Rubato

Rubato is a flexible audio sample rate conversion library for Rust, providing a choice of resamplers that can be optimized for either high quality or high speed. It processes audio in chunks, making it suitable for everything from real-time audio streams to offline batch processing.

The library allows for completely free selection of resampling ratios, which can even be updated on the fly. It features several resampler implementations, including high-quality asynchronous sinc resamplers and fast synchronous FFT-based resamplers.

Rubato is designed with real-time safety in mind, avoiding allocations during processing to ensure smooth and predictable performance. See Real-time considerations for more details.

Input and output data format

Input and output data is handled via Adapter and AdapterMut objects from the audioadapter crate. By using a suitable adapter, any sample layout and format can be used.

The audioadapter-buffers crate provides a selection of adapters for common data structures, and the audioadapter traits are kept simple in order to make it easy to implement them for new structures if needed.

For projects migrating from a previous version of rubato, the SequentialSliceOfVecs adapter is a good starting point, since it wraps the vector of vectors commonly used with rubato v0.16 and earlier.

Asynchronous resampling

Asynchronous resampling is when the input and output sample rates are not locked, and the ratio may vary slightly over time. This is common in real-time audio streams where the input and output devices have different clocks that may drift relative to each other. It allows for changing the resampling ratio at any time to compensate for this drift.

The asynchronous resamplers are available with and without anti-aliasing filters.

Resampling with anti-aliasing is based on band-limited interpolation using sinc interpolation filters. The sinc interpolation upsamples by an adjustable factor, and then the new sample points are calculated by interpolating between these points. The resampling ratio can be updated at any time.

Resampling without anti-aliasing omits the CPU-heavy sinc interpolation. This runs much faster but produces a lower quality result.

Synchronous resampling

Synchronous resampling is the case when the input and output sample rates are fixed and locked to each other. For example, converting a file from 44.1 kHz to 48 kHz. The ratio, 48 kHz / 44.1 kHz (equivalent to 160 / 147), is fixed and constant throughout the process.

Synchronous resampling is implemented via FFT (Fast Fourier Transform). The audio data is transformed into the frequency domain, the spectrum is scaled to match the target sample rate, and then transformed back to produce the resampled output. This type of resampler is considerably faster than sinc-based approaches but doesn't support changing the resampling ratio.

Choosing a resampler

Rubato provides two true resampler types. The synchronous Fft resampler is for a fixed ratio, while the asynchronous Async resampler allows the ratio to vary over time and comes in two interpolation modes. Which one to pick depends mainly on whether the ratio is fixed, and on how you want to trade CPU time for quality. In addition, the lightweight Slip clutch handles the special case of matching two rates that are almost identical.

Some common tasks and a suitable resampler for each:

• Convert an audio file from one fixed rate to another, for example 44.1 to 48 kHz as an offline batch job: use Fft with process_all().
• Resample a real-time stream between two devices whose clocks drift, such as a capture card feeding a sound card: use Async::new_sinc for the same high quality as Fft, and adjust the ratio to track the drift.
• Do the same on a tight CPU budget where a little aliasing is acceptable, for example in a game, a voice application such as a video call, or on an embedded target: use Async::new_poly.
• Change playback speed or pitch on the fly by sweeping the ratio while running: use an Async resampler, sinc for quality or polynomial for speed. The ratio is not limited to small adjustments; it can be swept over a wide range, up to the max_resample_ratio_relative factor chosen at construction.
• Keep a stream aligned to a sound card that is a few ppm off, without truly resampling: use Slip.
• Feed a downstream stage where quality barely matters, such as driving a VU meter or level display, or downsampling to a low rate to cut the cost of later processing: use Async::new_poly for the cheapest conversion.

