@layercode/js-sdk 1.0.3

package/dist/layercode-js-sdk.esm.js

@@ -0,0 +1,1827 @@
+/**
+ * Raw wav audio file contents
+ * @typedef {Object} WavPackerAudioType
+ * @property {Blob} blob
+ * @property {string} url
+ * @property {number} channelCount
+ * @property {number} sampleRate
+ * @property {number} duration
+ */
+
+/**
+ * Utility class for assembling PCM16 "audio/wav" data
+ * @class
+ */
+class WavPacker {
+  /**
+   * Converts Float32Array of amplitude data to ArrayBuffer in Int16Array format
+   * @param {Float32Array} float32Array
+   * @returns {ArrayBuffer}
+   */
+  static floatTo16BitPCM(float32Array) {
+    const buffer = new ArrayBuffer(float32Array.length * 2);
+    const view = new DataView(buffer);
+    let offset = 0;
+    for (let i = 0; i < float32Array.length; i++, offset += 2) {
+      let s = Math.max(-1, Math.min(1, float32Array[i]));
+      view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);
+    }
+    return buffer;
+  }
+
+  /**
+   * Concatenates two ArrayBuffers
+   * @param {ArrayBuffer} leftBuffer
+   * @param {ArrayBuffer} rightBuffer
+   * @returns {ArrayBuffer}
+   */
+  static mergeBuffers(leftBuffer, rightBuffer) {
+    const tmpArray = new Uint8Array(
+      leftBuffer.byteLength + rightBuffer.byteLength
+    );
+    tmpArray.set(new Uint8Array(leftBuffer), 0);
+    tmpArray.set(new Uint8Array(rightBuffer), leftBuffer.byteLength);
+    return tmpArray.buffer;
+  }
+
+  /**
+   * Packs data into an Int16 format
+   * @private
+   * @param {number} size 0 = 1x Int16, 1 = 2x Int16
+   * @param {number} arg value to pack
+   * @returns
+   */
+  _packData(size, arg) {
+    return [
+      new Uint8Array([arg, arg >> 8]),
+      new Uint8Array([arg, arg >> 8, arg >> 16, arg >> 24]),
+    ][size];
+  }
+
+  /**
+   * Packs audio into "audio/wav" Blob
+   * @param {number} sampleRate
+   * @param {{bitsPerSample: number, channels: Array<Float32Array>, data: Int16Array}} audio
+   * @returns {WavPackerAudioType}
+   */
+  pack(sampleRate, audio) {
+    if (!audio?.bitsPerSample) {
+      throw new Error(`Missing "bitsPerSample"`);
+    } else if (!audio?.channels) {
+      throw new Error(`Missing "channels"`);
+    } else if (!audio?.data) {
+      throw new Error(`Missing "data"`);
+    }
+    const { bitsPerSample, channels, data } = audio;
+    const output = [
+      // Header
+      'RIFF',
+      this._packData(
+        1,
+        4 + (8 + 24) /* chunk 1 length */ + (8 + 8) /* chunk 2 length */
+      ), // Length
+      'WAVE',
+      // chunk 1
+      'fmt ', // Sub-chunk identifier
+      this._packData(1, 16), // Chunk length
+      this._packData(0, 1), // Audio format (1 is linear quantization)
+      this._packData(0, channels.length),
+      this._packData(1, sampleRate),
+      this._packData(1, (sampleRate * channels.length * bitsPerSample) / 8), // Byte rate
+      this._packData(0, (channels.length * bitsPerSample) / 8),
+      this._packData(0, bitsPerSample),
+      // chunk 2
+      'data', // Sub-chunk identifier
+      this._packData(
+        1,
+        (channels[0].length * channels.length * bitsPerSample) / 8
+      ), // Chunk length
+      data,
+    ];
+    const blob = new Blob(output, { type: 'audio/mpeg' });
+    const url = URL.createObjectURL(blob);
+    return {
+      blob,
+      url,
+      channelCount: channels.length,
+      sampleRate,
+      duration: data.byteLength / (channels.length * sampleRate * 2),
+    };
+  }
+}
+
+globalThis.WavPacker = WavPacker;
+
+/**
+ * Constants for help with visualization
+ * Helps map frequency ranges from Fast Fourier Transform
+ * to human-interpretable ranges, notably music ranges and
+ * human vocal ranges.
+ */
+
+// Eighth octave frequencies
+const octave8Frequencies = [
+  4186.01, 4434.92, 4698.63, 4978.03, 5274.04, 5587.65, 5919.91, 6271.93,
+  6644.88, 7040.0, 7458.62, 7902.13,
+];
+
+// Labels for each of the above frequencies
+const octave8FrequencyLabels = [
+  'C',
+  'C#',
+  'D',
+  'D#',
+  'E',
+  'F',
+  'F#',
+  'G',
+  'G#',
+  'A',
+  'A#',
+  'B',
+];
+
+/**
+ * All note frequencies from 1st to 8th octave
+ * in format "A#8" (A#, 8th octave)
+ */
+const noteFrequencies = [];
+const noteFrequencyLabels = [];
+for (let i = 1; i <= 8; i++) {
+  for (let f = 0; f < octave8Frequencies.length; f++) {
+    const freq = octave8Frequencies[f];
+    noteFrequencies.push(freq / Math.pow(2, 8 - i));
+    noteFrequencyLabels.push(octave8FrequencyLabels[f] + i);
+  }
+}
+
+/**
+ * Subset of the note frequencies between 32 and 2000 Hz
+ * 6 octave range: C1 to B6
+ */
+const voiceFrequencyRange = [32.0, 2000.0];
+const voiceFrequencies = noteFrequencies.filter((_, i) => {
+  return (
+    noteFrequencies[i] > voiceFrequencyRange[0] &&
+    noteFrequencies[i] < voiceFrequencyRange[1]
+  );
+});
+const voiceFrequencyLabels = noteFrequencyLabels.filter((_, i) => {
+  return (
+    noteFrequencies[i] > voiceFrequencyRange[0] &&
+    noteFrequencies[i] < voiceFrequencyRange[1]
+  );
+});
+
+/**
+ * Output of AudioAnalysis for the frequency domain of the audio
+ * @typedef {Object} AudioAnalysisOutputType
+ * @property {Float32Array} values Amplitude of this frequency between {0, 1} inclusive
+ * @property {number[]} frequencies Raw frequency bucket values
+ * @property {string[]} labels Labels for the frequency bucket values
+ */
+
+/**
+ * Analyzes audio for visual output
+ * @class
+ */
+class AudioAnalysis {
+  /**
+   * Retrieves frequency domain data from an AnalyserNode adjusted to a decibel range
+   * returns human-readable formatting and labels
+   * @param {AnalyserNode} analyser
+   * @param {number} sampleRate
+   * @param {Float32Array} [fftResult]
+   * @param {"frequency"|"music"|"voice"} [analysisType]
+   * @param {number} [minDecibels] default -100
+   * @param {number} [maxDecibels] default -30
+   * @returns {AudioAnalysisOutputType}
+   */
+  static getFrequencies(
+    analyser,
+    sampleRate,
+    fftResult,
+    analysisType = 'frequency',
+    minDecibels = -100,
+    maxDecibels = -30,
+  ) {
+    if (!fftResult) {
+      fftResult = new Float32Array(analyser.frequencyBinCount);
+      analyser.getFloatFrequencyData(fftResult);
+    }
+    const nyquistFrequency = sampleRate / 2;
+    const frequencyStep = (1 / fftResult.length) * nyquistFrequency;
+    let outputValues;
+    let frequencies;
+    let labels;
+    if (analysisType === 'music' || analysisType === 'voice') {
+      const useFrequencies =
+        analysisType === 'voice' ? voiceFrequencies : noteFrequencies;
+      const aggregateOutput = Array(useFrequencies.length).fill(minDecibels);
+      for (let i = 0; i < fftResult.length; i++) {
+        const frequency = i * frequencyStep;
+        const amplitude = fftResult[i];
+        for (let n = useFrequencies.length - 1; n >= 0; n--) {
+          if (frequency > useFrequencies[n]) {
+            aggregateOutput[n] = Math.max(aggregateOutput[n], amplitude);
+            break;
+          }
+        }
+      }
+      outputValues = aggregateOutput;
+      frequencies =
+        analysisType === 'voice' ? voiceFrequencies : noteFrequencies;
+      labels =
+        analysisType === 'voice' ? voiceFrequencyLabels : noteFrequencyLabels;
+    } else {
+      outputValues = Array.from(fftResult);
+      frequencies = outputValues.map((_, i) => frequencyStep * i);
+      labels = frequencies.map((f) => `${f.toFixed(2)} Hz`);
+    }
+    // We normalize to {0, 1}
+    const normalizedOutput = outputValues.map((v) => {
+      return Math.max(
+        0,
+        Math.min((v - minDecibels) / (maxDecibels - minDecibels), 1),
