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AmigActive 6
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Emulation
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BasiliskII
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AmigaOS
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audio_amiga.cpp
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C/C++ Source or Header
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1999-11-03
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/*
* 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 __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/);
/*
* 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 __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--;
// 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)
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) {
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)
{
}