A few notes to guide the choice:

• The Fft resampler requires the fft_resampler feature, which is enabled by default. It is both fast and high quality, so it is the natural choice whenever the ratio is fixed.
• Use an Async resampler when the input and output clocks may drift relative to each other, or when you otherwise need to adjust the ratio while running (see Asynchronous resampling).
• Async::new_sinc matches the quality of the Fft resampler but is the most CPU-heavy option. Async::new_poly is much faster because it skips the sinc anti-aliasing filter; the quality loss is often subtle. See Resampling quality for the settings that tune each mode.
• Slip is not a true sample rate converter. It does not interpolate or filter; it just passes samples through and occasionally inserts or drops a single frame to keep two near-identical clocks in step, hiding each correction behind a short crossfade. In return it uses very little CPU, adds no delay, and leaves the passband untouched. Use it only for tracking tiny clock drift (a few ppm, ratio very close to 1.0); for any real ratio change, use an Async resampler instead.

If you are unsure, Fft is a good default for offline or fixed-rate work, and Async::new_sinc is a good default for real-time streams with clock drift.

Usage

The resamplers provided by this library are intended to support processing streams of audio. To enable this, they process audio in chunks. The optimal chunk size is determined by the application, but will likely end up somewhere between a few hundred to a few thousand frames. This gives a good compromise between efficiency and memory usage.

Chunk size and fixed size options

Rubato processes audio in chunks. The chunk_size parameter given to the resampler constructor sets the target size for the fixed side — the side that always has the same number of frames per call. The other side is variable and will differ from call to call.

The resamplers allow specifying which side should have a fixed size.

• Fixed input (FixedAsync::Input / FixedSync::Input): The input chunk size is fixed to chunk_size frames per call. The output chunk size varies depending on how many samples can be calculated from the available input data. This is convenient when the audio source delivers data in fixed-size chunks (e.g. a hardware capture callback).

• Fixed output (FixedAsync::Output / FixedSync::Output): The output chunk size is fixed to chunk_size frames per call. The input chunk size varies depending on how many new samples the resampler needs to fill the output. This is useful when the audio destination consumes fixed-size chunks (e.g. a hardware playback callback).

• Both input and output fixed (FixedSync::Both): Both input and output chunk sizes are fixed. This mode is only available for the synchronous resampler. In this mode, the chunk_size parameter is used as a hint, and the actual chunk sizes are calculated to fit the resampling ratio exactly. For example, a 44.1 kHz to 48 kHz resampler must use an input chunk size that is a multiple of 147, and an output chunk size that is a multiple of 160, in order to maintain the correct resampling ratio.

  This mode avoids some internal buffering compared to fixed input or fixed output modes, and is therefore somewhat more efficient.

  For asynchronous resamplers, fixing both input and output chunk sizes is not possible since the resampling ratio can change, requiring at least one side to be variable.

Input and output sizes per call

The chunk_size constructor parameter is only the target size for the fixed side. Always call input_frames_next() before providing data to process_into_buffer to find out the exact number of input frames required for that call. Similarly, call output_frames_next() to find the exact number of output frames that will be written.

• With fixed input, input_frames_next() always returns chunk_size. output_frames_next() varies and must be checked each call.
• With fixed output, output_frames_next() always returns chunk_size. input_frames_next() varies and must be checked each call.
• With FixedSync::Both, both values are fixed, but they may differ from chunk_size because they are rounded to fit the exact sample-rate ratio.

The input and output buffers must be large enough to hold at least the number of frames reported by input_frames_next() and output_frames_next() respectively. Both buffers may be larger than required — only the needed frames are read or written.

Resampling quality

The synchronous resampler has no quality settings; it always delivers the best quality.

When using cubic sinc interpolation, the quality of the asynchronous resampler is equivalent to the synchronous resampler. This mode is however computationally heavy, and therefore there are some settings that can be used when a different balance between speed and quality is required.

When using sinc interpolation, the length of the sinc function can be reduced. Each halving of the sinc function length nearly doubles the speed, at the cost of increased roll-off at high frequencies. It is also possible to lower the polynomial degree to quadratic or linear, which also increases the speed while producing higher amounts of distortion.