+      );
+    });
+    const values = new Float32Array(normalizedOutput);
+    return {
+      values,
+      frequencies,
+      labels,
+    };
+  }
+
+  /**
+   * Creates a new AudioAnalysis instance for an HTMLAudioElement
+   * @param {HTMLAudioElement} audioElement
+   * @param {AudioBuffer|null} [audioBuffer] If provided, will cache all frequency domain data from the buffer
+   * @returns {AudioAnalysis}
+   */
+  constructor(audioElement, audioBuffer = null) {
+    this.fftResults = [];
+    if (audioBuffer) {
+      /**
+       * Modified from
+       * https://stackoverflow.com/questions/75063715/using-the-web-audio-api-to-analyze-a-song-without-playing
+       *
+       * We do this to populate FFT values for the audio if provided an `audioBuffer`
+       * The reason to do this is that Safari fails when using `createMediaElementSource`
+       * This has a non-zero RAM cost so we only opt-in to run it on Safari, Chrome is better
+       */
+      const { length, sampleRate } = audioBuffer;
+      const offlineAudioContext = new OfflineAudioContext({
+        length,
+        sampleRate,
+      });
+      const source = offlineAudioContext.createBufferSource();
+      source.buffer = audioBuffer;
+      const analyser = offlineAudioContext.createAnalyser();
+      analyser.fftSize = 8192;
+      analyser.smoothingTimeConstant = 0.1;
+      source.connect(analyser);
+      // limit is :: 128 / sampleRate;
+      // but we just want 60fps - cuts ~1s from 6MB to 1MB of RAM
+      const renderQuantumInSeconds = 1 / 60;
+      const durationInSeconds = length / sampleRate;
+      const analyze = (index) => {
+        const suspendTime = renderQuantumInSeconds * index;
+        if (suspendTime < durationInSeconds) {
+          offlineAudioContext.suspend(suspendTime).then(() => {
+            const fftResult = new Float32Array(analyser.frequencyBinCount);
+            analyser.getFloatFrequencyData(fftResult);
+            this.fftResults.push(fftResult);
+            analyze(index + 1);
+          });
+        }
+        if (index === 1) {
+          offlineAudioContext.startRendering();
+        } else {
+          offlineAudioContext.resume();
+        }
+      };
+      source.start(0);
+      analyze(1);
+      this.audio = audioElement;
+      this.context = offlineAudioContext;
+      this.analyser = analyser;
+      this.sampleRate = sampleRate;
+      this.audioBuffer = audioBuffer;
+    } else {
+      const audioContext = new AudioContext();
+      const track = audioContext.createMediaElementSource(audioElement);
+      const analyser = audioContext.createAnalyser();
+      analyser.fftSize = 8192;
+      analyser.smoothingTimeConstant = 0.1;
+      track.connect(analyser);
+      analyser.connect(audioContext.destination);
+      this.audio = audioElement;
+      this.context = audioContext;
+      this.analyser = analyser;
+      this.sampleRate = this.context.sampleRate;
+      this.audioBuffer = null;
+    }
+  }
+
+  /**
+   * Gets the current frequency domain data from the playing audio track
+   * @param {"frequency"|"music"|"voice"} [analysisType]
+   * @param {number} [minDecibels] default -100
+   * @param {number} [maxDecibels] default -30
+   * @returns {AudioAnalysisOutputType}
+   */
+  getFrequencies(
+    analysisType = 'frequency',
+    minDecibels = -100,
+    maxDecibels = -30,
+  ) {
+    let fftResult = null;
+    if (this.audioBuffer && this.fftResults.length) {
+      const pct = this.audio.currentTime / this.audio.duration;
+      const index = Math.min(
+        (pct * this.fftResults.length) | 0,
+        this.fftResults.length - 1,
+      );
+      fftResult = this.fftResults[index];
+    }
+    return AudioAnalysis.getFrequencies(
+      this.analyser,
+      this.sampleRate,
+      fftResult,
+      analysisType,
+      minDecibels,
+      maxDecibels,
+    );
+  }
+
+  /**
+   * Resume the internal AudioContext if it was suspended due to the lack of
+   * user interaction when the AudioAnalysis was instantiated.
+   * @returns {Promise<true>}
+   */
+  async resumeIfSuspended() {
+    if (this.context.state === 'suspended') {
+      await this.context.resume();
+    }
+    return true;
+  }
+}
+
+globalThis.AudioAnalysis = AudioAnalysis;
+
+const StreamProcessorWorklet = `
+class StreamProcessor extends AudioWorkletProcessor {
+  constructor() {
+    super();
+    this.hasStarted = false;
+    this.hasInterrupted = false;
+    this.outputBuffers = [];
+    this.bufferLength = 128;
+    this.write = { buffer: new Float32Array(this.bufferLength), trackId: null };
+    this.writeOffset = 0;
+    this.trackSampleOffsets = {};
+    this.port.onmessage = (event) => {
+      if (event.data) {
+        const payload = event.data;
+        if (payload.event === 'write') {
+          const int16Array = payload.buffer;
+          const float32Array = new Float32Array(int16Array.length);
+          for (let i = 0; i < int16Array.length; i++) {
+            float32Array[i] = int16Array[i] / 0x8000; // Convert Int16 to Float32
+          }
+          this.writeData(float32Array, payload.trackId);
+        } else if (
+          payload.event === 'offset' ||
+          payload.event === 'interrupt'
+        ) {
+          const requestId = payload.requestId;
+          const trackId = this.write.trackId;
+          const offset = this.trackSampleOffsets[trackId] || 0;
+          this.port.postMessage({
+            event: 'offset',
+            requestId,
+            trackId,
+            offset,
+          });
+          if (payload.event === 'interrupt') {
+            this.hasInterrupted = true;
+          }
+        } else {
+          throw new Error(\`Unhandled event "\${payload.event}"\`);
+        }
+      }
+    };
+  }
+
+  writeData(float32Array, trackId = null) {
+    let { buffer } = this.write;
+    let offset = this.writeOffset;
+    for (let i = 0; i < float32Array.length; i++) {
+      buffer[offset++] = float32Array[i];
+      if (offset >= buffer.length) {
+        this.outputBuffers.push(this.write);
+        this.write = { buffer: new Float32Array(this.bufferLength), trackId };
+        buffer = this.write.buffer;
+        offset = 0;
+      }
+    }
+    this.writeOffset = offset;
+    return true;
+  }
+
+  process(inputs, outputs, parameters) {
+    const output = outputs[0];
+    const outputChannelData = output[0];
+    const outputBuffers = this.outputBuffers;
+    if (this.hasInterrupted) {
+      this.port.postMessage({ event: 'stop' });
+      return false;
+    } else if (outputBuffers.length) {
+      this.hasStarted = true;
+      const { buffer, trackId } = outputBuffers.shift();
+      for (let i = 0; i < outputChannelData.length; i++) {
+        outputChannelData[i] = buffer[i] || 0;
+      }
+      if (trackId) {
+        this.trackSampleOffsets[trackId] =
+          this.trackSampleOffsets[trackId] || 0;
+        this.trackSampleOffsets[trackId] += buffer.length;
+      }
+      return true;
+    } else if (this.hasStarted) {
+      this.port.postMessage({ event: 'stop' });
+      return false;
+    } else {
+      return true;
+    }
+  }
+}
+
+registerProcessor('stream_processor', StreamProcessor);
+`;
+
+const script$1 = new Blob([StreamProcessorWorklet], {
+  type: 'application/javascript',
+});
+const src$1 = URL.createObjectURL(script$1);
+const StreamProcessorSrc = src$1;
+
+/**
+ * Plays audio streams received in raw PCM16 chunks from the browser
+ * @class
+ */
+class WavStreamPlayer {
+  /**
+   * Creates a new WavStreamPlayer instance
+   * @param {{sampleRate?: number}} options
