mirror of
https://github.com/kanjitalk755/macemu.git
synced 2025-04-17 11:42:09 +00:00
- 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:
parent
c3df0cee5a
commit
348606cc5d
@ -8,6 +8,8 @@ V0.8 (snapshot) - <date>
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(default addressing mode, if possible)
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- Unix: added screen updates on SEGV signals [Gwenole Beauchesne]
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(activated through the "--enable-vosf" configure option)
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- Unix: added IRIX audio driver [Brian J. Johnson]
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- Unix: improved timing of periodic threads
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- AmigaOS: enabled floppy support, fixed floppy bugs [Jürgen Lachmann]
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- AmigaOS: Amiga mouse pointer is hidden inside windowed Mac displays
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- AmigaOS/sys_amiga.cpp: workaround for 2060scsi.device bug when
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@ -18,7 +20,6 @@ V0.8 (snapshot) - <date>
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FInfo/FXInfo, replaced get/set_finder_*() functions by get/set_finfo()
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- AmigaOS: added MacsBug support (tested with MacsBug6.6.1),
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fixed <move sr,(sp)> bug [Jürgen Lachmann]
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- Unix: improved timing of periodic threads
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- include/macos_util.h: defines FOURCC() macro to make MacOS-like
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four-character-codes, replaced most instances of multi-character
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constants in the sources by this macro to avoid compiler warnings
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413
BasiliskII/src/Unix/Irix/audio_irix.cpp
Normal file
413
BasiliskII/src/Unix/Irix/audio_irix.cpp
Normal file
@ -0,0 +1,413 @@
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/*
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* audio_irix.cpp - Audio support, SGI Irix implementation
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*
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* Basilisk II (C) 1997-2000 Christian Bauer
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "sysdeps.h"
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#include <sys/ioctl.h>
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#include <unistd.h>
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#include <errno.h>
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#include <pthread.h>
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#include <semaphore.h>
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#include <dmedia/audio.h>
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#include <dmedia/dmedia.h>
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#include "cpu_emulation.h"
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#include "main.h"
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#include "prefs.h"
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#include "user_strings.h"
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#include "audio.h"
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#include "audio_defs.h"
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#define DEBUG 0
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#include "debug.h"
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// Supported sample rates, sizes and channels (defaults)
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int audio_num_sample_rates = 1;
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uint32 audio_sample_rates[] = {44100 << 16};
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int audio_num_sample_sizes = 1;
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uint16 audio_sample_sizes[] = {16};
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int audio_num_channel_counts = 1;
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uint16 audio_channel_counts[] = {2};
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// Global variables
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static int audio_fd = -1; // fd from audio library
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static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
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static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
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static int sound_buffer_size; // Size of sound buffer in bytes
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static int sound_buffer_fill_point; // Fill buffer when this many frames are empty
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static uint8 silence_byte = 0; // Byte value to use to fill sound buffers with silence
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static pthread_t stream_thread; // Audio streaming thread
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static pthread_attr_t stream_thread_attr; // Streaming thread attributes
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static bool stream_thread_active = false; // Flag: streaming thread installed
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static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
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// IRIX libaudio control structures
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static ALconfig config;
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static ALport port;
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// Prototypes
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static void *stream_func(void *arg);
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/*
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* Initialization
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*/
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// Set AudioStatus to reflect current audio stream format
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static void set_audio_status_format(void)
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{
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AudioStatus.sample_rate = audio_sample_rates[0];
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AudioStatus.sample_size = audio_sample_sizes[0];
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AudioStatus.channels = audio_channel_counts[0];
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}
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// Init libaudio, returns false on error
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bool audio_init_al(void)
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{
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ALpv pv[2];
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printf("Using libaudio audio output\n");
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// Try to open the audio library
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config = alNewConfig();
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alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
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alSetWidth(config, AL_SAMPLE_16);
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alSetChannels(config, 2); // stereo
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alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
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port = alOpenPort("BasiliskII", "w", config);
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if (port == NULL) {
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fprintf(stderr, "ERROR: Cannot open audio port: %s\n",
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alGetErrorString(oserror()));
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return false;
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}
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// Set the sample rate
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pv[0].param = AL_RATE;
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pv[0].value.ll = alDoubleToFixed(audio_sample_rates[0] >> 16);
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pv[1].param = AL_MASTER_CLOCK;
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pv[1].value.i = AL_CRYSTAL_MCLK_TYPE;
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if (alSetParams(AL_DEFAULT_OUTPUT, pv, 2) < 0) {
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fprintf(stderr, "ERROR: libaudio setparams failed: %s\n",
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alGetErrorString(oserror()));
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alClosePort(port);
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return false;
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}
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// TODO: list all supported sample formats?