The fastest option is to use plain polynomial interpolation. This is significantly faster as it avoids the expensive sinc interpolation, but does not provide any anti-alias filtering. The effect of this is often subtle, and many applications can use this mode to save a significant amount of CPU time with little or no perceived quality loss.

Real-time considerations

Rubato is suitable for real-time applications when using the Resampler::process_into_buffer() method. This stores the output in a pre-allocated output buffer, and performs no allocations or other operations that may block the thread. Ensure that the resampler instance and any needed input and output buffers are created before entering time-sensitive parts of the application.

The log feature is disabled by default, and should not be enabled for real-time use.

Resampling a given audio clip

To resample a full audio clip that is already in memory, use either Resampler::process_all() or Resampler::process_all_into_buffer(). Both process the clip in suitably sized chunks internally and trim off the startup delay, so the result lines up with the input. Create a resampler of a suitable type, for example Fft which is fast and gives good quality. The chunk size can be chosen arbitrarily. Start with a chunk size of for example 1024. In this application, the exact input or output chunk sizes are not important and therefore the FixedSync::Both setting can be used for the Fft resampler.

• process_all() is the simplest option. It allocates the output, trims the delay, and returns an InterleavedOwned holding exactly the resampled frames.
• process_all_into_buffer() writes into a buffer you provide, avoiding the allocation. Call Resampler::process_all_needed_output_len() first to size the output buffer.

Do not create a resampler with the chunk size set to the whole clip length and call Resampler::process() once. process() is meant for a single chunk and does not trim the startup delay, so the output would start with silence and be missing the tail. Use process_all() for whole clips instead.

If there is more than one clip to resample from and to the same sample rates, the same resampler should be reused. Creating a new resampler is an expensive task and should be avoided when possible. process_all() resets the resampler for you; with process_all_into_buffer(), call Resampler::reset() between clips to prepare it for a new job.

Resampling a stream

When resampling a stream, the process is normally performed in real time, and either the input or output is some API that provides or consumes frames at a given rate.

Use case example, record from an audio API and save to a file

Audio APIs such as CoreAudio on MacOS, or the cross platform cpal crate, often use callback functions for data exchange.

Callback function

When capturing audio from these, the application passes a function to the audio API. The API then calls this function periodically, with a pointer to a data buffer containing new audio frames. The data buffer size is usually the same on every call, but that varies between APIs. It is important that the function does not block, since this would block some internal loop of the API and cause loss of some audio data. It is also recommended to keep the callback function light. No heavy processing such as resampling should be performed here. Ideally it should read the provided audio data from the buffer provided by the API, and optionally perform some light processing such as sample format conversion. It should then store the audio data to a shared buffer. The buffer may be a Arc<Mutex<VecDeque<T>>>, or something more advanced such as ringbuf.

Processing loop

A separate loop, running either in the main or a separate thread, should then read from that buffer, resample, and save to file. The resampler loop needs to wait for the needed number of frames to become available in the buffer, before reading and passing them to the resampler.

If the Audio API provides a fixed buffer size, then this number of frames is a good choice for the resampler input chunk size. If the size varies, the shared buffer can be used to adapt the chunk sizes of the audio API and the resampler. A good starting point for the resampler chunk size is to use an "easy" value, for example a power of two, near the average chunk size of the audio API. Make sure that the shared buffer is large enough to not get full in case the loop gets blocked waiting, for example for disk access.

The output of the resampler is then written to a file. The hound crate is a popular choice for reading and writing uncompressed audio formats.

SIMD acceleration

Asynchronous resampling with anti-aliasing

The asynchronous sinc resampler supports SIMD on x86_64 and on aarch64 (64-bit Arm). The SIMD capabilities of the CPU are determined at runtime. If no supported SIMD instruction set is available, it falls back to a scalar implementation.

