@layercode/js-sdk 1.0.6 → 1.0.7
This diff represents the content of publicly available package versions that have been released to one of the supported registries. The information contained in this diff is provided for informational purposes only and reflects changes between package versions as they appear in their respective public registries.
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{"version":3,"file":"layercode-js-sdk.esm.js","sources":["../src/wavtools/lib/wav_packer.js","../src/wavtools/lib/analysis/constants.js","../src/wavtools/lib/analysis/audio_analysis.js","../src/wavtools/lib/worklets/stream_processor.js","../src/wavtools/lib/wav_stream_player.js","../src/wavtools/lib/worklets/audio_processor.js","../src/wavtools/lib/wav_recorder.js","../src/utils.js","../src/index.ts"],"sourcesContent":["/**\n * Raw wav audio file contents\n * @typedef {Object} WavPackerAudioType\n * @property {Blob} blob\n * @property {string} url\n * @property {number} channelCount\n * @property {number} sampleRate\n * @property {number} duration\n */\n\n/**\n * Utility class for assembling PCM16 \"audio/wav\" data\n * @class\n */\nexport class WavPacker {\n /**\n * Converts Float32Array of amplitude data to ArrayBuffer in Int16Array format\n * @param {Float32Array} float32Array\n * @returns {ArrayBuffer}\n */\n static floatTo16BitPCM(float32Array) {\n const buffer = new ArrayBuffer(float32Array.length * 2);\n const view = new DataView(buffer);\n let offset = 0;\n for (let i = 0; i < float32Array.length; i++, offset += 2) {\n let s = Math.max(-1, Math.min(1, float32Array[i]));\n view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);\n }\n return buffer;\n }\n\n /**\n * Concatenates two ArrayBuffers\n * @param {ArrayBuffer} leftBuffer\n * @param {ArrayBuffer} rightBuffer\n * @returns {ArrayBuffer}\n */\n static mergeBuffers(leftBuffer, rightBuffer) {\n const tmpArray = new Uint8Array(\n leftBuffer.byteLength + rightBuffer.byteLength\n );\n tmpArray.set(new Uint8Array(leftBuffer), 0);\n tmpArray.set(new Uint8Array(rightBuffer), leftBuffer.byteLength);\n return tmpArray.buffer;\n }\n\n /**\n * Packs data into an Int16 format\n * @private\n * @param {number} size 0 = 1x Int16, 1 = 2x Int16\n * @param {number} arg value to pack\n * @returns\n */\n _packData(size, arg) {\n return [\n new Uint8Array([arg, arg >> 8]),\n new Uint8Array([arg, arg >> 8, arg >> 16, arg >> 24]),\n ][size];\n }\n\n /**\n * Packs audio into \"audio/wav\" Blob\n * @param {number} sampleRate\n * @param {{bitsPerSample: number, channels: Array<Float32Array>, data: Int16Array}} audio\n * @returns {WavPackerAudioType}\n */\n pack(sampleRate, audio) {\n if (!audio?.bitsPerSample) {\n throw new Error(`Missing \"bitsPerSample\"`);\n } else if (!audio?.channels) {\n throw new Error(`Missing \"channels\"`);\n } else if (!audio?.data) {\n throw new Error(`Missing \"data\"`);\n }\n const { bitsPerSample, channels, data } = audio;\n const output = [\n // Header\n 'RIFF',\n this._packData(\n 1,\n 4 + (8 + 24) /* chunk 1 length */ + (8 + 8) /* chunk 2 length */\n ), // Length\n 'WAVE',\n // chunk 1\n 'fmt ', // Sub-chunk identifier\n this._packData(1, 16), // Chunk length\n this._packData(0, 1), // Audio format (1 is linear quantization)\n this._packData(0, channels.length),\n this._packData(1, sampleRate),\n this._packData(1, (sampleRate * channels.length * bitsPerSample) / 8), // Byte rate\n this._packData(0, (channels.length * bitsPerSample) / 8),\n this._packData(0, bitsPerSample),\n // chunk 2\n 'data', // Sub-chunk identifier\n this._packData(\n 1,\n (channels[0].length * channels.length * bitsPerSample) / 8\n ), // Chunk length\n data,\n ];\n const blob = new Blob(output, { type: 'audio/mpeg' });\n const url = URL.createObjectURL(blob);\n return {\n blob,\n url,\n channelCount: channels.length,\n sampleRate,\n duration: data.byteLength / (channels.length * sampleRate * 2),\n };\n }\n}\n\nglobalThis.WavPacker = WavPacker;\n","/**\n * Constants for help with visualization\n * Helps map frequency ranges from Fast Fourier Transform\n * to human-interpretable ranges, notably music ranges and\n * human vocal ranges.\n */\n\n// Eighth octave frequencies\nconst octave8Frequencies = [\n 4186.01, 4434.92, 4698.63, 4978.03, 5274.04, 5587.65, 5919.91, 6271.93,\n 6644.88, 7040.0, 7458.62, 7902.13,\n];\n\n// Labels for each of the above frequencies\nconst octave8FrequencyLabels = [\n 'C',\n 'C#',\n 'D',\n 'D#',\n 'E',\n 'F',\n 'F#',\n 'G',\n 'G#',\n 'A',\n 'A#',\n 'B',\n];\n\n/**\n * All note frequencies from 1st to 8th octave\n * in format \"A#8\" (A#, 8th octave)\n */\nexport const noteFrequencies = [];\nexport const noteFrequencyLabels = [];\nfor (let i = 1; i <= 8; i++) {\n for (let f = 0; f < octave8Frequencies.length; f++) {\n const freq = octave8Frequencies[f];\n noteFrequencies.push(freq / Math.pow(2, 8 - i));\n noteFrequencyLabels.push(octave8FrequencyLabels[f] + i);\n }\n}\n\n/**\n * Subset of the note frequencies between 32 and 2000 Hz\n * 6 octave range: C1 to B6\n */\nconst voiceFrequencyRange = [32.0, 2000.0];\nexport const voiceFrequencies = noteFrequencies.filter((_, i) => {\n return (\n noteFrequencies[i] > voiceFrequencyRange[0] &&\n noteFrequencies[i] < voiceFrequencyRange[1]\n );\n});\nexport const voiceFrequencyLabels = noteFrequencyLabels.filter((_, i) => {\n return (\n noteFrequencies[i] > voiceFrequencyRange[0] &&\n noteFrequencies[i] < voiceFrequencyRange[1]\n );\n});\n","import {\n noteFrequencies,\n noteFrequencyLabels,\n voiceFrequencies,\n voiceFrequencyLabels,\n} from './constants.js';\n\n/**\n * Output of AudioAnalysis for the frequency domain of the audio\n * @typedef {Object} AudioAnalysisOutputType\n * @property {Float32Array} values Amplitude of this frequency between {0, 1} inclusive\n * @property {number[]} frequencies Raw frequency bucket values\n * @property {string[]} labels Labels for the frequency bucket values\n */\n\n/**\n * Analyzes audio for visual output\n * @class\n */\nexport class AudioAnalysis {\n /**\n * Retrieves frequency domain data from an AnalyserNode adjusted to a decibel range\n * returns human-readable formatting and labels\n * @param {AnalyserNode} analyser\n * @param {number} sampleRate\n * @param {Float32Array} [fftResult]\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {AudioAnalysisOutputType}\n */\n static getFrequencies(\n analyser,\n sampleRate,\n fftResult,\n analysisType = 'frequency',\n minDecibels = -100,\n maxDecibels = -30,\n ) {\n if (!fftResult) {\n fftResult = new Float32Array(analyser.frequencyBinCount);\n analyser.getFloatFrequencyData(fftResult);\n }\n const nyquistFrequency = sampleRate / 2;\n const frequencyStep = (1 / fftResult.length) * nyquistFrequency;\n let outputValues;\n let frequencies;\n let labels;\n if (analysisType === 'music' || analysisType === 'voice') {\n const useFrequencies =\n analysisType === 'voice' ? voiceFrequencies : noteFrequencies;\n const aggregateOutput = Array(useFrequencies.length).fill(minDecibels);\n for (let i = 0; i < fftResult.length; i++) {\n const frequency = i * frequencyStep;\n const amplitude = fftResult[i];\n for (let n = useFrequencies.length - 1; n >= 0; n--) {\n if (frequency > useFrequencies[n]) {\n aggregateOutput[n] = Math.max(aggregateOutput[n], amplitude);\n break;\n }\n }\n }\n outputValues = aggregateOutput;\n frequencies =\n analysisType === 'voice' ? voiceFrequencies : noteFrequencies;\n labels =\n analysisType === 'voice' ? voiceFrequencyLabels : noteFrequencyLabels;\n } else {\n outputValues = Array.from(fftResult);\n frequencies = outputValues.map((_, i) => frequencyStep * i);\n labels = frequencies.map((f) => `${f.toFixed(2)} Hz`);\n }\n // We normalize to {0, 1}\n const normalizedOutput = outputValues.map((v) => {\n