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    1.1.2 • Public • Published

    libltc wrapper


    This module gives you access to the libltc library from node.js.

    Bits per sample

    Internally libltc uses 8 bit unsigned audio data, but the decoder supports reading uint16, sint16 and float too. The encoder however only supports uint8.


    The following architectures are supported:

    • linux-x64
    • linux-armhf
    • linux-arm64
    • macos-x64
    • macos-arm64 (M1)
    • windows-x64

    Add the module to your project:

    npm i libltc-wrapper
    # or
    yarn add libltc-wrapper


    • Successfully encodes and decodes LTC audio
    • Tested successfully on Macbook Pro M1 (arm64) and Ubuntu 20.04 (x64)



        const { LTCDecoder } = require('libltc-wrapper');
        const decoder = new LTCDecoder(48000, 25, "u8"); // 48khz, 25 fps, unsigned 8 bit
        // Write audio buffer to the decoder
        // Check if there is any ltc frames ready
        let frame =;
        if (frame !== undefined) {
            // found a valid frame
            console.log("Frame: ", frame);


        const { LTCEncoder, LTC_USE_DATE } = require('libltc-wrapper');
        const encoder = new LTCEncoder(48000, 25, LTC_USE_DATE); // 48khz, 25 fps, LTC_USE_DATE flag
        let time = new Date();
          hours: time.getHours(),
          minutes: time.getMinutes(),
          seconds: time.getSeconds(),
          frame: time.getMilliseconds() / (1000 / 25),
          days: time.getDate(),
          months: time.getMonth() + 1,
          years: time.getFullYear() % 100,
          timezone: "+0000"
        // Every frame (25 times per second) increase the current frame
        // Get 1 frame worth of LTC audio (48khz 25fps would be 40ms audio)
        let buffer = encoder.getBuffer();

    Class methods

    LTCDecoder instance methods

    constructor(sampleRate: number, framerate: number, audioFormat: string)

    Samplerate in hz, for example 48000. Framerate should be less than or equal to 30 fps. Audioformat must be one of u8 for unsigned 8 bit, u16 for unsigned 16 bit, s16 for signed 16 bit, or float for 32 bit float.

    write(buffer: Buffer)

    Write audio data to the decoder, should be full frames, but any number of frames. They will be buffered up until it finds a LTC frame


    Reads the next frame from the decoder queue. If there are no frames in the queue it returns undefined. When a valid frame is found, it returns a LTCFrame object.

    LTCEncoder instance methods

    constructor(sampleRate: number, framerate: number, flags?: number);

    Samplerate in hz, for example 48000. Framerate should be less than or equal to 30. Flags should be one, or a combination of the following flags: LTC_USE_DATE, LTC_TC_CLOCK, LTC_BGF_DONT_TOUCH, LTC_NO_PARITY. See the librtc documentation for more information about the flags. The default flag is LTC_USE_DATE.

    setVolume(dBFS: number)

    Set the volume of the generated LTC signal, in dBFs

    setFilter(riseTime: number)

    Set encoder signal rise-time / signal filtering, in uS. Default is 40.

    setTimecode(timecode: LTCTimecode)

    Set encoder timecode, expects a LTCTimecode object. You only need to specify non-zero parameters. The default is 0 for any omitted object parameters.


    Get the current encoder timecode. Returns a LTCTimecode object with the current position.


    Write the next frame to the audio buffer. Fetch the buffer using getBuffer().


    Increment the current timecode by one frame.


    Decrement the timecode by one frame


    Returns the audio buffer for the current frame. The size of this buffer would be sample_rate / fps. So at 48000 khz, 25fps, that would be 1920 bytes of unsigned 8 bit audio.


    This node module is licensed under the MIT license. And libltc is licensed under the GNU LGPL 3 license, but is included in this project in binary form and dynamically linked. So this module should in theory be safe to use in any open source or closed source commercial products. I am not a lawyer. This must not be construed as legal advice. Read more here




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