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Fixed giltching from audio frame assembly from queue 

It wasn't copying the frame data from the sources to the output frame
correctly, and there was a small period of zeros at the start of the
frame.

I also converted to using a vec of tuples for the two channels to
make it easier to reason about, since some of the issues were related
to the differences sizes calculated from the clock based on the sample
rate
This commit is contained in:
transistor 2022-10-08 16:17:17 -07:00
parent e4cd3dcd63
commit 0876d85626
3 changed files with 43 additions and 180 deletions
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10
docs/log.txt
View File

@ -272,7 +272,15 @@ Audio
check before the audio devices write only account for one channel of audio instead of two, so the
buffer was over filling. Dividing the available buffer size by 2 fixed it
2022-10-08
- I replaced the circular buffer with queues of audio frames, which might be less able to handle time
dialation, but which uses fewer locks for less time, which seems to improve the occasional dropout
glitches. The next frame is also assembled by whichever sim audio source that pushes data sees
the queue is empty (audio frame taken by the output callback), so it then assembles the next frame
and pushes it to the queue. It definitely works better, but there are still some timing issues
- the PCM data in ym2612, looking at the recording of the waveform, shows it's offset by a bias, which
probably shouldn't be. The other audio sounds distorted but still sort of sounds like what it should,
so I don't think it's too far off