return Math.max(\n 0,\n Math.min((v - minDecibels) / (maxDecibels - minDecibels), 1),\n );\n });\n const values = new Float32Array(normalizedOutput);\n return {\n values,\n frequencies,\n labels,\n };\n }\n\n /**\n * Creates a new AudioAnalysis instance for an HTMLAudioElement\n * @param {HTMLAudioElement} audioElement\n * @param {AudioBuffer|null} [audioBuffer] If provided, will cache all frequency domain data from the buffer\n * @returns {AudioAnalysis}\n */\n constructor(audioElement, audioBuffer = null) {\n this.fftResults = [];\n if (audioBuffer) {\n /**\n * Modified from\n * https://stackoverflow.com/questions/75063715/using-the-web-audio-api-to-analyze-a-song-without-playing\n *\n * We do this to populate FFT values for the audio if provided an `audioBuffer`\n * The reason to do this is that Safari fails when using `createMediaElementSource`\n * This has a non-zero RAM cost so we only opt-in to run it on Safari, Chrome is better\n */\n const { length, sampleRate } = audioBuffer;\n const offlineAudioContext = new OfflineAudioContext({\n length,\n sampleRate,\n });\n const source = offlineAudioContext.createBufferSource();\n source.buffer = audioBuffer;\n const analyser = offlineAudioContext.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n source.connect(analyser);\n // limit is :: 128 / sampleRate;\n // but we just want 60fps - cuts ~1s from 6MB to 1MB of RAM\n const renderQuantumInSeconds = 1 / 60;\n const durationInSeconds = length / sampleRate;\n const analyze = (index) => {\n const suspendTime = renderQuantumInSeconds * index;\n if (suspendTime < durationInSeconds) {\n offlineAudioContext.suspend(suspendTime).then(() => {\n const fftResult = new Float32Array(analyser.frequencyBinCount);\n analyser.getFloatFrequencyData(fftResult);\n this.fftResults.push(fftResult);\n analyze(index + 1);\n });\n }\n if (index === 1) {\n offlineAudioContext.startRendering();\n } else {\n offlineAudioContext.resume();\n }\n };\n source.start(0);\n analyze(1);\n this.audio = audioElement;\n this.context = offlineAudioContext;\n this.analyser = analyser;\n this.sampleRate = sampleRate;\n this.audioBuffer = audioBuffer;\n } else {\n const audioContext = new AudioContext();\n const track = audioContext.createMediaElementSource(audioElement);\n const analyser = audioContext.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n track.connect(analyser);\n analyser.connect(audioContext.destination);\n this.audio = audioElement;\n this.context = audioContext;\n this.analyser = analyser;\n this.sampleRate = this.context.sampleRate;\n this.audioBuffer = null;\n }\n }\n\n /**\n * Gets the current frequency domain data from the playing audio track\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {AudioAnalysisOutputType}\n */\n getFrequencies(\n analysisType = 'frequency',\n minDecibels = -100,\n maxDecibels = -30,\n ) {\n let fftResult = null;\n if (this.audioBuffer && this.fftResults.length) {\n const pct = this.audio.currentTime / this.audio.duration;\n const index = Math.min(\n (pct * this.fftResults.length) | 0,\n this.fftResults.length - 1,\n );\n fftResult = this.fftResults[index];\n }\n return AudioAnalysis.getFrequencies(\n this.analyser,\n this.sampleRate,\n fftResult,\n analysisType,\n minDecibels,\n maxDecibels,\n );\n }\n\n /**\n * Resume the internal AudioContext if it was suspended due to the lack of\n * user interaction when the AudioAnalysis was instantiated.\n * @returns {Promise<true>}\n */\n async resumeIfSuspended() {\n if (this.context.state === 'suspended') {\n await this.context.resume();\n }\n return true;\n }\n}\n\nglobalThis.AudioAnalysis = AudioAnalysis;\n","export const StreamProcessorWorklet = `\nclass StreamProcessor extends AudioWorkletProcessor {\n constructor() {\n super();\n this.hasStarted = false;\n this.hasInterrupted = false;\n this.outputBuffers = [];\n this.bufferLength = 128;\n this.write = { buffer: new Float32Array(this.bufferLength), trackId: null };\n this.writeOffset = 0;\n this.trackSampleOffsets = {};\n this.port.onmessage = (event) => {\n if (event.data) {\n const payload = event.data;\n if (payload.event === 'write') {\n const int16Array = payload.buffer;\n const float32Array = new Float32Array(int16Array.length);\n for (let i = 0; i < int16Array.length; i++) {\n float32Array[i] = int16Array[i] / 0x8000; // Convert Int16 to Float32\n }\n this.writeData(float32Array, payload.trackId);\n } else if (\n payload.event === 'offset' ||\n payload.event === 'interrupt'\n ) {\n const requestId = payload.requestId;\n const trackId = this.write.trackId;\n const offset = this.trackSampleOffsets[trackId] || 0;\n this.port.postMessage({\n event: 'offset',\n requestId,\n trackId,\n offset,\n });\n if (payload.event === 'interrupt') {\n this.hasInterrupted = true;\n }\n } else {\n throw new Error(\\`Unhandled event \"\\${payload.event}\"\\`);\n }\n }\n };\n }\n\n writeData(float32Array, trackId = null) {\n let { buffer } = this.write;\n let offset = this.writeOffset;\n for (let i = 0; i < float32Array.length; i++) {\n buffer[offset++] = float32Array[i];\n if (offset >= buffer.length) {\n this.outputBuffers.push(this.write);\n this.write = { buffer: new Float32Array(this.bufferLength), trackId };\n buffer = this.write.buffer;\n offset = 0;\n }\n }\n this.writeOffset = offset;\n return true;\n }\n\n process(inputs, outputs, parameters) {\n const output = outputs[0];\n const outputChannelData = output[0];\n const outputBuffers = this.outputBuffers;\n if (this.hasInterrupted) {\n this.port.postMessage({ event: 'stop' });\n return false;\n } else if (outputBuffers.length) {\n this.hasStarted = true;\n const { buffer, trackId } = outputBuffers.shift();\n for (let i = 0; i < outputChannelData.length; i++) {\n outputChannelData[i] = buffer[i] || 0;\n }\n if (trackId) {\n this.trackSampleOffsets[trackId] =\n this.trackSampleOffsets[trackId] || 0;\n this.trackSampleOffsets[trackId] += buffer.length;\n }\n return true;\n } else if (this.hasStarted) {\n this.port.postMessage({ event: 'stop' });\n return false;\n } else {\n return true;\n }\n }\n}\n\nregisterProcessor('stream_processor', StreamProcessor);\n`;\n\nconst script = new Blob([StreamProcessorWorklet], {\n type: 'application/javascript',\n});\nconst src = URL.createObjectURL(script);\nexport const StreamProcessorSrc = src;\n","import { StreamProcessorSrc } from \"./worklets/stream_processor.js\";\nimport { AudioAnalysis } from \"./analysis/audio_analysis.js\";\n\n/**\n * Plays audio streams received in raw PCM16 chunks from the browser\n * @class\n */\nexport class WavStreamPlayer {\n /**\n * Creates a new WavStreamPlayer instance\n * @param {{sampleRate?: number}} options\n * @returns {WavStreamPlayer}\n */\n constructor({ finishedPlayingCallback = () => {}, sampleRate = 24000 } = {}) {\n this.scriptSrc = StreamProcessorSrc;\n this.sampleRate = sampleRate;\n this.context = null;\n this.stream = null;\n this.analyser = null;\n this.trackSampleOffsets = {};\n this.interruptedTrackIds = {};\n this.finishedPlayingCallback = finishedPlayingCallback;\n }\n\n /**\n * Connects the audio context and enables output to speakers\n * @returns {Promise<true>}\n */\n async connect() {\n this.context = new AudioContext({ sampleRate: this.sampleRate });\n if (this.context.state === \"suspended\") {\n await this.context.resume();\n }\n try {\n await this.context.audioWorklet.addModule(this.scriptSrc);\n } catch (e) {\n console.error(e);\n throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);\n }\n const analyser = this.context.