src/Unix/audio_oss_esd.cpp

/*
 *  audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
 *
 *  Basilisk II (C) 1997-2002 Christian Bauer
 *
 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation; either version 2 of the License, or
 *  (at your option) any later version.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */

#include "sysdeps.h"

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

#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"


// The currently selected audio parameters (indices in audio_sample_rates[] etc. vectors)
static int audio_sample_rate_index = 0;
static int audio_sample_size_index = 0;
static int audio_channel_count_index = 0;

// 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[audio_sample_rate_index];
        AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
        AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
}

// Init using /dev/dsp, returns false on error
static bool open_dsp(void)
{
        // Open /dev/dsp
        audio_fd = open(DSP_NAME, O_WRONLY);
        if (audio_fd < 0) {
                fprintf(stderr, "WARNING: Cannot open %s (%s)\n", DSP_NAME, strerror(errno));
                return false;
        }

        printf("Using " DSP_NAME " audio output\n");

        // Get supported sample formats
        if (audio_sample_sizes.empty()) {
                unsigned long format;
                ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
                if (format & AFMT_U8)
                        audio_sample_sizes.push_back(8);
                if (format & (AFMT_S16_BE | AFMT_S16_LE))
                        audio_sample_sizes.push_back(16);

                int stereo = 0;
                if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 0)
                        audio_channel_counts.push_back(1);
                stereo = 1;
                if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 1)
                        audio_channel_counts.push_back(2);

                if (audio_sample_sizes.empty() || audio_channel_counts.empty()) {
                        WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
                        close(audio_fd);
                        audio_fd = -1;
                        return false;
                }

                audio_sample_rates.push_back(11025 << 16);
                audio_sample_rates.push_back(22050 << 16);
                int rate = 44100;
                ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
                if (rate > 22050)
                        audio_sample_rates.push_back(rate << 16);

                // Default to highest supported values
                audio_sample_rate_index = audio_sample_rates.size() - 1;
                audio_sample_size_index = audio_sample_sizes.size() - 1;
                audio_channel_count_index = audio_channel_counts.size() - 1;
        }

        // Set DSP parameters
        unsigned long format;
        if (audio_sample_sizes[audio_sample_size_index] == 8) {
                format = AFMT_U8;
                little_endian = false;
                silence_byte = 0x80;
        } else {
                unsigned long sup_format;
                ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &sup_format);
                if (sup_format & AFMT_S16_BE) {
                        little_endian = false;
                        format = AFMT_S16_BE;
                } else {
                        little_endian = true;
                        format = AFMT_S16_LE;
                }
                silence_byte = 0;
        }
        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[audio_channel_count_index] == 2);
        ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
        int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
        ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);

        // Get sound buffer size
        ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
        D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
        return true;
}

// Init using ESD, returns false on error
static bool open_esd(void)
{
#ifdef ENABLE_ESD
        int rate;
        esd_format_t format = ESD_STREAM | ESD_PLAY;

        if (audio_sample_sizes.empty()) {

                // Default values
                rate = 44100;
                format |= (ESD_BITS16 | ESD_STEREO);

        } else {

                rate = audio_sample_rates[audio_sample_rate_index] >> 16;
                if (audio_sample_sizes[audio_sample_size_index] == 8)
                        format |= ESD_BITS8;
                else
                        format |= ESD_BITS16;
                if (audio_channel_counts[audio_channel_count_index] == 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, rate, NULL, NULL);
        if (audio_fd < 0) {
                fprintf(stderr, "WARNING: Cannot open ESD connection\n");
                return false;
        }

        printf("Using ESD audio output\n");

        // ESD supports a variety of twisted little audio formats, all different
        if (audio_sample_sizes.empty()) {

                // The reason we do this here is that we don't want to add sample
                // rates etc. unless the ESD server connection could be opened
                // (if ESD fails, /dev/dsp might be tried next)
                audio_sample_rates.push_back(11025 << 16);
                audio_sample_rates.push_back(22050 << 16);
                audio_sample_rates.push_back(44100 << 16);
                audio_sample_sizes.push_back(8);
                audio_sample_sizes.push_back(16);
                audio_channel_counts.push_back(1);
                audio_channel_counts.push_back(2);

