Unix/Irix/audio_irix.cpp

/*
 *  audio_irix.cpp - Audio support, SGI Irix implementation
 *
 *  Basilisk II (C) 1997-2000 Christian Bauer
 *
 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation; either version 2 of the License, or
 *  (at your option) any later version.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

#include "sysdeps.h"

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

#include <dmedia/audio.h>
#include <dmedia/dmedia.h>

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

#define DEBUG 0
#include "debug.h"


// Supported sample rates, sizes and channels (defaults)
int audio_num_sample_rates = 1;
uint32 audio_sample_rates[] = {44100 << 16};
int audio_num_sample_sizes = 1;
uint16 audio_sample_sizes[] = {16};
int audio_num_channel_counts = 1;
uint16 audio_channel_counts[] = {2};

// Global variables
static int audio_fd = -1;                                                       // fd from audio library
static sem_t audio_irq_done_sem;                                        // Signal from interrupt to streaming thread: data block read
static bool sem_inited = false;                                         // Flag: audio_irq_done_sem initialized
static int sound_buffer_size;                                           // Size of sound buffer in bytes
static int sound_buffer_fill_point;                                     // Fill buffer when this many frames are empty
static uint8 silence_byte = 0;                                          // Byte value to use to fill sound buffers with silence
static pthread_t stream_thread;                                         // Audio streaming thread
static pthread_attr_t stream_thread_attr;                       // Streaming thread attributes
static bool stream_thread_active = false;                       // Flag: streaming thread installed
static volatile bool stream_thread_cancel = false;      // Flag: cancel streaming thread

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


// Prototypes
static void *stream_func(void *arg);


/*
 *  Initialization
 */

// Set AudioStatus to reflect current audio stream format
static void set_audio_status_format(void)
{
        AudioStatus.sample_rate = audio_sample_rates[0];
        AudioStatus.sample_size = audio_sample_sizes[0];
        AudioStatus.channels = audio_channel_counts[0];
}

// Init libaudio, returns false on error
bool audio_init_al(void)
{
        ALpv     pv[2];

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

        // Try to open the audio library

        config = alNewConfig();
        alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
        alSetWidth(config, AL_SAMPLE_16);
        alSetChannels(config, 2);       // stereo
        alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
        
        port = alOpenPort("BasiliskII", "w", config);
        if (port == NULL) {
                fprintf(stderr, "ERROR: Cannot open audio port: %s\n", 
                                alGetErrorString(oserror()));
                return false;
        }

        // Set the sample rate

        pv[0].param = AL_RATE;
        pv[0].value.ll = alDoubleToFixed(audio_sample_rates[0] >> 16);
        pv[1].param = AL_MASTER_CLOCK;
        pv[1].value.i = AL_CRYSTAL_MCLK_TYPE;
        if (alSetParams(AL_DEFAULT_OUTPUT, pv, 2) < 0) {
                fprintf(stderr, "ERROR: libaudio setparams failed: %s\n",
                                alGetErrorString(oserror()));
                alClosePort(port);
                return false;
        }

        // TODO:  list all supported sample formats?

        // Set AudioStatus again because we now know more about the sound
        // system's capabilities
        set_audio_status_format();

        // Compute sound buffer size and libaudio refill point

        config = alGetConfig(port);
        audio_frames_per_block = alGetQueueSize(config);
        if (audio_frames_per_block < 0) {
                fprintf(stderr, "ERROR: couldn't get queue size: %s\n",
                                alGetErrorString(oserror()));
                alClosePort(port);
                return false;
        }
        D(bug("alGetQueueSize %d\n", audio_frames_per_block));

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

        // Put a limit on the Mac sound buffer size, to decrease delay
        if (audio_frames_per_block > 2048)
                audio_frames_per_block = 2048;
        // Try to keep the buffer pretty full.  5000 samples of slack works well.
        sound_buffer_fill_point = alGetQueueSize(config) - 5000;
        if (sound_buffer_fill_point < 0)
                sound_buffer_fill_point = alGetQueueSize(config) / 3;
        D(bug("fill point %d\n", sound_buffer_fill_point));

        sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;

