/* gameplaySP * * Copyright (C) 2006 Exophase * * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "common.h" direct_sound_struct direct_sound_channel[2]; gbc_sound_struct gbc_sound_channel[4]; const u32 sound_frequency = GBA_SOUND_FREQUENCY; u32 sound_on; static s16 sound_buffer[BUFFER_SIZE]; static u32 sound_buffer_base; static fixed16_16 gbc_sound_tick_step; /* Queue 4 samples to the top of the DS FIFO, wrap around circularly */ void sound_timer_queue32(u32 channel, u32 value) { direct_sound_struct *ds = &direct_sound_channel[channel]; ds->fifo[ds->fifo_top++] = value & 0xFF; ds->fifo_top &= 31; ds->fifo[ds->fifo_top++] = (value >> 8) & 0xFF; ds->fifo_top &= 31; ds->fifo[ds->fifo_top++] = (value >> 16) & 0xFF; ds->fifo_top &= 31; ds->fifo[ds->fifo_top++] = (value >> 24); ds->fifo_top &= 31; } unsigned sound_timer(fixed8_24 frequency_step, u32 channel) { int ret = 0; u32 sample_status = DIRECT_SOUND_INACTIVE; direct_sound_struct *ds = &direct_sound_channel[channel]; fixed8_24 fifo_fractional = ds->fifo_fractional; u32 buffer_index = ds->buffer_index; s16 current_sample, next_sample; current_sample = ds->fifo[ds->fifo_base] * 16; ds->fifo_base = (ds->fifo_base + 1) % 32; next_sample = ds->fifo[ds->fifo_base] * 16; if(sound_on == 1) { current_sample >>= ds->volume_halve; next_sample >>= ds->volume_halve; sample_status = ds->status; } // Unqueue 1 sample from the base of the DS FIFO and place it on the audio // buffer for as many samples as necessary. If the DS FIFO is 16 bytes or // smaller and if DMA is enabled for the sound channel initiate a DMA transfer // to the DS FIFO. switch(sample_status) { case DIRECT_SOUND_INACTIVE: /* render samples NULL */ while(fifo_fractional <= 0xFFFFFF) { fifo_fractional += frequency_step; buffer_index = (buffer_index + 2) % BUFFER_SIZE; } break; case DIRECT_SOUND_RIGHT: /* render samples RIGHT */ while(fifo_fractional <= 0xFFFFFF) { s16 dest_sample = current_sample + fp16_16_to_u32((next_sample - current_sample) * (fifo_fractional >> 8)); sound_buffer[buffer_index + 1] += dest_sample; fifo_fractional += frequency_step; buffer_index = (buffer_index + 2) % BUFFER_SIZE; } break; case DIRECT_SOUND_LEFT: /* render samples LEFT */ while(fifo_fractional <= 0xFFFFFF) { s16 dest_sample = current_sample + fp16_16_to_u32((next_sample - current_sample) * (fifo_fractional >> 8)); sound_buffer[buffer_index] += dest_sample; fifo_fractional += frequency_step; buffer_index = (buffer_index + 2) % BUFFER_SIZE; } break; case DIRECT_SOUND_LEFTRIGHT: /* render samples LEFT and RIGHT. */ while(fifo_fractional <= 0xFFFFFF) { s16 dest_sample = current_sample + fp16_16_to_u32((next_sample - current_sample) * (fifo_fractional >> 8)); sound_buffer[buffer_index] += dest_sample; sound_buffer[buffer_index + 1] += dest_sample; fifo_fractional += frequency_step; buffer_index = (buffer_index + 2) % BUFFER_SIZE; } break; } ds->buffer_index = buffer_index; ds->fifo_fractional = fp8_24_fractional_part(fifo_fractional); if(((ds->fifo_top - ds->fifo_base) % 32) <= 16) { if(dma[1].direct_sound_channel == channel) dma_transfer(1, &ret); if(dma[2].direct_sound_channel == channel) dma_transfer(2, &ret); } return ret; } void sound_reset_fifo(u32 channel) { memset(direct_sound_channel[channel].fifo, 0, 32); } // Initial pattern data = 4bits (signed) // Channel volume = 12bits // Envelope volume = 