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/* |
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* audio_amiga.cpp - Audio support, AmigaOS implementation using AHI |
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* |
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< |
* Basilisk II (C) 1997-1999 Christian Bauer |
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> |
* Basilisk II (C) 1997-2008 Christian Bauer |
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* |
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* This program is free software; you can redistribute it and/or modify |
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* it under the terms of the GNU General Public License as published by |
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#include <exec/types.h> |
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#include <exec/memory.h> |
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#include <devices/ahi.h> |
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#define __USE_SYSBASE |
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#include <proto/exec.h> |
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#include <proto/ahi.h> |
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#include <inline/exec.h> |
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#include <inline/ahi.h> |
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|
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#include "cpu_emulation.h" |
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#include "main.h" |
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#define DEBUG 0 |
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#include "debug.h" |
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|
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< |
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< |
// Supported sample rates, sizes and channels |
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< |
int audio_num_sample_rates = 1; |
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< |
uint32 audio_sample_rates[] = {22050 << 16}; |
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int audio_num_sample_sizes = 1; |
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< |
uint16 audio_sample_sizes[] = {16}; |
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int audio_num_channel_counts = 1; |
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uint16 audio_channel_counts[] = {2}; |
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> |
#define D1(x) ; |
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// Global variables |
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static long sound_buffer_size; // Size of one audio buffer in bytes |
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static int audio_block_fetched = 0; // Number of audio blocks fetched by interrupt routine |
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|
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static bool main_mute = false; |
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static bool speaker_mute = false; |
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static ULONG supports_volume_changes = false; |
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static ULONG supports_stereo_panning = false; |
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static ULONG current_main_volume; |
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static ULONG current_speaker_volume; |
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|
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|
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// Prototypes |
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< |
static __saveds __regargs ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/); |
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> |
static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/); |
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> |
void audio_set_sample_rate_byval(uint32 value); |
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> |
void audio_set_sample_size_byval(uint32 value); |
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> |
void audio_set_channels_byval(uint32 value); |
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/* |
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* Initialization |
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*/ |
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|
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// Set AudioStatus to reflect current audio stream format |
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static void set_audio_status_format(int sample_rate_index) |
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{ |
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AudioStatus.sample_rate = audio_sample_rates[sample_rate_index]; |
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AudioStatus.sample_size = audio_sample_sizes[0]; |
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AudioStatus.channels = audio_channel_counts[0]; |
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} |
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|
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void AudioInit(void) |
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{ |
