src/AmigaOS/audio_amiga.cpp

/*
 *  audio_amiga.cpp - Audio support, AmigaOS implementation using AHI
 *
 *  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 <exec/types.h>
#include <exec/memory.h>
#include <devices/ahi.h>
#define __USE_SYSBASE
#include <proto/exec.h>
#include <proto/ahi.h>
#include <inline/exec.h>
#include <inline/ahi.h>

#include "cpu_emulation.h"
#include "main.h"
#include "prefs.h"
#include "user_strings.h"
#include "audio.h"
#include "audio_defs.h"

#define DEBUG 0
#include "debug.h"

#define D1(x) ;


// Global variables
static ULONG ahi_id = AHI_DEFAULT_ID;                   // AHI audio ID
static struct AHIAudioCtrl *ahi_ctrl = NULL;
static struct AHISampleInfo sample[2];                  // Two sample infos for double-buffering
static struct Hook sf_hook;
static int play_buf = 0;                                                // Number of currently played buffer
static long sound_buffer_size;                                  // Size of one audio buffer in bytes
static int audio_block_fetched = 0;                             // Number of audio blocks fetched by interrupt routine

static bool main_mute = false;
static bool speaker_mute = false;
static ULONG supports_volume_changes = false;
static ULONG supports_stereo_panning = false;
static ULONG current_main_volume;
static ULONG current_speaker_volume;


// Prototypes
static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/);
void audio_set_sample_rate_byval(uint32 value);
void audio_set_sample_size_byval(uint32 value);
void audio_set_channels_byval(uint32 value);


/*
 *  Initialization
 */

// Set AudioStatus to reflect current audio stream format
static void set_audio_status_format(int sample_rate_index)
{
        AudioStatus.sample_rate = audio_sample_rates[sample_rate_index];
        AudioStatus.sample_size = audio_sample_sizes[0];
        AudioStatus.channels = audio_channel_counts[0];
}

void AudioInit(void)
{
        sample[0].ahisi_Address = sample[1].ahisi_Address = NULL;

        // Init audio status and feature flags
        audio_channel_counts.push_back(2);
//      set_audio_status_format();
        AudioStatus.mixer = 0;
        AudioStatus.num_sources = 0;
        audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut;

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

        // AHI available?
        if (AHIBase == NULL) {
                WarningAlert(GetString(STR_NO_AHI_WARN));
                return;
        }

        // Initialize callback hook
        sf_hook.h_Entry = (HOOKFUNC)audio_callback;

        // Read "sound" preferences
        const char *str = PrefsFindString("sound");
        if (str)
                sscanf(str, "ahi/%08lx", &ahi_id);

        // Open audio control structure
        if ((ahi_ctrl = AHI_AllocAudio(
                AHIA_AudioID, ahi_id,
                AHIA_MixFreq, AudioStatus.sample_rate >> 16,
                AHIA_Channels, 1,
                AHIA_Sounds, 2,
                AHIA_SoundFunc, (ULONG)&sf_hook,
                TAG_END)) == NULL) {
                WarningAlert(GetString(STR_NO_AHI_CTRL_WARN));
                return;
        }

        ULONG max_channels, sample_rate, frequencies, sample_rate_index;

        AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
                AHIDB_MaxChannels, (ULONG) &max_channels,
                AHIDB_Frequencies, (ULONG) &frequencies,
                TAG_END);

        D(bug("AudioInit: max_channels=%ld frequencies=%ld\n", max_channels, frequencies));

        for (int n=0; n<frequencies; n++)
                {
                AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
                        AHIDB_FrequencyArg, n,
                        AHIDB_Frequency, (ULONG) &sample_rate,
                        TAG_END);

                D(bug("AudioInit: f=%ld Hz\n", sample_rate));
                audio_sample_rates.push_back(sample_rate << 16);
                }

        ULONG sample_size_bits = 16;

        D(bug("AudioInit: sampe_rates=%ld\n", audio_sample_rates.size() ));

