src/Unix/audio_oss_esd.cpp

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

#include "sysdeps.h"

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

#ifdef __linux__
#include <linux/soundcard.h>
#endif

#ifdef __FreeBSD__
#include <machine/soundcard.h>
#endif

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

#ifdef ENABLE_ESD
#include <esd.h>
#endif

#define DEBUG 0
#include "debug.h"


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

// Constants
#define DSP_NAME "/dev/dsp"

// Global variables
static int audio_fd = -1;                                                       // fd of /dev/dsp or ESD
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 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
static uint8 silence_byte;                                                      // 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

// 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 using /dev/dsp, returns false on error
bool audio_init_dsp(void)
{
        printf("Using " DSP_NAME " audio output\n");

        // Get supported sample formats
        unsigned long format;
        ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
        if ((format & (AFMT_U8 | AFMT_S16_BE | AFMT_S16_LE)) == 0) {
                WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
                close(audio_fd);
                audio_fd = -1;
                return false;
        }
        if (format & (AFMT_S16_BE | AFMT_S16_LE)) {
                audio_sample_sizes[0] = 16;
                silence_byte = 0;
        } else {
                audio_sample_sizes[0] = 8;
                silence_byte = 0x80;
        }
        if (!(format & AFMT_S16_BE))
                little_endian = true;

        // Set DSP parameters
        format = audio_sample_sizes[0] == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
        ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
        int frag = 0x0004000c;          // Block size: 4096 frames
        ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
        int stereo = (audio_channel_counts[0] == 2);
        ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
        int rate = audio_sample_rates[0] >> 16;
        ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
        audio_sample_rates[0] = rate << 16;

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

        // Get sound buffer size
        ioctl(audio_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;
        return true;
}

// Init using ESD, returns false on error
bool audio_init_esd(void)
{
#ifdef ENABLE_ESD
        printf("Using ESD audio output\n");

        // ESD audio format
        esd_format_t format = ESD_STREAM | ESD_PLAY;
        if (AudioStatus.sample_size == 8)
                format |= ESD_BITS8;
        else
                format |= ESD_BITS16;
        if (AudioStatus.channels == 1)
                format |= ESD_MONO;
        else
                format |= ESD_STEREO;

#if WORDS_BIGENDIAN
        little_endian = false;
#else
        little_endian = true;
#endif
        silence_byte = 0;       // Is this correct for 8-bit mode?

        // Open connection to ESD server
        audio_fd = esd_play_stream(format, AudioStatus.sample_rate >> 16, NULL, NULL);
        if (audio_fd < 0) {
                char str[256];
                sprintf(str, GetString(STR_NO_ESD_WARN), strerror(errno));
                WarningAlert(str);
                return false;
        }

        // Sound buffer size = 4096 frames
        audio_frames_per_block = 4096;
        sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
        return true;
#else
        ErrorAlert("Basilisk II has been compiled with ESD support disabled.");
        return false;
#endif
}

void AudioInit(void)
{
        char str[256];

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

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

        // Try to open /dev/dsp
        audio_fd = open(DSP_NAME, O_WRONLY);
        if (audio_fd < 0) {
#ifdef ENABLE_ESD
                if (!audio_init_esd())
                        return;
#else
                sprintf(str, GetString(STR_NO_AUDIO_DEV_WARN), DSP_NAME, strerror(errno));
                WarningAlert(str);
                return;
#endif
        } else
                if (!audio_init_dsp())
                        return;

        // 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) {
                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 /dev/dsp
        if (audio_fd > 0)
                close(audio_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, silence_byte, sound_buffer_size);

        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"));

                        // 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(audio_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
                                                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);
                                        write(audio_fd, last_buffer, sound_buffer_size);
                                }
                                D(bug("stream: data written\n"));
                        } else
                                goto silence;

                } else {

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

bool audio_get_speaker_mute(void)
{
        return false;
}

uint32 audio_get_speaker_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;
}