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n this.analyser = analyser;\n return true;\n }\n\n /**\n * Gets the current frequency domain data from the playing track\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}\n */\n getFrequencies(analysisType = \"frequency\", minDecibels = -100, maxDecibels = -30) {\n if (!this.analyser) {\n throw new Error(\"Not connected, please call .connect() first\");\n }\n return AudioAnalysis.getFrequencies(this.analyser, this.sampleRate, null, analysisType, minDecibels, maxDecibels);\n }\n\n /**\n * Gets the real-time amplitude of the audio signal\n * @returns {number} Amplitude value between 0 and 1\n */\n getAmplitude() {\n if (!this.analyser) {\n throw new Error(\"AnalyserNode is not initialized. Please call connect() first.\");\n }\n\n const bufferLength = this.analyser.fftSize;\n const dataArray = new Uint8Array(bufferLength);\n this.analyser.getByteTimeDomainData(dataArray);\n\n // Calculate RMS (Root Mean Square) to get amplitude\n let sumSquares = 0;\n for (let i = 0; i < bufferLength; i++) {\n const normalized = (dataArray[i] - 128) / 128; // Normalize between -1 and 1\n sumSquares += normalized * normalized;\n }\n const rms = Math.sqrt(sumSquares / bufferLength);\n return rms;\n }\n\n /**\n * Starts amplitude monitoring\n * @param {function} callback - Function to call with amplitude value\n */\n startAmplitudeMonitoring(callback) {\n const monitor = () => {\n const amplitude = this.getAmplitude();\n callback(amplitude);\n requestAnimationFrame(monitor);\n };\n monitor();\n }\n\n /**\n * Starts audio streaming\n * @private\n * @returns {Promise<true>}\n */\n _start() {\n const streamNode = new AudioWorkletNode(this.context, \"stream_processor\");\n streamNode.connect(this.context.destination);\n streamNode.port.onmessage = (e) => {\n const { event } = e.data;\n if (event === \"stop\") {\n streamNode.disconnect();\n this.stream = null;\n this.finishedPlayingCallback();\n } else if (event === \"offset\") {\n const { requestId, trackId, offset } = e.data;\n const currentTime = offset / this.sampleRate;\n this.trackSampleOffsets[requestId] = { trackId, offset, currentTime };\n }\n };\n this.analyser.disconnect();\n streamNode.connect(this.analyser);\n this.stream = streamNode;\n return true;\n }\n\n /**\n * Adds 16BitPCM data to the currently playing audio stream\n * You can add chunks beyond the current play point and they will be queued for play\n * @param {ArrayBuffer|Int16Array} arrayBuffer\n * @param {string} [trackId]\n * @returns {Int16Array}\n */\n add16BitPCM(arrayBuffer, trackId = \"default\") {\n if (typeof trackId !== \"string\") {\n throw new Error(`trackId must be a string`);\n } else if (this.interruptedTrackIds[trackId]) {\n return;\n }\n if (!this.stream) {\n this._start();\n }\n let buffer;\n if (arrayBuffer instanceof Int16Array) {\n buffer = arrayBuffer;\n } else if (arrayBuffer instanceof ArrayBuffer) {\n buffer = new Int16Array(arrayBuffer);\n } else {\n throw new Error(`argument must be Int16Array or ArrayBuffer`);\n }\n this.stream.port.postMessage({ event: \"write\", buffer, trackId });\n return buffer;\n }\n\n /**\n * Gets the offset (sample count) of the currently playing stream\n * @param {boolean} [interrupt]\n * @returns {{trackId: string|null, offset: number, currentTime: number}}\n */\n async getTrackSampleOffset(interrupt = false) {\n if (!this.stream) {\n return null;\n }\n const requestId = crypto.randomUUID();\n this.stream.port.postMessage({\n event: interrupt ? \"interrupt\" : \"offset\",\n requestId,\n });\n let trackSampleOffset;\n while (!trackSampleOffset) {\n trackSampleOffset = this.trackSampleOffsets[requestId];\n await new Promise((r) => setTimeout(() => r(), 1));\n }\n const { trackId } = trackSampleOffset;\n if (interrupt && trackId) {\n this.interruptedTrackIds[trackId] = true;\n }\n return trackSampleOffset;\n }\n\n /**\n * Strips the current stream and returns the sample offset of the audio\n * @param {boolean} [interrupt]\n * @returns {{trackId: string|null, offset: number, currentTime: number}}\n */\n async interrupt() {\n return this.getTrackSampleOffset(true);\n }\n\n /**\n * Disconnects the audio context and cleans up resources\n * @returns {void}\n */\n disconnect() {\n if (this.stream) {\n this.stream.disconnect();\n this.stream = null;\n }\n\n if (this.analyser) {\n this.analyser.disconnect();\n }\n\n if (this.context) {\n this.context.close().catch((err) => console.error(\"Error closing audio context:\", err));\n }\n }\n}\n\nglobalThis.WavStreamPlayer = WavStreamPlayer;\n","const AudioProcessorWorklet = `\nclass AudioProcessor extends AudioWorkletProcessor {\n\n constructor() {\n super();\n this.port.onmessage = this.receive.bind(this);\n this.initialize();\n }\n\n initialize() {\n this.foundAudio = false;\n this.recording = false;\n this.chunks = [];\n }\n\n /**\n * Concatenates sampled chunks into channels\n * Format is chunk[Left[], Right[]]\n */\n readChannelData(chunks, channel = -1, maxChannels = 9) {\n let channelLimit;\n if (channel !== -1) {\n if (chunks[0] && chunks[0].length - 1 < channel) {\n throw new Error(\n \\`Channel \\${channel} out of range: max \\${chunks[0].length}\\`\n );\n }\n channelLimit = channel + 1;\n } else {\n channel = 0;\n channelLimit = Math.min(chunks[0] ? chunks[0].length : 1, maxChannels);\n }\n const channels = [];\n for (let n = channel; n < channelLimit; n++) {\n const length = chunks.reduce((sum, chunk) => {\n return sum + chunk[n].length;\n }, 0);\n const buffers = chunks.map((chunk) => chunk[n]);\n const result = new Float32Array(length);\n let offset = 0;\n for (let i = 0; i < buffers.length; i++) {\n result.set(buffers[i], offset);\n offset += buffers[i].length;\n }\n channels[n] = result;\n }\n return channels;\n }\n\n /**\n * Combines parallel audio data into correct format,\n * channels[Left[], Right[]] to float32Array[LRLRLRLR...]\n */\n formatAudioData(channels) {\n if (channels.length === 1) {\n // Simple case is only one channel\n const float32Array = channels[0].slice();\n const meanValues = channels[0].slice();\n return { float32Array, meanValues };\n } else {\n const float32Array = new Float32Array(\n channels[0].length * channels.length\n );\n const meanValues = new Float32Array(channels[0].length);\n for (let i = 0; i < channels[0].length; i++) {\n const offset = i * channels.length;\n let meanValue = 0;\n for (let n = 0; n < channels.length; n++) {\n float32Array[offset + n] = channels[n][i];\n meanValue += channels[n][i];\n }\n meanValues[i] = meanValue / channels.length;\n }\n return { float32Array, meanValues };\n }\n }\n\n /**\n * Converts 32-bit float data to 16-bit integers\n */\n floatTo16BitPCM(float32Array) {\n const buffer = new ArrayBuffer(float32Array.length * 2);\n const view = new DataView(buffer);\n let offset = 0;\n for (let i = 0; i < float32Array.length; i++, offset += 2) {\n let s = Math.max(-1, Math.min(1, float32Array[i]));\n view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);\n }\n return buffer;\n }\n\n /**\n * Retrieves the most recent amplitude values from the audio stream\n * @param {number} channel\n */\n getValues(channel = -1) {\n const channels = this.readChannelData(this.chunks, channel);\n const { meanValues } = this.formatAudioData(channels);\n return { meanValues, channels };\n }\n\n /**\n * Exports chunks as an audio/wav file\n */\n export() {\n const channels = this.readChannelData(this.chunks);\n const { float32Array, meanValues } = this.formatAudioData(channels);\n const audioData = this.floatTo16BitPCM(float32Array);\n return {\n meanValues: meanValues,\n audio: {\n bitsPerSample: 16,\n channels: channels,\n data: audioData,\n },\n };\n }\n\n receive(e) {\n const { event, id } = e.data;\n let receiptData = {};\n switch (event) {\n case 'start':\n this.recording = true;\n break;\n case 'stop':\n this.recording = false;\n break;\n case 'clear':\n this.initialize();\n break;\n case 'export':\n