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root/cebix/BasiliskII/src/AmigaOS/audio_amiga.cpp
Revision: 1.5
Committed: 1999-11-01T16:24:09Z (25 years ago) by cebix
Branch: MAIN
CVS Tags: snapshot-02111999
Changes since 1.4: +0 -8 lines
Log Message:
- AmigaOS: removed support for SAS/C
- AmigaOS: sys_amiga.cpp: supports 64-bit device access and respects
  device block size on writes
- AmigaOS: added support for resource forks and Finder info for ExtFS
- AmigaOS: added "ExtFS" gadget to prefs editor
- protection mask for all open()/creat()/mkdir() calls is now 0666 or
  0777

File Contents

# Content
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 __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/);
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 __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/)
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_speaker_mute(void)
283 {
284 return false;
285 }
286
287 uint32 audio_get_speaker_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_speaker_mute(bool mute)
301 {
302 }
303
304 void audio_set_speaker_volume(uint32 vol)
305 {
306 }