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_PCM, &p);
        }
}

void audio_set_speaker_mute(bool mute)
{
}

void audio_set_speaker_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);
        }
}
Revision:	1.5
Committed:	2000-07-13T13:47:06Z (24 years, 4 months ago) by cebix
Branch:	MAIN
CVS Tags:	snapshot-13072000
Changes since 1.4:	+3 -3 lines
Log Message:	- first version to run natively on NetBSD/m68k
#	Content
1	/*
2	* audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
3	*
4	* Basilisk II (C) 1997-2000 Christian Bauer
5	*
6	* This program is free software; you can redistribute it and/or modify
7	* it under the terms of the GNU General Public License as published by
8	* the Free Software Foundation; either version 2 of the License, or
9	* (at your option) any later version.
10	*
11	* This program is distributed in the hope that it will be useful,
12	* but WITHOUT ANY WARRANTY; without even the implied warranty of
13	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14	* GNU General Public License for more details.
15	*
16	* You should have received a copy of the GNU General Public License
17	* along with this program; if not, write to the Free Software
18	* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
19	*/
20
21	#include "sysdeps.h"
22
23	#include <sys/ioctl.h>
24	#include <unistd.h>
25	#include <errno.h>
26	#include <pthread.h>
27	#include <semaphore.h>
28
29	#ifdef __linux__
30	#include <linux/soundcard.h>
31	#endif
32
33	#ifdef __FreeBSD__
34	#include <machine/soundcard.h>
35	#endif
36
37	#include "cpu_emulation.h"
38	#include "main.h"
39	#include "prefs.h"
40	#include "user_strings.h"
41	#include "audio.h"
42	#include "audio_defs.h"
43
44	#ifdef ENABLE_ESD
45	#include <esd.h>
46	#endif
47
48	#define DEBUG 0
49	#include "debug.h"
50
51
52	// Supported sample rates, sizes and channels (defaults)
53	int audio_num_sample_rates = 1;
54	uint32 audio_sample_rates[] = {44100 << 16};
55	int audio_num_sample_sizes = 1;
56	uint16 audio_sample_sizes[] = {16};
57	int audio_num_channel_counts = 1;
58	uint16 audio_channel_counts[] = {2};
59
60	// Constants
61	#define DSP_NAME "/dev/dsp"
62
63	// Global variables
64	static int audio_fd = -1; // fd of /dev/dsp or ESD
65	static int mixer_fd = -1; // fd of /dev/mixer
66	static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
67	static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
68	static int sound_buffer_size; // Size of sound buffer in bytes
69	static bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
70	static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
71	static pthread_t stream_thread; // Audio streaming thread
72	static pthread_attr_t stream_thread_attr; // Streaming thread attributes
73	static bool stream_thread_active = false; // Flag: streaming thread installed
74	static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
75
76	// Prototypes
77	static void stream_func(void arg);
78
79
80	/*
81	* Initialization
82	*/
83
84	// Set AudioStatus to reflect current audio stream format
85	static void set_audio_status_format(void)
86	{
87	AudioStatus.sample_rate = audio_sample_rates[0];
88	AudioStatus.sample_size = audio_sample_sizes[0];
89	AudioStatus.channels = audio_channel_counts[0];
90	}
91
92	// Init using /dev/dsp, returns false on error
93	bool audio_init_dsp(void)
94	{
95	printf("Using " DSP_NAME " audio output\n");
96
97	// Get supported sample formats
98	unsigned long format;
99	ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
100	if ((format & (AFMT_U8 \| AFMT_S16_BE \| AFMT_S16_LE)) == 0) {
101	WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
102	close(audio_fd);
103	audio_fd = -1;
104	return false;
105	}
106	if (format & (AFMT_S16_BE \| AFMT_S16_LE)) {
107	audio_sample_sizes[0] = 16;
108	silence_byte = 0;
109	} else {
110	audio_sample_sizes[0] = 8;
111	silence_byte = 0x80;
112	}
113	if (!(format & AFMT_S16_BE))
114	little_endian = true;
115
116	// Set DSP parameters
117	format = audio_sample_sizes[0] == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
118	ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
119	int frag = 0x0004000c; // Block size: 4096 frames
120	ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
121	int stereo = (audio_channel_counts[0] == 2);