receiptData = this.export();\n break;\n case 'read':\n receiptData = this.getValues();\n break;\n default:\n break;\n }\n // Always send back receipt\n this.port.postMessage({ event: 'receipt', id, data: receiptData });\n }\n\n sendChunk(chunk) {\n const channels = this.readChannelData([chunk]);\n const { float32Array, meanValues } = this.formatAudioData(channels);\n const rawAudioData = this.floatTo16BitPCM(float32Array);\n const monoAudioData = this.floatTo16BitPCM(meanValues);\n this.port.postMessage({\n event: 'chunk',\n data: {\n mono: monoAudioData,\n raw: rawAudioData,\n },\n });\n }\n\n process(inputList, outputList, parameters) {\n // Copy input to output (e.g. speakers)\n // Note that this creates choppy sounds with Mac products\n const sourceLimit = Math.min(inputList.length, outputList.length);\n for (let inputNum = 0; inputNum < sourceLimit; inputNum++) {\n const input = inputList[inputNum];\n const output = outputList[inputNum];\n const channelCount = Math.min(input.length, output.length);\n for (let channelNum = 0; channelNum < channelCount; channelNum++) {\n input[channelNum].forEach((sample, i) => {\n output[channelNum][i] = sample;\n });\n }\n }\n const inputs = inputList[0];\n // There's latency at the beginning of a stream before recording starts\n // Make sure we actually receive audio data before we start storing chunks\n let sliceIndex = 0;\n if (!this.foundAudio) {\n for (const channel of inputs) {\n sliceIndex = 0; // reset for each channel\n if (this.foundAudio) {\n break;\n }\n if (channel) {\n for (const value of channel) {\n if (value !== 0) {\n // find only one non-zero entry in any channel\n this.foundAudio = true;\n break;\n } else {\n sliceIndex++;\n }\n }\n }\n }\n }\n if (inputs && inputs[0] && this.foundAudio && this.recording) {\n // We need to copy the TypedArray, because the \\`process\\`\n // internals will reuse the same buffer to hold each input\n const chunk = inputs.map((input) => input.slice(sliceIndex));\n this.chunks.push(chunk);\n this.sendChunk(chunk);\n }\n return true;\n }\n}\n\nregisterProcessor('audio_processor', AudioProcessor);\n`;\n\nconst script = new Blob([AudioProcessorWorklet], {\n type: 'application/javascript',\n});\nconst src = URL.createObjectURL(script);\nexport const AudioProcessorSrc = src;\n","import { AudioProcessorSrc } from './worklets/audio_processor.js';\nimport { AudioAnalysis } from './analysis/audio_analysis.js';\nimport { WavPacker } from './wav_packer.js';\n\n/**\n * Decodes audio into a wav file\n * @typedef {Object} DecodedAudioType\n * @property {Blob} blob\n * @property {string} url\n * @property {Float32Array} values\n * @property {AudioBuffer} audioBuffer\n */\n\n/**\n * Records live stream of user audio as PCM16 \"audio/wav\" data\n * @class\n */\nexport class WavRecorder {\n /**\n * Create a new WavRecorder instance\n * @param {{sampleRate?: number, outputToSpeakers?: boolean, debug?: boolean}} [options]\n * @returns {WavRecorder}\n */\n constructor({\n sampleRate = 24000,\n outputToSpeakers = false,\n debug = false,\n } = {}) {\n // Script source\n this.scriptSrc = AudioProcessorSrc;\n // Config\n this.sampleRate = sampleRate;\n this.outputToSpeakers = outputToSpeakers;\n this.debug = !!debug;\n this._deviceChangeCallback = null;\n this._devices = [];\n // State variables\n this.stream = null;\n this.processor = null;\n this.source = null;\n this.node = null;\n this.recording = false;\n // Event handling with AudioWorklet\n this._lastEventId = 0;\n this.eventReceipts = {};\n this.eventTimeout = 5000;\n // Process chunks of audio\n this._chunkProcessor = () => {};\n this._chunkProcessorSize = void 0;\n this._chunkProcessorBuffer = {\n raw: new ArrayBuffer(0),\n mono: new ArrayBuffer(0),\n };\n }\n\n /**\n * Decodes audio data from multiple formats to a Blob, url, Float32Array and AudioBuffer\n * @param {Blob|Float32Array|Int16Array|ArrayBuffer|number[]} audioData\n * @param {number} sampleRate\n * @param {number} fromSampleRate\n * @returns {Promise<DecodedAudioType>}\n */\n static async decode(audioData, sampleRate = 24000, fromSampleRate = -1) {\n const context = new AudioContext({ sampleRate });\n let arrayBuffer;\n let blob;\n if (audioData instanceof Blob) {\n if (fromSampleRate !== -1) {\n throw new Error(\n `Can not specify \"fromSampleRate\" when reading from Blob`,\n );\n }\n blob = audioData;\n arrayBuffer = await blob.arrayBuffer();\n } else if (audioData instanceof ArrayBuffer) {\n if (fromSampleRate !== -1) {\n throw new Error(\n `Can not specify \"fromSampleRate\" when reading from ArrayBuffer`,\n );\n }\n arrayBuffer = audioData;\n blob = new Blob([arrayBuffer], { type: 'audio/wav' });\n } else {\n let float32Array;\n let data;\n if (audioData instanceof Int16Array) {\n data = audioData;\n float32Array = new Float32Array(audioData.length);\n for (let i = 0; i < audioData.length; i++) {\n float32Array[i] = audioData[i] / 0x8000;\n }\n } else if (audioData instanceof Float32Array) {\n float32Array = audioData;\n } else if (audioData instanceof Array) {\n float32Array = new Float32Array(audioData);\n } else {\n throw new Error(\n `\"audioData\" must be one of: Blob, Float32Arrray, Int16Array, ArrayBuffer, Array<number>`,\n );\n }\n if (fromSampleRate === -1) {\n throw new Error(\n `Must specify \"fromSampleRate\" when reading from Float32Array, In16Array or Array`,\n );\n } else if (fromSampleRate < 3000) {\n throw new Error(`Minimum \"fromSampleRate\" is 3000 (3kHz)`);\n }\n if (!data) {\n data = WavPacker.floatTo16BitPCM(float32Array);\n }\n const audio = {\n bitsPerSample: 16,\n channels: [float32Array],\n data,\n };\n const packer = new WavPacker();\n const result = packer.pack(fromSampleRate, audio);\n blob = result.blob;\n arrayBuffer = await blob.arrayBuffer();\n }\n const audioBuffer = await context.decodeAudioData(arrayBuffer);\n const values = audioBuffer.getChannelData(0);\n const url = URL.createObjectURL(blob);\n return {\n blob,\n url,\n values,\n audioBuffer,\n };\n }\n\n /**\n * Logs data in debug mode\n * @param {...any} arguments\n * @returns {true}\n */\n log() {\n if (this.debug) {\n this.log(...arguments);\n }\n return true;\n }\n\n /**\n * Retrieves the current sampleRate for the recorder\n * @returns {number}\n */\n getSampleRate() {\n return this.sampleRate;\n }\n\n /**\n * Retrieves the current status of the recording\n * @returns {\"ended\"|\"paused\"|\"recording\"}\n */\n getStatus() {\n if (!this.processor) {\n return 'ended';\n } else if (!this.recording) {\n return 'paused';\n } else {\n return 'recording';\n }\n }\n\n /**\n * Sends an event to the AudioWorklet\n * @private\n * @param {string} name\n * @param {{[key: string]: any}} data\n * @param {AudioWorkletNode} [_processor]\n * @returns {Promise<{[key: string]: any}>}\n */\n async _event(name, data = {}, _processor = null) {\n _processor = _processor || this.processor;\n if (!_processor) {\n throw new Error('Can not send events without recording first');\n }\n const message = {\n event: name,\n id: this._lastEventId++,\n data,\n };\n _processor.port.postMessage(message);\n const t0 = new Date().valueOf();\n while (!this.eventReceipts[message.id]) {\n if (new Date().valueOf() - t0 > this.eventTimeout) {\n throw new Error(`Timeout waiting for \"${name}\" event`);\n }\n await new Promise((res) => setTimeout(() => res(true), 1));\n }\n const payload = this.eventReceipts[message.id];\n delete this.eventReceipts[message.id];\n return payload;\n }\n\n /**\n * Sets device change callback, remove if callback provided is `null`\n * @param {(Array<MediaDeviceInfo & {default: boolean}>): void|null} callback\n * @returns {true}\n */\n listenForDeviceChange(callback) {\n if (callback === null && this._deviceChangeCallback) {\n navigator.mediaDevices.removeEventListener(\n 'devicechange',\n this._deviceChangeCallback,\n );\n this._deviceChangeCallback = null;\n } else if (callback !