Can't get sound on Basillisk II

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eawo2k4
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Can't get sound on Basillisk II

Post by eawo2k4 »

I'm running on Linux which the OS I'm using is Zorin os and I download the window version with no full screen, and when I start it it says no audio device found will be disable, is there fix for this?
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adespoton
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Re: Can't get sound on Basillisk II

Post by adespoton »

Sounds like a Zorin issue -- are you using PulseAudio or ALSA?
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eawo2k4
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Re: Can't get sound on Basillisk II

Post by eawo2k4 »

I believe it's Pulse Audio
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adespoton
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Re: Can't get sound on Basillisk II

Post by adespoton »

It's possible BII may be looking for Alsa; it's been a while since I looked into that code.

https://github.com/cebix/macemu/blob/ma ... ss_esd.cpp

Looks like you may need to change your dsp settings in your prefs file.
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Cat_7
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Re: Can't get sound on Basillisk II

Post by Cat_7 »

Or check whether you can install a pulseaudio-alsa bridge from your distro.

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eawo2k4
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Re: Can't get sound on Basillisk II

Post by eawo2k4 »

So, I tried that audio_oss it's still giving me that error and tried with Pulseaudo alsa audio too, still doesn't see sound, would I have to download it again or try a different one?
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Re: Can't get sound on Basillisk II

Post by adespoton »

I don't think you're going to find a different download; what does the dsp line in your prefs file say? And does SheepShaver work correctly?
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eawo2k4
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Re: Can't get sound on Basillisk II

Post by eawo2k4 »

I don't know if you wanted me to post the code or not. but here ya go,. :smile:











#include "sysdeps.h"

#include <sys/ioctl.h>
#include <unistd.h>
#include <errno.h>
#include <pthread.h>
#include <semaphore.h>

#ifdef __linux__
#include <linux/soundcard.h>
#endif

#ifdef __FreeBSD__
#include <sys/soundcard.h>
#endif

#include "cpu_emulation.h"
#include "main.h"
#include "prefs.h"
#include "user_strings.h"
#include "audio.h"
#include "audio_defs.h"

#ifdef ENABLE_ESD
#include <esd.h>
#endif

#define DEBUG 0
#include "debug.h"


// The currently selected audio parameters (indices in audio_sample_rates[] etc. vectors)
static int audio_sample_rate_index = 0;
static int audio_sample_size_index = 0;
static int audio_channel_count_index = 0;

// Global variables
static bool is_dsp_audio = false; // Flag: is DSP audio
static int audio_fd = -1; // fd of dsp or ESD
static int mixer_fd = -1; // fd of mixer
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 bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
static uint8 silence_byte; // 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

// 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[audio_sample_rate_index];
AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
}

// Init using the dsp device, returns false on error
static bool open_dsp(void)
{
// Open the device
const char *dsp = PrefsFindString("dsp");
audio_fd = open(dsp, O_WRONLY);
if (audio_fd < 0) {
fprintf(stderr, "WARNING: Cannot open %s (%s)\n", dsp, strerror(errno));
return false;
}

printf("Using %s audio output\n", dsp);
is_dsp_audio = true;

// Get supported sample formats
if (audio_sample_sizes.empty()) {
unsigned long format;
ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
if (format & AFMT_U8)
audio_sample_sizes.push_back(8);
if (format & (AFMT_S16_BE | AFMT_S16_LE))
audio_sample_sizes.push_back(16);

int stereo = 0;
if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 0)
audio_channel_counts.push_back(1);
stereo = 1;
if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 1)
audio_channel_counts.push_back(2);

if (audio_sample_sizes.empty() || audio_channel_counts.empty()) {
WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
close(audio_fd);
audio_fd = -1;
return false;
}

audio_sample_rates.push_back(11025 << 16);
audio_sample_rates.push_back(22050 << 16);
int rate = 44100;
ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
if (rate > 22050)
audio_sample_rates.push_back(rate << 16);

// Default to highest supported values
audio_sample_rate_index = audio_sample_rates.size() - 1;
audio_sample_size_index = audio_sample_sizes.size() - 1;
audio_channel_count_index = audio_channel_counts.size() - 1;
}

