/* melder_audio.c
 *
 * Copyright (C) 1992-2005 Paul Boersma
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or (at
 * your option) any later version.
 *
 * This program is distributed in the hope that it will be useful, but
 * WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
 * See the GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
 */

/*
 * pb 2003/08/22 used getenv ("AUDIODEV") on Sun (thanks to Michel Scheffers)
 * pb 2003/10/22 fake mono for Linux drivers that do not support mono
 * pb 2003/12/06 use sys/soundcard.h instead of linux/soundcard.h for FreeBSD compatibility
 * pb 2004/05/07 removed motif_mac_setNullEventWaitingTime (we nowadays use 1 clock tick everywhere anyway)
 * pb 2004/08/10 fake mono for Linux drivers etc, also if not asynchronous
 * pb 2005/02/13 added O_NDELAY when opening /dev/dsp on Linux (suggestion by Rafael Laboissiere)
 * pb 2005/03/31 undid previous change (four complaints that sound stopped playing)
 * pb 2005/05/19 redid previous change (with fctrl fix suggested by Rafael Laboissiere)
 * pb 2005/10/13 edition for OpenBSD
 */

#include "melder.h"
#include "Gui.h"
#include "Resources.h"
#include "NUM.h"
#include <time.h>
#define my  me ->

#define Melder_HPUX_USE_AUDIO_SERVER

#if defined (sgi)
	/* For local audio: */
	#include <audio.h>
	#include <unistd.h>
	/* For the SPEXLAB audio server: */
	#include <ctype.h>
	#include <netdb.h>
	#include <sys/types.h>
	#include <sys/socket.h>
	#include <sys/ioctl.h>
	#include <sys/signal.h>
	#include <arpa/inet.h>
	#include <errno.h>
#elif defined (macintosh)
	#include "macport_on.h"
	#include <Sound.h>
	#ifndef __MACH__
		#include <SoundInput.h>
	#endif
	#ifdef __MACH__
		#include <sys/time.h>
	#else
		#include <Timer.h>
	#endif
	#include <Gestalt.h>
	#include "macport_off.h"
	#if TARGET_API_MAC_CARBON
		#define OSEventAvail  EventAvail
		#define USE_COREAUDIO  1
		#if USE_COREAUDIO
			#include <AudioUnit.h>
		#endif
	#endif
	#include <math.h>
#elif defined (_WIN32)
	#include <windows.h>
	#include <math.h>
#elif defined (sun)
	#include <fcntl.h>
	#include <stropts.h>
	#include <unistd.h>
	#include <sys/audioio.h>
	#include <sys/time.h>
	#include <signal.h>
#elif defined (HPUX)
	#include <fcntl.h>
	#include <ctype.h>
	#include <unistd.h>
	#include <sys/audio.h>
	#include <sys/ioctl.h>
	#include <sys/stat.h>
	#if defined (Melder_HPUX_USE_AUDIO_SERVER)
		#include "Alib.h"
	#endif
#elif defined (linux)
	#include <sys/types.h>
	#include <sys/stat.h>
	#include <sys/ioctl.h>
	#include <fcntl.h>
	#include <unistd.h>
	#if defined (__OpenBSD__) || defined (__NetBSD__)
		#include <soundcard.h>
	#else
		#include <sys/soundcard.h>
	#endif
	#include <errno.h>
	#include <sys/time.h>
	#include <signal.h>
#endif

#if defined (sgi)
	static int _melder_sgi_useAudioServer = -1;
	static void _melder_sgi_checkAudioServer (void) {
		if (_melder_sgi_useAudioServer != -1) return;   /* Already initialized. */
		_melder_sgi_useAudioServer = getenv ("PRAAT_USES_ADDA_SERVER") ? 1 : 0;
	}		
#endif

#ifdef UNIX
	static Bool predicateProcedure (Display *display, XEvent *event, char *arg) {
		char buffer [20];
		KeySym key;
		XComposeStatus compose;
		(void) display;
		(void) arg;
		if (event -> type != KeyPress) return False;
		XLookupString ((XKeyEvent *) event, buffer, 20, & key, & compose);
		return key == 0xFF1B || key == 0xFF09;   /* Escape key or Tab key. */
	}
#endif

static struct {
	int maximumAsynchronicity, useInternalSpeaker;
	double outputGain, zeroPadding;
} prefs = { Melder_ASYNCHRONOUS, TRUE, 1.0, 0.0 };

void Melder_setMaximumAsynchronicity (int maximumAsynchronicity) {
	Melder_stopPlaying (Melder_IMPLICIT);
	prefs. maximumAsynchronicity = maximumAsynchronicity;
}
int Melder_getMaximumAsynchronicity (void) { return prefs. maximumAsynchronicity; }

void Melder_setUseInternalSpeaker (int useInternalSpeaker) {
	Melder_stopPlaying (Melder_IMPLICIT);
	prefs. useInternalSpeaker = useInternalSpeaker;
}
int Melder_getUseInternalSpeaker (void) { return prefs. useInternalSpeaker; }

void Melder_setOutputGain (double gain) {
	Melder_stopPlaying (Melder_IMPLICIT);
	prefs. outputGain = gain;
}
double Melder_getOutputGain (void) { return prefs. outputGain; }

void Melder_setZeroPadding (double zeroPadding) {
	Melder_stopPlaying (Melder_IMPLICIT);
	prefs. zeroPadding = zeroPadding;
}
double Melder_getZeroPadding (void) { return prefs. zeroPadding; }

void Melder_audio_prefs (void) {
	Resources_addInt ("Audio.maximumAsynchronicity", & prefs. maximumAsynchronicity);
	Resources_addInt ("Audio.useInternalSpeaker", & prefs. useInternalSpeaker);
	Resources_addDouble ("Audio.outputGain", & prefs. outputGain);
	Resources_addDouble ("Audio.zeroPadding", & prefs. zeroPadding);
}

long Melder_getBestSampleRate (long fsamp) {
	#if defined (macintosh)
		return fsamp > 64000 ? 44100 : fsamp;
	#elif defined (sgi)
		_melder_sgi_checkAudioServer ();
		if (_melder_sgi_useAudioServer) return fsamp;
		return fsamp == 8000 || fsamp == 9800 || fsamp == 11025 || fsamp == 16000 || fsamp == 22050 ||
			fsamp == 32000 || fsamp == 44100 || fsamp == 48000 ? fsamp : 22050;
	#elif defined (_WIN32)
		return fsamp == 8000 || fsamp == 11025 || fsamp == 16000 || fsamp == 22050 ||
			fsamp == 32000 || fsamp == 44100 || fsamp == 48000 ? fsamp : 22050;
	#elif defined (HPUX) && defined (USE_AUDIO_SERVER)
		return fsamp == 5513 ? 5512 : fsamp;   /* 5512.5 should be given as 5512 */
	#elif defined (sun) || defined (HPUX)
		return fsamp == 8000 || fsamp == 11025 || fsamp == 16000 || fsamp == 22050 ||
			fsamp == 32000 || fsamp == 44100 || fsamp == 48000 ? fsamp : 22050;
	#elif defined (linux)
		return fsamp == 8000 || fsamp == 11025 || fsamp == 16000 || fsamp == 22050 ||
			fsamp == 32000 || fsamp == 44100 || fsamp == 48000 ? fsamp : 22050;
	#else
		return 22050;
	#endif
}

