src/Unix/audio_oss_esd.cpp

/*
 *  audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
 *
 *  Basilisk II (C) 1997-1999 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>

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

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

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

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

#define DEBUG 0
#include "debug.h"


// Supported sample rates, sizes and channels
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};

// Constants
#define DSP_NAME "/dev/dsp"

// Global variables
static int audio_fd = -1;                                                       // fd of /dev/dsp or ESD
static int mixer_fd = -1;                                                       // fd of /dev/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 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
 */

// Init using /dev/dsp, returns false on error
bool audio_init_dsp(void)
{
        printf("Using " DSP_NAME " audio output\n");

        // Get supported sample formats
        unsigned long format;
        ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
        if ((format & (AFMT_U8 | AFMT_S16_BE | AFMT_S16_LE)) == 0) {
                WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
                close(audio_fd);
                audio_fd = -1;
                return false;
        }
        if (format & (AFMT_S16_BE | AFMT_S16_LE))
                audio_sample_sizes[0] = 16;
        else
                audio_sample_sizes[0] = 8;
        if (!(format & AFMT_S16_BE))
                little_endian = true;

        // Set DSP parameters
        format = AudioStatus.sample_size == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
        ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
        int frag = 0x0004000c;          // Block size: 4096 frames
        ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
        int stereo = (AudioStatus.channels == 2);
        ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
        int rate = AudioStatus.sample_rate >> 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));
        sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
        return true;
}

// Init using ESD, returns false on error
bool audio_init_esd(void)
{
#if ENABLE_ESD
        printf("Using ESD audio output\n");

        // ESD audio format
        esd_format_t format = ESD_STREAM | ESD_PLAY;
        if (AudioStatus.sample_size == 8)
                format |= ESD_BITS8;
        else
                format |= ESD_BITS16;
        if (AudioStatus.channels == 1)
                format |= ESD_MONO;
        else
                format |= ESD_STEREO;

#if WORDS_BIGENDIAN
        little_endian = false;
#else
        little_endian = true;
#endif

        // Open connection to ESD server
        audio_fd = esd_play_stream(format, AudioStatus.sample_rate >> 16, NULL, NULL);
        if (audio_fd < 0) {
                char str[256];
                sprintf(str, GetString(STR_NO_ESD_WARN), strerror(errno));
                WarningAlert(str);
                return false;
        }

        // Sound buffer size = 4096 frames
        audio_frames_per_block = 4096;
        sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
        return true;
#else
        ErrorAlert("Basilisk II has been compiled with ESD support disabled.");
        return false;
#endif
}

void AudioInit(void)
{
        char str[256];

        // Init audio status (defaults) and feature flags
        AudioStatus.sample_rate = audio_sample_rates[0];
        AudioStatus.sample_size = audio_sample_sizes[0];
        AudioStatus.channels = audio_channel_counts[0];
        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 /dev/dsp
        audio_fd = open(DSP_NAME, O_WRONLY);
        if (audio_fd < 0) {
#if ENABLE_ESD
                if (!audio_init_esd())
                        return;
#else
                sprintf(str, GetString(STR_NO_AUDIO_DEV_WARN), DSP_NAME, strerror(errno));
                WarningAlert(str);
                return;
#endif
        } else
                if (!audio_init_dsp())
                        return;

        // Try to open /dev/mixer
        mixer_fd = open("/dev/mixer", O_RDWR);
        if (mixer_fd < 0)
                printf("WARNING: Cannot open /dev/mixer (%s)", strerror(errno));

        // 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 /dev/dsp
        if (audio_fd > 0)
                close(audio_fd);

        // Close /dev/mixer
        if (mixer_fd > 0)
                close(mixer_fd);
}


/*
 *  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 uint32 apple_stream_info;        // Mac address of SoundComponentData struct describing next buffer

