src/Unix/audio_oss_esd.cpp

/*
 *  audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
 *
 *  Basilisk II (C) 1997-2001 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 bool formats_known = false;                                      // Flag: available audio formats have been probed
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 (!formats_known) {
                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;
                formats_known = true;
        }

        // 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, &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
        // ESD supports a variety of audio formats
        if (!formats_known) {
                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 44.1kHz, 16-bit, stereo
                audio_sample_rate_index = 2;
                audio_sample_size_index = 1;
                audio_channel_count_index = 1;
                formats_known = true;
        }

        // ESD audio format
        esd_format_t format = ESD_STREAM | ESD_PLAY;
        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, audio_sample_rates[audio_sample_rate_index] >> 16, NULL, NULL);
        if (audio_fd < 0) {
                fprintf(stderr, "WARNING: Cannot open ESD connection\n");
                formats_known = false;
                return false;
        }

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

        // Sound buffer size = 4096 frames
        audio_frames_per_block = 4096;
        return true;
#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
        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_cancel = false;
        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)
{
        char str[256];

        // 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)
{
        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 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. 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:	1.7
Committed:	2001-07-05T20:30:51Z (22 years, 11 months ago) by cebix
Branch:	MAIN
Changes since 1.6:	+178 -90 lines
Log Message:	- supported audio formats are now kept in STL vectors - added run-time audio parameter switching for OSS/ESD audio output - setting ESPEAKER env variable causes B2 to try ESD before OSS
#	Content
1	/*
2	* audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
3	*
4	* Basilisk II (C) 1997-2001 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	// 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
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 bool formats_known = false; // Flag: available audio formats have been probed
64	static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
65	static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
66	static int sound_buffer_size; // Size of sound buffer in bytes
67	static bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
68	static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
69	static pthread_t stream_thread; // Audio streaming thread
70	static pthread_attr_t stream_thread_attr; // Streaming thread attributes
71	static bool stream_thread_active = false; // Flag: streaming thread installed
72	static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
73
74	// Prototypes
75	static void stream_func(void arg);
76
77
78	/*
79	* Initialization
80	*/
81
82	// Set AudioStatus to reflect current audio stream format
83	static void set_audio_status_format(void)
84	{
85	AudioStatus.sample_rate = audio_sample_rates[audio_sample_rate_index];
86	AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
87	AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
88	}
89
90	// Init using /dev/dsp, returns false on error
91	static bool open_dsp(void)
92	{
93	// Open /dev/dsp
94	audio_fd = open(DSP_NAME, O_WRONLY);
95	if (audio_fd < 0) {
96	fprintf(stderr, "WARNING: Cannot open %s (%s)\n", DSP_NAME, strerror(errno));
97	return false;
98	}
99
100	printf("Using " DSP_NAME " audio output\n");
101
