audio-nodes
DevelopmentC++ audio node engine for react-native-audio-api. Covers the AudioNode class hierarchy, the processNode() audio-thread contract (no allocs, no locks, no blocking I/O), AudioParam a-rate/k-rate processing, cross-thread communication patterns (CrossThreadEventScheduler, IAudioEventHandlerRegistry), and a step-by-step checklist for implementing a new node end-to-end. Use this skill when implementing a new Web Audio API node, modifying audio graph traversal or processing logic, or debugging audio rendering artifacts. Trigger phrases: "add a new node", "implement AudioNode", "processNode", "audio thread", "AudioParam automation".
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Skill: AudioNodes
Golden references: GainNode.h/.cpp (effect node), OscillatorNode.h/.cpp (scheduled source). Mirror their structure for any new node. See gainnode-example.md for an annotated header + .cpp.
If spec defaults or parameter ranges are unclear → fetch https://webaudio.github.io/web-audio-api/ before writing any constructor code.
Directory Structure
common/cpp/audioapi/core/
├── AudioNode.h / .cpp # Base class for all nodes
├── AudioParam.h / .cpp # Automatable parameter
├── BaseAudioContext.h / .cpp # Engine + node factory
├── AudioContext.h / .cpp # Real-time context
├── OfflineAudioContext.h / .cpp # Offline rendering context
├── sources/
│ ├── AudioScheduledSourceNode.h # Base for start/stop sources (INTERNAL)
│ ├── AudioBufferBaseSourceNode.h # Base for buffer playback (INTERNAL)
│ ├── OscillatorNode.h / .cpp
│ ├── AudioBufferSourceNode.h / .cpp
│ ├── AudioBufferQueueSourceNode.h / .cpp
│ ├── ConstantSourceNode.h / .cpp
│ ├── WorkletSourceNode.h / .cpp
│ └── RecorderAdapterNode.h / .cpp
├── effects/
│ ├── GainNode.h / .cpp
│ ├── BiquadFilterNode.h / .cpp
│ ├── DelayNode.h / .cpp
│ ├── IIRFilterNode.h / .cpp
│ ├── StereoPannerNode.h / .cpp
│ ├── WaveShaperNode.h / .cpp
│ ├── ConvolverNode.h / .cpp
│ ├── WorkletNode.h / .cpp
│ └── PeriodicWave.h / .cpp # Wave table (not a node)
├── analysis/
│ └── AnalyserNode.h / .cpp
├── destinations/
│ └── AudioDestinationNode.h / .cpp
├── inputs/
│ └── AudioRecorder.h / .cpp
└── utils/
└── AudioGraphManager.h / .cpp
The Audio Thread Contract
processNode() runs on the audio thread — the real-time rendering thread driven by the native audio driver (Oboe on Android, CoreAudio on iOS). This thread has strict requirements:
MUST NOT in processNode():
- Allocate or free memory (
new,delete,malloc,free,std::vector::push_backthat grows, etc.) - Acquire any mutex or lock (
std::mutex,std::lock_guard, etc.) - Make any blocking syscall (file I/O, socket,
sleep,wait) - Call into JavaScript — no JSI calls, no
callInvoker_->invokeSync() - Throw exceptions (or rely on exception unwinding paths that allocate)
Preallocate everything in the constructor:
// Constructor — JS thread, allocations OK
GainNode::GainNode(const std::shared_ptr<BaseAudioContext> &context, const GainOptions &options)
: AudioNode(context, options) {
// Preallocate the AudioBuffer used during processing
audioBuffer_ = std::make_shared<AudioBuffer>(channelCount_, context->getBufferSize());
// Preallocate params — they own their internal AudioBuffer too
gainParam_ = std::make_shared<AudioParam>(
options.gain, -3.4028234663852886e+38f, 3.4028234663852886e+38f, context);
}
// processNode — audio thread, NO allocations
std::shared_ptr<AudioBuffer> GainNode::processNode(
const std::shared_ptr<AudioBuffer> &processingBuffer,
int framesToProcess) {
// Already-allocated buffer reused each render quantum
auto gainValues = gainParam_->processARateParam(framesToProcess, time);
for (size_t i = 0; i < processingBuffer->getNumberOfChannels(); i++) {
processingBuffer->getChannel(i)->multiply(*gainValues->getChannel(0), framesToProcess);
}
return processingBuffer;
}
Processable State (reverse-topo pull)
Which nodes run each render quantum is decided by AudioGraph::settleProcessableState(), run inside Graph::process() after toposort/compaction and before the forward iter() pass. It is an audio-thread-only concern — never derived from HostGraph adjacency (that mutates on the JS thread under nodesMutex_).
GraphObject::PROCESSABLE_STATE:
ALWAYS_PROCESSABLE— seed / pull root. Set in ctors ofAudioDestinationNodeandAnalyserNode. Never reset by settle.CONDITIONAL_PROCESSABLE— on this quantum because something processable downstream pulls it. Recomputed from scratch every quantum.NOT_PROCESSABLE— idle / disconnected / default.
