF_SIZE * 2);
    
	outaudioport = ALopenport("esd", "w", audioconfig);
	if (outaudioport == (ALport) 0) {
	    switch (oserror()) {
	    case AL_BAD_NO_PORTS:
		printf( "system is out of ports\n");
		esd_audio_fd = -1;
		return esd_audio_fd;
		break;
	
	    case AL_BAD_DEVICE_ACCESS:
		printf("couldn't access audio device\n");
		esd_audio_fd = -1;
		return esd_audio_fd;
		break;
	
	    case AL_BAD_OUT_OF_MEM:
		printf("out of memory\n");
		esd_audio_fd = -1;
		return esd_audio_fd;
		break;
	    }
	    /* don't know how we got here, but it must be bad */
	    esd_audio_fd = -1;
	    return esd_audio_fd;
	}
	ALsetfillpoint(outaudioport, ESD_BUF_SIZE);

	esd_audio_fd = ALgetfd(outaudioport);

	/*
	 * If we are recording, open a second port to read from
	 * and return that fd instead
	 */
	if ( (esd_audio_format & ESD_MASK_FUNC) == ESD_RECORD ) {
	    inaudioport = ALopenport("esd", "r", audioconfig);
	    if (inaudioport == (ALport) 0) {
		switch (oserror()) {
		case AL_BAD_NO_PORTS:
		    printf( "system is out of ports\n");
		    esd_audio_fd = -1;
		    return esd_audio_fd;
		    break;
	
		case AL_BAD_DEVICE_ACCESS:
		    printf("couldn't access audio device\n");
		    esd_audio_fd = -1;
		    return esd_audio_fd;
		    break;
	
		case AL_BAD_OUT_OF_MEM:
		    printf("out of memory\n");
		    esd_audio_fd = -1;
		    return esd_audio_fd;
		    break;
		default:
		    printf( "Unknown error opening port\n" );
		}
				/* don't know how we got here, but it must be bad */
		esd_audio_fd = -1;
		return esd_audio_fd;
	    }
	    ALsetfillpoint(inaudioport, ESD_BUF_SIZE);
	    ALsetparams(AL_DEFAULT_DEVICE, (rate_params + 2), 2);