102	// Get supported sample formats
103	if (!formats_known) {
104	unsigned long format;
105	ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
106	if (format & AFMT_U8)
107	audio_sample_sizes.push_back(8);
108	if (format & (AFMT_S16_BE \| AFMT_S16_LE))
109	audio_sample_sizes.push_back(16);
110
111	int stereo = 0;
112	if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 0)
113	audio_channel_counts.push_back(1);
114	stereo = 1;
115	if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 1)
116	audio_channel_counts.push_back(2);
117
118	if (audio_sample_sizes.empty() \|\| audio_channel_counts.empty()) {
119	WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
120	close(audio_fd);
121	audio_fd = -1;
122	return false;
123	}
124
125	audio_sample_rates.push_back(11025 << 16);
126	audio_sample_rates.push_back(22050 << 16);
127	int rate = 44100;
128	ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
129	if (rate > 22050)
130	audio_sample_rates.push_back(rate << 16);
131
132	// Default to highest supported values
133	audio_sample_rate_index = audio_sample_rates.size() - 1;
134	audio_sample_size_index = audio_sample_sizes.size() - 1;
135	audio_channel_count_index = audio_channel_counts.size() - 1;
136	formats_known = true;
137	}
138
139	// Set DSP parameters
140	unsigned long format;
141	if (audio_sample_sizes[audio_sample_size_index] == 8) {
142	format = AFMT_U8;
143	little_endian = false;
144	silence_byte = 0x80;
145	} else {
146	unsigned long sup_format;
147	ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
148	if (sup_format & AFMT_S16_BE) {
149	little_endian = false;
150	format = AFMT_S16_BE;
151	} else {
152	little_endian = true;
153	format = AFMT_S16_LE;
154	}
155	silence_byte = 0;
156	}
157	ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
158	int frag = 0x0004000c; // Block size: 4096 frames
159	ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
160	int stereo = (audio_channel_counts[audio_channel_count_index] == 2);
161	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
162	int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
163	ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
164
165	// Get sound buffer size
166	ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
167	D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
168	return true;
169	}
170
171	// Init using ESD, returns false on error
172	static bool open_esd(void)
173	{
174	#ifdef ENABLE_ESD
175	// ESD supports a variety of audio formats
176	if (!formats_known) {
177	audio_sample_rates.push_back(11025 << 16);
178	audio_sample_rates.push_back(22050 << 16);
179	audio_sample_rates.push_back(44100 << 16);
180	audio_sample_sizes.push_back(8);
181	audio_sample_sizes.push_back(16);
182	audio_channel_counts.push_back(1);
183	audio_channel_counts.push_back(2);
184
185	// Default to 44.1kHz, 16-bit, stereo
186	audio_sample_rate_index = 2;
187	audio_sample_size_index = 1;
188	audio_channel_count_index = 1;
189	formats_known = true;
190	}
191
192	// ESD audio format
193	esd_format_t format = ESD_STREAM \| ESD_PLAY;
194	if (audio_sample_sizes[audio_sample_size_index] == 8)
195	format \|= ESD_BITS8;
196	else
197	format \|= ESD_BITS16;
198	if (audio_channel_counts[audio_channel_count_index] == 1)
199	format \|= ESD_MONO;
200	else
201	format \|= ESD_STEREO;
202
203	#if WORDS_BIGENDIAN
204	little_endian = false;
205	#else
206	little_endian = true;
207	#endif
208	silence_byte = 0; // Is this correct for 8-bit mode?
209
210	// Open connection to ESD server
211	audio_fd = esd_play_stream(format, audio_sample_rates[audio_sample_rate_index] >> 16, NULL, NULL);
212	if (audio_fd < 0) {
213	fprintf(stderr, "WARNING: Cannot open ESD connection\n");
214	formats_known = false;
215	return false;
216	}
217
218	printf("Using ESD audio output\n");
219
220	// Sound buffer size = 4096 frames
221	audio_frames_per_block = 4096;
222	return true;
223	#endif
224	}
225
226	static bool open_audio(void)
227	{
228	#ifdef ENABLE_ESD
229	// If ESPEAKER is set, the user probably wants to use ESD, so try that first