                // Default to highest supported values
                audio_sample_rate_index = audio_sample_rates.size() - 1;
                audio_sample_size_index = audio_sample_sizes.size() - 1;
                audio_channel_count_index = audio_channel_counts.size() - 1;
        }

        // Sound buffer size = 4096 frames
        audio_frames_per_block = 4096;
        return true;
#else
        // ESD is not enabled, shut up the compiler
        return false;
#endif
}

static bool open_audio(void)
{
#ifdef ENABLE_ESD
        // If ESPEAKER is set, the user probably wants to use ESD, so try that first
        if (getenv("ESPEAKER"))
                if (open_esd())
                        goto dev_opened;
#endif

        // Try to open /dev/dsp
        if (open_dsp())
                goto dev_opened;

#ifdef ENABLE_ESD
        // Hm, /dev/dsp failed so we try ESD again if ESPEAKER wasn't set
        if (!getenv("ESPEAKER"))
                if (open_esd())
                        goto dev_opened;
#endif

        // No audio device succeeded
        WarningAlert(GetString(STR_NO_AUDIO_WARN));
        return false;

        // Device opened, set AudioStatus
dev_opened:
        sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) * audio_channel_counts[audio_channel_count_index] * audio_frames_per_block;
        set_audio_status_format();

        // Start streaming thread
        Set_pthread_attr(&stream_thread_attr, 0);
        stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);

        // Everything went fine
        audio_open = true;
        return true;
}

void AudioInit(void)
{
        // Init audio status (reasonable defaults) and feature flags
        AudioStatus.sample_rate = 44100 << 16;
        AudioStatus.sample_size = 16;
        AudioStatus.channels = 2;
        AudioStatus.mixer = 0;
        AudioStatus.num_sources = 0;
        audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;

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

        // Init semaphore
        if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
                return;
        sem_inited = true;

        // 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));

        // Open and initialize audio device
        open_audio();
}


/*
 *  Deinitialization
 */

static void close_audio(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;
        }

        // Close /dev/dsp or ESD socket
        if (audio_fd >= 0) {
                close(audio_fd);
                audio_fd = -1;
        }

        audio_open = false;
}

void AudioExit(void)
{
        // Close audio device
        close_audio();

        // Delete semaphore
        if (sem_inited) {
                sem_destroy(&audio_irq_done_sem);
                sem_inited = false;
        }

        // Close /dev/mixer
        if (mixer_fd >= 0) {
                close(mixer_fd);
                mixer_fd = -1;
        }
}


/*
 *  First source added, start audio stream
 */

void audio_enter_stream()
{
        // Streaming thread is always running to avoid clicking noises
}


/*
 *  Last source removed, stop audio stream
 */

void audio_exit_stream()
{
        // Streaming thread is always running to avoid clicking noises
}


/*
 *  Streaming function
 */

static void *stream_func(void *arg)
{
        int16 *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. vectors
 *  It is guaranteed that AudioStatus.num_sources == 0
 */

bool audio_set_sample_rate(int index)
{
        close_audio();
        audio_sample_rate_index = index;
        return open_audio();
}

bool audio_set_sample_size(int index)
{
        close_audio();
        audio_sample_size_index = index;
        return open_audio();
}

bool audio_set_channels(int index)
{
        close_audio();
        audio_channel_count_index = index;
        return open_audio();
}


