const std = @import("std"); const SDL = @import("sdl2"); const io = @import("bus/io.zig"); const util = @import("../util.zig"); const Arm7tdmi = @import("cpu.zig").Arm7tdmi; const Scheduler = @import("scheduler.zig").Scheduler; const ToneSweep = @import("apu/ToneSweep.zig"); const Tone = @import("apu/Tone.zig"); const Wave = @import("apu/Wave.zig"); const Noise = @import("apu/Noise.zig"); const SoundFifo = std.fifo.LinearFifo(u8, .{ .Static = 0x20 }); const getHalf = util.getHalf; const setHalf = util.setHalf; const log = std.log.scoped(.APU); pub const host_rate = @import("../platform.zig").sample_rate; pub const host_format = @import("../platform.zig").sample_format; pub fn read(comptime T: type, apu: *const Apu, addr: u32) ?T { const byte_addr = @truncate(u8, addr); return switch (T) { u32 => switch (byte_addr) { 0x60 => @as(T, apu.ch1.sound1CntH()) << 16 | apu.ch1.sound1CntL(), 0x64 => apu.ch1.sound1CntX(), 0x68 => apu.ch2.sound2CntL(), 0x6C => apu.ch2.sound2CntH(), 0x70 => @as(T, apu.ch3.sound3CntH()) << 16 | apu.ch3.sound3CntL(), 0x74 => apu.ch3.sound3CntX(), 0x78 => apu.ch4.sound4CntL(), 0x7C => apu.ch4.sound4CntH(), 0x80 => @as(T, apu.dma_cnt.raw) << 16 | apu.psg_cnt.raw, // SOUNDCNT_H, SOUNDCNT_L 0x84 => apu.soundCntX(), 0x88 => apu.bias.raw, // SOUNDBIAS, high is unused 0x8C => null, 0x90, 0x94, 0x98, 0x9C => apu.ch3.wave_dev.read(T, apu.ch3.select, addr), 0xA0 => null, // FIFO_A 0xA4 => null, // FIFO_B else => util.io.read.err(T, log, "unaligned {} read from 0x{X:0>8}", .{ T, addr }), }, u16 => switch (byte_addr) { 0x60 => apu.ch1.sound1CntL(), 0x62 => apu.ch1.sound1CntH(), 0x64 => apu.ch1.sound1CntX(), 0x66 => 0x0000, // suite.gba expects 0x0000, not 0xDEAD 0x68 => apu.ch2.sound2CntL(), 0x6A => 0x0000, 0x6C => apu.ch2.sound2CntH(), 0x6E => 0x0000, 0x70 => apu.ch3.sound3CntL(), 0x72 => apu.ch3.sound3CntH(), 0x74 => apu.ch3.sound3CntX(), 0x76 => 0x0000, 0x78 => apu.ch4.sound4CntL(), 0x7A => 0x0000, 0x7C => apu.ch4.sound4CntH(), 0x7E => 0x0000, 0x80 => apu.soundCntL(), 0x82 => apu.soundCntH(), 0x84 => apu.soundCntX(), 0x86 => 0x0000, 0x88 => apu.bias.raw, // SOUNDBIAS 0x8A => 0x0000, 0x8C, 0x8E => null, 0x90, 0x92, 0x94, 0x96, 0x98, 0x9A, 0x9C, 0x9E => apu.ch3.wave_dev.read(T, apu.ch3.select, addr), 0xA0, 0xA2 => null, // FIFO_A 0xA4, 0xA6 => null, // FIFO_B else => util.io.read.err(T, log, "unaligned {} read from 0x{X:0>8}", .{ T, addr }), }, u8 => switch (byte_addr) { 0x60, 0x61 => @truncate(T, @as(u16, apu.ch1.sound1CntL()) >> getHalf(byte_addr)), 0x62, 0x63 => @truncate(T, apu.ch1.sound1CntH() >> getHalf(byte_addr)), 0x64, 0x65 => @truncate(T, apu.ch1.sound1CntX() >> getHalf(byte_addr)), 0x66, 0x67 => 0x00, // assuming behaviour is identical to that of 16-bit reads 0x68, 0x69 => @truncate(T, apu.ch2.sound2CntL() >> getHalf(byte_addr)), 0x6A, 0x6B => 0x00, 0x6C, 0x6D => @truncate(T, apu.ch2.sound2CntH() >> getHalf(byte_addr)), 0x6E, 0x6F => 0x00, 0x70, 0x71 => @truncate(T, @as(u16, apu.ch3.sound3CntL()) >> getHalf(byte_addr)), // SOUND3CNT_L 0x72, 0x73 => @truncate(T, apu.ch3.sound3CntH() >> getHalf(byte_addr)), 0x74, 0x75 => @truncate(T, apu.ch3.sound3CntX() >> getHalf(byte_addr)), // SOUND3CNT_L 0x76, 0x77 => 0x00, 0x78, 0x79 => @truncate(T, apu.ch4.sound4CntL() >> getHalf(byte_addr)), 0x7A, 0x7B => 0x00, 0x7C, 0x7D => @truncate(T, apu.ch4.sound4CntH() >> getHalf(byte_addr)), 0x7E, 0x7F => 0x00, 0x80, 0x81 => @truncate(T, apu.soundCntL() >> getHalf(byte_addr)), // SOUNDCNT_L 0x82, 0x83 => @truncate(T, apu.soundCntH() >> getHalf(byte_addr)), // SOUNDCNT_H 0x84, 0x85 => @truncate(T, @as(u16, apu.soundCntX()) >> getHalf(byte_addr)), 0x86, 0x87 => 0x00, 0x88, 0x89 => @truncate(T, apu.bias.raw >> getHalf(byte_addr)), // SOUNDBIAS 0x8A, 0x8B => 0x00, 0x8C...0x8F => null, 0x90...0x9F => apu.ch3.wave_dev.read(T, apu.ch3.select, addr), 0xA0, 0xA1, 0xA2, 0xA3 => null, // FIFO_A 0xA4, 0xA5, 0xA6, 0xA7 => null, // FIFO_B else => util.io.read.err(T, log, "unexpected {} read from 0x{X:0>8}", .{ T, addr }), }, else => @compileError("APU: Unsupported read width"), }; } pub fn write(comptime T: type, apu: *Apu, addr: u32, value: T) void { const byte_addr = @truncate(u8, addr); if (byte_addr <= 0x81 and !apu.cnt.apu_enable.read()) return; switch (T) { u32 => { // 0x80 and 0x81 handled in setSoundCnt if (byte_addr < 0x80 and !apu.cnt.apu_enable.read()) return; switch (byte_addr) { 0x60 => apu.ch1.setSound1Cnt(value), 0x64 => apu.ch1.setSound1CntX(&apu.fs, @truncate(u16, value)), 0x68 => apu.ch2.setSound2CntL(@truncate(u16, value)), 0x6C => apu.ch2.setSound2CntH(&apu.fs, @truncate(u16, value)), 0x70 => apu.ch3.setSound3Cnt(value), 0x74 => apu.ch3.setSound3CntX(&apu.fs, @truncate(u16, value)), 0x78 => apu.ch4.setSound4CntL(@truncate(u16, value)), 0x7C => apu.ch4.setSound4CntH(&apu.fs, @truncate(u16, value)), 0x80 => apu.setSoundCnt(value), 0x84 => apu.setSoundCntX(value >> 7 & 1 == 1), 0x88 => apu.bias.raw = @truncate(u16, value), 0x8C => {}, 0x90, 0x94, 0x98, 0x9C => apu.ch3.wave_dev.write(T, apu.ch3.select, addr, value), 0xA0 => apu.chA.push(value), // FIFO_A 0xA4 => apu.chB.push(value), // FIFO_B else => util.io.write.undef(log, "Tried to write 0x{X:0>8}{} to 0x{X:0>8}", .{ value, T, addr }), } }, u16 => { if (byte_addr <= 0x81 and !apu.cnt.apu_enable.read()) return; switch (byte_addr) { 0x60 => apu.ch1.setSound1CntL(@truncate(u8, value)), // SOUND1CNT_L 0x62 => apu.ch1.setSound1CntH(value), 0x64 => apu.ch1.setSound1CntX(&apu.fs, value), 0x66 => {}, 0x68 => apu.ch2.setSound2CntL(value), 0x6A => {}, 0x6C => apu.ch2.setSound2CntH(&apu.fs, value), 0x6E => {}, 0x70 => apu.ch3.setSound3CntL(@truncate(u8, value)), 0x72 => apu.ch3.setSound3CntH(value), 0x74 => apu.ch3.setSound3CntX(&apu.fs, value), 0x76 => {}, 0x78 => apu.ch4.setSound4CntL(value), 0x7A => {}, 0x7C => apu.ch4.setSound4CntH(&apu.fs, value), 0x7E => {}, 0x80 => apu.setSoundCntL(value), 0x82 => apu.setSoundCntH(value), 0x84 => apu.setSoundCntX(value >> 7 & 1 == 1), 0x86 => {}, 0x88 => apu.bias.raw = value, // SOUNDBIAS 0x8A, 0x8C, 0x8E => {}, 0x90, 0x92, 0x94, 0x96, 0x98, 0x9A, 0x9C, 0x9E => apu.ch3.wave_dev.write(T, apu.ch3.select, addr, value), 0xA0, 0xA2 => log.err("Tried to write 0x{X:0>4}{} to FIFO_A", .