CircuitPython's audio system for MicroPython and CPython — audiocore,
audiomixer, synthio (including MidiTrack), the effects modules
(audiofilters, audiodelays, audiofreeverb, audiospeed), audiomp3,
and CircuitPython play()/stop()/pause()/resume() output semantics.
Import names stay audiocore/synthio/etc., matching CircuitPython for
source compatibility.
MicroPython consumes this repository as USER_C_MODULES, and it is
fully standalone for both build flavors — no other repository is required.
./scripts/fetch_deps.sh puts the two pinned native dependencies (ulab,
mp3) under .deps/, and both micropython.mk and micropython.cmake
look there first, falling back to a sibling checkout beside this repo and
otherwise failing the build with an error naming both paths.
For a CMake port (esp32, rp2), point USER_C_MODULES straight at this
checkout:
git clone https://github.com/PyDevices/audioif ~/build/audioif
cd ~/build/audioif && ./scripts/fetch_deps.sh
idf.py build -DUSER_C_MODULES=~/build/audioifFor a Make port (unix, windows, webassembly), MicroPython's own build glob
looks one level down, so point USER_C_MODULES at this checkout's
parent directory instead:
git clone https://github.com/PyDevices/audioif ~/build/audioif
cd ~/build/audioif && ./scripts/fetch_deps.sh
git clone https://github.com/micropython/micropython ~/build/micropython
cd ~/build/micropython/mpy-cross && make
cd ~/build/micropython/ports/unix && make submodules
make USER_C_MODULES=~/buildThe two dependencies are ulab (so synthtools's import ulab.numpy
works) and mp3 (the audiomp3 tier's decoder), pinned by
DEPENDENCIES.lock. To build without them, skip the
fetch and set AUDIOIF_OPTIONAL_DEPS=1 on the build step instead — the
variable is read by micropython.mk/micropython.cmake, not by
fetch_deps.sh, and on CMake ports it is read as an environment
variable, not as a -D cache entry:
make USER_C_MODULES=~/build AUDIOIF_OPTIONAL_DEPS=1 # Make ports
AUDIOIF_OPTIONAL_DEPS=1 idf.py build -DUSER_C_MODULES=~/build/audioif # CMakeThat builds every module except audiomp3 with no clone beyond this
repository; synthtools' import ulab.numpy then fails at runtime, which
is the trade the flag buys. See docs/porting-plan.md
for the architecture, module tiers, phased plan, and testing strategy.
CPython 3.10+ installs from TestPyPI:
python -m pip install --index-url https://test.pypi.org/simple/ pydevices-audioifThis gets you audiocore, synthio, audiomixer, audiofilters,
audiodelays, audiofreeverb, audiospeed, audiodynamics, audioroute,
audiomath, audioecho, audioshaper, audioladder, audioconvolve,
audiobiquad, audioverb, and the
audiorender package.
audiomp3 remains firmware-only. The distribution declares no required
runtime dependencies and does not itself publish an audioif import; its
version is the VERSION file, which is also what _audioif.__version__
reports.
audiorender is the exception, and it is opt-in: it is numpy throughout,
so numpy comes with the render extra rather than with the wheel, keeping
the core dependency-free for boards and wasm. TestPyPI carries no usable
numpy, so the extra needs PyPI as a second index:
python -m pip install --index-url https://test.pypi.org/simple/ \
--extra-index-url https://pypi.org/simple/ "pydevices-audioif[render]"The instrument and effect libraries — audioinstruments (53 synthesizers,
keyboards and drum machines) and audioeffects (46 effect classes, racks
included) — are not part of this distribution. They live in the
audiocomponents repository
as their own packages, each depending on pydevices-audioif; see that
repository for how to install them.
One pure-Python tier sits on top of the CircuitPython-compatible core:
lib/audiorender/— renders a whole composition offline: tracks, tempo map, notes and automation in, a mixed stereo master and a level report out. This is the one part of the repository written for a desktop rather than a board — numpy throughout, a whole song in memory — so it ships in the wheel and is never frozen into firmware. It has its own README.
The instrument and effect libraries that used to sit beside it —
audioinstruments and audioeffects — now live in the
audiocomponents repository,
together with the audio component contract they implement: the metadata
manifest
(docs/audio-components.md)
and the runtime API
(docs/audio-component-api.md).
audiorender drives any component that speaks that API and does not need
those packages installed.
