664 lines
26 KiB
C
664 lines
26 KiB
C
/* gameplaySP
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*
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* Copyright (C) 2006 Exophase <exophase@gmail.com>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License as
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* published by the Free Software Foundation; either version 2 of
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* the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "common.h"
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u32 global_enable_audio = 1;
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direct_sound_struct direct_sound_channel[2];
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gbc_sound_struct gbc_sound_channel[4];
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u32 sound_frequency = GBA_SOUND_FREQUENCY;
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u32 sound_on;
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static s16 sound_buffer[BUFFER_SIZE];
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static u32 sound_buffer_base;
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static u32 sound_last_cpu_ticks;
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static fixed16_16 gbc_sound_tick_step;
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/* Queue 1 sample to the top of the DS FIFO, wrap around circularly */
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void sound_timer_queue8(u32 channel, u8 value)
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{
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direct_sound_struct *ds = direct_sound_channel + channel;
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*((s8 *)(ds->fifo + ds->fifo_top)) = value;
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ds->fifo_top = (ds->fifo_top + 1) % 32;
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}
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/* Queue 2 samples to the top of the DS FIFO, wrap around circularly */
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void sound_timer_queue16(u32 channel, u16 value)
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{
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direct_sound_struct *ds = direct_sound_channel + channel;
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*((s8 *)(ds->fifo + ds->fifo_top)) = value & 0xFF;
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ds->fifo_top = (ds->fifo_top + 1) % 32;
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*((s8 *)(ds->fifo + ds->fifo_top)) = value >> 8;
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ds->fifo_top = (ds->fifo_top + 1) % 32;
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}
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/* Queue 4 samples to the top of the DS FIFO, wrap around circularly */
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void sound_timer_queue32(u32 channel, u32 value)
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{
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direct_sound_struct *ds = direct_sound_channel + channel;
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*((s8 *)(ds->fifo + ds->fifo_top)) = value & 0xFF;
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ds->fifo_top = (ds->fifo_top + 1) % 32;
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*((s8 *)(ds->fifo + ds->fifo_top)) = (value >> 8) & 0xFF;
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ds->fifo_top = (ds->fifo_top + 1) % 32;
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*((s8 *)(ds->fifo + ds->fifo_top)) = (value >> 16) & 0xFF;
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ds->fifo_top = (ds->fifo_top + 1) % 32;
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*((s8 *)(ds->fifo + ds->fifo_top)) = (value >> 24);
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ds->fifo_top = (ds->fifo_top + 1) % 32;
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}
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void sound_timer(fixed8_24 frequency_step, u32 channel)
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{
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direct_sound_status_type sample_status = DIRECT_SOUND_INACTIVE;
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direct_sound_struct *ds = direct_sound_channel + channel;
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fixed8_24 fifo_fractional = ds->fifo_fractional;
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u32 buffer_index = ds->buffer_index;
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s16 current_sample, next_sample;
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current_sample = ds->fifo[ds->fifo_base] * 16;
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ds->fifo_base = (ds->fifo_base + 1) % 32;
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next_sample = ds->fifo[ds->fifo_base] * 16;
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if(sound_on == 1)
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{
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if(ds->volume == DIRECT_SOUND_VOLUME_50)
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{
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current_sample >>= 1;
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next_sample >>= 1;
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}
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sample_status = ds->status;
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}
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// Unqueue 1 sample from the base of the DS FIFO and place it on the audio
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// buffer for as many samples as necessary. If the DS FIFO is 16 bytes or
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// smaller and if DMA is enabled for the sound channel initiate a DMA transfer
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// to the DS FIFO.