        // Get a file descriptor we can select() on

        audio_fd = alGetFD(port);
        if (audio_fd < 0) {
                fprintf(stderr, "ERROR: couldn't get libaudio file descriptor: %s\n",
                                alGetErrorString(oserror()));
                alClosePort(port);
                return false;
        }

        return true;
}


/*
 *  Initialization
 */

void AudioInit(void)
{
        // Init audio status (defaults) and feature flags
        set_audio_status_format();
        AudioStatus.mixer = 0;
        AudioStatus.num_sources = 0;
        audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;

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

        // Try to open audio library
        if (!audio_init_al())
                        return;

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

        // Start streaming thread
        pthread_attr_init(&stream_thread_attr);
#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
        if (geteuid() == 0) {
                pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
                pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
                struct sched_param fifo_param;
                fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
                pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
        }
#endif
        stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);

        // Everything OK
        audio_open = true;
}


/*
 *  Deinitialization
 */

void AudioExit(void)
{
        // Stop stream and delete semaphore
        if (stream_thread_active) {
                stream_thread_cancel = true;
#ifdef HAVE_PTHREAD_CANCEL
                pthread_cancel(stream_thread);
#endif
                pthread_join(stream_thread, NULL);
                stream_thread_active = false;
        }
        if (sem_inited)
                sem_destroy(&audio_irq_done_sem);

        // Close audio library
        alClosePort(port);
}


/*
 *  First source added, start audio stream
 */

void audio_enter_stream()
{
        // Streaming thread is always running to avoid clicking noises
}


/*
 *  Last source removed, stop audio stream
 */

void audio_exit_stream()
{
        // Streaming thread is always running to avoid clicking noises
}


/*
 *  Streaming function
 */

static void *stream_func(void *arg)
{
        int16 *last_buffer = new int16[sound_buffer_size / 2];
        fd_set audio_fdset;
        int    numfds, was_error;

        numfds = audio_fd + 1;
        FD_ZERO(&audio_fdset);

        while (!stream_thread_cancel) {
                if (AudioStatus.num_sources) {

                        // Trigger audio interrupt to get new buffer
                        D(bug("stream: triggering irq\n"));
                        SetInterruptFlag(INTFLAG_AUDIO);
                        TriggerInterrupt();
                        D(bug("stream: waiting for ack\n"));
                        sem_wait(&audio_irq_done_sem);
                        D(bug("stream: ack received\n"));

                        uint32 apple_stream_info;       // Mac address of SoundComponentData struct describing next buffer
                        // Get size of audio data
                        apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);

                        if (apple_stream_info) {
                                int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
                                D(bug("stream: work_size %d\n", work_size));
                                if (work_size > sound_buffer_size)
                                        work_size = sound_buffer_size;
                                if (work_size == 0)
                                        goto silence;

                                // Send data to audio library
                                if (work_size == sound_buffer_size)
                                        alWriteFrames(port, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), audio_frames_per_block);
                                else {
                                        // Last buffer
                                        Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
                                        memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
                                        alWriteFrames(port, last_buffer, audio_frames_per_block);
                                }
                                D(bug("stream: data written\n"));
                        } else
                                goto silence;

                } else {

                        // Audio not active, play silence
                silence:        // D(bug("stream: silence\n"));
                        alZeroFrames(port, audio_frames_per_block);
                }

                // Wait for fill point to be reached (may be immediate)

                if (alSetFillPoint(port, sound_buffer_fill_point) < 0) {
                        fprintf(stderr, "ERROR: alSetFillPoint failed: %s\n",
                                        alGetErrorString(oserror()));
                        // Should stop the audio here....
                }

                do {
                        errno = 0;
                        FD_SET(audio_fd, &audio_fdset);
                        was_error = select(numfds, NULL, &audio_fdset, NULL, NULL);
                } while(was_error < 0 && (errno == EINTR));
                if (was_error < 0) {
                        fprintf(stderr, "ERROR: select returned %d, errno %d\n",
                                        was_error, errno);
                        // Should stop audio here....
                }
        }
        delete[] last_buffer;
        return NULL;
}


/*
 *  MacOS audio interrupt, read next data block
 */

void AudioInterrupt(void)
{
        D(bug("AudioInterrupt\n"));