/*
 *  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:	1.13
Committed:	2002-02-07T16:10:55Z (22 years, 9 months ago) by cebix
Branch:	MAIN
Changes since 1.12:	+1 -11 lines
Log Message:	cleaned up pthread attributes [Brian Johnson]
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
3			*
4	cebix	1.12	* Basilisk II (C) 1997-2002 Christian Bauer
5	cebix	1.1	*
6			* This program is free software; you can redistribute it and/or modify
7			* it under the terms of the GNU General Public License as published by
8			* the Free Software Foundation; either version 2 of the License, or
9			* (at your option) any later version.
10			*
11			* This program is distributed in the hope that it will be useful,
12			* but WITHOUT ANY WARRANTY; without even the implied warranty of
13			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14			* GNU General Public License for more details.
15			*
16			* You should have received a copy of the GNU General Public License
17			* along with this program; if not, write to the Free Software
18			* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
19			*/
20
21			#include "sysdeps.h"
22
23			#include <sys/ioctl.h>
24			#include <unistd.h>
25			#include <errno.h>
26			#include <pthread.h>
27			#include <semaphore.h>
28
29			#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	cebix	1.5	#ifdef ENABLE_ESD
45	cebix	1.1	#include <esd.h>
46			#endif
47
48			#define DEBUG 0
49			#include "debug.h"
50
51
52	cebix	1.7	// The currently selected audio parameters (indices in audio_sample_rates[] etc. vectors)
53			static int audio_sample_rate_index = 0;
54			static int audio_sample_size_index = 0;
55			static int audio_channel_count_index = 0;
56	cebix	1.1
57			// Constants
58			#define DSP_NAME "/dev/dsp"
59
60			// Global variables
61			static int audio_fd = -1; // fd of /dev/dsp or ESD
62			static int mixer_fd = -1; // fd of /dev/mixer
63			static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
64			static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
65			static int sound_buffer_size; // Size of sound buffer in bytes
66			static bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
67	cebix	1.2	static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
68	cebix	1.1	static pthread_t stream_thread; // Audio streaming thread
69			static pthread_attr_t stream_thread_attr; // Streaming thread attributes
70			static bool stream_thread_active = false; // Flag: streaming thread installed
71			static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
72
73			// Prototypes
74			static void stream_func(void arg);
75
76
77			/*
78			* Initialization
79			*/
80
81	cebix	1.2	// Set AudioStatus to reflect current audio stream format
82			static void set_audio_status_format(void)
83			{
84	cebix	1.7	AudioStatus.sample_rate = audio_sample_rates[audio_sample_rate_index];
85			AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
86			AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
87	cebix	1.2	}
88
89	cebix	1.1	// Init using /dev/dsp, returns false on error
90	cebix	1.7	static bool open_dsp(void)
91	cebix	1.1	{
92	cebix	1.7	// Open /dev/dsp
93			audio_fd = open(DSP_NAME, O_WRONLY);
94			if (audio_fd < 0) {
95			fprintf(stderr, "WARNING: Cannot open %s (%s)\n", DSP_NAME, strerror(errno));
96			return false;
97			}
98
99	cebix	1.1	printf("Using " DSP_NAME " audio output\n");
100
101			// Get supported sample formats
102	cebix	1.8	if (audio_sample_sizes.empty()) {
103	cebix	1.7	unsigned long format;
104			ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
105			if (format & AFMT_U8)
106			audio_sample_sizes.push_back(8);
107			if (format & (AFMT_S16_BE \| AFMT_S16_LE))
108			audio_sample_sizes.push_back(16);
109
110			int stereo = 0;
111			if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 0)
112			audio_channel_counts.push_back(1);
113			stereo = 1;
114			if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 1)
115			audio_channel_counts.push_back(2);
116
117			if (audio_sample_sizes.empty() \|\| audio_channel_counts.empty()) {
118			WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
119			close(audio_fd);
120			audio_fd = -1;
121			return false;
122			}
123