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switch(sample_status)
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{
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case DIRECT_SOUND_INACTIVE:
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/* render samples NULL */
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while(fifo_fractional <= 0xFFFFFF)
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{
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fifo_fractional += frequency_step;
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buffer_index = (buffer_index + 2) % BUFFER_SIZE;
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}
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break;
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case DIRECT_SOUND_RIGHT:
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/* render samples RIGHT */
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while(fifo_fractional <= 0xFFFFFF)
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{
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s16 dest_sample = current_sample +
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fp16_16_to_u32((next_sample - current_sample) * (fifo_fractional >> 8));
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sound_buffer[buffer_index + 1] += dest_sample;
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fifo_fractional += frequency_step;
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buffer_index = (buffer_index + 2) % BUFFER_SIZE;
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}
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break;
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case DIRECT_SOUND_LEFT:
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/* render samples LEFT */
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while(fifo_fractional <= 0xFFFFFF)
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{
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s16 dest_sample = current_sample +
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fp16_16_to_u32((next_sample - current_sample) * (fifo_fractional >> 8));
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sound_buffer[buffer_index] += dest_sample;
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fifo_fractional += frequency_step;
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buffer_index = (buffer_index + 2) % BUFFER_SIZE;
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}
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break;
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case DIRECT_SOUND_LEFTRIGHT:
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/* render samples LEFT and RIGHT. */
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while(fifo_fractional <= 0xFFFFFF)
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{
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s16 dest_sample = current_sample +
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fp16_16_to_u32((next_sample - current_sample) * (fifo_fractional >> 8));
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sound_buffer[buffer_index] += dest_sample;
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sound_buffer[buffer_index + 1] += dest_sample;
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fifo_fractional += frequency_step;
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buffer_index = (buffer_index + 2) % BUFFER_SIZE;
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}
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break;
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}
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ds->buffer_index = buffer_index;
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ds->fifo_fractional = fp8_24_fractional_part(fifo_fractional);
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if(((ds->fifo_top - ds->fifo_base) % 32) <= 16)
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{
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if(dma[1].direct_sound_channel == channel)
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dma_transfer(dma + 1);
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if(dma[2].direct_sound_channel == channel)
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dma_transfer(dma + 2);
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}
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}
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void sound_reset_fifo(u32 channel)
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{
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direct_sound_struct *ds = direct_sound_channel;
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memset(ds->fifo, 0, 32);
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}
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// Initial pattern data = 4bits (signed)
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// Channel volume = 12bits
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// Envelope volume = 14bits
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// Master volume = 2bits
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// Recalculate left and right volume as volume changes.
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// To calculate the current sample, use (sample * volume) >> 16
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// Square waves range from -8 (low) to 7 (high)
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const s8 square_pattern_duty[4][8] =
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{
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{ -8, -8, -8, -8, 7, -8, -8, -8 },
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{ -8, -8, -8, -8, 7, 7, -8, -8 },
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{ -8, -8, 7, 7, 7, 7, -8, -8 },
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{ 7, 7, 7, 7, -8, -8, 7, 7 },
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};
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s8 wave_samples[64];
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u32 noise_table15[1024];
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u32 noise_table7[4];
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u32 gbc_sound_master_volume_table[4] = { 1, 2, 4, 0 };
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u32 gbc_sound_channel_volume_table[8] =
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{
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fixed_div(0, 7, 12),
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fixed_div(1, 7, 12),
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fixed_div(2, 7, 12),
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fixed_div(3, 7, 12),
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fixed_div(4, 7, 12),
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fixed_div(5, 7, 12),
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fixed_div(6, 7, 12),
