experimental dynamic rate control feature in audio component (TBD: merge into SimpleSdl2Audio as an option)

This commit is contained in:
lifning 2021-12-11 03:10:39 -08:00
parent 72366248d7
commit b2c2c93537
2 changed files with 50 additions and 20 deletions

View File

@ -99,7 +99,7 @@ pub fn main() -> Result<(), Box<dyn std::error::Error + Send + Sync>> {
}
}
emu.register_component(SimpleSdl2AudioComponent::new(&mut sdl_context)?)?;
emu.register_component(Sdl2RateControlledAudioComponent::new(&mut sdl_context)?)?;
emu.register_component(SimpleSdl2KeyboardComponent::new(&mut sdl_context)?)?;
emu.register_component(SimpleSdl2GamepadComponent::new(&mut sdl_context))?;

View File

@ -12,13 +12,22 @@ use super::audio::resample;
pub struct Sdl2RateControlledAudioComponent {
src_freq: f64,
ratio: f64,
audio_buffer: Vec<i16>,
queue: AudioQueue<i16>,
avg_headroom: f64,
started: bool,
}
impl RetroCallbacks for Sdl2RateControlledAudioComponent {
fn video_refresh(&mut self, _frame: &VideoFrame) {
self.started = true;
}
fn audio_samples(&mut self, stereo_pcm: &[i16]) -> usize {
self.audio_buffer.extend(stereo_pcm);
if self.started {
self.audio_buffer.extend(stereo_pcm);
}
stereo_pcm.len() / 2
}
@ -33,25 +42,47 @@ impl RetroCallbacks for Sdl2RateControlledAudioComponent {
}
impl RetroComponent for Sdl2RateControlledAudioComponent {
fn post_load_game(&mut self, _retro: &mut LibretroWrapper, _rom: &Path) -> Result<(), Box<dyn Error + Send + Sync>> {
fn post_load_game(&mut self, retro: &mut LibretroWrapper, _rom: &Path) -> Result<(), Box<dyn Error + Send + Sync>> {
self.src_freq = retro.get_system_av_info().timing.sample_rate;
self.queue.resume();
self.started = false;
Ok(())
}
fn post_run(&mut self, _retro: &mut LibretroWrapper) -> ControlFlow {
if let Ok(converter) = Self::make_converter(self.src_freq, self.queue.spec()) {
let mut samples = std::mem::take(&mut self.audio_buffer);
samples = resample(&converter, samples);
self.audio_buffer = samples;
self.queue.queue(&self.audio_buffer);
self.audio_buffer.clear();
if self.src_freq != 0.0 {
if self.started {
let sample_headroom = (self.queue.size() / 4) as f64;
self.avg_headroom = ((self.avg_headroom * 9.0) + sample_headroom) / 10.0;
self.avg_headroom /= 2.0;
let room = self.src_freq / 100.0;
let ratio_raw = ((room - self.avg_headroom) / room).clamp(-1.0, 1.0);
self.ratio = 1.0 + (ratio_raw * 0.01);
}
match Self::make_converter(self.src_freq * self.ratio, self.queue.spec()) {
Ok(converter) => {
if !self.audio_buffer.is_empty() {
let mut samples = std::mem::take(&mut self.audio_buffer);
samples = resample(&converter, samples);
self.audio_buffer = samples;
self.queue.queue(&self.audio_buffer);
self.audio_buffer.clear();
}
ControlFlow::Continue
}
Err(e) => {
eprintln!("Audio sample rate conversion failed: {:?}", e);
ControlFlow::Break
}
}
} else {
ControlFlow::Continue
}
ControlFlow::Continue
}
}
impl Sdl2RateControlledAudioComponent {
pub fn new(sdl_context: &mut Sdl, retro: &LibretroWrapper) -> Result<Self, Box<dyn std::error::Error>> {
pub fn new(sdl_context: &mut Sdl) -> crate::base::Result<Self> {
let audio = sdl_context.audio().unwrap();
let desired_spec = AudioSpecDesired {
freq: None,
@ -61,31 +92,30 @@ impl Sdl2RateControlledAudioComponent {
let queue = AudioQueue::open_queue(&audio, None, &desired_spec)?;
let mut src_freq = retro.get_system_av_info().timing.sample_rate;
// HACK: some cores don't report this 'til we get an env call to set_system_av_info,
// so we can just default to the old libsnes default value.
if src_freq == 0.0 {
src_freq = 32040.5;
}
// default to the old libsnes default value until after load_game or set_system_av_info.
let src_freq = 32040.5;
Ok(Sdl2RateControlledAudioComponent {
src_freq,
ratio: 1.0,
audio_buffer: Default::default(),
queue,
avg_headroom: 0.0,
started: false,
})
}
fn make_converter(src_freq: f64, dest_spec: &AudioSpec) -> Result<AudioCVT, String> {
// note on the `* 64`: as long as the ratio between src_rate and dst_rate is right,
// note on the `* 16`: as long as the ratio between src_rate and dst_rate is right,
// we should be in the clear -- this is to make up for SDL not giving us floats for
// this, we can at least get some quasi-fixed-point precision going on...
AudioCVT::new(
AudioFormat::s16_sys(),
2,
(src_freq * 64.0).round() as i32,
(src_freq * 16.0).round() as i32,
dest_spec.format,
dest_spec.channels,
dest_spec.freq * 64,
dest_spec.freq * 16,
)
}
}