== null) {\n // Basically a debounce; we only want this called once when devices change\n // And we only want the most recent callback() to be executed\n // if a few are operating at the same time\n let lastId = 0;\n let lastDevices = [];\n const serializeDevices = (devices) =>\n devices\n .map((d) => d.deviceId)\n .sort()\n .join(',');\n const cb = async () => {\n let id = ++lastId;\n const devices = await this.listDevices();\n if (id === lastId) {\n if (serializeDevices(lastDevices) !== serializeDevices(devices)) {\n lastDevices = devices;\n callback(devices.slice());\n }\n }\n };\n navigator.mediaDevices.addEventListener('devicechange', cb);\n cb();\n this._deviceChangeCallback = cb;\n }\n return true;\n }\n\n /**\n * Manually request permission to use the microphone\n * @returns {Promise<true>}\n */\n async requestPermission() {\n const permissionStatus = await navigator.permissions.query({\n name: 'microphone',\n });\n if (permissionStatus.state === 'denied') {\n window.alert('You must grant microphone access to use this feature.');\n } else if (permissionStatus.state === 'prompt') {\n try {\n const stream = await navigator.mediaDevices.getUserMedia({\n audio: true,\n });\n const tracks = stream.getTracks();\n tracks.forEach((track) => track.stop());\n } catch (e) {\n window.alert('You must grant microphone access to use this feature.');\n }\n }\n return true;\n }\n\n /**\n * List all eligible devices for recording, will request permission to use microphone\n * @returns {Promise<Array<MediaDeviceInfo & {default: boolean}>>}\n */\n async listDevices() {\n if (\n !navigator.mediaDevices ||\n !('enumerateDevices' in navigator.mediaDevices)\n ) {\n throw new Error('Could not request user devices');\n }\n await this.requestPermission();\n const devices = await navigator.mediaDevices.enumerateDevices();\n const audioDevices = devices.filter(\n (device) => device.kind === 'audioinput',\n );\n const defaultDeviceIndex = audioDevices.findIndex(\n (device) => device.deviceId === 'default',\n );\n const deviceList = [];\n if (defaultDeviceIndex !== -1) {\n let defaultDevice = audioDevices.splice(defaultDeviceIndex, 1)[0];\n let existingIndex = audioDevices.findIndex(\n (device) => device.groupId === defaultDevice.groupId,\n );\n if (existingIndex !== -1) {\n defaultDevice = audioDevices.splice(existingIndex, 1)[0];\n }\n defaultDevice.default = true;\n deviceList.push(defaultDevice);\n }\n return deviceList.concat(audioDevices);\n }\n\n /**\n * Begins a recording session and requests microphone permissions if not already granted\n * Microphone recording indicator will appear on browser tab but status will be \"paused\"\n * @param {string} [deviceId] if no device provided, default device will be used\n * @returns {Promise<true>}\n */\n async begin(deviceId) {\n if (this.processor) {\n throw new Error(\n `Already connected: please call .end() to start a new session`,\n );\n }\n\n if (\n !navigator.mediaDevices ||\n !('getUserMedia' in navigator.mediaDevices)\n ) {\n throw new Error('Could not request user media');\n }\n try {\n const config = { audio: true };\n if (deviceId) {\n config.audio = { deviceId: { exact: deviceId } };\n }\n this.stream = await navigator.mediaDevices.getUserMedia(config);\n } catch (err) {\n throw new Error('Could not start media stream');\n }\n\n const context = new AudioContext({ sampleRate: this.sampleRate });\n const source = context.createMediaStreamSource(this.stream);\n // Load and execute the module script.\n try {\n await context.audioWorklet.addModule(this.scriptSrc);\n } catch (e) {\n console.error(e);\n throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);\n }\n const processor = new AudioWorkletNode(context, 'audio_processor');\n processor.port.onmessage = (e) => {\n const { event, id, data } = e.data;\n if (event === 'receipt') {\n this.eventReceipts[id] = data;\n } else if (event === 'chunk') {\n if (this._chunkProcessorSize) {\n const buffer = this._chunkProcessorBuffer;\n this._chunkProcessorBuffer = {\n raw: WavPacker.mergeBuffers(buffer.raw, data.raw),\n mono: WavPacker.mergeBuffers(buffer.mono, data.mono),\n };\n if (\n this._chunkProcessorBuffer.mono.byteLength >=\n this._chunkProcessorSize\n ) {\n this._chunkProcessor(this._chunkProcessorBuffer);\n this._chunkProcessorBuffer = {\n raw: new ArrayBuffer(0),\n mono: new ArrayBuffer(0),\n };\n }\n } else {\n this._chunkProcessor(data);\n }\n }\n };\n\n const node = source.connect(processor);\n const analyser = context.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n node.connect(analyser);\n if (this.outputToSpeakers) {\n // eslint-disable-next-line no-console\n console.warn(\n 'Warning: Output to speakers may affect sound quality,\\n' +\n 'especially due to system audio feedback preventative measures.\\n' +\n 'use only for debugging',\n );\n analyser.connect(context.destination);\n }\n\n this.source = source;\n this.node = node;\n this.analyser = analyser;\n this.processor = processor;\n return true;\n }\n\n /**\n * Gets the current frequency domain data from the recording track\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}\n */\n getFrequencies(\n analysisType = 'frequency',\n minDecibels = -100,\n maxDecibels = -30,\n ) {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n return AudioAnalysis.getFrequencies(\n this.analyser,\n this.sampleRate,\n null,\n analysisType,\n minDecibels,\n maxDecibels,\n );\n }\n\n\n /**\n * Gets the real-time amplitude of the audio signal\n * @returns {number} Amplitude value between 0 and 1\n */\n getAmplitude() {\n if (!this.analyser) {\n throw new Error('AnalyserNode is not initialized. Please call connect() first.');\n }\n\n const bufferLength = this.analyser.fftSize;\n const dataArray = new Uint8Array(bufferLength);\n this.analyser.getByteTimeDomainData(dataArray);\n\n // Calculate RMS (Root Mean Square) to get amplitude\n let sumSquares = 0;\n for (let i = 0; i < bufferLength; i++) {\n const normalized = (dataArray[i] - 128) / 128; // Normalize between -1 and 1\n sumSquares += normalized * normalized;\n }\n const rms = Math.sqrt(sumSquares / bufferLength);\n return rms;\n }\n\n /**\n * Starts amplitude monitoring\n * @param {function} callback - Function to call with amplitude value\n */\n startAmplitudeMonitoring(callback) {\n const monitor = () => {\n const amplitude = this.getAmplitude();\n callback(amplitude);\n requestAnimationFrame(monitor);\n };\n monitor();\n } \n\n /**\n * Pauses the recording\n * Keeps microphone stream open but halts storage of audio\n * @returns {Promise<true>}\n */\n async pause() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n } else if (!this.recording) {\n throw new Error('Already paused: please call .record() first');\n }\n if (this._chunkProcessorBuffer.raw.byteLength) {\n this._chunkProcessor(this._chunkProcessorBuffer);\n }\n this.log('Pausing ...');\n await this._event('stop');\n this.recording = false;\n return true;\n }\n\n /**\n * Start recording stream and storing to memory from the connected audio source\n * @param {(data: { mono: Int16Array; raw: Int16Array }) => any} [chunkProcessor]\n * @param {number} [chunkSize] chunkProcessor will not be triggered until this size threshold met in mono audio\n * @returns {Promise<true>}\n */\n async record(chunkProcessor = () => {}, chunkSize = 8192) {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n } else if (this.recording) {\n throw new Error('Already recording: please call .pause() first');\n } else if (typeof chunkProcessor !