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sample[0].ahisi_Address = sample[1].ahisi_Address = NULL; |
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|
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// Init audio status and feature flags |
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AudioStatus.sample_rate = audio_sample_rates[0]; |
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AudioStatus.sample_size = audio_sample_sizes[0]; |
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AudioStatus.channels = audio_channel_counts[0]; |
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> |
audio_channel_counts.push_back(2); |
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// set_audio_status_format(); |
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AudioStatus.mixer = 0; |
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AudioStatus.num_sources = 0; |
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audio_component_flags = cmpWantsRegisterMessage | kStereoOut | k16BitOut; |
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return; |
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} |
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|
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ULONG max_channels, sample_rate, frequencies, sample_rate_index; |
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|
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AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
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AHIDB_MaxChannels, (ULONG) &max_channels, |
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AHIDB_Frequencies, (ULONG) &frequencies, |
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TAG_END); |
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|
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D(bug("AudioInit: max_channels=%ld frequencies=%ld\n", max_channels, frequencies)); |
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|
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for (int n=0; n<frequencies; n++) |
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{ |
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AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
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AHIDB_FrequencyArg, n, |
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AHIDB_Frequency, (ULONG) &sample_rate, |
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TAG_END); |
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|
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D(bug("AudioInit: f=%ld Hz\n", sample_rate)); |
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audio_sample_rates.push_back(sample_rate << 16); |
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} |
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|
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ULONG sample_size_bits = 16; |
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|
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D(bug("AudioInit: sampe_rates=%ld\n", audio_sample_rates.size() )); |
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|
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// get index of sample rate closest to 22050 Hz |
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AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
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AHIDB_IndexArg, 22050, |
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AHIDB_Bits, (ULONG) &sample_size_bits, |
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AHIDB_Index, (ULONG) &sample_rate_index, |
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AHIDB_Volume, (ULONG) &supports_volume_changes, |
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AHIDB_Panning, (ULONG) &supports_stereo_panning, |
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TAG_END); |
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|
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audio_sample_sizes.push_back(16); |
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|
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set_audio_status_format(sample_rate_index); |
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|
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// 2048 frames per block |
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audio_frames_per_block = 2048; |
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sound_buffer_size = (AudioStatus.sample_size >> 3) * AudioStatus.channels * audio_frames_per_block; |
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sample[1].ahisi_Address = AllocVec(sound_buffer_size, MEMF_PUBLIC | MEMF_CLEAR); |
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if (sample[0].ahisi_Address == NULL || sample[1].ahisi_Address == NULL) |
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return; |
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|
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AHI_LoadSound(0, AHIST_DYNAMICSAMPLE, &sample[0], ahi_ctrl); |
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AHI_LoadSound(1, AHIST_DYNAMICSAMPLE, &sample[1], ahi_ctrl); |
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|
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// Set parameters |
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play_buf = 0; |
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AHI_SetVol(0, 0x10000, 0x8000, ahi_ctrl, AHISF_IMM); |
