haudio.c

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      ATransStatus trst;
      APauseAudio(a->audio,a->tid,&trst,&st);
      /*
        AStopAudio(a->audio,a->tid,ASMThisTrans,&trst,&st);
      */
   }
#endif
#ifdef OSS_AUDIO
   if (ioctl(audio_fd, SNDCTL_DSP_RESET, 0) < 0)
      HError(6006, "StopAudi: error stopping OSS input audio port");
#endif
   a->isActive = ADS_STOPPED;
   a->curVol = 0;
}


#ifdef MMAPI_AUDIO
static int OutSamples(AudioOut a);
static AudioOut sAudioOut=NULL;

void CALLBACK callBackOut(HWAVE hwaveIn, UINT msg, 
                          DWORD magic, DWORD fullBufHdr, DWORD c)
{
   AudioOut a;
   mmApiBuf *p;

   a = sAudioOut;
   if (a==NULL || a->magic!=magic) return;
   switch(msg) {
   case MM_WOM_DONE:
      for (p=a->pHead;p!=NULL;p=p->next)
         if (p->index==a->current) break;
      if (p!=NULL) a->total-=p->n;
      if (p!=NULL && p->next!=NULL) a->current=p->next->index;
      else a->current=-1; /* Done or fail */
      break;
   default:
      break;
   }
}
#endif

/* InitAudo: initialise the given audio output device */
static void InitAudo(AudioOut a, HTime *sampPeriod)
{
#ifdef MMAPI_AUDIO
   {
      /* Initialise */
      if (sAudioOut!=NULL)
         HError(6006,"InitAudo: MMAPI audio output already in use");
      sAudioOut=a;
      a->total=0; a->current=-1; a->pHead=a->pTail=NULL; 
      a->magic=sMagic; sMagic+=12345;

      /* Allocate special structures */
      a->waveFmt = mmeAllocMem(sizeof(PCMWAVEFORMAT));
      a->wavePos = mmeAllocMem(sizeof(MMTIME));

      /* Set up required format */
      a->waveFmt->wf.wFormatTag = WAVE_FORMAT_PCM;
      a->waveFmt->wf.nChannels = 1;
      if(*sampPeriod == 0.0){
         *sampPeriod = 1.0E+07 / (float)16000;
         a->waveFmt->wf.nSamplesPerSec = 16000;
      }else
         a->waveFmt->wf.nSamplesPerSec = 1.0E+07 / *sampPeriod;
      a->waveFmt->wf.nBlockAlign = sizeof(short);
      a->waveFmt->wf.nAvgBytesPerSec = 
         a->waveFmt->wf.nBlockAlign*a->waveFmt->wf.nSamplesPerSec;
      a->waveFmt->wBitsPerSample = 16;

      /* Set up position query */
      a->wavePos->wType = TIME_SAMPLES;

      /* Open wave device */
      if ((a->mmError=waveOutOpen(&a->waveOut, WAVE_MAPPER, 
                                  (LPWAVEFORMAT)a->waveFmt, callBackOut, a->magic,
                                  CALLBACK_FUNCTION))!=MMSYSERR_NOERROR)
         HError(6006,"InitAudo: Cannot open MMAPI audio output [ERR=%d]",a->mmError);
#ifdef DEC_AUDIO
      if ((a->mmError=waveOutGetID(a->waveOut, 
                                   &a->waveOutDev))!=MMSYSERR_NOERROR)
         HError(6006,"InitAudo: Cannot get MMAPI audio output device number [ERR=%d]",a->mmError);
      /* Only needed with DEC API */
      if (mmeCheckForCallbacks())
         mmeProcessCallbacks();
#endif
      if (trace & T_AUD) {
         printf(" Initialised MMAPI audio output at %.2fkHz\n",
                a->waveFmt->wf.nSamplesPerSec*1E-3);
         fflush(stdout);
      }
   }
#endif
#ifdef RS6000_AUDIO                            
   {
      long sampRate;
      