14bits // Master volume = 2bits // Recalculate left and right volume as volume changes. // To calculate the current sample, use (sample * volume) >> 16 // Square waves range from -8 (low) to 7 (high) const s8 square_pattern_duty[4][8] = { { -8, -8, -8, -8, 7, -8, -8, -8 }, { -8, -8, -8, -8, 7, 7, -8, -8 }, { -8, -8, 7, 7, 7, 7, -8, -8 }, { 7, 7, 7, 7, -8, -8, 7, 7 }, }; s8 wave_samples[64]; u32 noise_table15[1024]; u32 noise_table7[4]; const u32 gbc_sound_master_volume_table[4] = { 1, 2, 4, 0 }; const u32 gbc_sound_channel_volume_table[8] = { fixed_div(0, 7, 12), fixed_div(1, 7, 12), fixed_div(2, 7, 12), fixed_div(3, 7, 12), fixed_div(4, 7, 12), fixed_div(5, 7, 12), fixed_div(6, 7, 12), fixed_div(7, 7, 12) }; const u32 gbc_sound_envelope_volume_table[16] = { fixed_div(0, 15, 14), fixed_div(1, 15, 14), fixed_div(2, 15, 14), fixed_div(3, 15, 14), fixed_div(4, 15, 14), fixed_div(5, 15, 14), fixed_div(6, 15, 14), fixed_div(7, 15, 14), fixed_div(8, 15, 14), fixed_div(9, 15, 14), fixed_div(10, 15, 14), fixed_div(11, 15, 14), fixed_div(12, 15, 14), fixed_div(13, 15, 14), fixed_div(14, 15, 14), fixed_div(15, 15, 14) }; u32 gbc_sound_buffer_index = 0; u32 gbc_sound_last_cpu_ticks = 0; u32 gbc_sound_partial_ticks = 0; u32 gbc_sound_master_volume_left; u32 gbc_sound_master_volume_right; u32 gbc_sound_master_volume; #define update_volume_channel_envelope(channel) \ volume_##channel = gbc_sound_envelope_volume_table[envelope_volume] * \ gbc_sound_channel_volume_table[gbc_sound_master_volume_##channel] * \ gbc_sound_master_volume_table[gbc_sound_master_volume] \ #define update_volume_channel_noenvelope(channel) \ volume_##channel = gs->wave_volume * \ gbc_sound_channel_volume_table[gbc_sound_master_volume_##channel] * \ gbc_sound_master_volume_table[gbc_sound_master_volume] \ #define update_volume(type) \ update_volume_channel_##type(left); \ update_volume_channel_##type(right) \ #define update_tone_sweep() \ if(gs->sweep_status) \ { \ u32 sweep_ticks = gs->sweep_ticks - 1; \ \ if(sweep_ticks == 0) \ { \ u32 rate = gs->rate; \ \ if(gs->sweep_direction) \ rate = rate - (rate >> gs->sweep_shift); \ else \ rate = rate + (rate >> gs->sweep_shift); \ \ if(rate > 2047) { \ rate = 2047; \ gs->active_flag = 0; \ break; \ } \ \ frequency_step = float_to_fp16_16(((131072.0f / (2048 - rate)) * 8.0f) \ / sound_frequency); \ \ gs->frequency_step = frequency_step; \ gs->rate = rate; \ \ sweep_ticks = gs->sweep_initial_ticks; \ } \ gs->sweep_ticks = sweep_ticks; \ } \ #define update_tone_nosweep() \ #define update_tone_envelope() \ if(gs->envelope_status) \ { \ u32 envelope_ticks = gs->envelope_ticks - 1; \ envelope_volume = gs->envelope_volume; \ \ if(envelope_ticks == 0) \ { \ if(gs->envelope_direction) \ { \ if(envelope_volume != 15) \ envelope_volume = gs->envelope_volume + 1; \ } \ else \ { \ if(envelope_volume != 0) \ envelope_volume = gs->envelope_volume - 1; \ } \ \ update_volume(envelope); \ \ gs->envelope_volume = envelope_volume; \ gs->envelope_ticks = gs->envelope_initial_ticks; \ } \ else \ gs->envelope_ticks = envelope_ticks; \ } \ #define update_tone_noenvelope() \ #define update_tone_counters(envelope_op, sweep_op) \ tick_counter += gbc_sound_tick_step; \ if(tick_counter > 0xFFFF) \ { \ if(gs->length_status) \ { \ u32 length_ticks = gs->length_ticks - 1; \ gs->length_ticks = length_ticks; \ \ if(length_ticks == 0) \ { \ gs->active_flag = 0; \ break; \ } \ } \ \ update_tone_##envelope_op(); \ update_tone_##sweep_op(); \ \ tick_counter &= 0xFFFF; \ } \ #define gbc_sound_render_sample_right() \ sound_buffer[buffer_index + 1] += (current_sample * volume_right) >> 22 \ #define gbc_sound_render_sample_left() \ sound_buffer[buffer_index] += (current_sample * volume_left) >> 22 \ #define gbc_sound_render_sample_both() \ gbc_sound_render_sample_right(); \ gbc_sound_render_sample_left() \ #define gbc_sound_render_samples(type, sample_length, envelope_op, sweep_op) \ for(i = 0; i < buffer_ticks; i++) \ { \ current_sample = \ sample_data[fp16_16_to_u32(sample_index) % sample_length]; \ gbc_sound_render_sample_##type(); \ \ sample_index += frequency_step; \ buffer_index = (buffer_index + 2) % BUFFER_SIZE; \ \ update_tone_counters(envelope_op, sweep_op); \ } \ #define gbc_noise_wrap_full 32767 #define gbc_noise_wrap_half 126 #define get_noise_sample_full() \ current_sample = \ ((s32)(noise_table15[fp16_16_to_u32(sample_index) >> 5] << \ (fp16_16_to_u32(sample_index) & 0x1F)) >> 31) ^ 0x07 \ #define get_noise_sample_half() \ current_sample = \ ((s32)(noise_table7[fp16_16_to_u32(sample_index) >> 5] << \ (fp16_16_to_u32(sample_index) & 0x1F)) >> 31) ^ 0x07 \ #define gbc_sound_render_noise(type, noise_type, envelope_op, sweep_op) \ for(i = 0; i < buffer_ticks; i++) \ { \ get_noise_sample_##noise_type(); \ gbc_sound_render_sample_##type(); \ \ sample_index += frequency_step; \ \ if(sample_index >= u32_to_fp16_16(gbc_noise_wrap_##noise_type)) \ sample_index -= u32_to_fp16_16(gbc_noise_wrap_##noise_type); \ \ buffer_index = (buffer_index + 2) % BUFFER_SIZE; \ update_tone_counters(envelope_op, sweep_op); \ } \ #define gbc_sound_render_channel(type, sample_length, envelope_op, sweep_op) \ buffer_index = gbc_sound_buffer_index; \ sample_index = gs->sample_index; \ frequency_step = gs->frequency_step; \ tick_counter = gs->tick_counter; \ \ update_volume(envelope_op); \ \ switch(gs->status) \ { \ case GBC_SOUND_INACTIVE: \ break; \ \ case GBC_SOUND_LEFT: \ gbc_sound_render_##type(left, sample_length, envelope_op, sweep_op); \ break; \ \ case GBC_SOUND_RIGHT: \ gbc_sound_render_##type(right, sample_length, envelope_op, sweep_op); \ break; \ \ case GBC_SOUND_LEFTRIGHT: \ gbc_sound_render_##type(both, sample_length, envelope_op, sweep_op); \ break; \ } \ \ gs->sample_index = sample_index; \ gs->tick_counter = tick_counter; \ void render_gbc_sound() { u32 i, i2; gbc_sound_struct *gs = gbc_sound_channel; fixed16_16 sample_index, frequency_step; fixed16_16 tick_counter; u32 buffer_index; s32 volume_left, volume_right; u32 envelope_volume; s32 current_sample; u16 sound_status = read_ioreg(REG_SOUNDCNT_X) & 0xFFF0; const s8 *sample_data; u32 tick_delta = cpu_ticks - gbc_sound_last_cpu_ticks; fixed16_16 buffer_ticks = float_to_fp16_16((float)(tick_delta) * sound_frequency / GBC_BASE_RATE); if (!tick_delta) return; gbc_update_count++; gbc_sound_partial_ticks += fp16_16_fractional_part(buffer_ticks); buffer_ticks = fp16_16_to_u32(buffer_ticks); if(gbc_sound_partial_ticks > 0xFFFF) { buffer_ticks += 1; gbc_sound_partial_ticks &= 0xFFFF; } if(sound_on == 1) { s8 *wave_bank; gs = gbc_sound_channel + 0; if(gs->active_flag) { sound_status |= 0x01; sample_data = &square_pattern_duty[gs->sample_table_idx][0]; envelope_volume = gs->envelope_volume; gbc_sound_render_channel(samples, 8, envelope, sweep); } gs = gbc_sound_channel + 1; if(gs->active_flag) { sound_status |= 0x02; sample_data = &square_pattern_duty[gs->sample_table_idx][0]; envelope_volume = gs->envelope_volume; gbc_sound_render_channel(samples, 8, envelope, nosweep); } gs = gbc_sound_channel + 2; if(gbc_sound_wave_update) { unsigned bank = (gs->wave_bank == 1) ? 