Unix/Irix/audio_irix.cpp

/*
 *  audio_irix.cpp - Audio support, SGI Irix implementation
 *
 *  Basilisk II (C) 1997-2008 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"


// 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 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

static bool current_main_mute = false;                          // Flag: output muted
static bool current_speaker_mute = false;                       // Flag: speaker muted
static uint32 current_main_volume = 0;                          // Output volume
static uint32 current_speaker_volume = 0;                       // Speaker volume

// IRIX libaudio control structures
static ALconfig config;
static ALport   port;


// Prototypes
static void *stream_func(void *arg);
static uint32 read_volume(void);
static bool read_mute(void);
static void set_mute(bool mute);


/*
 *  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];
}

bool open_audio(void)
{
        ALpv     pv[2];

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

        // Get supported sample formats

        if (audio_sample_sizes.empty()) {
                // All sample sizes are supported
                audio_sample_sizes.push_back(8);
                audio_sample_sizes.push_back(16);

                // Assume at least two channels are supported.  Some IRIX boxes
                // can do 4 or more...  MacOS only handles up to 2.
                audio_channel_counts.push_back(1);
                audio_channel_counts.push_back(2);

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

                audio_sample_rates.push_back( 8000 << 16);
                audio_sample_rates.push_back(11025 << 16);
                audio_sample_rates.push_back(22050 << 16);
                audio_sample_rates.push_back(44100 << 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 the sample format

        D(bug("Size %d, channels %d, rate %d\n",
                  audio_sample_sizes[audio_sample_size_index],
                  audio_channel_counts[audio_channel_count_index],
                  audio_sample_rates[audio_sample_rate_index] >> 16));
        config = alNewConfig();
        alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
        if (audio_sample_sizes[audio_sample_size_index] == 8) {
                alSetWidth(config, AL_SAMPLE_8);
        }
        else {
                alSetWidth(config, AL_SAMPLE_16);
        }
        alSetChannels(config, audio_channel_counts[audio_channel_count_index]);
        alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
        
        // Try to open the audio library

        port = alOpenPort("BasiliskII", "w", config);
        if (port == NULL) {
                fprintf(stderr, "ERROR: Cannot open audio port: %s\n", 
                                alGetErrorString(oserror()));
                WarningAlert(GetString(STR_NO_AUDIO_WARN));
                return false;
        }
        
        // Set the sample rate

        pv[0].param = AL_RATE;
        pv[0].value.ll = alDoubleToFixed(audio_sample_rates[audio_sample_rate_index] >> 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;
        }

        // 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, width %d, channels %d\n",
                  audio_frames_per_block,
                  alGetWidth(config),
                  alGetChannels(config)));

        // Put a limit on the Mac sound buffer size, to decrease delay
#define AUDIO_BUFFER_MSEC 50    // milliseconds of sound to buffer
        int target_frames_per_block = 
                (audio_sample_rates[audio_sample_rate_index] >> 16) *
                AUDIO_BUFFER_MSEC / 1000;
        if (audio_frames_per_block > target_frames_per_block)
                audio_frames_per_block = target_frames_per_block;
        D(bug("frames per block %d\n", audio_frames_per_block));

        alZeroFrames(port, audio_frames_per_block);     // so we don't underflow

        // Try to keep the buffer pretty full
        sound_buffer_fill_point = alGetQueueSize(config) - 
                2 * audio_frames_per_block;
        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 = (audio_sample_sizes[audio_sample_size_index] >> 3) *
                audio_channel_counts[audio_channel_count_index] * 
                audio_frames_per_block;
        set_audio_status_format();

        // 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;
        }

        // Initialize volume, mute settings
        current_main_volume = current_speaker_volume = read_volume();
        current_main_mute = current_speaker_mute = read_mute();


        // 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;

        // 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;
                stream_thread_cancel = false;
        }

        // Close audio library
        alClosePort(port);

        audio_open = false;
}

void AudioExit(void)
{
        // Close audio device
        close_audio();