        // get index of sample rate closest to 22050 Hz
        AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
                AHIDB_IndexArg, 22050,
                AHIDB_Bits, (ULONG) &sample_size_bits,
                AHIDB_Index, (ULONG) &sample_rate_index,
                AHIDB_Volume, (ULONG) &supports_volume_changes,
                AHIDB_Panning, (ULONG) &supports_stereo_panning,
                TAG_END);

        audio_sample_sizes.push_back(16);

        set_audio_status_format(sample_rate_index);

        // 2048 frames per block
        audio_frames_per_block = 2048;
        sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;

        // Prepare SampleInfos and load sounds (two sounds for double buffering)
        sample[0].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
        sample[0].ahisi_Length = audio_frames_per_block;
        sample[0].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC | MEMF_CLEAR);
        sample[1].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
        sample[1].ahisi_Length = audio_frames_per_block;
        sample[1].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC | MEMF_CLEAR);
        if (sample[0].ahisi_Address == NULL || sample[1].ahisi_Address == NULL)
                return;

        AHI_LoadSound(0, AHIST_DYNAMICSAMPLE, &sample[0], ahi_ctrl);
        AHI_LoadSound(1, AHIST_DYNAMICSAMPLE, &sample[1], ahi_ctrl);

        // Set parameters
        play_buf = 0;
        current_main_volume = current_speaker_volume = 0x10000;
        AHI_SetVol(0, current_speaker_volume, 0x8000, ahi_ctrl, AHISF_IMM);

        AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, AHISF_IMM);
        AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, AHISF_IMM);

        // Everything OK
        audio_open = true;
}


/*
 *  Deinitialization
 */

void AudioExit(void)
{
        // Free everything
        if (ahi_ctrl != NULL) {
                AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
                AHI_FreeAudio(ahi_ctrl);
        }

        FreeVec(sample[0].ahisi_Address);
        FreeVec(sample[1].ahisi_Address);
}


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

void audio_enter_stream()
{
        AHI_ControlAudio(ahi_ctrl, AHIC_Play, TRUE, TAG_END);
}


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

void audio_exit_stream()
{
        AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
}


/*
 *  AHI sound callback, request next buffer
 */

static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/)
{
        play_buf ^= 1;

        // New buffer available?
        if (audio_block_fetched)
                {
                audio_block_fetched--;

                if (main_mute || speaker_mute)
                        {
                        memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size);
                        }
                else
                        {
                        // Get size of audio data
                        uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
                        if (apple_stream_info) {
                                int32 sample_count = ReadMacInt32(apple_stream_info + scd_sampleCount);

                                uint32 num_channels = ReadMacInt16(apple_stream_info + scd_numChannels);
                                uint32 sample_size = ReadMacInt16(apple_stream_info + scd_sampleSize);
                                uint32 sample_rate = ReadMacInt32(apple_stream_info + scd_sampleRate);

                                D(bug("stream: sample_count=%ld  num_channels=%ld  sample_size=%ld  sample_rate=%ld\n", sample_count, num_channels, sample_size, sample_rate >> 16));

                                // Yes, this can happen.
                                if(sample_count != 0) {
                                        if(sample_rate != AudioStatus.sample_rate) {
                                                audio_set_sample_rate_byval(sample_rate);
                                        }
                                        if(num_channels != AudioStatus.channels) {
                                                audio_set_channels_byval(num_channels);
                                        }
                                        if(sample_size != AudioStatus.sample_size) {
                                                audio_set_sample_size_byval(sample_size);
                                        }
                                }

                                if (sample_count < 0)
                                        sample_count = 0;

                                int work_size = sample_count * num_channels * (sample_size>>3);
                                D(bug("stream: work_size=%ld  sound_buffer_size=%ld\n", work_size, sound_buffer_size));

                                if (work_size > sound_buffer_size)
                                        work_size = sound_buffer_size;

                                // Put data into AHI buffer (convert 8-bit data unsigned->signed)
                                if (AudioStatus.sample_size == 16)
                                        Mac2Host_memcpy(sample[play_buf].ahisi_Address, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
                                else {
                                        uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
                                        uint32 *q = (uint32 *)sample[play_buf].ahisi_Address;
                                        int r = work_size >> 2;
                                        while (r--)
                                                *q++ = *p++ ^ 0x80808080;
                                }
                                if (work_size != sound_buffer_size)
                                        memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size);
                        }
                }

