Unix/Linux/audio_linux.cpp

/*
 *  audio_linux.cpp - Audio support, Linux (OSS) implementation
 *
 *  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 <linux/soundcard.h>
#include <unistd.h>
#include <errno.h>
#include <pthread.h>
#include <semaphore.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
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 dsp_fd = -1;                                         // fd of /dev/dsp
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 pthread_t stream_thread;                         // Audio streaming thread
static pthread_attr_t stream_thread_attr;       // Streaming thread attributes
static bool stream_thread_active = false;
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

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


/*
 *  Initialization
 */

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;

        // Open /dev/dsp
        dsp_fd = open("/dev/dsp", O_WRONLY);
        if (dsp_fd < 0) {
                sprintf(str, GetString(STR_NO_AUDIO_DEV_WARN), "/dev/dsp", strerror(errno));
                WarningAlert(str);
                return;
        }

        // Get supported sample formats
        unsigned long format;
        ioctl(dsp_fd, SNDCTL_DSP_GETFMTS, &format);
        if ((format & (AFMT_U8 | AFMT_S16_BE | AFMT_S16_LE)) == 0) {
                WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
                close(dsp_fd);
                dsp_fd = -1;
                return;
        }
        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(dsp_fd, SNDCTL_DSP_SETFMT, &format);
        int frag = 0x0004000c;          // Block size: 4096 frames
        ioctl(dsp_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
        int stereo = (AudioStatus.channels == 2);
        ioctl(dsp_fd, SNDCTL_DSP_STEREO, &stereo);
        int rate = AudioStatus.sample_rate >> 16;
        ioctl(dsp_fd, SNDCTL_DSP_SPEED, &rate);

        // Get sound buffer size
        ioctl(dsp_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;

        // 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) {
                pthread_cancel(stream_thread);
                pthread_join(stream_thread, NULL);
                stream_thread_active = false;
        }
        if (sem_inited)
                sem_destroy(&audio_irq_done_sem);

        // Close /dev/dsp
        if (dsp_fd > 0)
                close(dsp_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);

        for (;;) {
                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(dsp_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(dsp_fd, last_buffer, sound_buffer_size);
                                }
                                D(bug("stream: data written\n"));
                        } else
                                goto silence;

                } else {

                        // Audio not active, play silence
silence:        write(dsp_fd, silent_buffer, sound_buffer_size);
                }
        }
        ioctl(dsp_fd, SNDCTL_DSP_SYNC);
        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_VOLUME, &vol) == 0) {
                        int left = vol >> 8;
                        int right = vol & 0xff;
                        return ((left * 256 / 100) << 16) | (right * 256 / 100);
                }
        }
        return 0x01000100;
}