Ten things here are not CircuitPython's. audiodynamics (compression,
limiting, expansion, gating, transient shaping) and audioroute.Splitter
(fan one stream out to parallel branches) come from micropython-vst3's audio
engine, which had them and CircuitPython does not. audiomath (multiply one
stream by another — ring and amplitude modulation; and divide one down in
frequency — the analog octave divider), audioecho (a delay with a filter,
a soft-clip and a cross-feed inside its feedback loop), audioshaper (a
waveshaper whose curve is data, applied above the sample rate),
audioladder (a transistor ladder filter — four one-pole stages round a
feedback loop with an odd saturator inside it), audioconvolve (apply a
measured or synthesized impulse response, by partitioned FFT), audioverb
(a reverberation tank whose line lengths and output taps come from Python),
audioroute.MidSide (scale the difference between a stereo pair's channels)
and audiobiquad (below) have no ancestor anywhere and are audioif's own.
apply_cp_patches.sh adds every one of them to a CircuitPython tree too.
audiofilters.Filter (over synthio.Biquad) and audiofilters.Phaser are
ported CircuitPython and their recursions are integer. Both have states that
reproduce themselves: fed silence after a burst, they can hold a constant of
one to a few LSB for ever. audiobiquad is the same two filters with
float state and a flush of anything below 1e-20, so a tail decays to zero
and stays there — and its all-pass feedback is not clamped to 0.1..0.9, so
zero is zero and a feedback-free phaser's notches are true nulls.
import audiobiquad, synthio
eq = audiobiquad.Biquad(mode=audiobiquad.PEAKING_EQ, frequency=3200,
Q=1.2, gain_db=-4.0, sample_rate=48000)
eq.play(source)
sweep = synthio.LFO(rate=0.4, scale=400.0, offset=900.0)
phase = audiobiquad.AllPass(stages=4, frequency=sweep, feedback=0.0,
mix=0.5, sample_rate=48000)
phase.play(eq)Biquad |
AllPass |
|
|---|---|---|
| constructor | mode, frequency, Q, gain_db, mix, sample_rate, channel_count |
stages, frequency, feedback, mix, sample_rate, channel_count |
| block inputs | frequency, Q, gain_db, mix |
frequency, feedback, mix |
| fixed at construction | sample_rate, channel_count |
those, plus stages (it sizes the state) |
| methods | play, stop, clear |
play, stop, clear |
| read-only | playing, coefficients |
playing, stages, coefficient |
| latency | 0 samples | 0 samples |
mode is one of LOW_PASS, HIGH_PASS, BAND_PASS, NOTCH,
PEAKING_EQ, LOW_SHELF, HIGH_SHELF — numbered exactly as
synthio.FilterMode numbers them, so either can be handed to either.
mix crossfades: 0 is a wire, 1 is the filtered signal alone; for an
all-pass cascade 0.5 is the equal sum a script phaser makes, and where its
notches are deepest. AllPass's frequency really is the frequency at
which one stage's phase passes −90°, so nothing has to pre-warp it. Both
nodes hand out 256 frames at a time, sit in an audiosample chain like any
other effect, and never report themselves finished — a starved chain gets
silence. Q is bounded to 0.05..60 and feedback to ±0.99: a pole on the
unit circle is a filter that never stops ringing, which is the one thing
this module exists to avoid. See
docs/upstream-diff.md for the measurements.
audiodynamics.Dynamics carries twenty-one options beyond what the engine
gave it, added for the effects program and every one of them default-off: an
RMS detector, a feedback detector topology and an external key input with Key
Listen, a 4x true-peak reconstruction, a two-pole key band with both corners
settable, a settable expander/gate depth, a four-stage gate envelope with
hold and hysteresis, a relative-threshold gain computer, a program-dependent
attack, and the transient shaper's four detector time constants, a second
envelope pair and a peak-hold on the slow one. What each is for, what it was
measured doing, and what it cost:
docs/upstream-diff.md.
audioecho.FeedbackDelay takes four further options, each off at its
default. wow_shape swaps the built-in modulation sine for one period of
your own — int16 Q15, a power-of-two count from 2 to 4096 — because a
bucket brigade's delay is its line length over its clock, so a triangle on
the clock is a reciprocal on the delay and no sine is that. delay_slew
walks the read head to a new delay_ms at a constant rate (delay-seconds
per second) instead of jumping to it, which is both an Echoplex's varispeed
and the reason a delay-time knob can be turned mid-take without a click.