== 'function') {\n throw new Error(`chunkProcessor must be a function`);\n }\n this._chunkProcessor = chunkProcessor;\n this._chunkProcessorSize = chunkSize;\n this._chunkProcessorBuffer = {\n raw: new ArrayBuffer(0),\n mono: new ArrayBuffer(0),\n };\n this.log('Recording ...');\n await this._event('start');\n this.recording = true;\n return true;\n }\n\n /**\n * Clears the audio buffer, empties stored recording\n * @returns {Promise<true>}\n */\n async clear() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n await this._event('clear');\n return true;\n }\n\n /**\n * Reads the current audio stream data\n * @returns {Promise<{meanValues: Float32Array, channels: Array<Float32Array>}>}\n */\n async read() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n this.log('Reading ...');\n const result = await this._event('read');\n return result;\n }\n\n /**\n * Saves the current audio stream to a file\n * @param {boolean} [force] Force saving while still recording\n * @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}\n */\n async save(force = false) {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n if (!force && this.recording) {\n throw new Error(\n 'Currently recording: please call .pause() first, or call .save(true) to force',\n );\n }\n this.log('Exporting ...');\n const exportData = await this._event('export');\n const packer = new WavPacker();\n const result = packer.pack(this.sampleRate, exportData.audio);\n return result;\n }\n\n /**\n * Ends the current recording session and saves the result\n * @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}\n */\n async end() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n\n const _processor = this.processor;\n\n this.log('Stopping ...');\n await this._event('stop');\n this.recording = false;\n const tracks = this.stream.getTracks();\n tracks.forEach((track) => track.stop());\n\n this.log('Exporting ...');\n const exportData = await this._event('export', {}, _processor);\n\n this.processor.disconnect();\n this.source.disconnect();\n this.node.disconnect();\n this.analyser.disconnect();\n this.stream = null;\n this.processor = null;\n this.source = null;\n this.node = null;\n\n const packer = new WavPacker();\n const result = packer.pack(this.sampleRate, exportData.audio);\n return result;\n }\n\n /**\n * Performs a full cleanup of WavRecorder instance\n * Stops actively listening via microphone and removes existing listeners\n * @returns {Promise<true>}\n */\n async quit() {\n this.listenForDeviceChange(null);\n if (this.processor) {\n await this.end();\n }\n return true;\n }\n}\n\nglobalThis.WavRecorder = WavRecorder;\n","/**\n * Converts a base64 string to an ArrayBuffer.\n * @param {string} base64 - The base64 string to convert.\n * @returns {ArrayBuffer} The resulting ArrayBuffer.\n */\nexport function base64ToArrayBuffer(base64) {\n const binaryString = atob(base64);\n const len = binaryString.length;\n const bytes = new Uint8Array(len);\n for (let i = 0; i < len; i++) {\n bytes[i] = binaryString.charCodeAt(i);\n }\n return bytes.buffer;\n}\n\n/**\n * Converts an ArrayBuffer to a base64 string.\n * @param {ArrayBuffer|Float32Array|Int16Array} arrayBuffer - The ArrayBuffer to convert.\n * @returns {string} The resulting base64 string.\n */\nexport function arrayBufferToBase64(arrayBuffer) {\n if (arrayBuffer instanceof Float32Array) {\n arrayBuffer = this.floatTo16BitPCM(arrayBuffer);\n } else if (arrayBuffer instanceof Int16Array) {\n arrayBuffer = arrayBuffer.buffer;\n }\n let binary = '';\n let bytes = new Uint8Array(arrayBuffer);\n const chunkSize = 0x8000; // 32KB chunk size\n for (let i = 0; i < bytes.length; i += chunkSize) {\n let chunk = bytes.subarray(i, i + chunkSize);\n binary += String.fromCharCode.apply(null, chunk);\n }\n return btoa(binary);\n}\n","/* eslint-env browser */\nimport { WavRecorder, WavStreamPlayer } from './wavtools/index.js';\nimport { base64ToArrayBuffer, arrayBufferToBase64 } from './utils.js';\n\n/**\n * Interface for LayercodeClient constructor options\n */\ninterface LayercodeClientOptions {\n /** The ID of the Layercode pipeline to connect to */\n pipelineId: string;\n /** The ID of the session to connect to */\n sessionId?: string | null;\n /** The endpoint URL for the audio agent API */\n authorizeSessionEndpoint: string;\n /** Metadata to send with webhooks */\n metadata?: Record<string, any>;\n /** Callback when connection is established */\n onConnect?: ({ sessionId }: { sessionId: string | null }) => void;\n /** Callback when connection is closed */\n onDisconnect?: () => void;\n /** Callback when an error occurs */\n onError?: (error: Error) => void;\n /** Callback for data messages */\n onDataMessage?: (message: any) => void;\n /** Callback for user audio amplitude changes */\n onUserAmplitudeChange?: (amplitude: number) => void;\n /** Callback for agent audio amplitude changes */\n onAgentAmplitudeChange?: (amplitude: number) => void;\n /** Callback when connection status changes */\n onStatusChange?: (status: string) => void;\n}\n\n/**\n * @class LayercodeClient\n * @classdesc Core client for Layercode audio pipeline that manages audio recording, WebSocket communication, and speech processing.\n */\nclass LayercodeClient {\n private options: Required<LayercodeClientOptions>;\n private wavRecorder: WavRecorder;\n private wavPlayer: WavStreamPlayer;\n private ws: WebSocket | null;\n private AMPLITUDE_MONITORING_SAMPLE_RATE: number;\n private assistantIsSpeaking: boolean;\n private assistantHasBeenInterrupted: boolean;\n private pushToTalkActive: boolean;\n _websocketUrl: string;\n status: string;\n userAudioAmplitude: number;\n agentAudioAmplitude: number;\n sessionId: string | null;\n // private currentTurnId: string | null = null;\n // private lastDeltaIdPlayed: string | null = null;\n\n /**\n * Creates an instance of LayercodeClient.\n * @param {Object} options - Configuration options\n */\n constructor(options: LayercodeClientOptions) {\n this.options = {\n pipelineId: options.pipelineId,\n sessionId: options.sessionId || null,\n authorizeSessionEndpoint: options.authorizeSessionEndpoint,\n metadata: options.metadata || {},\n onConnect: options.onConnect || (() => {}),\n onDisconnect: options.onDisconnect || (() => {}),\n onError: options.onError || (() => {}),\n onDataMessage: options.onDataMessage || (() => {}),\n onUserAmplitudeChange: options.onUserAmplitudeChange || (() => {}),\n onAgentAmplitudeChange: options.onAgentAmplitudeChange || (() => {}),\n onStatusChange: options.onStatusChange || (() => {}),\n };\n\n this.AMPLITUDE_MONITORING_SAMPLE_RATE = 10;\n this._websocketUrl = 'wss://api.layercode.com/v1/pipelines/websocket';\n\n this.wavRecorder = new WavRecorder({ sampleRate: 8000 }); // TODO should be set my fetched pipeline config\n this.wavPlayer = new WavStreamPlayer({\n finishedPlayingCallback: this._clientResponseAudioReplayFinished.bind(this),\n sampleRate: 16000, // TODO should be set my fetched pipeline config\n });\n\n this.ws = null;\n this.status = 'disconnected';\n this.userAudioAmplitude = 0;\n this.agentAudioAmplitude = 0;\n this.assistantIsSpeaking = false;\n this.assistantHasBeenInterrupted = false;\n this.sessionId = options.sessionId || null;\n this.pushToTalkActive = false;\n // this.currentTurnId = null; // TODO implement\n // this.lastDeltaIdPlayed = null; // TODO implement\n\n // Bind event handlers\n this._handleWebSocketMessage = this._handleWebSocketMessage.bind(this);\n this._handleDataAvailable = this._handleDataAvailable.bind(this);\n }\n\n /**\n * Updates the connection status and triggers the callback\n * @param {string} status - New status value\n * @private\n */\n private _setStatus(status: string): void {\n this.status = status;\n this.options.onStatusChange(status);\n }\n\n /**\n * Handles when agent audio finishes playing\n * @private\n */\n private _clientResponseAudioReplayFinished(): void {\n this.assistantIsSpeaking = false;\n this._wsSend({\n type: 'trigger.response.audio.replay_finished',\n reason: 'completed',\n // last_delta_id_played: this.lastDeltaIdPlayed, // TODO implement\n // turn_id: this.currentTurnId, // TODO implement\n });\n }\n\n private async _setupWavPlayer(): Promise<void> {\n this.wavPlayer = new WavStreamPlayer({\n finishedPlayingCallback: this._clientResponseAudioReplayFinished.bind(this),\n sampleRate: 16000, // TODO should be set my fetched pipeline config\n });\n await this.wavPlayer.connect();\n\n // Set up amplitude monitoring only if callbacks are provided\n if (this.options.onAgentAmplitudeChange !