feat: add audio resampler
Also implement extremely naive audio sync
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parent
d270ec711f
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c3611a0f00
41
src/apu.zig
41
src/apu.zig
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@ -11,8 +11,7 @@ const AudioDeviceId = SDL.SDL_AudioDeviceID;
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const intToBytes = @import("util.zig").intToBytes;
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const log = std.log.scoped(.APU);
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const sample_rate = 32768;
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const sample_ticks = (1 << 24) / sample_rate;
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pub const host_sample_rate = 1 << 15;
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pub const Apu = struct {
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const Self = @This();
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@ -29,7 +28,8 @@ pub const Apu = struct {
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dma_cnt: io.DmaSoundControl,
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cnt: io.SoundControl,
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dev: ?AudioDeviceId,
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sampling_cycle: u2,
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stream: *SDL.SDL_AudioStream,
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sched: *Scheduler,
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pub fn init(sched: *Scheduler) Self {
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@ -46,18 +46,16 @@ pub const Apu = struct {
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.cnt = .{ .raw = 0 },
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.bias = .{ .raw = 0x0200 },
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.dev = null,
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.sampling_cycle = 0b00,
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.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_F32, 2, 1 << 15, SDL.AUDIO_F32, 2, host_sample_rate) orelse unreachable,
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.sched = sched,
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};
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sched.push(.SampleAudio, sched.now() + sample_ticks);
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sched.push(.SampleAudio, sched.now() + apu.sampleTicks());
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return apu;
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}
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pub fn attachAudioDevice(self: *Self, dev: AudioDeviceId) void {
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self.dev = dev;
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}
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pub fn setDmaCnt(self: *Self, value: u16) void {
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const new: io.DmaSoundControl = .{ .raw = value };
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@ -97,9 +95,30 @@ pub const Apu = struct {
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const left = (chA_left + chB_left) / 2;
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const right = (chA_right + chB_right) / 2;
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if (self.dev) |dev| _ = SDL.SDL_QueueAudio(dev, &[2]f32{ left, right }, 2 * @sizeOf(f32));
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if (self.sampling_cycle != self.bias.sampling_cycle.read()) {
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log.warn("Sampling Cycle changed from {} to {}", .{ self.sampling_cycle, self.bias.sampling_cycle.read() });
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self.sched.push(.SampleAudio, self.sched.now() + sample_ticks - late);
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// Sample Rate Changed, Create a new Resampler since i can't figure out how to change
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// the parameters of the old one
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const old = self.stream;
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defer SDL.SDL_FreeAudioStream(old);
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self.sampling_cycle = self.bias.sampling_cycle.read();
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self.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_F32, 2, @intCast(c_int, self.sampleRate()), SDL.AUDIO_F32, 2, host_sample_rate) orelse unreachable;
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}
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while (SDL.SDL_AudioStreamAvailable(self.stream) > (@sizeOf(f32) * 2 * 0x800)) {}
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_ = SDL.SDL_AudioStreamPut(self.stream, &[2]f32{ left, right }, 2 * @sizeOf(f32));
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self.sched.push(.SampleAudio, self.sched.now() + self.sampleTicks() - late);
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}
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inline fn sampleTicks(self: *const Self) u64 {
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return (1 << 24) / self.sampleRate();
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}
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inline fn sampleRate(self: *const Self) u64 {
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return @as(u64, 1) << (15 + @as(u6, self.bias.sampling_cycle.read()));
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}
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pub fn handleTimerOverflow(self: *Self, cpu: *Arm7tdmi, tim_id: u3) void {
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42
src/main.zig
42
src/main.zig
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@ -5,6 +5,7 @@ const known_folders = @import("known_folders");
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const emu = @import("emu.zig");
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const Bus = @import("Bus.zig");
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const Apu = @import("apu.zig").Apu;
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const Arm7tdmi = @import("cpu.zig").Arm7tdmi;
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const Scheduler = @import("scheduler.zig").Scheduler;
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const FpsAverage = @import("util.zig").FpsAverage;
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@ -20,6 +21,8 @@ const gba_height = @import("ppu.zig").height;
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const framebuf_pitch = @import("ppu.zig").framebuf_pitch;
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const expected_rate = @import("emu.zig").frame_rate;
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const sample_rate = @import("apu.zig").host_sample_rate;
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pub const enable_logging: bool = false;
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const is_binary: bool = false;
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const log = std.log.scoped(.GUI);
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@ -70,10 +73,6 @@ pub fn main() anyerror!void {
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_ = initSdl2();
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defer SDL.SDL_Quit();
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// Initialize SDL Audio
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const audio_dev = initAudio();
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defer SDL.SDL_CloseAudioDevice(audio_dev);
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// Initialize Emulator
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var scheduler = Scheduler.init(alloc);
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defer scheduler.deinit();
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@ -81,10 +80,12 @@ pub fn main() anyerror!void {
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const paths = .{ .bios = bios_path, .rom = rom_path, .save = save_path };
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var cpu = try Arm7tdmi.init(alloc, &scheduler, paths);
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defer cpu.deinit();
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cpu.bus.apu.attachAudioDevice(audio_dev);
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cpu.fastBoot();
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// Initialize SDL Audio
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const audio_dev = initAudio(&cpu.bus.apu);
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defer SDL.SDL_CloseAudioDevice(audio_dev);
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const log_file: ?File = if (enable_logging) blk: {
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const file = try std.fs.cwd().createFile(if (is_binary) "zba.bin" else "zba.log", .{});
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cpu.useLogger(&file, is_binary);
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@ -155,6 +156,7 @@ pub fn main() anyerror!void {
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SDL.SDLK_s => io.keyinput.shoulder_r.set(),
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SDL.SDLK_RETURN => io.keyinput.start.set(),
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SDL.SDLK_RSHIFT => io.keyinput.select.set(),
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SDL.SDLK_i => std.debug.print("{} samples\n", .{@intCast(u32, SDL.SDL_AudioStreamAvailable(cpu.bus.apu.stream)) / (2 * @sizeOf(f32))}),
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else => {},
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}
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},
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@ -238,19 +240,31 @@ fn createTexture(renderer: *SDL.SDL_Renderer, width: c_int, height: c_int) *SDL.
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) orelse sdlPanic();
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}
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fn initAudio() SDL.SDL_AudioDeviceID {
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fn initAudio(apu: *Apu) SDL.SDL_AudioDeviceID {
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var have: SDL.SDL_AudioSpec = undefined;
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var want = std.mem.zeroes(SDL.SDL_AudioSpec);
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want.freq = 32768;
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want.format = SDL.AUDIO_F32;
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want.channels = 2;
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want.samples = 0x100;
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want.callback = null;
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var want: SDL.SDL_AudioSpec = .{
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.freq = sample_rate,
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.format = SDL.AUDIO_F32,
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.channels = 2,
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.samples = 0x100,
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.callback = audioCallback,
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.userdata = apu,
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.silence = undefined,
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.size = undefined,
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.padding = undefined,
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};
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const dev = SDL.SDL_OpenAudioDevice(null, 0, &want, &have, 0);
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if (dev == 0) sdlPanic();
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// Start Playback on the Audio evice
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// Start Playback on the Audio device
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SDL.SDL_PauseAudioDevice(dev, 0);
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return dev;
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}
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export fn audioCallback(userdata: ?*anyopaque, stream: [*c]u8, len: c_int) void {
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const apu = @ptrCast(*Apu, @alignCast(8, userdata));
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const result = SDL.SDL_AudioStreamGet(apu.stream, stream, len);
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if (result < 0) log.err("Audio Callback Underflow", .{});
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}
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