122	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
123	int rate = audio_sample_rates[0] >> 16;
124	ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
125	audio_sample_rates[0] = rate << 16;
126
127	// Set AudioStatus again because we now know more about the sound
128	// system's capabilities
129	set_audio_status_format();
130
131	// Get sound buffer size
132	ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
133	D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
134	sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
135	return true;
136	}
137
138	// Init using ESD, returns false on error
139	bool audio_init_esd(void)
140	{
141	#ifdef ENABLE_ESD
142	printf("Using ESD audio output\n");
143
144	// ESD audio format
145	esd_format_t format = ESD_STREAM \| ESD_PLAY;
146	if (AudioStatus.sample_size == 8)
147	format \|= ESD_BITS8;
148	else
149	format \|= ESD_BITS16;
150	if (AudioStatus.channels == 1)
151	format \|= ESD_MONO;
152	else
153	format \|= ESD_STEREO;
154
155	#if WORDS_BIGENDIAN
156	little_endian = false;
157	#else
158	little_endian = true;
159	#endif
160	silence_byte = 0; // Is this correct for 8-bit mode?
161
162	// Open connection to ESD server
163	audio_fd = esd_play_stream(format, AudioStatus.sample_rate >> 16, NULL, NULL);
164	if (audio_fd < 0) {
165	char str[256];
166	sprintf(str, GetString(STR_NO_ESD_WARN), strerror(errno));
167	WarningAlert(str);
168	return false;
169	}
170
171	// Sound buffer size = 4096 frames
172	audio_frames_per_block = 4096;
173	sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
174	return true;
175	#else
176	ErrorAlert("Basilisk II has been compiled with ESD support disabled.");
177	return false;
178	#endif
179	}
180
181	void AudioInit(void)
182	{
183	char str[256];
184
185	// Init audio status (defaults) and feature flags
186	set_audio_status_format();
187	AudioStatus.mixer = 0;
188	AudioStatus.num_sources = 0;
189	audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
190
191	// Sound disabled in prefs? Then do nothing
192	if (PrefsFindBool("nosound"))
193	return;
194
195	// Try to open /dev/dsp
196	audio_fd = open(DSP_NAME, O_WRONLY);
197	if (audio_fd < 0) {
198	#ifdef ENABLE_ESD
199	if (!audio_init_esd())
200	return;
201	#else
202	sprintf(str, GetString(STR_NO_AUDIO_DEV_WARN), DSP_NAME, strerror(errno));
203	WarningAlert(str);
204	return;
205	#endif
206	} else
207	if (!audio_init_dsp())
208	return;
209
210	// Try to open /dev/mixer
211	mixer_fd = open("/dev/mixer", O_RDWR);
212	if (mixer_fd < 0)
213	printf("WARNING: Cannot open /dev/mixer (%s)", strerror(errno));
214
215	// Init semaphore
216	if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
217	return;
218	sem_inited = true;
219
220	// Start streaming thread
221	pthread_attr_init(&stream_thread_attr);
222	#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
223	if (geteuid() == 0) {
224	pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
225	pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
226	struct sched_param fifo_param;
227	fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
228	pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
229	}
230	#endif
231	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
232
233	// Everything OK
234	audio_open = true;
235	}
236
237
238	/*
239	* Deinitialization
240	*/
241
242	void AudioExit(void)
243	{
244	// Stop stream and delete semaphore
245	if (stream_thread_active) {
246	stream_thread_cancel = true;
247	#ifdef HAVE_PTHREAD_CANCEL
248	pthread_cancel(stream_thread);
249	#endif
250	pthread_join(stream_thread, NULL);
251	stream_thread_active = false;
252	}
253	if (sem_inited)
254	sem_destroy(&audio_irq_done_sem);
255
256	// Close /dev/dsp
257	if (audio_fd > 0)
258	close(audio_fd);
259
260	// Close /dev/mixer
261	if (mixer_fd > 0)
262	close(mixer_fd);
263	}
264
265
266	/*
267	* First source added, start audio stream
268	*/
269
270	void audio_enter_stream()
271	{
272	// Streaming thread is always running to avoid clicking noises
273	}
274
275
276	/*
277	* Last source removed, stop audio stream
278	*/
279
280	void audio_exit_stream()
281	{
282	// Streaming thread is always running to avoid clicking noises
283	}
284
285
286	/*
287	* Streaming function
288	*/
289
290	static uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
291
292	static void stream_func(void arg)