== (() => {})) {\n let agentUpdateCounter = 0;\n this.wavPlayer.startAmplitudeMonitoring((amplitude: number) => {\n if (agentUpdateCounter == this.AMPLITUDE_MONITORING_SAMPLE_RATE) {\n this.agentAudioAmplitude = amplitude;\n this.options.onAgentAmplitudeChange(amplitude);\n agentUpdateCounter = 0; // Reset after each sample\n }\n agentUpdateCounter++;\n });\n }\n }\n\n private async _clientInterruptAssistantReplay(): Promise<void> {\n if (this.assistantIsSpeaking) {\n await this.wavPlayer.interrupt();\n console.log('interrupting assistant replay');\n console.log('setting assistantIsSpeaking to false');\n console.log('setting assistantHasBeenInterrupted to true');\n this.assistantIsSpeaking = false;\n this.assistantHasBeenInterrupted = true;\n this._wsSend({\n type: 'trigger.response.audio.replay_finished',\n reason: 'interrupted',\n // last_delta_id_played: this.lastDeltaIdPlayed, // TODO implement\n // turn_id: this.currentTurnId, // TODO implement\n });\n }\n }\n\n async triggerUserTurnStarted(): Promise<void> {\n if (!this.pushToTalkActive) {\n this.pushToTalkActive = true;\n this._wsSend({ type: 'trigger.turn.start', role: 'user' });\n await this._clientInterruptAssistantReplay();\n }\n }\n\n async triggerUserTurnFinished(): Promise<void> {\n if (this.pushToTalkActive) {\n this.pushToTalkActive = false;\n this._wsSend({ type: 'trigger.turn.end', role: 'user' });\n }\n }\n\n /**\n * Handles incoming WebSocket messages\n * @param {MessageEvent} event - The WebSocket message event\n * @private\n */\n private async _handleWebSocketMessage(event: MessageEvent): Promise<void> {\n try {\n const message = JSON.parse(event.data);\n if (message.type !== 'response.audio') {\n console.log('received ws msg:', message);\n }\n\n switch (message.type) {\n case 'turn.start':\n // 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.\n if (message.role === 'user') {\n // Interrupt any playing assistant audio\n // await this._clientInterruptAssistantReplay(); // TODO work whether to call interrupt or not here\n }\n break;\n\n case 'response.audio':\n // console.log(\"received response.audio\");\n if (!this.assistantHasBeenInterrupted) {\n // If the assistant has been interrupted, ignore the rest of the audio chunks for this response (turn)\n // TODO: scope audio chunks we ignore based on the turn_id\n if (!this.assistantIsSpeaking) {\n // If we have switch from assistant not speaking to now speaking, this is the start of the assistant's response\n this.assistantIsSpeaking = true;\n console.log('assistantIsSpeaking is currently false, received audio chunk, so setting to true');\n }\n const audioBuffer = base64ToArrayBuffer(message.content);\n this.wavPlayer.add16BitPCM(audioBuffer, 'default');\n } else {\n console.log('ignoring response.audio because assistant has been interrupted');\n }\n break;\n\n case 'response.end':\n console.log('received response.end');\n console.log('setting assistantHasBeenInterrupted to false');\n this.assistantHasBeenInterrupted = false;\n break;\n\n case 'response.data':\n console.log('received response.data', message);\n this.options.onDataMessage(message);\n break;\n\n default:\n console.error('Unknown message type received:', message);\n break;\n }\n } catch (error) {\n console.error('Error processing WebSocket message:', error);\n this.options.onError(error instanceof Error ? error : new Error(String(error)));\n }\n }\n\n /**\n * Handles available client browser microphone audio data and sends it over the WebSocket\n * @param {ArrayBuffer} data - The audio data buffer\n * @private\n */\n private _handleDataAvailable(data: { mono: ArrayBuffer }): void {\n try {\n const base64 = arrayBufferToBase64(data.mono);\n this._wsSend({ type: 'client.audio', content: base64 });\n } catch (error) {\n console.error('Error processing audio:', error);\n this.options.onError(error instanceof Error ? error : new Error(String(error)));\n }\n }\n\n private _wsSend(message: any): void {\n if (message.type !== 'client.audio') {\n console.log('sent ws msg:', message);\n }\n const messageString = JSON.stringify(message);\n if (this.ws?.readyState === WebSocket.OPEN) {\n this.ws.send(messageString);\n } else {\n // console.error(\"WebSocket is not open. Did not send message:\", messageString);\n }\n }\n\n /**\n * Connects to the Layercode pipeline and starts the audio session\n * @async\n * @returns {Promise<void>}\n */\n async connect(): Promise<void> {\n try {\n this._setStatus('connecting');\n\n // Get session key from server\n let authorizeSessionRequestBody = {\n pipeline_id: this.options.pipelineId,\n metadata: this.options.metadata,\n } as { pipeline_id: string; metadata: Record<string, any>; session_id?: string };\n // 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.\n if (this.options.sessionId) {\n authorizeSessionRequestBody.session_id = this.options.sessionId;\n }\n const authorizeSessionResponse = await fetch(this.options.authorizeSessionEndpoint, {\n method: 'POST',\n headers: {\n 'Content-Type': 'application/json',\n },\n body: JSON.stringify(authorizeSessionRequestBody),\n });\n if (!authorizeSessionResponse.ok) {\n throw new Error(`Failed to authorize session: ${authorizeSessionResponse.statusText}`);\n }\n const authorizeSessionResponseBody = await authorizeSessionResponse.json();\n this.sessionId = authorizeSessionResponseBody.session_id; // Save the session_id for use in future reconnects\n\n // Connect WebSocket\n this.ws = new WebSocket(\n `${this._websocketUrl}?${new URLSearchParams({\n client_session_key: authorizeSessionResponseBody.client_session_key,\n })}`\n );\n\n this.ws.onopen = () => {\n console.log('WebSocket connection established');\n this._setStatus('connected');\n this.options.onConnect({ sessionId: this.sessionId });\n };\n\n this.ws.onclose = () => {\n console.log('WebSocket connection closed');\n this._setStatus('disconnected');\n this.options.onDisconnect();\n };\n\n this.ws.onerror = (error: Event) => {\n console.error('WebSocket error:', error);\n this._setStatus('error');\n this.options.onError(new Error('WebSocket connection error'));\n };\n\n this.ws.onmessage = this._handleWebSocketMessage;\n\n // Initialize audio\n await this.wavRecorder.begin();\n await this.wavRecorder.record(this._handleDataAvailable);\n\n if (this.options.onUserAmplitudeChange !