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;

// Global variables
static int audio_fd = -1;                                                       // fd of dsp or ESD
static int mixer_fd = -1;                                                       // fd of 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 the dsp device, returns false on error
static bool open_dsp(void)
{
        // Open the device
        const char *dsp = PrefsFindString("dsp");
        audio_fd = open(dsp, O_WRONLY);
        if (audio_fd < 0) {
                fprintf(stderr, "WARNING: Cannot open %s (%s)\n", dsp, strerror(errno));
                return false;
        }

        printf("Using %s audio output\n", dsp);

        // 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, 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 dsp
        if (open_dsp())
                goto dev_opened;

#ifdef ENABLE_ESD
        // Hm, 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 the mixer device
        const char *mixer = PrefsFindString("mixer");
        mixer_fd = open(mixer, O_RDWR);
        if (mixer_fd < 0)
                printf("WARNING: Cannot open %s (%s)\n", mixer, 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 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 mixer device
        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:	1.15
Committed:	2002-10-15T16:25:04Z (21 years, 8 months ago) by cebix
Branch:	MAIN
CVS Tags:	nigel-build-12, nigel-build-13
Changes since 1.14:	+18 -19 lines
Log Message:	Unix: new prefs options "dsp" and "mixer" to set the OSS device names instead of the hardcoded '/dev/dsp' and '/dev/mixer'
#	Content
1	/*
2	* audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
3	*
4	* Basilisk II (C) 1997-2002 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	// Global variables
58	static int audio_fd = -1; // fd of dsp or ESD
59	static int mixer_fd = -1; // fd of mixer
60	static sem_t audio_irq_done_sem; // Signal from interrupt to streaming thread: data block read
61	static bool sem_inited = false; // Flag: audio_irq_done_sem initialized
62	static int sound_buffer_size; // Size of sound buffer in bytes
63	static bool little_endian = false; // Flag: DSP accepts only little-endian 16-bit sound data
64	static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
65	static pthread_t stream_thread; // Audio streaming thread
66	static pthread_attr_t stream_thread_attr; // Streaming thread attributes
67	static bool stream_thread_active = false; // Flag: streaming thread installed
68	static volatile bool stream_thread_cancel = false; // Flag: cancel streaming thread
69
70	// Prototypes
71	static void stream_func(void arg);
72
73
74	/*
75	* Initialization
76	*/
77
78	// Set AudioStatus to reflect current audio stream format
79	static void set_audio_status_format(void)
80	{
81	AudioStatus.sample_rate = audio_sample_rates[audio_sample_rate_index];
82	AudioStatus.sample_size = audio_sample_sizes[audio_sample_size_index];
83	AudioStatus.channels = audio_channel_counts[audio_channel_count_index];
84	}
85
86	// Init using the dsp device, returns false on error
87	static bool open_dsp(void)
88	{
89	// Open the device
90	const char *dsp = PrefsFindString("dsp");
91	audio_fd = open(dsp, O_WRONLY);
92	if (audio_fd < 0) {
93	fprintf(stderr, "WARNING: Cannot open %s (%s)\n", dsp, strerror(errno));
94	return false;
95	}
96
97	printf("Using %s audio output\n", dsp);
98
99	// Get supported sample formats
100	if (audio_sample_sizes.empty()) {
101	unsigned long format;
102	ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &format);
103	if (format & AFMT_U8)
104	audio_sample_sizes.push_back(8);
105	if (format & (AFMT_S16_BE \| AFMT_S16_LE))
106	audio_sample_sizes.push_back(16);
