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:	1.1.1.1 (vendor branch)
Committed:	1999-10-03T14:16:25Z (25 years, 1 month ago) by cebix
Branch:	cebix
CVS Tags:	start, release-0_7-2
Changes since 1.1:	+0 -0 lines
Log Message:	Imported sources
#	Content
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	}