Unix/Irix/audio_irix.cpp

/*
 *  audio_irix.cpp - Audio support, SGI Irix implementation
 *
 *  Basilisk II (C) 1997-2002 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"


// 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
        audio_sample_rates.push_back(44100 << 16);
        audio_sample_sizes.push_back(16);
        audio_channel_counts.push_back(2);
        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
 */

bool audio_set_sample_rate(int index)
{
        return true;
}

bool audio_set_sample_size(int index)
{
        return true;
}

bool audio_set_channels(int index)
{
        return true;
}


/*
 *  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:	1.5
Committed:	2002-01-15T14:58:38Z (22 years, 10 months ago) by cebix
Branch:	MAIN
CVS Tags:	snapshot-15012002
Changes since 1.4:	+1 -1 lines
Log Message:	- documentation updates - 2001 -> 2002 - version 0.9 -> 1.0
#	Content
1	/*
2	* audio_irix.cpp - Audio support, SGI Irix implementation
3	*
4	* Basilisk II (C) 1997-2002 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	// Global variables
44	static int audio_fd = -1; // fd from audio library
45	static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
46	static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
47	static int sound_buffer_size; // Size of sound buffer in bytes
48	static int sound_buffer_fill_point; // Fill buffer when this many frames are empty
49	static uint8 silence_byte = 0; // Byte value to use to fill sound buffers with silence
50	static pthread_t stream_thread; // Audio streaming thread
51	static pthread_attr_t stream_thread_attr; // Streaming thread attributes
52	static bool stream_thread_active = false; // Flag: streaming thread installed
53	static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
54
55	// IRIX libaudio control structures
56	static ALconfig config;
57	static ALport port;
58
59
60	// Prototypes
61	static void stream_func(void arg);
62
63
64	/*
65	* Initialization
66	*/
67
68	// Set AudioStatus to reflect current audio stream format
69	static void set_audio_status_format(void)
70	{
71	AudioStatus.sample_rate = audio_sample_rates[0];
72	AudioStatus.sample_size = audio_sample_sizes[0];
73	AudioStatus.channels = audio_channel_counts[0];
74	}
75
76	// Init libaudio, returns false on error
77	bool audio_init_al(void)
78	{
79	ALpv pv[2];
80
81	printf("Using libaudio audio output\n");
82
83	// Try to open the audio library
84
85	config = alNewConfig();
86	alSetSampFmt(config, AL_SAMPFMT_TWOSCOMP);
87	alSetWidth(config, AL_SAMPLE_16);
88	alSetChannels(config, 2); // stereo
89	alSetDevice(config, AL_DEFAULT_OUTPUT); // Allow selecting via prefs?
90
91	port = alOpenPort("BasiliskII", "w", config);
92	if (port == NULL) {
93	fprintf(stderr, "ERROR: Cannot open audio port: %s\n",
94	alGetErrorString(oserror()));
95	return false;
96	}
97
98	// Set the sample rate
99
100	pv[0].param = AL_RATE;
101	pv[0].value.ll = alDoubleToFixed(audio_sample_rates[0] >> 16);
102	pv[1].param = AL_MASTER_CLOCK;
103	pv[1].value.i = AL_CRYSTAL_MCLK_TYPE;
104	if (alSetParams(AL_DEFAULT_OUTPUT, pv, 2) < 0) {
105	fprintf(stderr, "ERROR: libaudio setparams failed: %s\n",
106	alGetErrorString(oserror()));
107	alClosePort(port);
108	return false;
109	}
110
111	// TODO: list all supported sample formats?
112
113	// Set AudioStatus again because we now know more about the sound
114	// system's capabilities
115	set_audio_status_format();
116
117	// Compute sound buffer size and libaudio refill point
118
119	config = alGetConfig(port);
120	audio_frames_per_block = alGetQueueSize(config);
121	if (audio_frames_per_block < 0) {
122	fprintf(stderr, "ERROR: couldn't get queue size: %s\n",
123	alGetErrorString(oserror()));
124	alClosePort(port);
125	return false;
126	}
127	D(bug("alGetQueueSize %d\n", audio_frames_per_block));
128
129	alZeroFrames(port, audio_frames_per_block); // so we don't underflow
130
131	// Put a limit on the Mac sound buffer size, to decrease delay
132	if (audio_frames_per_block > 2048)
133	audio_frames_per_block = 2048;
134	// Try to keep the buffer pretty full. 5000 samples of slack works well.
135	sound_buffer_fill_point = alGetQueueSize(config) - 5000;
136	if (sound_buffer_fill_point < 0)
137	sound_buffer_fill_point = alGetQueueSize(config) / 3;
138	D(bug("fill point %d\n", sound_buffer_fill_point));
139
140	sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
141
142	// Get a file descriptor we can select() on
143
144	audio_fd = alGetFD(port);
145	if (audio_fd < 0) {
146	fprintf(stderr, "ERROR: couldn't get libaudio file descriptor: %s\n",
147	alGetErrorString(oserror()));
148	alClosePort(port);
149	return false;
150	}
151
152	return true;
153	}
154
155
156	/*
157	* Initialization
158	*/
159
160	void AudioInit(void)
161	{
162	// Init audio status (defaults) and feature flags
163	audio_sample_rates.push_back(44100 << 16);
164	audio_sample_sizes.push_back(16);
165	audio_channel_counts.push_back(2);
166	set_audio_status_format();
167	AudioStatus.mixer = 0;
168	AudioStatus.num_sources = 0;
169	audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
170
171	// Sound disabled in prefs? Then do nothing
172	if (PrefsFindBool("nosound"))
173	return;
174
175	// Try to open audio library
176	if (!audio_init_al())
177	return;
178
179	// Init semaphore
180	if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
181	return;
182	sem_inited = true;
183