wow_am_depth puts the same oscillator on the wet gain, dipping only.
loop_semitones pitch-shifts the line read inside the loop, so every
repeat rises again — a shimmer, which chaining audiodelays.PitchShift
after a delay is not. See
docs/upstream-diff.md for what each was asked for
and what it measures.
audioroute.MidSide(source, width=1.0, sample_rate=48000, channel_count=2)
takes a stereo pair apart into its mono sum and the difference between its
channels, scales the difference, and puts the pair back together. width=0
collapses to mono, 1 passes through, 2 doubles the sides; values outside
0..2 clamp to those rails. play(sample) sets the source, set(width=...)
moves the width mid-stream. There is no state and no latency — and at
width=1 the output bytes are the input bytes, exactly, for every int16 pair,
so the node costs nothing to leave in a chain that is not using it. A mono
source passes through: there is no difference to scale. It exists for the
drive classes as much as for stereo width: a nonlinearity applied to a stereo
pair intermodulates its channels, so a saturator that wants to keep its image
drives the mid and leaves the side alone.
audioshaper.Waveshaper(sample_rate=…, curve=…, oversample=4, channel_count=2, **options) is the newest of them, and everything but the
first four moves in set(). curve is int16 Q15, at least two points,
spanning −1..+1 of input: compute it once on a desktop and ship it as data,
never rebuild it on a board, whose float is single-precision where the
desktop's is double. oversample is 1, 2, 4 or 8 — the shaping happens there,
between a matched pair of polyphase all-pass half-bands, because a
nonlinearity at the base rate folds the harmonics it makes above Nyquist
straight back onto the signal. pre_gain is the drive knob (gain into one
normalised curve, never a curve rebuilt per knob move), bias moves the
operating point, post_gain follows the curve, and mix is a straight 0..1
crossfade. hysteresis is off by default; above zero it gives the curve a
memory, so a slow signal in and out traces two paths and encloses an area,
with hysteresis_width its half-width as a fraction of full scale at the
input and hysteresis_bias splitting that between the rising and falling
branches. audioshaper.GROUP_DELAY_SAMPLES carries the measured group delay
per factor — 0, 2.2, 3.3 and 3.9 base samples — for a component that has to
report its latency. Why each of those is the shape it is, and what the
oversampling measures:
docs/upstream-diff.md, "audioshaper: audioif's own,
and the two things a fixed curve cannot be".
audioladder is the newest, and the one whose reason for existing is least
obvious next to a module CircuitPython already has. audiofilters.Filter
is a better resonant low-pass than this: a cascade of biquads tracks the
analytic response to a fraction of a decibel. It is also linear, so it never
sustains a tone of its own, has no harmonics to speak of, and sounds exactly
the same however hard you hit it — and those three are what a ladder is. The
whole surface:
import audioladder
ladder = audioladder.Ladder(
sample_rate=48000, # and channel_count=1 or 2
cutoff_hz=800.0, # where the filter turns over
resonance=3.6, # 0..4.2; at 4 it sustains a tone at the cutoff
drive=2.5, # linear gain into the loop, 1 being unity
poles=4, # tap stage 1..4: 6, 12, 18 or 24 dB an octave
passband_comp=0.0, # 0..1, how much of the passband to give back
oversample=2, # 1 or 2; 2 folds back fewer of its harmonics
mix=1.0, # 0 a wire, 1 the filter
)
ladder.play(source)
ladder.set(cutoff_hz=1200.0) # any option, mid-stream, once a block
ladder.clear() # empty the integrators; stops a sustained toneSee docs/upstream-diff.md for why its feedback loop is solved rather than delayed, and what that was measured to be worth.