{ value, T }), 0xA4, 0xA6 => log.err("Tried to write 0x{X:0>4}{} to FIFO_B", .{ value, T }), else => util.io.write.undef(log, "Tried to write 0x{X:0>4}{} to 0x{X:0>8}", .{ value, T, addr }), } }, u8 => { if (byte_addr <= 0x81 and !apu.cnt.apu_enable.read()) return; switch (byte_addr) { 0x60 => apu.ch1.setSound1CntL(value), 0x61 => {}, 0x62 => apu.ch1.setNr11(value), 0x63 => apu.ch1.setNr12(value), 0x64 => apu.ch1.setNr13(value), 0x65 => apu.ch1.setNr14(&apu.fs, value), 0x66, 0x67 => {}, 0x68 => apu.ch2.setNr21(value), 0x69 => apu.ch2.setNr22(value), 0x6A, 0x6B => {}, 0x6C => apu.ch2.setNr23(value), 0x6D => apu.ch2.setNr24(&apu.fs, value), 0x6E, 0x6F => {}, 0x70 => apu.ch3.setSound3CntL(value), // NR30 0x71 => {}, 0x72 => apu.ch3.setNr31(value), 0x73 => apu.ch3.vol.raw = value, // NR32 0x74 => apu.ch3.setNr33(value), 0x75 => apu.ch3.setNr34(&apu.fs, value), 0x76, 0x77 => {}, 0x78 => apu.ch4.setNr41(value), 0x79 => apu.ch4.setNr42(value), 0x7A, 0x7B => {}, 0x7C => apu.ch4.poly.raw = value, // NR 43 0x7D => apu.ch4.setNr44(&apu.fs, value), 0x7E, 0x7F => {}, 0x80, 0x81 => apu.setSoundCntL(setHalf(u16, apu.psg_cnt.raw, byte_addr, value)), 0x82, 0x83 => apu.setSoundCntH(setHalf(u16, apu.dma_cnt.raw, byte_addr, value)), 0x84 => apu.setSoundCntX(value >> 7 & 1 == 1), 0x85 => {}, 0x86, 0x87 => {}, 0x88, 0x89 => apu.bias.raw = setHalf(u16, apu.bias.raw, byte_addr, value), // SOUNDBIAS 0x8A...0x8F => {}, 0x90...0x9F => apu.ch3.wave_dev.write(T, apu.ch3.select, addr, value), 0xA0...0xA3 => log.err("Tried to write 0x{X:0>2}{} to FIFO_A", .{ value, T }), 0xA4...0xA7 => log.err("Tried to write 0x{X:0>2}{} to FIFO_B", .{ value, T }), else => util.io.write.undef(log, "Tried to write 0x{X:0>2}{} to 0x{X:0>8}", .{ value, T, addr }), } }, else => @compileError("APU: Unsupported write width"), } } pub const Apu = struct { const Self = @This(); ch1: ToneSweep, ch2: Tone, ch3: Wave, ch4: Noise, chA: DmaSound(.A), chB: DmaSound(.B), bias: io.SoundBias, /// NR50, NR51 psg_cnt: io.ChannelVolumeControl, dma_cnt: io.DmaSoundControl, cnt: io.SoundControl, sampling_cycle: u2, stream: *SDL.SDL_AudioStream, sched: *Scheduler, fs: FrameSequencer, capacitor: f32, is_buffer_full: bool, pub const Tick = enum { Length, Envelope, Sweep }; pub fn init(sched: *Scheduler) Self { const apu: Self = .{ .ch1 = ToneSweep.init(sched), .ch2 = Tone.init(sched), .ch3 = Wave.init(sched), .ch4 = Noise.init(sched), .chA = DmaSound(.A).init(), .chB = DmaSound(.B).init(), .psg_cnt = .{ .raw = 0 }, .dma_cnt = .{ .raw = 0 }, .cnt = .{ .raw = 0 }, .bias = .