- added audio support for IRIX [Brian J. Johnson]

- improved Delay_usec() under FreeBSD and IRIX
- fixed typo ("HAVE_PTHREDS") in video_x.cpp
This commit is contained in:
cebix 2000-11-02 14:45:17 +00:00
parent c3df0cee5a
commit 348606cc5d
6 changed files with 443 additions and 8 deletions

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@ -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

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@ -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)
{
}

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@ -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=""

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@ -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

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@ -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));
}

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@ -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