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alWriteBuffers(3dm)					   alWriteBuffers(3dm)


NAME    [Toc]    [Back]

     alWriteBuffers - write flexibly interleaved or non-interleaved audio data
     to	an audio port

SYNOPSIS    [Toc]    [Back]

     #include <dmedia/audio.h>

     int alWriteBuffers(const ALport port, void	**bufs,	int *strides, const int	framecount)

PARAMETER    [Toc]    [Back]

     port	  is the audio output port to which you	want to	write samples.
		  This is the returned value of	an alOpenPort(3dm) call.

     bufs	  is an	array of pointers to sample buffers, each element of
		  which	corresponds to a single	channel	of audio output.

     strides	  is an	array of integers, one corresponding to	each output
		  channel. Each	element	indicates the number of	interleaved
		  channels in the buffer directed at that channel.

     framecount	  is the number	of sample frames that you want to write	to the
		  audio	port.

DESCRIPTION    [Toc]    [Back]

     alWriteBuffers transfers data to an audio port from a set of buffers, or
     from different locations in a single buffer. alWriteBuffers allows	the
     application to specify how	the data is interleaved.

     bufs is an	array of pointers to sample buffers. Each element of bufs
     corresponds to one	output channel.	If the element is 0, that channel will
     be	zeroed on output. This allows an application to	only direct audio at
     the channels of interest.

     The sample	buffers	can be arbitrarily interleaved;	the strides parameter
     indicates the number of channels in each source sample buffer. For
     example, if strides[n] is 1, then bufs[n] is mono;	if strides[n] is 2,
     then bufs[n] is interleaved stereo, and alWriteFrames will	read every
     other sample from that buffer.  The elements of strides can have any
     value. If strides is 0, all the sample buffers are	considered mono, and
     alWriteBuffers does non-interleaved output.

     There must	be exactly as many elements in bufs and	strides	as the number
     of	channels specified for the port	with alSetChannels(3dm).

     alWriteBuffers blocks until framecount sample frames have been written to
     the port.	If you do not wish to block, make sure that framecount is less
     than the return value of alGetFillable(3dm).

     The expected format of each sample	depends	upon the configuration of the
     audio port. Each sample can be an 8-, 16-,	or 32-bit integer, or a
     single- or	double-precision floating-point	value or subcode data; see
     alSetSampFmt(3dm) and alSetWidth(3dm) for a description of	how these



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alWriteBuffers(3dm)					   alWriteBuffers(3dm)



     formats work. By default, the sample format is 16-bit integer (short).

     Note that since an	audio port contains an internal	queue, samples written
     to	the port will not immediately come out the associated audio device or
     devices. For precise synchronization of audio and other media, use
     alGetFrameTime(3dm) and alGetFrameNumber(3dm) to determine	when samples
     will actually be output.

     In	order to achieve the best possible performance,	alWriteBuffers does
     not attempt to verify that	port, bufs, or strides are valid. You should
     make certain these	values are valid before	passing	them as	arguments to
     alWriteBuffers.

EXAMPLES    [Toc]    [Back]

     The following code	fragment opens an 8-channel audio output port and
     writes 8 separate mono buffers to it. Then	it mutes channels 3 and	5 and
     writes the	same data.

	  ALport p;
	  short	buf[8][1000];
	  void *bufs[8];
	  int i,j;
	  ALconfig c;

	  c = alNewConfig();
	  if (!c) {
	      printf("config create failed:%s\n", alGetErrorString(oserror()));
	      exit(-1);
	  }

	  alSetChannels(c, 8);

	  /* open a port with our configuration	*/
	  p = alOpenPort("alWriteBuffers example","w",c);

	  if (!p) {
	      printf("port open	failed:%s\n", alGetErrorString(oserror()));
	      exit(-1);
	  }

	  /*
	   * Fill our buffer. This isn't real audio data here; it's DC on each
	   * channel. In a real	application, we'd want more meaningful data in
	   * each of the buffers.
	   */
	  for (i = 0; i	< 8; i++) {
	       /* fill channel i */
		  for (j = 0; j	< 1000;	j++) {
			  buf[i][j] = i	* 2000;
		  }
		  bufs[i] = buf[i];
	  }



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alWriteBuffers(3dm)					   alWriteBuffers(3dm)



	  alWriteBuffers(p, bufs, 0, 1000);	     /*	write 1000 8-channel frames */

	  /*
	   * Now mute channels 3 and 5
	   */
	  bufs[2] = 0;	 /* channel 3 */
	  bufs[4] = 0;	 /* channel 5 */

	  alWriteBuffers(p, bufs, 0, 1000);	     /*	write 1000 8-channel frames */
	  }


     The following example writes a single 16-channel buffer to	2 8-channel
     devices. This is somewhat simplified, since it does not actually
     synchronize the audio between the devices.	For examples of	how to do
     that, see the example "scrub" in /usr/share/src/dmedia/audio.