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


/*
 *  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)
{
        int32 *last_buffer = new int32[sound_buffer_size / 4];
        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"));

                        // Get size of audio data
                        uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
                        if (!current_main_mute &&
                                !current_speaker_mute &&
                                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.  Convert 8-bit data
                                // unsigned->signed, using same algorithm as audio_amiga.cpp.
                                // It works fine for 8-bit mono, but not stereo.
                                if (AudioStatus.sample_size == 8) {
                                        uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
                                        uint32 *q = (uint32 *)last_buffer;
                                        int r = work_size >> 2;
                                        // XXX not quite right....
                                        while (r--)
                                                *q++ = *p++ ^ 0x80808080;
                                        if (work_size != sound_buffer_size)
                                                memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
                                        alWriteFrames(port, last_buffer, audio_frames_per_block);
                                }
                                else 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;
}


/*
 * Read or set the current output volume using the audio library
 */

static uint32 read_volume(void)
{
        ALpv x[2];
        ALfixed gain[8];
        double maxgain, mingain;
        ALparamInfo pi;
        uint32 ret = 0x01000100;        // default, maximum value
        int dev = alGetDevice(config);

        // Fetch the maximum and minimum gain settings

        alGetParamInfo(dev, AL_GAIN, &pi);
        maxgain = alFixedToDouble(pi.max.ll);
        mingain = alFixedToDouble(pi.min.ll);
//      printf("maxgain = %lf dB, mingain = %lf dB\n", maxgain, mingain);

        // Get the current gain values

        x[0].param = AL_GAIN;
        x[0].value.ptr = gain;
        x[0].sizeIn = sizeof(gain) / sizeof(gain[0]);
        x[1].param = AL_CHANNELS;
        if (alGetParams(dev, x, 2) < 0) {
                printf("alGetParams failed: %s\n", alGetErrorString(oserror()));
        }
        else {
                if (x[0].sizeOut < 0) {
                        printf("AL_GAIN was an unrecognized parameter\n");
                }
                else {
                        double v;
                        uint32 left, right;

                        // Left
                        v = alFixedToDouble(gain[0]);
                        if (v < mingain)
                                v = mingain;    // handle gain == -inf
                        v = (v - mingain) / (maxgain - mingain); // scale to 0..1
                        left = (uint32)(v * (double)256); // convert to 8.8 fixed point

                        // Right
                        if (x[0].sizeOut <= 1) {        // handle a mono interface
                                right = left;
                        }
                        else {
                                v = alFixedToDouble(gain[1]);
                                if (v < mingain)
                                        v = mingain; // handle gain == -inf
                                v = (v - mingain) / (maxgain - mingain); // scale to 0..1
                                right = (uint32)(v * (double)256); // convert to 8.8 fixed point
                        }

                        ret = (left << 16) | right;
                }
        }

        return ret;
}

static void set_volume(uint32 vol)
{
        ALpv x[1];
        ALfixed gain[2];                        // left and right
        double maxgain, mingain;
        ALparamInfo pi;
        int dev = alGetDevice(config);

        // Fetch the maximum and minimum gain settings

        alGetParamInfo(dev, AL_GAIN, &pi);
        maxgain = alFixedToDouble(pi.max.ll);
        mingain = alFixedToDouble(pi.min.ll);           

        // Set the new gain values

        x[0].param = AL_GAIN;
        x[0].value.ptr = gain;
        x[0].sizeIn = sizeof(gain) / sizeof(gain[0]);

        uint32 left = vol >> 16;
        uint32 right = vol & 0xffff;
        double lv, rv;

        if (left == 0 && pi.specialVals & AL_NEG_INFINITY_BIT) {
                lv = AL_NEG_INFINITY;
        }
        else {
                lv = ((double)left / 256) * (maxgain - mingain) + mingain;
        }

        if (right == 0 && pi.specialVals & AL_NEG_INFINITY_BIT) {
                rv = AL_NEG_INFINITY;
        }
        else {
                rv = ((double)right / 256) * (maxgain - mingain) + mingain;
        }