1 : 0; u8 *wave_ram = ((u8 *)io_registers) + 0x90; wave_bank = wave_samples + (bank * 32); for(i = 0, i2 = 0; i < 16; i++, i2 += 2) { current_sample = wave_ram[i]; wave_bank[i2] = (((current_sample >> 4) & 0x0F) - 8); wave_bank[i2 + 1] = ((current_sample & 0x0F) - 8); } gbc_sound_wave_update = 0; } if((gs->active_flag) && (gs->master_enable)) { 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; 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->volume_halve = 1; 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); memset(&gbc_sound_channel[0], 0, sizeof(gbc_sound_channel)); for(i = 0; i < 4; i++, gs++) { gs->status = GBC_SOUND_INACTIVE; gs->sample_table_idx = 2; gs->active_flag = 0; } } void init_sound() { 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); reset_sound(); } bool sound_check_savestate(const u8 *src) { static const char *gvars[] = { "on", "buf-base", "gbc-buf-idx", "gbc-last-cpu-ticks", "gbc-partial-ticks", "gbc-ms-vol-left", "gbc-ms-vol-right", "gbc-ms-vol" }; static const char *dsvars[] = { "status", "volume", "fifo-base", "fifo-top", "fifo-frac", "buf-idx" }; static const char *gsvars[] = { "status", "rate", "freq-step", "sample-idx", "tick-cnt", "volume", "active", "enable", "env-vol0", "env-vol", "env-dir", "env-status", "env-ticks0", "env-ticks", "sweep-status", "sweep-dir", "sweep-ticks0", "sweep-ticks","sweep-shift", "wav-type", "wav-bank", "wav-vol", "len-status", "len-ticks", "noise-type", "sample-tbl" }; int i; const u8 *snddoc = bson_find_key(src, "sound"); if (!snddoc) return false; for (i = 0; i < sizeof(gvars)/sizeof(gvars[0]); i++) if (!bson_contains_key(snddoc, gvars[i], BSON_TYPE_INT32)) return false; if (!bson_contains_key(snddoc, "wav-samples", BSON_TYPE_BIN)) return false; for (i = 0; i < 2; i++) { char tn[4] = {'d', 's', '0' + i, 0}; const u8 *sndchan = bson_find_key(snddoc, tn); if (!sndchan) return false; for (i = 0; i < sizeof(dsvars)/sizeof(dsvars[0]); i++) if (!bson_contains_key(sndchan, dsvars[i], BSON_TYPE_INT32)) return false; if (!bson_contains_key(sndchan, "fifo-bytes", BSON_TYPE_BIN)) return false; } for (i = 0; i < 4; i++) { char tn[4] = {'g', 's', '0' + i, 0}; const u8 *sndchan = bson_find_key(snddoc, tn); if (!sndchan) return false; for (i = 0; i < sizeof(gsvars)/sizeof(gsvars[0]); i++) if (!bson_contains_key(sndchan, gsvars[i], BSON_TYPE_INT32)) return false; } return true; } bool sound_read_savestate(const u8 *src) { int i; const u8 *snddoc = bson_find_key(src, "sound"); if (!( bson_read_int32(snddoc, "on", &sound_on) && bson_read_int32(snddoc, "buf-base", &sound_buffer_base) && bson_read_int32(snddoc, "gbc-buf-idx", &gbc_sound_buffer_index) && bson_read_int32(snddoc, "gbc-last-cpu-ticks", &gbc_sound_last_cpu_ticks) && bson_read_int32(snddoc, "gbc-partial-ticks", &gbc_sound_partial_ticks) && bson_read_int32(snddoc, "gbc-ms-vol-left", &gbc_sound_master_volume_left) && bson_read_int32(snddoc, "gbc-ms-vol-right", &gbc_sound_master_volume_right) && bson_read_int32(snddoc, "gbc-ms-vol", &gbc_sound_master_volume) && bson_read_bytes(snddoc, "wav-samples", wave_samples, sizeof(wave_samples)))) return false; for (i = 0; i < 2; i++) { direct_sound_struct *ds = &direct_sound_channel[i]; char tn[4] = {'d', 's', '0' + i, 0}; const u8 *sndchan = bson_find_key(snddoc, tn); if (!