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
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_irix.cpp - Audio support, SGI Irix implementation
3			*
4	cebix	1.5	* Basilisk II (C) 1997-2002 Christian Bauer
5	cebix	1.1	*
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	cebix	1.3	audio_sample_rates.push_back(44100 << 16);
164			audio_sample_sizes.push_back(16);
165			audio_channel_counts.push_back(2);
166	cebix	1.1	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	cebix	1.4	bool audio_set_sample_rate(int index)
356	cebix	1.1	{
357	cebix	1.4	return true;
358	cebix	1.1	}
359
360	cebix	1.4	bool audio_set_sample_size(int index)
361	cebix	1.1	{
362	cebix	1.4	return true;
363	cebix	1.1	}
364
365	cebix	1.4	bool audio_set_channels(int index)
366	cebix	1.1	{
367	cebix	1.4	return true;
368	cebix	1.1	}
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			}