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 (22 years, 1 month 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'
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_oss_esd.cpp - Audio support, implementation for OSS and ESD (Linux and FreeBSD)
3			*
4	cebix	1.12	* Basilisk II (C) 1997-2002 Christian Bauer
5	cebix	1.1	*
6			* This program is free software; you can redistribute it and/or modify
7			* it under the terms of the GNU General Public License as published by
8			* the Free Software Foundation; either version 2 of the License, or
9			* (at your option) any later version.
10			*
11			* This program is distributed in the hope that it will be useful,
12			* but WITHOUT ANY WARRANTY; without even the implied warranty of
13			* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14			* GNU General Public License for more details.
15			*
16			* You should have received a copy of the GNU General Public License
17			* along with this program; if not, write to the Free Software
18			* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
19			*/
20
21			#include "sysdeps.h"
22
23			#include <sys/ioctl.h>
24			#include <unistd.h>
25			#include <errno.h>
26			#include <pthread.h>
27			#include <semaphore.h>
28
29			#ifdef __linux__
30			#include <linux/soundcard.h>
31			#endif
32
33			#ifdef __FreeBSD__
34	cebix	1.15	#include <machine/soundcard.h>
35	cebix	1.1	#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			// Global variables
58	cebix	1.15	static int audio_fd = -1; // fd of dsp or ESD
59			static int mixer_fd = -1; // fd of mixer
60	cebix	1.1	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	cebix	1.2	static uint8 silence_byte; // Byte value to use to fill sound buffers with silence
65	cebix	1.1	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	cebix	1.2	// Set AudioStatus to reflect current audio stream format
79			static void set_audio_status_format(void)
80			{
81	cebix	1.7	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	cebix	1.2	}
85
86	cebix	1.15	// Init using the dsp device, returns false on error
87	cebix	1.7	static bool open_dsp(void)
88	cebix	1.1	{
89	cebix	1.15	// Open the device
90			const char *dsp = PrefsFindString("dsp");
91			audio_fd = open(dsp, O_WRONLY);
92	cebix	1.7	if (audio_fd < 0) {
93	cebix	1.15	fprintf(stderr, "WARNING: Cannot open %s (%s)\n", dsp, strerror(errno));
94	cebix	1.7	return false;
95			}
96
97	cebix	1.15	printf("Using %s audio output\n", dsp);
98	cebix	1.1
99			// Get supported sample formats
100	cebix	1.8	if (audio_sample_sizes.empty()) {
101	cebix	1.7	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	cebix	1.1	unsigned long format;
137	cebix	1.7	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	cebix	1.9	ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &sup_format);
144	cebix	1.7	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	cebix	1.2	silence_byte = 0;
152			}
153	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_SETFMT, &format);
154			int frag = 0x0004000c; // Block size: 4096 frames
155			ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &frag);
156	cebix	1.7	int stereo = (audio_channel_counts[audio_channel_count_index] == 2);
157	cebix	1.1	ioctl(audio_fd, SNDCTL_DSP_STEREO, &stereo);
158	cebix	1.7	int rate = audio_sample_rates[audio_sample_rate_index] >> 16;
159	cebix	1.1	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	cebix	1.7	static bool open_esd(void)
169	cebix	1.1	{
170	cebix	1.5	#ifdef ENABLE_ESD
171	cebix	1.8	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	cebix	1.7
182	cebix	1.8	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	cebix	1.7	}
192	cebix	1.1
193			#if WORDS_BIGENDIAN
194			little_endian = false;
195			#else
196			little_endian = true;
197			#endif
198	cebix	1.2	silence_byte = 0; // Is this correct for 8-bit mode?
