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
#ifdef __GNUC__
static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/);
#else
static __saveds __regargs ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/);
#endif


/*
 *  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
 */

#ifdef __GNUC__
static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/)
#else
static __saveds __regargs ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/)
#endif
{
        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_speaker_mute(void)
{
        return false;
}

uint32 audio_get_speaker_volume(void)
{
        return 0x01000100;
}

void audio_set_main_mute(bool mute)
{
}

void audio_set_main_volume(uint32 vol)
{
}

void audio_set_speaker_mute(bool mute)
{
}

void audio_set_speaker_volume(uint32 vol)
{
}
Revision:	1.4
Committed:	1999-10-21T22:39:56Z (25 years ago) by cebix
Branch:	MAIN
Changes since 1.3:	+8 -0 lines
Log Message:	- ExtFS works under AmigaOS - fixed erroneous __regargs attributes in prefs_editor_amiga.cpp and audio_amiga.cpp for GCC
#	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	cebix	1.4	#ifdef __GNUC__
61			static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook hook /a0/, struct AHISoundMessage msg /a1/, struct AHIAudioCtrl ahi_ctrl /a2*/);
62			#else
63	cebix	1.2	static __saveds __regargs ULONG audio_callback(struct Hook hook /a0/, struct AHISoundMessage msg /a1/, struct AHIAudioCtrl ahi_ctrl /a2*/);
64	cebix	1.4	#endif
65	cebix	1.1
66
67			/*
68			* Initialization
69			*/
70
71			void AudioInit(void)
72			{
73			sample[0].ahisi_Address = sample[1].ahisi_Address = NULL;
74
75			// Init audio status and feature flags
76			AudioStatus.sample_rate = audio_sample_rates[0];
77			AudioStatus.sample_size = audio_sample_sizes[0];
78			AudioStatus.channels = audio_channel_counts[0];
79			AudioStatus.mixer = 0;
80			AudioStatus.num_sources = 0;
81			audio_component_flags = cmpWantsRegisterMessage \| kStereoOut \| k16BitOut;
82
83			// Sound disabled in prefs? Then do nothing
84			if (PrefsFindBool("nosound"))
85			return;
86
87			// AHI available?
88			if (AHIBase == NULL) {
89			WarningAlert(GetString(STR_NO_AHI_WARN));
90			return;
91			}
92
93			// Initialize callback hook
94			sf_hook.h_Entry = (HOOKFUNC)audio_callback;
95
96			// Read "sound" preferences
97			const char *str = PrefsFindString("sound");
98			if (str)
99			sscanf(str, "ahi/%08lx", &ahi_id);
100
101			// Open audio control structure
102			if ((ahi_ctrl = AHI_AllocAudio(
103			AHIA_AudioID, ahi_id,
104			AHIA_MixFreq, AudioStatus.sample_rate >> 16,
105			AHIA_Channels, 1,
106			AHIA_Sounds, 2,
107			AHIA_SoundFunc, (ULONG)&sf_hook,
108			TAG_END)) == NULL) {
109			WarningAlert(GetString(STR_NO_AHI_CTRL_WARN));
110			return;
111			}
112
113			// 2048 frames per block
114			audio_frames_per_block = 2048;
115			sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block;
116
117			// Prepare SampleInfos and load sounds (two sounds for double buffering)
118			sample[0].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
119			sample[0].ahisi_Length = audio_frames_per_block;
120			sample[0].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC \| MEMF_CLEAR);
121			sample[1].ahisi_Type = AudioStatus.sample_size == 16 ? AHIST_S16S : AHIST_S8S;
122			sample[1].ahisi_Length = audio_frames_per_block;
123			sample[1].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC \| MEMF_CLEAR);
124			if (sample[0].ahisi_Address == NULL \|\| sample[1].ahisi_Address == NULL)
125			return;
126			AHI_LoadSound(0, AHIST_DYNAMICSAMPLE, &sample[0], ahi_ctrl);
127			AHI_LoadSound(1, AHIST_DYNAMICSAMPLE, &sample[1], ahi_ctrl);
128
129			// Set parameters
130			play_buf = 0;
131			AHI_SetVol(0, 0x10000, 0x8000, ahi_ctrl, AHISF_IMM);
132			AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, AHISF_IMM);
133			AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, AHISF_IMM);
134
135			// Everything OK
136			audio_open = true;
137			}
138
139
140			/*
141			* Deinitialization
142			*/
143
144			void AudioExit(void)
145			{
146			// Free everything
147			if (ahi_ctrl != NULL) {
148			AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
149			AHI_FreeAudio(ahi_ctrl);
150			}
151
152			FreeVec(sample[0].ahisi_Address);