On x86_64, it will try to use AVX. If AVX isn't available, it will instead try SSE3.

On aarch64, it will use Neon if available.

Synchronous resampling

The synchronous FFT resampler benefits from the SIMD support of the RustFFT library.

Cargo features

fft_resampler: Enable the FFT based synchronous resampler

This feature is enabled by default. Disable it if the FFT resampler is not needed, to save compile time and reduce the resulting binary size.

log: Enable logging

This feature enables logging via the log crate. This is intended for debugging purposes, when working on rubato itself or investigating issues. Applications using this library should normally keep this feature disabled to avoid cluttering logs with unnecessary messages.

Note that outputting a log message allocates a String, and most logging implementations involve various other system calls. These calls may take some (unpredictable) time to return, during which the application is blocked. This means that logging should be avoided if using this library in a realtime application.

The log feature can be enabled when running tests, which can be very useful when debugging. The logging level can be set via the RUST_LOG environment variable.

Example:

RUST_LOG=trace cargo test --features log

Example

Resample stereo audio from 44100 to 48000 Hz, one chunk at a time. The input is processed in a loop and can come from anywhere: a file read chunk by chunk, or a live stream that keeps running indefinitely. This uses the Async resampler with polynomial interpolation, which is always available and needs no optional features. See also the "process_raw" example that can be used to process a file from disk.

use rubato::{
    Resampler, Async, FixedAsync, PolynomialDegree, Indexing
};
use audioadapter_buffers::direct::InterleavedSlice;

let channels = 2;
let chunk_size = 1024;

let mut resampler = Async::<f64>::new_poly(
    48000.0 / 44100.0,
    1.1,
    PolynomialDegree::Cubic,
    chunk_size,
    channels,
    FixedAsync::Input,
).unwrap();

// Reusable buffers for a single chunk, assuming interleaved f64 samples.
// With `FixedAsync::Input` every call consumes `chunk_size` input frames and
// produces at most `output_frames_max()` output frames.
let mut indata = vec![0.0; channels * chunk_size];
let mut outdata = vec![0.0; channels * resampler.output_frames_max()];
let outdata_capacity = outdata.len() / channels;

let indexing = Indexing::new();

// Keep processing for as long as there is more audio to handle.
// Here the source is a dummy counter that stops after a few chunks;
// in a real application this stands in for "is there more data?".
let mut chunks_left = 10;
loop {
    // Fetch the next `input_frames_next()` frames from the source into `indata`.
    // For a file, break out of the loop once the end is reached (a shorter final
    // chunk is handled by setting `partial_len` on the indexing struct, see the
    // `process_raw` example). For an endless stream, simply never break.
    if chunks_left == 0 {
        break;
    }
    chunks_left -= 1;
    let frames_to_read = resampler.input_frames_next();
    // (read `frames_to_read` frames from the file or stream into `indata` here)

    let input_adapter = InterleavedSlice::new(&indata, channels, frames_to_read).unwrap();
    let mut output_adapter =
        InterleavedSlice::new_mut(&mut outdata, channels, outdata_capacity).unwrap();

    let (_frames_read, frames_written) = resampler
        .process_into_buffer(&input_adapter, &mut output_adapter, Some(&indexing))
        .unwrap();

    // Write the `frames_written` output frames to the destination file or stream.
    let _ = frames_written;
}

Included examples

The examples directory contains a few sample applications for testing the resamplers. There are also Python scripts for generating simple test signals as well as analyzing the resampled results. Run any of them with --help for the full list of options.

• resample_wav reads and writes .wav files directly, and converts to any sample format supported by the waveadapter crate.
• process_raw converts between two fixed sample rates, using any of the resampler types. The resampling runs in f32 or f64, chosen with --precision, for comparing the two.
• adjust_ratio_f64 applies a small constant rate offset, the clock drift case.
• ramp_ratio_f64 ramps the ratio while processing.