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// Set AudioStatus again because we now know more about the sound
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// system's capabilities
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set_audio_status_format();
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// Compute sound buffer size and libaudio refill point
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config = alGetConfig(port);
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audio_frames_per_block = alGetQueueSize(config);
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if (audio_frames_per_block < 0) {
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fprintf(stderr, "ERROR: couldn't get queue size: %s\n",
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alGetErrorString(oserror()));
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alClosePort(port);
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return false;
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}
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D(bug("alGetQueueSize %d\n", audio_frames_per_block));
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alZeroFrames(port, audio_frames_per_block); // so we don't underflow
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// Put a limit on the Mac sound buffer size, to decrease delay
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if (audio_frames_per_block > 2048)
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audio_frames_per_block = 2048;
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// Try to keep the buffer pretty full. 5000 samples of slack works well.
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sound_buffer_fill_point = alGetQueueSize(config) - 5000;
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if (sound_buffer_fill_point < 0)
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sound_buffer_fill_point = alGetQueueSize(config) / 3;
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D(bug("fill point %d\n", sound_buffer_fill_point));
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sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
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// Get a file descriptor we can select() on
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audio_fd = alGetFD(port);
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if (audio_fd < 0) {
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fprintf(stderr, "ERROR: couldn't get libaudio file descriptor: %s\n",
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alGetErrorString(oserror()));
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alClosePort(port);
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return false;
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}
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return true;
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}
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/*
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* Initialization
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*/
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void AudioInit(void)
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{
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// Init audio status (defaults) and feature flags
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set_audio_status_format();
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AudioStatus.mixer = 0;
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AudioStatus.num_sources = 0;
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audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;
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// Sound disabled in prefs? Then do nothing
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if (PrefsFindBool("nosound"))
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return;
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// Try to open audio library
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if (!audio_init_al())
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return;
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// Init semaphore
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if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
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return;
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sem_inited = true;
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// Start streaming thread
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pthread_attr_init(&stream_thread_attr);
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#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
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if (geteuid() == 0) {
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pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
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pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
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struct sched_param fifo_param;
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fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
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pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
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}
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#endif
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stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
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// Everything OK
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audio_open = true;
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}
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/*
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* Deinitialization
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*/
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void AudioExit(void)
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{
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// Stop stream and delete semaphore
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if (stream_thread_active) {
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stream_thread_cancel = true;
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#ifdef HAVE_PTHREAD_CANCEL
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pthread_cancel(stream_thread);
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#endif
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pthread_join(stream_thread, NULL);
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stream_thread_active = false;
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}
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if (sem_inited)
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sem_destroy(&audio_irq_done_sem);
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// Close audio library
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alClosePort(port);
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}
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/*
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* First source added, start audio stream
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*/
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void audio_enter_stream()
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{
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// Streaming thread is always running to avoid clicking noises
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}
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/*
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* Last source removed, stop audio stream
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*/
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void audio_exit_stream()
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{
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// Streaming thread is always running to avoid clicking noises
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}
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/*
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* Streaming function
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*/
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static void *stream_func(void *arg)
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{
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int16 *last_buffer = new int16[sound_buffer_size / 2];
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fd_set audio_fdset;
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int numfds, was_error;
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numfds = audio_fd + 1;
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FD_ZERO(&audio_fdset);
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while (!stream_thread_cancel) {
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if (AudioStatus.num_sources) {
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// Trigger audio interrupt to get new buffer
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D(bug("stream: triggering irq\n"));
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SetInterruptFlag(INTFLAG_AUDIO);
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TriggerInterrupt();
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D(bug("stream: waiting for ack\n"));
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sem_wait(&audio_irq_done_sem);
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D(bug("stream: ack received\n"));
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uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
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// Get size of audio data
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apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
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if (apple_stream_info) {
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int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
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D(bug("stream: work_size %d\n", work_size));
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if (work_size > sound_buffer_size)
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work_size = sound_buffer_size;
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if (work_size == 0)
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goto silence;
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// Send data to audio library
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if (work_size == sound_buffer_size)
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alWriteFrames(port, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), audio_frames_per_block);
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else {
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// Last buffer
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Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
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memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
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alWriteFrames(port, last_buffer, audio_frames_per_block);
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}
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D(bug("stream: data written\n"));
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} else
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goto silence;
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} else {
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// Audio not active, play silence
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silence: // D(bug("stream: silence\n"));
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alZeroFrames(port, audio_frames_per_block);
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}
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// Wait for fill point to be reached (may be immediate)
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if (alSetFillPoint(port, sound_buffer_fill_point) < 0) {
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fprintf(stderr, "ERROR: alSetFillPoint failed: %s\n",
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alGetErrorString(oserror()));
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// Should stop the audio here....