int Melder_isPlaying;

static struct MelderPlay {
	const short *buffer;
	long sampleRate, numberOfSamples, samplesLeft, samplesSent, samplesPlayed;
	int asynchronicity, numberOfChannels, explicit, fakeMono;
	int (*callback) (void *closure, long samplesPlayed);
	void *closure;
	XtWorkProcId workProcId;
	#if defined (sgi)
		/* For local audio: */
		ALconfig config;
		ALport port;
		/* For the SPEXLAB audio server: */
		int inited, status, portNumber, sokket;
		char hostName [64];
		long defchan, channels;
	#elif defined (_WIN32)
		HWAVEOUT hWaveOut;
		WAVEHDR waveHeader;
		MMRESULT status;
		clock_t startingTime;
	#elif defined (macintosh)
		SndChannelPtr soundChannel;
		/*static void channelCallback (SndChannelPtr channel, SndCommand cmd) {} */
		#ifndef __MACH__
			TMTask tmTask;
			clock_t startingTime;
		#endif
	#elif defined (HPUX) && defined (Melder_HPUX_USE_AUDIO_SERVER)
		long status;
		Audio *audio;
		SBucket *bucket;
		ATransID xid;
	#elif defined (HPUX)
		int audio_fd;
	#elif defined (sun)
		int audio_fd;
		audio_info_t audioInfo;
	#elif defined (linux)
		int audio_fd, val, err;
	#endif
} thePlay;

long Melder_getSamplesPlayed (void) {
	return thePlay. samplesPlayed;
}

int Melder_stopWasExplicit (void) {
	return thePlay. explicit;
}

#if defined (linux) || defined (sun) || defined (HPUX) || defined (__MACH__)
	static struct itimerval theTide;
	static void Melder_startClock (void) {
		theTide.it_value.tv_sec = 20000;
		theTide.it_value.tv_usec = 0;
		signal (SIGALRM, SIG_IGN);
		setitimer (ITIMER_REAL, & theTide, NULL);
	}
	static double Melder_clock (void) {
		getitimer (ITIMER_REAL, & theTide);
		return (20000 - theTide.it_value.tv_sec) - theTide.it_value.tv_usec / 1000000.0;
	}
	static void Melder_stopClock (void) {
		theTide.it_value.tv_sec = 0;
		theTide.it_value.tv_usec = 0;
		setitimer (ITIMER_REAL, & theTide, NULL);
		signal (SIGALRM, SIG_DFL);
	}
#endif

/*
 * The flush () procedure will always have to be invoked after normal play, i.e. in the following cases:
 * 1. After synchronous play (asynchronicity = 0, 1, or 2).
 * 2. After interruption of asynchronicity 2 by the ESCAPE key.
 * 3. After asynchronous play, by the workProc.
 * 4. After interruption of asynchronicity 3 by Melder_stopPlaying ().
 */
static Boolean flush (void) {
	struct MelderPlay *me = & thePlay;
	#if defined (sgi)
		_melder_sgi_checkAudioServer ();
		if (_melder_sgi_useAudioServer) {
			close (my sokket);
			signal (SIGPIPE, SIG_DFL);
		} else {
			ALcloseport (my port), my port = 0;
		}
	#elif defined (_WIN32)
		/*
		 * FIX: Do not reset the sound card if played to the end:
		 * the last 20 milliseconds may be truncated!
		 * This used to happen on Barbertje's Dell PC, not with SoundBlaster.
		 */
		if (my samplesPlayed != my numberOfSamples || Melder_debug == 2)
			waveOutReset (my hWaveOut);
		my status = waveOutUnprepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR));
		if (/* Melder_debug == 3 && */ my status == WAVERR_STILLPLAYING) {
			waveOutReset (my hWaveOut);
			waveOutUnprepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR));
		}
		waveOutClose (my hWaveOut), my hWaveOut = 0;
	#elif defined (macintosh)
		if (my asynchronicity == Melder_ASYNCHRONOUS) {   /* Other asynchronicities are handled within Melder_play16 (). */
			SndDisposeChannel (my soundChannel, 1), my soundChannel = NULL;
			#ifdef __MACH__
				my samplesPlayed = Melder_clock () * my sampleRate;
			#else
				RmvTime ((QElemPtr) & my tmTask);
				/*if (my tmTask. tmCount > 0)
					my samplesPlayed = ((80000000 - my tmTask. tmCount) / 1000.0) * my sampleRate;*/
			#endif
		}
		#ifdef __MACH__
			Melder_stopClock ();
		#endif
	#elif defined (HPUX) && defined (Melder_HPUX_USE_AUDIO_SERVER)
		if (my audio) {
			AStopAudio (my audio, my xid, ASMThisTrans, NULL, & my status);
			/*
			 * Unhack the max_length and audio_data attributes first.
			 */
			my bucket -> max_length = 0;
			my bucket -> audio_data = NULL;
			ADestroySBucket (my audio, my bucket, & my status), my bucket = NULL;
			ACloseAudio (my audio, & my status), my audio = NULL;
		}
		Melder_stopClock ();
	#elif defined (sun)
		/*
		 * On Sun, draining is performed by calling close ().
		 * If this has happened, we should not close again.
		 * Fortunately, we signalled the closing by setting the file descriptor pointer to zero.
		 */
		if (my audio_fd) {
			ioctl (my audio_fd, I_FLUSH, FLUSHW);   /* Discard what's in the buffers! */
			close (my audio_fd), my audio_fd = 0;
		}
		Melder_stopClock ();
	#elif defined (linux)
		/*
		 * As on Sun.
		 */
		if (my audio_fd) {
			ioctl (my audio_fd, SNDCTL_DSP_RESET, (my val = 0, & my val));
			close (my audio_fd), my audio_fd = 0;
		}
		Melder_stopClock ();
	#endif
	if (my fakeMono) {
		NUMsvector_free ((short *) my buffer, 0);
	}
	Melder_isPlaying = 0;
	if (my samplesPlayed >= my numberOfSamples)
		my samplesPlayed = my numberOfSamples;
	if (my samplesPlayed <= 0)
		my samplesPlayed = 1;
	/*
	 * Call the callback for the last time, which is recognizable by the value of Melder_isPlaying.
	 * In this way, the caller of Melder_play16 can be notified.
	 * The caller can examine the actual number of samples played by testing Melder_getSamplesPlayed ().
	 */
	if (my callback)
		my callback (my closure, my samplesPlayed);
	my callback = 0;
	my closure = 0;
	return True;   /* Remove workProc if called from workProc. */
}

int Melder_stopPlaying (int explicit) {
	struct MelderPlay *me = & thePlay;
	my explicit = explicit;
	if (! Melder_isPlaying || my asynchronicity < Melder_ASYNCHRONOUS) return 0;
	(void) flush ();
	XtRemoveWorkProc (thePlay. workProcId);
	return 1;
}