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current_main_volume = current_speaker_volume = 0x10000; |
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AHI_SetVol(0, current_speaker_volume, 0x8000, ahi_ctrl, AHISF_IMM); |
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|
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AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, AHISF_IMM); |
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AHI_SetSound(0, play_buf, 0, 0, ahi_ctrl, AHISF_IMM); |
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|
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* AHI sound callback, request next buffer |
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*/ |
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|
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< |
static __saveds __regargs ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/) |
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> |
static __saveds __attribute__((regparm(3))) ULONG audio_callback(struct Hook *hook /*a0*/, struct AHISoundMessage *msg /*a1*/, struct AHIAudioCtrl *ahi_ctrl /*a2*/) |
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{ |
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play_buf ^= 1; |
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|
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// New buffer available? |
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if (audio_block_fetched) { |
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> |
if (audio_block_fetched) |
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> |
{ |
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audio_block_fetched--; |
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|
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< |
// Get size of audio data |
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< |
uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo); |
| 241 |
< |
if (apple_stream_info) { |
| 242 |
< |
int work_size = ReadMacInt32(apple_stream_info + scd_sampleCount) * (AudioStatus.sample_size >> 3) * AudioStatus.channels; |
| 243 |
< |
D(bug("stream: work_size %d\n", work_size)); |
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< |
if (work_size > sound_buffer_size) |
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< |
work_size = sound_buffer_size; |
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< |
|
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< |
// Put data into AHI buffer (convert 8-bit data unsigned->signed) |
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< |
if (AudioStatus.sample_size == 16) |
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< |
memcpy(sample[play_buf].ahisi_Address, Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)), work_size); |
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< |
else { |
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< |
uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)); |
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< |
uint32 *q = (uint32 *)sample[play_buf].ahisi_Address; |
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< |
int r = work_size >> 2; |
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< |
while (r--) |
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< |
*q++ = *p++ ^ 0x80808080; |
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> |
if (main_mute || speaker_mute) |
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> |
{ |
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> |
memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size); |
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> |
} |
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else |
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{ |
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> |
// Get size of audio data |
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> |
uint32 apple_stream_info = ReadMacInt32(audio_data + adatStreamInfo); |
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> |
if (apple_stream_info) { |
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> |
int32 sample_count = ReadMacInt32(apple_stream_info + scd_sampleCount); |
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> |
|
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> |
uint32 num_channels = ReadMacInt16(apple_stream_info + scd_numChannels); |
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> |
uint32 sample_size = ReadMacInt16(apple_stream_info + scd_sampleSize); |
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> |
uint32 sample_rate = ReadMacInt32(apple_stream_info + scd_sampleRate); |
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> |
|
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> |
D(bug("stream: sample_count=%ld num_channels=%ld sample_size=%ld sample_rate=%ld\n", sample_count, num_channels, sample_size, sample_rate >> 16)); |
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> |
|
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> |
// Yes, this can happen. |
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> |
if(sample_count != 0) { |
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> |
if(sample_rate != AudioStatus.sample_rate) { |
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> |
audio_set_sample_rate_byval(sample_rate); |
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> |