        D(bug("set_volume:  left=%lf dB, right=%lf dB\n", lv, rv));

        gain[0] = alDoubleToFixed(lv);
        gain[1] = alDoubleToFixed(rv);

        if (alSetParams(dev, x, 1) < 0) {
                printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
        }
}


/*
 * Read or set the mute setting using the audio library
 */

static bool read_mute(void)
{
        bool ret;
        int dev = alGetDevice(config);
        ALpv x;
        x.param = AL_MUTE;

        if (alGetParams(dev, &x, 1) < 0) {
                printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
                return current_main_mute; // Or just return false?
        }

        ret = x.value.i;

        D(bug("read_mute:  mute=%d\n", ret));
        return ret;
}

static void set_mute(bool mute)
{
        D(bug("set_mute: mute=%ld\n", mute));

        int dev = alGetDevice(config);
        ALpv x;
        x.param = AL_MUTE;
        x.value.i = mute;

        if (alSetParams(dev, &x, 1) < 0) {
                printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
        }
}


/*
 *  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)
{
        D(bug("audio_get_main_mute:  mute=%ld\n", current_main_mute));

        return current_main_mute;
}

uint32 audio_get_main_volume(void)
{
        uint32 ret = current_main_volume;

        D(bug("audio_get_main_volume:  vol=0x%x\n", ret));

        return ret;
}

bool audio_get_speaker_mute(void)
{
        D(bug("audio_get_speaker_mute:  mute=%ld\n", current_speaker_mute));

        return current_speaker_mute;
}

uint32 audio_get_speaker_volume(void)
{
        uint32 ret = current_speaker_volume;

        D(bug("audio_get_speaker_volume:  vol=0x%x\n", ret));

        return ret;
}

void audio_set_main_mute(bool mute)
{
        D(bug("audio_set_main_mute: mute=%ld\n", mute));

        if (mute != current_main_mute) {
                current_main_mute = mute;
        }

        set_mute(current_main_mute);
}

void audio_set_main_volume(uint32 vol)
{

        D(bug("audio_set_main_volume:  vol=%x\n", vol));

        current_main_volume = vol;

        set_volume(vol);
}

void audio_set_speaker_mute(bool mute)
{
        D(bug("audio_set_speaker_mute: mute=%ld\n", mute));

        if (mute != current_speaker_mute) {
                current_speaker_mute = mute;
        }

        set_mute(current_speaker_mute);
}

void audio_set_speaker_volume(uint32 vol)
{
        D(bug("audio_set_speaker_volume:  vol=%x\n", vol));

        current_speaker_volume = vol;