293	{
294	int16 *silent_buffer = new int16[sound_buffer_size / 2];
295	int16 *last_buffer = new int16[sound_buffer_size / 2];
296	memset(silent_buffer, silence_byte, sound_buffer_size);
297
298	while (!stream_thread_cancel) {
299	if (AudioStatus.num_sources) {
300
301	// Trigger audio interrupt to get new buffer
302	D(bug("stream: triggering irq\n"));
303	SetInterruptFlag(INTFLAG_AUDIO);
304	TriggerInterrupt();
305	D(bug("stream: waiting for ack\n"));
306	sem_wait(&audio_irq_done_sem);
307	D(bug("stream: ack received\n"));
308
309	// Get size of audio data
310	uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
311	if (apple_stream_info) {
312	int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
313	D(bug("stream: work_size %d\n", work_size));
314	if (work_size > sound_buffer_size)
315	work_size = sound_buffer_size;
316	if (work_size == 0)
317	goto silence;
318
319	// Send data to DSP
320	if (work_size == sound_buffer_size && !little_endian)
321	write(audio_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
322	else {
323	// Last buffer or little-endian DSP
324	if (little_endian) {
325	int16 p = (int16 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
326	for (int i=0; i<work_size/2; i++)
327	last_buffer[i] = ntohs(p[i]);
328	} else
329	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
330	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
331	write(audio_fd, last_buffer, sound_buffer_size);
332	}
333	D(bug("stream: data written\n"));
334	} else
335	goto silence;
336
337	} else {
338
339	// Audio not active, play silence
340	silence: write(audio_fd, silent_buffer, sound_buffer_size);
341	}
342	}
343	delete[] silent_buffer;
344	delete[] last_buffer;
345	return NULL;
346	}
347
348
349	/*
350	* MacOS audio interrupt, read next data block
351	*/
352
353	void AudioInterrupt(void)
354	{
355	D(bug("AudioInterrupt\n"));
356
357	// Get data from apple mixer
358	if (AudioStatus.mixer) {
359	M68kRegisters r;
360	r.a[0] = audio_data + adatStreamInfo;
361	r.a[1] = AudioStatus.mixer;
362	Execute68k(audio_data + adatGetSourceData, &r);
363	D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
364	} else
365	WriteMacInt32(audio_data + adatStreamInfo, 0);
366
367	// Signal stream function
368	sem_post(&audio_irq_done_sem);
369	D(bug("AudioInterrupt done\n"));
370	}
371
372
373	/*
374	* Set sampling parameters
375	* "index" is an index into the audio_sample_rates[] etc. arrays
376	* It is guaranteed that AudioStatus.num_sources == 0
377	*/
378
379	void audio_set_sample_rate(int index)
380	{
381	}
382
383	void audio_set_sample_size(int index)
384	{
385	}
386
387	void audio_set_channels(int index)
388	{
389	}
390
391
392	/*
393	* Get/set volume controls (volume values received/returned have the left channel
394	* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
395	* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
396	*/
397
398	bool audio_get_main_mute(void)
399	{
400	return false;
401	}
402
403	uint32 audio_get_main_volume(void)
404	{
405	if (mixer_fd >= 0) {
406	int vol;
407	if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
408	int left = vol >> 8;
409	int right = vol & 0xff;
410	return ((left * 256 / 100) << 16) \| (right * 256 / 100);
411	}
412	}
413	return 0x01000100;
414	}
415
416	bool audio_get_speaker_mute(void)
417	{
418	return false;
419	}
420
421	uint32 audio_get_speaker_volume(void)
422	{
423	if (mixer_fd >= 0) {
424	int vol;
425	if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
426	int left = vol >> 8;
427	int right = vol & 0xff;
428	return ((left * 256 / 100) << 16) \| (right * 256 / 100);
429	}
430	}
431	return 0x01000100;
432	}
433
434	void audio_set_main_mute(bool mute)
435	{
436	}
437
438	void audio_set_main_volume(uint32 vol)
439	{
440	if (mixer_fd >= 0) {
441	int left = vol >> 16;
442	int right = vol & 0xffff;
443	int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
444	ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
445	}
446	}
447
448	void audio_set_speaker_mute(bool mute)
449	{
450	}
451
452	void audio_set_speaker_volume(uint32 vol)
453	{
454	if (mixer_fd >= 0) {
455	int left = vol >> 16;
456	int right = vol & 0xffff;
457	int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
458	ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
459	}
460	}