	    esd_audio_fd = ALgetfd(inaudioport);
	}

    }
    return esd_audio_fd;
}
예제 #11
0
/*--------------------------------------------------------------------------------------*/
PaError PaHost_OpenStream(internalPortAudioStream *past)
{
	PaError                 result = paNoError;
	PaHostSoundControl      *pahsc;
	unsigned int            minNumBuffers;
	internalPortAudioDevice *padIN, *padOUT;        /* For looking up native AL-numbers. */
    ALconfig                sgiALconfig = NULL;     /* IRIX-datatype.   */
    long                    pvbuf[8];               /* To get/set hardware configs.      */
    long                    sr, ALqsize;
    DBUG(("PaHost_OpenStream() called.\n"));        /* Alloc FASTMEM and init host data. */
    if (!past)
        {
        ERR_RPT(("Streampointer NULL!\n"));
        result = paBadStreamPtr;        goto done;
        }
	pahsc = (PaHostSoundControl*)PaHost_AllocateFastMemory(sizeof(PaHostSoundControl));
	if (pahsc == NULL)
	    {
        ERR_RPT(("FAST Memory allocation failed.\n"));  /* Pass trough some ERR_RPT-exit-  */
        result = paInsufficientMemory;  goto done;      /* code (nothing will be freed).   */
	    }
	memset(pahsc, 0, sizeof(PaHostSoundControl));
/*  pahsc->pahsc_threadPID = -1;                       Should pahsc_threadPID be inited to */
	past->past_DeviceData = (void*)pahsc;           /* -1 instead of 0 ??                  */
    /*--------------------------------------------------- Allocate native buffers: --------*/
    pahsc->pahsc_SamplesPerInputBuffer = past->past_FramesPerUserBuffer * /* Needed by the */
                                         past->past_NumInputChannels;     /* audio-thread. */
	pahsc->pahsc_BytesPerInputBuffer   = pahsc->pahsc_SamplesPerInputBuffer * sizeof(short);
	if (past->past_NumInputChannels > 0)                       /* Assumes short = 16 bits! */
	    {
		pahsc->pahsc_NativeInputBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerInputBuffer);        
		if( pahsc->pahsc_NativeInputBuffer == NULL )
		    {
            ERR_RPT(("Fast memory allocation for input-buffer failed.\n"));
			result = paInsufficientMemory;  goto done;
		    }
	    }
    pahsc->pahsc_SamplesPerOutputBuffer = past->past_FramesPerUserBuffer * /* Needed by the */
                                          past->past_NumOutputChannels;    /* audio-thread. */
	pahsc->pahsc_BytesPerOutputBuffer   = pahsc->pahsc_SamplesPerOutputBuffer * sizeof(short);
	if (past->past_NumOutputChannels > 0)                       /* Assumes short = 16 bits! */
	    {
		pahsc->pahsc_NativeOutputBuffer = (short*)PaHost_AllocateFastMemory(pahsc->pahsc_BytesPerOutputBuffer);
		if (pahsc->pahsc_NativeOutputBuffer == NULL)
		    {
            ERR_RPT(("Fast memory allocation for output-buffer failed.\n"));
			result = paInsufficientMemory;  goto done;
		    }
	    }
    /*------------------------------------------ Manipulate hardware if necessary and allowed: --*/
    ALseterrorhandler(0);                           /* 0 = turn off the default error handler.   */
    pvbuf[0] = AL_INPUT_RATE;
    pvbuf[2] = AL_INPUT_COUNT;
    pvbuf[4] = AL_OUTPUT_RATE;              /* TO FIX: rates may be logically, not always in Hz! */
    pvbuf[6] = AL_OUTPUT_COUNT;
    sr = (long)(past->past_SampleRate + 0.5);   /* Common for input and output :-)               */
    if (past->past_NumInputChannels > 0)                        /* We need to lookup the corre-  */
        {                                                       /* sponding native AL-number(s). */
        padIN = Pa_GetInternalDevice(past->past_InputDeviceID);
        if (!padIN)
            {
            ERR_RPT(("Pa_GetInternalDevice() for input failed.\n"));
	        result = paHostError;  goto done;
            }
        if (ALgetparams(padIN->pad_ALdevice, &pvbuf[0], 4)) /* Although input and output will both be on */
            goto sgiError;                                  /* the same AL-device, the AL-library might  */
        if (pvbuf[1] != sr)                                 /* contain more than AL_DEFAULT_DEVICE in    */
            {  /* Rate different from current harware-rate?    the future. Therefore 2 seperate queries. */
            if (pvbuf[3] > 0)     /* Means, there's other clients using AL-input-ports */
                {
                ERR_RPT(("Sorry, not allowed to switch input-hardware to %ld Hz because \
another process is currently using input at %ld kHz.\n", sr, pvbuf[1]));
                result = paHostError;   goto done;
                }
            pvbuf[1] = sr;                  /* Then set input-rate. */
            if (ALsetparams(padIN->pad_ALdevice, &pvbuf[0], 2))
                goto sgiError;      /* WHETHER THIS SAMPLERATE WAS REALLY PRESENT IN OUR ARRAY OF RATES, */
            }                       /* IS NOT CHECKED, AT LEAST NOT BY ME, WITHIN THIS FILE! Does PA do? */
예제 #12
0
void *Audio_Irix::open (AudioConfig& cfg, const char *)
{
    // Copy input parameters. May later be replaced with driver defaults.
    _settings = cfg;
    
    _config = ALnewconfig();

    // Set sample format
    ALsetsampfmt(_config, AL_SAMPFMT_TWOSCOMP);

    // stereo or mono mode
    _settings.channels = cfg.channels >= 2 ? 2 : 1;
    if (_settings.channels == 2)
        ALsetchannels(_config, AL_STEREO);
    else
        ALsetchannels(_config, AL_MONO);

    // 16 or 8 bit sample
    _settings.precision = cfg.precision >= 16 ? 16 : 8;
    if (_settings.precision == 16)
        ALsetwidth(_config, AL_SAMPLE_16);
    else
        ALsetwidth(_config, AL_SAMPLE_8);