+   * @returns {WavStreamPlayer}
+   */
+  constructor({ finishedPlayingCallback = () => {}, sampleRate = 24000 } = {}) {
+    this.scriptSrc = StreamProcessorSrc;
+    this.sampleRate = sampleRate;
+    this.context = null;
+    this.stream = null;
+    this.analyser = null;
+    this.trackSampleOffsets = {};
+    this.interruptedTrackIds = {};
+    this.finishedPlayingCallback = finishedPlayingCallback;
+  }
+
+  /**
+   * Connects the audio context and enables output to speakers
+   * @returns {Promise<true>}
+   */
+  async connect() {
+    this.context = new AudioContext({ sampleRate: this.sampleRate });
+    if (this.context.state === "suspended") {
+      await this.context.resume();
+    }
+    try {
+      await this.context.audioWorklet.addModule(this.scriptSrc);
+    } catch (e) {
+      console.error(e);
+      throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);
+    }
+    const analyser = this.context.createAnalyser();
+    analyser.fftSize = 8192;
+    analyser.smoothingTimeConstant = 0.1;
+    this.analyser = analyser;
+    return true;
+  }
+
+  /**
+   * Gets the current frequency domain data from the playing track
+   * @param {"frequency"|"music"|"voice"} [analysisType]
+   * @param {number} [minDecibels] default -100
+   * @param {number} [maxDecibels] default -30
+   * @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}
+   */
+  getFrequencies(analysisType = "frequency", minDecibels = -100, maxDecibels = -30) {
+    if (!this.analyser) {
+      throw new Error("Not connected, please call .connect() first");
+    }
+    return AudioAnalysis.getFrequencies(this.analyser, this.sampleRate, null, analysisType, minDecibels, maxDecibels);
+  }
+
+  /**
+   * Gets the real-time amplitude of the audio signal
+   * @returns {number} Amplitude value between 0 and 1
+   */
+  getAmplitude() {
+    if (!this.analyser) {
+      throw new Error("AnalyserNode is not initialized. Please call connect() first.");
+    }
+
+    const bufferLength = this.analyser.fftSize;
+    const dataArray = new Uint8Array(bufferLength);
+    this.analyser.getByteTimeDomainData(dataArray);
+
+    // Calculate RMS (Root Mean Square) to get amplitude
+    let sumSquares = 0;
+    for (let i = 0; i < bufferLength; i++) {
+      const normalized = (dataArray[i] - 128) / 128; // Normalize between -1 and 1
+      sumSquares += normalized * normalized;
+    }
+    const rms = Math.sqrt(sumSquares / bufferLength);
+    return rms;
+  }
+
+  /**
+   * Starts amplitude monitoring
+   * @param {function} callback - Function to call with amplitude value
+   */
+  startAmplitudeMonitoring(callback) {
+    const monitor = () => {
+      const amplitude = this.getAmplitude();
+      callback(amplitude);
+      requestAnimationFrame(monitor);
+    };
+    monitor();
+  }
+
+  /**
+   * Starts audio streaming
+   * @private
+   * @returns {Promise<true>}
+   */
+  _start() {
+    const streamNode = new AudioWorkletNode(this.context, "stream_processor");
+    streamNode.connect(this.context.destination);
+    streamNode.port.onmessage = (e) => {
+      const { event } = e.data;
+      if (event === "stop") {
+        streamNode.disconnect();
+        this.stream = null;
+        this.finishedPlayingCallback();
+      } else if (event === "offset") {
+        const { requestId, trackId, offset } = e.data;
+        const currentTime = offset / this.sampleRate;
+        this.trackSampleOffsets[requestId] = { trackId, offset, currentTime };
+      }
+    };
+    this.analyser.disconnect();
+    streamNode.connect(this.analyser);
+    this.stream = streamNode;
+    return true;
+  }
+
+  /**
+   * Adds 16BitPCM data to the currently playing audio stream
+   * You can add chunks beyond the current play point and they will be queued for play
+   * @param {ArrayBuffer|Int16Array} arrayBuffer
+   * @param {string} [trackId]
+   * @returns {Int16Array}
+   */
+  add16BitPCM(arrayBuffer, trackId = "default") {
+    if (typeof trackId !== "string") {
+      throw new Error(`trackId must be a string`);
+    } else if (this.interruptedTrackIds[trackId]) {
+      return;
+    }
+    if (!this.stream) {
+      this._start();
+    }
+    let buffer;
+    if (arrayBuffer instanceof Int16Array) {
+      buffer = arrayBuffer;
+    } else if (arrayBuffer instanceof ArrayBuffer) {
+      buffer = new Int16Array(arrayBuffer);
+    } else {
+      throw new Error(`argument must be Int16Array or ArrayBuffer`);
+    }
+    this.stream.port.postMessage({ event: "write", buffer, trackId });
+    return buffer;
+  }
+
+  /**
+   * Gets the offset (sample count) of the currently playing stream
+   * @param {boolean} [interrupt]
+   * @returns {{trackId: string|null, offset: number, currentTime: number}}
+   */
+  async getTrackSampleOffset(interrupt = false) {
+    if (!this.stream) {
+      return null;
+    }
+    const requestId = crypto.randomUUID();
+    this.stream.port.postMessage({
+      event: interrupt ? "interrupt" : "offset",
+      requestId,
+    });
+    let trackSampleOffset;
+    while (!trackSampleOffset) {
+      trackSampleOffset = this.trackSampleOffsets[requestId];
+      await new Promise((r) => setTimeout(() => r(), 1));
+    }
+    const { trackId } = trackSampleOffset;
+    if (interrupt && trackId) {
+      this.interruptedTrackIds[trackId] = true;
+    }
+    return trackSampleOffset;
+  }
+
+  /**
+   * Strips the current stream and returns the sample offset of the audio
+   * @param {boolean} [interrupt]
+   * @returns {{trackId: string|null, offset: number, currentTime: number}}
+   */
+  async interrupt() {
+    return this.getTrackSampleOffset(true);
+  }
+
+  /**
+   * Disconnects the audio context and cleans up resources
+   * @returns {void}
+   */
+  disconnect() {
+    if (this.stream) {
+      this.stream.disconnect();
+      this.stream = null;
+    }
+
+    if (this.analyser) {
+      this.analyser.disconnect();
+    }
+
+    if (this.context) {
+      this.context.close().catch((err) => console.error("Error closing audio context:", err));
+    }
+  }
+}
+
+globalThis.WavStreamPlayer = WavStreamPlayer;
+
+const AudioProcessorWorklet = `
+class AudioProcessor extends AudioWorkletProcessor {
+
+  constructor() {
+    super();
+    this.port.onmessage = this.receive.bind(this);
+    this.initialize();
+  }
+
+  initialize() {
+    this.foundAudio = false;
+    this.recording = false;
+    this.chunks = [];
+  }
+
+  /**
+   * Concatenates sampled chunks into channels
+   * Format is chunk[Left[], Right[]]
+   */
+  readChannelData(chunks, channel = -1, maxChannels = 9) {
+    let channelLimit;
+    if (channel !== -1) {
+      if (chunks[0] && chunks[0].length - 1 < channel) {
+        throw new Error(
+          \`Channel \${channel} out of range: max \${chunks[0].length}\`
+        );
+      }
+      channelLimit = channel + 1;
+    } else {
+      channel = 0;
+      channelLimit = Math.min(chunks[0] ? chunks[0].length : 1, maxChannels);
+    }
+    const channels = [];
+    for (let n = channel; n < channelLimit; n++) {
+      const length = chunks.reduce((sum, chunk) => {
+        return sum + chunk[n].length;
+      }, 0);
+      const buffers = chunks.map((chunk) => chunk[n]);
+      const result = new Float32Array(length);
+      let offset = 0;
+      for (let i = 0; i < buffers.length; i++) {
+        result.set(buffers[i], offset);
+        offset += buffers[i].length;
+      }
+      channels[n] = result;
+    }
+    return channels;
+  }
+
+  /**
+   * Combines parallel audio data into correct format,
+   * channels[Left[], Right[]] to float32Array[LRLRLRLR...]