bool audio_get_dac_mute(void)
{
        return false;
}

uint32 audio_get_dac_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;
}

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_VOLUME, &p);
        }
}

void audio_set_dac_mute(bool mute)
{
}

void audio_set_dac_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);
        }
}
Revision:	1.1
Committed:	1999-10-03T14:16:25Z (25 years, 1 month ago) by cebix
Branch:	MAIN
Branch point for:	cebix
Log Message:	Initial revision
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_linux.cpp - Audio support, Linux (OSS) implementation
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 <linux/soundcard.h>
25			#include <unistd.h>
26			#include <errno.h>
27			#include <pthread.h>
28			#include <semaphore.h>
29
30			#include "cpu_emulation.h"
31			#include "main.h"
32			#include "prefs.h"
33			#include "user_strings.h"
34			#include "audio.h"
35			#include "audio_defs.h"
36
37			#define DEBUG 0
38			#include "debug.h"
39
40
41			// Supported sample rates, sizes and channels
42			int audio_num_sample_rates = 1;
43			uint32 audio_sample_rates[] = {44100 << 16};
44			int audio_num_sample_sizes = 1;
45			uint16 audio_sample_sizes[] = {16};
46			int audio_num_channel_counts = 1;
47			uint16 audio_channel_counts[] = {2};
48
49			// Global variables
50			static int dsp_fd = -1; // fd of /dev/dsp
51			static int mixer_fd = -1; // fd of /dev/mixer
52			static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
53			static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
54			static pthread_t stream_thread; // Audio streaming thread
55			static pthread_attr_t stream_thread_attr; // Streaming thread attributes
56			static bool stream_thread_active = false;
57			static int sound_buffer_size; // Size of sound buffer in bytes
58			static bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
59
60			// Prototypes
61			static void stream_func(void arg);
62
63
64			/*
65			* Initialization
66			*/
67
68			void AudioInit(void)
69			{
70			char str[256];
71
72			// Init audio status (defaults) and feature flags
73			AudioStatus.sample_rate = audio_sample_rates[0];
74			AudioStatus.sample_size = audio_sample_sizes[0];
75			AudioStatus.channels = audio_channel_counts[0];
76			AudioStatus.mixer = 0;
77			AudioStatus.num_sources = 0;
78			audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
79
80			// Sound disabled in prefs? Then do nothing
81			if (PrefsFindBool("nosound"))
82			return;
83
84			// Open /dev/dsp
85			dsp_fd = open("/dev/dsp", O_WRONLY);
86			if (dsp_fd < 0) {
87			sprintf(str, GetString(STR_NO_AUDIO_DEV_WARN), "/dev/dsp", strerror(errno));
88			WarningAlert(str);
89			return;
90			}
91
92			// Get supported sample formats
93			unsigned long format;
94			ioctl(dsp_fd, SNDCTL_DSP_GETFMTS, &format);
95			if ((format & (AFMT_U8 \| AFMT_S16_BE \| AFMT_S16_LE)) == 0) {
96			WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
97			close(dsp_fd);
98			dsp_fd = -1;
99			return;
100			}
101			if (format & (AFMT_S16_BE \| AFMT_S16_LE))
102			audio_sample_sizes[0] = 16;
103			else
104			audio_sample_sizes[0] = 8;
105			if (!(format & AFMT_S16_BE))
106			little_endian = true;
107
108			// Set DSP parameters
109			format = AudioStatus.sample_size == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
110			ioctl(dsp_fd, SNDCTL_DSP_SETFMT, &format);
111			int frag = 0x0004000c; // Block size: 4096 frames
112			ioctl(dsp_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
113			int stereo = (AudioStatus.channels == 2);
114			ioctl(dsp_fd, SNDCTL_DSP_STEREO, &stereo);
115			int rate = AudioStatus.sample_rate >> 16;
116			ioctl(dsp_fd, SNDCTL_DSP_SPEED, &rate);
117
118			// Get sound buffer size
119			ioctl(dsp_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
120			D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
121			sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
122
123			// Try to open /dev/mixer
124			mixer_fd = open("/dev/mixer", O_RDWR);
125			if (mixer_fd < 0)
126			printf("WARNING: Cannot open /dev/mixer (%s)", strerror(errno));
127
128			// Init semaphore
129			if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
130			return;
131			sem_inited = true;
132
133			// Start streaming thread
134			pthread_attr_init(&stream_thread_attr);
135			#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
136			if (geteuid() == 0) {
137			pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
138			pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
139			struct sched_param fifo_param;
140			fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
141			pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
142			}
143			#endif
144			stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
145
146			// Everything OK
147			audio_open = true;
148			}
149
150
151			/*
152			* Deinitialization
153			*/
154
155			void AudioExit(void)
156			{
157			// Stop stream and delete semaphore
158			if (stream_thread_active) {
159			pthread_cancel(stream_thread);
160			pthread_join(stream_thread, NULL);
161			stream_thread_active = false;
162			}
163			if (sem_inited)
164			sem_destroy(&audio_irq_done_sem);
165
166			// Close /dev/dsp
167			if (dsp_fd > 0)
168			close(dsp_fd);
169
