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 (23 years, 2 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
#	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	1.6	* Basilisk II (C) 1997-2001 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	cebix	1.7	static bool formats_known = false; // Flag: available audio formats have been probed
64	cebix	1.1	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	cebix	1.2	static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
69	cebix	1.1	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	cebix	1.2	// Set AudioStatus to reflect current audio stream format
83			static void set_audio_status_format(void)
84			{
85	cebix	1.7	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	cebix	1.2	}
89
90	cebix	1.1	// Init using /dev/dsp, returns false on error
91	cebix	1.7	static bool open_dsp(void)
92	cebix	1.1	{
93	cebix	1.7	// 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	cebix	1.1	printf("Using " DSP_NAME " audio output\n");
101
102			// Get supported sample formats
103	cebix	1.7	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	cebix	1.1	unsigned long format;
141	cebix	1.7	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	cebix	1.2	silence_byte = 0;
156			}
157	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
158			int frag = 0x0004000c; // Block size: 4096 frames
159			ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
160	cebix	1.7	int stereo = (audio_channel_counts[audio_channel_count_index] == 2);
161	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
162	cebix	1.7	int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
163	cebix	1.1	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	cebix	1.7	static bool open_esd(void)
173	cebix	1.1	{
174	cebix	1.5	#ifdef ENABLE_ESD
175	cebix	1.7	// 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	cebix	1.1
192			// ESD audio format
193			esd_format_t format = ESD_STREAM \| ESD_PLAY;
194	cebix	1.7	if (audio_sample_sizes[audio_sample_size_index] == 8)
195	cebix	1.1	format \|= ESD_BITS8;
196			else
197			format \|= ESD_BITS16;
198	cebix	1.7	if (audio_channel_counts[audio_channel_count_index] == 1)
199	cebix	1.1	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	cebix	1.2	silence_byte = 0; // Is this correct for 8-bit mode?
209	cebix	1.1
210			// Open connection to ESD server
211	cebix	1.7	audio_fd = esd_play_stream(format, audio_sample_rates[audio_sample_rate_index] >> 16, NULL, NULL);
212	cebix	1.1	if (audio_fd < 0) {
213	cebix	1.7	fprintf(stderr, "WARNING: Cannot open ESD connection\n");
214			formats_known = false;
215	cebix	1.1	return false;
216			}
217
218	cebix	1.7	printf("Using ESD audio output\n");
219
220	cebix	1.1	// Sound buffer size = 4096 frames
221			audio_frames_per_block = 4096;
222			return true;
223			#endif
224			}
225
226	cebix	1.7	static bool open_audio(void)
227	cebix	1.1	{
228	cebix	1.7	#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	cebix	1.1
235	cebix	1.7	// Try to open /dev/dsp
236			if (open_dsp())
237			goto dev_opened;
238	cebix	1.1
239	cebix	1.5	#ifdef ENABLE_ESD
240	cebix	1.7	// 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	cebix	1.1	#endif
245
246	cebix	1.7	// No audio device succeeded
247			WarningAlert(GetString(STR_NO_AUDIO_WARN));
248			return false;
249	cebix	1.1
250	cebix	1.7	// 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	cebix	1.1
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	cebix	1.7	stream_thread_cancel = false;
267	cebix	1.1	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
268
269	cebix	1.7	// Everything went fine
270	cebix	1.1	audio_open = true;
271	cebix	1.7	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	cebix	1.1	}
303
304
305			/*
306			* Deinitialization
307			*/
308
309	cebix	1.7	static void close_audio(void)
310	cebix	1.1	{
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	cebix	1.7	// Close /dev/dsp or ESD socket
322			if (audio_fd >= 0) {
323	cebix	1.1	close(audio_fd);
324	cebix	1.7	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	cebix	1.1
337			// Close /dev/mixer
338	cebix	1.7	if (mixer_fd >= 0) {
339	cebix	1.1	close(mixer_fd);
340	cebix	1.7	mixer_fd = -1;
341			}
342	cebix	1.1	}
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	cebix	1.2	memset(silent_buffer, silence_byte, sound_buffer_size);
376	cebix	1.1
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	cebix	1.3	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
409	cebix	1.2	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
410	cebix	1.1	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	cebix	1.7	* "index" is an index into the audio_sample_rates[] etc. vectors
455	cebix	1.1	* It is guaranteed that AudioStatus.num_sources == 0
456			*/
457
458	cebix	1.7	bool audio_set_sample_rate(int index)
459	cebix	1.1	{
460	cebix	1.7	close_audio();
461			audio_sample_rate_index = index;
462			return open_audio();
463	cebix	1.1	}
464
465	cebix	1.7	bool audio_set_sample_size(int index)
466	cebix	1.1	{
467	cebix	1.7	close_audio();
468			audio_sample_size_index = index;
469			return open_audio();
470	cebix	1.1	}
471
472	cebix	1.7	bool audio_set_channels(int index)
473	cebix	1.1	{
474	cebix	1.7	close_audio();
475			audio_channel_count_index = index;
476			return open_audio();
477	cebix	1.1	}
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			}