feat: add audio resampler

Also implement extremely naive audio sync
This commit is contained in:
Rekai Nyangadzayi Musuka 2022-04-20 04:43:25 -03:00
parent d270ec711f
commit c3611a0f00
2 changed files with 58 additions and 25 deletions

View File

@ -11,8 +11,7 @@ const AudioDeviceId = SDL.SDL_AudioDeviceID;
const intToBytes = @import("util.zig").intToBytes;
const log = std.log.scoped(.APU);
const sample_rate = 32768;
const sample_ticks = (1 << 24) / sample_rate;
pub const host_sample_rate = 1 << 15;
pub const Apu = struct {
const Self = @This();
@ -29,7 +28,8 @@ pub const Apu = struct {
dma_cnt: io.DmaSoundControl,
cnt: io.SoundControl,
dev: ?AudioDeviceId,
sampling_cycle: u2,
stream: *SDL.SDL_AudioStream,
sched: *Scheduler,
pub fn init(sched: *Scheduler) Self {
@ -46,18 +46,16 @@ pub const Apu = struct {
.cnt = .{ .raw = 0 },
.bias = .{ .raw = 0x0200 },
.dev = null,
.sampling_cycle = 0b00,
.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_F32, 2, 1 << 15, SDL.AUDIO_F32, 2, host_sample_rate) orelse unreachable,
.sched = sched,
};
sched.push(.SampleAudio, sched.now() + sample_ticks);
sched.push(.SampleAudio, sched.now() + apu.sampleTicks());
return apu;
}
pub fn attachAudioDevice(self: *Self, dev: AudioDeviceId) void {
self.dev = dev;
}
pub fn setDmaCnt(self: *Self, value: u16) void {
const new: io.DmaSoundControl = .{ .raw = value };
@ -97,9 +95,30 @@ pub const Apu = struct {
const left = (chA_left + chB_left) / 2;
const right = (chA_right + chB_right) / 2;
if (self.dev) |dev| _ = SDL.SDL_QueueAudio(dev, &[2]f32{ left, right }, 2 * @sizeOf(f32));
if (self.sampling_cycle != self.bias.sampling_cycle.read()) {
log.warn("Sampling Cycle changed from {} to {}", .{ self.sampling_cycle, self.bias.sampling_cycle.read() });
self.sched.push(.SampleAudio, self.sched.now() + sample_ticks - late);
// Sample Rate Changed, Create a new Resampler since i can't figure out how to change
// the parameters of the old one
const old = self.stream;
defer SDL.SDL_FreeAudioStream(old);
self.sampling_cycle = self.bias.sampling_cycle.read();
self.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_F32, 2, @intCast(c_int, self.sampleRate()), SDL.AUDIO_F32, 2, host_sample_rate) orelse unreachable;
}
while (SDL.SDL_AudioStreamAvailable(self.stream) > (@sizeOf(f32) * 2 * 0x800)) {}
_ = SDL.SDL_AudioStreamPut(self.stream, &[2]f32{ left, right }, 2 * @sizeOf(f32));
self.sched.push(.SampleAudio, self.sched.now() + self.sampleTicks() - late);
}
inline fn sampleTicks(self: *const Self) u64 {
return (1 << 24) / self.sampleRate();
}
inline fn sampleRate(self: *const Self) u64 {
return @as(u64, 1) << (15 + @as(u6, self.bias.sampling_cycle.read()));
}
pub fn handleTimerOverflow(self: *Self, cpu: *Arm7tdmi, tim_id: u3) void {

View File

@ -5,6 +5,7 @@ const known_folders = @import("known_folders");
const emu = @import("emu.zig");
const Bus = @import("Bus.zig");
const Apu = @import("apu.zig").Apu;
const Arm7tdmi = @import("cpu.zig").Arm7tdmi;
const Scheduler = @import("scheduler.zig").Scheduler;
const FpsAverage = @import("util.zig").FpsAverage;
@ -20,6 +21,8 @@ const gba_height = @import("ppu.zig").height;
const framebuf_pitch = @import("ppu.zig").framebuf_pitch;
const expected_rate = @import("emu.zig").frame_rate;
const sample_rate = @import("apu.zig").host_sample_rate;
pub const enable_logging: bool = false;
const is_binary: bool = false;
const log = std.log.scoped(.GUI);
@ -70,10 +73,6 @@ pub fn main() anyerror!void {
_ = initSdl2();
defer SDL.SDL_Quit();
// Initialize SDL Audio
const audio_dev = initAudio();
defer SDL.SDL_CloseAudioDevice(audio_dev);
// Initialize Emulator
var scheduler = Scheduler.init(alloc);
defer scheduler.deinit();
@ -81,10 +80,12 @@ pub fn main() anyerror!void {
const paths = .{ .bios = bios_path, .rom = rom_path, .save = save_path };
var cpu = try Arm7tdmi.init(alloc, &scheduler, paths);
defer cpu.deinit();
cpu.bus.apu.attachAudioDevice(audio_dev);
cpu.fastBoot();
// Initialize SDL Audio
const audio_dev = initAudio(&cpu.bus.apu);
defer SDL.SDL_CloseAudioDevice(audio_dev);
const log_file: ?File = if (enable_logging) blk: {
const file = try std.fs.cwd().createFile(if (is_binary) "zba.bin" else "zba.log", .{});
cpu.useLogger(&file, is_binary);
@ -155,6 +156,7 @@ pub fn main() anyerror!void {
SDL.SDLK_s => io.keyinput.shoulder_r.set(),
SDL.SDLK_RETURN => io.keyinput.start.set(),
SDL.SDLK_RSHIFT => io.keyinput.select.set(),
SDL.SDLK_i => std.debug.print("{} samples\n", .{@intCast(u32, SDL.SDL_AudioStreamAvailable(cpu.bus.apu.stream)) / (2 * @sizeOf(f32))}),
else => {},
}
},
@ -238,19 +240,31 @@ fn createTexture(renderer: *SDL.SDL_Renderer, width: c_int, height: c_int) *SDL.
) orelse sdlPanic();
}
fn initAudio() SDL.SDL_AudioDeviceID {
fn initAudio(apu: *Apu) SDL.SDL_AudioDeviceID {
var have: SDL.SDL_AudioSpec = undefined;
var want = std.mem.zeroes(SDL.SDL_AudioSpec);
want.freq = 32768;
want.format = SDL.AUDIO_F32;
want.channels = 2;
want.samples = 0x100;
want.callback = null;
var want: SDL.SDL_AudioSpec = .{
.freq = sample_rate,
.format = SDL.AUDIO_F32,
.channels = 2,
.samples = 0x100,
.callback = audioCallback,
.userdata = apu,
.silence = undefined,
.size = undefined,
.padding = undefined,
};
const dev = SDL.SDL_OpenAudioDevice(null, 0, &want, &have, 0);
if (dev == 0) sdlPanic();
// Start Playback on the Audio evice
// Start Playback on the Audio device
SDL.SDL_PauseAudioDevice(dev, 0);
return dev;
}
export fn audioCallback(userdata: ?*anyopaque, stream: [*c]u8, len: c_int) void {
const apu = @ptrCast(*Apu, @alignCast(8, userdata));
const result = SDL.SDL_AudioStreamGet(apu.stream, stream, len);
if (result < 0) log.err("Audio Callback Underflow", .{});
}