Dattorro's plate reverberator, with the network handed in rather than compiled
in — which is the difference from audiofreeverb.Freeverb, whose comb and
all-pass lengths are constants in a CircuitPython-ported kernel.
plate = audioverb.Tank(
sample_rate=48000, channel_count=2, max_predelay_ms=200,
decay=0.7, diffusion=0.75, bandwidth_hz=9000, damping_hz=4500,
mod_rate_hz=1.0, mod_depth_ms=0.27, mix=0.35)
plate.play(source)
audio_out.play(plate)Construction fixes the allocation and the topology: sample_rate,
channel_count (1 or 2), max_predelay_ms, delays and taps. delays is
twelve line lengths in frames — the four input diffusers, then each tank
half's modulated all-pass, delay, all-pass and delay. taps is four values
per tap (channel, line index, offset in frames, gain), at most 32. Both
default to Dattorro's published table, scaled from 29761 Hz to sample_rate.
set() moves the rest, mid-stream and without emptying the lines: decay,
diffusion, damping_hz, bandwidth_hz, low_cut_hz, predelay_ms,
mod_depth_ms, mod_rate_hz, drive, width, tone_db, mix. Every
filter is out of the path at zero, so a bare Tank() is that network with
nothing added to it; mix follows audiodelays.Echo's 0..2 convention, dry
at unity until 1. clear() empties every line and every filter. Latency is
zero dry-to-wet, and the tail reaches exact zero rather than sitting at one
LSB — every line write is a magnitude truncation, which is the standard cure
for a recirculating fixed-point network's limit cycles. See
docs/upstream-diff.md for the measurement, the cost
and the memory the default table needs.
MicroPython: all module tiers ported and oracle-diffed byte-for-byte
against bin/circuitpython on unix; DSP parity re-verified on windows and
wasm; built and measured on two real mcu targets, ESP32-P4
(hardware-confirmed by ear) and RP2040 (build-only). See
docs/porting-plan.md for the full phased history
and docs/upstream-diff.md for every deliberate
deviation from upstream CircuitPython.
CPython: the public surface and wheel plumbing are present, and the committed synthesis, mixer, MIDI, streaming, and effects fixtures match CircuitPython 10.2.1 PCM byte-for-byte. Import/API smoke success is not used as a substitute for those oracle comparisons.
ulab and mp3 (the vendored Adafruit_MP3/Helix decoder audiomp3
depends on, RPSL/RCSL-licensed — not MIT, carried unmodified per upstream's
own terms) are consumed as cloned sibling dependencies in the parent
workspace, same pattern as pygraphics/displayif, not vendored into this
repo.
The playback-facing pull protocol (audiocore.get_buffer/reset_buffer)
is consumed by pydevices' lib/audiodev package (AudioOut in
sample_out.py), which pumps any audiosample into any of audiodev's
existing push-PCM transports (sdl2/win/wasm/i2s/emulated) — see that
package's own docs for the playback-side contract this repo's protocol
implementation is built to satisfy.
Acceptance target: todbot's
synthtools running
unmodified, with rendered PCM diffed against bin/circuitpython's own unix
coverage build (which already contains the entire DSP stack — the parity
oracle for this whole port).
The API is our contract with you: class names, signatures, metadata, and macro surfaces stay stable and change only deliberately. The sound is not part of that contract. These components sound great, but they are not all as accurate as they could be, and implementations will keep being refined as the library matures — so a component may render audibly differently from one release to the next. If a composition depends on the exact sound of a release, pin that release rather than tracking the latest; the code of every release stays available for exactly this reason.
Beneath the components sits a harder guarantee: the audioif core — the
CircuitPython-compatible synthio/audiocore/effects-module layer — is
held bit-exact to CircuitPython itself, verified by parity gates, and
that never changes release to release. Where we find CircuitPython and
audioif disagree, we treat it as a bug and report it upstream. The
components are where the sound evolves; the floor they stand on does
not.
audioif is MIT licensed — see LICENSE.
The attribution that goes with it lives in NOTICE: substantial
portions of this code are ported from CircuitPython under its own MIT
license, CircuitPython is a trademark of Adafruit Industries and this
project is not affiliated with or endorsed by them, and the vendored test
corpus under tests/vendor/ carries its own licenses. The two are separate
files on purpose — GitHub and PyPI only detect the license when LICENSE holds
the MIT text and nothing else, so please do not fold NOTICE back into it.
NOTICE is in setuptools' default license-files glob, so it ships in the
wheel and sdist alongside LICENSE.
The Helix MP3 decoder that audiomp3 wraps is RPSL/RCSL-licensed, not MIT;
it is a cloned sibling dependency, not part of this repository. See
docs/upstream-diff.md.