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:	1.1.1.1 (vendor branch)
Committed:	1999-10-03T14:16:25Z (25 years, 1 month ago) by cebix
Branch:	cebix
CVS Tags:	release-0_7-2, start
Changes since 1.1:	+0 -0 lines
Error occurred while calculating annotation data.
Log Message:	Imported sources
#	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 __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	}