{ .raw = 0x0200 }, .sampling_cycle = 0b00, .stream = SDL.SDL_NewAudioStream(SDL.AUDIO_U16, 2, 1 << 15, host_format, 2, host_rate).?, .sched = sched, .capacitor = 0, .fs = FrameSequencer.init(), .is_buffer_full = false, }; sched.push(.SampleAudio, apu.interval()); sched.push(.{ .ApuChannel = 0 }, @import("apu/signal/Square.zig").interval); sched.push(.{ .ApuChannel = 1 }, @import("apu/signal/Square.zig").interval); sched.push(.{ .ApuChannel = 2 }, @import("apu/signal/Wave.zig").interval); sched.push(.{ .ApuChannel = 3 }, @import("apu/signal/Lfsr.zig").interval); sched.push(.FrameSequencer, FrameSequencer.interval); return apu; } /// Used when resetting the emulator pub fn reset(self: *Self) void { // FIXME: These reset functions are meant to emulate obscure APU behaviour. Write proper emu reset fns self.ch1.reset(); self.ch2.reset(); self.ch3.reset(); self.ch4.reset(); self.chA.reset(); self.chB.reset(); self.psg_cnt = .{ .raw = 0 }; self.dma_cnt = .{ .raw = 0 }; self.cnt = .{ .raw = 0 }; self.bias = .{ .raw = 0x200 }; self.sampling_cycle = 0; self.fs.reset(); } /// Emulates the reset behaviour of the APU fn _reset(self: *Self) void { // All PSG Registers between 0x0400_0060..0x0400_0081 are zeroed // 0x0400_0082 and 0x0400_0088 retain their values self.ch1.reset(); self.ch2.reset(); self.ch3.reset(); self.ch4.reset(); // GBATEK says 4000060h..4000081h I take this to mean inclusive self.psg_cnt.raw = 0x0000; } /// SOUNDCNT fn setSoundCnt(self: *Self, value: u32) void { if (self.cnt.apu_enable.read()) self.setSoundCntL(@truncate(u16, value)); self.setSoundCntH(@truncate(u16, value >> 16)); } /// SOUNDCNT_L pub fn soundCntL(self: *const Self) u16 { return self.psg_cnt.raw & 0xFF77; } /// SOUNDCNT_L pub fn setSoundCntL(self: *Self, value: u16) void { self.psg_cnt.raw = value; } /// SOUNDCNT_H pub fn setSoundCntH(self: *Self, value: u16) void { const new: io.DmaSoundControl = .{ .raw = value }; // Reinitializing instead of resetting is fine because // the FIFOs I'm using are stack allocated and 0x20 bytes big if (new.chA_reset.read()) self.chA.fifo = SoundFifo.init(); if (new.chB_reset.read()) self.chB.fifo = SoundFifo.init(); self.dma_cnt = new; } /// SOUNDCNT_H pub fn soundCntH(self: *const Self) u16 { return self.dma_cnt.raw & 0x770F; } /// NR52 pub fn setSoundCntX(self: *Self, value: bool) void { self.cnt.apu_enable.write(value); if (value) { self.fs.step = 0; // Reset Frame Sequencer // Reset Square Wave Offsets self.ch1.square.reset(); self.ch2.square.reset(); // Reset Wave self.ch3.wave_dev.reset(); // Rest Noise self.ch4.lfsr.reset(); } else { self._reset(); } } /// NR52 pub fn soundCntX(self: *const Self) u8 { const apu_enable: u8 = @boolToInt(self.cnt.apu_enable.read()); const ch1_enable: u8 = @boolToInt(self.ch1.enabled); const ch2_enable: u8 = @boolToInt(self.ch2.enabled); const ch3_enable: u8 = @boolToInt(self.ch3.enabled); const ch4_enable: u8 = @boolToInt(self.ch4.enabled); return apu_enable << 7 | ch4_enable << 3 | ch3_enable << 2 | ch2_enable << 1 | ch1_enable; } pub fn sampleAudio(self: *Self, late: u64) void { self.sched.push(.SampleAudio, self.interval() -| late); // Whether the APU is busy or not is determined by the main loop in emu.zig // This should only ever be true (because this side of the emu is single threaded) // When audio sync is disaabled if (self.is_buffer_full) return; var left: i16 = 0; var right: i16 = 0; // SOUNDCNT_L Channel Enable flags const ch_left: u4 = self.psg_cnt.ch_left.read(); const ch_right: u4 = self.psg_cnt.ch_right.read(); // Determine SOUNDCNT_H volume modifications const gba_vol: u4 = switch (self.dma_cnt.ch_vol.read()) { 0b00 => 2, 0b01 => 1, else => 0, }; // Add all PSG channels together left += if (ch_left & 1 == 1) @as(i16, self.ch1.sample) else 0; left += if (ch_left >> 1 & 1 == 1) @as(i16, self.ch2.sample) else 0; left += if (ch_left >> 2 & 1 == 1) @as(i16, self.ch3.sample) else 0; left += if (ch_left >> 3 == 1) @as(i16, self.ch4.sample) else 0; right += if (ch_right & 1 == 1) @as(i16, self.ch1.sample) else 0; right += if (ch_right >> 1 & 1 == 1) @as(i16, self.ch2.sample) else 0; right += if (ch_right >> 2 & 1 == 1) @as(i16, self.ch3.sample) else 0; right += if (ch_right >> 3 == 1) @as(i16, self.ch4.sample) else 0; // Multiply by master channel volume left *= 1 + @as(i16, self.psg_cnt.left_vol.read()); right *= 1 + @as(i16, self.psg_cnt.right_vol.read()); // Apply GBA volume modifications to PSG Channels left >>= gba_vol; right >>= gba_vol; const chA_sample = self.chA.amplitude() << if (self.dma_cnt.chA_vol.read()) @as(u4, 2) else 1; const chB_sample = self.chB.amplitude() << if (self.dma_cnt.chB_vol.read()) @as(u4, 2) else 1; left += if (self.dma_cnt.chA_left.read()) chA_sample else 0; left += if (self.dma_cnt.chB_left.read()) chB_sample else 0; right += if (self.dma_cnt.chA_right.read()) chA_sample else 0; right += if (self.dma_cnt.chB_right.read()) chB_sample else 0; // Add SOUNDBIAS // FIXME: SOUNDBIAS is 10-bit but The waveform is centered around 0 if I treat it as 11-bit const bias = @as(i16, self.bias.level.read()) << 2; left += bias; right += bias; const clamped_left = std.math.clamp(@bitCast(u16, left), std.math.minInt(u11), std.math.maxInt(u11)); const clamped_right = std.math.clamp(@bitCast(u16, right), std.math.minInt(u11), std.math.maxInt(u11)); // Extend to 16-bit signed audio samples const ext_left = (clamped_left << 5) | (clamped_left >> 6); const ext_right = (clamped_right << 5) | (clamped_right >> 6); if (self.sampling_cycle != self.bias.sampling_cycle.read()) self.replaceSDLResampler(); _ = SDL.SDL_AudioStreamPut(self.stream, &[2]u16{ ext_left, ext_right }, 2 * @sizeOf(u16)); } fn replaceSDLResampler(self: *Self) void { @setCold(true); const sample_rate = Self.sampleRate(self.bias.sampling_cycle.read()); log.info("Sample Rate changed from {}Hz to {}Hz", .{ Self.sampleRate(self.sampling_cycle), sample_rate }); // Sampling Cycle (Sample Rate) changed, Craete a new SDL Audio Resampler // FIXME: Replace SDL's Audio Resampler with either a custom or more reliable one const old_stream = self.stream; defer SDL.SDL_FreeAudioStream(old_stream); self.sampling_cycle = self.bias.sampling_cycle.read(); self.