+   */
+  formatAudioData(channels) {
+    if (channels.length === 1) {
+      // Simple case is only one channel
+      const float32Array = channels[0].slice();
+      const meanValues = channels[0].slice();
+      return { float32Array, meanValues };
+    } else {
+      const float32Array = new Float32Array(
+        channels[0].length * channels.length
+      );
+      const meanValues = new Float32Array(channels[0].length);
+      for (let i = 0; i < channels[0].length; i++) {
+        const offset = i * channels.length;
+        let meanValue = 0;
+        for (let n = 0; n < channels.length; n++) {
+          float32Array[offset + n] = channels[n][i];
+          meanValue += channels[n][i];
+        }
+        meanValues[i] = meanValue / channels.length;
+      }
+      return { float32Array, meanValues };
+    }
+  }
+
+  /**
+   * Converts 32-bit float data to 16-bit integers
+   */
+  floatTo16BitPCM(float32Array) {
+    const buffer = new ArrayBuffer(float32Array.length * 2);
+    const view = new DataView(buffer);
+    let offset = 0;
+    for (let i = 0; i < float32Array.length; i++, offset += 2) {
+      let s = Math.max(-1, Math.min(1, float32Array[i]));
+      view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);
+    }
+    return buffer;
+  }
+
+  /**
+   * Retrieves the most recent amplitude values from the audio stream
+   * @param {number} channel
+   */
+  getValues(channel = -1) {
+    const channels = this.readChannelData(this.chunks, channel);
+    const { meanValues } = this.formatAudioData(channels);
+    return { meanValues, channels };
+  }
+
+  /**
+   * Exports chunks as an audio/wav file
+   */
+  export() {
+    const channels = this.readChannelData(this.chunks);
+    const { float32Array, meanValues } = this.formatAudioData(channels);
+    const audioData = this.floatTo16BitPCM(float32Array);
+    return {
+      meanValues: meanValues,
+      audio: {
+        bitsPerSample: 16,
+        channels: channels,
+        data: audioData,
+      },
+    };
+  }
+
+  receive(e) {
+    const { event, id } = e.data;
+    let receiptData = {};
+    switch (event) {
+      case 'start':
+        this.recording = true;
+        break;
+      case 'stop':
+        this.recording = false;
+        break;
+      case 'clear':
+        this.initialize();
+        break;
+      case 'export':
+        receiptData = this.export();
+        break;
+      case 'read':
+        receiptData = this.getValues();
+        break;
+      default:
+        break;
+    }
+    // Always send back receipt
+    this.port.postMessage({ event: 'receipt', id, data: receiptData });
+  }
+
+  sendChunk(chunk) {
+    const channels = this.readChannelData([chunk]);
+    const { float32Array, meanValues } = this.formatAudioData(channels);
+    const rawAudioData = this.floatTo16BitPCM(float32Array);
+    const monoAudioData = this.floatTo16BitPCM(meanValues);
+    this.port.postMessage({
+      event: 'chunk',
+      data: {
+        mono: monoAudioData,
+        raw: rawAudioData,
+      },
+    });
+  }
+
+  process(inputList, outputList, parameters) {
+    // Copy input to output (e.g. speakers)
+    // Note that this creates choppy sounds with Mac products
+    const sourceLimit = Math.min(inputList.length, outputList.length);
+    for (let inputNum = 0; inputNum < sourceLimit; inputNum++) {
+      const input = inputList[inputNum];
+      const output = outputList[inputNum];
+      const channelCount = Math.min(input.length, output.length);
+      for (let channelNum = 0; channelNum < channelCount; channelNum++) {
+        input[channelNum].forEach((sample, i) => {
+          output[channelNum][i] = sample;
+        });
+      }
+    }
+    const inputs = inputList[0];
+    // There's latency at the beginning of a stream before recording starts
+    // Make sure we actually receive audio data before we start storing chunks
+    let sliceIndex = 0;
+    if (!this.foundAudio) {
+      for (const channel of inputs) {
+        sliceIndex = 0; // reset for each channel
+        if (this.foundAudio) {
+          break;
+        }
+        if (channel) {
+          for (const value of channel) {
+            if (value !== 0) {
+              // find only one non-zero entry in any channel
+              this.foundAudio = true;
+              break;
+            } else {
+              sliceIndex++;
+            }
+          }
+        }
+      }
+    }
+    if (inputs && inputs[0] && this.foundAudio && this.recording) {
+      // We need to copy the TypedArray, because the \`process\`
+      // internals will reuse the same buffer to hold each input
+      const chunk = inputs.map((input) => input.slice(sliceIndex));
+      this.chunks.push(chunk);
+      this.sendChunk(chunk);
+    }
+    return true;
+  }
+}
+
+registerProcessor('audio_processor', AudioProcessor);
+`;
+
+const script = new Blob([AudioProcessorWorklet], {
+  type: 'application/javascript',
+});
+const src = URL.createObjectURL(script);
+const AudioProcessorSrc = src;
+
+/**
+ * Decodes audio into a wav file
+ * @typedef {Object} DecodedAudioType
+ * @property {Blob} blob
+ * @property {string} url
+ * @property {Float32Array} values
+ * @property {AudioBuffer} audioBuffer
+ */
+
+/**
+ * Records live stream of user audio as PCM16 "audio/wav" data
+ * @class
+ */
+class WavRecorder {
+  /**
+   * Create a new WavRecorder instance
+   * @param {{sampleRate?: number, outputToSpeakers?: boolean, debug?: boolean}} [options]
+   * @returns {WavRecorder}
+   */
+  constructor({
+    sampleRate = 24000,
+    outputToSpeakers = false,
+    debug = false,
+  } = {}) {
+    // Script source
+    this.scriptSrc = AudioProcessorSrc;
+    // Config
+    this.sampleRate = sampleRate;
+    this.outputToSpeakers = outputToSpeakers;
+    this.debug = !!debug;
+    this._deviceChangeCallback = null;
+    this._devices = [];
+    // State variables
+    this.stream = null;
+    this.processor = null;
+    this.source = null;
+    this.node = null;
+    this.recording = false;
+    // Event handling with AudioWorklet
+    this._lastEventId = 0;
+    this.eventReceipts = {};
+    this.eventTimeout = 5000;
+    // Process chunks of audio
+    this._chunkProcessor = () => {};
+    this._chunkProcessorSize = void 0;
+    this._chunkProcessorBuffer = {
+      raw: new ArrayBuffer(0),
+      mono: new ArrayBuffer(0),
+    };
+  }
+
+  /**
+   * Decodes audio data from multiple formats to a Blob, url, Float32Array and AudioBuffer
+   * @param {Blob|Float32Array|Int16Array|ArrayBuffer|number[]} audioData
+   * @param {number} sampleRate
+   * @param {number} fromSampleRate
+   * @returns {Promise<DecodedAudioType>}
+   */
+  static async decode(audioData, sampleRate = 24000, fromSampleRate = -1) {
+    const context = new AudioContext({ sampleRate });
+    let arrayBuffer;
+    let blob;
+    if (audioData instanceof Blob) {
+      if (fromSampleRate !== -1) {
+        throw new Error(
+          `Can not specify "fromSampleRate" when reading from Blob`,
+        );
+      }
+      blob = audioData;
+      arrayBuffer = await blob.arrayBuffer();
+    } else if (audioData instanceof ArrayBuffer) {
+      if (fromSampleRate !== -1) {
+        throw new Error(
+          `Can not specify "fromSampleRate" when reading from ArrayBuffer`,
+        );
+      }
+      arrayBuffer = audioData;
+      blob = new Blob([arrayBuffer], { type: 'audio/wav' });
+    } else {
+      let float32Array;
+      let data;
+      if (audioData instanceof Int16Array) {
+        data = audioData;