} |
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> |
if(num_channels != AudioStatus.channels) { |
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> |
audio_set_channels_byval(num_channels); |
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> |
} |
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> |
if(sample_size != AudioStatus.sample_size) { |
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> |
audio_set_sample_size_byval(sample_size); |
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> |
} |
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> |
} |
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> |
|
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> |
if (sample_count < 0) |
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> |
sample_count = 0; |
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> |
|
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> |
int work_size = sample_count * num_channels * (sample_size>>3); |
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> |
D(bug("stream: work_size=%ld sound_buffer_size=%ld\n", work_size, sound_buffer_size)); |
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> |
|
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> |
if (work_size > sound_buffer_size) |
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> |
work_size = sound_buffer_size; |
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> |
|
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> |
// Put data into AHI buffer (convert 8-bit data unsigned->signed) |
| 279 |
> |
if (AudioStatus.sample_size == 16) |
| 280 |
> |
Mac2Host_memcpy(sample[play_buf].ahisi_Address, ReadMacInt32(apple_stream_info + scd_buffer), work_size); |
| 281 |
> |
else { |
| 282 |
> |
uint32 *p = (uint32 *)Mac2HostAddr(ReadMacInt32(apple_stream_info + scd_buffer)); |
| 283 |
> |
uint32 *q = (uint32 *)sample[play_buf].ahisi_Address; |
| 284 |
> |
int r = work_size >> 2; |
| 285 |
> |
while (r--) |
| 286 |
> |
*q++ = *p++ ^ 0x80808080; |
| 287 |
> |
} |
| 288 |
> |
if (work_size != sound_buffer_size) |
| 289 |
> |
memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size); |
| 290 |
|
} |
| 203 |
– |
if (work_size != sound_buffer_size) |
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– |
memset((uint8 *)sample[play_buf].ahisi_Address + work_size, 0, sound_buffer_size - work_size); |
| 291 |
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} |
| 292 |
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|
| 293 |
< |
} else |
| 293 |
> |
} |
| 294 |
> |
else |
| 295 |
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memset(sample[play_buf].ahisi_Address, 0, sound_buffer_size); |
| 296 |
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|
| 297 |
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// Play next buffer |
| 299 |
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|
| 300 |
|
// Trigger audio interrupt to get new buffer |
| 301 |
|
if (AudioStatus.num_sources) { |
| 302 |
< |
D(bug("stream: triggering irq\n")); |
| 302 |
> |
D1(bug("stream: triggering irq\n")); |
| 303 |
|
SetInterruptFlag(INTFLAG_AUDIO); |
| 304 |
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TriggerInterrupt(); |
| 305 |
|
} |
| 313 |
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|
| 314 |
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void AudioInterrupt(void) |
| 315 |
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{ |
| 316 |
< |
D(bug("AudioInterrupt\n")); |
| 316 |
> |
D1(bug("AudioInterrupt\n")); |
| 317 |
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|
| 318 |
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// Get data from apple mixer |
| 319 |
|
if (AudioStatus.mixer) { |
| 321 |
|
r.a[0] = audio_data + adatStreamInfo; |
| 322 |
|
r.a[1] = AudioStatus.mixer; |
| 323 |
|
Execute68k(audio_data + adatGetSourceData, &r); |
| 324 |
< |
D(bug(" GetSourceData() returns %08lx\n", r.d[0])); |
| 324 |
> |
D1(bug(" GetSourceData() returns %08lx\n", r.d[0])); |
| 325 |
|
} else |
| 326 |
|
WriteMacInt32(audio_data + adatStreamInfo, 0); |
| 327 |
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|
| 328 |
|
// Signal stream function |
| 329 |
|
audio_block_fetched++; |
| 330 |
< |
D(bug("AudioInterrupt done\n")); |
| 330 |
> |
D1(bug("AudioInterrupt done\n")); |
| 331 |
|
} |
| 332 |
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|
| 333 |
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|
| 337 |
|
* It is guaranteed that AudioStatus.num_sources == 0 |
| 338 |
|
*/ |
| 339 |
|
|
| 340 |
< |
void audio_set_sample_rate(int index) |
| 340 |
> |
void audio_set_sample_rate_byval(uint32 value) |
| 341 |
|
{ |
| 342 |
+ |
bool changed = (AudioStatus.sample_rate != value); |
| 343 |
+ |
if(changed) |
| 344 |
+ |
{ |
| 345 |
+ |
ULONG sample_rate_index; |
| 346 |
+ |
|
| 347 |
+ |
// get index of sample rate closest to <value> Hz |
| 348 |
+ |
AHI_GetAudioAttrs(ahi_id, ahi_ctrl, |
| 349 |
+ |
AHIDB_IndexArg, value >> 16, |
| 350 |
+ |
AHIDB_Index, (ULONG) &sample_rate_index, |
| 351 |
+ |
TAG_END); |
| 352 |
+ |
|
| 353 |
+ |
D(bug(" audio_set_sample_rate_byval requested rate=%ld Hz\n", value >> 16)); |
| 354 |
+ |
|
| 355 |
+ |
AudioStatus.sample_rate = audio_sample_rates[sample_rate_index]; |
| 356 |
+ |
|
| 357 |
+ |
AHI_SetFreq(0, AudioStatus.sample_rate >> 16, ahi_ctrl, 0); |
| 358 |
+ |
} |
| 359 |
+ |
|
| 360 |
+ |