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fixed_div(7, 7, 12)
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};
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u32 gbc_sound_envelope_volume_table[16] =
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{
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fixed_div(0, 15, 14),
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fixed_div(1, 15, 14),
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fixed_div(2, 15, 14),
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fixed_div(3, 15, 14),
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fixed_div(4, 15, 14),
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fixed_div(5, 15, 14),
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fixed_div(6, 15, 14),
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fixed_div(7, 15, 14),
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fixed_div(8, 15, 14),
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fixed_div(9, 15, 14),
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fixed_div(10, 15, 14),
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fixed_div(11, 15, 14),
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fixed_div(12, 15, 14),
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fixed_div(13, 15, 14),
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fixed_div(14, 15, 14),
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fixed_div(15, 15, 14)
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};
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u32 gbc_sound_buffer_index = 0;
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u32 gbc_sound_last_cpu_ticks = 0;
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u32 gbc_sound_partial_ticks = 0;
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u32 gbc_sound_master_volume_left;
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u32 gbc_sound_master_volume_right;
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u32 gbc_sound_master_volume;
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#define update_volume_channel_envelope(channel) \
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volume_##channel = gbc_sound_envelope_volume_table[envelope_volume] * \
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gbc_sound_channel_volume_table[gbc_sound_master_volume_##channel] * \
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gbc_sound_master_volume_table[gbc_sound_master_volume] \
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#define update_volume_channel_noenvelope(channel) \
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volume_##channel = gs->wave_volume * \
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gbc_sound_channel_volume_table[gbc_sound_master_volume_##channel] * \
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gbc_sound_master_volume_table[gbc_sound_master_volume] \
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#define update_volume(type) \
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update_volume_channel_##type(left); \
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update_volume_channel_##type(right) \
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#define update_tone_sweep() \
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if(gs->sweep_status) \
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{ \
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u32 sweep_ticks = gs->sweep_ticks - 1; \
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\
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if(sweep_ticks == 0) \
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{ \
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u32 rate = gs->rate; \
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\
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if(gs->sweep_direction) \
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rate = rate - (rate >> gs->sweep_shift); \
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else \
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rate = rate + (rate >> gs->sweep_shift); \
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\
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if(rate > 2047) { \
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rate = 2047; \
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gs->active_flag = 0; \
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break; \
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} \
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\
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frequency_step = float_to_fp16_16(((131072.0f / (2048 - rate)) * 8.0f) \
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/ sound_frequency); \
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\
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gs->frequency_step = frequency_step; \
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gs->rate = rate; \
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\
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sweep_ticks = gs->sweep_initial_ticks; \
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} \
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gs->sweep_ticks = sweep_ticks; \
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} \
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#define update_tone_nosweep() \
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#define update_tone_envelope() \
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if(gs->envelope_status) \
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{ \
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u32 envelope_ticks = gs->envelope_ticks - 1; \
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envelope_volume = gs->envelope_volume; \
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\
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if(envelope_ticks == 0) \
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{ \
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if(gs->envelope_direction) \
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{ \
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if(envelope_volume != 15) \
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envelope_volume = gs->envelope_volume + 1; \
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} \
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else \
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{ \
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if(envelope_volume != 0) \
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envelope_volume = gs->envelope_volume - 1; \
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} \
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\
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update_volume(envelope); \
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\
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gs->envelope_volume = envelope_volume; \