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_U16, 2, @intCast(c_int, sample_rate), host_format, 2, host_rate).?; } fn interval(self: *const Self) u64 { return (1 << 24) / Self.sampleRate(self.bias.sampling_cycle.read()); } fn sampleRate(cycle: u2) u64 { return @as(u64, 1) << (15 + @as(u6, cycle)); } pub fn onSequencerTick(self: *Self, late: u64) void { self.fs.tick(); switch (self.fs.step) { 7 => self.tick(.Envelope), // Clock Envelope 0, 4 => self.tick(.Length), // Clock Length 2, 6 => { // Clock Length and Sweep self.tick(.Length); self.tick(.Sweep); }, 1, 3, 5 => {}, } self.sched.push(.FrameSequencer, ((1 << 24) / 512) -| late); } fn tick(self: *Self, comptime kind: Tick) void { self.ch1.tick(kind); switch (kind) { .Length => { self.ch2.tick(kind); self.ch3.tick(kind); self.ch4.tick(kind); }, .Envelope => { self.ch2.tick(kind); self.ch4.tick(kind); }, .Sweep => {}, // Already handled above (only for Ch1) } } pub fn onDmaAudioSampleRequest(self: *Self, cpu: *Arm7tdmi, tim_id: u3) void { if (!self.cnt.apu_enable.read()) return; if (@boolToInt(self.dma_cnt.chA_timer.read()) == tim_id) { if (!self.chA.enabled) return; self.chA.updateSample(); if (self.chA.len() <= 15) cpu.bus.dma[1].requestAudio(0x0400_00A0); } if (@boolToInt(self.dma_cnt.chB_timer.read()) == tim_id) { if (!self.chB.enabled) return; self.chB.updateSample(); if (self.chB.len() <= 15) cpu.bus.dma[2].requestAudio(0x0400_00A4); } } }; pub fn DmaSound(comptime kind: DmaSoundKind) type { return struct { const Self = @This(); fifo: SoundFifo, kind: DmaSoundKind, sample: i8, enabled: bool, fn init() Self { return .{ .fifo = SoundFifo.init(), .kind = kind, .sample = 0, .enabled = false, }; } /// Used when resetting hte emulator (not emulation code) fn reset(self: *Self) void { self.* = Self.init(); } pub fn push(self: *Self, value: u32) void { if (!self.enabled) self.enable(); self.fifo.write(std.mem.asBytes(&value)) catch |e| log.err("{} Error: {}", .{ kind, e }); } fn enable(self: *Self) void { @setCold(true); self.enabled = true; } pub fn len(self: *const Self) usize { return self.fifo.readableLength(); } pub fn updateSample(self: *Self) void { if (self.fifo.readItem()) |sample| self.sample = @bitCast(i8, sample); } pub fn amplitude(self: *const Self) i16 { return @as(i16, self.sample); } }; } const DmaSoundKind = enum { A, B, }; pub const FrameSequencer = struct { const Self = @This(); pub const interval = (1 << 24) / 512; step: u3 = 0, pub fn init() Self { return .{}; } pub fn reset(self: *Self) void { self.* = .{}; } pub fn tick(self: *Self) void { self.step +%= 1; } pub fn isLengthNext(self: *const Self) bool { return (self.step +% 1) & 1 == 0; // Steps, 0, 2, 4, and 6 clock length } pub fn isEnvelopeNext(self: *const Self) bool { return (self.step +% 1) == 7; } };