src/AmigaOS/audio_amiga.cpp

/*
 *  audio_amiga.cpp - Audio support, AmigaOS implementation using AHI
 *
 *  Basilisk II (C) 1997-1999 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>
#include <proto/exec.h>
#include <proto/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"


// Supported sample rates, sizes and channels
int audio_num_sample_rates = 1;
uint32 audio_sample_rates[] = {22050 << 16};
int audio_num_sample_sizes = 1;
uint16 audio_sample_sizes[] = {16};
int audio_num_channel_counts = 1;
uint16 audio_channel_counts[] = {2};


// 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


// Prototypes
static __saveds __asm ULONG audio_callback(register __a0 struct Hook *hook, register __a2 struct AHIAudioCtrl *ahi_ctrl, register __a1 struct AHISoundMessage *msg);


/*
 *  Initialization
 */

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

        // Init audio status and feature flags
        AudioStatus.sample_rate = audio_sample_rates[0];
        AudioStatus.sample_size = audio_sample_sizes[0];
        AudioStatus.channels = audio_channel_counts[0];
        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;
        }

        // 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;
        AHI_SetVol(0, 0x10000, 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 __asm ULONG audio_callback(register __a0 struct Hook *hook, register __a2 struct AHIAudioCtrl *ahi_ctrl, register __a1 struct AHISoundMessage *msg)
{
        play_buf ^= 1;

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

                // 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;

                        // Put data into AHI buffer (convert 8-bit data unsigned->signed)
                        if (AudioStatus.sample_size == 16)
                                memcpy(sample[play_buf].ahisi_Address, Mac2HostAddr(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) {
                D(bug("stream: triggering irq\n"));
                SetInterruptFlag(INTFLAG_AUDIO);
                TriggerInterrupt();
        }
        return 0;
}


/*
 *  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
        audio_block_fetched++;
        D(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(int index)
{
}

void audio_set_sample_size(int index)
{
}

void audio_set_channels(int index)
{
}


/*
 *  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)
{
        return 0x01000100;
}