( bson_read_int32(sndchan, "status", &ds->status) && bson_read_int32(sndchan, "volume", &ds->volume_halve) && bson_read_int32(sndchan, "fifo-base", &ds->fifo_base) && bson_read_int32(sndchan, "fifo-top", &ds->fifo_top) && bson_read_int32(sndchan, "fifo-frac", &ds->fifo_fractional) && bson_read_bytes(sndchan, "fifo-bytes", ds->fifo, sizeof(ds->fifo)) && bson_read_int32(sndchan, "buf-idx", &ds->buffer_index))) return false; } for (i = 0; i < 4; i++) { gbc_sound_struct *gs = &gbc_sound_channel[i]; char tn[4] = {'g', 's', '0' + i, 0}; const u8 *sndchan = bson_find_key(snddoc, tn); if (!( bson_read_int32(sndchan, "status", &gs->status) && bson_read_int32(sndchan, "rate", &gs->rate) && bson_read_int32(sndchan, "freq-step", &gs->frequency_step) && bson_read_int32(sndchan, "sample-idx", &gs->sample_index) && bson_read_int32(sndchan, "tick-cnt", &gs->tick_counter) && bson_read_int32(sndchan, "volume", &gs->total_volume) && bson_read_int32(sndchan, "active", &gs->active_flag) && bson_read_int32(sndchan, "enable", &gs->master_enable) && bson_read_int32(sndchan, "env-vol0", &gs->envelope_initial_volume) && bson_read_int32(sndchan, "env-vol", &gs->envelope_volume) && bson_read_int32(sndchan, "env-dir", &gs->envelope_direction) && bson_read_int32(sndchan, "env-status", &gs->envelope_status) && bson_read_int32(sndchan, "env-ticks0", &gs->envelope_initial_ticks) && bson_read_int32(sndchan, "env-ticks", &gs->envelope_ticks) && bson_read_int32(sndchan, "sweep-status", &gs->sweep_status) && bson_read_int32(sndchan, "sweep-dir", &gs->sweep_direction) && bson_read_int32(sndchan, "sweep-ticks0", &gs->sweep_initial_ticks) && bson_read_int32(sndchan, "sweep-ticks", &gs->sweep_ticks) && bson_read_int32(sndchan, "sweep-shift", &gs->sweep_shift) && bson_read_int32(sndchan, "wav-type", &gs->wave_type) && bson_read_int32(sndchan, "wav-bank", &gs->wave_bank) && bson_read_int32(sndchan, "wav-vol", &gs->wave_volume) && bson_read_int32(sndchan, "len-status", &gs->length_status) && bson_read_int32(sndchan, "len-ticks", &gs->length_ticks) && bson_read_int32(sndchan, "noise-type", &gs->noise_type) && bson_read_int32(sndchan, "sample-tbl", &gs->sample_table_idx))) return false; } return true; } unsigned sound_write_savestate(u8 *dst) { int i; u8 *wbptr, *startp = dst; bson_start_document(dst, "sound", wbptr); bson_write_int32(dst, "on", sound_on); bson_write_int32(dst, "buf-base", sound_buffer_base); bson_write_int32(dst, "gbc-buf-idx", gbc_sound_buffer_index); bson_write_int32(dst, "gbc-last-cpu-ticks", gbc_sound_last_cpu_ticks); bson_write_int32(dst, "gbc-partial-ticks", gbc_sound_partial_ticks); bson_write_int32(dst, "gbc-ms-vol-left", gbc_sound_master_volume_left); bson_write_int32(dst, "gbc-ms-vol-right", gbc_sound_master_volume_right); bson_write_int32(dst, "gbc-ms-vol", gbc_sound_master_volume); bson_write_bytes(dst, "wav-samples", wave_samples, sizeof(wave_samples)); for (i = 0; i < 2; i++) { u8 *wbptr2; char tn[4] = {'d', 's', '0' + i, 