diff --git a/BasiliskII/ChangeLog b/BasiliskII/ChangeLog
index 83c93648..3c1ff4dc 100644
--- a/BasiliskII/ChangeLog
+++ b/BasiliskII/ChangeLog
@@ -8,6 +8,8 @@ V0.8 (snapshot) - <date>
 	  (default addressing mode, if possible)
 	- Unix: added screen updates on SEGV signals [Gwenole Beauchesne]
 	  (activated through the "--enable-vosf" configure option)
+	- Unix: added IRIX audio driver [Brian J. Johnson]
+	- Unix: improved timing of periodic threads
 	- AmigaOS: enabled floppy support, fixed floppy bugs [Jürgen Lachmann]
 	- AmigaOS: Amiga mouse pointer is hidden inside windowed Mac displays
 	- AmigaOS/sys_amiga.cpp: workaround for 2060scsi.device bug when
@@ -18,7 +20,6 @@ V0.8 (snapshot) - <date>
 	  FInfo/FXInfo, replaced get/set_finder_*() functions by get/set_finfo()
 	- AmigaOS: added MacsBug support (tested with MacsBug6.6.1),
 	  fixed <move sr,(sp)> bug [Jürgen Lachmann]
-	- Unix: improved timing of periodic threads
 	- include/macos_util.h: defines FOURCC() macro to make MacOS-like
 	  four-character-codes, replaced most instances of multi-character
 	  constants in the sources by this macro to avoid compiler warnings
diff --git a/BasiliskII/src/Unix/Irix/audio_irix.cpp b/BasiliskII/src/Unix/Irix/audio_irix.cpp
new file mode 100644
index 00000000..c4b9ecab
--- /dev/null
+++ b/BasiliskII/src/Unix/Irix/audio_irix.cpp
@@ -0,0 +1,413 @@
+/*
+ *  audio_irix.cpp - Audio support, SGI Irix implementation
+ *
+ *  Basilisk II (C) 1997-2000 Christian Bauer
+ *
+ *  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., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include "sysdeps.h"
+
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <errno.h>
+#include <pthread.h>
+#include <semaphore.h>
+
+#include <dmedia/audio.h>
+#include <dmedia/dmedia.h>
+
+#include "cpu_emulation.h"
+#include "main.h"
+#include "prefs.h"
+#include "user_strings.h"
+#include "audio.h"
+#include "audio_defs.h"
+
+#define DEBUG 0
+#include "debug.h"
+
+
+// Supported sample rates, sizes and channels (defaults)
+int audio_num_sample_rates = 1;
+uint32 audio_sample_rates[] = {44100 << 16};
+int audio_num_sample_sizes = 1;
+uint16 audio_sample_sizes[] = {16};
+int audio_num_channel_counts = 1;
+uint16 audio_channel_counts[] = {2};
+
+// Global variables
+static int audio_fd = -1;							// fd from audio library
+static sem_t audio_irq_done_sem;					// Signal from interrupt to streaming thread: data block read
+static bool sem_inited = false;						// Flag: audio_irq_done_sem initialized
+static int sound_buffer_size;						// Size of sound buffer in bytes
+static int sound_buffer_fill_point;					// Fill buffer when this many frames are empty
+static uint8 silence_byte = 0;						// Byte value to use to fill sound buffers with silence
+static pthread_t stream_thread;						// Audio streaming thread
+static pthread_attr_t stream_thread_attr;			// Streaming thread attributes
+static bool stream_thread_active = false;			// Flag: streaming thread installed
+static volatile bool stream_thread_cancel = false;	// Flag: cancel streaming thread
+
+// IRIX libaudio control structures
+static ALconfig config;
+static ALport   port;
+
+
+// Prototypes
+static void *stream_func(void *arg);
+
+
+/*
+ *  Initialization
+ */
+
+// Set AudioStatus to reflect current audio stream format
+static void set_audio_status_format(void)
+{
+	AudioStatus.sample_rate = audio_sample_rates[0];
+	AudioStatus.sample_size = audio_sample_sizes[0];
+	AudioStatus.channels = audio_channel_counts[0];
+}
+
+// Init libaudio, returns false on error
+bool audio_init_al(void)
+{
+	ALpv     pv[2];
+
+	printf("Using libaudio audio output\n");
+
+	// Try to open the audio library
+
+	config = alNewConfig();
+	alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
+	alSetWidth(config, AL_SAMPLE_16);
+	alSetChannels(config, 2);	// stereo
+	alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
+	
+	port = alOpenPort("BasiliskII", "w", config);
+	if (port == NULL) {
+		fprintf(stderr, "ERROR: Cannot open audio port: %s\n", 
+				alGetErrorString(oserror()));
+		return false;
+	}
+
+	// Set the sample rate
+
+	pv[0].param = AL_RATE;
+	pv[0].value.ll = alDoubleToFixed(audio_sample_rates[0] >> 16);
+	pv[1].param = AL_MASTER_CLOCK;
+	pv[1].value.i = AL_CRYSTAL_MCLK_TYPE;
+	if (alSetParams(AL_DEFAULT_OUTPUT, pv, 2) < 0) {
+		fprintf(stderr, "ERROR: libaudio setparams failed: %s\n",
+				alGetErrorString(oserror()));
+		alClosePort(port);
+		return false;
+	}
+
+	// TODO:  list all supported sample formats?
+
+	// Set AudioStatus again because we now know more about the sound
+	// system's capabilities
+	set_audio_status_format();
+
+	// Compute sound buffer size and libaudio refill point
+
+	config = alGetConfig(port);
+	audio_frames_per_block = alGetQueueSize(config);
+	if (audio_frames_per_block < 0) {
+		fprintf(stderr, "ERROR: couldn't get queue size: %s\n",
+				alGetErrorString(oserror()));
+		alClosePort(port);
+		return false;
+	}
+	D(bug("alGetQueueSize %d\n", audio_frames_per_block));
+
+	alZeroFrames(port, audio_frames_per_block);	// so we don't underflow
+
+	// Put a limit on the Mac sound buffer size, to decrease delay
+	if (audio_frames_per_block > 2048)
+		audio_frames_per_block = 2048;
+	// Try to keep the buffer pretty full.  5000 samples of slack works well.
+	sound_buffer_fill_point = alGetQueueSize(config) - 5000;
+	if (sound_buffer_fill_point < 0)
+		sound_buffer_fill_point = alGetQueueSize(config) / 3;
+	D(bug("fill point %d\n", sound_buffer_fill_point));
+
+	sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
+
+	// Get a file descriptor we can select() on
+
+	audio_fd = alGetFD(port);
+	if (audio_fd < 0) {
+		fprintf(stderr, "ERROR: couldn't get libaudio file descriptor: %s\n",
+				alGetErrorString(oserror()));
+		alClosePort(port);
+		return false;
+	}
+
+	return true;
+}
+
+
+/*
+ *  Initialization
+ */
+
+void AudioInit(void)
+{
+	// Init audio status (defaults) and feature flags