124			audio_sample_rates.push_back(11025 << 16);
125			audio_sample_rates.push_back(22050 << 16);
126			int rate = 44100;
127			ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
128			if (rate > 22050)
129			audio_sample_rates.push_back(rate << 16);
130
131			// Default to highest supported values
132			audio_sample_rate_index = audio_sample_rates.size() - 1;
133			audio_sample_size_index = audio_sample_sizes.size() - 1;
134			audio_channel_count_index = audio_channel_counts.size() - 1;
135			}
136
137			// Set DSP parameters
138	cebix	1.1	unsigned long format;
139	cebix	1.7	if (audio_sample_sizes[audio_sample_size_index] == 8) {
140			format = AFMT_U8;
141			little_endian = false;
142			silence_byte = 0x80;
143			} else {
144			unsigned long sup_format;
145	cebix	1.9	ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &sup_format);
146	cebix	1.7	if (sup_format & AFMT_S16_BE) {
147			little_endian = false;
148			format = AFMT_S16_BE;
149			} else {
150			little_endian = true;
151			format = AFMT_S16_LE;
152			}
153	cebix	1.2	silence_byte = 0;
154			}
155	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
156			int frag = 0x0004000c; // Block size: 4096 frames
157			ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
158	cebix	1.7	int stereo = (audio_channel_counts[audio_channel_count_index] == 2);
159	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
160	cebix	1.7	int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
161	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
162
163			// Get sound buffer size
164			ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
165			D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
166			return true;
167			}
168
169			// Init using ESD, returns false on error
170	cebix	1.7	static bool open_esd(void)
171	cebix	1.1	{
172	cebix	1.5	#ifdef ENABLE_ESD
173	cebix	1.8	int rate;
174			esd_format_t format = ESD_STREAM \| ESD_PLAY;
175
176			if (audio_sample_sizes.empty()) {
177
178			// Default values
179			rate = 44100;
180			format \|= (ESD_BITS16 \| ESD_STEREO);
181
182			} else {
183	cebix	1.7
184	cebix	1.8	rate = audio_sample_rates[audio_sample_rate_index] >> 16;
185			if (audio_sample_sizes[audio_sample_size_index] == 8)
186			format \|= ESD_BITS8;
187			else
188			format \|= ESD_BITS16;
189			if (audio_channel_counts[audio_channel_count_index] == 1)
190			format \|= ESD_MONO;
191			else
192			format \|= ESD_STEREO;
193	cebix	1.7	}
194	cebix	1.1
195			#if WORDS_BIGENDIAN
196			little_endian = false;
197			#else
198			little_endian = true;
199			#endif
200	cebix	1.2	silence_byte = 0; // Is this correct for 8-bit mode?
201	cebix	1.1
202			// Open connection to ESD server
203	cebix	1.8	audio_fd = esd_play_stream(format, rate, NULL, NULL);
204	cebix	1.1	if (audio_fd < 0) {
205	cebix	1.7	fprintf(stderr, "WARNING: Cannot open ESD connection\n");
206	cebix	1.1	return false;
207			}
208
209	cebix	1.7	printf("Using ESD audio output\n");
210	cebix	1.8
211			// ESD supports a variety of twisted little audio formats, all different
212			if (audio_sample_sizes.empty()) {
213
214			// The reason we do this here is that we don't want to add sample
215			// rates etc. unless the ESD server connection could be opened
216			// (if ESD fails, /dev/dsp might be tried next)
217			audio_sample_rates.push_back(11025 << 16);
218			audio_sample_rates.push_back(22050 << 16);
219			audio_sample_rates.push_back(44100 << 16);
220			audio_sample_sizes.push_back(8);
221			audio_sample_sizes.push_back(16);
222			audio_channel_counts.push_back(1);
223			audio_channel_counts.push_back(2);
224
225			// Default to highest supported values
226			audio_sample_rate_index = audio_sample_rates.size() - 1;
227			audio_sample_size_index = audio_sample_sizes.size() - 1;
228			audio_channel_count_index = audio_channel_counts.size() - 1;
229			}
230	cebix	1.7
231	cebix	1.1	// Sound buffer size = 4096 frames
232			audio_frames_per_block = 4096;
233			return true;
234	gbeauche	1.11	#else
235			// ESD is not enabled, shut up the compiler
236			return false;
237	cebix	1.1	#endif
238			}
239
240	cebix	1.7	static bool open_audio(void)
241	cebix	1.1	{
242	cebix	1.7	#ifdef ENABLE_ESD