Settle algorithm (allocation-free):
- Reverse pull: walk the topo-sorted node array sinks → sources; for every
ALWAYS_/CONDITIONAL_PROCESSABLEnode, mark its inputs (and processable-links)CONDITIONAL_PROCESSABLE. Iterates to a fixpoint for processable-links. - End-of-quantum demotion: after
processInputs(), each node that wasCONDITIONAL_PROCESSABLEflips back toNOT_PROCESSABLEinGraphObject::process(). That replaces a global reset at the start of settle — nodes that ran last quantum are already idle when the next pull begins.
Key invariants:
- Pull from
processableState_, neverAudioNode::isProcessable(). A tail-bearing node (Delay/Convolver/Biquad) overridesisProcessable()to staytruewhile its tail drains after a disconnect; using that for the pull would wrongly re-activate its whole upstream cone. The tail node stays scheduled via that override; itsprocessableState_isNOT_PROCESSABLE, so it correctly does not pull upstream. disable()is sticky.AudioNode::disable()setsNOT_PROCESSABLEandexcludeFromProcessablePull_ = true, so a finished source still wired to a live consumer is not re-activated by the every-quantum pull. Sources calldisable()from the audio thread when playback finishes.- DelayReader → DelayWriter have no audio edge (they share a ring buffer).
Graph::linkNodes(reader, writer)records a processable-link, mirrored ontoAudioGraph::Node::link_head. Settle follows links so pulling the reader also pulls the writer and the writer's inputs. Links are NOT part of the topological sort (that would create a cycle for feedback delays).
Class Hierarchy
classDiagram
direction TD
class AudioScheduledSourceNode {
<<internal base>>
start(when)
stop(when)
}
class AudioBufferBaseSourceNode {
<<internal base>>
playbackRate AudioParam
detune AudioParam
}
AudioNode <|-- AudioScheduledSourceNode
AudioNode <|-- GainNode
AudioNode <|-- BiquadFilterNode
AudioNode <|-- DelayNode
AudioNode <|-- IIRFilterNode
AudioNode <|-- StereoPannerNode
AudioNode <|-- WaveShaperNode
AudioNode <|-- ConvolverNode
AudioNode <|-- WorkletNode
AudioNode <|-- AnalyserNode
AudioNode <|-- AudioDestinationNode
AudioNode <|-- AudioRecorder
AudioScheduledSourceNode <|-- AudioBufferBaseSourceNode
AudioScheduledSourceNode <|-- OscillatorNode
AudioScheduledSourceNode <|-- ConstantSourceNode
AudioScheduledSourceNode <|-- WorkletSourceNode
AudioBufferBaseSourceNode <|-- AudioBufferSourceNode
AudioBufferBaseSourceNode <|-- AudioBufferQueueSourceNode
AudioScheduledSourceNode (internal only — not exposed to JS directly)
Base class for source nodes that have a scheduled start and stop time. Not instantiated directly.
// Playback state machine
enum class PlaybackState {
UNSCHEDULED, // before start() called
SCHEDULED, // start() called, waiting for startTime_
PLAYING, // actively producing audio
STOP_SCHEDULED, // stop() called, waiting for stopTime_
FINISHED // done, node will be disabled
};
Subclasses call updatePlaybackInfo(currentTime, framesToProcess) at the top of processNode() to transition the state machine and handle sample-accurate start/stop.
When the node finishes, fire the ENDED event to JS via audioEventHandlerRegistry_->invokeHandlerWithEventBody(AudioEvent::ENDED, {}).
processNode() Signature
protected:
// Audio-thread only
virtual std::shared_ptr<AudioBuffer> processNode(
const std::shared_ptr<AudioBuffer> &processingBuffer,
int framesToProcess) = 0;
processingBuffer— already contains the mixed input from all connected input nodes. Modify in-place and return it.framesToProcess— number of samples per channel to process, typically 128 (RENDER_QUANTUM_SIZE).- Called by
AudioNode::processAudio()which handles input mixing, channel count modes, and deduplication (vialastRenderedFrame_).
Thread Annotations in Header Files
Annotate every method with the thread it is safe to call from. Use comments in the header:
class MyNode : public AudioNode {
public:
// JS-thread only
void setSomething(float value);
float getSomething() const;
protected:
// Audio-thread only
std::shared_ptr<AudioBuffer> processNode(
const std::shared_ptr<AudioBuffer> &processingBuffer,
int framesToProcess) override;
};
In AudioParam.h the pattern is:
/// JS-Thread only methods
[[nodiscard]] inline float getValue() const noexcept { ... }
void setValue(float value);
void setValueAtTime(float value, double startTime);
/// Audio-Thread only methods
std::shared_ptr<AudioBuffer> processARateParam(int framesToProcess, double time);
float processKRateParam(int framesToProcess, double time);
AudioParam — Automatable Parameters
Every automatable property (frequency, gain, detune, Q, etc.) is an AudioParam.