static Boolean workProc (XtPointer closure) {
	struct MelderPlay *me = & thePlay;
	#if defined (sgi)
		_melder_sgi_checkAudioServer ();
		if (_melder_sgi_useAudioServer) {
		} else {
			if (my samplesLeft > 0) {
				long dsamples = my samplesLeft > 1000 ? 1000 : my samplesLeft;
				ALwritesamps (my port, (void *) & my buffer [my samplesSent * my numberOfChannels], dsamples * my numberOfChannels);
				my samplesLeft -= dsamples;
				my samplesSent += dsamples;
				my samplesPlayed = my samplesSent - ALgetfilled (my port) / my numberOfChannels;
				if (my callback && ! my callback (my closure, my samplesPlayed))
					return flush ();
			} else {
				long n = ALgetfilled (my port);
				if (n == 0) {
					my samplesPlayed = my numberOfSamples;
					return flush ();
				} else {
					my samplesPlayed = my numberOfSamples - n / my numberOfChannels;
					if (my callback && ! my callback (my closure, my samplesPlayed))
						return flush ();
				}
				sginap (1);
			}
		}
	#elif defined (_WIN32)
  		if (my waveHeader. dwFlags & WHDR_DONE) {
			my samplesPlayed = my numberOfSamples;
			return flush ();
  		} else {
  			static long previousTime = 0;
  			unsigned long currentTime = clock ();
  			if (Melder_debug == 1) {
				my samplesPlayed = ((clock () - my startingTime) / (double) CLOCKS_PER_SEC) * my sampleRate;
  			} else {
	  			MMTIME mmtime;
	  			mmtime. wType = TIME_BYTES;
	  			waveOutGetPosition (my hWaveOut, & mmtime, sizeof (MMTIME));
				my samplesPlayed = mmtime. u.cb / (2 * my numberOfChannels);
			}
			if (/* Melder_debug != 4 || */ currentTime - previousTime > CLOCKS_PER_SEC / 100) {
				previousTime = currentTime;
				if (my callback && ! my callback (my closure, my samplesPlayed))
					return flush ();
			}
  		}
	#elif defined (macintosh)
		SCStatus status;
		SndChannelStatus (my soundChannel, sizeof (SCStatus), & status);
		if (status. scChannelBusy) {
			#ifdef __MACH__
				my samplesPlayed = Melder_clock () * my sampleRate;
			#else
				my samplesPlayed = ((clock () - my startingTime) / (double) CLOCKS_PER_SEC) * my sampleRate;
			#endif
			if (my callback && ! my callback (my closure, my samplesPlayed))
				return flush ();
		} else {
			my samplesPlayed = my numberOfSamples;
			return flush ();
		}
	#elif defined (HPUX) && defined (Melder_HPUX_USE_AUDIO_SERVER)
		AEvent audioEvent;
		int eventAvailable = ACheckEvent (my audio, & audioEvent, & my status);
		if (eventAvailable && audioEvent. any_event. type == AETTransCompleted && audioEvent. any_event. xid == my xid) {
			my samplesPlayed = my numberOfSamples;
			return flush ();
		} else {
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed)) {
				/*APauseAudio (my audio, my xid, NULL, & my status);*/
				/*ASetCloseDownMode (my audio, ADestroyAll, & my status);*/
				return flush ();
			}
		}
	#elif defined (sun)
		if (my samplesLeft > 0) {
			int dsamples = my samplesLeft > 2000 ? 2000 : my samplesLeft;
			write (my audio_fd, (char *) & my buffer [my samplesSent * my numberOfChannels], 2 * dsamples * my numberOfChannels);
			my samplesLeft -= dsamples;
			my samplesSent += dsamples;
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				return flush ();
			if (my samplesLeft == 0) {
				/*
				 * Write a zero-length buffer (end-of-file record) to the output stream,
				 * so that we can use the play.eof attribute.
				 */
				my audioInfo. play. eof = 0;
				write (my audio_fd, NULL, 0);
			}
		} else if (ioctl (my audio_fd, AUDIO_GETINFO, & my audioInfo), my audioInfo. play. eof) {
			my samplesPlayed = my numberOfSamples;
			return flush ();
  		} else {
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				return flush ();
		}
	#elif defined (linux)
		if (my samplesLeft > 0) {
			int dsamples = my samplesLeft > 500 ? 500 : my samplesLeft;
			write (my audio_fd, (char *) & my buffer [my samplesSent * my numberOfChannels], 2 * dsamples * my numberOfChannels);
			my samplesLeft -= dsamples;
			my samplesSent += dsamples;
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				return flush ();
		} else /*if (my samplesPlayed >= my numberOfSamples)*/ {
			close (my audio_fd), my audio_fd = 0;
			my samplesPlayed = my numberOfSamples;
			return flush ();
  		/*} else {
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				return flush ();*/
		}
	#endif
	(void) closure;
	return False;
}

#ifndef macintosh
static void cancel (void) {
	struct MelderPlay *me = & thePlay;
	#if defined (sgi)
		if (_melder_sgi_useAudioServer) {
			if (my sokket) close (my sokket), my sokket = 0;
			signal (SIGPIPE, SIG_DFL);
		}
	#elif defined (HPUX) && defined (Melder_HPUX_USE_AUDIO_SERVER)
		if (my bucket) {
			my bucket -> max_length = 0;
			my bucket -> audio_data = NULL;
			ADestroySBucket (my audio, my bucket, & my status), my bucket = NULL;
		}
		if (my audio) ACloseAudio (my audio, & my status), my audio = NULL;
	#elif defined (HPUX)
		if (my audio_fd) close (my audio_fd), my audio_fd = 0;
	#elif defined (sun)
		if (my audio_fd) close (my audio_fd), my audio_fd = 0;
	#elif defined (linux)
		if (my audio_fd) close (my audio_fd), my audio_fd = 0;
	#endif
	Melder_isPlaying = 0;
}
#endif

#ifdef macintosh
# define FloatToUnsigned(f)  \
	 ((unsigned long)(((long)(f - 2147483648.0)) + 2147483647L + 1))
static void double2real10 (double x, unsigned char *bytes) {
	int sign, exponent;
	double fMantissa, fsMantissa;
	unsigned long highMantissa, lowMantissa;
	if (x < 0.0) { sign = 0x8000; x *= -1; }
	else sign = 0;
	if (x == 0.0) { exponent = 0; highMantissa = 0; lowMantissa = 0; }
	else {
		fMantissa = frexp (x, & exponent);
		if ((exponent > 16384) || ! (fMantissa < 1))   /* Infinity or Not-a-Number. */
			{ exponent = sign | 0x7FFF; highMantissa = 0; lowMantissa = 0; }   /* Infinity. */
		else {   /* Finite */
			exponent += 16382;   /* Add bias. */
			if (exponent < 0) {   /* Denormalized. */
				fMantissa = ldexp (fMantissa, exponent);
				exponent = 0;
			}
			exponent |= sign;
			fMantissa = ldexp (fMantissa, 32);          
			fsMantissa = floor (fMantissa); 
			highMantissa = FloatToUnsigned (fsMantissa);
			fMantissa = ldexp (fMantissa - fsMantissa, 32); 
			fsMantissa = floor (fMantissa); 
			lowMantissa = FloatToUnsigned (fsMantissa);
		}
	}
	bytes [0] = exponent >> 8;
	bytes [1] = exponent;
	bytes [2] = highMantissa >> 24;
	bytes [3] = highMantissa >> 16;
	bytes [4] = highMantissa >> 8;
	bytes [5] = highMantissa;
	bytes [6] = lowMantissa >> 24;
	bytes [7] = lowMantissa >> 16;
	bytes [8] = lowMantissa >> 8;
	bytes [9] = lowMantissa;
}
#endif