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}
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do {
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errno = 0;
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FD_SET(audio_fd, &audio_fdset);
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was_error = select(numfds, NULL, &audio_fdset, NULL, NULL);
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} while(was_error < 0 && (errno == EINTR));
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if (was_error < 0) {
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fprintf(stderr, "ERROR: select returned %d, errno %d\n",
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was_error, errno);
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// Should stop audio here....
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}
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}
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delete[] last_buffer;
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return NULL;
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}
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/*
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* MacOS audio interrupt, read next data block
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*/
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void AudioInterrupt(void)
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{
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D(bug("AudioInterrupt\n"));
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// Get data from apple mixer
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if (AudioStatus.mixer) {
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M68kRegisters r;
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r.a[0] = audio_data + adatStreamInfo;
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r.a[1] = AudioStatus.mixer;
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Execute68k(audio_data + adatGetSourceData, &r);
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D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
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} else
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WriteMacInt32(audio_data + adatStreamInfo, 0);
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// Signal stream function
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sem_post(&audio_irq_done_sem);
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D(bug("AudioInterrupt done\n"));
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}
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/*
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* Set sampling parameters
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* "index" is an index into the audio_sample_rates[] etc. arrays
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* It is guaranteed that AudioStatus.num_sources == 0
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*/
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void audio_set_sample_rate(int index)
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{
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}
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void audio_set_sample_size(int index)
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{
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}
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void audio_set_channels(int index)
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{
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}
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/*
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* Get/set volume controls (volume values received/returned have the left channel
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* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
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* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
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*/
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bool audio_get_main_mute(void)
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{
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return false;
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}
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uint32 audio_get_main_volume(void)
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{
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return 0x01000100;
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}
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bool audio_get_speaker_mute(void)
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{
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return false;
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}
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uint32 audio_get_speaker_volume(void)
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{
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return 0x01000100;
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}
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void audio_set_main_mute(bool mute)
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{
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}
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void audio_set_main_volume(uint32 vol)
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{
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}
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void audio_set_speaker_mute(bool mute)
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{
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}
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void audio_set_speaker_volume(uint32 vol)
|
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{
|
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}
|
6
BasiliskII/src/Unix/aclocal.m4
vendored
6
BasiliskII/src/Unix/aclocal.m4
vendored
@ -245,6 +245,8 @@ AC_ARG_ENABLE(esdtest, [ --disable-esdtest Do not try to compile and run
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if test "$ESD_CONFIG" = "no" ; then
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no_esd=yes
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else
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AC_LANG_SAVE
|
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AC_LANG_C
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ESD_CFLAGS=`$ESD_CONFIG $esdconf_args --cflags`
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ESD_LIBS=`$ESD_CONFIG $esdconf_args --libs`
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|
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@ -321,6 +323,7 @@ int main ()
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],, no_esd=yes,[echo $ac_n "cross compiling; assumed OK... $ac_c"])
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CFLAGS="$ac_save_CFLAGS"
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LIBS="$ac_save_LIBS"
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AC_LANG_RESTORE
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fi
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fi
|
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if test "x$no_esd" = x ; then
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@ -340,6 +343,8 @@ int main ()
|
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echo "*** Could not run ESD test program, checking why..."
|
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CFLAGS="$CFLAGS $ESD_CFLAGS"
|
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LIBS="$LIBS $ESD_LIBS"
|
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AC_LANG_SAVE
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AC_LANG_C
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AC_TRY_LINK([
|
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#include <stdio.h>
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#include <esd.h>
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@ -359,6 +364,7 @@ int main ()
|
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echo "*** may want to edit the esd-config script: $ESD_CONFIG" ])
|
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CFLAGS="$ac_save_CFLAGS"
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LIBS="$ac_save_LIBS"
|
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AC_LANG_RESTORE
|
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fi
|
||||
fi
|
||||
ESD_CFLAGS=""
|
||||
|
@ -259,10 +259,11 @@ solaris*)
|
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DEFINES="$DEFINES -DBSD_COMP -D_POSIX_PTHREAD_SEMANTICS"
|
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;;
|
||||
irix*)
|
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AUDIOSRC=Irix/audio_irix.cpp
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EXTRASYSSRCS=Irix/unaligned.c
|
||||
dnl IRIX headers work fine, but somehow don't define or use "STDC_HEADERS"
|
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DEFINES="$DEFINES -DCRTSCTS=CNEW_RTSCTS -DB230400=B115200 -DSTDC_HEADERS"
|
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LIBS="$LIBS -lm"
|
||||
LIBS="$LIBS -laudio"
|
||||
;;
|
||||
esac
|
||||
|
||||
|
@ -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));
|
||||
}
|
||||
|
||||
|
@ -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
|
||||
|
||||
|
Loading…
x
Reference in New Issue
Block a user