      a->evout = somGetGlobalEnvironment();
      a->adevout = UMSBAUDDeviceNew();
      if ((a->rc = UMSAudioDevice_open(a->adevout,a->evout,AUDIO_DEV,"PLAY",
                                       UMSAudioDevice_BlockingIO))!=rOK)
         HError(6006,"InitAudo: Cannot open RS6000 audio output device");
      a->channels = 1;
      a->bits  = 16 ;
      if (*sampPeriod == 0.0){
         UMSAudioDevice_get_sample_rate(a->adevout,a->evout,&sampRate); 
         *sampPeriod= 1.0E+07 / sampRate;                  
      }
      else
         sampRate = 1.0E+07 / *sampPeriod;
      a->rc = UMSAudioDevice_set_sample_rate(a->adevout,a->evout,sampRate,&a->osamples);
      a->rc = UMSAudioDevice_set_bits_per_sample(a->adevout,a->evout,a->bits);
      a->rc = UMSAudioDevice_set_number_of_channels(a->adevout,a->evout,a->channels);
      a->rc = UMSAudioDevice_set_audio_format_type(a->adevout,a->evout,"PCM");
      a->rc = UMSAudioDevice_set_byte_order(a->adevout,a->evout,"MSB");
      a->rc=UMSAudioDevice_set_number_format(a->adevout,a->evout,"TWOS COMPLEMENT");
      a->rc = UMSAudioDevice_get_byte_order(a->adevout,a->evout,&a->obyte_order);
      /* you have to free the string after the query */
      if (a->obyte_order) free(a->obyte_order);
      a->rc = UMSAudioDevice_set_balance(a->adevout,a->evout,0);
      a->rc=UMSAudioDevice_set_time_format(a->adevout,a->evout,UMSAudioTypes_Samples);
      a->lgain = 100; /*maximum left input gain*/
      a->rgain = 100; /*maimum right input gain*/
      if (lineOut)
         strcpy(a->outConn, "LINE_OUT");
      if (phonesOut)
         strcpy(a->outConn, "LINE_OUT"); /* Don't have a phones out in RS6000 */
      if (speakerOut)
         strcpy(a->outConn, "INTERNAL_SPEAKER");
      a->rc = UMSAudioDevice_enable_output(a->adevout,a->evout,a->outConn,
                                           &a->lgain,&a->rgain);
      a->rc = UMSAudioDevice_initialize(a->adevout,a->evout);
   }
#endif   
#ifdef SGI_AUDIO
   {
      a->params[0] = AL_OUTPUT_RATE;
      a->params[2] = AL_LEFT_SPEAKER_GAIN;
      a->params[4] = AL_RIGHT_SPEAKER_GAIN;
      
      ALgetparams(AL_DEFAULT_DEVICE, a->params, 6);
      if (*sampPeriod == 0 ) 
         *sampPeriod =  1.0E+07 / (float) a->params[1] + 0.5;
      else {
         a->params[1] = (long int) (1.0E+07 / *sampPeriod + 0.5);
         ALsetparams(AL_DEFAULT_DEVICE,a->params,2);
      }
      a->config = ALnewconfig();
      ALsetchannels(a->config, AL_MONO);
      ALsetwidth(a->config, AL_SAMPLE_16);
      if ((a->playPort = ALopenport("HAudio_output", "w", a->config)) == NULL)
         HError(6006,"InitAudo: Cannot initialise SGI output audio port");
   }
#endif
#ifdef SUN16_AUDIO
   {
      int f,g;
      
      AUDIO_INITINFO(&a->audio_info);
      if ((a->audio_ctld = open(AUDIO_IO, O_WRONLY + O_NDELAY)) < 0) 
         HError(6006,"InitAudo: Cannot open Sun audio output%s",
                (errno==EBUSY?" [Already in use]":""));
      if (ioctl(a->audio_ctld, AUDIO_GETINFO, &a->audio_info) < 0)
         HError(6006,"InitAudo: Cannot interrogate Sun audio output");
      if (*sampPeriod == 0 )
         *sampPeriod = 1.0E+07 / (float) a->audio_info.play.sample_rate;
      AUDIO_INITINFO(&a->audio_info);
      f = (int) (1.0E+07 / *sampPeriod);
      a->audio_info.play.sample_rate = f;
      a->audio_info.play.channels = 1;
      a->audio_info.play.precision = 16;
      a->audio_info.play.balance = AUDIO_MID_BALANCE;
      a->audio_info.play.encoding = AUDIO_ENCODING_LINEAR;
      a->audio_info.play.port = 
         (lineOut?AUDIO_LINE_OUT:0)+
         (speakerOut?AUDIO_SPEAKER:0)+
         (phonesOut?AUDIO_HEADPHONE:0);
      if (ioctl(a->audio_ctld, AUDIO_SETINFO, &a->audio_info) < 0) {
         g = TrimSampFreq(f);
         if (f==g)
            HError(6006,"InitAudo: Cannot initialise Sun audio output [%d]",errno);
         a->audio_info.play.sample_rate = g;
         if (ioctl(a->audio_ctld, AUDIO_SETINFO, &a->audio_info) < 0)
            HError(6006,"InitAudo: Cannot initialise Sun audio output [%d]",errno);
         HError(-6006,"InitAudo: adjusting sampling frequency %d -> %d", f, g);
      }
      a->numSamples = a->numWrites = 0;
   }
#endif
#ifdef HPUX_AUDIO
   {
      long st;
      int i;
      ATransStatus atst;
      AudioAttributes *attr;
      AudioAttrMask mask;
      