int Melder_play16 (const short *buffer, long sampleRate, long numberOfSamples, int numberOfChannels,
	int (*playCallback) (void *playClosure, long numberOfSamplesPlayed), void *playClosure)
{
	struct MelderPlay *me = & thePlay;
	#ifdef _WIN32
		int wasPlaying = Melder_isPlaying;
	#endif
	if (Melder_isPlaying) Melder_stopPlaying (Melder_IMPLICIT);   /* Otherwise, keep "explicit" tag. */
	my buffer = buffer;
	my sampleRate = sampleRate;
	my numberOfSamples = numberOfSamples;
	my numberOfChannels = numberOfChannels;
	my callback = playCallback;
	my closure = playClosure;
	my asynchronicity = Melder_batch ? Melder_SYNCHRONOUS : Melder_backgrounding ? Melder_INTERRUPTABLE : Melder_ASYNCHRONOUS;
	if (my asynchronicity > prefs. maximumAsynchronicity)
		my asynchronicity = prefs. maximumAsynchronicity;
	my explicit = Melder_IMPLICIT;
	my fakeMono = FALSE;

	my samplesLeft = numberOfSamples;
	my samplesSent = 0;
	my samplesPlayed = 0;
	Melder_isPlaying = 1;
#if defined (sgi)
{
	_melder_sgi_checkAudioServer ();
	if (_melder_sgi_useAudioServer) {
		/*
		 * This is code to access the SPEXLAB audio server, which is still used in Nijmegen.
		 */
		struct { long mark, pid, request, sampleRate, numberOfBits, numberOfSamples, channels; } header;
		char message [64];
		int val;
		struct sockaddr_in remoteHost;

		if (! my inited) {
			char *daChannel = getenv ("DA_CHANNEL"), *addaHost = getenv ("ADDA_HOST"), *atSign;
			if (daChannel) {
				unsigned int channel = atoi (daChannel);
				if (channel <= 47) my channels = (channel << 16) + channel / 16;
			}
			if (! addaHost) return cancel (), Melder_error ("Environment variable ADDA_HOST not set:\n"
					"Cannot determine host name and port number for audio output.");
			if ((atSign = strchr (addaHost, '@')) != NULL) {   /* ADDA_HOST is something like "17@fonsg6": port 1517. */
				*atSign = '\0';
				my portNumber = atoi (addaHost) + 1500;
				strcpy (my hostName, atSign + 1);
			} else {   /* ADDA_HOST is something like "fonsg6": port 1500. */
				strcpy (my hostName, addaHost);
				my portNumber = 1500;
			}
			if (getenv ("ADDA_CHANNEL")) my defchan = atoi (getenv ("ADDA_CHANNEL"));
			my inited = 1;
		}
		if ((my channels & 0xFFFF0000) == 0) my channels |= my defchan << 16;

		/*
		 * Ignore any "broken pipe" signals.
		 */
		signal (SIGPIPE, SIG_IGN);

		if (isdigit (my hostName [0])) {
			if ((remoteHost. sin_addr. s_addr = inet_addr (my hostName)) == 0xFFFFFFFF)
				return cancel (), Melder_error ("Invalid audio host name \"%s\".", my hostName);
			remoteHost. sin_family = AF_INET;
		} else {
			struct hostent *host = gethostbyname (my hostName);
			if (host == NULL)
				return cancel (), Melder_error ("Unknown audio host \"%s\".", my hostName);
			remoteHost. sin_family = host -> h_addrtype;
			memcpy (& remoteHost. sin_addr, host -> h_addr, host -> h_length);   /* BUG: sin_addr is a struct or union */
		}
		remoteHost. sin_port = htons (my portNumber);

		if ((my sokket = socket (AF_INET, SOCK_STREAM, IPPROTO_TCP)) == -1)
			return cancel (), Melder_error ("Cannot create socket for audio.");
		if (ioctl (my sokket, FIONBIO, (val = 1, & val)) == -1 )
			return cancel (), Melder_error ("Cannot set non-blocking on for audio socket.");
		for (;;) {
			int status = connect (my sokket, & remoteHost, sizeof (struct sockaddr_in));
			if (status == 0 || errno == EISCONN)   /* Success. */
				break;
			else if (errno != EINPROGRESS && errno != EALREADY)
				return cancel (), Melder_error ("Cannot connect socket to audio host.");
		}
		if (ioctl (my sokket, FIONBIO, (val = 0, & val)) == -1)
			return cancel (), Melder_error ("Cannot set non-blocking off for audio socket.");
		/* Now we are connected to port 'ntohs (remoteHost.sin_port)'
		 * on host 'inet_ntoa (remoteHost.sin_addr)'.
		 */
		header. mark = 1;
		header. pid = getpid ();
		header. request = 2;   /* Audio output (1 is input). */
		header. sampleRate = my sampleRate;
		header. numberOfBits = 16;
		header. numberOfSamples = my numberOfSamples;
		header. channels = my channels;

		if (write (my sokket, (char *) & header, sizeof header) < 0)
			return cancel (), Melder_error ("Audio server: error writing header.");

		/*
		 * Read the accept or refuse message from the server.
		 */
		/* Set socket to non-blocking I/O so we can interrupt if it takes too long. */
		if (ioctl (my sokket, FIONBIO, (val = 1, & val)) == -1)
			return cancel (), Melder_error ("Audio server cannot set blocking off.");
		/* Wait for read message. */
		for (;;) {
			int nread = read (my sokket, message, sizeof (message));
			if (nread > 0)
				break;   /* Success. */
			else if (errno != EWOULDBLOCK)
				return cancel (), Melder_error ("Audio server error #%d reading message.", errno);
		}
		/* Set socket back to blocking mode. */
		if (ioctl (my sokket, FIONBIO, (val = 0, & val)) == -1)
			return cancel (), Melder_error ("Audio server cannot set blocking on.");
		if (message [0] == 'A')
			;   /* Audio server accepts message. */
		else if (message [0] == 'R')
			return cancel (), Melder_error ("Audio server refuses message.");
		else
			return cancel (), Melder_error ("Audio server: unknown message \"%s\".", message);