// Set DSP parameters
unsigned long format;
if (audio_sample_sizes[audio_sample_size_index] == 8) {
format = AFMT_U8;
little_endian = false;
silence_byte = 0x80;
} else {
unsigned long sup_format;
ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &sup_format);
if (sup_format & AFMT_S16_BE) {
little_endian = false;
format = AFMT_S16_BE;
} else {
little_endian = true;
format = AFMT_S16_LE;
}
silence_byte = 0;
}
ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
int frag = 0x0004000c; // Block size: 4096 frames
ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
int stereo = (audio_channel_counts[audio_channel_count_index] == 2);
ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);

// Get sound buffer size
ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
return true;
}

// Init using ESD, returns false on error
static bool open_esd(void)
{
#ifdef ENABLE_ESD
int rate;
esd_format_t format = ESD_STREAM | ESD_PLAY;

if (audio_sample_sizes.empty()) {

// Default values
rate = 44100;
format |= (ESD_BITS16 | ESD_STEREO);

} else {

rate = audio_sample_rates[audio_sample_rate_index] >> 16;
if (audio_sample_sizes[audio_sample_size_index] == 8)
format |= ESD_BITS8;
else
format |= ESD_BITS16;
if (audio_channel_counts[audio_channel_count_index] == 1)
format |= ESD_MONO;
else
format |= ESD_STEREO;
}

#if WORDS_BIGENDIAN
little_endian = false;
#else
little_endian = true;
#endif
silence_byte = 0; // Is this correct for 8-bit mode?

// Open connection to ESD server
audio_fd = esd_play_stream(format, rate, NULL, NULL);
if (audio_fd < 0) {
fprintf(stderr, "WARNING: Cannot open ESD connection\n");
return false;
}

printf("Using ESD audio output\n");

// ESD supports a variety of twisted little audio formats, all different
if (audio_sample_sizes.empty()) {

// The reason we do this here is that we don't want to add sample
// rates etc. unless the ESD server connection could be opened
// (if ESD fails, dsp might be tried next)
audio_sample_rates.push_back(11025 << 16);
audio_sample_rates.push_back(22050 << 16);
audio_sample_rates.push_back(44100 << 16);
audio_sample_sizes.push_back(8);
audio_sample_sizes.push_back(16);
audio_channel_counts.push_back(1);
audio_channel_counts.push_back(2);

// Default to highest supported values
audio_sample_rate_index = audio_sample_rates.size() - 1;
audio_sample_size_index = audio_sample_sizes.size() - 1;
audio_channel_count_index = audio_channel_counts.size() - 1;
}

// Sound buffer size = 4096 frames
audio_frames_per_block = 4096;
return true;
#else
// ESD is not enabled, shut up the compiler
return false;
#endif
}

static bool open_audio(void)
{
#ifdef ENABLE_ESD
// If ESPEAKER is set, the user probably wants to use ESD, so try that first
if (getenv("ESPEAKER"))
if (open_esd())
goto dev_opened;
#endif

// Try to open dsp
if (open_dsp())
goto dev_opened;

#ifdef ENABLE_ESD
// Hm, dsp failed so we try ESD again if ESPEAKER wasn't set
if (!getenv("ESPEAKER"))
if (open_esd())
goto dev_opened;
#endif

// No audio device succeeded
WarningAlert(GetString(STR_NO_AUDIO_WARN));
return false;

// Device opened, set AudioStatus
dev_opened:
sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) * audio_channel_counts[audio_channel_count_index] * audio_frames_per_block;
set_audio_status_format();

// Start streaming thread
Set_pthread_attr(&stream_thread_attr, 0);
stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);

// Everything went fine
audio_open = true;
return true;
}

void AudioInit(void)
{
// Init audio status (reasonable defaults) and feature flags
AudioStatus.sample_rate = 44100 << 16;
AudioStatus.sample_size = 16;
AudioStatus.channels = 2;
AudioStatus.mixer = 0;
AudioStatus.num_sources = 0;
audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;

// Sound disabled in prefs? Then do nothing
if (PrefsFindBool("nosound"))
return;