+	set_audio_status_format();
+	AudioStatus.mixer = 0;
+	AudioStatus.num_sources = 0;
+	audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;
+
+	// Sound disabled in prefs? Then do nothing
+	if (PrefsFindBool("nosound"))
+		return;
+
+	// Try to open audio library
+	if (!audio_init_al())
+			return;
+
+	// Init semaphore
+	if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
+		return;
+	sem_inited = true;
+
+	// Start streaming thread
+	pthread_attr_init(&stream_thread_attr);
+#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
+	if (geteuid() == 0) {
+		pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
+		pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
+		struct sched_param fifo_param;
+		fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
+		pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
+	}
+#endif
+	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
+
+	// Everything OK
+	audio_open = true;
+}
+
+
+/*
+ *  Deinitialization
+ */
+
+void AudioExit(void)
+{
+	// Stop stream and delete semaphore
+	if (stream_thread_active) {
+		stream_thread_cancel = true;
+#ifdef HAVE_PTHREAD_CANCEL
+		pthread_cancel(stream_thread);
+#endif
+		pthread_join(stream_thread, NULL);
+		stream_thread_active = false;
+	}
+	if (sem_inited)
+		sem_destroy(&audio_irq_done_sem);
+
+	// Close audio library
+	alClosePort(port);
+}
+
+
+/*
+ *  First source added, start audio stream
+ */
+
+void audio_enter_stream()
+{
+	// Streaming thread is always running to avoid clicking noises
+}
+
+
+/*
+ *  Last source removed, stop audio stream
+ */
+
+void audio_exit_stream()
+{
+	// Streaming thread is always running to avoid clicking noises
+}
+
+
+/*
+ *  Streaming function
+ */
+
+static void *stream_func(void *arg)
+{
+	int16 *last_buffer = new int16[sound_buffer_size / 2];
+	fd_set audio_fdset;
+	int    numfds, was_error;
+
+	numfds = audio_fd + 1;
+	FD_ZERO(&audio_fdset);
+
+	while (!stream_thread_cancel) {
+		if (AudioStatus.num_sources) {
+
+			// Trigger audio interrupt to get new buffer
+			D(bug("stream: triggering irq\n"));
+			SetInterruptFlag(INTFLAG_AUDIO);
+			TriggerInterrupt();
+			D(bug("stream: waiting for ack\n"));
+			sem_wait(&audio_irq_done_sem);
+			D(bug("stream: ack received\n"));
+
+			uint32 apple_stream_info;	// Mac address of SoundComponentData struct describing next buffer
+			// Get size of audio data
+			apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
+
+			if (apple_stream_info) {
+				int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
+				D(bug("stream: work_size %d\n", work_size));
+				if (work_size > sound_buffer_size)
+					work_size = sound_buffer_size;
+				if (work_size == 0)
+					goto silence;
+
+				// Send data to audio library
+				if (work_size == sound_buffer_size)
+					alWriteFrames(port, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), audio_frames_per_block);
+				else {
+					// Last buffer
+					Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
+					memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
+					alWriteFrames(port, last_buffer, audio_frames_per_block);
+				}
+				D(bug("stream: data written\n"));
+			} else
+				goto silence;
+
+		} else {
+
+			// Audio not active, play silence
+		silence:	// D(bug("stream: silence\n"));
+			alZeroFrames(port, audio_frames_per_block);
+		}
+
+		// Wait for fill point to be reached (may be immediate)
+
+		if (alSetFillPoint(port, sound_buffer_fill_point) < 0) {
+			fprintf(stderr, "ERROR: alSetFillPoint failed: %s\n",
+					alGetErrorString(oserror()));
+			// Should stop the audio here....
+		}
+
+		do {
+			errno = 0;
+			FD_SET(audio_fd, &audio_fdset);
+			was_error = select(numfds, NULL, &audio_fdset, NULL, NULL);
+		} while(was_error < 0 && (errno == EINTR));
+		if (was_error < 0) {
+			fprintf(stderr, "ERROR: select returned %d, errno %d\n",
+					was_error, errno);
+			// Should stop audio here....
+		}
+	}
+	delete[] last_buffer;
+	return NULL;
+}
+
+
+/*
+ *  MacOS audio interrupt, read next data block
+ */
+
+void AudioInterrupt(void)
+{
+	D(bug("AudioInterrupt\n"));
+
+	// Get data from apple mixer
+	if (AudioStatus.mixer) {
+		M68kRegisters r;
+		r.a[0] = audio_data + adatStreamInfo;
+		r.a[1] = AudioStatus.mixer;
+		Execute68k(audio_data + adatGetSourceData, &r);
+		D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
+	} else
+		WriteMacInt32(audio_data + adatStreamInfo, 0);
+
+	// Signal stream function
+	sem_post(&audio_irq_done_sem);
+	D(bug("AudioInterrupt done\n"));
+}
+
+
+/*
+ *  Set sampling parameters
+ *  "index" is an index into the audio_sample_rates[] etc. arrays
+ *  It is guaranteed that AudioStatus.num_sources == 0
+ */
+
+void audio_set_sample_rate(int index)
+{
+}
+
+void audio_set_sample_size(int index)
+{
+}
+
+void audio_set_channels(int index)
+{
+}
+
+
+/*
+ *  Get/set volume controls (volume values received/returned have the left channel
+ *  volume in the upper 16 bits and the right channel volume in the lower 16 bits;
+ *  both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
+ */
+
+bool audio_get_main_mute(void)
+{
+	return false;
+}
+
+uint32 audio_get_main_volume(void)
+{
+	return 0x01000100;
+}
+
+bool audio_get_speaker_mute(void)
+{
+	return false;
+}
+
+uint32 audio_get_speaker_volume(void)
+{
+	return 0x01000100;
+}
+
+void audio_set_main_mute(bool mute)
+{
+}
+
+void audio_set_main_volume(uint32 vol)
+{
+}
+
+void audio_set_speaker_mute(bool mute)
+{
+}
+
+void audio_set_speaker_volume(uint32 vol)
+{
+}
diff --git a/BasiliskII/src/Unix/aclocal.m4 b/BasiliskII/src/Unix/aclocal.m4
index 3243d88f..7d10bf23 100644
--- a/BasiliskII/src/Unix/aclocal.m4
+++ b/BasiliskII/src/Unix/aclocal.m4
@@ -245,6 +245,8 @@ AC_ARG_ENABLE(esdtest, [  --disable-esdtest       Do not try to compile and run
   if test "$ESD_CONFIG" = "no" ; then
     no_esd=yes
   else
+    AC_LANG_SAVE
+    AC_LANG_C
     ESD_CFLAGS=`$ESD_CONFIG $esdconf_args --cflags`
     ESD_LIBS=`$ESD_CONFIG $esdconf_args --libs`
 