		my samplesLeft = my numberOfSamples;
		my samplesSent = 0;
		while (my samplesLeft > 0) {
			long dsamples = my samplesLeft > 10240 ? 10240 : my samplesLeft, dbytes;
			dbytes = write (my sokket, (char *) & my buffer [my samplesSent], dsamples * 2);   /* BUG: mono */
			if (dbytes < 0)
				return cancel (), Melder_error ("Audio server: error writing data.");
			my samplesLeft -= dbytes / 2;   /* BUG: what if odd ? */
			my samplesSent += dbytes / 2;
		}
		my samplesPlayed = my numberOfSamples - my samplesLeft;
	} else {
		/*
		 * Normal local SGI audio output.
		 */
		int interrupted = 0;
		long pvbuf [2], n;
		if (! my config) my config = ALnewconfig ();
		pvbuf [0] = AL_OUTPUT_RATE;
		pvbuf [1] = my sampleRate;
		ALsetparams (AL_DEFAULT_DEVICE, pvbuf, 2);
		ALsetchannels (my config, my numberOfChannels == 1 ? AL_MONO : AL_STEREO);
		ALsetwidth (my config, AL_SAMPLE_16);
		if (! (my port = ALopenport ("Praat play", "w", my config))) {
			Melder_isPlaying = 0;
			return Melder_error ("Cannot open audio port.");
		}
		if (my asynchronicity == Melder_SYNCHRONOUS) {
			ALwritesamps (my port, (void *) my buffer, my numberOfSamples * my numberOfChannels);
			while (ALgetfilled (my port) > 0) { }   /* Drain. */
			my samplesPlayed = my numberOfSamples;
		} else if (my asynchronicity <= Melder_INTERRUPTABLE) {
			while (my samplesLeft > 0 && ! interrupted) {
				XEvent event;
				long dsamples = my samplesLeft > 1000 ? 1000 : my samplesLeft;
				ALwritesamps (my port, (void *) & my buffer [my samplesSent * my numberOfChannels], dsamples * my numberOfChannels);
				my samplesLeft -= dsamples;
				my samplesSent += dsamples;
				my samplesPlayed = my samplesSent - ALgetfilled (my port) / my numberOfChannels;
				if (my callback && ! my callback (my closure, my samplesPlayed))
					interrupted = 1;
				if (my asynchronicity == Melder_INTERRUPTABLE &&
				    XCheckIfEvent (XtDisplay (Melder_topShell), & event, predicateProcedure, 0))
					my explicit = Melder_EXPLICIT, interrupted = 1;
			}
			while ((n = ALgetfilled (my port)) > 0 && ! interrupted) {
				XEvent event;
				sginap (1);
				my samplesPlayed = my numberOfSamples - n / my numberOfChannels;
				if (my callback && ! my callback (my closure, my samplesPlayed))
					interrupted = 1;
				if (my asynchronicity == Melder_INTERRUPTABLE &&
				    XCheckIfEvent (XtDisplay (Melder_topShell), & event, predicateProcedure, 0))
					my explicit = Melder_EXPLICIT, interrupted = 1;
			}
		} else /* my asynchronicity == Melder_ASYNCHRONOUS */ {
			my workProcId = XtAppAddWorkProc (Melder_appContext, workProc, (XtPointer) me);
			return 1;
		}
	}
	flush ();
	return 1;
}
#elif defined (_WIN32)
{
	WAVEFORMATEX waveFormat;
	MMRESULT err;
	waveFormat. wFormatTag = WAVE_FORMAT_PCM;
	waveFormat. nChannels = my numberOfChannels;
	waveFormat. nSamplesPerSec = my sampleRate;
	waveFormat. wBitsPerSample = 16;
	waveFormat. nBlockAlign = my numberOfChannels * waveFormat. wBitsPerSample / 8;
	waveFormat. nAvgBytesPerSec = waveFormat. nBlockAlign * waveFormat. nSamplesPerSec;
	waveFormat. cbSize = 0;
	err = waveOutOpen (& my hWaveOut, WAVE_MAPPER, & waveFormat, 0, 0, CALLBACK_NULL | WAVE_ALLOWSYNC);
	if (err != MMSYSERR_NOERROR) {
		Melder_isPlaying = 0;
		return Melder_error (
			err == MMSYSERR_ALLOCATED ?
				"(Melder_play16:) Previous sound is still playing? Should not occur!\n"
				"Report bug to the author: %ld; wasPlaying: %d" :
			err == MMSYSERR_BADDEVICEID ?
				"(Melder_play16:) Cannot play a sound. Perhaps another program is playing a sound at the same time?" :
			err == MMSYSERR_NODRIVER ?
				"(Melder_play16:) This machine probably has no sound card." :
			err == MMSYSERR_NOMEM ?
				"(Melder_play16:) Not enough free memory to play any sound at all." :
			err == WAVERR_BADFORMAT ?
				"(Melder_play16:) Bad sound format? Should not occur! Report bug to the author!" :
			"(Melder_play16:) Unknown error %ld while trying to play a sound? Report bug to the author!", err, wasPlaying);
	}

	my waveHeader. dwFlags = 0;
	my waveHeader. lpData = (char *) my buffer;
	my waveHeader. dwBufferLength = my numberOfSamples * 2 * my numberOfChannels;
	my waveHeader. dwLoops = 1;
	my waveHeader. lpNext = NULL;
	my waveHeader. reserved = 0;
	err = waveOutPrepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR));
//waveOutReset (my hWaveOut);
	if (err != MMSYSERR_NOERROR) {
		waveOutClose (my hWaveOut), my hWaveOut = 0;
		Melder_isPlaying = 0;
		return Melder_error (
			err == MMSYSERR_INVALHANDLE ?
				"(Melder_play16:) No device? Should not occur!" :
			err == MMSYSERR_NODRIVER ?
				"(Melder_play16:) No driver? Should not occur!" :
			err == MMSYSERR_NOMEM ?
				"(Melder_play16:) Not enough free memory to play this sound.\n"
				"Remove some objects, play a shorter sound, or buy more memory." :
			"(Melder_play16:) Unknown error %ld while preparing header? Should not occur!", err);
	}

	my startingTime = clock ();
	err = waveOutWrite (my hWaveOut, & my waveHeader, sizeof (WAVEHDR));
	if (err != MMSYSERR_NOERROR) {
		waveOutReset (my hWaveOut);
		waveOutUnprepareHeader (my hWaveOut, & my waveHeader, sizeof (WAVEHDR));
		waveOutClose (my hWaveOut), my hWaveOut = 0;
		Melder_isPlaying = 0;
		return Melder_error ("(Melder_play16:) Error %d while writing.", err);   /* BUG: should flush */
	}