        set_volume(vol);
}
Revision:	1.10
Committed:	2008-01-01T09:40:33Z (16 years, 10 months ago) by gbeauche
Branch:	MAIN
CVS Tags:	HEAD
Changes since 1.9:	+1 -1 lines
Log Message:	Happy New Year!
#	Content
1	/*
2	* audio_irix.cpp - Audio support, SGI Irix implementation
3	*
4	* Basilisk II (C) 1997-2008 Christian Bauer
5	*
6	* This program is free software; you can redistribute it and/or modify
7	* it under the terms of the GNU General Public License as published by
8	* the Free Software Foundation; either version 2 of the License, or
9	* (at your option) any later version.
10	*
11	* This program is distributed in the hope that it will be useful,
12	* but WITHOUT ANY WARRANTY; without even the implied warranty of
13	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14	* GNU General Public License for more details.
15	*
16	* You should have received a copy of the GNU General Public License
17	* along with this program; if not, write to the Free Software
18	* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
19	*/
20
21	#include "sysdeps.h"
22
23	#include <sys/ioctl.h>
24	#include <unistd.h>
25	#include <errno.h>
26	#include <pthread.h>
27	#include <semaphore.h>
28
29	#include <dmedia/audio.h>
30	#include <dmedia/dmedia.h>
31
32	#include "cpu_emulation.h"
33	#include "main.h"
34	#include "prefs.h"
35	#include "user_strings.h"
36	#include "audio.h"
37	#include "audio_defs.h"
38
39	#define DEBUG 0
40	#include "debug.h"
41
42
43	// The currently selected audio parameters (indices in audio_sample_rates[]
44	// etc. vectors)
45	static int audio_sample_rate_index = 0;
46	static int audio_sample_size_index = 0;
47	static int audio_channel_count_index = 0;
48
49	// Global variables
50	static int audio_fd = -1; // fd from audio library
51	static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
52	static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
53	static int sound_buffer_size; // Size of sound buffer in bytes
54	static int sound_buffer_fill_point; // Fill buffer when this many frames are empty
55	static uint8 silence_byte = 0; // Byte value to use to fill sound buffers with silence
56	static pthread_t stream_thread; // Audio streaming thread
57	static pthread_attr_t stream_thread_attr; // Streaming thread attributes
58	static bool stream_thread_active = false; // Flag: streaming thread installed
59	static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
60
61	static bool current_main_mute = false; // Flag: output muted
62	static bool current_speaker_mute = false; // Flag: speaker muted
63	static uint32 current_main_volume = 0; // Output volume
64	static uint32 current_speaker_volume = 0; // Speaker volume
65
66	// IRIX libaudio control structures
67	static ALconfig config;
68	static ALport port;
69
70
71	// Prototypes
72	static void stream_func(void arg);
73	static uint32 read_volume(void);
74	static bool read_mute(void);
75	static void set_mute(bool mute);
76
77
78	/*
79	* Initialization
80	*/
81
82	// Set AudioStatus to reflect current audio stream format
83	static void set_audio_status_format(void)
84	{
85	AudioStatus.sample_rate = audio_sample_rates[audio_sample_rate_index];
86	AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
87	AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
88	}
89
90	bool open_audio(void)
91	{
92	ALpv pv[2];
93
94	printf("Using libaudio audio output\n");
95
96	// Get supported sample formats
97
98	if (audio_sample_sizes.empty()) {
99	// All sample sizes are supported
100	audio_sample_sizes.push_back(8);
101	audio_sample_sizes.push_back(16);
102
103	// Assume at least two channels are supported. Some IRIX boxes
104	// can do 4 or more... MacOS only handles up to 2.
105	audio_channel_counts.push_back(1);
106	audio_channel_counts.push_back(2);
107