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, 0, 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[i] = ntohs(p[i]);
                                        } else
                                                memcpy(last_buffer, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size);
                                        memset((uint8 *)last_buffer + work_size, 0, 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. 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)
{
        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);
        }
}
Revision:	1.1
Committed:	1999-10-23T17:57:45Z (24 years, 8 months ago) by cebix
Branch:	MAIN
Log Message:	- audio_linux.cpp renamed to audio_oss_esd.cpp (now also used under FreeBSD) and added support for ESD - medium removal is allowed for CD-ROM on exit - added mkinstalldirs to "make install" target
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
3			*
4			* Basilisk II (C) 1997-1999 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			#ifdef __linux__
30			#include <linux/soundcard.h>
31			#endif
32
33			#ifdef __FreeBSD__
34			#include <machine/soundcard.h>
35			#endif
36
37			#include "cpu_emulation.h"
38			#include "main.h"
39			#include "prefs.h"
40			#include "user_strings.h"
41			#include "audio.h"
42			#include "audio_defs.h"
43
44			#if ENABLE_ESD
45			#include <esd.h>
46			#endif
47
48			#define DEBUG 0
49			#include "debug.h"
50
51
52			// Supported sample rates, sizes and channels
53			int audio_num_sample_rates = 1;
54			uint32 audio_sample_rates[] = {44100 << 16};
55			int audio_num_sample_sizes = 1;
56			uint16 audio_sample_sizes[] = {16};
57			int audio_num_channel_counts = 1;
58			uint16 audio_channel_counts[] = {2};
59
60			// Constants
61			#define DSP_NAME "/dev/dsp"
62
63			// Global variables
64			static int audio_fd = -1; // fd of /dev/dsp or ESD
65			static int mixer_fd = -1; // fd of /dev/mixer
66			static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
67			static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
68			static int sound_buffer_size; // Size of sound buffer in bytes
69			static bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
70			static pthread_t stream_thread; // Audio streaming thread
71			static pthread_attr_t stream_thread_attr; // Streaming thread attributes
72			static bool stream_thread_active = false; // Flag: streaming thread installed
73			static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
74
75			// Prototypes
76			static void stream_func(void arg);
77
78
79			/*
80			* Initialization
81			*/
82
83			// Init using /dev/dsp, returns false on error
84			bool audio_init_dsp(void)
85			{
86			printf("Using " DSP_NAME " audio output\n");
87
88			// Get supported sample formats
89			unsigned long format;
90			ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
91			if ((format & (AFMT_U8 \| AFMT_S16_BE \| AFMT_S16_LE)) == 0) {
92			WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
93			close(audio_fd);
94			audio_fd = -1;
95			return false;
96			}
97			if (format & (AFMT_S16_BE \| AFMT_S16_LE))
98			audio_sample_sizes[0] = 16;
99			else
100			audio_sample_sizes[0] = 8;
101			if (!(format & AFMT_S16_BE))
102			little_endian = true;
103
104			// Set DSP parameters
105			format = AudioStatus.sample_size == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
106			ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
107			int frag = 0x0004000c; // Block size: 4096 frames
108			ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
109			int stereo = (AudioStatus.channels == 2);
110			ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
111			int rate = AudioStatus.sample_rate >> 16;
112			ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
113
114			// Get sound buffer size
115			ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
116			D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
117			sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
118			return true;
119			}
120
121			// Init using ESD, returns false on error
122			bool audio_init_esd(void)
123			{
124			#if ENABLE_ESD
125			printf("Using ESD audio output\n");
126
127			// ESD audio format
128			esd_format_t format = ESD_STREAM \| ESD_PLAY;