	  #include <audio.h>

	  #define NPORTS 2
	  #define NCHANS_PER_PORT 8
	  #define BUF_NCHANS (NPORTS*NCHANS_PER_PORT)

	  main()
	  {
	      ALport p[NPORTS];
	      float buf[1000 * BUF_NCHANS];
	      void *bufs[NPORTS][NCHANS_PER_PORT];
	      int strides[NCHANS_PER_PORT];
	      ALconfig c;
	      int i, j,	k = 0;

	      /*
	       * Set up	an array of device resource ID's. To do	multi-device
	       * I/O meaningfully,  we'd want these to be different resource ID's.
	       * This here will	open 2 ports to	the same resource ID, and the
	       * audio system will mix the 2 ports.
	       */
	      int device[NPORTS]={AL_DEFAULT_OUTPUT, AL_DEFAULT_OUTPUT};

	      for (i = 0; i < 16000; i++) {
	       buf[i] =	(i & 15) * 1000.0;
	      }

	      c	= alNewConfig();
	      if (!c) {
	       printf("config create failed:%s\n", alGetErrorString(oserror()));
	       exit(-1);
	      }

	      alSetChannels(c, NCHANS_PER_PORT);
	      alSetFloatMax(c, 32767);



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alWriteBuffers(3dm)					   alWriteBuffers(3dm)



	      alSetSampFmt(c, AL_SAMPFMT_FLOAT);

	      /*
	       * Let's assume that in this case	we know	our floating-point data
	       * to be in range. We turn limiting off for (potentially)	increased
	       * performance.
	       */
	      alSetLimiting(c, 0);

	      /*
	       * Set up	bufs & strides to make different NCHANS-channel	"windows"
	       * into our buffer.
	       */
	      k	= 0;
	      for (j=0;	j < NPORTS; j++) {
	       for (i=0; i < NCHANS_PER_PORT; i++) {
		   bufs[j][i] =	&buf[k++];
	       }
	      }
	      for (i=0;	i < NCHANS_PER_PORT; i++) {
	       strides[i] = BUF_NCHANS;	/* 16, by default */
	      }

	      /*
	       * Open up NPORTS	audio ports. Note that we aren't doing
	       * anything special to synchronize the ports here; they won't
	       * in general be sync'ed.
	       */
	      for (i = 0; i < NPORTS; i++) {
	       alSetDevice(c, device[i]);

	       p[i] = alOpenPort("alWriteBuffers example","w",c);

	       if (!p[i]) {
		   printf("port	open failed:%s\n", alGetErrorString(oserror()));
		   exit(-1);
	       }
	      }

	      /*
	       * Write 1000 frames of data to all the audio ports
	       */
	      while (1)
	      for (i =0; i < NPORTS; i++) {
	       alWriteBuffers(p[i], bufs[i], strides, 1000);
	      }
	  }








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alWriteBuffers(3dm)					   alWriteBuffers(3dm)



DIAGNOSTICS
     alWriteBuffers always returns 0.

NOTES    [Toc]    [Back]

     On	output,	the data from all ports	on the system writing to a particular
     output device will	be mixed together, except in the case of subcode data.
     Because subcode data is treated as	inherently logical information,	no
     amount mathematics	can be applied to perform operations such as mixing.

     This function was introduced via patch to IRIX 6.3	and 6.4, and is
     present by	default	in later OS releases.  You should ensure that the
     target system will	have the functionality before calling this function;
     otherwise,	your program will crash	when you attempt to make the function
     call. To determine	if the feature is present, check the value of
     AL_VERSION	on the system resource.	The parameter must be present and its
     value must	be at least 6.

		  pv.param = AL_VERSION;
		  alGetParams(AL_SYSTEM,&pv,1);
		  if (pv.sizeOut < 0 ||	pv.value.i < 6)	{
		      /* feature not present */
		  }


SEE ALSO    [Toc]    [Back]

      
      
     alOpenPort(3dm), alGetFillable(3dm), alGetFilled(3dm),
     alSetChannels(3dm), alSetWidth(3dm), alReadFrames(3dm),
     alZeroFrames(3dm),	alSetConfig(3dm), alSetQueueSize(3dm),
     alSetSampFmt(3dm),	alSetFloatMax(3dm)


									PPPPaaaaggggeeee 5555
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 Similar pages
Name OS Title
alReadBuffers IRIX read flexibly interleaved or non-interleaved audio data from an audio port
alWriteFrames IRIX write interleaved sample frames to an audio port
alReadFrames IRIX read interleaved sample frames from an audio port
AFgetchannels IRIX get the number of interleaved track / virtual channels from an AFfilehandle structure for an audio track
passthru IRIX pass audio sample data from an input port to an output port
ALwritesamps IRIX (obsolete) write samples to an audio port
alZeroFrames IRIX write zero-valued sample frames to an audio port
AFwriteframes IRIX write audio sample frames to a specified track in an audio file
alDiscardFrames IRIX discard audio from an audio port
AFreadmisc IRIX read from / write to / move logical read/write pointer for data in a miscellaneous chunk in an audio file
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