108	if (audio_sample_sizes.empty() \|\| audio_channel_counts.empty()) {
109	WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
110	alClosePort(port);
111	audio_fd = -1;
112	return false;
113	}
114
115	audio_sample_rates.push_back( 8000 << 16);
116	audio_sample_rates.push_back(11025 << 16);
117	audio_sample_rates.push_back(22050 << 16);
118	audio_sample_rates.push_back(44100 << 16);
119
120	// Default to highest supported values
121	audio_sample_rate_index = audio_sample_rates.size() - 1;
122	audio_sample_size_index = audio_sample_sizes.size() - 1;
123	audio_channel_count_index = audio_channel_counts.size() - 1;
124	}
125
126	// Set the sample format
127
128	D(bug("Size %d, channels %d, rate %d\n",
129	audio_sample_sizes[audio_sample_size_index],
130	audio_channel_counts[audio_channel_count_index],
131	audio_sample_rates[audio_sample_rate_index] >> 16));
132	config = alNewConfig();
133	alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
134	if (audio_sample_sizes[audio_sample_size_index] == 8) {
135	alSetWidth(config, AL_SAMPLE_8);
136	}
137	else {
138	alSetWidth(config, AL_SAMPLE_16);
139	}
140	alSetChannels(config, audio_channel_counts[audio_channel_count_index]);
141	alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
142
143	// Try to open the audio library
144
145	port = alOpenPort("BasiliskII", "w", config);
146	if (port == NULL) {
147	fprintf(stderr, "ERROR: Cannot open audio port: %s\n",
148	alGetErrorString(oserror()));
149	WarningAlert(GetString(STR_NO_AUDIO_WARN));
150	return false;
151	}
152
153	// Set the sample rate
154
155	pv[0].param = AL_RATE;
156	pv[0].value.ll = alDoubleToFixed(audio_sample_rates[audio_sample_rate_index] >> 16);
157	pv[1].param = AL_MASTER_CLOCK;
158	pv[1].value.i = AL_CRYSTAL_MCLK_TYPE;
159	if (alSetParams(AL_DEFAULT_OUTPUT, pv, 2) < 0) {
160	fprintf(stderr, "ERROR: libaudio setparams failed: %s\n",
161	alGetErrorString(oserror()));
162	alClosePort(port);
163	return false;
164	}
165
166	// Compute sound buffer size and libaudio refill point
167
168	config = alGetConfig(port);
169	audio_frames_per_block = alGetQueueSize(config);
170	if (audio_frames_per_block < 0) {
171	fprintf(stderr, "ERROR: couldn't get queue size: %s\n",
172	alGetErrorString(oserror()));
173	alClosePort(port);
174	return false;
175	}
176	D(bug("alGetQueueSize %d, width %d, channels %d\n",
177	audio_frames_per_block,
178	alGetWidth(config),
179	alGetChannels(config)));
180
181	// Put a limit on the Mac sound buffer size, to decrease delay
182	#define AUDIO_BUFFER_MSEC 50 // milliseconds of sound to buffer
183	int target_frames_per_block =
184	(audio_sample_rates[audio_sample_rate_index] >> 16) *
185	AUDIO_BUFFER_MSEC / 1000;
186	if (audio_frames_per_block > target_frames_per_block)
187	audio_frames_per_block = target_frames_per_block;
188	D(bug("frames per block %d\n", audio_frames_per_block));
189
190	alZeroFrames(port, audio_frames_per_block); // so we don't underflow
191
192	// Try to keep the buffer pretty full
193	sound_buffer_fill_point = alGetQueueSize(config) -
194	2 * audio_frames_per_block;
195	if (sound_buffer_fill_point < 0)
196	sound_buffer_fill_point = alGetQueueSize(config) / 3;
197	D(bug("fill point %d\n", sound_buffer_fill_point));
198
199	sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) *
200	audio_channel_counts[audio_channel_count_index] *
201	audio_frames_per_block;
202	set_audio_status_format();
203
204	// Get a file descriptor we can select() on
205
206	audio_fd = alGetFD(port);
207	if (audio_fd < 0) {
208	fprintf(stderr, "ERROR: couldn't get libaudio file descriptor: %s\n",
209	alGetErrorString(oserror()));
210	alClosePort(port);
211	return false;
212	}
213
214	// Initialize volume, mute settings
215	current_main_volume = current_speaker_volume = read_volume();
