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feat: reimplement audio sync

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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
2022-10-21 05:12:57 -03:00
parent b2386a6a2b
commit d2a50cf9d2
3 changed files with 58 additions and 33 deletions
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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);
_ = SDL.SDL_AudioStreamPut(self.stream, &[2]u16{ ext_left, ext_right }, 2 * @sizeOf(u16));
}
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;
}
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 });
_ = SDL.SDL_AudioStreamPut(self.stream, &[2]u16{ final_left, final_right }, 2 * @sizeOf(u16));
self.sched.push(.SampleAudio, self.sampleTicks() -| late);
// 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 {