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feat: reimplement audio sync 

APU will now drop samples if the Audio Queue is already full, therefore
creating a "sped-up" effect when the emulator runs faster than 100%
This commit is contained in:
Rekai Nyangadzayi Musuka 2022-09-08 20:38:42 -03:00
parent fa862f095a
commit 65cfc97f28
3 changed files with 58 additions and 33 deletions
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4
src/Gui.zig
View File

@ -180,11 +180,11 @@ const Audio = struct {
export fn callback(userdata: ?*anyopaque, stream: [*c]u8, len: c_int) void {
const apu = @ptrCast(*Apu, @alignCast(@alignOf(*Apu), userdata));
const written = SDL.SDL_AudioStreamGet(apu.stream, stream, len);
_ = SDL.SDL_AudioStreamGet(apu.stream, stream, len);
// If we don't write anything, play silence otherwise garbage will be played
// FIXME: I don't think this hack to remove DC Offset is acceptable :thinking:
if (written == 0) std.mem.set(u8, stream[0..@intCast(usize, len)], 0x40);
// if (written == 0) std.mem.set(u8, stream[0..@intCast(usize, len)], 0x40);
}
};

46
src/core/apu.zig
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@ -163,6 +163,8 @@ pub const Apu = struct {
fs: FrameSequencer,
capacitor: f32,
is_buffer_full: bool,
pub fn init(sched: *Scheduler) Self {
const apu: Self = .{
.ch1 = ToneSweep.init(sched),
@ -178,11 +180,12 @@ pub const Apu = struct {
.bias = .{ .raw = 0x0200 },
.sampling_cycle = 0b00,
.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_U16, 2, 1 << 15, SDL.AUDIO_U16, 2, host_sample_rate) orelse unreachable,
.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_U16, 2, 1 << 15, SDL.AUDIO_U16, 2, host_sample_rate).?,
.sched = sched,
.capacitor = 0,
.fs = FrameSequencer.init(),
.is_buffer_full = false,
};
sched.push(.SampleAudio, apu.sampleTicks());
@ -277,6 +280,13 @@ pub const Apu = struct {
}
pub fn sampleAudio(self: *Self, late: u64) void {
self.sched.push(.SampleAudio, self.sampleTicks() -| 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;
@ -325,28 +335,30 @@ pub const Apu = struct {
left += bias;
right += bias;
const tmp_left = std.math.clamp(@bitCast(u16, left), std.math.minInt(u11), std.math.maxInt(u11));
const tmp_right = std.math.clamp(@bitCast(u16, right), std.math.minInt(u11), std.math.maxInt(u11));
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 final_left = (tmp_left << 5) | (tmp_left >> 6);
const final_right = (tmp_right << 5) | (tmp_right >> 6);
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()) {
const new_sample_rate = Self.sampleRate(self.bias.sampling_cycle.read());
log.info("Sample Rate changed from {}Hz to {}Hz", .{ Self.sampleRate(self.sampling_cycle), new_sample_rate });
// FIXME: This rarely happens
if (self.sampling_cycle != self.bias.sampling_cycle.read()) self.replaceSDLResampler();
// 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_U16, 2, @intCast(c_int, new_sample_rate), SDL.AUDIO_U16, 2, host_sample_rate) orelse unreachable;
_ = SDL.SDL_AudioStreamPut(self.stream, &[2]u16{ ext_left, ext_right }, 2 * @sizeOf(u16));
}
_ = SDL.SDL_AudioStreamPut(self.stream, &[2]u16{ final_left, final_right }, 2 * @sizeOf(u16));
self.sched.push(.SampleAudio, self.sampleTicks() -| late);
fn replaceSDLResampler(self: *Self) void {
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), SDL.AUDIO_U16, 2, host_sample_rate).?;
}
fn sampleTicks(self: *const Self) u64 {