216	current_main_mute = current_speaker_mute = read_mute();
217
218
219	// Start streaming thread
220	Set_pthread_attr(&stream_thread_attr, 0);
221	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
222
223	// Everything went fine
224	audio_open = true;
225	return true;
226	}
227
228	void AudioInit(void)
229	{
230	// Init audio status (reasonable defaults) and feature flags
231	AudioStatus.sample_rate = 44100 << 16;
232	AudioStatus.sample_size = 16;
233	AudioStatus.channels = 2;
234	AudioStatus.mixer = 0;
235	AudioStatus.num_sources = 0;
236	audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
237
238	// Sound disabled in prefs? Then do nothing
239	if (PrefsFindBool("nosound"))
240	return;
241
242	// Init semaphore
243	if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
244	return;
245	sem_inited = true;
246
247	// Open and initialize audio device
248	open_audio();
249	}
250
251
252	/*
253	* Deinitialization
254	*/
255
256	static void close_audio(void)
257	{
258	// Stop stream and delete semaphore
259	if (stream_thread_active) {
260	stream_thread_cancel = true;
261	#ifdef HAVE_PTHREAD_CANCEL
262	pthread_cancel(stream_thread);
263	#endif
264	pthread_join(stream_thread, NULL);
265	stream_thread_active = false;
266	stream_thread_cancel = false;
267	}
268
269	// Close audio library
270	alClosePort(port);
271
272	audio_open = false;
273	}
274
275	void AudioExit(void)
276	{
277	// Close audio device
278	close_audio();
279
280	// Delete semaphore
281	if (sem_inited) {
282	sem_destroy(&audio_irq_done_sem);
283	sem_inited = false;
284	}
285	}
286
287
288	/*
289	* First source added, start audio stream
290	*/
291
292	void audio_enter_stream()
293	{
294	// Streaming thread is always running to avoid clicking noises
295	}
296
297
298	/*
299	* Last source removed, stop audio stream
300	*/
301
302	void audio_exit_stream()
303	{
304	// Streaming thread is always running to avoid clicking noises
305	}
306
307
308	/*
309	* Streaming function
310	*/
311
312	static void stream_func(void arg)
313	{
314	int32 *last_buffer = new int32[sound_buffer_size / 4];
315	fd_set audio_fdset;
316	int numfds, was_error;
317
318	numfds = audio_fd + 1;
319	FD_ZERO(&audio_fdset);
320
321	while (!stream_thread_cancel) {
322	if (AudioStatus.num_sources) {
323
324	// Trigger audio interrupt to get new buffer
325	D(bug("stream: triggering irq\n"));
326	SetInterruptFlag(INTFLAG_AUDIO);
327	TriggerInterrupt();
328	D(bug("stream: waiting for ack\n"));
329	sem_wait(&audio_irq_done_sem);
330	D(bug("stream: ack received\n"));
331
332	// Get size of audio data
333	uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
334	if (!current_main_mute &&
335	!current_speaker_mute &&
336	apple_stream_info) {
337	int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
338	D(bug("stream: work_size %d\n", work_size));
339	if (work_size > sound_buffer_size)
340	work_size = sound_buffer_size;
341	if (work_size == 0)
342	goto silence;
343
344	// Send data to audio library. Convert 8-bit data
345	// unsigned->signed, using same algorithm as audio_amiga.cpp.
346	// It works fine for 8-bit mono, but not stereo.
347	if (AudioStatus.sample_size == 8) {
348	uint32 p = (uint32 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
349	uint32 q = (uint32 )last_buffer;
350	int r = work_size >> 2;
351	// XXX not quite right....
352	while (r--)
353	q++ = p++ ^ 0x80808080;
354	if (work_size != sound_buffer_size)
355	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
356	alWriteFrames(port, last_buffer, audio_frames_per_block);
357	}
358	else if (work_size == sound_buffer_size)
359	alWriteFrames(port, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), audio_frames_per_block);
360	else {
361	// Last buffer
362	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