                }
        else
                memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size);

        // Play next buffer
        AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, 0);

        // Trigger audio interrupt to get new buffer
        if (AudioStatus.num_sources) {
                D1(bug("stream: triggering irq\n"));
                SetInterruptFlag(INTFLAG_AUDIO);
                TriggerInterrupt();
        }
        return 0;
}


/*
 *  MacOS audio interrupt, read next data block
 */

void AudioInterrupt(void)
{
        D1(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);
                D1(bug(" GetSourceData() returns %08lx\n", r.d[0]));
        } else
                WriteMacInt32(audio_data + adatStreamInfo, 0);

        // Signal stream function
        audio_block_fetched++;
        D1(bug("AudioInterrupt done\n"));
}


/*
 *  Set sampling parameters
 *  "index" is an index into the audio_sample_rates[] etc. arrays
 *  It is guaranteed that AudioStatus.num_sources == 0
 */

void audio_set_sample_rate_byval(uint32 value)
{
        bool changed = (AudioStatus.sample_rate != value);
        if(changed)
                {
                ULONG sample_rate_index;

                // get index of sample rate closest to <value> Hz
                AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
                        AHIDB_IndexArg, value >> 16,
                        AHIDB_Index, (ULONG) &sample_rate_index,
                        TAG_END);

                D(bug(" audio_set_sample_rate_byval requested rate=%ld Hz\n", value >> 16));

                AudioStatus.sample_rate = audio_sample_rates[sample_rate_index];

                AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, 0);
                }

        D(bug(" audio_set_sample_rate_byval rate=%ld Hz\n", AudioStatus.sample_rate >> 16));
}

void audio_set_sample_size_byval(uint32 value)
{
        bool changed = (AudioStatus.sample_size != value);
        if(changed) {
//              AudioStatus.sample_size = value;
//              update_sound_parameters();
//              WritePrivateProfileInt( "Audio", "SampleSize", AudioStatus.sample_size, ini_file_name );
        }
        D(bug(" audio_set_sample_size_byval %d\n", AudioStatus.sample_size));
}

void audio_set_channels_byval(uint32 value)
{
        bool changed = (AudioStatus.channels != value);
        if(changed) {
//              AudioStatus.channels = value;
//              update_sound_parameters();
//              WritePrivateProfileInt( "Audio", "Channels", AudioStatus.channels, ini_file_name );
        }
        D(bug(" audio_set_channels_byval %d\n", AudioStatus.channels));
}

bool audio_set_sample_rate(int index)
{
        if(index >= 0 && index < audio_sample_rates.size() ) {
                audio_set_sample_rate_byval( audio_sample_rates[index] );
                D(bug(" audio_set_sample_rate index=%ld rate=%ld\n", index, AudioStatus.sample_rate >> 16));
        }

        return true;
}

bool audio_set_sample_size(int index)
{
        if(index >= 0 && index < audio_sample_sizes.size()  ) {
                audio_set_sample_size_byval( audio_sample_sizes[index] );
                D(bug(" audio_set_sample_size %d,%d\n", index,AudioStatus.sample_size));
        }

        return true;
}

bool audio_set_channels(int index)
{
        if(index >= 0 && index < audio_channel_counts.size()   ) {
                audio_set_channels_byval( audio_channel_counts[index] );
                D(bug(" audio_set_channels %d,%d\n", index,AudioStatus.channels));
        }

        return true;
}


/*
 *  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)
{
        D(bug("audio_get_main_mute:  mute=%ld\n", main_mute));

        return main_mute;
}

uint32 audio_get_main_volume(void)
{
        D(bug("audio_get_main_volume\n"));

