tmp: removed audio resampler
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@@ -15,12 +15,10 @@ const SoundFifo = std.fifo.LinearFifo(u8, .{ .Static = 0x20 });
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const getHalf = util.getHalf;
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const setHalf = util.setHalf;
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const intToBytes = util.intToBytes;
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const RingBuffer = util.RingBuffer;
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const log = std.log.scoped(.APU);
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pub const host_rate = @import("../platform.zig").sample_rate;
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pub const host_format = @import("../platform.zig").sample_format;
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pub fn read(comptime T: type, apu: *const Apu, addr: u32) ?T {
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const byte_addr = @truncate(u8, addr);
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@@ -246,17 +244,20 @@ pub const Apu = struct {
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sampling_cycle: u2,
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stream: *SDL.SDL_AudioStream,
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sample_queue: RingBuffer(u16),
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sched: *Scheduler,
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fs: FrameSequencer,
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capacitor: f32,
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is_buffer_full: bool,
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pub const Tick = enum { Length, Envelope, Sweep };
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pub fn init(sched: *Scheduler) Self {
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const NUM_CHANNELS: usize = 2;
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const allocator = std.heap.c_allocator;
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const sample_buf = allocator.alloc(u16, 0x800 * NUM_CHANNELS) catch @panic("failed to allocate sample buffer");
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const apu: Self = .{
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.ch1 = ToneSweep.init(sched),
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.ch2 = Tone.init(sched),
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@@ -271,12 +272,11 @@ pub const Apu = struct {
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.bias = .{ .raw = 0x0200 },
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.sampling_cycle = 0b00,
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.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_U16, 2, 1 << 15, host_format, 2, host_rate).?,
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.sample_queue = RingBuffer(u16).init(sample_buf),
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.sched = sched,
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.capacitor = 0,
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.fs = FrameSequencer.init(),
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.is_buffer_full = false,
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};
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sched.push(.SampleAudio, apu.interval());
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@@ -370,11 +370,6 @@ pub const Apu = struct {
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pub fn sampleAudio(self: *Self, late: u64) void {
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self.sched.push(.SampleAudio, self.interval() -| late);
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// Whether the APU is busy or not is determined by the main loop in emu.zig
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// This should only ever be true (because this side of the emu is single threaded)
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// When audio sync is disaabled
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if (self.is_buffer_full) return;
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var left: i16 = 0;
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var right: i16 = 0;
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@@ -430,23 +425,7 @@ pub const Apu = struct {
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const ext_left = (clamped_left << 5) | (clamped_left >> 6);
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const ext_right = (clamped_right << 5) | (clamped_right >> 6);
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if (self.sampling_cycle != self.bias.sampling_cycle.read()) self.replaceSDLResampler();
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_ = SDL.SDL_AudioStreamPut(self.stream, &[2]u16{ ext_left, ext_right }, 2 * @sizeOf(u16));
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}
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fn replaceSDLResampler(self: *Self) void {
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@setCold(true);
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const sample_rate = Self.sampleRate(self.bias.sampling_cycle.read());
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log.info("Sample Rate changed from {}Hz to {}Hz", .{ Self.sampleRate(self.sampling_cycle), sample_rate });
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// Sampling Cycle (Sample Rate) changed, Craete a new SDL Audio Resampler
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// FIXME: Replace SDL's Audio Resampler with either a custom or more reliable one
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const old_stream = self.stream;
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defer SDL.SDL_FreeAudioStream(old_stream);
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self.sampling_cycle = self.bias.sampling_cycle.read();
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self.stream = SDL.SDL_NewAudioStream(SDL.AUDIO_U16, 2, @intCast(c_int, sample_rate), host_format, 2, host_rate).?;
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self.sample_queue.push(ext_left, ext_right) catch {};
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}
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fn interval(self: *const Self) u64 {
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@@ -5,6 +5,7 @@ const config = @import("../config.zig");
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const Scheduler = @import("scheduler.zig").Scheduler;
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const Arm7tdmi = @import("cpu.zig").Arm7tdmi;
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const FpsTracker = @import("../util.zig").FpsTracker;
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const RingBuffer = @import("../util.zig").RingBuffer;
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const Timer = std.time.Timer;
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const Atomic = std.atomic.Atomic;
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@@ -58,7 +59,7 @@ fn inner(comptime kind: RunKind, audio_sync: bool, quit: *Atomic(bool), schedule
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while (!quit.load(.Monotonic)) {
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runFrame(scheduler, cpu);
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audioSync(audio_sync, cpu.bus.apu.stream, &cpu.bus.apu.is_buffer_full);
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audioSync(audio_sync, &cpu.bus.apu.sample_queue);
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if (kind == .UnlimitedFPS) tracker.?.tick();
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}
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@@ -77,7 +78,7 @@ fn inner(comptime kind: RunKind, audio_sync: bool, quit: *Atomic(bool), schedule
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// the amount of time needed for audio to catch up rather than
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// our expected wake-up time
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audioSync(audio_sync, cpu.bus.apu.stream, &cpu.bus.apu.is_buffer_full);
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audioSync(audio_sync, &cpu.bus.apu.sample_queue);
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if (!audio_sync) spinLoop(&timer, wake_time);
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wake_time = new_wake_time;
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@@ -104,22 +105,13 @@ pub fn runFrame(sched: *Scheduler, cpu: *Arm7tdmi) void {
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}
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}
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fn audioSync(audio_sync: bool, stream: *SDL.SDL_AudioStream, is_buffer_full: *bool) void {
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fn audioSync(audio_sync: bool, sample_queue: *RingBuffer(u16)) void {
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comptime std.debug.assert(@import("../platform.zig").sample_format == SDL.AUDIO_U16);
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const sample_size = 2 * @sizeOf(u16);
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const max_buf_size: c_int = 0x400;
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// const sample_size = 2 * @sizeOf(u16);
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// const max_buf_size: c_int = 0x400;
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// Determine whether the APU is busy right at this moment
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var still_full: bool = SDL.SDL_AudioStreamAvailable(stream) > sample_size * if (is_buffer_full.*) max_buf_size >> 1 else max_buf_size;
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defer is_buffer_full.* = still_full; // Update APU Busy status right before exiting scope
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// If Busy is false, there's no need to sync here
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if (!still_full) return;
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while (true) {
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still_full = SDL.SDL_AudioStreamAvailable(stream) > sample_size * max_buf_size >> 1;
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if (!audio_sync or !still_full) break;
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}
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_ = audio_sync;
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_ = sample_queue;
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}
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fn videoSync(timer: *Timer, wake_time: u64) u64 {
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