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gs->envelope_ticks = gs->envelope_initial_ticks; \
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} \
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else \
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gs->envelope_ticks = envelope_ticks; \
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} \
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#define update_tone_noenvelope() \
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#define update_tone_counters(envelope_op, sweep_op) \
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tick_counter += gbc_sound_tick_step; \
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if(tick_counter > 0xFFFF) \
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{ \
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if(gs->length_status) \
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{ \
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u32 length_ticks = gs->length_ticks - 1; \
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gs->length_ticks = length_ticks; \
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\
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if(length_ticks == 0) \
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{ \
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gs->active_flag = 0; \
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break; \
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} \
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} \
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\
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update_tone_##envelope_op(); \
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update_tone_##sweep_op(); \
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\
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tick_counter &= 0xFFFF; \
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} \
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#define gbc_sound_render_sample_right() \
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sound_buffer[buffer_index + 1] += (current_sample * volume_right) >> 22 \
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#define gbc_sound_render_sample_left() \
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sound_buffer[buffer_index] += (current_sample * volume_left) >> 22 \
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#define gbc_sound_render_sample_both() \
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gbc_sound_render_sample_right(); \
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gbc_sound_render_sample_left() \
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#define gbc_sound_render_samples(type, sample_length, envelope_op, sweep_op) \
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for(i = 0; i < buffer_ticks; i++) \
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{ \
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current_sample = \
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sample_data[fp16_16_to_u32(sample_index) % sample_length]; \
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gbc_sound_render_sample_##type(); \
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\
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sample_index += frequency_step; \
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buffer_index = (buffer_index + 2) % BUFFER_SIZE; \
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\
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update_tone_counters(envelope_op, sweep_op); \
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} \
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#define gbc_noise_wrap_full 32767
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#define gbc_noise_wrap_half 126
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#define get_noise_sample_full() \
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current_sample = \
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((s32)(noise_table15[fp16_16_to_u32(sample_index) >> 5] << \
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(fp16_16_to_u32(sample_index) & 0x1F)) >> 31) & 0x0F \
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#define get_noise_sample_half() \
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current_sample = \
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((s32)(noise_table7[fp16_16_to_u32(sample_index) >> 5] << \
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(fp16_16_to_u32(sample_index) & 0x1F)) >> 31) & 0x0F \
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#define gbc_sound_render_noise(type, noise_type, envelope_op, sweep_op) \
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for(i = 0; i < buffer_ticks; i++) \
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{ \
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get_noise_sample_##noise_type(); \
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gbc_sound_render_sample_##type(); \
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\
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sample_index += frequency_step; \
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\
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if(sample_index >= u32_to_fp16_16(gbc_noise_wrap_##noise_type)) \
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sample_index -= u32_to_fp16_16(gbc_noise_wrap_##noise_type); \
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\
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buffer_index = (buffer_index + 2) % BUFFER_SIZE; \
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update_tone_counters(envelope_op, sweep_op); \
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} \
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#define gbc_sound_render_channel(type, sample_length, envelope_op, sweep_op) \
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buffer_index = gbc_sound_buffer_index; \
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sample_index = gs->sample_index; \
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frequency_step = gs->frequency_step; \
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tick_counter = gs->tick_counter; \
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\
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update_volume(envelope_op); \
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\
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switch(gs->status) \
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{ \
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case GBC_SOUND_INACTIVE: \
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break; \
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\
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case GBC_SOUND_LEFT: \
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gbc_sound_render_##type(left, sample_length, envelope_op, sweep_op); \