                ULONG volume = current_main_volume >> 8;        // 0x10000 => 0x100

                D(bug("audio_get_main_volume: volume=%08lx\n", volume));

                return (volume << 16) + volume;

        return 0x01000100;
}

bool audio_get_speaker_mute(void)
{
        D(bug("audio_get_speaker_mute:  mute=%ld\n", speaker_mute));

        return speaker_mute;
}

uint32 audio_get_speaker_volume(void)
{
        D(bug("audio_get_speaker_volume: \n"));

        if (audio_open)
                {
                ULONG volume = current_speaker_volume >> 8;     // 0x10000 => 0x100

                D(bug("audio_get_speaker_volume: volume=%08lx\n", volume));

                return (volume << 16) + volume;
                }

        return 0x01000100;
}

void audio_set_main_mute(bool mute)
{
        D(bug("audio_set_main_mute: mute=%ld\n", mute));

        if (mute != main_mute)
                {
                main_mute = mute;
                }
}

void audio_set_main_volume(uint32 vol)
{
        D(bug("audio_set_main_volume: vol=%08lx\n", vol));

        if (audio_open && supports_volume_changes)
                {
                ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff));

                D(bug("audio_set_main_volume: volume=%08lx\n", volume));

                current_main_volume = volume;

                AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM);
                }
}

void audio_set_speaker_mute(bool mute)
{
        D(bug("audio_set_speaker_mute: mute=%ld\n", mute));

        if (mute != speaker_mute)
                {
                speaker_mute = mute;
                }
}

void audio_set_speaker_volume(uint32 vol)
{
        D(bug("audio_set_speaker_volume: vol=%08lx\n", vol));

        if (audio_open && supports_volume_changes)
                {
                ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff));

                D(bug("audio_set_speaker_volume: volume=%08lx\n", volume));

                current_speaker_volume = volume;

                AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM);
                }
}
Revision:	1.11
Committed:	2002-06-23T08:27:05Z (22 years, 4 months ago) by jlachmann
Branch:	MAIN
CVS Tags:	nigel-build-19, nigel-build-12, nigel-build-13, nigel-build-16, nigel-build-17, nigel-build-15
Changes since 1.10:	+240 -34 lines
Log Message:	Adapted to OO video scheme; Audio volume/muting/sample rate now settable
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_amiga.cpp - Audio support, AmigaOS implementation using AHI
3			*
4	jlachmann	1.11	* 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 <exec/types.h>
24			#include <exec/memory.h>
25			#include <devices/ahi.h>
26	jlachmann	1.11	#define __USE_SYSBASE
27	cebix	1.1	#include <proto/exec.h>
28			#include <proto/ahi.h>
29	jlachmann	1.11	#include <inline/exec.h>
30			#include <inline/ahi.h>
31	cebix	1.1
32			#include "cpu_emulation.h"
33			#include "main.h"
34			#include "prefs.h"
35			#include "user_strings.h"
36			#include "audio.h"
37			#include "audio_defs.h"
38
39			#define DEBUG 0
40			#include "debug.h"
41
42	jlachmann	1.11	#define D1(x) ;
43
44	cebix	1.1
45			// Global variables
46			static ULONG ahi_id = AHI_DEFAULT_ID; // AHI audio ID
47			static struct AHIAudioCtrl *ahi_ctrl = NULL;
48			static struct AHISampleInfo sample[2]; // Two sample infos for double-buffering
49			static struct Hook sf_hook;
50			static int play_buf = 0; // Number of currently played buffer
51			static long sound_buffer_size; // Size of one audio buffer in bytes
52			static int audio_block_fetched = 0; // Number of audio blocks fetched by interrupt routine
53
54	jlachmann	1.11	static bool main_mute = false;
55			static bool speaker_mute = false;
56			static ULONG supports_volume_changes = false;
57			static ULONG supports_stereo_panning = false;
58			static ULONG current_main_volume;
59			static ULONG current_speaker_volume;
60
61	cebix	1.1
62			// Prototypes
63	cebix	1.4	static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook hook /a0/, struct AHISoundMessage msg /a1/, struct AHIAudioCtrl ahi_ctrl /a2*/);
64	jlachmann	1.11	void audio_set_sample_rate_byval(uint32 value);
65			void audio_set_sample_size_byval(uint32 value);
66			void audio_set_channels_byval(uint32 value);
67	cebix	1.1
68
69			/*
70			* Initialization
71			*/
72
73	cebix	1.9	// Set AudioStatus to reflect current audio stream format
74	jlachmann	1.11	static void set_audio_status_format(int sample_rate_index)
75	cebix	1.9	{
76	jlachmann	1.11	AudioStatus.sample_rate = audio_sample_rates[sample_rate_index];
77	cebix	1.9	AudioStatus.sample_size = audio_sample_sizes[0];
78			AudioStatus.channels = audio_channel_counts[0];
79			}
80
81	cebix	1.1	void AudioInit(void)
82			{
83			sample[0].ahisi_Address = sample[1].ahisi_Address = NULL;
84
85			// Init audio status and feature flags
86	cebix	1.9	audio_channel_counts.push_back(2);
87	jlachmann	1.11	// set_audio_status_format();
88	cebix	1.1	AudioStatus.mixer = 0;
89			AudioStatus.num_sources = 0;
90			audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
91
92			// Sound disabled in prefs? Then do nothing
93			if (PrefsFindBool("nosound"))
94			return;
95
96			// AHI available?
97			if (AHIBase == NULL) {
98			WarningAlert(GetString(STR_NO_AHI_WARN));
99			return;
100			}
101
102			// Initialize callback hook
103			sf_hook.h_Entry = (HOOKFUNC)audio_callback;
104
105			// Read "sound" preferences
106			const char *str = PrefsFindString("sound");
107			if (str)
108			sscanf(str, "ahi/%08lx", &ahi_id);
109
110			// Open audio control structure
111			if ((ahi_ctrl = AHI_AllocAudio(
112			AHIA_AudioID, ahi_id,
113			AHIA_MixFreq, AudioStatus.sample_rate >> 16,
114			AHIA_Channels, 1,
115			AHIA_Sounds, 2,
116			AHIA_SoundFunc, (ULONG)&sf_hook,
117			TAG_END)) == NULL) {
118			WarningAlert(GetString(STR_NO_AHI_CTRL_WARN));
119			return;
120			}
121
122	jlachmann	1.11	ULONG max_channels, sample_rate, frequencies, sample_rate_index;
123
124			AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
125			AHIDB_MaxChannels, (ULONG) &max_channels,
