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- /*
- * audio_irix.cpp - Audio support, SGI Irix implementation
- *
- * Basilisk II (C) 1997-2001 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)
- {
- }
-