The Python scripts make test signals in the raw 64-bit float format the examples read, and analyze the results. They need numpy, and analyze_result.py also needs matplotlib.

• make_sine.py writes a sine, taking the sample rate, channel count, length and frequency as arguments.
• make_tone_scale.py writes a scale of stepped pure sine tones, which makes the artefacts of a slowly adjusted ratio easy to hear. It also prints the ppm offset to give adjust_ratio_f64 for a chosen correction rate.
• make_multitone.py writes a comb of equally spaced tones, for looking at aliasing and intermodulation across the whole band at once. Same arguments as make_sine.py, with the tone grid set by --first, --last and --spacing.
• analyze_result.py plots the spectrum of a resampled file, taking the file name, channel count, sample rate and sample format as arguments.

Apart from resample_wav, the examples read and write raw audio data as 64-bit floats. They can be used to process .wav files if the files are first converted to the right format. Example, use sox to convert a .wav to 64-bit float raw samples:

sox some_file.wav -e floating-point -b 64 some_file_f64.raw

After processing with for instance the process_raw example, the result can be converted back to a new .wav. This command converts the 64-bit floats to 16-bits at 44.1 kHz:

sox -e floating-point -b 64 -r 44100 -c 2 resampler_output.raw -e signed-integer -b 16 some_file_resampled.wav

Many audio editors, for example Audacity, are also able to directly import and export the raw samples.

Compatibility

The rubato crate requires rustc version 1.87 or newer.

Migrating from 3.x to 4.0

Version 4.0 has a handful of breaking changes. The common ones, with before/after:

audioadapter 4.0. The Adapter and AdapterMut traits no longer carry a lifetime parameter. Update any explicit uses, and bump your own audioadapter / audioadapter-buffers dependencies to 4.0 to match the versions rubato re-exports.

// before
fn resample(buf: &dyn Adapter<'_, f32>) { /* ... */ }
// after
fn resample(buf: &dyn Adapter<f32>) { /* ... */ }

Resampler::process takes an Option<&Indexing> instead of separate input_offset and active_channels_mask arguments, matching process_into_buffer.

// before
let out = resampler.process(&input, 0, None)?;
// after
let out = resampler.process(&input, None)?;

// before: with an offset and mask
let out = resampler.process(&input, offset, Some(&mask))?;
// after
let indexing = Indexing::new().input_offset(offset).active_channels_mask(mask);
let out = resampler.process(&input, Some(&indexing))?;

SincInterpolationParameters::f_cutoff is now an Option<f32>. Prefer SincInterpolationParameters::new(sinc_len, window), which derives the cutoff automatically. When building the struct directly, use None for the automatic cutoff or wrap a manual value in Some.

// before
let params = SincInterpolationParameters {
    sinc_len: 256,
    f_cutoff: 0.95,
    oversampling_factor: 128,
    interpolation: SincInterpolationType::Cubic,
    window: WindowFunction::BlackmanHarris2,
};
// after (recommended): cutoff derived from sinc_len and window
let params = SincInterpolationParameters::new(256, WindowFunction::BlackmanHarris2);
// after (struct form): None for automatic, or Some(value) to override
let params = SincInterpolationParameters { f_cutoff: None, ..params };

Fft::new no longer takes sub_chunks. It picks a value automatically and uses a default window. For control over either, use the new Fft::new_custom.

// before
let r = Fft::<f64>::new(rate_in, rate_out, chunk_size, sub_chunks, channels, fixed)?;
// after
let r = Fft::<f64>::new(rate_in, rate_out, chunk_size, channels, fixed)?;
// or, with explicit sub_chunks and window:
let r = Fft::<f64>::new_custom(rate_in, rate_out, chunk_size, sub_chunks, channels,
    WindowFunction::BlackmanHarris2, fixed)?;

Ratio and chunk-size changes moved to capability traits. set_resample_ratio, set_resample_ratio_relative are now on the Adjustable trait, and set_chunk_size is on Resizable. On a concrete resampler, just bring the trait into scope. On a dyn Resampler, recover the capability with as_adjustable() / as_resizable() (this replaces the old SyncNotAdjustable / ChunkSizeNotAdjustable errors, which are removed). To query the capability through a shared &dyn Resampler, use is_adjustable() / is_resizable().