    // Frequency
    long chpars[] = {AL_OUTPUT_RATE, 0};
    if (cfg.frequency > 48000)
        chpars[1] = AL_RATE_48000;
    else if (cfg.frequency > 44100)
        chpars[1] = AL_RATE_44100;
    else if (cfg.frequency > 32000)
        chpars[1] = AL_RATE_32000;
    else if (cfg.frequency > 22050)
        chpars[1] = AL_RATE_22050;
    else if (cfg.frequency > 16000)
        chpars[1] = AL_RATE_16000;
    else
        chpars[1] = AL_RATE_11025;
    ALsetparams(AL_DEFAULT_DEVICE, chpars, 2);
    ALgetparams(AL_DEFAULT_DEVICE, chpars, 2);
    _settings.frequency = (uint_least32_t) chpars[1];

    // Allocate sound buffers and set audio queue
    ALsetqueuesize(_config, chpars[1]);

    // open audio device
    _audio = ALopenport("SIDPLAY2 sound", "w", _config);
    if (_audio == NULL)
    {
        perror("AUDIO:");
        _errorString = "ERROR: Could not open audio device.\n       See standard error output.";
        ALfreeconfig(_config);
        return 0;
    }

    // Setup internal Config
    _settings.encoding  = AUDIO_SIGNED_PCM;
    _settings.bufSize   = (uint_least32_t) chpars[1];

    // Update the users settings
    getConfig (cfg);

    // Allocate memory same size as buffer
#ifdef HAVE_EXCEPTIONS
    _sampleBuffer = new(std::nothrow) int_least8_t[chpars[1]];
#else
    _sampleBuffer = new int_least8_t[chpars[1]];
#endif

    _errorString = "OK";
    return (void *) _sampleBuffer;
}
예제 #13
0
static AudioContext
initialize_audio_port (AudioContext desc)
{
  /* we can't use the same port for mono and stereo */
  static AudioContextRec mono_port_state
    = { { 0, 0, 0, 0 },
	{ (ALport) 0, AFunknown, 1, 0 },
	{ (void *) 0, (unsigned long) 0 } };
#if HAVE_STEREO
  static AudioContextRec stereo_port_state
    = { { 0, 0, 0, 0 },
	{ (ALport) 0, AFunknown, 2, 0 },
	{ (void *) 0, (unsigned long) 0 } };
  static AudioContext return_ac;

  switch (desc->ac_nchan)
    {
    case 1:  return_ac = & mono_port_state; break;
    case 2:  return_ac = & stereo_port_state; break;
    default: return (AudioContext) 0;
    }
#else /* not HAVE_STEREO */
  static AudioContext return_ac = & mono_port_state;
#endif /* not HAVE_STEREO */

  return_ac->device = desc->device;
  return_ac->buffer = desc->buffer;
  return_ac->ac_format = desc->ac_format;
  return_ac->ac_queue_size = desc->ac_queue_size;

  if (return_ac->ac_port==(ALport) 0)
    {
      if ((open_audio_port (return_ac, desc))==-1)
	{
	  report_file_error ("Open audio port", Qnil);
	  return (AudioContext) 0;
	}
    }
  else
    {
      ALconfig config = ALgetconfig (return_ac->ac_port);
      int changed = 0;
      long params[2];

      params[0] = AL_OUTPUT_RATE;
      ALgetparams (return_ac->ac_device, params, 2);
      return_ac->ac_output_rate = params[1];

      if (return_ac->ac_output_rate != desc->ac_output_rate)
	{
	  return_ac->ac_output_rate = params[1] = desc->ac_output_rate;
	  ALsetparams (return_ac->ac_device, params, 2);
	}
      if ((changed = set_output_format (config, return_ac->ac_format))==-1)
	return (AudioContext) 0;
      return_ac->ac_format = desc->ac_format;
      if (changed)
	ALsetconfig (return_ac->ac_port, config);
    }
  return_ac->ac_write_chunk_function = desc->ac_write_chunk_function;
  get_current_volumes (& return_ac->device);
  if (return_ac->ac_left_speaker_gain != desc->ac_left_speaker_gain
      || return_ac->ac_right_speaker_gain != desc->ac_right_speaker_gain)
    adjust_audio_volume (& desc->device);
  return return_ac;
}