        // Get data from apple mixer
        if (AudioStatus.mixer) {
                M68kRegisters r;
                r.a[0] = audio_data + adatStreamInfo;
                r.a[1] = AudioStatus.mixer;
                Execute68k(audio_data + adatGetSourceData, &r);
                D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
        } else
                WriteMacInt32(audio_data + adatStreamInfo, 0);

        // Signal stream function
        sem_post(&audio_irq_done_sem);
        D(bug("AudioInterrupt done\n"));
}


/*
 *  Set sampling parameters
 *  "index" is an index into the audio_sample_rates[] etc. arrays
 *  It is guaranteed that AudioStatus.num_sources == 0
 */

void audio_set_sample_rate(int index)
{
}

void audio_set_sample_size(int index)
{
}

void audio_set_channels(int index)
{
}


/*
 *  Get/set volume controls (volume values received/returned have the left channel
 *  volume in the upper 16 bits and the right channel volume in the lower 16 bits;
 *  both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
 */

bool audio_get_main_mute(void)
{
        return false;
}

uint32 audio_get_main_volume(void)
{
        return 0x01000100;
}

bool audio_get_speaker_mute(void)
{
        return false;
}

uint32 audio_get_speaker_volume(void)
{
        return 0x01000100;
}

void audio_set_main_mute(bool mute)
{
}

void audio_set_main_volume(uint32 vol)
{
}

void audio_set_speaker_mute(bool mute)
{
}

void audio_set_speaker_volume(uint32 vol)
{
}
Revision:	1.1
Committed:	2000-11-02T14:45:17Z (24 years ago) by cebix
Branch:	MAIN
Log Message:	- added audio support for IRIX [Brian J. Johnson] - improved Delay_usec() under FreeBSD and IRIX - fixed typo ("HAVE_PTHREDS") in video_x.cpp
#	Content
1	/*
2	* audio_irix.cpp - Audio support, SGI Irix implementation
3	*
4	* Basilisk II (C) 1997-2000 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	// Supported sample rates, sizes and channels (defaults)
44	int audio_num_sample_rates = 1;
45	uint32 audio_sample_rates[] = {44100 << 16};
46	int audio_num_sample_sizes = 1;
47	uint16 audio_sample_sizes[] = {16};
48	int audio_num_channel_counts = 1;
49	uint16 audio_channel_counts[] = {2};
50
51	// Global variables
52	static int audio_fd = -1; // fd from audio library
53	static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
54	static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
55	static int sound_buffer_size; // Size of sound buffer in bytes
56	static int sound_buffer_fill_point; // Fill buffer when this many frames are empty
57	static uint8 silence_byte = 0; // Byte value to use to fill sound buffers with silence
58	static pthread_t stream_thread; // Audio streaming thread
59	static pthread_attr_t stream_thread_attr; // Streaming thread attributes
60	static bool stream_thread_active = false; // Flag: streaming thread installed
61	static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
62
63	// IRIX libaudio control structures
64	static ALconfig config;
65	static ALport port;
66
67
68	// Prototypes
69	static void stream_func(void arg);
70
71
72	/*
73	* Initialization
74	*/
75
76	// Set AudioStatus to reflect current audio stream format
77	static void set_audio_status_format(void)
78	{
79	AudioStatus.sample_rate = audio_sample_rates[0];
80	AudioStatus.sample_size = audio_sample_sizes[0];
81	AudioStatus.channels = audio_channel_counts[0];
82	}
83
84	// Init libaudio, returns false on error
85	bool audio_init_al(void)
86	{
87	ALpv pv[2];
88
89	printf("Using libaudio audio output\n");
90
91	// Try to open the audio library
92
93	config = alNewConfig();
94	alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
95	alSetWidth(config, AL_SAMPLE_16);
96	alSetChannels(config, 2); // stereo
97	alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
98
99	port = alOpenPort("BasiliskII", "w", config);
100	if (port == NULL) {
101	fprintf(stderr, "ERROR: Cannot open audio port: %s\n",
