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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) 1997-2001 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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|
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#include "sysdeps.h" |
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|
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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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|
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#include <dmedia/audio.h> |
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#include <dmedia/dmedia.h> |
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|
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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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|
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#define DEBUG 0 |
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#include "debug.h" |
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|
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|
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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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|
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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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|
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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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|
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|
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// Prototypes |
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static void *stream_func(void *arg); |
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|
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|
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/* |
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* Initialization |
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*/ |
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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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|
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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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|
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printf("Using libaudio audio output\n"); |
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|
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// Try to open the audio library |
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|
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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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|
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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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|
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// Set the sample rate |
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|
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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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|
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// TODO: list all supported sample formats? |
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|
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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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|
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// Compute sound buffer size and libaudio refill point |
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|
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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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|
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alZeroFrames(port, audio_frames_per_block); // so we don't underflow |
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|
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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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|
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sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block; |
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|
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// Get a file descriptor we can select() on |
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|
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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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|
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return true; |
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} |
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|
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|
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/* |
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* Initialization |
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*/ |
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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/* |
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* Deinitialization |
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*/ |
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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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|
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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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|
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/* |
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* First source added, start audio stream |
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*/ |
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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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|
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/* |
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* Last source removed, stop audio stream |
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*/ |
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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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|
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/* |
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* Streaming function |
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*/ |
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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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|
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numfds = audio_fd + 1; |
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FD_ZERO(&audio_fdset); |
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|
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while (!stream_thread_cancel) { |
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if (AudioStatus.num_sources) { |
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|
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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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|
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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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|
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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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|
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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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|
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} else { |
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|
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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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|
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// Wait for fill point to be reached (may be immediate) |
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|
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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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|
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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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|
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/* |
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* MacOS audio interrupt, read next data block |
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*/ |
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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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|
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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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|
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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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|
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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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|
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void audio_set_sample_rate(int index) |
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{ |
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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void audio_set_main_mute(bool mute) |
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{ |
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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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|
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void audio_set_speaker_mute(bool mute) |
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{ |
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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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} |