// before: on a Box<dyn Resampler>, with a runtime error for synchronous resamplers
resampler.set_resample_ratio(new_ratio, true)?;
// after: None means "synchronous, nothing to adjust"
if let Some(adjustable) = resampler.as_adjustable() {
    adjustable.set_resample_ratio(new_ratio, true)?;
}

// before: on a concrete Async resampler
async_resampler.set_resample_ratio_relative(0.95, true)?;
// after: same call, but the Adjustable trait must be in scope
use rubato::Adjustable;
async_resampler.set_resample_ratio_relative(0.95, true)?;

The error enums are now #[non_exhaustive]. If you match on ResampleError or ResamplerConstructionError, add a _ => ... arm.

Changelog

• Unreleased
  • Reject a non-finite resample_ratio or max_resample_ratio_relative when constructing an asynchronous resampler. Comparisons against NaN and infinity are false, so both slipped through the range checks and left the resampler reporting sizes it could not deliver.
  • Panic, instead of reporting a usize::MAX sized buffer, when the combination of chunk_size, resample_ratio and max_resample_ratio_relative gives a size that does not fit in a usize. The float to integer conversion saturates, so such a size used to look like an ordinary answer from input_frames_max, output_frames_max, input_frames_next and output_frames_next, and the internal buffer length could wrap to a value far too small.
  • Reject a relative ratio in Adjustable::set_resample_ratio_relative when multiplying it by the original ratio does not give a finite, positive result. The product could overflow to infinity while the relative ratio itself was inside the allowed range.
• v5.0.0
  • Fix an out-of-bounds panic in the asynchronous resamplers when the resampling ratio is changed by a large factor with ramp = true. The input and output size estimates now average the step sizes rather than the ratios, and correct for the ramp overshoot, so the estimate is never lower than the number of frames the processing loop actually consumes.
  • Speed up the polynomial interpolation in Async::new_poly by evaluating the polynomials in Horner form.
  • Update to audioadapter 5.0 and audioadapter-buffers 5.1. Bump your own audioadapter dependencies to match the versions rubato re-exports.
  • Raise the minimum supported rustc version to 1.87.
• v4.0.0
  • Update to audioadapter 4.0, which removes the lifetime parameter from the Adapter and AdapterMut traits.
  • Add a fluent way to construct an Indexing: Indexing::new plus chainable setters, and implement Default for Indexing.
  • Add a fluent way to construct SincInterpolationParameters: SincInterpolationParameters::new(sinc_len, window) plus chainable setters.
  • Change SincInterpolationParameters::f_cutoff to an Option<f32>. Leave it None (the default) to let the resampler derive the cutoff from sinc_len and window with calculate_cutoff, or set Some(value) to override it.
  • Implement Default for SincInterpolationParameters (sinc_len 256, automatic cutoff, oversampling_factor 128, Cubic interpolation, BlackmanHarris2 window).
  • Add Resampler::process_all, an allocating one-shot method for resampling a whole clip. It resets the resampler, trims the startup delay, and returns an InterleavedOwned holding exactly the resampled frames. This is the convenient counterpart to process_all_into_buffer.
  • Change Resampler::process to take an Option<&Indexing> instead of separate input_offset and active_channels_mask arguments, matching process_into_buffer. This adds partial_len support (for a short final chunk) and makes the common call process(&input, None). The Indexing output_offset field is ignored here.
  • Let the synchronous Fft resampler choose the anti-aliasing window. Fft::new is simplified (it drops sub_chunks, picking a value automatically, and uses a default window), and a new Fft::new_custom exposes both sub_chunks and the window function.
  • Derive Clone, Copy and PartialEq for ResampleError and ResamplerConstructionError, and mark both #[non_exhaustive].
