src/Unix/audio_oss_esd.cpp

/*
 *  audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
 *
 *  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 <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"

#if 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)
{
#if 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) {
#if 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
                                                memcpy(last_buffer, Mac2HostAddr(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:	1.2
Committed:	1999-10-25T19:01:34Z (25 years ago) by cebix
Branch:	MAIN
CVS Tags:	snapshot-02111999
Changes since 1.1:	+27 -11 lines
Log Message:	- fixes to audio_oss_esd.cpp from Alexander R. Pruss (8-bit mode) - added configuration summary to configure script - prefs_editor_amiga.cpp: output of SCSI prefs was broken
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
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 <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			#if ENABLE_ESD
45			#include <esd.h>
46			#endif
47
48			#define DEBUG 0
49			#include "debug.h"
50
51
52	cebix	1.2	// Supported sample rates, sizes and channels (defaults)
53	cebix	1.1	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	cebix	1.2	static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
71	cebix	1.1	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	cebix	1.2	// 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	cebix	1.1	// 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	cebix	1.2	if (format & (AFMT_S16_BE \| AFMT_S16_LE)) {
107	cebix	1.1	audio_sample_sizes[0] = 16;
108	cebix	1.2	silence_byte = 0;
109			} else {
110	cebix	1.1	audio_sample_sizes[0] = 8;
111	cebix	1.2	silence_byte = 0x80;
112			}
113	cebix	1.1	if (!(format & AFMT_S16_BE))
114			little_endian = true;
115
116			// Set DSP parameters
117	cebix	1.2	format = audio_sample_sizes[0] == 8 ? AFMT_U8 : (little_endian ? AFMT_S16_LE : AFMT_S16_BE);
118	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
119			int frag = 0x0004000c; // Block size: 4096 frames
120			ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
121	cebix	1.2	int stereo = (audio_channel_counts[0] == 2);
122	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
123	cebix	1.2	int rate = audio_sample_rates[0] >> 16;
124	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
125	cebix	1.2	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	cebix	1.1
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			#if 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	cebix	1.2	silence_byte = 0; // Is this correct for 8-bit mode?
161	cebix	1.1
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	cebix	1.2	set_audio_status_format();
187	cebix	1.1	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			#if 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	cebix	1.2	memset(silent_buffer, silence_byte, sound_buffer_size);
297	cebix	1.1
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			memcpy(last_buffer, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size);
330	cebix	1.2	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
331	cebix	1.1	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			}