243			// If ESPEAKER is set, the user probably wants to use ESD, so try that first
244			if (getenv("ESPEAKER"))
245			if (open_esd())
246			goto dev_opened;
247			#endif
248	cebix	1.1
249	cebix	1.7	// Try to open /dev/dsp
250			if (open_dsp())
251			goto dev_opened;
252	cebix	1.1
253	cebix	1.5	#ifdef ENABLE_ESD
254	cebix	1.7	// Hm, /dev/dsp failed so we try ESD again if ESPEAKER wasn't set
255			if (!getenv("ESPEAKER"))
256			if (open_esd())
257			goto dev_opened;
258	cebix	1.1	#endif
259
260	cebix	1.7	// No audio device succeeded
261			WarningAlert(GetString(STR_NO_AUDIO_WARN));
262			return false;
263	cebix	1.1
264	cebix	1.7	// Device opened, set AudioStatus
265			dev_opened:
266			sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) * audio_channel_counts[audio_channel_count_index] * audio_frames_per_block;
267			set_audio_status_format();
268	cebix	1.1
269			// Start streaming thread
270	cebix	1.13	Set_pthread_attr(&stream_thread_attr, 0);
271	cebix	1.1	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
272
273	cebix	1.7	// Everything went fine
274	cebix	1.1	audio_open = true;
275	cebix	1.7	return true;
276			}
277
278			void AudioInit(void)
279			{
280			// Init audio status (reasonable defaults) and feature flags
281			AudioStatus.sample_rate = 44100 << 16;
282			AudioStatus.sample_size = 16;
283			AudioStatus.channels = 2;
284			AudioStatus.mixer = 0;
285			AudioStatus.num_sources = 0;
286			audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
287
288			// Sound disabled in prefs? Then do nothing
289			if (PrefsFindBool("nosound"))
290			return;
291
292			// Init semaphore
293			if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
294			return;
295			sem_inited = true;
296
297			// Try to open /dev/mixer
298			mixer_fd = open("/dev/mixer", O_RDWR);
299			if (mixer_fd < 0)
300			printf("WARNING: Cannot open /dev/mixer (%s)", strerror(errno));
301
302			// Open and initialize audio device
303			open_audio();
304	cebix	1.1	}
305
306
307			/*
308			* Deinitialization
309			*/
310
311	cebix	1.7	static void close_audio(void)
312	cebix	1.1	{
313			// Stop stream and delete semaphore
314			if (stream_thread_active) {
315			stream_thread_cancel = true;
316			#ifdef HAVE_PTHREAD_CANCEL
317			pthread_cancel(stream_thread);
318			#endif
319			pthread_join(stream_thread, NULL);
320			stream_thread_active = false;
321			}
322
323	cebix	1.7	// Close /dev/dsp or ESD socket
324			if (audio_fd >= 0) {
325	cebix	1.1	close(audio_fd);
326	cebix	1.7	audio_fd = -1;
327			}
328
329			audio_open = false;
330			}
331
332			void AudioExit(void)
333			{
334	cebix	1.10	// Close audio device
335			close_audio();
336
337			// Delete semaphore
338	cebix	1.7	if (sem_inited) {
339			sem_destroy(&audio_irq_done_sem);
340			sem_inited = false;
341			}
342	cebix	1.1
343			// Close /dev/mixer
344	cebix	1.7	if (mixer_fd >= 0) {
345	cebix	1.1	close(mixer_fd);
346	cebix	1.7	mixer_fd = -1;
347			}
348	cebix	1.1	}
349
350
351			/*
352			* First source added, start audio stream
353			*/
354
355			void audio_enter_stream()
356			{
357			// Streaming thread is always running to avoid clicking noises
358			}
359
360
361			/*
362			* Last source removed, stop audio stream
363			*/
364
365			void audio_exit_stream()
366			{
367			// Streaming thread is always running to avoid clicking noises
368			}
369
370
371			/*
372			* Streaming function
373			*/
374
375			static void stream_func(void arg)
376			{
377			int16 *silent_buffer = new int16[sound_buffer_size / 2];
378			int16 *last_buffer = new int16[sound_buffer_size / 2];
379	cebix	1.2	memset(silent_buffer, silence_byte, sound_buffer_size);
380	cebix	1.1
381			while (!stream_thread_cancel) {
382			if (AudioStatus.num_sources) {
383
384			// Trigger audio interrupt to get new buffer
385			D(bug("stream: triggering irq\n"));
386			SetInterruptFlag(INTFLAG_AUDIO);
387			TriggerInterrupt();
388			D(bug("stream: waiting for ack\n"));
389			sem_wait(&audio_irq_done_sem);
390			D(bug("stream: ack received\n"));
391
392			// Get size of audio data
393			uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
394			if (apple_stream_info) {
395			int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
396			D(bug("stream: work_size %d\n", work_size));