363	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
364	alWriteFrames(port, last_buffer, audio_frames_per_block);
365	}
366	D(bug("stream: data written\n"));
367	} else
368	goto silence;
369
370	} else {
371
372	// Audio not active, play silence
373	silence: // D(bug("stream: silence\n"));
374	alZeroFrames(port, audio_frames_per_block);
375	}
376
377	// Wait for fill point to be reached (may be immediate)
378
379	if (alSetFillPoint(port, sound_buffer_fill_point) < 0) {
380	fprintf(stderr, "ERROR: alSetFillPoint failed: %s\n",
381	alGetErrorString(oserror()));
382	// Should stop the audio here....
383	}
384
385	do {
386	errno = 0;
387	FD_SET(audio_fd, &audio_fdset);
388	was_error = select(numfds, NULL, &audio_fdset, NULL, NULL);
389	} while(was_error < 0 && (errno == EINTR));
390	if (was_error < 0) {
391	fprintf(stderr, "ERROR: select returned %d, errno %d\n",
392	was_error, errno);
393	// Should stop audio here....
394	}
395	}
396	delete[] last_buffer;
397	return NULL;
398	}
399
400
401	/*
402	* Read or set the current output volume using the audio library
403	*/
404
405	static uint32 read_volume(void)
406	{
407	ALpv x[2];
408	ALfixed gain[8];
409	double maxgain, mingain;
410	ALparamInfo pi;
411	uint32 ret = 0x01000100; // default, maximum value
412	int dev = alGetDevice(config);
413
414	// Fetch the maximum and minimum gain settings
415
416	alGetParamInfo(dev, AL_GAIN, &pi);
417	maxgain = alFixedToDouble(pi.max.ll);
418	mingain = alFixedToDouble(pi.min.ll);
419	// printf("maxgain = %lf dB, mingain = %lf dB\n", maxgain, mingain);
420
421	// Get the current gain values
422
423	x[0].param = AL_GAIN;
424	x[0].value.ptr = gain;
425	x[0].sizeIn = sizeof(gain) / sizeof(gain[0]);
426	x[1].param = AL_CHANNELS;
427	if (alGetParams(dev, x, 2) < 0) {
428	printf("alGetParams failed: %s\n", alGetErrorString(oserror()));
429	}
430	else {
431	if (x[0].sizeOut < 0) {
432	printf("AL_GAIN was an unrecognized parameter\n");
433	}
434	else {
435	double v;
436	uint32 left, right;
437
438	// Left
439	v = alFixedToDouble(gain[0]);
440	if (v < mingain)
441	v = mingain; // handle gain == -inf
442	v = (v - mingain) / (maxgain - mingain); // scale to 0..1
443	left = (uint32)(v * (double)256); // convert to 8.8 fixed point
444
445	// Right
446	if (x[0].sizeOut <= 1) { // handle a mono interface
447	right = left;
448	}
449	else {
450	v = alFixedToDouble(gain[1]);
451	if (v < mingain)
452	v = mingain; // handle gain == -inf
453	v = (v - mingain) / (maxgain - mingain); // scale to 0..1
454	right = (uint32)(v * (double)256); // convert to 8.8 fixed point
455	}
456
457	ret = (left << 16) \| right;
458	}
459	}
460
461	return ret;
462	}
463
464	static void set_volume(uint32 vol)
465	{
466	ALpv x[1];
467	ALfixed gain[2]; // left and right
468	double maxgain, mingain;
469	ALparamInfo pi;
470	int dev = alGetDevice(config);
471
472	// Fetch the maximum and minimum gain settings
473
474	alGetParamInfo(dev, AL_GAIN, &pi);
475	maxgain = alFixedToDouble(pi.max.ll);
476	mingain = alFixedToDouble(pi.min.ll);
477
478	// Set the new gain values
479
480	x[0].param = AL_GAIN;
481	x[0].value.ptr = gain;
482	x[0].sizeIn = sizeof(gain) / sizeof(gain[0]);
483
484	uint32 left = vol >> 16;
485	uint32 right = vol & 0xffff;
486	double lv, rv;
487
488	if (left == 0 && pi.specialVals & AL_NEG_INFINITY_BIT) {
489	lv = AL_NEG_INFINITY;
490	}
491	else {
492	lv = ((double)left / 256) * (maxgain - mingain) + mingain;
493	}
494
495	if (right == 0 && pi.specialVals & AL_NEG_INFINITY_BIT) {
496	rv = AL_NEG_INFINITY;
497	}
498	else {
499	rv = ((double)right / 256) * (maxgain - mingain) + mingain;
500	}
501
502	D(bug("set_volume: left=%lf dB, right=%lf dB\n", lv, rv));
503
504	gain[0] = alDoubleToFixed(lv);
505	gain[1] = alDoubleToFixed(rv);
506
