Optional
options: Partial<FrequencyShifterOptions>Readonly
contextThe context belonging to the node.
Set this debug flag to log all events that happen in this class.
Readonly
frequencyThe ring modulators carrier frequency. This frequency determines by how many Hertz the input signal will be shifted up or down. Default is 0.
The effect input node
Readonly
nameThe effect output
The wet control is how much of the effected will pass through to the output. 1 = 100% effected signal, 0 = 100% dry signal.
Static
versionThe version number semver
The number of seconds of 1 processing block (128 samples)
console.log(Tone.Destination.blockTime);
channelCount is the number of channels used when up-mixing and down-mixing connections to any inputs to the node. The default value is 2 except for specific nodes where its value is specially determined.
channelCountMode determines how channels will be counted when up-mixing and down-mixing connections to any inputs to the node. The default value is "max". This attribute has no effect for nodes with no inputs.
channelInterpretation determines how individual channels will be treated when up-mixing and down-mixing connections to any inputs to the node. The default value is "speakers".
Indicates if the instance was disposed. 'Disposing' an instance means that all of the Web Audio nodes that were created for the instance are disconnected and freed for garbage collection.
The number of inputs feeding into the AudioNode. For source nodes, this will be 0.
const node = new Tone.Gain();
console.log(node.numberOfInputs);
The number of outputs of the AudioNode.
const node = new Tone.Gain();
console.log(node.numberOfOutputs);
The duration in seconds of one sample.
Connect the output of this node to the rest of the nodes in series.
Rest
...nodes: InputNode[]const player = new Tone.Player("https://tonejs.github.io/audio/drum-samples/handdrum-loop.mp3");
player.autostart = true;
const filter = new Tone.AutoFilter(4).start();
const distortion = new Tone.Distortion(0.5);
// connect the player to the filter, distortion and then to the master output
player.chain(filter, distortion, Tone.Destination);
connect the output of a ToneAudioNode to an AudioParam, AudioNode, or ToneAudioNode
The output to connect to
The output to connect from
The input to connect to
disconnect the output
Optional
destination: InputNodeconnect the output of this node to the rest of the nodes in parallel.
Rest
...nodes: InputNode[]const player = new Tone.Player("https://tonejs.github.io/audio/drum-samples/conga-rhythm.mp3");
player.autostart = true;
const pitchShift = new Tone.PitchShift(4).toDestination();
const filter = new Tone.Filter("G5").toDestination();
// connect a node to the pitch shift and filter in parallel
player.fan(pitchShift, filter);
Set multiple properties at once with an object.
const filter = new Tone.Filter().toDestination();
// set values using an object
filter.set({
frequency: "C6",
type: "highpass"
});
const player = new Tone.Player("https://tonejs.github.io/audio/berklee/Analogsynth_octaves_highmid.mp3").connect(filter);
player.autostart = true;
Convert the incoming time to seconds. This is calculated against the current TransportClass bpm
const gain = new Tone.Gain();
setInterval(() => console.log(gain.toSeconds("4n")), 100);
// ramp the tempo to 60 bpm over 30 seconds
Tone.getTransport().bpm.rampTo(60, 30);
Static
get
FrequencyShifter can be used to shift all frequencies of a signal by a fixed amount. The amount can be changed at audio rate and the effect is applied in real time. The frequency shifting is implemented with a technique called single side band modulation using a ring modulator. Note: Contrary to pitch shifting, all frequencies are shifted by the same amount, destroying the harmonic relationship between them. This leads to the classic ring modulator timbre distortion. The algorithm will produces some aliasing towards the high end, especially if your source material contains a lot of high frequencies. Unfortunatelly the webaudio API does not support resampling buffers in real time, so it is not possible to fix it properly. Depending on the use case it might be an option to low pass filter your input before frequency shifting it to get ride of the aliasing. You can find a very detailed description of the algorithm here: https://larzeitlin.github.io/RMFS/
Example