126			AHIDB_Frequencies, (ULONG) &frequencies,
127			TAG_END);
128
129			D(bug("AudioInit: max_channels=%ld frequencies=%ld\n", max_channels, frequencies));
130
131			for (int n=0; n<frequencies; n++)
132			{
133			AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
134			AHIDB_FrequencyArg, n,
135			AHIDB_Frequency, (ULONG) &sample_rate,
136			TAG_END);
137
138			D(bug("AudioInit: f=%ld Hz\n", sample_rate));
139			audio_sample_rates.push_back(sample_rate << 16);
140			}
141
142			ULONG sample_size_bits = 16;
143
144			D(bug("AudioInit: sampe_rates=%ld\n", audio_sample_rates.size() ));
145
146			// get index of sample rate closest to 22050 Hz
147			AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
148			AHIDB_IndexArg, 22050,
149			AHIDB_Bits, (ULONG) &sample_size_bits,
150			AHIDB_Index, (ULONG) &sample_rate_index,
151			AHIDB_Volume, (ULONG) &supports_volume_changes,
152			AHIDB_Panning, (ULONG) &supports_stereo_panning,
153			TAG_END);
154
155			audio_sample_sizes.push_back(16);
156
157			set_audio_status_format(sample_rate_index);
158
159	cebix	1.1	// 2048 frames per block
160			audio_frames_per_block = 2048;
161			sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
162
163			// Prepare SampleInfos and load sounds (two sounds for double buffering)
164			sample[0].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
165			sample[0].ahisi_Length = audio_frames_per_block;
166			sample[0].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC \| MEMF_CLEAR);
167			sample[1].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
168			sample[1].ahisi_Length = audio_frames_per_block;
169			sample[1].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC \| MEMF_CLEAR);
170			if (sample[0].ahisi_Address == NULL \|\| sample[1].ahisi_Address == NULL)
171			return;
172	jlachmann	1.11
173	cebix	1.1	AHI_LoadSound(0, AHIST_DYNAMICSAMPLE, &sample[0], ahi_ctrl);
174			AHI_LoadSound(1, AHIST_DYNAMICSAMPLE, &sample[1], ahi_ctrl);
175
176			// Set parameters
177			play_buf = 0;
178	jlachmann	1.11	current_main_volume = current_speaker_volume = 0x10000;
179			AHI_SetVol(0, current_speaker_volume, 0x8000, ahi_ctrl, AHISF_IMM);
180
181	cebix	1.1	AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, AHISF_IMM);
182			AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, AHISF_IMM);
183
184			// Everything OK
185			audio_open = true;
186			}
187
188
189			/*
190			* Deinitialization
191			*/
192
193			void AudioExit(void)
194			{
195			// Free everything
196			if (ahi_ctrl != NULL) {
197			AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
198			AHI_FreeAudio(ahi_ctrl);
199			}
200
201			FreeVec(sample[0].ahisi_Address);
202			FreeVec(sample[1].ahisi_Address);
203			}
204
205
206			/*
207			* First source added, start audio stream
208			*/
209
210			void audio_enter_stream()
211			{
212			AHI_ControlAudio(ahi_ctrl, AHIC_Play, TRUE, TAG_END);
213			}
214
215
216			/*
217			* Last source removed, stop audio stream
218			*/
219
220			void audio_exit_stream()
221			{
222			AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
223			}
224
225
226			/*
227			* AHI sound callback, request next buffer
228			*/
229
230	cebix	1.4	static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook hook /a0/, struct AHISoundMessage msg /a1/, struct AHIAudioCtrl ahi_ctrl /a2*/)
231	cebix	1.1	{
232			play_buf ^= 1;
233
234			// New buffer available?
235	jlachmann	1.11	if (audio_block_fetched)
236			{
237	cebix	1.1	audio_block_fetched--;
238
239	jlachmann	1.11	if (main_mute \|\| speaker_mute)
240			{
241			memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size);
242			}
243			else
244			{
245			// Get size of audio data
246			uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
247			if (apple_stream_info) {
248			int32 sample_count = ReadMacInt32(apple_stream_info + scd_sampleCount);
249
250			uint32 num_channels = ReadMacInt16(apple_stream_info + scd_numChannels);
251			uint32 sample_size = ReadMacInt16(apple_stream_info + scd_sampleSize);
