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feat: add audio resampler

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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
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41
src/apu.zig
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@@ -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 {