	if (my asynchronicity == Melder_SYNCHRONOUS) {
		while (! (my waveHeader. dwFlags & WHDR_DONE)) { (void) clock (); }
		my samplesPlayed = my numberOfSamples;
  	} else if (my asynchronicity <= Melder_INTERRUPTABLE) {
  		while (! (my waveHeader. dwFlags & WHDR_DONE)) {
			MSG event;
			my samplesPlayed = ((clock () - my startingTime) / (double) CLOCKS_PER_SEC) * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				break;
			if (my asynchronicity == Melder_INTERRUPTABLE &&
			    PeekMessage (& event, 0, 0, 0, PM_REMOVE) && event. message == WM_KEYDOWN) {
				if (LOWORD (event. wParam) == VK_ESCAPE) { my explicit = Melder_EXPLICIT; break; }
			}
		}
	} else {
		my workProcId = XtAppAddWorkProc (0, workProc, 0);
		return 1;
	}
	flush ();
	return 1;
}
#elif defined (macintosh)
{
	if (my asynchronicity == Melder_SYNCHRONOUS) {
		static char listResource [500];
		SndListPtr data = (SndListPtr) & listResource [0];
		ExtSoundHeaderPtr soundHeader;
		data -> format = 1;
		data -> numModifiers = 1;
		data -> modifierPart [0]. modNumber = sampledSynth;
		data -> modifierPart [0]. modInit = initStereo;
		data -> numCommands = 1;
		data -> commandPart [0]. cmd = bufferCmd + dataOffsetFlag;
		data -> commandPart [0]. param1 = 0;
		data -> commandPart [0]. param2 = 20;
		soundHeader = (void *) & data -> dataPart;
		soundHeader -> samplePtr = (char *) my buffer;
		soundHeader -> numChannels = my numberOfChannels;
		soundHeader -> sampleRate = (UnsignedFixed) (unsigned long) floor (65536.0 * my sampleRate);
		soundHeader -> loopStart = 0L;
		soundHeader -> loopEnd = 0L;
		soundHeader -> encode = extSH;
		soundHeader -> baseFrequency = 60;
		soundHeader -> numFrames = my numberOfSamples;
		soundHeader -> sampleSize = 16;
		(void) SndPlay (NULL, (void *) (Handle) & data, false);
		my samplesPlayed = my numberOfSamples;
	} else {
	/*	long soundFeatures;
		int can16bit = Gestalt (gestaltSoundAttr, & soundFeatures) == 0 &&
			(soundFeatures & (1 << gestalt16BitAudioSupport)) != 0;
		*/
		static char hdr [300];
		SndCommand cmd;
		ExtSoundHeaderPtr soundHeader = (ExtSoundHeaderPtr) & hdr;
		soundHeader -> samplePtr = (char *) my buffer;
		soundHeader -> numChannels = my numberOfChannels;
		soundHeader -> sampleRate = (UnsignedFixed) (unsigned long) floor (65536.0 * my sampleRate);
		double2real10 (my sampleRate, (unsigned char *) & soundHeader -> AIFFSampleRate);
		soundHeader -> loopStart = 0L;
		soundHeader -> loopEnd = 0L;
		soundHeader -> encode = extSH;
		soundHeader -> baseFrequency = 60;
		soundHeader -> numFrames = my numberOfSamples;
		soundHeader -> sampleSize = 16;
		SndNewChannel (& my soundChannel, sampledSynth, initStereo, NULL /*NewSndCallBackProc (channelCallback)*/);
		cmd. cmd = bufferCmd;
		cmd. param1 = 0;
		cmd. param2 = (long) soundHeader;
		#ifdef __MACH__
			Melder_startClock ();
		#else
			my startingTime = clock ();
			my tmTask. qLink = 0;
			my tmTask. qType = 0;
			my tmTask. tmAddr = NULL;
			my tmTask. tmWakeUp = 0;
			my tmTask. tmReserved = 0;
			InsXTime ((QElemPtr) & my tmTask);
			PrimeTime ((QElemPtr) & my tmTask, 80000000);
		#endif
		SndDoCommand (my soundChannel, & cmd, 0);
		if (my asynchronicity <= Melder_INTERRUPTABLE) {
			SCStatus status;
			while (SndChannelStatus (my soundChannel, sizeof (SCStatus), & status), status. scChannelBusy) {
				int interrupted = 0;
				EventRecord event;
				#ifdef __MACH__
					my samplesPlayed = Melder_clock () * my sampleRate;
				#else
					my samplesPlayed = ((clock () - my startingTime) / (double) CLOCKS_PER_SEC) * my sampleRate;
				#endif
				if (my callback && ! my callback (my closure, my samplesPlayed))
					interrupted = 1;
				/*
				 * Safe operation: only listen to key-down events.
				 * Do this on the lowest level that will work.
				 */
				if (my asynchronicity == Melder_INTERRUPTABLE && ! interrupted && OSEventAvail (keyDownMask, & event)) {
					/*
					 * Remove the event, even if it was a different key.
					 * Otherwise, the key will block the future availability of the Escape key.
					 */
					FlushEvents (keyDownMask, 0);
					/*
					 * Catch Escape and Command-period.
					 */
					if ((event. message & charCodeMask) == 27 || (event. modifiers & cmdKey) && (event. message & charCodeMask) == '.')
						my explicit = Melder_EXPLICIT, interrupted = 1;
				}
				if (interrupted) {
					#ifdef __MACH__
						my samplesPlayed = Melder_clock () * my sampleRate;
					#else
						RmvTime ((QElemPtr) & my tmTask);
						if (my tmTask. tmCount > 0)
							my samplesPlayed = ((80000000 - my tmTask. tmCount) / 1000.0) * my sampleRate;
						else
							my samplesPlayed = my numberOfSamples;
					#endif
					/*FlushEvents (everyEvent, 0);*/
					SndDisposeChannel (my soundChannel, 1), my soundChannel = NULL;
					flush ();
					return 1;
				}
			}
			SndDisposeChannel (my soundChannel, 0), my soundChannel = NULL;
			#ifndef __MACH__
				RmvTime ((QElemPtr) & my tmTask);
			#endif
			my samplesPlayed = my numberOfSamples;
		} else /* my asynchronicity == Melder_ASYNCHRONOUS */ {
			my workProcId = XtAppAddWorkProc (0, workProc, 0);
			return 1;
		}
	}
	flush ();
	return 1;
}
#elif defined (HPUX) && defined (Melder_HPUX_USE_AUDIO_SERVER)
{
	AudioAttributes attributes;
	SBPlayParams params;
	AGainEntry gainEntry;
	AEvent audioEvent;
	attributes. type = ATSampled;
	attributes. attr. sampled_attr. data_format = ADFLin16;
	attributes. attr. sampled_attr. bits_per_sample = 16;
	attributes. attr. sampled_attr. sampling_rate = my sampleRate;
	attributes. attr. sampled_attr. channels = my numberOfChannels;
	attributes. attr. sampled_attr. duration. type = ATTSamples;
	attributes. attr. sampled_attr. duration. u. samples = my numberOfSamples;   /* BUG ? times 2 */
	if (! my audio)
		my audio = AOpenAudio (NULL, & my status);
	if (my status) return cancel (), Melder_error ("Cannot open audio. Check audio server (\"man aserver\").");
	my bucket = ACreateSBucket (my audio, ASDataFormatMask | ASBitsPerSampleMask |
		ASSamplingRateMask | ASChannelsMask | ASDurationMask, & attributes, & my status);
	if (my status) return cancel (), Melder_error ("Cannot create sound bucket.");

	/*
	 * According to the Audio man pages, the bucket created by ACreateSBucket is for recording;
	 * a bucket for playing would be returned by ALoadAFile only.
	 * Apparently, they suppose that everybody only plays sounds from disk.
	 * We are more interested in playing a sound from memory, so we will use the empty bucket
	 * just created. We still call ACreateSBucket instead of using an in-line struct,
	 * because HP may add fields in the future.
	 * However, we are going to hack the max_length and audio_data attributes of the bucket.
	 * This is not especially compatible with future versions of the audio library,
	 * but if they only add attributes at the end, our code will still work without recompiling.
	 */
	my bucket -> max_length = my numberOfSamples * my numberOfChannels * 2;
	my bucket -> audio_data = (char *) my buffer;

	params. gain_matrix. type = AGMTOutput;
	params. gain_matrix. num_entries = 1;   /* BUG */
	params. gain_matrix. gain_entries = & gainEntry;
	gainEntry. u. o. out_dst = prefs. useInternalSpeaker ? AODTMonoIntSpeaker : AODTMonoJack;
	gainEntry. u. o. out_ch = AOCTMono;   /* BUG */
	gainEntry. gain = AUnityGain;
	params. play_volume = AUnityGain;
	params. pause_first = FALSE;
	params. start_offset. type = ATTSamples;
	params. start_offset. u. samples = 0;
	params. duration. type = ATTSamples;
	params. duration. u. samples = my numberOfSamples;   /* BUG ? */
	params. loop_count = 0;
	params. priority = APriorityNormal;
	params. previous_transaction = 0;
	params. event_mask = ATransCompletedMask;
	my xid = APlaySBucket (my audio, my bucket, & params, & my status);
	if (my status) return cancel (), Melder_error ("Cannot play sound bucket.");