  • Derive PartialEq, Eq and Hash for the configuration enums WindowFunction, SincInterpolationType, PolynomialDegree, FixedSync and FixedAsync.
  • Return WrongNumberOfMaskChannels instead of panicking when the active_channels_mask passed to a process method has the wrong length.
  • Split the capability-specific methods out of Resampler into the Adjustable trait (set_resample_ratio, set_resample_ratio_relative) and the Resizable trait (set_chunk_size). Resampler gains as_adjustable() and as_resizable() to recover these capabilities from a trait object, and is_adjustable() / is_resizable() to query them through a shared reference. The SyncNotAdjustable and ChunkSizeNotAdjustable error variants are removed, since calling these methods is now a compile-time capability.
  • Add Slip, a very cheap resampler for matching two almost-equal sample rates by occasionally slipping (inserting or dropping) a frame, hidden by a short crossfade, rather than running a full resampler. It is meant for compensating small clock differences: it adds no delay and no high-frequency roll-off, and its ratio is meant to be adjusted at runtime through Adjustable by a feedback loop.
• v3.0.0
  • Use separate lifetimes for buffer_in and buffer_out in process_into_buffer.
  • Improve sinc resampler performance with smarter dot product calculation.
  • Improve SIMD performance (AVX, SSE, NEON) using multiple accumulators.
  • Switch dot product strategy based on channel count.
  • More aggressive inlining of hot paths.
• v2.0.0
  • Update to audioadapter 3.0.
  • Add re-export of audioadapter-buffers.
• v1.0.1
  • Fix calculation in process_all_needed_output_len method.
• v1.0.0
  • New API using the AudioAdapter crate to handle different buffer layouts and sample formats.
  • Merged the FixedIn, FixedOut and FixedInOut resamplers into single types that supports all modes.
  • Merged the sinc and polynomial asynchronous resamplers into one type that supports both interpolation modes.
• v0.16.2
  • Fix issues when using on 32-bit systems.
• v0.16.1
  • Fix issue in test suite when building without FFT resamplers.
• v0.16.0
  • Add support for changing the fixed input or output size of the asynchronous resamplers.
• v0.15.0
  • Make FFT resamplers optional via fft_resampler feature.
  • Fix calculation of input and output sizes when creating FftFixedInOut resampler.
  • Fix panic when using very small chunksizes (less than 5).
• v0.14.1
  • More bugfixes for buffer allocation and max output length calculation.
  • Fix building with log feature.
• v0.14.0
  • Add argument to let input/output_buffer_allocate() optionally pre-fill buffers with zeros.
  • Add convenience methods for managing buffers.
  • Bugfixes for buffer allocation and max output length calculation.
• v0.13.0
  • Switch to slices of references for input and output data.
  • Add faster (lower quality) asynchronous resamplers.
  • Add a macro to help implement custom object safe resamplers.
  • Optional smooth ramping of ratio changes to avoid audible steps.
  • Add convenience methods for handling last frames in a stream.
  • Add resampler reset method.
  • Refactoring for a more logical structure.
  • Add helper function for calculating cutoff frequency.
  • Add quadratic interpolation for sinc resampler.
  • Add method to get the delay through a resampler as a number of output frames.
• v0.12.0
  • Always enable all simd acceleration (and remove the simd Cargo features).
• v0.11.0
  • New api to allow use in realtime applications.
  • Configurable adjust range of asynchronous resamplers.
• v0.10.1
  • Fix compiling with neon feature after changes in latest nightly.
• v0.10.0
  • Add an object-safe wrapper trait for Resampler.
• v0.9.0
  • Accept any AsRef<[T]> as input.

License

Licensed under either of

• Apache License, Version 2.0 (LICENSE-APACHE)
• MIT license (LICENSE-MIT)

at your option.