397			if (work_size > sound_buffer_size)
398			work_size = sound_buffer_size;
399			if (work_size == 0)
400			goto silence;
401
402			// Send data to DSP
403			if (work_size == sound_buffer_size && !little_endian)
404			write(audio_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
405			else {
406			// Last buffer or little-endian DSP
407			if (little_endian) {
408			int16 p = (int16 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
409			for (int i=0; i<work_size/2; i++)
410			last_buffer[i] = ntohs(p[i]);
411			} else
412	cebix	1.3	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
413	cebix	1.2	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
414	cebix	1.1	write(audio_fd, last_buffer, sound_buffer_size);
415			}
416			D(bug("stream: data written\n"));
417			} else
418			goto silence;
419
420			} else {
421
422			// Audio not active, play silence
423			silence: write(audio_fd, silent_buffer, sound_buffer_size);
424			}
425			}
426			delete[] silent_buffer;
427			delete[] last_buffer;
428			return NULL;
429			}
430
431
432			/*
433			* MacOS audio interrupt, read next data block
434			*/
435
436			void AudioInterrupt(void)
437			{
438			D(bug("AudioInterrupt\n"));
439
440			// Get data from apple mixer
441			if (AudioStatus.mixer) {
442			M68kRegisters r;
443			r.a[0] = audio_data + adatStreamInfo;
444			r.a[1] = AudioStatus.mixer;
445			Execute68k(audio_data + adatGetSourceData, &r);
446			D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
447			} else
448			WriteMacInt32(audio_data + adatStreamInfo, 0);
449
450			// Signal stream function
451			sem_post(&audio_irq_done_sem);
452			D(bug("AudioInterrupt done\n"));
453			}
454
455
456			/*
457			* Set sampling parameters
458	cebix	1.7	* "index" is an index into the audio_sample_rates[] etc. vectors
459	cebix	1.1	* It is guaranteed that AudioStatus.num_sources == 0
460			*/
461
462	cebix	1.7	bool audio_set_sample_rate(int index)
463	cebix	1.1	{
464	cebix	1.7	close_audio();
465			audio_sample_rate_index = index;
466			return open_audio();
467	cebix	1.1	}
468
469	cebix	1.7	bool audio_set_sample_size(int index)
470	cebix	1.1	{
471	cebix	1.7	close_audio();
472			audio_sample_size_index = index;
473			return open_audio();
474	cebix	1.1	}
475
476	cebix	1.7	bool audio_set_channels(int index)
477	cebix	1.1	{
478	cebix	1.7	close_audio();
479			audio_channel_count_index = index;
480			return open_audio();
481	cebix	1.1	}
482
483
484			/*
485			* Get/set volume controls (volume values received/returned have the left channel
486			* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
487			* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
488			*/
489
490			bool audio_get_main_mute(void)
491			{
492			return false;
493			}
494
495			uint32 audio_get_main_volume(void)
496			{
497			if (mixer_fd >= 0) {
498			int vol;
499			if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
500			int left = vol >> 8;
501			int right = vol & 0xff;
502			return ((left * 256 / 100) << 16) \| (right * 256 / 100);
503			}
504			}
505			return 0x01000100;
506			}
507
508			bool audio_get_speaker_mute(void)
509			{
510			return false;
511			}
512
513			uint32 audio_get_speaker_volume(void)
514			{
515			if (mixer_fd >= 0) {
516			int vol;
517			if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
518			int left = vol >> 8;
519			int right = vol & 0xff;
520			return ((left * 256 / 100) << 16) \| (right * 256 / 100);
521			}
522			}
523			return 0x01000100;
524			}
525
526			void audio_set_main_mute(bool mute)
527			{
528			}
529
530			void audio_set_main_volume(uint32 vol)
531			{
532			if (mixer_fd >= 0) {
533			int left = vol >> 16;
534			int right = vol & 0xffff;
535			int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
536			ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
537			}
538			}
539
540			void audio_set_speaker_mute(bool mute)
541			{
542			}
543
544			void audio_set_speaker_volume(uint32 vol)
545			{
546			if (mixer_fd >= 0) {
547			int left = vol >> 16;
548			int right = vol & 0xffff;
549			int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
550			ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
551			}
552			}