gainParam_ = std::make_shared<AudioParam>(
defaultValue,
minValue,
maxValue,
context
);
A-rate vs K-rate
-
A-rate (audio-rate): one value per sample — use when the parameter can change significantly within a render quantum (e.g. frequency modulation)
// Call processARateParam() for per-sample values — returns AudioBuffer, no allocation auto gainValues = gainParam_->processARateParam(framesToProcess, time); float *values = gainValues->getChannel(0)->getData(); // values[i] is the gain for frame i -
K-rate (control-rate): one value per render quantum — use when the parameter changes slowly
// Call processKRateParam() for a single block-wide value float gain = gainParam_->processKRateParam(framesToProcess, time); // Single value for the whole block
JS → Audio Thread parameter updates
CrossThreadEventScheduler<T> is a lock-free SPSC channel. When JS calls param.setValueAtTime(...), it enqueues a lambda on the scheduler. The audio thread drains the queue at the start of each processARateParam / processKRateParam call.
// JS-thread (in AudioParam):
void AudioParam::setValueAtTime(float value, double startTime) {
eventScheduler_.scheduleEvent([value, startTime](AudioParam ¶m) {
param.eventsQueue_.insertEvent(...);
});
}
// Audio-thread (inside processARateParam):
eventScheduler_.processAllEvents(*this); // drain all pending events
Important: HostObject setters forward to the node/param asynchronously through this scheduler. By the time processNode() runs, the queued update may or may not have been applied yet, depending on timing. Design accordingly — never assume immediate consistency.
Cross-Thread Communication Patterns
JS → Audio (parameter/graph updates)
Use CrossThreadEventScheduler (lock-free SPSC queue). See utils/CrossThreadEventScheduler.hpp.
Audio → JS (events like ended, loopEnded, positionChanged)
Use IAudioEventHandlerRegistry::invokeHandlerWithEventBody() which internally calls callInvoker_->invokeAsync() — this safely schedules the JS callback on the JS thread from the audio thread.
// Audio-thread: fire 'ended' event
audioEventHandlerRegistry_->invokeHandlerWithEventBody(
AudioEvent::ENDED, {});
Callback IDs are stored as std::atomic<uint64_t> on the node. 0 means no listener registered.
JS → Audio (graph mutations: connect/disconnect)
All graph mutations are queued via AudioGraphManager using its own SPSC channel (addPendingNodeConnection, addPendingParamConnection). The audio thread calls graphManager_->preProcessGraph() before each render pass to apply pending changes.
Implementing a New Node — Checklist
-
Subclass the right base
AudioNode— standard effect or analysis nodeAudioScheduledSourceNode— source with start/stop schedulingAudioBufferBaseSourceNode— source that plays back an AudioBuffer with pitch control
-
Header file (
core/<category>/MyNode.h)- Annotate every method with
// JS-thread onlyor// Audio-thread only - Declare
processNode()inprotected: - Declare
AudioParammembers for automatable properties - Preallocate all buffers you'll need in
private:state
- Annotate every method with
-
Constructor (runs on JS thread)
- Call
AudioNode(context, options)base constructor with correctnumberOfInputs,numberOfOutputs - Create all
AudioParaminstances with correct default/min/max values from the Web Audio spec - Preallocate any DSP state buffers (IIR delay lines, ring buffers, etc.)
- Do NOT call
context_->...inprocessNode()for anything that could block
- Call
-
processNode() (runs on audio thread)
- Call
context_.lock()to get ashared_ptr<BaseAudioContext>— return early if null - Call
context->getCurrentTime()for automation timing - Use
processARateParam()orprocessKRateParam()to read param values - Process samples in-place on
processingBuffer - No allocations, no locks, no blocking I/O
- Call
-
HostObject (see the
host-objectsskill)- Create
MyNodeHostObjectextendingAudioNodeHostObject - Add factory method to
BaseAudioContextHostObject(createMyNode) - Add factory method to
BaseAudioContextC++ class
- Create
-
TypeScript API (see the
turbo-modulesskill)- Add TS class in
src/core/ - Export from package index
- Add TS class in
-
Spec compliance
- Check the Web Audio API spec for default values, parameter ranges, and behavior
- See
web-audio-api.mdskill
-
Tests and docs — see the
flowskill
See full GainNode example for a complete header + .cpp reference implementation.
Web Audio API Spec Reference
All node behavior (parameter names, default values, valid ranges, processing semantics) must match the spec:
Key spec-defined constraints already encoded in the codebase:
AudioParammin/max values come from spec tablesGainNode.gaindefault = 1.0, no clampingBiquadFilterNode.frequencydefault = 350 Hz, range [Nyquist - epsilon, Nyquist]OscillatorNode.frequencydefault = 440 Hz- Render quantum = 128 frames
Maintenance: see maintenance.md.