fixed_resample/channel.rs
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use std::{
num::NonZeroUsize,
sync::{
atomic::{AtomicBool, Ordering},
Arc,
},
};
use ringbuf::traits::{Consumer, Observer, Producer, Split};
use rubato::Sample;
use crate::{ResampleQuality, ResamplerType, RtResampler};
/// Additional options for a resampling channel.
#[derive(Debug, Clone, Copy, PartialEq)]
pub struct ResamplingChannelConfig {
/// The amount of latency added in seconds between the input stream and the
/// output stream. If this value is too small, then underflows may occur.
///
/// The default value is `0.15` (150 ms).
pub latency_seconds: f64,
/// The capacity of the channel in seconds. If this is too small, then
/// overflows may occur. This should be at least twice as large as
/// `latency_seconds`.
///
/// The default value is `0.4` (400 ms).
pub capacity_seconds: f64,
/// The quality of the resampling alrgorithm to use if needed.
///
/// The default value is `ResampleQuality::Normal`.
pub quality: ResampleQuality,
}
impl Default for ResamplingChannelConfig {
fn default() -> Self {
Self {
latency_seconds: 0.15,
capacity_seconds: 0.4,
quality: ResampleQuality::Normal,
}
}
}
/// Create a new realtime-safe spsc channel for sending samples across streams.
///
/// If the input and output samples rates differ, then this will automatically
/// resample the input stream to match the output stream. If the sample rates
/// match, then no resampling will occur.
///
/// Internally this uses the `ringbuf` crate.
///
/// * `in_sample_rate` - The sample rate of the input stream.
/// * `out_sample_rate` - The sample rate of the output stream.
/// * `num_channels` - The number of channels in the stream.
/// * `config` - Additional options for the resampling channel.
///
/// # Panics
///
/// Panics when any of the following are true:
///
/// * `in_sample_rate == 0`
/// * `out_sample_rate == 0`
/// * `num_channels == 0`
/// * `config.latency_seconds <= 0.0`
/// * `config.capacity_seconds <= 0.0`
pub fn resampling_channel<T: Sample>(
in_sample_rate: u32,
out_sample_rate: u32,
num_channels: usize,
config: ResamplingChannelConfig,
) -> (ResamplingProd<T>, ResamplingCons<T>) {
let resampler = if in_sample_rate != out_sample_rate {
Some(RtResampler::<T>::new(
in_sample_rate,
out_sample_rate,
num_channels,
true,
config.quality,
))
} else {
None
};
resampling_channel_inner(
resampler,
in_sample_rate,
out_sample_rate,
num_channels,
config,
)
}
/// Create a new realtime-safe spsc channel for sending samples across streams
/// using the custom resampler.
///
/// If the input and output samples rates differ, then this will automatically
/// resample the input stream to match the output stream. If the sample rates
/// match, then no resampling will occur.
///
/// Internally this uses the `ringbuf` crate.
///
/// * `resampler` - The custom rubato resampler.
/// * `in_sample_rate` - The sample rate of the input stream.
/// * `out_sample_rate` - The sample rate of the output stream.
/// * `num_channels` - The number of channels in the stream.
/// * `config` - Additional options for the resampling channel. Note that
/// `config.quality` will be ignored.
///
/// # Panics
///
/// Panics when any of the following are true:
///
/// * `resampler.num_channels() != num_channels`
/// * `in_sample_rate == 0`
/// * `out_sample_rate == 0`
/// * `num_channels == 0`
/// * `config.latency_seconds <= 0.0`
/// * `config.capacity_seconds <= 0.0`
pub fn resampling_channel_custom<T: Sample>(
resampler: impl Into<ResamplerType<T>>,
in_sample_rate: u32,
out_sample_rate: u32,
num_channels: usize,
config: ResamplingChannelConfig,
) -> (ResamplingProd<T>, ResamplingCons<T>) {
let resampler: ResamplerType<T> = resampler.into();
assert_eq!(resampler.num_channels(), num_channels);
let resampler = if in_sample_rate != out_sample_rate {
Some(RtResampler::<T>::from_custom(resampler, true))
} else {
None
};
resampling_channel_inner(
resampler,
in_sample_rate,
out_sample_rate,
num_channels,
config,
)
}
fn resampling_channel_inner<T: Sample>(
resampler: Option<RtResampler<T>>,
in_sample_rate: u32,
out_sample_rate: u32,
num_channels: usize,
config: ResamplingChannelConfig,
) -> (ResamplingProd<T>, ResamplingCons<T>) {
assert_ne!(in_sample_rate, 0);
assert_ne!(out_sample_rate, 0);
assert_ne!(num_channels, 0);
assert!(config.latency_seconds > 0.0);
assert!(config.capacity_seconds > 0.0);
let latency_frames = ((in_sample_rate as f64 * config.latency_seconds).round() as usize).max(1);
let buffer_capacity_frames = ((in_sample_rate as f64 * config.capacity_seconds).round()
as usize)
.max(latency_frames * 2);
let (mut prod, cons) = ringbuf::HeapRb::<T>::new(buffer_capacity_frames * num_channels).split();
// Pad the beginning of the buffer with zeros to create the desired latency.