102	alGetErrorString(oserror()));
103	return false;
104	}
105
106	// Set the sample rate
107
108	pv[0].param = AL_RATE;
109	pv[0].value.ll = alDoubleToFixed(audio_sample_rates[0] >> 16);
110	pv[1].param = AL_MASTER_CLOCK;
111	pv[1].value.i = AL_CRYSTAL_MCLK_TYPE;
112	if (alSetParams(AL_DEFAULT_OUTPUT, pv, 2) < 0) {
113	fprintf(stderr, "ERROR: libaudio setparams failed: %s\n",
114	alGetErrorString(oserror()));
115	alClosePort(port);
116	return false;
117	}
118
119	// TODO: list all supported sample formats?
120
121	// Set AudioStatus again because we now know more about the sound
122	// system's capabilities
123	set_audio_status_format();
124
125	// Compute sound buffer size and libaudio refill point
126
127	config = alGetConfig(port);
128	audio_frames_per_block = alGetQueueSize(config);
129	if (audio_frames_per_block < 0) {
130	fprintf(stderr, "ERROR: couldn't get queue size: %s\n",
131	alGetErrorString(oserror()));
132	alClosePort(port);
133	return false;
134	}
135	D(bug("alGetQueueSize %d\n", audio_frames_per_block));
136
137	alZeroFrames(port, audio_frames_per_block); // so we don't underflow
138
139	// Put a limit on the Mac sound buffer size, to decrease delay
140	if (audio_frames_per_block > 2048)
141	audio_frames_per_block = 2048;
142	// Try to keep the buffer pretty full. 5000 samples of slack works well.
143	sound_buffer_fill_point = alGetQueueSize(config) - 5000;
144	if (sound_buffer_fill_point < 0)
145	sound_buffer_fill_point = alGetQueueSize(config) / 3;
146	D(bug("fill point %d\n", sound_buffer_fill_point));
147
148	sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
149
150	// Get a file descriptor we can select() on
151
152	audio_fd = alGetFD(port);
153	if (audio_fd < 0) {
154	fprintf(stderr, "ERROR: couldn't get libaudio file descriptor: %s\n",
155	alGetErrorString(oserror()));
156	alClosePort(port);
157	return false;
158	}
159
160	return true;
161	}
162
163
164	/*
165	* Initialization
166	*/
167
168	void AudioInit(void)
169	{
170	// Init audio status (defaults) and feature flags
171	set_audio_status_format();
172	AudioStatus.mixer = 0;
173	AudioStatus.num_sources = 0;
174	audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
175
176	// Sound disabled in prefs? Then do nothing
177	if (PrefsFindBool("nosound"))
178	return;
179
180	// Try to open audio library
181	if (!audio_init_al())
182	return;
183
184	// Init semaphore
185	if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
186	return;
187	sem_inited = true;
188
189	// Start streaming thread
190	pthread_attr_init(&stream_thread_attr);
191	#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
192	if (geteuid() == 0) {
193	pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
194	pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
195	struct sched_param fifo_param;
196	fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
197	pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
198	}
199	#endif
200	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
201
202	// Everything OK
203	audio_open = true;
204	}
205
206
207	/*
208	* Deinitialization
209	*/
210
211	void AudioExit(void)
212	{
213	// Stop stream and delete semaphore
214	if (stream_thread_active) {
215	stream_thread_cancel = true;
216	#ifdef HAVE_PTHREAD_CANCEL
217	pthread_cancel(stream_thread);
218	#endif
219	pthread_join(stream_thread, NULL);
220	stream_thread_active = false;
221	}
222	if (sem_inited)
223	sem_destroy(&audio_irq_done_sem);
224
225	// Close audio library
226	alClosePort(port);
227	}
228
229
230	/*
231	* First source added, start audio stream
232	*/
233
234	void audio_enter_stream()
235	{
236	// Streaming thread is always running to avoid clicking noises
237	}
238
239
240	/*
241	* Last source removed, stop audio stream
242	*/
243
244	void audio_exit_stream()
245	{
246	// Streaming thread is always running to avoid clicking noises
247	}
248
249
250	/*
251	* Streaming function
252	*/
253