D(bug(" audio_set_sample_rate_byval rate=%ld Hz\n", AudioStatus.sample_rate >> 16)); |
| 361 |
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} |
| 362 |
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|
| 363 |
< |
void audio_set_sample_size(int index) |
| 363 |
> |
void audio_set_sample_size_byval(uint32 value) |
| 364 |
|
{ |
| 365 |
+ |
bool changed = (AudioStatus.sample_size != value); |
| 366 |
+ |
if(changed) { |
| 367 |
+ |
// AudioStatus.sample_size = value; |
| 368 |
+ |
// update_sound_parameters(); |
| 369 |
+ |
// WritePrivateProfileInt( "Audio", "SampleSize", AudioStatus.sample_size, ini_file_name ); |
| 370 |
+ |
} |
| 371 |
+ |
D(bug(" audio_set_sample_size_byval %d\n", AudioStatus.sample_size)); |
| 372 |
|
} |
| 373 |
|
|
| 374 |
< |
void audio_set_channels(int index) |
| 374 |
> |
void audio_set_channels_byval(uint32 value) |
| 375 |
|
{ |
| 376 |
+ |
bool changed = (AudioStatus.channels != value); |
| 377 |
+ |
if(changed) { |
| 378 |
+ |
// AudioStatus.channels = value; |
| 379 |
+ |
// update_sound_parameters(); |
| 380 |
+ |
// WritePrivateProfileInt( "Audio", "Channels", AudioStatus.channels, ini_file_name ); |
| 381 |
+ |
} |
| 382 |
+ |
D(bug(" audio_set_channels_byval %d\n", AudioStatus.channels)); |
| 383 |
+ |
} |
| 384 |
+ |
|
| 385 |
+ |
bool audio_set_sample_rate(int index) |
| 386 |
+ |
{ |
| 387 |
+ |
if(index >= 0 && index < audio_sample_rates.size() ) { |
| 388 |
+ |
audio_set_sample_rate_byval( audio_sample_rates[index] ); |
| 389 |
+ |
D(bug(" audio_set_sample_rate index=%ld rate=%ld\n", index, AudioStatus.sample_rate >> 16)); |
| 390 |
+ |
} |
| 391 |
+ |
|
| 392 |
+ |
return true; |
| 393 |
+ |
} |
| 394 |
+ |
|
| 395 |
+ |
bool audio_set_sample_size(int index) |
| 396 |
+ |
{ |
| 397 |
+ |
if(index >= 0 && index < audio_sample_sizes.size() ) { |
| 398 |
+ |
audio_set_sample_size_byval( audio_sample_sizes[index] ); |
| 399 |
+ |
D(bug(" audio_set_sample_size %d,%d\n", index,AudioStatus.sample_size)); |
| 400 |
+ |
} |
| 401 |
+ |
|
| 402 |
+ |
return true; |
| 403 |
+ |
} |
| 404 |
+ |
|
| 405 |
+ |
bool audio_set_channels(int index) |
| 406 |
+ |
{ |
| 407 |
+ |
if(index >= 0 && index < audio_channel_counts.size() ) { |
| 408 |
+ |
audio_set_channels_byval( audio_channel_counts[index] ); |
| 409 |
+ |
D(bug(" audio_set_channels %d,%d\n", index,AudioStatus.channels)); |
| 410 |
+ |
} |
| 411 |
+ |
|
| 412 |
+ |
return true; |
| 413 |
|
} |
| 414 |
|
|
| 415 |
|
|
| 421 |
|
|
| 422 |
|
bool audio_get_main_mute(void) |
| 423 |
|
{ |
| 424 |
< |
return false; |
| 424 |
> |
D(bug("audio_get_main_mute: mute=%ld\n", main_mute)); |
| 425 |
> |
|
| 426 |
> |
return main_mute; |
| 427 |
|
} |
| 428 |
|
|
| 429 |
|
uint32 audio_get_main_volume(void) |
| 430 |
|
{ |
| 431 |
+ |
D(bug("audio_get_main_volume\n")); |
| 432 |
+ |
|
| 433 |
+ |
ULONG volume = current_main_volume >> 8; // 0x10000 => 0x100 |
| 434 |
+ |
|
| 435 |
+ |
D(bug("audio_get_main_volume: volume=%08lx\n", volume)); |
| 436 |
+ |
|
| 437 |
+ |
return (volume << 16) + volume; |
| 438 |
+ |
|
| 439 |
|
return 0x01000100; |
| 440 |
|
} |
| 441 |
|
|
| 442 |
|
bool audio_get_speaker_mute(void) |
| 443 |
|
{ |
| 444 |
< |
return false; |
| 444 |
> |
D(bug("audio_get_speaker_mute: mute=%ld\n", speaker_mute)); |
| 445 |
> |
|
| 446 |
> |
return speaker_mute; |
| 447 |
|
} |
| 448 |
|
|
| 449 |
|
uint32 audio_get_speaker_volume(void) |
| 450 |
|
{ |
| 451 |
+ |
D(bug("audio_get_speaker_volume: \n")); |
| 452 |
+ |
|
| 453 |
+ |
if (audio_open) |
| 454 |
+ |
{ |
| 455 |
+ |
ULONG volume = current_speaker_volume >> 8; // 0x10000 => 0x100 |
| 456 |
+ |
|
| 457 |
+ |
D(bug("audio_get_speaker_volume: volume=%08lx\n", volume)); |
| 458 |
+ |
|
| 459 |
+ |
return (volume << 16) + volume; |
| 460 |
+ |
} |
| 461 |
+ |
|
| 462 |
|
return 0x01000100; |
| 463 |
|
} |
| 464 |
|
|
| 465 |
|
void audio_set_main_mute(bool mute) |
| 466 |
|
{ |
| 467 |
+ |
D(bug("audio_set_main_mute: mute=%ld\n", mute)); |
| 468 |
+ |
|
| 469 |
+ |
if (mute != main_mute) |
| 470 |
+ |
{ |
| 471 |
+ |
main_mute = mute; |
| 472 |
+ |
} |
| 473 |
|
} |
| 474 |
|
|
| 475 |
|
void audio_set_main_volume(uint32 vol) |
| 476 |
|
{ |
| 477 |
+ |
D(bug("audio_set_main_volume: vol=%08lx\n", vol)); |
| 478 |
+ |
|
| 479 |
+ |
if (audio_open && supports_volume_changes) |
| 480 |
+ |
{ |
| 481 |
+ |
ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff)); |
| 482 |
+ |
|
| 483 |
+ |
D(bug("audio_set_main_volume: volume=%08lx\n", volume)); |
| 484 |
+ |
|
| 485 |
+ |
current_main_volume = volume; |
| 486 |
+ |
|
| 487 |
+ |
AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM); |
| 488 |
+ |
} |
| 489 |
|
} |
| 490 |
|
|
| 491 |
|
void audio_set_speaker_mute(bool mute) |
| 492 |
|
{ |
| 493 |
+ |
D(bug("audio_set_speaker_mute: mute=%ld\n", mute)); |
| 494 |
+ |
|
| 495 |
+ |
if (mute != speaker_mute) |
| 496 |
+ |
{ |
| 497 |
+ |
speaker_mute = mute; |
| 498 |
+ |
} |
| 499 |
|
} |
| 500 |
|
|
| 501 |
|
void audio_set_speaker_volume(uint32 vol) |
| 502 |
|
{ |
| 503 |
+ |
D(bug("audio_set_speaker_volume: vol=%08lx\n", vol)); |
| 504 |
+ |
|
| 505 |
+ |
if (audio_open && supports_volume_changes) |
| 506 |
+ |
{ |
| 507 |
+ |
ULONG volume = 0x80 * ((vol >> 16) + (vol & 0xffff)); |
| 508 |
+ |
|
| 509 |
+ |
D(bug("audio_set_speaker_volume: volume=%08lx\n", volume)); |
| 510 |
+ |
|
| 511 |
+ |
current_speaker_volume = volume; |
| 512 |
+ |
|
| 513 |
+ |
AHI_SetVol(0, volume, 0x8000, ahi_ctrl, AHISF_IMM); |
| 514 |
+ |
} |
| 515 |
|
} |