252			uint32 sample_rate = ReadMacInt32(apple_stream_info + scd_sampleRate);
253
254			D(bug("stream: sample_count=%ld num_channels=%ld sample_size=%ld sample_rate=%ld\n", sample_count, num_channels, sample_size, sample_rate >> 16));
255
256			// Yes, this can happen.
257			if(sample_count != 0) {
258			if(sample_rate != AudioStatus.sample_rate) {
259			audio_set_sample_rate_byval(sample_rate);
260			}
261			if(num_channels != AudioStatus.channels) {
262			audio_set_channels_byval(num_channels);
263			}
264			if(sample_size != AudioStatus.sample_size) {
265			audio_set_sample_size_byval(sample_size);
266			}
267			}
268
269			if (sample_count < 0)
270			sample_count = 0;
271
272			int work_size = sample_count * num_channels * (sample_size>>3);
273			D(bug("stream: work_size=%ld sound_buffer_size=%ld\n", work_size, sound_buffer_size));
274
275			if (work_size > sound_buffer_size)
276			work_size = sound_buffer_size;
277
278			// Put data into AHI buffer (convert 8-bit data unsigned->signed)
279			if (AudioStatus.sample_size == 16)
280			Mac2Host_memcpy(sample[play_buf].ahisi_Address, ReadMacInt32(apple_stream_info + scd_buffer), work_size);
281			else {
282			uint32 p = (uint32 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
283			uint32 q = (uint32 )sample[play_buf].ahisi_Address;
284			int r = work_size >> 2;
285			while (r--)
286			q++ = p++ ^ 0x80808080;
287			}
288			if (work_size != sound_buffer_size)
289			memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size);
290	cebix	1.1	}
291			}
292
293	jlachmann	1.11	}
294			else
295	cebix	1.1	memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size);
296
297			// Play next buffer
298			AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, 0);
299
300			// Trigger audio interrupt to get new buffer
301			if (AudioStatus.num_sources) {
302	jlachmann	1.11	D1(bug("stream: triggering irq\n"));
303	cebix	1.1	SetInterruptFlag(INTFLAG_AUDIO);
304			TriggerInterrupt();
305			}
306			return 0;
307			}
308
309
310			/*
311			* MacOS audio interrupt, read next data block
312			*/
313
314			void AudioInterrupt(void)
315			{
316	jlachmann	1.11	D1(bug("AudioInterrupt\n"));
317	cebix	1.1
318			// Get data from apple mixer
319			if (AudioStatus.mixer) {
320			M68kRegisters r;
321			r.a[0] = audio_data + adatStreamInfo;
322			r.a[1] = AudioStatus.mixer;
323			Execute68k(audio_data + adatGetSourceData, &r);
324	jlachmann	1.11	D1(bug(" GetSourceData() returns %08lx\n", r.d[0]));
325	cebix	1.1	} else
326			WriteMacInt32(audio_data + adatStreamInfo, 0);
327
328			// Signal stream function
329			audio_block_fetched++;
330	jlachmann	1.11	D1(bug("AudioInterrupt done\n"));
331	cebix	1.1	}
332
333
334			/*
335			* Set sampling parameters
336			* "index" is an index into the audio_sample_rates[] etc. arrays
337			* It is guaranteed that AudioStatus.num_sources == 0
338			*/
339
340	jlachmann	1.11	void audio_set_sample_rate_byval(uint32 value)
341			{
342			bool changed = (AudioStatus.sample_rate != value);
343			if(changed)
344			{
345			ULONG sample_rate_index;
346
347			// get index of sample rate closest to <value> Hz
348			AHI_GetAudioAttrs(ahi_id, ahi_ctrl,
349			AHIDB_IndexArg, value >> 16,
350			AHIDB_Index, (ULONG) &sample_rate_index,
351			TAG_END);
352
353			D(bug(" audio_set_sample_rate_byval requested rate=%ld Hz\n", value >> 16));
354
355			AudioStatus.sample_rate = audio_sample_rates[sample_rate_index];
356
357			AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, 0);
358			}
359
360			D(bug(" audio_set_sample_rate_byval rate=%ld Hz\n", AudioStatus.sample_rate >> 16));
361			}
362
363			void audio_set_sample_size_byval(uint32 value)
364			{
365			bool changed = (AudioStatus.sample_size != value);
366			if(changed) {
367			// AudioStatus.sample_size = value;
368			// update_sound_parameters();
369			// WritePrivateProfileInt( "Audio", "SampleSize", AudioStatus.sample_size, ini_file_name );
370			}
371			D(bug(" audio_set_sample_size_byval %d\n", AudioStatus.sample_size));
372			}
373
374			void audio_set_channels_byval(uint32 value)
375			{