// Init semaphore
if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
return;
sem_inited = true;

// Try to open the mixer device
const char *mixer = PrefsFindString("mixer");
mixer_fd = open(mixer, O_RDWR);
if (mixer_fd < 0)
printf("WARNING: Cannot open %s (%s)\n", mixer, strerror(errno));

// Open and initialize audio device
open_audio();
}


/*
* Deinitialization
*/

static void close_audio(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;
}

// Close dsp or ESD socket
if (audio_fd >= 0) {
close(audio_fd);
audio_fd = -1;
}

audio_open = false;
}

void AudioExit(void)
{
// Stop the device immediately. Otherwise, close() sends
// SNDCTL_DSP_SYNC, which may hang
if (is_dsp_audio)
ioctl(audio_fd, SNDCTL_DSP_RESET, 0);

// Close audio device
close_audio();

// Delete semaphore
if (sem_inited) {
sem_destroy(&audio_irq_done_sem);
sem_inited = false;
}

// Close mixer device
if (mixer_fd >= 0) {
close(mixer_fd);
mixer_fd = -1;
}
}


/*
* 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 *silent_buffer = new int16[sound_buffer_size / 2];
int16 *last_buffer = new int16[sound_buffer_size / 2];
memset(silent_buffer, silence_byte, sound_buffer_size);

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"));

// Get size of audio data
uint32 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 DSP
if (work_size == sound_buffer_size && !little_endian)
write(audio_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
else {
// Last buffer or little-endian DSP
if (little_endian) {
int16 *p = (int16 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
for (int i=0; i<work_size/2; i++)
last_buffer = ntohs(p);
} else
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);
write(audio_fd, last_buffer, sound_buffer_size);
}
D(bug("stream: data written\n"));
} else
goto silence;

} else {

// Audio not active, play silence
silence: write(audio_fd, silent_buffer, sound_buffer_size);
}
}
delete[] silent_buffer;
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. vectors
* It is guaranteed that AudioStatus.num_sources == 0
*/

bool audio_set_sample_rate(int index)
{
close_audio();
audio_sample_rate_index = index;
return open_audio();
}

bool audio_set_sample_size(int index)
{
close_audio();
audio_sample_size_index = index;
return open_audio();
}

bool audio_set_channels(int index)
{
close_audio();
audio_channel_count_index = index;
return open_audio();
}


/*
* 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)
{
if (mixer_fd >= 0) {
int vol;
if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
int left = vol >> 8;
int right = vol & 0xff;
return ((left * 256 / 100) << 16) | (right * 256 / 100);
}
}
return 0x01000100;
}

bool audio_get_speaker_mute(void)
{
return false;
}

uint32 audio_get_speaker_volume(void)
{
if (mixer_fd >= 0) {
int vol;
if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
int left = vol >> 8;
int right = vol & 0xff;
return ((left * 256 / 100) << 16) | (right * 256 / 100);
}
}
return 0x01000100;
}

void audio_set_main_mute(bool mute)
{
}

void audio_set_main_volume(uint32 vol)
{
if (mixer_fd >= 0) {
int left = vol >> 16;
int right = vol & 0xffff;
int p = ((left * 100 / 256) << 8) | (right * 100 / 256);
ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
}
}

void audio_set_speaker_mute(bool mute)
{
}

void audio_set_speaker_volume(uint32 vol)
{
if (mixer_fd >= 0) {
int left = vol >> 16;
int right = vol & 0xffff;
int p = ((left * 100 / 256) << 8) | (right * 100 / 256);
ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
}
}
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adespoton
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Re: Can't get sound on Basillisk II

Post by adespoton »

Sorry, I guess I wasn't clear enough -- what I was looking for was your Basilisk II preferences/config file; I already linked the source code earlier in the thread.
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eawo2k4
Student Driver
Posts: 16
Joined: Thu Jul 27, 2017 12:23 am

Re: Can't get sound on Basillisk II

Post by eawo2k4 »

oh sorry about that, b I found the problem in Zorin OS, so the sound stuff was missing so I updated my system and fix it, even though it was already updated :roll:
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