153			FreeVec(sample[1].ahisi_Address);
154			}
155
156
157			/*
158			* First source added, start audio stream
159			*/
160
161			void audio_enter_stream()
162			{
163			AHI_ControlAudio(ahi_ctrl, AHIC_Play, TRUE, TAG_END);
164			}
165
166
167			/*
168			* Last source removed, stop audio stream
169			*/
170
171			void audio_exit_stream()
172			{
173			AHI_ControlAudio(ahi_ctrl, AHIC_Play, FALSE, TAG_END);
174			}
175
176
177			/*
178			* AHI sound callback, request next buffer
179			*/
180
181	cebix	1.4	#ifdef __GNUC__
182			static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook hook /a0/, struct AHISoundMessage msg /a1/, struct AHIAudioCtrl ahi_ctrl /a2*/)
183			#else
184	cebix	1.2	static __saveds __regargs ULONG audio_callback(struct Hook hook /a0/, struct AHISoundMessage msg /a1/, struct AHIAudioCtrl ahi_ctrl /a2*/)
185	cebix	1.4	#endif
186	cebix	1.1	{
187			play_buf ^= 1;
188
189			// New buffer available?
190			if (audio_block_fetched) {
191			audio_block_fetched--;
192
193			// Get size of audio data
194			uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo);
195			if (apple_stream_info) {
196			int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels;
197			D(bug("stream: work_size %d\n", work_size));
198			if (work_size > sound_buffer_size)
199			work_size = sound_buffer_size;
200
201			// Put data into AHI buffer (convert 8-bit data unsigned->signed)
202			if (AudioStatus.sample_size == 16)
203			memcpy(sample[play_buf].ahisi_Address, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size);
204			else {
205			uint32 p = (uint32 )Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer));
206			uint32 q = (uint32 )sample[play_buf].ahisi_Address;
207			int r = work_size >> 2;
208			while (r--)
209			q++ = p++ ^ 0x80808080;
210			}
211			if (work_size != sound_buffer_size)
212			memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size);
213			}
214
215			} else
216			memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size);
217
218			// Play next buffer
219			AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, 0);
220
221			// Trigger audio interrupt to get new buffer
222			if (AudioStatus.num_sources) {
223			D(bug("stream: triggering irq\n"));
224			SetInterruptFlag(INTFLAG_AUDIO);
225			TriggerInterrupt();
226			}
227			return 0;
228			}
229
230
231			/*
232			* MacOS audio interrupt, read next data block
233			*/
234
235			void AudioInterrupt(void)
236			{
237			D(bug("AudioInterrupt\n"));
238
239			// Get data from apple mixer
240			if (AudioStatus.mixer) {
241			M68kRegisters r;
242			r.a[0] = audio_data + adatStreamInfo;
243			r.a[1] = AudioStatus.mixer;
244			Execute68k(audio_data + adatGetSourceData, &r);
245			D(bug(" GetSourceData() returns %08lx\n", r.d[0]));
246			} else
247			WriteMacInt32(audio_data + adatStreamInfo, 0);
248
249			// Signal stream function
250			audio_block_fetched++;
251			D(bug("AudioInterrupt done\n"));
252			}
253
254
255			/*
256			* Set sampling parameters
257			* "index" is an index into the audio_sample_rates[] etc. arrays
258			* It is guaranteed that AudioStatus.num_sources == 0
259			*/
260
261			void audio_set_sample_rate(int index)
262			{
263			}
264
265			void audio_set_sample_size(int index)
266			{
267			}
268
269			void audio_set_channels(int index)
270			{
271			}
272
273
274			/*
275			* Get/set volume controls (volume values received/returned have the left channel
276			* volume in the upper 16 bits and the right channel volume in the lower 16 bits;
277			* both volumes are 8.8 fixed point values with 0x0100 meaning "maximum volume"))
278			*/
279
280			bool audio_get_main_mute(void)
281			{
282			return false;
283			}
284
285			uint32 audio_get_main_volume(void)
286			{
287			return 0x01000100;
288			}
289
290	cebix	1.3	bool audio_get_speaker_mute(void)
291	cebix	1.1	{
292			return false;
293			}
294
295	cebix	1.3	uint32 audio_get_speaker_volume(void)
296	cebix	1.1	{
297			return 0x01000100;
298			}
299
300			void audio_set_main_mute(bool mute)
301			{
302			}
303
304			void audio_set_main_volume(uint32 vol)
305			{
306			}
307
308	cebix	1.3	void audio_set_speaker_mute(bool mute)
309	cebix	1.1	{
310			}
311
312	cebix	1.3	void audio_set_speaker_volume(uint32 vol)
313	cebix	1.1	{
314			}