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break; \
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\
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case GBC_SOUND_RIGHT: \
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gbc_sound_render_##type(right, sample_length, envelope_op, sweep_op); \
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break; \
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\
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case GBC_SOUND_LEFTRIGHT: \
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gbc_sound_render_##type(both, sample_length, envelope_op, sweep_op); \
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break; \
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} \
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\
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gs->sample_index = sample_index; \
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gs->tick_counter = tick_counter; \
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void update_gbc_sound(u32 cpu_ticks)
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{
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fixed16_16 buffer_ticks = float_to_fp16_16((float)(cpu_ticks -
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gbc_sound_last_cpu_ticks) * sound_frequency / GBC_BASE_RATE);
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u32 i, i2;
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gbc_sound_struct *gs = gbc_sound_channel;
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fixed16_16 sample_index, frequency_step;
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fixed16_16 tick_counter;
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u32 buffer_index;
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s32 volume_left, volume_right;
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u32 envelope_volume;
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s32 current_sample;
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u16 sound_status = read_ioreg(REG_SOUNDCNT_X) & 0xFFF0;
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const s8 *sample_data;
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s8 *wave_bank;
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u8 *wave_ram = ((u8 *)io_registers) + 0x90;
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gbc_sound_partial_ticks += fp16_16_fractional_part(buffer_ticks);
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buffer_ticks = fp16_16_to_u32(buffer_ticks);
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if(gbc_sound_partial_ticks > 0xFFFF)
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{
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buffer_ticks += 1;
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gbc_sound_partial_ticks &= 0xFFFF;
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}
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if(sound_on == 1)
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{
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gs = gbc_sound_channel + 0;
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if(gs->active_flag)
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{
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sound_status |= 0x01;
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sample_data = gs->sample_data;
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envelope_volume = gs->envelope_volume;
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gbc_sound_render_channel(samples, 8, envelope, sweep);
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}
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gs = gbc_sound_channel + 1;
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if(gs->active_flag)
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{
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sound_status |= 0x02;
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sample_data = gs->sample_data;
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envelope_volume = gs->envelope_volume;
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gbc_sound_render_channel(samples, 8, envelope, nosweep);
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}
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gs = gbc_sound_channel + 2;
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if(gbc_sound_wave_update)
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{
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unsigned bank = (gs->wave_bank == 1) ? 1 : 0;
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wave_bank = wave_samples + (bank * 32);
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for(i = 0, i2 = 0; i < 16; i++, i2 += 2)
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{
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current_sample = wave_ram[i];
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wave_bank[i2] = (((current_sample >> 4) & 0x0F) - 8);
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wave_bank[i2 + 1] = ((current_sample & 0x0F) - 8);
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}
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gbc_sound_wave_update = 0;
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}
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if((gs->active_flag) && (gs->master_enable))
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{
|
|
sound_status |= 0x04;
|
|
sample_data = wave_samples;
|
|
if(gs->wave_type == 0)
|
|
{
|
|
if(gs->wave_bank == 1)
|
|
sample_data += 32;
|
|
|
|
gbc_sound_render_channel(samples, 32, noenvelope, nosweep);
|
|
}
|
|
else
|
|
{
|
|
gbc_sound_render_channel(samples, 64, noenvelope, nosweep);
|
|
}
|
|
}
|
|
|
|
gs = gbc_sound_channel + 3;
|
|
if(gs->active_flag)
|
|
{
|
|
sound_status |= 0x08;
|
|
envelope_volume = gs->envelope_volume;
|
|
|
|
if(gs->noise_type == 1)
|
|
{
|
|
gbc_sound_render_channel(noise, half, envelope, nosweep);
|
|
}
|
|
else
|
|
{
|
|
gbc_sound_render_channel(noise, full, envelope, nosweep);
|
|
}
|
|
}
|
|
}
|
|
|
|
write_ioreg(REG_SOUNDCNT_X, sound_status);
|
|
|
|
gbc_sound_last_cpu_ticks = cpu_ticks;
|
|
gbc_sound_buffer_index =
|
|
(gbc_sound_buffer_index + (buffer_ticks * 2)) % BUFFER_SIZE;
|
|
}
|
|
|
|
// Special thanks to blarrg for the LSFR frequency used in Meridian, as posted
|
|
// on the forum at http://meridian.overclocked.org:
|
|
// http://meridian.overclocked.org/cgi-bin/wwwthreads/showpost.pl?Board=merid
|
|
// angeneraldiscussion&Number=2069&page=0&view=expanded&mode=threaded&sb=4
|
|
// Hope you don't mind me borrowing it ^_-
|
|
|
|
static void init_noise_table(u32 *table, u32 period, u32 bit_length)
|
|
{
|
|
u32 shift_register = 0xFF;
|
|
u32 mask = ~(1 << bit_length);
|
|
s32 table_pos, bit_pos;
|
|
u32 current_entry;
|
|
u32 table_period = (period + 31) / 32;
|
|
|
|
// Bits are stored in reverse order so they can be more easily moved to
|
|
// bit 31, for sign extended shift down.