+        float32Array = new Float32Array(audioData.length);
+        for (let i = 0; i < audioData.length; i++) {
+          float32Array[i] = audioData[i] / 0x8000;
+        }
+      } else if (audioData instanceof Float32Array) {
+        float32Array = audioData;
+      } else if (audioData instanceof Array) {
+        float32Array = new Float32Array(audioData);
+      } else {
+        throw new Error(
+          `"audioData" must be one of: Blob, Float32Arrray, Int16Array, ArrayBuffer, Array<number>`,
+        );
+      }
+      if (fromSampleRate === -1) {
+        throw new Error(
+          `Must specify "fromSampleRate" when reading from Float32Array, In16Array or Array`,
+        );
+      } else if (fromSampleRate < 3000) {
+        throw new Error(`Minimum "fromSampleRate" is 3000 (3kHz)`);
+      }
+      if (!data) {
+        data = WavPacker.floatTo16BitPCM(float32Array);
+      }
+      const audio = {
+        bitsPerSample: 16,
+        channels: [float32Array],
+        data,
+      };
+      const packer = new WavPacker();
+      const result = packer.pack(fromSampleRate, audio);
+      blob = result.blob;
+      arrayBuffer = await blob.arrayBuffer();
+    }
+    const audioBuffer = await context.decodeAudioData(arrayBuffer);
+    const values = audioBuffer.getChannelData(0);
+    const url = URL.createObjectURL(blob);
+    return {
+      blob,
+      url,
+      values,
+      audioBuffer,
+    };
+  }
+
+  /**
+   * Logs data in debug mode
+   * @param {...any} arguments
+   * @returns {true}
+   */
+  log() {
+    if (this.debug) {
+      this.log(...arguments);
+    }
+    return true;
+  }
+
+  /**
+   * Retrieves the current sampleRate for the recorder
+   * @returns {number}
+   */
+  getSampleRate() {
+    return this.sampleRate;
+  }
+
+  /**
+   * Retrieves the current status of the recording
+   * @returns {"ended"|"paused"|"recording"}
+   */
+  getStatus() {
+    if (!this.processor) {
+      return 'ended';
+    } else if (!this.recording) {
+      return 'paused';
+    } else {
+      return 'recording';
+    }
+  }
+
+  /**
+   * Sends an event to the AudioWorklet
+   * @private
+   * @param {string} name
+   * @param {{[key: string]: any}} data
+   * @param {AudioWorkletNode} [_processor]
+   * @returns {Promise<{[key: string]: any}>}
+   */
+  async _event(name, data = {}, _processor = null) {
+    _processor = _processor || this.processor;
+    if (!_processor) {
+      throw new Error('Can not send events without recording first');
+    }
+    const message = {
+      event: name,
+      id: this._lastEventId++,
+      data,
+    };
+    _processor.port.postMessage(message);
+    const t0 = new Date().valueOf();
+    while (!this.eventReceipts[message.id]) {
+      if (new Date().valueOf() - t0 > this.eventTimeout) {
+        throw new Error(`Timeout waiting for "${name}" event`);
+      }
+      await new Promise((res) => setTimeout(() => res(true), 1));
+    }
+    const payload = this.eventReceipts[message.id];
+    delete this.eventReceipts[message.id];
+    return payload;
+  }
+
+  /**
+   * Sets device change callback, remove if callback provided is `null`
+   * @param {(Array<MediaDeviceInfo & {default: boolean}>): void|null} callback
+   * @returns {true}
+   */
+  listenForDeviceChange(callback) {
+    if (callback === null && this._deviceChangeCallback) {
+      navigator.mediaDevices.removeEventListener(
+        'devicechange',
+        this._deviceChangeCallback,
+      );
+      this._deviceChangeCallback = null;
+    } else if (callback !== null) {
+      // Basically a debounce; we only want this called once when devices change
+      // And we only want the most recent callback() to be executed
+      // if a few are operating at the same time
+      let lastId = 0;
+      let lastDevices = [];
+      const serializeDevices = (devices) =>
+        devices
+          .map((d) => d.deviceId)
+          .sort()
+          .join(',');
+      const cb = async () => {
+        let id = ++lastId;
+        const devices = await this.listDevices();
+        if (id === lastId) {
+          if (serializeDevices(lastDevices) !== serializeDevices(devices)) {
+            lastDevices = devices;
+            callback(devices.slice());
+          }
+        }
+      };
+      navigator.mediaDevices.addEventListener('devicechange', cb);
+      cb();
+      this._deviceChangeCallback = cb;
+    }
+    return true;
+  }
+
+  /**
+   * Manually request permission to use the microphone
+   * @returns {Promise<true>}
+   */
+  async requestPermission() {
+    const permissionStatus = await navigator.permissions.query({
+      name: 'microphone',
+    });
+    if (permissionStatus.state === 'denied') {
+      window.alert('You must grant microphone access to use this feature.');
+    } else if (permissionStatus.state === 'prompt') {
+      try {
+        const stream = await navigator.mediaDevices.getUserMedia({
+          audio: true,
+        });
+        const tracks = stream.getTracks();
+        tracks.forEach((track) => track.stop());
+      } catch (e) {
+        window.alert('You must grant microphone access to use this feature.');
+      }
+    }
+    return true;
+  }
+
+  /**
+   * List all eligible devices for recording, will request permission to use microphone
+   * @returns {Promise<Array<MediaDeviceInfo & {default: boolean}>>}
+   */
+  async listDevices() {
+    if (
+      !navigator.mediaDevices ||
+      !('enumerateDevices' in navigator.mediaDevices)
+    ) {
+      throw new Error('Could not request user devices');
+    }
+    await this.requestPermission();
+    const devices = await navigator.mediaDevices.enumerateDevices();
+    const audioDevices = devices.filter(
+      (device) => device.kind === 'audioinput',
+    );
+    const defaultDeviceIndex = audioDevices.findIndex(
+      (device) => device.deviceId === 'default',
+    );
+    const deviceList = [];
+    if (defaultDeviceIndex !== -1) {
+      let defaultDevice = audioDevices.splice(defaultDeviceIndex, 1)[0];
+      let existingIndex = audioDevices.findIndex(
+        (device) => device.groupId === defaultDevice.groupId,
+      );
+      if (existingIndex !== -1) {
+        defaultDevice = audioDevices.splice(existingIndex, 1)[0];
+      }
+      defaultDevice.default = true;
+      deviceList.push(defaultDevice);
+    }
+    return deviceList.concat(audioDevices);
+  }
+
+  /**
+   * Begins a recording session and requests microphone permissions if not already granted
+   * Microphone recording indicator will appear on browser tab but status will be "paused"
+   * @param {string} [deviceId] if no device provided, default device will be used
+   * @returns {Promise<true>}
+   */
+  async begin(deviceId) {
+    if (this.processor) {
+      throw new Error(
+        `Already connected: please call .end() to start a new session`,
+      );
+    }
+
+    if (
+      !navigator.mediaDevices ||
+      !('getUserMedia' in navigator.mediaDevices)
+    ) {
+      throw new Error('Could not request user media');
+    }
+    try {
+      const config = { audio: true };
+      if (deviceId) {
+        config.audio = { deviceId: { exact: deviceId } };
+      }
+      this.stream = await navigator.mediaDevices.getUserMedia(config);
+    } catch (err) {
+      throw new Error('Could not start media stream');
+    }
+
+    const context = new AudioContext({ sampleRate: this.sampleRate });
+    const source = context.createMediaStreamSource(this.stream);
+    // Load and execute the module script.