prod.push_slice(&vec![T::zero(); latency_frames * num_channels]);
let reset_flag = Arc::new(AtomicBool::new(false));
let in_sample_rate_recip = (in_sample_rate as f64).recip();
(
ResamplingProd {
prod,
num_channels: NonZeroUsize::new(num_channels).unwrap(),
latency_seconds: config.latency_seconds,
in_sample_rate_recip,
reset_flag: Arc::clone(&reset_flag),
},
ResamplingCons {
cons,
resampler,
num_channels: NonZeroUsize::new(num_channels).unwrap(),
latency_frames,
is_waiting_for_frames: true,
latency_seconds: config.latency_seconds,
in_sample_rate: in_sample_rate as f64,
in_sample_rate_recip,
reset_flag,
},
)
}
/// The producer end of a realtime-safe spsc channel for sending samples across
/// streams.
///
/// If the input and output samples rates differ, then this will automatically
/// resample the input stream to match the output stream. If the sample rates
/// match, then no resampling will occur.
///
/// Internally this uses the `ringbuf` crate.
pub struct ResamplingProd<T: Sample> {
prod: ringbuf::HeapProd<T>,
num_channels: NonZeroUsize,
latency_seconds: f64,
in_sample_rate_recip: f64,
reset_flag: Arc<AtomicBool>,
}
impl<T: Sample> ResamplingProd<T> {
/// Push the given data in interleaved format.
///
/// Returns the number of frames (not samples) that were successfully pushed.
/// If this number is less than the number of frames in `data`, then it means
/// an overflow has occured.
pub fn push(&mut self, data: &[T]) -> usize {
let data_frames = data.len() / self.num_channels.get();
let pushed_samples = self
.prod
.push_slice(&data[..data_frames * self.num_channels.get()]);
pushed_samples / self.num_channels.get()
}
/// Returns the number of frames that are currently available to be pushed
/// to the buffer.
pub fn available_frames(&self) -> usize {
self.prod.vacant_len() / self.num_channels.get()
}
/// The number of channels configured for this stream.
pub fn num_channels(&self) -> NonZeroUsize {
self.num_channels
}
/// An number describing the current amount of jitter in seconds between the
/// input and output streams. A value of `0.0` means the two channels are
/// perfectly synced, a value less than `0.0` means the input channel is
/// slower than the input channel, and a value greater than `0.0` means the
/// input channel is faster than the output channel.
///
/// This value can be used to correct for jitter and avoid underflows/
/// overflows. For example, if this value goes below a certain threshold,
/// then you can push an extra packet of data to correct for the jitter.
///
/// This number will be in the range `[-latency_seconds, capacity_seconds - latency_seconds]`,
/// where `latency_seconds` and `capacity_seconds` are the values passed in
/// [`ResamplingChannelConfig`] when this channel was constructed.
///
/// Note, it is typical for the jitter value to be around plus or minus
/// `out_max_block_frames / out_sample_rate` or `data_frames / in_sample_rate`
/// (whichever is higher) even when the streams are perfectly in sync
/// (`data_frames` being the typical length in frames of a packet of data pushed
/// to [`ResamplingProd::push`]).
pub fn jitter_seconds(&self) -> f64 {
((self.prod.occupied_len() / self.num_channels.get()) as f64 * self.in_sample_rate_recip)
- self.latency_seconds
}
/// Tell the consumer to clear all queued frames in the buffer.
pub fn reset(&mut self) {
self.reset_flag.store(true, Ordering::Relaxed);
}
}
/// The consumer end of a realtime-safe spsc channel for sending samples across
/// streams.
///
/// If the input and output samples rates differ, then this will automatically
/// resample the input stream to match the output stream. If the sample rates
/// match, then no resampling will occur.
///
/// Internally this uses the `ringbuf` crate.
pub struct ResamplingCons<T: Sample> {
cons: ringbuf::HeapCons<T>,
resampler: Option<RtResampler<T>>,
num_channels: NonZeroUsize,
latency_frames: usize,
is_waiting_for_frames: bool,
latency_seconds: f64,
in_sample_rate: f64,
in_sample_rate_recip: f64,
reset_flag: Arc<AtomicBool>,
}
impl<T: Sample> ResamplingCons<T> {
/// The number of channels configured for this stream.
pub fn num_channels(&self) -> NonZeroUsize {
self.num_channels
}
/// Returns `true` if resampling is occurring, `false` if the input and output
/// sample rates match.
pub fn is_resampling(&self) -> bool {
self.resampler.is_some()
}
/// Get the delay of the internal resampler, reported as a number of output
/// frames.