230	if (getenv("ESPEAKER"))
231	if (open_esd())
232	goto dev_opened;
233	#endif
234
235	// Try to open /dev/dsp
236	if (open_dsp())
237	goto dev_opened;
238
239	#ifdef ENABLE_ESD
240	// Hm, /dev/dsp failed so we try ESD again if ESPEAKER wasn't set
241	if (!getenv("ESPEAKER"))
242	if (open_esd())
243	goto dev_opened;
244	#endif
245
246	// No audio device succeeded
247	WarningAlert(GetString(STR_NO_AUDIO_WARN));
248	return false;
249
250	// Device opened, set AudioStatus
251	dev_opened:
252	sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) * audio_channel_counts[audio_channel_count_index] * audio_frames_per_block;
253	set_audio_status_format();
254
255	// Start streaming thread
256	pthread_attr_init(&stream_thread_attr);
257	#if defined(_POSIX_THREAD_PRIORITY_SCHEDULING)
258	if (geteuid() == 0) {
259	pthread_attr_setinheritsched(&stream_thread_attr, PTHREAD_EXPLICIT_SCHED);
260	pthread_attr_setschedpolicy(&stream_thread_attr, SCHED_FIFO);
261	struct sched_param fifo_param;
262	fifo_param.sched_priority = (sched_get_priority_min(SCHED_FIFO) + sched_get_priority_max(SCHED_FIFO)) / 2;
263	pthread_attr_setschedparam(&stream_thread_attr, &fifo_param);
264	}
265	#endif
266	stream_thread_cancel = false;
267	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
268
269	// Everything went fine
270	audio_open = true;
271	return true;
272	}
273
274	void AudioInit(void)
275	{
276	char str[256];
277
278	// Init audio status (reasonable defaults) and feature flags
279	AudioStatus.sample_rate = 44100 << 16;
280	AudioStatus.sample_size = 16;
281	AudioStatus.channels = 2;
282	AudioStatus.mixer = 0;
283	AudioStatus.num_sources = 0;
284	audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
285
286	// Sound disabled in prefs? Then do nothing
287	if (PrefsFindBool("nosound"))
288	return;
289
290	// Init semaphore
291	if (sem_init(&audio_irq_done_sem, 0, 0) < 0)
292	return;
293	sem_inited = true;
294
295	// Try to open /dev/mixer
296	mixer_fd = open("/dev/mixer", O_RDWR);
297	if (mixer_fd < 0)
298	printf("WARNING: Cannot open /dev/mixer (%s)", strerror(errno));
299
300	// Open and initialize audio device
301	open_audio();
302	}
303
304
305	/*
306	* Deinitialization
307	*/
308
309	static void close_audio(void)
310	{
311	// Stop stream and delete semaphore
312	if (stream_thread_active) {
313	stream_thread_cancel = true;
314	#ifdef HAVE_PTHREAD_CANCEL
315	pthread_cancel(stream_thread);
316	#endif
317	pthread_join(stream_thread, NULL);
318	stream_thread_active = false;
319	}
320
321	// Close /dev/dsp or ESD socket
322	if (audio_fd >= 0) {
323	close(audio_fd);
324	audio_fd = -1;
325	}
326
327	audio_open = false;
328	}
329
330	void AudioExit(void)
331	{
332	if (sem_inited) {
333	sem_destroy(&audio_irq_done_sem);
334	sem_inited = false;
335	}
336
337	// Close /dev/mixer
338	if (mixer_fd >= 0) {
339	close(mixer_fd);
340	mixer_fd = -1;
341	}
342	}
343
344
345	/*
346	* First source added, start audio stream
347	*/
348
349	void audio_enter_stream()
350	{
351	// Streaming thread is always running to avoid clicking noises
352	}
353
354
355	/*
356	* Last source removed, stop audio stream
357	*/
358
359	void audio_exit_stream()
360	{
361	// Streaming thread is always running to avoid clicking noises
362	}
363
364
365	/*
366	* Streaming function
367	*/
368
369	static uint32 apple_stream_info; // Mac address of SoundComponentData struct describing next buffer
370
371	static void stream_func(void arg)
372	{
373	int16 *silent_buffer = new int16[sound_buffer_size / 2];
374	int16 *last_buffer = new int16[sound_buffer_size / 2];
375	memset(silent_buffer, silence_byte, sound_buffer_size);
376
377	while (!stream_thread_cancel) {
378	if (AudioStatus.num_sources) {
379
380	// Trigger audio interrupt to get new buffer
381	D(bug("stream: triggering irq\n"));
382	SetInterruptFlag(INTFLAG_AUDIO);
383	TriggerInterrupt();
384	D(bug("stream: waiting for ack\n"));
385	sem_wait(&audio_irq_done_sem);
386	D(bug("stream: ack received\n"));