|
|
|
|
for(table_pos = 0; table_pos < table_period; table_pos++)
|
|
{
|
|
current_entry = 0;
|
|
for(bit_pos = 31; bit_pos >= 0; bit_pos--)
|
|
{
|
|
current_entry |= (shift_register & 0x01) << bit_pos;
|
|
|
|
shift_register =
|
|
((1 & (shift_register ^ (shift_register >> 1))) << bit_length) |
|
|
((shift_register >> 1) & mask);
|
|
}
|
|
|
|
table[table_pos] = current_entry;
|
|
}
|
|
}
|
|
|
|
void reset_sound(void)
|
|
{
|
|
direct_sound_struct *ds = direct_sound_channel;
|
|
gbc_sound_struct *gs = gbc_sound_channel;
|
|
u32 i;
|
|
|
|
sound_on = 0;
|
|
sound_buffer_base = 0;
|
|
sound_last_cpu_ticks = 0;
|
|
memset(sound_buffer, 0, sizeof(sound_buffer));
|
|
|
|
for(i = 0; i < 2; i++, ds++)
|
|
{
|
|
ds->buffer_index = 0;
|
|
ds->status = DIRECT_SOUND_INACTIVE;
|
|
ds->fifo_top = 0;
|
|
ds->fifo_base = 0;
|
|
ds->fifo_fractional = 0;
|
|
ds->last_cpu_ticks = 0;
|
|
memset(ds->fifo, 0, 32);
|
|
}
|
|
|
|
gbc_sound_buffer_index = 0;
|
|
gbc_sound_last_cpu_ticks = 0;
|
|
gbc_sound_partial_ticks = 0;
|
|
|
|
gbc_sound_master_volume_left = 0;
|
|
gbc_sound_master_volume_right = 0;
|
|
gbc_sound_master_volume = 0;
|
|
memset(wave_samples, 0, 64);
|
|
|
|
for(i = 0; i < 4; i++, gs++)
|
|
{
|
|
gs->status = GBC_SOUND_INACTIVE;
|
|
gs->sample_data = square_pattern_duty[2];
|
|
gs->active_flag = 0;
|
|
}
|
|
}
|
|
|
|
void init_sound(int need_reset)
|
|
{
|
|
gbc_sound_tick_step =
|
|
float_to_fp16_16(256.0f / sound_frequency);
|
|
|
|
init_noise_table(noise_table15, 32767, 14);
|
|
init_noise_table(noise_table7, 127, 6);
|
|
|
|
if (need_reset)
|
|
reset_sound();
|
|
}
|
|
|
|
#define sound_savestate_builder(type) \
|
|
void sound_##type##_savestate(void) \
|
|
{ \
|
|
state_mem_##type##_variable(sound_on); \
|
|
state_mem_##type##_variable(sound_buffer_base); \
|
|
state_mem_##type##_variable(sound_last_cpu_ticks); \
|
|
state_mem_##type##_variable(gbc_sound_buffer_index); \
|
|
state_mem_##type##_variable(gbc_sound_last_cpu_ticks); \
|
|
state_mem_##type##_variable(gbc_sound_partial_ticks); \
|
|
state_mem_##type##_variable(gbc_sound_master_volume_left); \
|
|
state_mem_##type##_variable(gbc_sound_master_volume_right); \
|
|
state_mem_##type##_variable(gbc_sound_master_volume); \
|
|
state_mem_##type##_array(wave_samples); \
|
|
state_mem_##type##_array(direct_sound_channel); \
|
|
state_mem_##type##_array(gbc_sound_channel); \
|
|
}
|
|
|
|
sound_savestate_builder(read)
|
|
sound_savestate_builder(write)
|
|
|
|
|
|
#include "libretro.h"
|
|
|
|
static retro_audio_sample_batch_t audio_batch_cb;
|
|
void retro_set_audio_sample(retro_audio_sample_t cb) { }
|
|
void retro_set_audio_sample_batch(retro_audio_sample_batch_t cb) { audio_batch_cb = cb; }
|
|
|
|
void render_audio(void)
|
|
{
|
|
static s16 stream_base[512];
|
|
s16 *source;
|
|
u32 i;
|
|
|
|
while (((gbc_sound_buffer_index - sound_buffer_base) & BUFFER_SIZE_MASK) > 512)
|
|
{
|
|
source = (s16 *)(sound_buffer + sound_buffer_base);
|
|
for(i = 0; i < 512; i++)
|
|
{
|
|
s32 current_sample = source[i];
|
|
if(current_sample > 2047)
|
|
current_sample = 2047;
|
|
if(current_sample < -2048)
|
|
current_sample = -2048;
|
|
stream_base[i] = current_sample * 16;
|
|
source[i] = 0;
|
|
}
|
|
audio_batch_cb(stream_base, 256);
|
|
sound_buffer_base += 512;
|
|
sound_buffer_base &= BUFFER_SIZE_MASK;
|
|
}
|
|
}
|