170			// Close /dev/mixer
171			if (mixer_fd > 0)
172			close(mixer_fd);
173			}
174
175
176			/*
177			* First source added, start audio stream
178			*/
179
180			void audio_enter_stream()
181			{
182			// Streaming thread is always running to avoid clicking noises
183			}
184
185
186			/*
187			* Last source removed, stop audio stream
188			*/
189
190			void audio_exit_stream()
191			{
192			// Streaming thread is always running to avoid clicking noises
193			}
194
195
196			/*
197			* Streaming function
198			*/
199
200			static uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
201
202			static void stream_func(void arg)
203			{
204			int16 *silent_buffer = new int16[sound_buffer_size / 2];
205			int16 *last_buffer = new int16[sound_buffer_size / 2];
206			memset(silent_buffer, 0, sound_buffer_size);
207
208			for (;;) {
209			if (AudioStatus.num_sources) {
210
211			// Trigger audio interrupt to get new buffer
212			D(bug("stream: triggering irq\n"));
213			SetInterruptFlag(INTFLAG_AUDIO);
214			TriggerInterrupt();
215			D(bug("stream: waiting for ack\n"));
216			sem_wait(&audio_irq_done_sem);
217			D(bug("stream: ack received\n"));
218
219			// Get size of audio data
220			uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
221			if (apple_stream_info) {
222			int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
223			D(bug("stream: work_size %d\n", work_size));
224			if (work_size > sound_buffer_size)
225			work_size = sound_buffer_size;
226			if (work_size == 0)
227			goto silence;
228
229			// Send data to DSP
230			if (work_size == sound_buffer_size && !little_endian)
231			write(dsp_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
232			else {
233			// Last buffer or little-endian DSP
234			if (little_endian) {
235			int16 p = (int16 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
236			for (int i=0; i<work_size/2; i++)
237			last_buffer[i] = ntohs(p[i]);
238			} else
239			memcpy(last_buffer, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size);
240			memset((uint8 *)last_buffer + work_size, 0, sound_buffer_size - work_size);
241			write(dsp_fd, last_buffer, sound_buffer_size);
242			}
243			D(bug("stream: data written\n"));
244			} else
245			goto silence;
246
247			} else {
248
249			// Audio not active, play silence
250			silence: write(dsp_fd, silent_buffer, sound_buffer_size);
251			}
252			}
253			ioctl(dsp_fd, SNDCTL_DSP_SYNC);
254			delete[] silent_buffer;
255			delete[] last_buffer;
256			return NULL;
257			}
258
259
260			/*
261			* MacOS audio interrupt, read next data block
262			*/
263
264			void AudioInterrupt(void)
265			{
266			D(bug("AudioInterrupt\n"));
267
268			// Get data from apple mixer
269			if (AudioStatus.mixer) {
270			M68kRegisters r;
271			r.a[0] = audio_data + adatStreamInfo;
272			r.a[1] = AudioStatus.mixer;
273			Execute68k(audio_data + adatGetSourceData, &r);
274			D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
275			} else
276			WriteMacInt32(audio_data + adatStreamInfo, 0);
277
278			// Signal stream function
279			sem_post(&audio_irq_done_sem);
280			D(bug("AudioInterrupt done\n"));
281			}
282
283
284			/*
285			* Set sampling parameters
286			* "index" is an index into the audio_sample_rates[] etc. arrays
287			* It is guaranteed that AudioStatus.num_sources == 0
288			*/
289
290			void audio_set_sample_rate(int index)
291			{
292			}
293
294			void audio_set_sample_size(int index)
295			{
296			}
297
298			void audio_set_channels(int index)
299			{
300			}
301
302
303			/*
304			* Get/set volume controls (volume values received/returned have the left channel
305			* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
306			* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
307			*/
308
309			bool audio_get_main_mute(void)
310			{
311			return false;
312			}
313
314			uint32 audio_get_main_volume(void)
315			{
316			if (mixer_fd >= 0) {
317			int vol;
318			if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
319			int left = vol >> 8;
320			int right = vol & 0xff;
321			return ((left * 256 / 100) << 16) \| (right * 256 / 100);
322			}
323			}
324			return 0x01000100;
325			}
326
327			bool audio_get_dac_mute(void)
328			{
329			return false;
330			}
331
332			uint32 audio_get_dac_volume(void)
333			{
334			if (mixer_fd >= 0) {
335			int vol;
336			if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
337			int left = vol >> 8;
338			int right = vol & 0xff;
339			return ((left * 256 / 100) << 16) \| (right * 256 / 100);
340			}
341			}
342			return 0x01000100;
343			}
344
345			void audio_set_main_mute(bool mute)
346			{
347			}
348
349			void audio_set_main_volume(uint32 vol)
350			{
351			if (mixer_fd >= 0) {
352			int left = vol >> 16;
353			int right = vol & 0xffff;
354			int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
355			ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
356			}
357			}
358
359			void audio_set_dac_mute(bool mute)
360			{
361			}
362
363			void audio_set_dac_volume(uint32 vol)
364			{
365			if (mixer_fd >= 0) {
366			int left = vol >> 16;
367			int right = vol & 0xffff;
368			int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
369			ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
370			}
371			}