bool audio_get_dac_mute(void)
{
        return false;
}

uint32 audio_get_dac_volume(void)
{
        return 0x01000100;
}

void audio_set_main_mute(bool mute)
{
}

void audio_set_main_volume(uint32 vol)
{
}

void audio_set_dac_mute(bool mute)
{
}

void audio_set_dac_volume(uint32 vol)
{
}
Revision:	1.1
Committed:	1999-10-03T14:16:25Z (25 years, 1 month ago) by cebix
Branch:	MAIN
Branch point for:	cebix
Log Message:	Initial revision
#	User	Rev	Content
1	cebix	1.1	/*
2			* audio_amiga.cpp - Audio support, AmigaOS implementation using AHI
3			*
4			* Basilisk II (C) 1997-1999 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 <exec/types.h>
24			#include <exec/memory.h>
25			#include <devices/ahi.h>
26			#include <proto/exec.h>
27			#include <proto/ahi.h>
28
29			#include "cpu_emulation.h"
30			#include "main.h"
31			#include "prefs.h"
32			#include "user_strings.h"
33			#include "audio.h"
34			#include "audio_defs.h"
35
36			#define DEBUG 0
37			#include "debug.h"
38
39
40			// Supported sample rates, sizes and channels
41			int audio_num_sample_rates = 1;
42			uint32 audio_sample_rates[] = {22050 << 16};
43			int audio_num_sample_sizes = 1;
44			uint16 audio_sample_sizes[] = {16};
45			int audio_num_channel_counts = 1;
46			uint16 audio_channel_counts[] = {2};
47
48
49			// Global variables
50			static ULONG ahi_id = AHI_DEFAULT_ID; // AHI audio ID
51			static struct AHIAudioCtrl *ahi_ctrl = NULL;
52			static struct AHISampleInfo sample[2]; // Two sample infos for double-buffering
53			static struct Hook sf_hook;
54			static int play_buf = 0; // Number of currently played buffer
55			static long sound_buffer_size; // Size of one audio buffer in bytes
56			static int audio_block_fetched = 0; // Number of audio blocks fetched by interrupt routine
57
58
59			// Prototypes
60			static __saveds __asm ULONG audio_callback(register __a0 struct Hook hook, register __a2 struct AHIAudioCtrl ahi_ctrl, register __a1 struct AHISoundMessage *msg);
61
62
63			/*
64			* Initialization
65			*/
66
67			void AudioInit(void)
68			{
69			sample[0].ahisi_Address = sample[1].ahisi_Address = NULL;
70
71			// Init audio status and feature flags
72			AudioStatus.sample_rate = audio_sample_rates[0];
73			AudioStatus.sample_size = audio_sample_sizes[0];
74			AudioStatus.channels = audio_channel_counts[0];
75			AudioStatus.mixer = 0;
76			AudioStatus.num_sources = 0;
77			audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
78
79			// Sound disabled in prefs? Then do nothing
80			if (PrefsFindBool("nosound"))
81			return;
82
83			// AHI available?
84			if (AHIBase == NULL) {
85			WarningAlert(GetString(STR_NO_AHI_WARN));
86			return;
87			}
88
89			// Initialize callback hook
90			sf_hook.h_Entry = (HOOKFUNC)audio_callback;
91
92			// Read "sound" preferences
93			const char *str = PrefsFindString("sound");
94			if (str)
95			sscanf(str, "ahi/%08lx", &ahi_id);
96
97			// Open audio control structure
98			if ((ahi_ctrl = AHI_AllocAudio(
99			AHIA_AudioID, ahi_id,
100			AHIA_MixFreq, AudioStatus.sample_rate >> 16,
101			AHIA_Channels, 1,
102			AHIA_Sounds, 2,
103			AHIA_SoundFunc, (ULONG)&sf_hook,
104			TAG_END)) == NULL) {
105			WarningAlert(GetString(STR_NO_AHI_CTRL_WARN));
106			return;
107			}
108
109			// 2048 frames per block
110			audio_frames_per_block = 2048;
111			sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
112
113			// Prepare SampleInfos and load sounds (two sounds for double buffering)
114			sample[0].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
115			sample[0].ahisi_Length = audio_frames_per_block;
116			sample[0].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC \| MEMF_CLEAR);
117			sample[1].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
118			sample[1].ahisi_Length = audio_frames_per_block;
119			sample[1].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC \| MEMF_CLEAR);
120			if (sample[0].ahisi_Address == NULL \|\| sample[1].ahisi_Address == NULL)
121			return;
122			AHI_LoadSound(0, AHIST_DYNAMICSAMPLE, &sample[0], ahi_ctrl);
123			AHI_LoadSound(1, AHIST_DYNAMICSAMPLE, &sample[1], ahi_ctrl);
124
125			// Set parameters
126			play_buf = 0;
127			AHI_SetVol(0, 0x10000, 0x8000, ahi_ctrl, AHISF_IMM);
128			AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, AHISF_IMM);
129			AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, AHISF_IMM);
130
131			// Everything OK
132			audio_open = true;
133			}
134
135
136			/*
137			* Deinitialization
138			*/
139
140			void AudioExit(void)
141			{
142			// Free everything
143			if (ahi_ctrl != NULL) {
144			AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
145			AHI_FreeAudio(ahi_ctrl);
146			}
147
148			FreeVec(sample[0].ahisi_Address);
149			FreeVec(sample[1].ahisi_Address);
150			}
151
152
153			/*
154			* First source added, start audio stream
155			*/
156
157			void audio_enter_stream()
158			{
159			AHI_ControlAudio(ahi_ctrl, AHIC_Play, TRUE, TAG_END);
160			}
161
162
163			/*
164			* Last source removed, stop audio stream
165			*/
166
167			void audio_exit_stream()
168			{
169			AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
170			}
171
172
173			/*
174			* AHI sound callback, request next buffer
175			*/
176
177			static __saveds __asm ULONG audio_callback(register __a0 struct Hook hook, register __a2 struct AHIAudioCtrl ahi_ctrl, register __a1 struct AHISoundMessage *msg)
178			{
179			play_buf ^= 1;
180
181			// New buffer available?
182			if (audio_block_fetched) {
183			audio_block_fetched--;
184
185			// Get size of audio data
186			uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
187			if (apple_stream_info) {
188			int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
189			D(bug("stream: work_size %d\n", work_size));
190			if (work_size > sound_buffer_size)
191			work_size = sound_buffer_size;
192
193			// Put data into AHI buffer (convert 8-bit data unsigned->signed)
194			if (AudioStatus.sample_size == 16)
195			memcpy(sample[play_buf].ahisi_Address, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size);
196			else {
197			uint32 p = (uint32 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
198			uint32 q = (uint32 )sample[play_buf].ahisi_Address;
199			int r = work_size >> 2;
200			while (r--)
201			q++ = p++ ^ 0x80808080;
202			}
203			if (work_size != sound_buffer_size)
204			memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size);
205			}
206
207			} else
208			memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size);
209
210			// Play next buffer
211			AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, 0);
212
213			// Trigger audio interrupt to get new buffer
214			if (AudioStatus.num_sources) {
215			D(bug("stream: triggering irq\n"));
216			SetInterruptFlag(INTFLAG_AUDIO);
217			TriggerInterrupt();
218			}
219			return 0;
220			}
221
222
223			/*
224			* MacOS audio interrupt, read next data block
225			*/
226
227			void AudioInterrupt(void)
228			{
229			D(bug("AudioInterrupt\n"));
230
231			// Get data from apple mixer
232			if (AudioStatus.mixer) {
233			M68kRegisters r;
234			r.a[0] = audio_data + adatStreamInfo;
235			r.a[1] = AudioStatus.mixer;
236			Execute68k(audio_data + adatGetSourceData, &r);
237			D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
238			} else
239			WriteMacInt32(audio_data + adatStreamInfo, 0);
240
241			// Signal stream function
242			audio_block_fetched++;
243			D(bug("AudioInterrupt done\n"));
244			}
245
246
247			/*
248			* Set sampling parameters
249			* "index" is an index into the audio_sample_rates[] etc. arrays
250			* It is guaranteed that AudioStatus.num_sources == 0
251			*/
252
253			void audio_set_sample_rate(int index)
254			{
255			}
256
257			void audio_set_sample_size(int index)
258			{
259			}
260
261			void audio_set_channels(int index)
262			{
263			}
264
265
266			/*
267			* Get/set volume controls (volume values received/returned have the left channel
268			* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
269			* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
270			*/
271
272			bool audio_get_main_mute(void)
273			{
274			return false;
275			}
276
277			uint32 audio_get_main_volume(void)
278			{
279			return 0x01000100;
280			}
281
282			bool audio_get_dac_mute(void)
283			{
284			return false;
285			}
286
287			uint32 audio_get_dac_volume(void)
288			{
289			return 0x01000100;
290			}
291
292			void audio_set_main_mute(bool mute)
293			{
294			}
295
296			void audio_set_main_volume(uint32 vol)
297			{
298			}
299
300			void audio_set_dac_mute(bool mute)
301			{
302			}
303
304			void audio_set_dac_volume(uint32 vol)
305			{
306			}