	/*
	 * One of the following is superfluous (loop plus ASetCloseDownMode).
	 */
	Melder_startClock ();
	if (my asynchronicity == Melder_SYNCHRONOUS) {
		for (;;) {
			ANextEvent (my audio, & audioEvent, & my status);
			if (audioEvent. any_event. type == AETTransCompleted && audioEvent. any_event. xid == my xid)
				break;
		}
		my samplesPlayed = my numberOfSamples;
	} else if (my asynchronicity <= Melder_INTERRUPTABLE) {
		for (;;) {
			XEvent event;
			int eventAvailable = ACheckEvent (my audio, & audioEvent, & my status);
			if (eventAvailable && audioEvent. any_event. type == AETTransCompleted && audioEvent. any_event. xid == my xid) {
				my samplesPlayed = my numberOfSamples;
				break;
			}
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				break;
			if (my asynchronicity == Melder_INTERRUPTABLE && XCheckIfEvent (XtDisplay (Melder_topShell), & event, predicateProcedure, 0))
				{ my explicit = Melder_EXPLICIT; break; }
		}
	} else {
		my workProcId = XtAppAddWorkProc (Melder_appContext, workProc, 0);
		return 1;
	}
	#if 0
	AStopAudio (my audio, my xid, ASMThisTrans, NULL, & my status);
	ASetCloseDownMode (my audio, AKeepTransactions, & my status);
	my bucket -> max_length = 0;
	my bucket -> audio_data = NULL;
	ADestroySBucket (my audio, my bucket, & my status);
	ACloseAudio (my audio, & my status), my audio = NULL;
	#endif

	flush ();
	return 1;
}
#elif defined (HPUX)
{
	struct audio_describe audioInfo;
	struct audio_gains audioGains;
	struct audio_beep_type oldBeep;
	int bytesLeft = 2 * numberOfChannels * numberOfSamples, bytesWritten = 0;
	int err, channels, wantedOutput, currentOutput, dataFormat, oldSampleRate;

	my audio_fd = open ("/dev/audio", O_WRONLY);
	if (my audio_fd == -1) return cancel (), Melder_error ("Cannot open audio output device.");
	ioctl (my audio_fd, AUDIO_RESET, RESET_TX_BUF | RESET_TX_UNF);   /* Flush buffers? */

	ioctl (my audio_fd, AUDIO_GET_CHANNELS, & channels);
	if (channels != 1 && ioctl (my audio_fd, AUDIO_SET_CHANNELS, 1) == -1)
		return cancel (), Melder_error ("Cannot set mono.");

	wantedOutput = ( prefs. useInternalSpeaker ? AUDIO_OUT_SPEAKER : AUDIO_OUT_HEADPHONE ) | AUDIO_OUT_LINE;
	ioctl (my audio_fd, AUDIO_GET_OUTPUT, & currentOutput);
	if (wantedOutput != currentOutput && ioctl (my audio_fd, AUDIO_SET_OUTPUT, wantedOutput) == -1)
		return cancel (), Melder_error ("Cannot set output device.");

	ioctl (my audio_fd, AUDIO_GET_DATA_FORMAT, & dataFormat);
	if (dataFormat != AUDIO_FORMAT_LINEAR16BIT && ioctl (my audio_fd, AUDIO_SET_DATA_FORMAT, AUDIO_FORMAT_LINEAR16BIT) == -1)
		return cancel (), Melder_error ("Cannot set to 16-bit linear.");

	/*{
		struct audio_beep_type beep;
		if (ioctl (audio_fd, AUDIO_GET_BEEP, & oldBeep) == -1)
			return cancel (), Melder_error ("Cannot get old beep.");
		beep. type = 1;
		beep. datalen = 1;
		beep. data = "";
		beep. returnlen = 0;
		beep. repeat_count = 1;
		beep. sample_rate = 8000;
		beep. nchannels = 1;
		if (ioctl (audio_fd, AUDIO_SET_BEEP, & beep) == -1)
			return cancel (), Melder_error ("Cannot remove beep.");
	}*/
	/*ioctl (audio_fd, AUDIO_DESCRIBE, & audioInfo);
	audioGains. transmit_gain = 9 + 20 * log10 (
		prefs. outputGain > 1.0 ? 1.0 : prefs. outputGain < 1e-30 ? 1e-30 : prefs. outputGain);
	if (audioGains. transmit_gain < -84) audioGains. transmit_gain = -84;
	audioGains. monitor_gain = audioInfo. min_monitor_gain;
	if (ioctl (audio_fd, AUDIO_SET_GAINS, & audioGains) == -1)
		return cancel (), Melder_error ("Cannot set gains.");
	}*/

	ioctl (audio_fd, AUDIO_GET_SAMPLE_RATE, & oldSampleRate);
	if (sampleRate != oldSampleRate && ioctl (audio_fd, AUDIO_SET_SAMPLE_RATE, sampleRate) == -1)
		return cancel (), Melder_error ("Cannot set the sampling frequency to %ld Hz.", sampleRate);

	while (bytesLeft) {
		int dbytes = bytesLeft > 262144 ? 262144 : bytesLeft;
		if (write (audio_fd, (char *) & buffer [0] + bytesWritten, dbytes) == -1)
			return cancel (), Melder_error ("Cannot write audio output.");
		bytesLeft -= dbytes;
		bytesWritten += dbytes;
	}
	close (audio_fd);

	/*audio_fd = open ("/dev/audio", O_WRONLY);
	if (ioctl (audio_fd, AUDIO_SET_BEEP, & oldBeep) == -1)
		return cancel (), Melder_error ("Cannot reset old beep.");
	close (audio_fd);*/

	return 1;
}
#elif defined (sun)
{
	if ((my audio_fd = open (getenv ("AUDIODEV") ? getenv ("AUDIODEV") : "/dev/audio", O_WRONLY)) == -1)
		return cancel (), Melder_error ("Cannot open audio output device.");
	AUDIO_INITINFO (& my audioInfo);
	my audioInfo. play. sample_rate = my sampleRate;
	my audioInfo. play. channels = my numberOfChannels;
	my audioInfo. play. precision = 16;   /* Bits per sample. */
	my audioInfo. play. encoding = AUDIO_ENCODING_LINEAR;
	my audioInfo. play. port = ( prefs. useInternalSpeaker ? AUDIO_SPEAKER : 0 ) | AUDIO_HEADPHONE | AUDIO_LINE_OUT;
	if (ioctl (my audio_fd, AUDIO_SETINFO, & my audioInfo) == -1)
		return cancel (), Melder_error ("Cannot initialize audio output.");