+    try {
+      await context.audioWorklet.addModule(this.scriptSrc);
+    } catch (e) {
+      console.error(e);
+      throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);
+    }
+    const processor = new AudioWorkletNode(context, 'audio_processor');
+    processor.port.onmessage = (e) => {
+      const { event, id, data } = e.data;
+      if (event === 'receipt') {
+        this.eventReceipts[id] = data;
+      } else if (event === 'chunk') {
+        if (this._chunkProcessorSize) {
+          const buffer = this._chunkProcessorBuffer;
+          this._chunkProcessorBuffer = {
+            raw: WavPacker.mergeBuffers(buffer.raw, data.raw),
+            mono: WavPacker.mergeBuffers(buffer.mono, data.mono),
+          };
+          if (
+            this._chunkProcessorBuffer.mono.byteLength >=
+            this._chunkProcessorSize
+          ) {
+            this._chunkProcessor(this._chunkProcessorBuffer);
+            this._chunkProcessorBuffer = {
+              raw: new ArrayBuffer(0),
+              mono: new ArrayBuffer(0),
+            };
+          }
+        } else {
+          this._chunkProcessor(data);
+        }
+      }
+    };
+
+    const node = source.connect(processor);
+    const analyser = context.createAnalyser();
+    analyser.fftSize = 8192;
+    analyser.smoothingTimeConstant = 0.1;
+    node.connect(analyser);
+    if (this.outputToSpeakers) {
+      // eslint-disable-next-line no-console
+      console.warn(
+        'Warning: Output to speakers may affect sound quality,\n' +
+          'especially due to system audio feedback preventative measures.\n' +
+          'use only for debugging',
+      );
+      analyser.connect(context.destination);
+    }
+
+    this.source = source;
+    this.node = node;
+    this.analyser = analyser;
+    this.processor = processor;
+    return true;
+  }
+
+  /**
+   * Gets the current frequency domain data from the recording track
+   * @param {"frequency"|"music"|"voice"} [analysisType]
+   * @param {number} [minDecibels] default -100
+   * @param {number} [maxDecibels] default -30
+   * @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}
+   */
+  getFrequencies(
+    analysisType = 'frequency',
+    minDecibels = -100,
+    maxDecibels = -30,
+  ) {
+    if (!this.processor) {
+      throw new Error('Session ended: please call .begin() first');
+    }
+    return AudioAnalysis.getFrequencies(
+      this.analyser,
+      this.sampleRate,
+      null,
+      analysisType,
+      minDecibels,
+      maxDecibels,
+    );
+  }
+
+
+    /**
+   * Gets the real-time amplitude of the audio signal
+   * @returns {number} Amplitude value between 0 and 1
+   */
+  getAmplitude() {
+    if (!this.analyser) {
+      throw new Error('AnalyserNode is not initialized. Please call connect() first.');
+    }
+
+    const bufferLength = this.analyser.fftSize;
+    const dataArray = new Uint8Array(bufferLength);
+    this.analyser.getByteTimeDomainData(dataArray);
+
+    // Calculate RMS (Root Mean Square) to get amplitude
+    let sumSquares = 0;
+    for (let i = 0; i < bufferLength; i++) {
+      const normalized = (dataArray[i] - 128) / 128; // Normalize between -1 and 1
+      sumSquares += normalized * normalized;
+    }
+    const rms = Math.sqrt(sumSquares / bufferLength);
+    return rms;
+  }
+
+  /**
+   * Starts amplitude monitoring
+   * @param {function} callback - Function to call with amplitude value
+   */
+  startAmplitudeMonitoring(callback) {
+    const monitor = () => {
+      const amplitude = this.getAmplitude();
+      callback(amplitude);
+      requestAnimationFrame(monitor);
+    };
+    monitor();
+  }  
+
+  /**
+   * Pauses the recording
+   * Keeps microphone stream open but halts storage of audio
+   * @returns {Promise<true>}
+   */
+  async pause() {
+    if (!this.processor) {
+      throw new Error('Session ended: please call .begin() first');
+    } else if (!this.recording) {
+      throw new Error('Already paused: please call .record() first');
+    }
+    if (this._chunkProcessorBuffer.raw.byteLength) {
+      this._chunkProcessor(this._chunkProcessorBuffer);
+    }
+    this.log('Pausing ...');
+    await this._event('stop');
+    this.recording = false;
+    return true;
+  }
+
+  /**
+   * Start recording stream and storing to memory from the connected audio source
+   * @param {(data: { mono: Int16Array; raw: Int16Array }) => any} [chunkProcessor]
+   * @param {number} [chunkSize] chunkProcessor will not be triggered until this size threshold met in mono audio
+   * @returns {Promise<true>}
+   */
+  async record(chunkProcessor = () => {}, chunkSize = 8192) {
+    if (!this.processor) {
+      throw new Error('Session ended: please call .begin() first');
+    } else if (this.recording) {
+      throw new Error('Already recording: please call .pause() first');
+    } else if (typeof chunkProcessor !== 'function') {
+      throw new Error(`chunkProcessor must be a function`);
+    }
+    this._chunkProcessor = chunkProcessor;
+    this._chunkProcessorSize = chunkSize;
+    this._chunkProcessorBuffer = {
+      raw: new ArrayBuffer(0),
+      mono: new ArrayBuffer(0),
+    };
+    this.log('Recording ...');
+    await this._event('start');
+    this.recording = true;
+    return true;
+  }
+
+  /**
+   * Clears the audio buffer, empties stored recording
+   * @returns {Promise<true>}
+   */
+  async clear() {
+    if (!this.processor) {
+      throw new Error('Session ended: please call .begin() first');
+    }
+    await this._event('clear');
+    return true;
+  }
+
+  /**
+   * Reads the current audio stream data
+   * @returns {Promise<{meanValues: Float32Array, channels: Array<Float32Array>}>}
+   */
+  async read() {
+    if (!this.processor) {
+      throw new Error('Session ended: please call .begin() first');
+    }
+    this.log('Reading ...');
+    const result = await this._event('read');
+    return result;
+  }
+
+  /**
+   * Saves the current audio stream to a file
+   * @param {boolean} [force] Force saving while still recording
+   * @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}
+   */
+  async save(force = false) {
+    if (!this.processor) {
+      throw new Error('Session ended: please call .begin() first');
+    }
+    if (!force && this.recording) {
+      throw new Error(
+        'Currently recording: please call .pause() first, or call .save(true) to force',
+      );
+    }
+    this.log('Exporting ...');
+    const exportData = await this._event('export');
+    const packer = new WavPacker();
+    const result = packer.pack(this.sampleRate, exportData.audio);
+    return result;
+  }
+
+  /**
+   * Ends the current recording session and saves the result
+   * @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}
+   */
+  async end() {
+    if (!this.processor) {
+      throw new Error('Session ended: please call .begin() first');
+    }
+
+    const _processor = this.processor;
+
+    this.log('Stopping ...');
+    await this._event('stop');
+    this.recording = false;
+    const tracks = this.stream.getTracks();
+    tracks.forEach((track) => track.stop());
+
+    this.log('Exporting ...');
+    const exportData = await this._event('export', {}, _processor);
+
+    this.processor.disconnect();
+    this.source.disconnect();
+    this.node.disconnect();
+    this.analyser.disconnect();
+    this.stream = null;
+    this.processor = null;
+    this.source = null;
+    this.node = null;
+
+    const packer = new WavPacker();
+    const result = packer.pack(this.sampleRate, exportData.audio);
+    return result;
+  }
+
+  /**
+   * Performs a full cleanup of WavRecorder instance
+   * Stops actively listening via microphone and removes existing listeners
+   * @returns {Promise<true>}
+   */
+  async quit() {
+    this.listenForDeviceChange(null);
+    if (this.processor) {
+      await this.end();
+    }
+    return true;
+  }
+}
+
+globalThis.WavRecorder = WavRecorder;
+
+/**
+ * Converts a base64 string to an ArrayBuffer.