0}; bson_start_document(dst, tn, wbptr2); bson_write_int32(dst, "status", direct_sound_channel[i].status); bson_write_int32(dst, "volume", direct_sound_channel[i].volume_halve); bson_write_int32(dst, "fifo-base", direct_sound_channel[i].fifo_base); bson_write_int32(dst, "fifo-top", direct_sound_channel[i].fifo_top); bson_write_int32(dst, "fifo-frac", direct_sound_channel[i].fifo_fractional); bson_write_bytes(dst, "fifo-bytes", direct_sound_channel[i].fifo, sizeof(direct_sound_channel[i].fifo)); bson_write_int32(dst, "buf-idx", direct_sound_channel[i].buffer_index); bson_finish_document(dst, wbptr2); } for (i = 0; i < 4; i++) { gbc_sound_struct *gs = &gbc_sound_channel[i]; u8 *wbptr2; char tn[4] = {'g', 's', '0' + i, 0}; bson_start_document(dst, tn, wbptr2); bson_write_int32(dst, "status", gs->status); bson_write_int32(dst, "rate", gs->rate); bson_write_int32(dst, "freq-step", gs->frequency_step); bson_write_int32(dst, "sample-idx", gs->sample_index); bson_write_int32(dst, "tick-cnt", gs->tick_counter); bson_write_int32(dst, "volume", gs->total_volume); bson_write_int32(dst, "active", gs->active_flag); bson_write_int32(dst, "enable", gs->master_enable); bson_write_int32(dst, "env-vol0", gs->envelope_initial_volume); bson_write_int32(dst, "env-vol", gs->envelope_volume); bson_write_int32(dst, "env-dir", gs->envelope_direction); bson_write_int32(dst, "env-status", gs->envelope_status); bson_write_int32(dst, "env-ticks0", gs->envelope_initial_ticks); bson_write_int32(dst, "env-ticks", gs->envelope_ticks); bson_write_int32(dst, "sweep-status", gs->sweep_status); bson_write_int32(dst, "sweep-dir", gs->sweep_direction); bson_write_int32(dst, "sweep-ticks0", gs->sweep_initial_ticks); bson_write_int32(dst, "sweep-ticks", gs->sweep_ticks); bson_write_int32(dst, "sweep-shift", gs->sweep_shift); bson_write_int32(dst, "wav-type", gs->wave_type); bson_write_int32(dst, "wav-bank", gs->wave_bank); bson_write_int32(dst, "wav-vol", gs->wave_volume); bson_write_int32(dst, "len-status", gs->length_status); bson_write_int32(dst, "len-ticks", gs->length_ticks); bson_write_int32(dst, "noise-type", gs->noise_type); bson_write_int32(dst, "sample-tbl", gs->sample_table_idx); // No longer used fields, keep for backwards compatibility. bson_write_int32(dst, "env-step", 0); bson_finish_document(dst, wbptr2); } bson_finish_document(dst, wbptr); return (unsigned int)(dst - startp); } u32 sound_read_samples(s16 *out, u32 frames) { u32 i; u32 samples_to_read = frames << 1; /* Get total number of samples in the buffer */ u32 samples_available = (gbc_sound_buffer_index - sound_buffer_base) & BUFFER_SIZE_MASK; /* The last 512 samples are 'in use', and cannot * be read out yet */ samples_available = (samples_available > 512) ? (samples_available - 512) : 0; /* Available sample count must be an even number */ samples_available = (samples_available >> 1) << 1; if (samples_to_read > samples_available) samples_to_read = samples_available; for(i = 0; i < samples_to_read; i++) { u32 source_index = (sound_buffer_base + i) & BUFFER_SIZE_MASK; s32 current_sample = sound_buffer[source_index]; sound_buffer[source_index] = 0; if(current_sample > 2047) current_sample = 2047; if(current_sample < -2048) current_sample = -2048; out[i] = current_sample * 16; } sound_buffer_base += samples_to_read; sound_buffer_base &= BUFFER_SIZE_MASK; /* Function returns number of frames read */ return (samples_to_read >> 1); }