205
emulator/frontends/common/src/audio.rs
View File

@ -5,14 +5,13 @@ use cpal::{Sample, Stream, SampleRate, SampleFormat, StreamConfig, traits::{Devi
use moa_core::Clock;
use moa_core::host::{HostData, Audio, ClockedQueue};
use crate::circularbuf::CircularBuffer;
const SAMPLE_RATE: usize = 48000;
#[derive(Clone)]
pub struct AudioFrame {
data: Vec<f32>,
data: Vec<(f32, f32)>,
}
pub struct AudioSource {
@ -21,7 +20,6 @@ pub struct AudioSource {
frame_size: usize,
sequence_num: usize,
mixer: Arc<Mutex<AudioMixer>>,
buffer: CircularBuffer<f32>,
queue: ClockedQueue<AudioFrame>,
}
@ -37,7 +35,6 @@ impl AudioSource {
mixer.frame_size(),
)
};
let buffer = CircularBuffer::new(frame_size * 2, 0.0);
Self {
id,
@ -45,29 +42,24 @@ impl AudioSource {
frame_size,
sequence_num: 0,
mixer,
buffer,
queue,
}
}
pub fn space_available(&self) -> usize {
//self.buffer.free_space() / 2
self.frame_size / 2
}
pub fn fill_with(&mut self, clock: Clock, buffer: &[f32]) {
let mut data = vec![];
//if self.buffer.free_space() > buffer.len() * 2 {
for sample in buffer.iter() {
// TODO this is here to keep it quiet for testing, but should be removed later
let sample = 0.5 * *sample;
data.push(sample);
data.push(sample);
}
//}
for sample in buffer.iter() {
// TODO this is here to keep it quiet for testing, but should be removed later
let sample = 0.5 * *sample;
data.push((sample, sample));
}
let frame = AudioFrame {
data, //: Vec::from(buffer)
data,
};
//println!("synthesized {}: {:?}", self.id, frame.data);
@ -80,7 +72,6 @@ impl AudioSource {
self.mixer.lock().unwrap().check_next_frame();
}
/*
pub fn fill_with(&mut self, buffer: &[f32]) {
if self.buffer.free_space() > buffer.len() * 2 {
@ -119,12 +110,6 @@ impl AudioSource {
*/
}
// could have the audio source use the circular buffer and then publish to its queue, and then call the mixer to flush if possible,
// and have the mixer (from the sim thread effectively) build the frame and publish it to its output. Frames in the source queues
// could even be 1ms, and the assembler could just fetch multiple frames, adjusting for sim time
// you could either only use the circular buffer, or only use the source queue
impl Audio for AudioSource {
fn samples_per_second(&self) -> usize {
@ -144,8 +129,6 @@ impl Audio for AudioSource {
}
}
use moa_core::host::audio::SquareWave;
#[derive(Clone)]
pub struct AudioMixer {
sample_rate: usize,
@ -155,7 +138,6 @@ pub struct AudioMixer {
sources: Vec<ClockedQueue<AudioFrame>>,
buffer_underrun: bool,
output: Arc<Mutex<AudioOutput>>,
test: SquareWave,
}
impl AudioMixer {
@ -168,7 +150,6 @@ impl AudioMixer {
sources: vec![],
buffer_underrun: false,
output: AudioOutput::new(),
test: SquareWave::new(600.0, sample_rate),
}))
}
@ -194,7 +175,6 @@ impl AudioMixer {
}
pub fn frame_size(&self) -> usize {
//self.buffer.len()
self.frame_size
}
@ -203,9 +183,6 @@ impl AudioMixer {
}
pub fn resize_frame(&mut self, newlen: usize) {
//if self.buffer.len() != newlen {
// self.buffer = vec![0.0; newlen];
//}
self.frame_size = newlen;
}
@ -219,7 +196,7 @@ impl AudioMixer {
self.frame_size = self.output.lock().unwrap().frame_size;
let nanos_per_sample = self.nanos_per_sample();
let mut data: Vec<f32> = vec![0.0; self.frame_size];
let mut data: Vec<(f32, f32)> = vec![(0.0, 0.0); self.frame_size];
if self.buffer_underrun {
self.buffer_underrun = false;
@ -230,14 +207,6 @@ impl AudioMixer {
return;
}
/*
for i in (0..data.len()).step_by(2) {
let sample = self.test.next().unwrap() * 0.5;
data[i] = sample;
data[i + 1] = sample;
}
*/
let lowest_clock = self.sources
.iter()
.fold(self.clock, |lowest_clock, source|
@ -260,18 +229,28 @@ impl AudioMixer {
//println!("clock: {} - {} = {}", clock, self.clock, clock - self.clock);
//if clock > self.clock {
let start = ((2 * (clock - self.clock) / nanos_per_sample) as usize).min(data.len() - 1);
let start = (((clock - self.clock) / nanos_per_sample) as usize).min(data.len() - 1);
let length = frame.data.len().min(data.len() - start);
//println!("source: {}, clock: {}, start: {}, end: {}, length: {}", id, clock, start, start + length, length);
data[start..start + length].iter_mut().zip(frame.data[..length].iter()).for_each(|(d, s)| *d = (*d + s).clamp(-1.0, 1.0));
data[start..start + length].iter_mut()
.zip(frame.data[..length].iter())
.for_each(|(d, s)|
*d = (
(d.0 + s.0).clamp(-1.0, 1.0),
(d.1 + s.1).clamp(-1.0, 1.0)
)
);
//println!("{} {} {} {}", i, start, length, frame.data.len());
if length < frame.data.len() {
let adjusted_clock = clock + nanos_per_sample * (length / 2) as Clock;
//println!("unpopping {} {}", clock, adjusted_clock);
let adjusted_clock = clock + nanos_per_sample * length as Clock;
//println!("unpopping at clock {}, length {}", adjusted_clock, frame.data.len() - length);
source.unpop(adjusted_clock, AudioFrame { data: frame.data[length..].to_vec() });
}
//}
// TODO we need to handle the opposite case
i += length;
i = start + length;
//println!("{}", i);
}
}
self.clock += nanos_per_sample * data.len() as Clock;
@ -279,79 +258,6 @@ impl AudioMixer {
//println!("{:?}", data);
self.output.lock().unwrap().add_frame(AudioFrame { data });
}
/*
pub fn assembly_frame(&mut self, data: &mut [f32]) {
self.resize_frame(data.len());
println!("assemble audio frame {}", self.sequence_num);
//for i in 0..data.len() {
// data[i] = Sample::from(&self.buffer[i]);
// self.buffer[i] = 0.0;
//}
//self.sources
// .iter()
// .filter_map(|queue| queue.pop_latest())
// .fold(data, |data, frame| {
// data.iter_mut()
// .zip(frame.1.data.iter())
// .for_each(|(d, s)| *d = (*d + s).clamp(-1.0, 1.0));
// data
// });
if let Some((_, last)) = self.output.pop_latest() {
self.last_frame = Some(last);
}
if let Some(last) = &self.last_frame {
data.copy_from_slice(&last.data);
}
println!("frame {} sent", self.sequence_num);
self.sequence_num = self.sequence_num.wrapping_add(1);
/*
let mut buffer = vec![0.0; data.len()];
for source in &self.sources {
let mut locked_source = source.lock();
// TODO these are quick hacks to delay or shrink the buffer if it's too small or big
if locked_source.used_space() < data.len() {
continue;
}
let excess = locked_source.used_space() - (data.len() * 2);
if excess > 0 {
locked_source.drop_next(excess);
}
for addr in buffer.iter_mut() {
*addr += locked_source.next().unwrap_or(0.0);
}
}
for i in 0..data.len() {
let sample = buffer[i] / self.sources.len() as f32;
data[i] = Sample::from(&sample);
}
*/
/*
let mut locked_source = self.sources[1].lock();
for i in 0..data.len() {
let sample = locked_source.next().unwrap_or(0.0);
data[i] = Sample::from(&sample);
}
*/
}
*/
// TODO you need a way to add data to the mixer... the question is do you need to keep track of real time
// If you have a counter that calculates the amount of time until the next sample based on the size of
// the buffer given to the data_callback, then when submitting data, the audio sources can know that they
// the next place to write to is a given position in the mixer buffer (maybe not the start of the buffer).
// But what do you do if there needs to be some skipping. If the source is generating data in 1 to 10 ms
// chunks according to simulated time, there might be a case where it tries to write too much data because
// it's running fast. (If it's running slow, you can insert silence)
}
#[allow(dead_code)]
@ -365,7 +271,7 @@ pub struct AudioOutput {
impl AudioOutput {
pub fn new() -> Arc<Mutex<Self>> {
Arc::new(Mutex::new(Self {
frame_size: 1280,
frame_size: 0,
sequence_num: 0,
last_frame: None,
output: VecDeque::with_capacity(2),
@ -414,19 +320,19 @@ impl CpalAudioOutput {
let data_callback = move |data: &mut [f32], _: &cpal::OutputCallbackInfo| {
let result = if let Ok(mut output) = output.lock() {
output.frame_size = data.len();
output.frame_size = data.len() / 2;
output.pop_next()
} else {
return;
};
if let Some(frame) = result {
//println!("needs {}, gets {}", data.len(), frame.data.len());
//println!("{:?}", frame.data);
let length = frame.data.len().min(data.len());
data[..length].copy_from_slice(&frame.data[..length]);
let (start, middle, end) = unsafe { frame.data.align_to::<f32>() };
//assert!(start.len() == 0 && end.len() == 0);
let length = middle.len().min(data.len());
data[..length].copy_from_slice(&middle[..length]);
} else {
println!("missed a frame");
println!("missed an audio frame");
}
};
@ -444,56 +350,5 @@ impl CpalAudioOutput {
stream,
}
}
/*
pub fn create_audio_output2(mut updater: Box<dyn AudioUpdater>) -> AudioOutput {
let device = cpal::default_host()
.default_output_device()
.expect("No sound output device available");
let config: StreamConfig = device
.supported_output_configs()
.expect("error while querying configs")
.find(|config| config.sample_format() == SampleFormat::F32 && config.channels() == 2)
.expect("no supported config?!")
.with_sample_rate(SampleRate(SAMPLE_RATE as u32))
.into();
let channels = config.channels as usize;
let mixer = AudioMixer::new(SAMPLE_RATE);
let data_callback = {
let mixer = mixer.clone();
move |data: &mut [f32], _: &cpal::OutputCallbackInfo| {
let samples = data.len() / 2;
let mut buffer = vec![0.0; samples];
updater.update_audio_frame(samples, mixer.lock().sample_rate(), &mut buffer);
for (i, channels) in data.chunks_mut(2).enumerate() {
let sample = Sample::from(&buffer[i]);
channels[0] = sample;
channels[1] = sample;
}
}
};
let stream = device.build_output_stream(
&config,
data_callback,
move |err| {
// react to errors here.
println!("ERROR");
},
).unwrap();
stream.play().unwrap();
AudioOutput {
stream,
mixer,
}
}
*/
}