+ * @param {string} base64 - The base64 string to convert.
+ * @returns {ArrayBuffer} The resulting ArrayBuffer.
+ */
+function base64ToArrayBuffer(base64) {
+  const binaryString = atob(base64);
+  const len = binaryString.length;
+  const bytes = new Uint8Array(len);
+  for (let i = 0; i < len; i++) {
+    bytes[i] = binaryString.charCodeAt(i);
+  }
+  return bytes.buffer;
+}
+
+/**
+ * Converts an ArrayBuffer to a base64 string.
+ * @param {ArrayBuffer|Float32Array|Int16Array} arrayBuffer - The ArrayBuffer to convert.
+ * @returns {string} The resulting base64 string.
+ */
+function arrayBufferToBase64(arrayBuffer) {
+  if (arrayBuffer instanceof Float32Array) {
+    arrayBuffer = this.floatTo16BitPCM(arrayBuffer);
+  } else if (arrayBuffer instanceof Int16Array) {
+    arrayBuffer = arrayBuffer.buffer;
+  }
+  let binary = '';
+  let bytes = new Uint8Array(arrayBuffer);
+  const chunkSize = 0x8000; // 32KB chunk size
+  for (let i = 0; i < bytes.length; i += chunkSize) {
+    let chunk = bytes.subarray(i, i + chunkSize);
+    binary += String.fromCharCode.apply(null, chunk);
+  }
+  return btoa(binary);
+}
+
+/* eslint-env browser */
+/**
+ * @class LayercodeClient
+ * @classdesc Core client for Layercode audio pipeline that manages audio recording, WebSocket communication, and speech processing.
+ */
+class LayercodeClient {
+    // private currentTurnId: string | null = null;
+    // private lastDeltaIdPlayed: string | null = null;
+    /**
+     * Creates an instance of LayercodeClient.
+     * @param {Object} options - Configuration options
+     */
+    constructor(options) {
+        this.options = {
+            pipelineId: options.pipelineId,
+            sessionId: options.sessionId || null,
+            authorizeSessionEndpoint: options.authorizeSessionEndpoint,
+            metadata: options.metadata || {},
+            onConnect: options.onConnect || (() => { }),
+            onDisconnect: options.onDisconnect || (() => { }),
+            onError: options.onError || (() => { }),
+            onDataMessage: options.onDataMessage || (() => { }),
+            onUserAmplitudeChange: options.onUserAmplitudeChange || (() => { }),
+            onAgentAmplitudeChange: options.onAgentAmplitudeChange || (() => { }),
+            onStatusChange: options.onStatusChange || (() => { }),
+        };
+        this.AMPLITUDE_MONITORING_SAMPLE_RATE = 10;
+        this.websocketUrl = 'wss://api.layercode.com/v1/pipelines/websocket';
+        this.wavRecorder = new WavRecorder({ sampleRate: 8000 }); // TODO should be set my fetched pipeline config
+        this.wavPlayer = new WavStreamPlayer({
+            finishedPlayingCallback: this._clientResponseAudioReplayFinished.bind(this),
+            sampleRate: 16000, // TODO should be set my fetched pipeline config
+        });
+        this.ws = null;
+        this.status = 'disconnected';
+        this.userAudioAmplitude = 0;
+        this.agentAudioAmplitude = 0;
+        this.assistantIsSpeaking = false;
+        this.assistantHasBeenInterrupted = false;
+        this.sessionId = options.sessionId || null;
+        this.pushToTalkActive = false;
+        // this.currentTurnId = null; // TODO implement
+        // this.lastDeltaIdPlayed = null; // TODO implement
+        // Bind event handlers
+        this._handleWebSocketMessage = this._handleWebSocketMessage.bind(this);
+        this._handleDataAvailable = this._handleDataAvailable.bind(this);
+    }
+    /**
+     * Updates the connection status and triggers the callback
+     * @param {string} status - New status value
+     * @private
+     */
+    _setStatus(status) {
+        this.status = status;
+        this.options.onStatusChange(status);
+    }
+    /**
+     * Handles when agent audio finishes playing
+     * @private
+     */
+    _clientResponseAudioReplayFinished() {
+        this.assistantIsSpeaking = false;
+        this._wsSend({
+            type: 'trigger.response.audio.replay_finished',
+            reason: 'completed',
+            // last_delta_id_played: this.lastDeltaIdPlayed, // TODO implement
+            // turn_id: this.currentTurnId, // TODO implement
+        });
+    }
+    async _setupWavPlayer() {
+        this.wavPlayer = new WavStreamPlayer({
+            finishedPlayingCallback: this._clientResponseAudioReplayFinished.bind(this),
+            sampleRate: 16000, // TODO should be set my fetched pipeline config
+        });
+        await this.wavPlayer.connect();
+        // Set up amplitude monitoring only if callbacks are provided
+        if (this.options.onAgentAmplitudeChange !== (() => { })) {
+            let agentUpdateCounter = 0;
+            this.wavPlayer.startAmplitudeMonitoring((amplitude) => {
+                if (agentUpdateCounter == this.AMPLITUDE_MONITORING_SAMPLE_RATE) {
+                    this.agentAudioAmplitude = amplitude;
+                    this.options.onAgentAmplitudeChange(amplitude);
+                    agentUpdateCounter = 0; // Reset after each sample
+                }
+                agentUpdateCounter++;
+            });
+        }
+    }
+    async _clientInterruptAssistantReplay() {
+        if (this.assistantIsSpeaking) {
+            await this.wavPlayer.interrupt();
+            this.wavPlayer.disconnect();
+            await this._setupWavPlayer();
+            console.log('interrupting assistant replay');
+            console.log('setting assistantIsSpeaking to false');
+            console.log('setting assistantHasBeenInterrupted to true');
+            this.assistantIsSpeaking = false;
+            this.assistantHasBeenInterrupted = true;
+            this._wsSend({
+                type: 'trigger.response.audio.replay_finished',
+                reason: 'interrupted',
+                // last_delta_id_played: this.lastDeltaIdPlayed, // TODO implement
+                // turn_id: this.currentTurnId, // TODO implement
+            });
+        }
+    }
+    async triggerUserTurnStarted() {
+        if (!this.pushToTalkActive) {
+            this.pushToTalkActive = true;
+            this._wsSend({ type: 'trigger.turn.start', role: 'user' });
+            await this._clientInterruptAssistantReplay();
+        }
+    }
+    async triggerUserTurnFinished() {
+        if (this.pushToTalkActive) {
+            this.pushToTalkActive = false;
+            this._wsSend({ type: 'trigger.turn.end', role: 'user' });
+        }
+    }
+    /**
+     * Handles incoming WebSocket messages
+     * @param {MessageEvent} event - The WebSocket message event
+     * @private
+     */
+    async _handleWebSocketMessage(event) {
+        try {
+            const message = JSON.parse(event.data);
+            if (message.type !== 'response.audio') {
+                console.log('received ws msg:', message);
+            }
+            switch (message.type) {
+                case 'turn.start':
+                    // Sent from the server to this client when a new user turn is detected. The client should interrupt any playing assistant audio and send a "trigger.response.audio.replay_finished" event to the server.