507	if (alSetParams(dev, x, 1) < 0) {
508	printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
509	}
510	}
511
512
513	/*
514	* Read or set the mute setting using the audio library
515	*/
516
517	static bool read_mute(void)
518	{
519	bool ret;
520	int dev = alGetDevice(config);
521	ALpv x;
522	x.param = AL_MUTE;
523
524	if (alGetParams(dev, &x, 1) < 0) {
525	printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
526	return current_main_mute; // Or just return false?
527	}
528
529	ret = x.value.i;
530
531	D(bug("read_mute: mute=%d\n", ret));
532	return ret;
533	}
534
535	static void set_mute(bool mute)
536	{
537	D(bug("set_mute: mute=%ld\n", mute));
538
539	int dev = alGetDevice(config);
540	ALpv x;
541	x.param = AL_MUTE;
542	x.value.i = mute;
543
544	if (alSetParams(dev, &x, 1) < 0) {
545	printf("alSetParams failed: %s\n", alGetErrorString(oserror()));
546	}
547	}
548
549
550
551	/*
552	* MacOS audio interrupt, read next data block
553	*/
554
555	void AudioInterrupt(void)
556	{
557	D(bug("AudioInterrupt\n"));
558
559	// Get data from apple mixer
560	if (AudioStatus.mixer) {
561	M68kRegisters r;
562	r.a[0] = audio_data + adatStreamInfo;
563	r.a[1] = AudioStatus.mixer;
564	Execute68k(audio_data + adatGetSourceData, &r);
565	D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
566	} else
567	WriteMacInt32(audio_data + adatStreamInfo, 0);
568
569	// Signal stream function
570	sem_post(&audio_irq_done_sem);
571	D(bug("AudioInterrupt done\n"));
572	}
573
574
575	/*
576	* Set sampling parameters
577	* "index" is an index into the audio_sample_rates[] etc. vectors
578	* It is guaranteed that AudioStatus.num_sources == 0
579	*/
580
581	bool audio_set_sample_rate(int index)
582	{
583	close_audio();
584	audio_sample_rate_index = index;
585	return open_audio();
586	}
587
588	bool audio_set_sample_size(int index)
589	{
590	close_audio();
591	audio_sample_size_index = index;
592	return open_audio();
593	}
594
595	bool audio_set_channels(int index)
596	{
597	close_audio();
598	audio_channel_count_index = index;
599	return open_audio();
600	}
601
602
603	/*
604	* Get/set volume controls (volume values received/returned have the left channel
605	* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
606	* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
607	*/
608
609	bool audio_get_main_mute(void)
610	{
611	D(bug("audio_get_main_mute: mute=%ld\n", current_main_mute));
612
613	return current_main_mute;
614	}
615
616	uint32 audio_get_main_volume(void)
617	{
618	uint32 ret = current_main_volume;
619
620	D(bug("audio_get_main_volume: vol=0x%x\n", ret));
621
622	return ret;
623	}
624
625	bool audio_get_speaker_mute(void)
626	{
627	D(bug("audio_get_speaker_mute: mute=%ld\n", current_speaker_mute));
628
629	return current_speaker_mute;
630	}
631
632	uint32 audio_get_speaker_volume(void)
633	{
634	uint32 ret = current_speaker_volume;
635
636	D(bug("audio_get_speaker_volume: vol=0x%x\n", ret));
637
638	return ret;
639	}
640
641	void audio_set_main_mute(bool mute)
642	{
643	D(bug("audio_set_main_mute: mute=%ld\n", mute));
644
645	if (mute != current_main_mute) {
646	current_main_mute = mute;
647	}
648
649	set_mute(current_main_mute);
650	}
651
652	void audio_set_main_volume(uint32 vol)
653	{
654
655	D(bug("audio_set_main_volume: vol=%x\n", vol));
656
657	current_main_volume = vol;
658
659	set_volume(vol);
660	}
661
662	void audio_set_speaker_mute(bool mute)
663	{
664	D(bug("audio_set_speaker_mute: mute=%ld\n", mute));
665
666	if (mute != current_speaker_mute) {
667	current_speaker_mute = mute;
668	}
669
670	set_mute(current_speaker_mute);
671	}
672
673	void audio_set_speaker_volume(uint32 vol)
674	{
675	D(bug("audio_set_speaker_volume: vol=%x\n", vol));
676
677	current_speaker_volume = vol;
678
679	set_volume(vol);
680	}