8
emulator/frontends/minifb/src/lib.rs
View File

@ -246,13 +246,13 @@ impl MiniFrontend {
// Limit to max ~60 fps update rate
window.limit_update_rate(Some(Duration::from_micros(16600)));
let mut average_time = 0;
//let mut average_time = 0;
let mut update_timer = Instant::now();
let mut last_frame = Frame::new(size.0, size.1);
while window.is_open() && !window.is_key_down(Key::Escape) {
let frame_time = update_timer.elapsed();
update_timer = Instant::now();
println!("new frame after {:?}us", frame_time.as_micros());
//println!("new frame after {:?}us", frame_time.as_micros());
let run_timer = Instant::now();
if let Some(system) = system.as_mut() {
@ -261,8 +261,8 @@ impl MiniFrontend {
//system.run_until_break().unwrap();
}
let sim_time = run_timer.elapsed().as_micros();
average_time = (average_time + sim_time) / 2;
println!("ran simulation for {:?}us in {:?}us (avg: {:?}us)", frame_time.as_nanos() / 1_000, sim_time, average_time);
//average_time = (average_time + sim_time) / 2;
//println!("ran simulation for {:?}us in {:?}us (avg: {:?}us)", frame_time.as_nanos() / 1_000, sim_time, average_time);
if let Some(keys) = window.get_keys_pressed(minifb::KeyRepeat::No) {
for key in keys {