254	static void stream_func(void arg)
255	{
256	int16 *last_buffer = new int16[sound_buffer_size / 2];
257	fd_set audio_fdset;
258	int numfds, was_error;
259
260	numfds = audio_fd + 1;
261	FD_ZERO(&audio_fdset);
262
263	while (!stream_thread_cancel) {
264	if (AudioStatus.num_sources) {
265
266	// Trigger audio interrupt to get new buffer
267	D(bug("stream: triggering irq\n"));
268	SetInterruptFlag(INTFLAG_AUDIO);
269	TriggerInterrupt();
270	D(bug("stream: waiting for ack\n"));
271	sem_wait(&audio_irq_done_sem);
272	D(bug("stream: ack received\n"));
273
274	uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
275	// Get size of audio data
276	apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
277
278	if (apple_stream_info) {
279	int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
280	D(bug("stream: work_size %d\n", work_size));
281	if (work_size > sound_buffer_size)
282	work_size = sound_buffer_size;
283	if (work_size == 0)
284	goto silence;
285
286	// Send data to audio library
287	if (work_size == sound_buffer_size)
288	alWriteFrames(port, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), audio_frames_per_block);
289	else {
290	// Last buffer
291	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
292	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
293	alWriteFrames(port, last_buffer, audio_frames_per_block);
294	}
295	D(bug("stream: data written\n"));
296	} else
297	goto silence;
298
299	} else {
300
301	// Audio not active, play silence
302	silence: // D(bug("stream: silence\n"));
303	alZeroFrames(port, audio_frames_per_block);
304	}
305
306	// Wait for fill point to be reached (may be immediate)
307
308	if (alSetFillPoint(port, sound_buffer_fill_point) < 0) {
309	fprintf(stderr, "ERROR: alSetFillPoint failed: %s\n",
310	alGetErrorString(oserror()));
311	// Should stop the audio here....
312	}
313
314	do {
315	errno = 0;
316	FD_SET(audio_fd, &audio_fdset);
317	was_error = select(numfds, NULL, &audio_fdset, NULL, NULL);
318	} while(was_error < 0 && (errno == EINTR));
319	if (was_error < 0) {
320	fprintf(stderr, "ERROR: select returned %d, errno %d\n",
321	was_error, errno);
322	// Should stop audio here....
323	}
324	}
325	delete[] last_buffer;
326	return NULL;
327	}
328
329
330	/*
331	* MacOS audio interrupt, read next data block
332	*/
333
334	void AudioInterrupt(void)
335	{
336	D(bug("AudioInterrupt\n"));
337
338	// Get data from apple mixer
339	if (AudioStatus.mixer) {
340	M68kRegisters r;
341	r.a[0] = audio_data + adatStreamInfo;
342	r.a[1] = AudioStatus.mixer;
343	Execute68k(audio_data + adatGetSourceData, &r);
344	D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
345	} else
346	WriteMacInt32(audio_data + adatStreamInfo, 0);
347
348	// Signal stream function
349	sem_post(&audio_irq_done_sem);
350	D(bug("AudioInterrupt done\n"));
351	}
352
353
354	/*
355	* Set sampling parameters
356	* "index" is an index into the audio_sample_rates[] etc. arrays
357	* It is guaranteed that AudioStatus.num_sources == 0
358	*/
359
360	void audio_set_sample_rate(int index)
361	{
362	}
363
364	void audio_set_sample_size(int index)
365	{
366	}
367
368	void audio_set_channels(int index)
369	{
370	}
371
372
373	/*
374	* Get/set volume controls (volume values received/returned have the left channel
375	* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
376	* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
377	*/
378
379	bool audio_get_main_mute(void)
380	{
381	return false;
382	}
383
384	uint32 audio_get_main_volume(void)
385	{
386	return 0x01000100;
387	}
388
389	bool audio_get_speaker_mute(void)
390	{
391	return false;
392	}
393
394	uint32 audio_get_speaker_volume(void)
395	{
396	return 0x01000100;
397	}
398
399	void audio_set_main_mute(bool mute)
400	{
401	}
402
403	void audio_set_main_volume(uint32 vol)
404	{
405	}
406
407	void audio_set_speaker_mute(bool mute)
408	{
409	}
410
411	void audio_set_speaker_volume(uint32 vol)
412	{
413	}