@@ -321,6 +323,7 @@ int main ()
 ],, no_esd=yes,[echo $ac_n "cross compiling; assumed OK... $ac_c"])
        CFLAGS="$ac_save_CFLAGS"
        LIBS="$ac_save_LIBS"
+       AC_LANG_RESTORE
      fi
   fi
   if test "x$no_esd" = x ; then
@@ -340,6 +343,8 @@ int main ()
           echo "*** Could not run ESD test program, checking why..."
           CFLAGS="$CFLAGS $ESD_CFLAGS"
           LIBS="$LIBS $ESD_LIBS"
+          AC_LANG_SAVE
+          AC_LANG_C
           AC_TRY_LINK([
 #include <stdio.h>
 #include <esd.h>
@@ -359,6 +364,7 @@ int main ()
           echo "*** may want to edit the esd-config script: $ESD_CONFIG" ])
           CFLAGS="$ac_save_CFLAGS"
           LIBS="$ac_save_LIBS"
+          AC_LANG_RESTORE
        fi
      fi
      ESD_CFLAGS=""
diff --git a/BasiliskII/src/Unix/configure.in b/BasiliskII/src/Unix/configure.in
index a0d7e190..e94dd8fd 100644
--- a/BasiliskII/src/Unix/configure.in
+++ b/BasiliskII/src/Unix/configure.in
@@ -259,10 +259,11 @@ solaris*)
   DEFINES="$DEFINES -DBSD_COMP -D_POSIX_PTHREAD_SEMANTICS"
   ;;
 irix*)
+  AUDIOSRC=Irix/audio_irix.cpp
   EXTRASYSSRCS=Irix/unaligned.c
   dnl IRIX headers work fine, but somehow don't define or use "STDC_HEADERS"
   DEFINES="$DEFINES -DCRTSCTS=CNEW_RTSCTS -DB230400=B115200 -DSTDC_HEADERS"
-  LIBS="$LIBS -lm"
+  LIBS="$LIBS -laudio"
   ;;
 esac
 