///
/// If no resampler is active, then this will return `0`.
pub fn output_delay(&self) -> usize {
self.resampler
.as_ref()
.map(|r| r.output_delay())
.unwrap_or(0)
}
/// The number of frames that are currently available to read from the buffer.
pub fn available_frames(&self) -> usize {
self.cons.occupied_len() / self.num_channels.get()
}
/// An number describing the current amount of jitter in seconds between the
/// input and output streams. A value of `0.0` means the two channels are
/// perfectly synced, a value less than `0.0` means the input channel is
/// slower than the input channel, and a value greater than `0.0` means the
/// input channel is faster than the output channel.
///
/// This value can be used to correct for jitter and avoid underflows/
/// overflows. For example, if this value goes above a certain threshold,
/// then you can read an extra packet of data or call
/// [`ResamplingCons::discard_frames`] or [`ResamplingCons::discard_jitter`]
/// to correct for the jitter.
///
/// This number will be in the range `[-latency_seconds, capacity_seconds - latency_seconds]`,
/// where `latency_seconds` and `capacity_seconds` are the values passed in
/// [`ResamplingChannelConfig`] when this channel was constructed.
///
/// Note, it is typical for the jitter value to be around plus or minus
/// `out_max_block_frames / out_sample_rate` or `data_frames / in_sample_rate`
/// (whichever is higher) even when the streams are perfectly in sync
/// (`data_frames` being the typical length in frames of a packet of data pushed
/// to [`ResamplingProd::push`]).
pub fn jitter_seconds(&self) -> f64 {
(self.available_frames() as f64 * self.in_sample_rate_recip) - self.latency_seconds
}
/// Clear all queued frames in the buffer.
pub fn reset(&mut self) {
if let Some(resampler) = &mut self.resampler {
resampler.reset();
}
self.cons.clear();
self.is_waiting_for_frames = true;
}
/// Discard a certian number of input frames from the buffer. This can be used to
/// correct for jitter and avoid overflows.
///
/// This will discard `frames.min(self.available_frames())` frames.
///
/// If `frames` is `None`, then the amount of frames to return the jitter count
/// to `0.0` will be discarded.
///
/// Returns the number of input frames that were discarded.
pub fn discard_frames(&mut self, frames: usize) -> usize {
self.cons
.skip(frames.min(self.available_frames()) * self.num_channels.get())
/ self.num_channels.get()
}
/// If the value of [`ResamplingCons::jitter_seconds`] is greater than the
/// given threshold in seconds, then discard the number of frames needed to
/// bring the jitter value back to `0.0` to avoid overflows.
///
/// Note, it is typical for the jitter value to be around plus or minus
/// `out_max_block_frames / out_sample_rate` or `data_frames / in_sample_rate`
/// (whichever is higher) even when the streams are perfectly in sync
/// (`data_frames` being the typical length in frames of a packet of data pushed
/// to [`ResamplingProd::push`]).
///
/// Returns the number of input frames that were discarded.
pub fn discard_jitter(&mut self, threshold_seconds: f64) -> usize {
assert!(threshold_seconds >= 0.0);
let jitter_secs = self.jitter_seconds();
if jitter_secs > threshold_seconds.max(0.0) {
let frames = (jitter_secs * self.in_sample_rate).round() as usize;
self.discard_frames(frames)
} else {
0
}
}
/// Read from the channel and store the results into the output buffer
/// in interleaved format.
pub fn read(&mut self, output: &mut [T]) -> ReadStatus {
let num_channels = self.num_channels.get();
let out_frames = output.len() / num_channels;
if self.reset_flag.swap(false, Ordering::Relaxed) {
self.reset();
}
if self.is_waiting_for_frames {
if self.available_frames() >= self.latency_frames {
self.is_waiting_for_frames = false;
} else {
return ReadStatus::WaitingForFrames;
}
}
let mut status = ReadStatus::Ok;
if let Some(resampler) = &mut self.resampler {
resampler.process_interleaved(
|in_buf| {
// Completely fill the buffer with new data.
// If the requested number of samples cannot be appended (i.e.
// an underflow occured), then fill the rest with zeros.
let samples = self.cons.pop_slice(in_buf);
if samples < in_buf.len() {
status = ReadStatus::Underflow;
in_buf[samples..].fill(T::zero());
self.is_waiting_for_frames = true;
}
},
&mut output[..out_frames * num_channels],
);
} else {
// Simply copy the input stream to the output.
let samples = self
.cons
.pop_slice(&mut output[..out_frames * num_channels]);
if samples < output.len() {
status = ReadStatus::Underflow;
output[samples..].fill(T::zero());
self.is_waiting_for_frames = true;
}
}
status
}
}
/// The status of reading data from [`ResamplingCons::read`].
#[derive(Debug, Clone, Copy, PartialEq, Eq)]
pub enum ReadStatus {
/// The buffer was fully filled with samples from the input
/// stream.
Ok,
/// An input underflow occured. This may result in audible audio
/// glitches.
Underflow,
/// The channel is waiting for a certain number of frames to be
/// filled in the buffer before continuing after an underflow
/// or a reset. The output will contain silence.
WaitingForFrames,
}