107
108	int stereo = 0;
109	if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 0)
110	audio_channel_counts.push_back(1);
111	stereo = 1;
112	if (ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo) == 0 && stereo == 1)
113	audio_channel_counts.push_back(2);
114
115	if (audio_sample_sizes.empty() \|\| audio_channel_counts.empty()) {
116	WarningAlert(GetString(STR_AUDIO_FORMAT_WARN));
117	close(audio_fd);
118	audio_fd = -1;
119	return false;
120	}
121
122	audio_sample_rates.push_back(11025 << 16);
123	audio_sample_rates.push_back(22050 << 16);
124	int rate = 44100;
125	ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
126	if (rate > 22050)
127	audio_sample_rates.push_back(rate << 16);
128
129	// Default to highest supported values
130	audio_sample_rate_index = audio_sample_rates.size() - 1;
131	audio_sample_size_index = audio_sample_sizes.size() - 1;
132	audio_channel_count_index = audio_channel_counts.size() - 1;
133	}
134
135	// Set DSP parameters
136	unsigned long format;
137	if (audio_sample_sizes[audio_sample_size_index] == 8) {
138	format = AFMT_U8;
139	little_endian = false;
140	silence_byte = 0x80;
141	} else {
142	unsigned long sup_format;
143	ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &sup_format);
144	if (sup_format & AFMT_S16_BE) {
145	little_endian = false;
146	format = AFMT_S16_BE;
147	} else {
148	little_endian = true;
149	format = AFMT_S16_LE;
150	}
151	silence_byte = 0;
152	}
153	ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
154	int frag = 0x0004000c; // Block size: 4096 frames
155	ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
156	int stereo = (audio_channel_counts[audio_channel_count_index] == 2);
157	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
158	int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
159	ioctl(audio_fd, SNDCTL_DSP_SPEED, &rate);
160
161	// Get sound buffer size
162	ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &audio_frames_per_block);
163	D(bug("DSP_GETBLKSIZE %d\n", audio_frames_per_block));
164	return true;
165	}
166
167	// Init using ESD, returns false on error
168	static bool open_esd(void)
169	{
170	#ifdef ENABLE_ESD
171	int rate;
172	esd_format_t format = ESD_STREAM \| ESD_PLAY;
173
174	if (audio_sample_sizes.empty()) {
175
176	// Default values
177	rate = 44100;
178	format \|= (ESD_BITS16 \| ESD_STEREO);
179
180	} else {
181
182	rate = audio_sample_rates[audio_sample_rate_index] >> 16;
183	if (audio_sample_sizes[audio_sample_size_index] == 8)
184	format \|= ESD_BITS8;
185	else
186	format \|= ESD_BITS16;
187	if (audio_channel_counts[audio_channel_count_index] == 1)
188	format \|= ESD_MONO;
189	else
190	format \|= ESD_STEREO;
191	}
192
193	#if WORDS_BIGENDIAN
194	little_endian = false;
195	#else
196	little_endian = true;
197	#endif
198	silence_byte = 0; // Is this correct for 8-bit mode?
199
200	// Open connection to ESD server
201	audio_fd = esd_play_stream(format, rate, NULL, NULL);
202	if (audio_fd < 0) {
203	fprintf(stderr, "WARNING: Cannot open ESD connection\n");
204	return false;
205	}
206
207	printf("Using ESD audio output\n");
208
209	// ESD supports a variety of twisted little audio formats, all different
210	if (audio_sample_sizes.empty()) {
211
212	// The reason we do this here is that we don't want to add sample
213	// rates etc. unless the ESD server connection could be opened
214	// (if ESD fails, dsp might be tried next)
215	audio_sample_rates.push_back(11025 << 16);
216	audio_sample_rates.push_back(22050 << 16);
217	audio_sample_rates.push_back(44100 << 16);
218	audio_sample_sizes.push_back(8);
219	audio_sample_sizes.push_back(16);
220	audio_channel_counts.push_back(1);
221	audio_channel_counts.push_back(2);
222
223	// Default to highest supported values