      if (audio_dev==NULL)
         audio_dev=AOpenAudio(NULL,&st);
      else
         st=AENoError;
      a->audio=audio_dev;
      audio_cnt++;
      if (a->audio==NULL || st!=AENoError)
         HError(6006,"InitAudo: Cannot open HP audio output ERR=%d",st);
      ASetCloseDownMode(a->audio,AKeepTransactions,NULL);
      a->attr=*ABestAudioAttributes(a->audio);
      a->attr.type=ATSampled;
      a->attr.attr.sampled_attr.data_format = ADFLin16;
      a->attr.attr.sampled_attr.bits_per_sample = 16;
      if (*sampPeriod == 0.0)
         *sampPeriod= 1.0E+07 / a->attr.attr.sampled_attr.sampling_rate;
      else
         a->attr.attr.sampled_attr.sampling_rate = (long unsigned int) (1.0E+07 / *sampPeriod);
      a->attr.attr.sampled_attr.channels = 1;
      a->attr.attr.sampled_attr.interleave = 1;
      
      i=0;
      if (speakerOut) {
         a->gains[i].u.o.out_ch = AOCTMono;
         a->gains[i].gain = AUnityGain;
         a->gains[i].u.o.out_dst = AODTMonoIntSpeaker;
         i++;
      }
      if (phonesOut) {
         a->gains[i].u.o.out_ch = AOCTMono;
         a->gains[i].gain = AUnityGain;
         a->gains[i].u.o.out_dst = AODTMonoHeadphone;
         i++;
      }
      if (lineOut || i==0) {
         a->gains[i].u.o.out_ch = AOCTMono;
         a->gains[i].gain = AUnityGain;
         a->gains[i].u.o.out_dst = AODTMonoJack;
         i++;
      }
      a->parms.gain_matrix.type = AGMTOutput;
      a->parms.gain_matrix.num_entries = i;
      a->parms.gain_matrix.gain_entries = a->gains;
      a->parms.play_volume = AUnityGain;
      a->parms.priority = APriorityNormal;
      a->parms.event_mask = 0;
   }
#endif
#ifdef OSS_AUDIO
   {
      int f, g;
      short *p;
      
      if (audio_io==0) {
         if ((audio_fd = open(AUDIO_DEV, O_WRONLY, 0)) < 0)
            HError(6006, "InitAudo: Cannot open OSS audio device %s", AUDIO_DEV);
         if ((mixer_fd = open(MIXER_DEV, O_WRONLY, 0)) < 0)
            HError(6006, "InitAudo: unable to open OSS audio mixer %s", MIXER_DEV);
         for (f=0, p=zero_buf; f<BUF_SIZE; f++, p++) *p=0;
         f = 2;
         if (ioctl(audio_fd, SNDCTL_DSP_SUBDIVIDE, &f) < 0)
            HError(-6006, "InitAudo: error dividing buffer");
      }
      audio_io = audio_io | AUDIO_WR;
      if (ioctl (audio_fd, SNDCTL_DSP_SYNC, NULL) < 0) {
         HError(6006, "InitAudo: unable to sync audio device");
      }     
      f = g = (IsVAXOrder ()) ? AFMT_S16_LE : AFMT_S16_BE;
      if (ioctl(audio_fd, SNDCTL_DSP_SETFMT, &f) < 0)
         HError(6006, "InitAudo: error setting sample format");
      if (f != g)
         HError(6006, "InitAudo: unable to set 16 bit sample format");
      f = 0;
      if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &f) < 0)
         HError(6006, "InitAudo: error setting audio channel");
      f = g = (int) (1.0E+07 / (float) (*sampPeriod));
      if (ioctl(audio_fd, SNDCTL_DSP_SPEED, &f)==-1)
         HError(6006, "InitAudo: error setting sampling rate");
      if (ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &audio_info) < 0)
         HError(6006, "InitAudo: error getting fragment size");
      frag_size = audio_info.fragsize / sizeof(short);
      if (trace&T_AUD) {
         printf("InitAudo: info.fragments   %d\n", audio_info.fragments);
         printf("InitAudo: info.fragsize    %d\n", audio_info.fragsize);
         printf("InitAudo: info.bytes       %d\n", audio_info.bytes);
      }
   }
#endif
#ifdef NO_AUDIO
   *sampPeriod = 0.0;
#endif
   a->isActive = FALSE;
}