+                    if (message.role === 'user') {
+                        // Interrupt any playing assistant audio
+                        // await this._clientInterruptAssistantReplay(); // TODO work whether to call interrupt or not here
+                    }
+                    break;
+                case 'response.audio':
+                    // console.log("received response.audio");
+                    if (!this.assistantHasBeenInterrupted) {
+                        // If the assistant has been interrupted, ignore the rest of the audio chunks for this response (turn)
+                        // TODO: scope audio chunks we ignore based on the turn_id
+                        if (!this.assistantIsSpeaking) {
+                            // If we have switch from assistant not speaking to now speaking, this is the start of the assistant's response
+                            this.assistantIsSpeaking = true;
+                            console.log('assistantIsSpeaking is currently false, received audio chunk, so setting to true');
+                        }
+                        console.log('adding audio chunk to wavPlayer');
+                        console.log('message.content length:', message.content.length);
+                        const audioBuffer = base64ToArrayBuffer(message.content);
+                        this.wavPlayer.add16BitPCM(audioBuffer, 'default');
+                    }
+                    else {
+                        console.log('ignoring response.audio because assistant has been interrupted');
+                    }
+                    break;
+                case 'response.end':
+                    console.log('received response.end');
+                    console.log('setting assistantHasBeenInterrupted to false');
+                    this.assistantHasBeenInterrupted = false;
+                    break;
+                case 'response.data':
+                    this.options.onDataMessage(message);
+                    break;
+                default:
+                    console.error('Unknown message type received:', message);
+                    break;
+            }
+        }
+        catch (error) {
+            console.error('Error processing WebSocket message:', error);
+            this.options.onError(error instanceof Error ? error : new Error(String(error)));
+        }
+    }
+    /**
+     * Handles available client browser microphone audio data and sends it over the WebSocket
+     * @param {ArrayBuffer} data - The audio data buffer
+     * @private
+     */
+    _handleDataAvailable(data) {
+        try {
+            const base64 = arrayBufferToBase64(data.mono);
+            this._wsSend({ type: 'client.audio', content: base64 });
+        }
+        catch (error) {
+            console.error('Error processing audio:', error);
+            this.options.onError(error instanceof Error ? error : new Error(String(error)));
+        }
+    }
+    _wsSend(message) {
+        var _a;
+        if (message.type !== 'client.audio') {
+            console.log('sent ws msg:', message);
+        }
+        const messageString = JSON.stringify(message);
+        if (((_a = this.ws) === null || _a === void 0 ? void 0 : _a.readyState) === WebSocket.OPEN) {
+            this.ws.send(messageString);
+        }
+    }
+    /**
+     * Connects to the Layercode pipeline and starts the audio session
+     * @async
+     * @returns {Promise<void>}
+     */
+    async connect() {
+        try {
+            this._setStatus('connecting');
+            // Get session key from server
+            let authorizeSessionRequestBody = {
+                pipeline_id: this.options.pipelineId,
+                metadata: this.options.metadata,
+            };
+            // If we're reconnecting to a previous session, we need to include the session_id in the request. Otherwise we don't send session_id, and a new session will be created and the session_id will be returned in the response.
+            if (this.options.sessionId) {
+                authorizeSessionRequestBody.session_id = this.options.sessionId;
+            }
+            const authorizeSessionResponse = await fetch(this.options.authorizeSessionEndpoint, {
+                method: 'POST',
+                headers: {
+                    'Content-Type': 'application/json',
+                },
+                body: JSON.stringify(authorizeSessionRequestBody),
+            });
+            if (!authorizeSessionResponse.ok) {
+                throw new Error(`Failed to authorize session: ${authorizeSessionResponse.statusText}`);
+            }
+            const authorizeSessionResponseBody = await authorizeSessionResponse.json();
+            this.sessionId = authorizeSessionResponseBody.session_id; // Save the session_id for use in future reconnects
+            // Connect WebSocket
+            this.ws = new WebSocket(`${this.websocketUrl}?${new URLSearchParams({
+                client_session_key: authorizeSessionResponseBody.client_session_key,
+            })}`);
+            this.ws.onopen = () => {
+                console.log('WebSocket connection established');
+                this._setStatus('connected');
+                this.options.onConnect({ sessionId: this.sessionId });
+            };
+            this.ws.onclose = () => {
+                console.log('WebSocket connection closed');
+                this._setStatus('disconnected');
+                this.options.onDisconnect();
+            };
+            this.ws.onerror = (error) => {
+                console.error('WebSocket error:', error);
+                this._setStatus('error');
+                this.options.onError(new Error('WebSocket connection error'));
+            };
+            this.ws.onmessage = this._handleWebSocketMessage;
+            // Initialize audio
+            await this.wavRecorder.begin();
+            await this.wavRecorder.record(this._handleDataAvailable);
+            if (this.options.onUserAmplitudeChange !== (() => { })) {
+                let userUpdateCounter = 0;
+                this.wavRecorder.startAmplitudeMonitoring((amplitude) => {
+                    if (userUpdateCounter == this.AMPLITUDE_MONITORING_SAMPLE_RATE) {
+                        this.userAudioAmplitude = amplitude;
+                        this.options.onUserAmplitudeChange(amplitude);
+                        userUpdateCounter = 0; // Reset after each sample
+                    }
+                    userUpdateCounter++;
+                });
+            }
+            await this._setupWavPlayer();
+            // Handle page unload
+            window.addEventListener('beforeunload', () => {
+                this.disconnect();
+            });
+        }
+        catch (error) {
+            console.error('Error connecting to Layercode pipeline:', error);
+            this._setStatus('error');
+            this.options.onError(error instanceof Error ? error : new Error(String(error)));
+            throw error;
+        }
+    }
+    /**
+     * Disconnects from the Layercode pipeline and stops audio recording
+     */
+    disconnect() {
+        console.log('disconnecting');
+        if (this.ws) {
+            this.ws.close();
+            this.ws = null;
+        }
+        // Stop recording user microphone audio
+        this.wavRecorder.stop();
+        // Handle wavPlayer cleanup without calling disconnect directly
+        if (this.wavPlayer) {
+            // Use type assertion to access internal properties
+            const player = this.wavPlayer;
+            // Clean up any audio resources manually
+            if (player.stream) {
+                player.stream.disconnect();
+                player.stream = null;
+            }
+            if (player.analyser) {
+                player.analyser.disconnect();
+            }
+            if (player.context) {
+                player.context.close().catch((err) => console.error('Error closing audio context:', err));
+            }
+        }
+        this._setStatus('disconnected');
+    }
+    /**
+     * Mutes or unmutes the microphone
+     * @param {boolean} mute - Whether to mute the microphone
+     */
+    setMuteMic(mute) {
+        if (mute) {
+            this.wavRecorder.mute();
+        }
+        else {
+            this.wavRecorder.unmute();
+        }
+    }
+}
+
+export { LayercodeClient as default };
+//# sourceMappingURL=layercode-js-sdk.esm.js.map