184	// Start streaming thread
185	pthread_attr_init(&stream_thread_attr);
186	#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
187	if (geteuid() == 0) {
188	pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
189	pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
190	struct sched_param fifo_param;
191	fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
192	pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
193	}
194	#endif
195	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
196
197	// Everything OK
198	audio_open = true;
199	}
200
201
202	/*
203	* Deinitialization
204	*/
205
206	void AudioExit(void)
207	{
208	// Stop stream and delete semaphore
209	if (stream_thread_active) {
210	stream_thread_cancel = true;
211	#ifdef HAVE_PTHREAD_CANCEL
212	pthread_cancel(stream_thread);
213	#endif
214	pthread_join(stream_thread, NULL);
215	stream_thread_active = false;
216	}
217	if (sem_inited)
218	sem_destroy(&audio_irq_done_sem);
219
220	// Close audio library
221	alClosePort(port);
222	}
223
224
225	/*
226	* First source added, start audio stream
227	*/
228
229	void audio_enter_stream()
230	{
231	// Streaming thread is always running to avoid clicking noises
232	}
233
234
235	/*
236	* Last source removed, stop audio stream
237	*/
238
239	void audio_exit_stream()
240	{
241	// Streaming thread is always running to avoid clicking noises
242	}
243
244
245	/*
246	* Streaming function
247	*/
248
249	static void stream_func(void arg)
250	{
251	int16 *last_buffer = new int16[sound_buffer_size / 2];
252	fd_set audio_fdset;
253	int numfds, was_error;
254
255	numfds = audio_fd + 1;
256	FD_ZERO(&audio_fdset);
257
258	while (!stream_thread_cancel) {
259	if (AudioStatus.num_sources) {
260
261	// Trigger audio interrupt to get new buffer
262	D(bug("stream: triggering irq\n"));
263	SetInterruptFlag(INTFLAG_AUDIO);
264	TriggerInterrupt();
265	D(bug("stream: waiting for ack\n"));
266	sem_wait(&audio_irq_done_sem);
267	D(bug("stream: ack received\n"));
268
269	uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
270	// Get size of audio data
271	apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
272
273	if (apple_stream_info) {
274	int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
275	D(bug("stream: work_size %d\n", work_size));
276	if (work_size > sound_buffer_size)
277	work_size = sound_buffer_size;
278	if (work_size == 0)
279	goto silence;
280
281	// Send data to audio library
282	if (work_size == sound_buffer_size)
283	alWriteFrames(port, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), audio_frames_per_block);
284	else {
285	// Last buffer
286	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
287	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
288	alWriteFrames(port, last_buffer, audio_frames_per_block);
289	}
290	D(bug("stream: data written\n"));
291	} else
292	goto silence;
293
294	} else {
295
296	// Audio not active, play silence
297	silence: // D(bug("stream: silence\n"));
298	alZeroFrames(port, audio_frames_per_block);
299	}
300
301	// Wait for fill point to be reached (may be immediate)
302
303	if (alSetFillPoint(port, sound_buffer_fill_point) < 0) {
304	fprintf(stderr, "ERROR: alSetFillPoint failed: %s\n",
305	alGetErrorString(oserror()));
306	// Should stop the audio here....
307	}
308
309	do {
310	errno = 0;
311	FD_SET(audio_fd, &audio_fdset);
312	was_error = select(numfds, NULL, &audio_fdset, NULL, NULL);
313	} while(was_error < 0 && (errno == EINTR));
314	if (was_error < 0) {
315	fprintf(stderr, "ERROR: select returned %d, errno %d\n",
316	was_error, errno);
317	// Should stop audio here....
318	}
319	}
320	delete[] last_buffer;
321	return NULL;
322	}
323
324
325	/*
326	* MacOS audio interrupt, read next data block
327	*/
328
329	void AudioInterrupt(void)
330	{
331	D(bug("AudioInterrupt\n"));
332
333	// Get data from apple mixer
334	if (AudioStatus.mixer) {
335	M68kRegisters r;
336	r.a[0] = audio_data + adatStreamInfo;
337	r.a[1] = AudioStatus.mixer;
338	Execute68k(audio_data + adatGetSourceData, &r);
339	D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
340	} else
341	WriteMacInt32(audio_data + adatStreamInfo, 0);
342
343	// Signal stream function
344	sem_post(&audio_irq_done_sem);
345	D(bug("AudioInterrupt done\n"));
346	}
347
348
349	/*
350	* Set sampling parameters
351	* "index" is an index into the audio_sample_rates[] etc. arrays
352	* It is guaranteed that AudioStatus.num_sources == 0
353	*/
354
355	bool audio_set_sample_rate(int index)
356	{
357	return true;
358	}
359
360	bool audio_set_sample_size(int index)
361	{
362	return true;
363	}
364
365	bool audio_set_channels(int index)
366	{
367	return true;
368	}
369
370
371	/*
372	* Get/set volume controls (volume values received/returned have the left channel
373	* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
374	* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
375	*/
376
377	bool audio_get_main_mute(void)
378	{
379	return false;
380	}
381
382	uint32 audio_get_main_volume(void)
383	{
384	return 0x01000100;
385	}
386
387	bool audio_get_speaker_mute(void)
388	{
389	return false;
390	}
391
392	uint32 audio_get_speaker_volume(void)
393	{
394	return 0x01000100;
395	}
396
397	void audio_set_main_mute(bool mute)
398	{
399	}
400
401	void audio_set_main_volume(uint32 vol)
402	{
403	}
404
405	void audio_set_speaker_mute(bool mute)
406	{
407	}
408
409	void audio_set_speaker_volume(uint32 vol)
410	{
411	}