	Melder_startClock ();
	if (my asynchronicity == Melder_SYNCHRONOUS) {
		if (write (my audio_fd, & buffer [0], 2 * my numberOfChannels * my numberOfSamples) == -1)
			return cancel (), Melder_error ("Cannot write audio output.");
		close (my audio_fd), my audio_fd = 0;   /* Drain. */
		my samplesPlayed = my numberOfSamples;
	} else if (my asynchronicity <= Melder_INTERRUPTABLE) {
		int interrupted = 0;
		while (my samplesLeft && ! interrupted) {
			int dsamples = my samplesLeft > 2000 ? 2000 : my samplesLeft;
			XEvent event;
			if (write (my audio_fd, (char *) & buffer [my samplesSent * my numberOfChannels], 2 * dsamples * my numberOfChannels) == -1)
				return cancel (), Melder_error ("Cannot write audio output.");
			my samplesLeft -= dsamples;
			my samplesSent += dsamples;
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				interrupted = 1;
			if (my asynchronicity == Melder_INTERRUPTABLE && XCheckIfEvent (XtDisplay (Melder_topShell), & event, predicateProcedure, 0))
				my explicit = Melder_EXPLICIT, interrupted = 1;
		}
		if (! interrupted) {
			/*
			 * Write a zero-length buffer (end-of-file record) to the output stream,
			 * so that we can use the play.eof attribute to drain the rest of the buffer.
			 */
			my audioInfo. play. eof = 0;
			write (my audio_fd, NULL, 0);
			while (ioctl (my audio_fd, AUDIO_GETINFO, & my audioInfo), my audioInfo. play. eof == 0) {
				XEvent event;
				my samplesPlayed = Melder_clock () * my sampleRate;
				if (my callback && ! my callback (my closure, my samplesPlayed))
					break;
				if (my asynchronicity == Melder_INTERRUPTABLE && XCheckIfEvent (XtDisplay (Melder_topShell), & event, predicateProcedure, 0))
					{ my explicit = Melder_EXPLICIT; break; }
			}
		}
	} else /* my asynchronicity == Melder_ASYNCHRONOUS */ {
		my workProcId = XtAppAddWorkProc (Melder_appContext, workProc, 0);
		return 1;
	}
	flush ();
	return 1;
}
#elif defined (linux)
{
	/* O_NDELAY option added by Rafael Laboissiere, May 19, 2005 */
	if ((my audio_fd = open ("/dev/dsp", O_WRONLY | (Melder_debug == 16 ? 0 : O_NDELAY))) == -1)
		return cancel (), Melder_error (errno == EBUSY ? "Audio device already in use." :
			"Cannot open audio device.\nConsult /usr/doc/HOWTO/Sound-HOWTO.");
	fcntl (my audio_fd, F_SETFL, 0);   /* Added by Rafael Laboissiere, May 19, 2005 */
	if (ioctl (my audio_fd, SNDCTL_DSP_SAMPLESIZE, (my val = 16, & my val)) == -1 ||   /* Error? */
	    my val != 16)   /* Has sound card overridden our sample size? */
	{
		return cancel (), Melder_error ("Cannot set sample size to 16 bit.");
	}
	if (ioctl (my audio_fd, SNDCTL_DSP_CHANNELS, (my val = my numberOfChannels, & my val)) == -1 ||   /* Error? */
	    my val != my numberOfChannels)   /* Has sound card overridden our number of channels? */
	{
		/*
		 * There is one specific case in which we can work around the current failure,
		 * namely when we are trying to play in mono but the driver of the sound card supports stereo only
		 * and notified us of this by overriding our number of channels.
		 */
		if (my numberOfChannels == 1 && my val == 2) {
			long isamp;
			short *newBuffer;
			my fakeMono = TRUE;
			if ((newBuffer = NUMsvector (0, 2 * numberOfSamples - 1)) == NULL)
				return cancel (), Melder_error ("Cannot fake mono.");
			for (isamp = 0; isamp < numberOfSamples; isamp ++) {
				newBuffer [isamp + isamp] = newBuffer [isamp + isamp + 1] = buffer [isamp];
			}
			my buffer = (const short *) newBuffer;
			my numberOfChannels = 2;
		} else {
			return cancel (), Melder_error ("Cannot set number of channels to %d.", my numberOfChannels);
		}
	}
	if (ioctl (my audio_fd, SNDCTL_DSP_SPEED, (my val = my sampleRate, & my val)) == -1 ||    /* Error? */
	    my val != my sampleRate)   /* Has sound card overridden our sampling frequency? */
	{
		return cancel (), Melder_error ("Cannot set sampling frequency to %d.", my sampleRate);
	}

	Melder_startClock ();
	if (my asynchronicity == Melder_SYNCHRONOUS) {
		if (write (my audio_fd, & my buffer [0], 2 * numberOfChannels * numberOfSamples) == -1)
			return cancel (), Melder_error ("Cannot write audio output.");
		close (my audio_fd), my audio_fd = 0;   /* Drain. Set to zero in order to notify flush (). */
		my samplesPlayed = my numberOfSamples;
	} else if (my asynchronicity <= Melder_INTERRUPTABLE) {
		int interrupted = 0;
		while (my samplesLeft && ! interrupted) {
			int dsamples = my samplesLeft > 500 ? 500 : my samplesLeft;
			XEvent event;
			if (write (my audio_fd, (char *) & my buffer [my samplesSent * my numberOfChannels], 2 * dsamples * my numberOfChannels) == -1)
				return cancel (), Melder_error ("Cannot write audio output.");
			my samplesLeft -= dsamples;
			my samplesSent += dsamples;
			my samplesPlayed = Melder_clock () * my sampleRate;
			if (my callback && ! my callback (my closure, my samplesPlayed))
				interrupted = 1;
			if (my asynchronicity == Melder_INTERRUPTABLE && XCheckIfEvent (XtDisplay (Melder_topShell), & event, predicateProcedure, 0))
				my explicit = Melder_EXPLICIT, interrupted = 1;
		}
		if (! interrupted) {
			/*
			 * Wait for playing to end.
			 */
			close (my audio_fd), my audio_fd = 0;   /* BUG: should do a loop. */
			my samplesPlayed = my numberOfSamples;
		}
	} else /* my asynchronicity == Melder_ASYNCHRONOUS */ {
		my workProcId = XtAppAddWorkProc (Melder_appContext, workProc, 0);
		return 1;
	}
	flush ();
	return 1;
}
#else
{
	return Melder_error ("Cannot play a sound on this machine.");
}
#endif
}

/********** WAITING FOR SOUND INPUT **********/

void Melder_audioTrigger (void) {
#ifdef sgi
	ALconfig audio = ALnewconfig ();
	ALport poort;
	short buffer;
	ALsetchannels (audio, AL_MONO);
	ALsetwidth (audio, AL_SAMPLE_16);
	poort = ALopenport ("praat", "r", audio);
	for (;;) {
		ALreadsamps (poort, & buffer, 1);
		if (buffer < -8000 || buffer > +8000) {
			ALcloseport (poort);
			ALfreeconfig (audio);
			return;
		}
	}
#endif
}

/* End of file melder_audio.c */