387
388	// Get size of audio data
389	uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
390	if (apple_stream_info) {
391	int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
392	D(bug("stream: work_size %d\n", work_size));
393	if (work_size > sound_buffer_size)
394	work_size = sound_buffer_size;
395	if (work_size == 0)
396	goto silence;
397
398	// Send data to DSP
399	if (work_size == sound_buffer_size && !little_endian)
400	write(audio_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
401	else {
402	// Last buffer or little-endian DSP
403	if (little_endian) {
404	int16 p = (int16 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
405	for (int i=0; i<work_size/2; i++)
406	last_buffer[i] = ntohs(p[i]);
407	} else
408	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
409	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
410	write(audio_fd, last_buffer, sound_buffer_size);
411	}
412	D(bug("stream: data written\n"));
413	} else
414	goto silence;
415
416	} else {
417
418	// Audio not active, play silence
419	silence: write(audio_fd, silent_buffer, sound_buffer_size);
420	}
421	}
422	delete[] silent_buffer;
423	delete[] last_buffer;
424	return NULL;
425	}
426
427
428	/*
429	* MacOS audio interrupt, read next data block
430	*/
431
432	void AudioInterrupt(void)
433	{
434	D(bug("AudioInterrupt\n"));
435
436	// Get data from apple mixer
437	if (AudioStatus.mixer) {
438	M68kRegisters r;
439	r.a[0] = audio_data + adatStreamInfo;
440	r.a[1] = AudioStatus.mixer;
441	Execute68k(audio_data + adatGetSourceData, &r);
442	D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
443	} else
444	WriteMacInt32(audio_data + adatStreamInfo, 0);
445
446	// Signal stream function
447	sem_post(&audio_irq_done_sem);
448	D(bug("AudioInterrupt done\n"));
449	}
450
451
452	/*
453	* Set sampling parameters
454	* "index" is an index into the audio_sample_rates[] etc. vectors
455	* It is guaranteed that AudioStatus.num_sources == 0
456	*/
457
458	bool audio_set_sample_rate(int index)
459	{
460	close_audio();
461	audio_sample_rate_index = index;
462	return open_audio();
463	}
464
465	bool audio_set_sample_size(int index)
466	{
467	close_audio();
468	audio_sample_size_index = index;
469	return open_audio();
470	}
471
472	bool audio_set_channels(int index)
473	{
474	close_audio();
475	audio_channel_count_index = index;
476	return open_audio();
477	}
478
479
480	/*
481	* Get/set volume controls (volume values received/returned have the left channel
482	* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
483	* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
484	*/
485
486	bool audio_get_main_mute(void)
487	{
488	return false;
489	}
490
491	uint32 audio_get_main_volume(void)
492	{
493	if (mixer_fd >= 0) {
494	int vol;
495	if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
496	int left = vol >> 8;
497	int right = vol & 0xff;
498	return ((left * 256 / 100) << 16) \| (right * 256 / 100);
499	}
500	}
501	return 0x01000100;
502	}
503
504	bool audio_get_speaker_mute(void)
505	{
506	return false;
507	}
508
509	uint32 audio_get_speaker_volume(void)
510	{
511	if (mixer_fd >= 0) {
512	int vol;
513	if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
514	int left = vol >> 8;
515	int right = vol & 0xff;
516	return ((left * 256 / 100) << 16) \| (right * 256 / 100);
517	}
518	}
519	return 0x01000100;
520	}
521
522	void audio_set_main_mute(bool mute)
523	{
524	}
525
526	void audio_set_main_volume(uint32 vol)
527	{
528	if (mixer_fd >= 0) {
529	int left = vol >> 16;
530	int right = vol & 0xffff;
531	int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
532	ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
533	}
534	}
535
536	void audio_set_speaker_mute(bool mute)
537	{
538	}
539
540	void audio_set_speaker_volume(uint32 vol)
541	{
542	if (mixer_fd >= 0) {
543	int left = vol >> 16;
544	int right = vol & 0xffff;
545	int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
546	ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
547	}
548	}