224	audio_sample_rate_index = audio_sample_rates.size() - 1;
225	audio_sample_size_index = audio_sample_sizes.size() - 1;
226	audio_channel_count_index = audio_channel_counts.size() - 1;
227	}
228
229	// Sound buffer size = 4096 frames
230	audio_frames_per_block = 4096;
231	return true;
232	#else
233	// ESD is not enabled, shut up the compiler
234	return false;
235	#endif
236	}
237
238	static bool open_audio(void)
239	{
240	#ifdef ENABLE_ESD
241	// If ESPEAKER is set, the user probably wants to use ESD, so try that first
242	if (getenv("ESPEAKER"))
243	if (open_esd())
244	goto dev_opened;
245	#endif
246
247	// Try to open dsp
248	if (open_dsp())
249	goto dev_opened;
250
251	#ifdef ENABLE_ESD
252	// Hm, dsp failed so we try ESD again if ESPEAKER wasn't set
253	if (!getenv("ESPEAKER"))
254	if (open_esd())
255	goto dev_opened;
256	#endif
257
258	// No audio device succeeded
259	WarningAlert(GetString(STR_NO_AUDIO_WARN));
260	return false;
261
262	// Device opened, set AudioStatus
263	dev_opened:
264	sound_buffer_size = (audio_sample_sizes[audio_sample_size_index] >> 3) * audio_channel_counts[audio_channel_count_index] * audio_frames_per_block;
265	set_audio_status_format();
266
267	// Start streaming thread
268	Set_pthread_attr(&stream_thread_attr, 0);
269	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
270
271	// Everything went fine
272	audio_open = true;
273	return true;
274	}
275
276	void AudioInit(void)
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 the mixer device
296	const char *mixer = PrefsFindString("mixer");
297	mixer_fd = open(mixer, O_RDWR);
298	if (mixer_fd < 0)
299	printf("WARNING: Cannot open %s (%s)\n", mixer, strerror(errno));
300
301	// Open and initialize audio device
302	open_audio();
303	}
304
305
306	/*
307	* Deinitialization
308	*/
309
310	static void close_audio(void)
311	{
312	// Stop stream and delete semaphore
313	if (stream_thread_active) {
314	stream_thread_cancel = true;
315	#ifdef HAVE_PTHREAD_CANCEL
316	pthread_cancel(stream_thread);
317	#endif
318	pthread_join(stream_thread, NULL);
319	stream_thread_active = false;
320	}
321
322	// Close dsp or ESD socket
323	if (audio_fd >= 0) {
324	close(audio_fd);
325	audio_fd = -1;
326	}
327
328	audio_open = false;
329	}
330
331	void AudioExit(void)
332	{
333	// Close audio device
334	close_audio();
335
336	// Delete semaphore
337	if (sem_inited) {
338	sem_destroy(&audio_irq_done_sem);
339	sem_inited = false;
340	}
341
342	// Close mixer device
343	if (mixer_fd >= 0) {
344	close(mixer_fd);
345	mixer_fd = -1;
346	}
347	}
348
349
350	/*
351	* First source added, start audio stream
352	*/
353
354	void audio_enter_stream()
355	{
356	// Streaming thread is always running to avoid clicking noises
357	}
358
359
360	/*
361	* Last source removed, stop audio stream
362	*/
363
364	void audio_exit_stream()
365	{
366	// Streaming thread is always running to avoid clicking noises
367	}
368
369
370	/*
371	* Streaming function
372	*/
373
374	static void stream_func(void arg)
375	{
376	int16 *silent_buffer = new int16[sound_buffer_size / 2];
377	int16 *last_buffer = new int16[sound_buffer_size / 2];
378	memset(silent_buffer, silence_byte, sound_buffer_size);
379
380	while (!stream_thread_cancel) {
381	if (AudioStatus.num_sources) {
382
383	// Trigger audio interrupt to get new buffer
384	D(bug("stream: triggering irq\n"));
385	SetInterruptFlag(INTFLAG_AUDIO);
386	TriggerInterrupt();
387	D(bug("stream: waiting for ack\n"));
388	sem_wait(&audio_irq_done_sem);
389	D(bug("stream: ack received\n"));
390
391	// Get size of audio data
392	uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
393	if (apple_stream_info) {
394	int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
395	D(bug("stream: work_size %d\n", work_size));
396	if (work_size > sound_buffer_size)