diff --git a/BasiliskII/src/Unix/main_unix.cpp b/BasiliskII/src/Unix/main_unix.cpp
index 9f2293d6..49263ca2 100644
--- a/BasiliskII/src/Unix/main_unix.cpp
+++ b/BasiliskII/src/Unix/main_unix.cpp
@@ -860,27 +860,35 @@ uint64 GetTicks_usec(void)
 
 /*
  *  Delay by specified number of microseconds (<1 second)
- *  (adapted from SDL_Delay() source)
+ *  (adapted from SDL_Delay() source; this function is designed to provide
+ *  the highest accuracy possible)
  */
 
 void Delay_usec(uint32 usec)
 {
 	int was_error;
-#ifndef __linux__	// Non-Linux implementations need to calculate time left
+
+#if defined(linux)
+	struct timeval tv;
+#elif defined(__FreeBSD__) || defined(sgi)
+	struct timespec elapsed, tv;
+#else	// Non-Linux implementations need to calculate time left
 	uint64 then, now, elapsed;
 #endif
-	struct timeval tv;
 
 	// Set the timeout interval - Linux only needs to do this once
-#ifdef __linux__
+#if defined(linux)
 	tv.tv_sec = 0;
 	tv.tv_usec = usec;
+#elif defined(__FreeBSD__)
+	elapsed.tv_sec = 0;
+	elapsed.tv_nsec = usec * 1000;
 #else
 	then = GetTicks_usec();
 #endif
 	do {
 		errno = 0;
-#ifndef __linux__
+#if !defined(linux) && !defined(__FreeBSD__) && !defined(sgi)
 		/* Calculate the time interval left (in case of interrupt) */
 		now = GetTicks_usec();
 		elapsed = now - then;
@@ -891,7 +899,13 @@ void Delay_usec(uint32 usec)
 		tv.tv_sec = 0;
 		tv.tv_usec = usec;
 #endif
+#if defined(__FreeBSD__) || defined(sgi)
+		tv.tv_sec = elapsed.tv_sec;
+		tv.tv_nsec = elapsed.tv_nsec;
+		was_error = nanosleep(&tv, &elapsed);
+#else
 		was_error = select(0, NULL, NULL, NULL, &tv);
+#endif
 	} while (was_error && (errno == EINTR));
 }
 
diff --git a/BasiliskII/src/Unix/video_x.cpp b/BasiliskII/src/Unix/video_x.cpp
index 83ec7b86..b223deb1 100644
--- a/BasiliskII/src/Unix/video_x.cpp
+++ b/BasiliskII/src/Unix/video_x.cpp
@@ -1239,7 +1239,7 @@ void VideoInterrupt(void)
 
 void video_set_palette(uint8 *pal)
 {
-#ifdef HAVE_PTHREDS
+#ifdef HAVE_PTHREADS
 	pthread_mutex_lock(&palette_lock);
 #endif