376			bool changed = (AudioStatus.channels != value);
377			if(changed) {
378			// AudioStatus.channels = value;
379			// update_sound_parameters();
380			// WritePrivateProfileInt( "Audio", "Channels", AudioStatus.channels, ini_file_name );
381			}
382			D(bug(" audio_set_channels_byval %d\n", AudioStatus.channels));
383			}
384
385	cebix	1.9	bool audio_set_sample_rate(int index)
386	cebix	1.1	{
387	jlachmann	1.11	if(index >= 0 && index < audio_sample_rates.size() ) {
388			audio_set_sample_rate_byval( audio_sample_rates[index] );
389			D(bug(" audio_set_sample_rate index=%ld rate=%ld\n", index, AudioStatus.sample_rate >> 16));
390			}
391
392	cebix	1.9	return true;
393	cebix	1.1	}
394
395	cebix	1.9	bool audio_set_sample_size(int index)
396	cebix	1.1	{
397	jlachmann	1.11	if(index >= 0 && index < audio_sample_sizes.size() ) {
398			audio_set_sample_size_byval( audio_sample_sizes[index] );
399			D(bug(" audio_set_sample_size %d,%d\n", index,AudioStatus.sample_size));
400			}
401
402	cebix	1.9	return true;
403	cebix	1.1	}
404
405	cebix	1.9	bool audio_set_channels(int index)
406	cebix	1.1	{
407	jlachmann	1.11	if(index >= 0 && index < audio_channel_counts.size() ) {
408			audio_set_channels_byval( audio_channel_counts[index] );
409			D(bug(" audio_set_channels %d,%d\n", index,AudioStatus.channels));
410			}
411
412	cebix	1.9	return true;
413	cebix	1.1	}
414
415
416			/*
417			* Get/set volume controls (volume values received/returned have the left channel
418			* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
419			* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
420			*/
421
422			bool audio_get_main_mute(void)
423			{
424	jlachmann	1.11	D(bug("audio_get_main_mute: mute=%ld\n", main_mute));
425
426			return main_mute;
427	cebix	1.1	}
428
429			uint32 audio_get_main_volume(void)
430			{
431	jlachmann	1.11	D(bug("audio_get_main_volume\n"));
432
433			ULONG volume = current_main_volume >> 8; // 0x10000 => 0x100
434
435			D(bug("audio_get_main_volume: volume=%08lx\n", volume));
436
437			return (volume << 16) + volume;
438
439	cebix	1.1	return 0x01000100;
440			}
441
442	cebix	1.3	bool audio_get_speaker_mute(void)
443	cebix	1.1	{
444	jlachmann	1.11	D(bug("audio_get_speaker_mute: mute=%ld\n", speaker_mute));
445
446			return speaker_mute;
447	cebix	1.1	}
448
449	cebix	1.3	uint32 audio_get_speaker_volume(void)
450	cebix	1.1	{
451	jlachmann	1.11	D(bug("audio_get_speaker_volume: \n"));
452
453			if (audio_open)
454			{
455			ULONG volume = current_speaker_volume >> 8; // 0x10000 => 0x100
456
457			D(bug("audio_get_speaker_volume: volume=%08lx\n", volume));
458
459			return (volume << 16) + volume;
460			}
461
462	cebix	1.1	return 0x01000100;
463			}
464
465			void audio_set_main_mute(bool mute)
466			{
467	jlachmann	1.11	D(bug("audio_set_main_mute: mute=%ld\n", mute));
468
469			if (mute != main_mute)
470			{
471			main_mute = mute;
472			}
473	cebix	1.1	}
474
475			void audio_set_main_volume(uint32 vol)
476			{
477	jlachmann	1.11	D(bug("audio_set_main_volume: vol=%08lx\n", vol));
478
479			if (audio_open && supports_volume_changes)
480			{
481			ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff));
482
483			D(bug("audio_set_main_volume: volume=%08lx\n", volume));
484
485			current_main_volume = volume;
486
487			AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM);
488			}
489	cebix	1.1	}
490
491	cebix	1.3	void audio_set_speaker_mute(bool mute)
492	cebix	1.1	{
493	jlachmann	1.11	D(bug("audio_set_speaker_mute: mute=%ld\n", mute));
494
495			if (mute != speaker_mute)
496			{
497			speaker_mute = mute;
498			}
499	cebix	1.1	}
500
501	cebix	1.3	void audio_set_speaker_volume(uint32 vol)
502	cebix	1.1	{
503	jlachmann	1.11	D(bug("audio_set_speaker_volume: vol=%08lx\n", vol));
504
505			if (audio_open && supports_volume_changes)
506			{
507			ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff));
508
509			D(bug("audio_set_speaker_volume: volume=%08lx\n", volume));
510
511			current_speaker_volume = volume;
512
513			AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM);
514			}
515	cebix	1.1	}