129			if (AudioStatus.sample_size == 8)
130			format \|= ESD_BITS8;
131			else
132			format \|= ESD_BITS16;
133			if (AudioStatus.channels == 1)
134			format \|= ESD_MONO;
135			else
136			format \|= ESD_STEREO;
137
138			#if WORDS_BIGENDIAN
139			little_endian = false;
140			#else
141			little_endian = true;
142			#endif
143
144			// Open connection to ESD server
145			audio_fd = esd_play_stream(format, AudioStatus.sample_rate >> 16, NULL, NULL);
146			if (audio_fd < 0) {
147			char str[256];
148			sprintf(str, GetString(STR_NO_ESD_WARN), strerror(errno));
149			WarningAlert(str);
150			return false;
151			}
152
153			// Sound buffer size = 4096 frames
154			audio_frames_per_block = 4096;
155			sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
156			return true;
157			#else
158			ErrorAlert("Basilisk II has been compiled with ESD support disabled.");
159			return false;
160			#endif
161			}
162
163			void AudioInit(void)
164			{
165			char str[256];
166
167			// Init audio status (defaults) and feature flags
168			AudioStatus.sample_rate = audio_sample_rates[0];
169			AudioStatus.sample_size = audio_sample_sizes[0];
170			AudioStatus.channels = audio_channel_counts[0];
171			AudioStatus.mixer = 0;
172			AudioStatus.num_sources = 0;
173			audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
174
175			// Sound disabled in prefs? Then do nothing
176			if (PrefsFindBool("nosound"))
177			return;
178
179			// Try to open /dev/dsp
180			audio_fd = open(DSP_NAME, O_WRONLY);
181			if (audio_fd < 0) {
182			#if ENABLE_ESD
183			if (!audio_init_esd())
184			return;
185			#else
186			sprintf(str, GetString(STR_NO_AUDIO_DEV_WARN), DSP_NAME, strerror(errno));
187			WarningAlert(str);
188			return;
189			#endif
190			} else
191			if (!audio_init_dsp())
192			return;
193
194			// Try to open /dev/mixer
195			mixer_fd = open("/dev/mixer", O_RDWR);
196			if (mixer_fd < 0)
197			printf("WARNING: Cannot open /dev/mixer (%s)", strerror(errno));
198
199			// Init semaphore
200			if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
201			return;
202			sem_inited = true;
203
204			// Start streaming thread
205			pthread_attr_init(&stream_thread_attr);
206			#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
207			if (geteuid() == 0) {
208			pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
209			pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
210			struct sched_param fifo_param;
211			fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
212			pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
213			}
214			#endif
215			stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
216
217			// Everything OK
218			audio_open = true;
219			}
220
221
222			/*
223			* Deinitialization
224			*/
225
226			void AudioExit(void)
227			{
228			// Stop stream and delete semaphore
229			if (stream_thread_active) {
230			stream_thread_cancel = true;
231			#ifdef HAVE_PTHREAD_CANCEL
232			pthread_cancel(stream_thread);
233			#endif
234			pthread_join(stream_thread, NULL);
235			stream_thread_active = false;
236			}
237			if (sem_inited)
238			sem_destroy(&audio_irq_done_sem);
239
240			// Close /dev/dsp
241			if (audio_fd > 0)
242			close(audio_fd);
243
244			// Close /dev/mixer
245			if (mixer_fd > 0)
246			close(mixer_fd);
247			}
248
249
250			/*
251			* First source added, start audio stream
252			*/
253
254			void audio_enter_stream()
255			{
256			// Streaming thread is always running to avoid clicking noises
257			}
258
259
260			/*
261			* Last source removed, stop audio stream
262			*/
263
264			void audio_exit_stream()
265			{
266			// Streaming thread is always running to avoid clicking noises
267			}
268
269
270			/*
271			* Streaming function
272			*/
273
274			static uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
275
276			static void stream_func(void arg)
277			{
278			int16 *silent_buffer = new int16[sound_buffer_size / 2];
279			int16 *last_buffer = new int16[sound_buffer_size / 2];
280			memset(silent_buffer, 0, sound_buffer_size);
281
282			while (!stream_thread_cancel) {
283			if (AudioStatus.num_sources) {
284
285			// Trigger audio interrupt to get new buffer
286			D(bug("stream: triggering irq\n"));