41
src/core/emu.zig
View File

@ -70,12 +70,21 @@ pub fn runFrame(sched: *Scheduler, cpu: *Arm7tdmi) void {
}
}
fn syncToAudio(cpu: *const Arm7tdmi) void {
const stream = cpu.bus.apu.stream;
const min_sample_count = 0x800;
fn syncToAudio(stream: *SDL.SDL_AudioStream, is_buffer_full: *bool) void {
const sample_size = 2 * @sizeOf(u16);
const max_buf_size: c_int = 0x400;
// Busy Loop while we wait for the Audio system to catch up
while (SDL.SDL_AudioStreamAvailable(stream) > (@sizeOf(u16) * 2) * min_sample_count) {}
// Determine whether the APU is busy right at this moment
var still_full: bool = SDL.SDL_AudioStreamAvailable(stream) > sample_size * if (is_buffer_full.*) max_buf_size >> 1 else max_buf_size;
defer is_buffer_full.* = still_full; // Update APU Busy status right before exiting scope
// If Busy is false, there's no need to sync here
if (!still_full) return;
while (true) {
still_full = SDL.SDL_AudioStreamAvailable(stream) > sample_size * max_buf_size >> 1;
if (!sync_audio or !still_full) break;
}
}
pub fn runUnsynchronized(quit: *Atomic(bool), sched: *Scheduler, cpu: *Arm7tdmi, fps: ?*FpsTracker) void {
@ -86,21 +95,21 @@ pub fn runUnsynchronized(quit: *Atomic(bool), sched: *Scheduler, cpu: *Arm7tdmi,
while (!quit.load(.SeqCst)) {
runFrame(sched, cpu);
if (sync_audio) syncToAudio(cpu);
syncToAudio(cpu.bus.apu.stream, &cpu.bus.apu.is_buffer_full);
tracker.tick();
}
} else {
while (!quit.load(.SeqCst)) {
runFrame(sched, cpu);
if (sync_audio) syncToAudio(cpu);
syncToAudio(cpu.bus.apu.stream, &cpu.bus.apu.is_buffer_full);
}
}
}
pub fn runSynchronized(quit: *Atomic(bool), sched: *Scheduler, cpu: *Arm7tdmi, fps: ?*FpsTracker) void {
log.info("Emulation thread w/ video sync", .{});
var timer = Timer.start() catch unreachable;
var timer = Timer.start() catch std.debug.panic("Failed to initialize std.timer.Timer", .{});
var wake_time: u64 = frame_period;
if (fps) |tracker| {
@ -108,13 +117,14 @@ pub fn runSynchronized(quit: *Atomic(bool), sched: *Scheduler, cpu: *Arm7tdmi, f
while (!quit.load(.SeqCst)) {
runFrame(sched, cpu);
const new_wake_time = syncToVideo(&timer, wake_time);
const new_wake_time = blockOnVideo(&timer, wake_time);
// Spin to make up the difference of OS scheduler innacuracies
// If we happen to also be syncing to audio, we choose to spin on
// the amount of time needed for audio to catch up rather than
// our expected wake-up time
if (sync_audio) syncToAudio(cpu) else spinLoop(&timer, wake_time);
syncToAudio(cpu.bus.apu.stream, &cpu.bus.apu.is_buffer_full);
if (!sync_audio) spinLoop(&timer, wake_time);
wake_time = new_wake_time;
tracker.tick();
@ -122,16 +132,17 @@ pub fn runSynchronized(quit: *Atomic(bool), sched: *Scheduler, cpu: *Arm7tdmi, f
} else {
while (!quit.load(.SeqCst)) {
runFrame(sched, cpu);
const new_wake_time = syncToVideo(&timer, wake_time);
// see above comment
if (sync_audio) syncToAudio(cpu) else spinLoop(&timer, wake_time);
const new_wake_time = blockOnVideo(&timer, wake_time);
// see above comment
syncToAudio(cpu.bus.apu.stream, &cpu.bus.apu.is_buffer_full);
if (!sync_audio) spinLoop(&timer, wake_time);
wake_time = new_wake_time;
}
}
}
inline fn syncToVideo(timer: *Timer, wake_time: u64) u64 {
inline fn blockOnVideo(timer: *Timer, wake_time: u64) u64 {
// Use the OS scheduler to put the emulation thread to sleep
const maybe_recalc_wake_time = sleep(timer, wake_time);
@ -149,6 +160,8 @@ pub fn runBusyLoop(quit: *Atomic(bool), sched: *Scheduler, cpu: *Arm7tdmi) void
runFrame(sched, cpu);
spinLoop(&timer, wake_time);
syncToAudio(cpu.bus.apu.stream, &cpu.bus.apu.is_buffer_full);
// Update to the new wake time
wake_time += frame_period;
}