397	work_size = sound_buffer_size;
398	if (work_size == 0)
399	goto silence;
400
401	// Send data to DSP
402	if (work_size == sound_buffer_size && !little_endian)
403	write(audio_fd, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), sound_buffer_size);
404	else {
405	// Last buffer or little-endian DSP
406	if (little_endian) {
407	int16 p = (int16 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
408	for (int i=0; i<work_size/2; i++)
409	last_buffer[i] = ntohs(p[i]);
410	} else
411	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
412	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
413	write(audio_fd, last_buffer, sound_buffer_size);
414	}
415	D(bug("stream: data written\n"));
416	} else
417	goto silence;
418
419	} else {
420
421	// Audio not active, play silence
422	silence: write(audio_fd, silent_buffer, sound_buffer_size);
423	}
424	}
425	delete[] silent_buffer;
426	delete[] last_buffer;
427	return NULL;
428	}
429
430
431	/*
432	* MacOS audio interrupt, read next data block
433	*/
434
435	void AudioInterrupt(void)
436	{
437	D(bug("AudioInterrupt\n"));
438
439	// Get data from apple mixer
440	if (AudioStatus.mixer) {
441	M68kRegisters r;
442	r.a[0] = audio_data + adatStreamInfo;
443	r.a[1] = AudioStatus.mixer;
444	Execute68k(audio_data + adatGetSourceData, &r);
445	D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
446	} else
447	WriteMacInt32(audio_data + adatStreamInfo, 0);
448
449	// Signal stream function
450	sem_post(&audio_irq_done_sem);
451	D(bug("AudioInterrupt done\n"));
452	}
453
454
455	/*
456	* Set sampling parameters
457	* "index" is an index into the audio_sample_rates[] etc. vectors
458	* It is guaranteed that AudioStatus.num_sources == 0
459	*/
460
461	bool audio_set_sample_rate(int index)
462	{
463	close_audio();
464	audio_sample_rate_index = index;
465	return open_audio();
466	}
467
468	bool audio_set_sample_size(int index)
469	{
470	close_audio();
471	audio_sample_size_index = index;
472	return open_audio();
473	}
474
475	bool audio_set_channels(int index)
476	{
477	close_audio();
478	audio_channel_count_index = index;
479	return open_audio();
480	}
481
482
483	/*
484	* Get/set volume controls (volume values received/returned have the left channel
485	* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
486	* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
487	*/
488
489	bool audio_get_main_mute(void)
490	{
491	return false;
492	}
493
494	uint32 audio_get_main_volume(void)
495	{
496	if (mixer_fd >= 0) {
497	int vol;
498	if (ioctl(mixer_fd, SOUND_MIXER_READ_PCM, &vol) == 0) {
499	int left = vol >> 8;
500	int right = vol & 0xff;
501	return ((left * 256 / 100) << 16) \| (right * 256 / 100);
502	}
503	}
504	return 0x01000100;
505	}
506
507	bool audio_get_speaker_mute(void)
508	{
509	return false;
510	}
511
512	uint32 audio_get_speaker_volume(void)
513	{
514	if (mixer_fd >= 0) {
515	int vol;
516	if (ioctl(mixer_fd, SOUND_MIXER_READ_VOLUME, &vol) == 0) {
517	int left = vol >> 8;
518	int right = vol & 0xff;
519	return ((left * 256 / 100) << 16) \| (right * 256 / 100);
520	}
521	}
522	return 0x01000100;
523	}
524
525	void audio_set_main_mute(bool mute)
526	{
527	}
528
529	void audio_set_main_volume(uint32 vol)
530	{
531	if (mixer_fd >= 0) {
532	int left = vol >> 16;
533	int right = vol & 0xffff;
534	int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
535	ioctl(mixer_fd, SOUND_MIXER_WRITE_PCM, &p);
536	}
537	}
538
539	void audio_set_speaker_mute(bool mute)
540	{
541	}
542
543	void audio_set_speaker_volume(uint32 vol)
544	{
545	if (mixer_fd >= 0) {
546	int left = vol >> 16;
547	int right = vol & 0xffff;
548	int p = ((left * 100 / 256) << 8) \| (right * 100 / 256);
549	ioctl(mixer_fd, SOUND_MIXER_WRITE_VOLUME, &p);
550	}
551	}