199	cebix	1.1
200			// Open connection to ESD server
201	cebix	1.8	audio_fd = esd_play_stream(format, rate, NULL, NULL);
202	cebix	1.1	if (audio_fd < 0) {
203	cebix	1.7	fprintf(stderr, "WARNING: Cannot open ESD connection\n");
204	cebix	1.1	return false;
205			}
206
207	cebix	1.7	printf("Using ESD audio output\n");
208	cebix	1.8
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	cebix	1.15	// (if ESD fails, dsp might be tried next)
215	cebix	1.8	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	cebix	1.7
229	cebix	1.1	// Sound buffer size = 4096 frames
230			audio_frames_per_block = 4096;
231			return true;
232	gbeauche	1.11	#else
233			// ESD is not enabled, shut up the compiler
234			return false;
235	cebix	1.1	#endif
236			}
237
238	cebix	1.7	static bool open_audio(void)
239	cebix	1.1	{
240	cebix	1.7	#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	cebix	1.1
247	cebix	1.15	// Try to open dsp
248	cebix	1.7	if (open_dsp())
249			goto dev_opened;
250	cebix	1.1
251	cebix	1.5	#ifdef ENABLE_ESD
252	cebix	1.15	// Hm, dsp failed so we try ESD again if ESPEAKER wasn't set
253	cebix	1.7	if (!getenv("ESPEAKER"))
254			if (open_esd())
255			goto dev_opened;
256	cebix	1.1	#endif
257
258	cebix	1.7	// No audio device succeeded
259			WarningAlert(GetString(STR_NO_AUDIO_WARN));
260			return false;
261	cebix	1.1
262	cebix	1.7	// 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	cebix	1.1
267			// Start streaming thread
268	cebix	1.13	Set_pthread_attr(&stream_thread_attr, 0);
269	cebix	1.1	stream_thread_active = (pthread_create(&stream_thread, &stream_thread_attr, stream_func, NULL) == 0);
270
271	cebix	1.7	// Everything went fine
272	cebix	1.1	audio_open = true;
273	cebix	1.7	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	cebix	1.15	// Try to open the mixer device
296			const char *mixer = PrefsFindString("mixer");
297			mixer_fd = open(mixer, O_RDWR);
298	cebix	1.7	if (mixer_fd < 0)
299	cebix	1.15	printf("WARNING: Cannot open %s (%s)\n", mixer, strerror(errno));
300	cebix	1.7
301			// Open and initialize audio device
302			open_audio();
303	cebix	1.1	}
304
305
306			/*
307			* Deinitialization
308			*/
309
310	cebix	1.7	static void close_audio(void)
311	cebix	1.1	{
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	cebix	1.15	// Close dsp or ESD socket
323	cebix	1.7	if (audio_fd >= 0) {
324	cebix	1.1	close(audio_fd);
325	cebix	1.7	audio_fd = -1;
326			}
327
328			audio_open = false;
329			}
330
331			void AudioExit(void)
332			{
333	cebix	1.10	// Close audio device
334			close_audio();
335
336			// Delete semaphore
337	cebix	1.7	if (sem_inited) {
338			sem_destroy(&audio_irq_done_sem);
339			sem_inited = false;
340			}
341	cebix	1.1
342	cebix	1.15	// Close mixer device
343	cebix	1.7	if (mixer_fd >= 0) {
344	cebix	1.1	close(mixer_fd);
345	cebix	1.7	mixer_fd = -1;
346			}
347	cebix	1.1	}
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	cebix	1.2	memset(silent_buffer, silence_byte, sound_buffer_size);
379	cebix	1.1
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	cebix	1.3	Mac2Host_memcpy(last_buffer, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
412	cebix	1.2	memset((uint8 *)last_buffer + work_size, silence_byte, sound_buffer_size - work_size);
413	cebix	1.1	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	cebix	1.7	* "index" is an index into the audio_sample_rates[] etc. vectors
458	cebix	1.1	* It is guaranteed that AudioStatus.num_sources == 0
459			*/
460
461	cebix	1.7	bool audio_set_sample_rate(int index)
462	cebix	1.1	{
463	cebix	1.7	close_audio();
464			audio_sample_rate_index = index;
465			return open_audio();
466	cebix	1.1	}
467
468	cebix	1.7	bool audio_set_sample_size(int index)
469	cebix	1.1	{
470	cebix	1.7	close_audio();
471			audio_sample_size_index = index;
472			return open_audio();
473	cebix	1.1	}
474
475	cebix	1.7	bool audio_set_channels(int index)
476	cebix	1.1	{
477	cebix	1.7	close_audio();
478			audio_channel_count_index = index;
479			return open_audio();
480	cebix	1.1	}
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			}