/* CloseAudo: close the given audio output device */
static void CloseAudo(AudioOut a)
{
#ifdef MMAPI_AUDIO 
   {
      mmApiBuf *p;

      while(a->current>=0) {
#ifdef DEC_AUDIO
         /* Only needed with DEC API */
         mmeWaitForCallbacks();
         mmeProcessCallbacks();
#endif
         if (OutSamples(a)==0) break;
      }  /* Block until finished playing */
      if((a->mmError=waveOutReset(a->waveOut))!=MMSYSERR_NOERROR)
         HError(6006,"CloseAudo: Cannot reset MMAPI output audio device [ERR=%d]",a->mmError);
      for(p=a->pHead;p!=NULL;p=p->next) {
#ifdef WIN32_AUDIO
         if((a->mmError=waveOutUnprepareHeader(a->waveOut, p->waveHdr, 
                                               sizeof(WAVEHDR))) != MMSYSERR_NOERROR)
            HError(6006,"CloseAudo: MMAPI Header unpreparation failed [ERR=%d]",a->mmError);
#endif
         if (mmeFreeMem(p->waveHdr)!=TRUE)
            HError(6006,"CloseAudo: MMAPI Header Free failed");
         if (mmeFreeMem(p->waveData)!=TRUE)
            HError(6006,"CloseAudo: MMAPI Data Free failed");
         /* Note mmApiBufs will be freed with AudioOut structure */
      }
      a->pHead=a->pTail=NULL;
      if((a->mmError=waveOutClose( a->waveOut )) != MMSYSERR_NOERROR) 
         HError(6006,"CloseAudo: Cannot close MMAPI output audio device [ERR=%d]",a->mmError);
      if (mmeFreeMem(a->waveFmt)!=TRUE)
         HError(6006,"CloseAudo: MMAPI wave foramt free failed");
      if (mmeFreeBuffer(a->wavePos)!=TRUE)
         HError(6006,"CloseAudo: MMAPI wave position free failed");
#ifdef DEC_AUDIO
      /* Only needed with DEC API */
      if (mmeCheckForCallbacks())
         mmeProcessCallbacks();
#endif
      if (trace & T_AUD) {
         printf(" Closing MMAPI audio output\n");
         fflush(stdout);
      }
      sAudioOut=NULL;
   }
#endif
#ifdef RS6000_AUDIO     
   {
      UMSAudioDevice_stop(a->adevout,a->evout);
      UMSAudioDevice_close(a->adevout,a->evout);
      _somFree(a->adevout);
   }
#endif   
#ifdef SGI_AUDIO
   {
      ALfreeconfig(a->config);
      ALcloseport(a->playPort);
   }
#endif
#ifdef SUN16_AUDIO
   {
      close(a->audio_ctld);
      a->audio_ctld=-1;
   }
#endif
#ifdef HPUX_AUDIO
   {
      long st;
      a->audio=NULL;
      audio_cnt--;
      if (audio_cnt<=0) {
         ASetCloseDownMode(audio_dev,AKeepTransactions,NULL);
         ACloseAudio(audio_dev,&st);
         audio_cnt=0;audio_dev=NULL;
      }
   }
#endif
#ifdef OSS_AUDIO
   {
      if (audio_io&AUDIO_WR) {
         audio_io^=AUDIO_WR;
         if (audio_io==0) {
            if (ioctl (audio_fd, SNDCTL_DSP_SYNC, NULL) < 0) {
               HError(6006, "CloseAudo: unable to sync audio device");
            }     
            close(audio_fd); 
            close(mixer_fd);
         }
      }
   }
#endif
   a->isActive = FALSE;
}

/* OutSamples: return num samples left to play in output device */
static int OutSamples(AudioOut a)
{
   if (!a->isActive) return 0;
#ifdef MMAPI_AUDIO
   {
#ifdef DEC_AUDIO
      /* Only needed with DEC API */
      if (mmeCheckForCallbacks())
         mmeProcessCallbacks();
#endif
      if (a->current<0) return(0);
      if((a->mmError=waveOutGetPosition( a->waveOut, a->wavePos,
                                         sizeof(MMTIME))) != MMSYSERR_NOERROR) 
         HError(6006,"OutSamples: Cannot get current play back position");
      return(a->total - a->wavePos->u.sample);
   }
#endif
#ifdef RS6000_AUDIO
   {
      long samps=0;
      
      if ((a->rc=UMSAudioDevice_write_buff_used(a->adevout,a->evout,&samps))!=rOK){
         if (a->rc==rUR)
            samps=0;