287			SetInterruptFlag(INTFLAG_AUDIO);
288			TriggerInterrupt();
289			D(bug("stream: waiting for ack\n"));
290			sem_wait(&audio_irq_done_sem);
291			D(bug("stream: ack received\n"));
292
293			// Get size of audio data
294			uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
295			if (apple_stream_info) {
296			int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
297			D(bug("stream: work_size %d\n", work_size));
298			if (work_size > sound_buffer_size)
299			work_size = sound_buffer_size;
300			if (work_size == 0)
301			goto silence;
302
303			// Send data to DSP
304			if (work_size == sound_buffer_size && !little_endian)
305			write(audio_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
306			else {
307			// Last buffer or little-endian DSP
308			if (little_endian) {
309			int16 p = (int16 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
310			for (int i=0; i<work_size/2; i++)
311			last_buffer[i] = ntohs(p[i]);
312			} else
313			memcpy(last_buffer, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size);
314			memset((uint8 *)last_buffer + work_size, 0, sound_buffer_size - work_size);
315			write(audio_fd, last_buffer, sound_buffer_size);
316			}
317			D(bug("stream: data written\n"));
318			} else
319			goto silence;
320
321			} else {
322
323			// Audio not active, play silence
324			silence: write(audio_fd, silent_buffer, sound_buffer_size);
325			}
326			}
327			delete[] silent_buffer;
328			delete[] last_buffer;
329			return NULL;
330			}
331
332
333			/*
334			* MacOS audio interrupt, read next data block
335			*/
336
337			void AudioInterrupt(void)
338			{
339			D(bug("AudioInterrupt\n"));
340
341			// Get data from apple mixer
342			if (AudioStatus.mixer) {
343			M68kRegisters r;
344			r.a[0] = audio_data + adatStreamInfo;
345			r.a[1] = AudioStatus.mixer;
346			Execute68k(audio_data + adatGetSourceData, &r);
347			D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
348			} else
349			WriteMacInt32(audio_data + adatStreamInfo, 0);
350
351			// Signal stream function
352			sem_post(&audio_irq_done_sem);
353			D(bug("AudioInterrupt done\n"));
354			}
355
356
357			/*
358			* Set sampling parameters
359			* "index" is an index into the audio_sample_rates[] etc. arrays
360			* It is guaranteed that AudioStatus.num_sources == 0
361			*/
362
363			void audio_set_sample_rate(int index)
364			{
365			}
366
367			void audio_set_sample_size(int index)
368			{
369			}
370
371			void audio_set_channels(int index)
372			{
373			}
374
375
376			/*
377			* Get/set volume controls (volume values received/returned have the left channel
378			* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
379			* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
380			*/
381
382			bool audio_get_main_mute(void)
383			{
384			return false;
385			}
386
387			uint32 audio_get_main_volume(void)
388			{
389			if (mixer_fd >= 0) {
390			int vol;
391			if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
392			int left = vol >> 8;
393			int right = vol & 0xff;
394			return ((left * 256 / 100) << 16) \| (right * 256 / 100);
395			}
396			}
397			return 0x01000100;
398			}
399
400			bool audio_get_speaker_mute(void)
401			{
402			return false;
403			}
404
405			uint32 audio_get_speaker_volume(void)
406			{
407			if (mixer_fd >= 0) {
408			int vol;
409			if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
410			int left = vol >> 8;
411			int right = vol & 0xff;
412			return ((left * 256 / 100) << 16) \| (right * 256 / 100);
413			}
414			}
415			return 0x01000100;
416			}
417
418			void audio_set_main_mute(bool mute)
419			{
420			}
421
422			void audio_set_main_volume(uint32 vol)
423			{
424			if (mixer_fd >= 0) {
425			int left = vol >> 16;
426			int right = vol & 0xffff;
427			int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
428			ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
429			}
430			}
431
432			void audio_set_speaker_mute(bool mute)
433			{
434			}
435
436			void audio_set_speaker_volume(uint32 vol)
437			{
438			if (mixer_fd >= 0) {
439			int left = vol >> 16;
440			int right = vol & 0xffff;
441			int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
442			ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
443			}
444			}