== (() => {})) {\n let userUpdateCounter = 0;\n this.wavRecorder.startAmplitudeMonitoring((amplitude: number) => {\n if (userUpdateCounter == this.AMPLITUDE_MONITORING_SAMPLE_RATE) {\n this.userAudioAmplitude = amplitude;\n this.options.onUserAmplitudeChange(amplitude);\n userUpdateCounter = 0; // Reset after each sample\n }\n userUpdateCounter++;\n });\n }\n\n await this._setupWavPlayer();\n\n // Handle page unload\n window.addEventListener('beforeunload', () => {\n this.disconnect();\n });\n } catch (error) {\n console.error('Error connecting to Layercode pipeline:', error);\n this._setStatus('error');\n this.options.onError(error instanceof Error ? error : new Error(String(error)));\n throw error;\n }\n }\n\n /**\n * Disconnects from the Layercode pipeline and stops audio recording\n */\n disconnect(): void {\n console.log('disconnecting');\n if (this.ws) {\n this.ws.close();\n this.ws = null;\n }\n\n // Stop recording user microphone audio\n this.wavRecorder.stop();\n\n // Handle wavPlayer cleanup without calling disconnect directly\n if (this.wavPlayer) {\n // Use type assertion to access internal properties\n const player = this.wavPlayer as any;\n\n // Clean up any audio resources manually\n if (player.stream) {\n player.stream.disconnect();\n player.stream = null;\n }\n\n if 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{"version":3,"file":"layercode-js-sdk.esm.js","sources":["../src/wavtools/lib/wav_packer.js","../src/wavtools/lib/analysis/constants.js","../src/wavtools/lib/analysis/audio_analysis.js","../src/wavtools/lib/worklets/stream_processor.js","../src/wavtools/lib/wav_stream_player.js","../src/wavtools/lib/worklets/audio_processor.js","../src/wavtools/lib/wav_recorder.js","../src/utils.js","../src/index.ts"],"sourcesContent":["/**\n * Raw wav audio file contents\n * @typedef {Object} WavPackerAudioType\n * @property {Blob} blob\n * @property {string} url\n * @property {number} channelCount\n * @property {number} sampleRate\n * @property {number} duration\n */\n\n/**\n * Utility class for assembling PCM16 \"audio/wav\" data\n * @class\n */\nexport class WavPacker {\n /**\n * Converts Float32Array of amplitude data to ArrayBuffer in Int16Array format\n * @param {Float32Array} float32Array\n * @returns {ArrayBuffer}\n */\n static floatTo16BitPCM(float32Array) {\n const buffer = new ArrayBuffer(float32Array.length * 2);\n const view = new DataView(buffer);\n let offset = 0;\n for (let i = 0; i < float32Array.length; i++, offset += 2) {\n let s = Math.max(-1, Math.min(1, float32Array[i]));\n view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);\n }\n return buffer;\n }\n\n /**\n * Concatenates two ArrayBuffers\n * @param {ArrayBuffer} leftBuffer\n * @param {ArrayBuffer} rightBuffer\n * @returns {ArrayBuffer}\n */\n static mergeBuffers(leftBuffer, rightBuffer) {\n const tmpArray = new Uint8Array(\n leftBuffer.byteLength + rightBuffer.byteLength\n );\n tmpArray.set(new Uint8Array(leftBuffer), 0);\n tmpArray.set(new Uint8Array(rightBuffer), leftBuffer.byteLength);\n return tmpArray.buffer;\n }\n\n /**\n * Packs data into an Int16 format\n * @private\n * @param {number} size 0 = 1x Int16, 1 = 2x Int16\n * @param {number} arg value to pack\n * @returns\n */\n _packData(size, arg) {\n return [\n new Uint8Array([arg, arg >> 8]),\n new Uint8Array([arg, arg >> 8, arg >> 16, arg >> 24]),\n ][size];\n }\n\n /**\n * Packs audio into \"audio/wav\" Blob\n * @param {number} sampleRate\n * @param {{bitsPerSample: number, channels: Array<Float32Array>, data: Int16Array}} audio\n * @returns {WavPackerAudioType}\n */\n pack(sampleRate, audio) {\n if (!audio?.bitsPerSample) {\n throw new Error(`Missing \"bitsPerSample\"`);\n } else if (!audio?.channels) {\n throw new Error(`Missing \"channels\"`);\n } else if (!audio?.data) {\n throw new Error(`Missing \"data\"`);\n }\n const { bitsPerSample, channels, data } = audio;\n const output = [\n // Header\n 'RIFF',\n this._packData(\n 1,\n 4 + (8 + 24) /* chunk 1 length */ + (8 + 8) /* chunk 2 length */\n ), // Length\n 'WAVE',\n // chunk 1\n 'fmt ', // Sub-chunk identifier\n this._packData(1, 16), // Chunk length\n this._packData(0, 1), // Audio format (1 is linear quantization)\n this._packData(0, channels.length),\n this._packData(1, sampleRate),\n this._packData(1, (sampleRate * channels.length * bitsPerSample) / 8), // Byte rate\n this._packData(0, (channels.length * bitsPerSample) / 8),\n this._packData(0, bitsPerSample),\n // chunk 2\n 'data', // Sub-chunk identifier\n this._packData(\n 1,\n (channels[0].length * channels.length * bitsPerSample) / 8\n ), // Chunk length\n data,\n ];\n const blob = new Blob(output, { type: 'audio/mpeg' });\n const url = URL.createObjectURL(blob);\n return {\n blob,\n url,\n channelCount: channels.length,\n sampleRate,\n duration: data.byteLength / (channels.length * sampleRate * 2),\n };\n }\n}\n\nglobalThis.WavPacker = WavPacker;\n","/**\n * Constants for help with visualization\n * Helps map frequency ranges from Fast Fourier Transform\n * to human-interpretable ranges, notably music ranges and\n * human vocal ranges.\n */\n\n// Eighth octave frequencies\nconst octave8Frequencies = [\n 4186.01, 4434.92, 4698.63, 4978.03, 5274.04, 5587.65, 5919.91, 6271.93,\n 6644.88, 7040.0, 7458.62, 7902.13,\n];\n\n// Labels for each of the above frequencies\nconst octave8FrequencyLabels = [\n 'C',\n 'C#',\n 'D',\n 'D#',\n 'E',\n 'F',\n 'F#',\n 'G',\n 'G#',\n 'A',\n 'A#',\n 'B',\n];\n\n/**\n * All note frequencies from 1st to 8th octave\n * in format \"A#8\" (A#, 8th octave)\n */\nexport const noteFrequencies = [];\nexport const noteFrequencyLabels = [];\nfor (let i = 1; i <= 8; i++) {\n for (let f = 0; f < octave8Frequencies.length; f++) {\n const freq = octave8Frequencies[f];\n noteFrequencies.push(freq / Math.pow(2, 8 - i));\n noteFrequencyLabels.push(octave8FrequencyLabels[f] + i);\n }\n}\n\n/**\n * Subset of the note frequencies between 32 and 2000 Hz\n * 6 octave range: C1 to B6\n */\nconst voiceFrequencyRange = [32.0, 2000.0];\nexport const voiceFrequencies = noteFrequencies.filter((_, i) => {\n return (\n noteFrequencies[i] > voiceFrequencyRange[0] &&\n noteFrequencies[i] < voiceFrequencyRange[1]\n );\n});\nexport const voiceFrequencyLabels = noteFrequencyLabels.filter((_, i) => {\n return (\n noteFrequencies[i] > voiceFrequencyRange[0] &&\n noteFrequencies[i] < voiceFrequencyRange[1]\n );\n});\n","import {\n noteFrequencies,\n noteFrequencyLabels,\n voiceFrequencies,\n voiceFrequencyLabels,\n} from './constants.js';\n\n/**\n * Output of AudioAnalysis for the frequency domain of the audio\n * @typedef {Object} AudioAnalysisOutputType\n * @property {Float32Array} values Amplitude of this frequency between {0, 1} inclusive\n * @property {number[]} frequencies Raw frequency bucket values\n * @property {string[]} labels Labels for the frequency bucket values\n */\n\n/**\n * Analyzes audio for visual output\n * @class\n */\nexport class AudioAnalysis {\n /**\n * Retrieves frequency domain data from an AnalyserNode adjusted to a decibel range\n * returns human-readable formatting and labels\n * @param {AnalyserNode} analyser\n * @param {number} sampleRate\n * @param {Float32Array} [fftResult]\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {AudioAnalysisOutputType}\n */\n static getFrequencies(\n analyser,\n sampleRate,\n fftResult,\n analysisType = 'frequency',\n minDecibels = -100,\n maxDecibels = -30,\n ) {\n if (!fftResult) {\n fftResult = new Float32Array(analyser.frequencyBinCount);\n analyser.getFloatFrequencyData(fftResult);\n }\n const nyquistFrequency = sampleRate / 2;\n const frequencyStep = (1 / fftResult.length) * nyquistFrequency;\n let outputValues;\n let frequencies;\n let labels;\n if (analysisType === 'music' || analysisType === 'voice') {\n const useFrequencies =\n analysisType === 'voice' ? voiceFrequencies : noteFrequencies;\n const aggregateOutput = Array(useFrequencies.length).fill(minDecibels);\n for (let i = 0; i < fftResult.length; i++) {\n const frequency = i * frequencyStep;\n const amplitude = fftResult[i];\n for (let n = useFrequencies.length - 1; n >= 0; n--) {\n if (frequency > useFrequencies[n]) {\n aggregateOutput[n] = Math.max(aggregateOutput[n], amplitude);\n break;\n }\n }\n }\n outputValues = aggregateOutput;\n frequencies =\n analysisType === 'voice' ? voiceFrequencies : noteFrequencies;\n labels =\n analysisType === 'voice' ? voiceFrequencyLabels : noteFrequencyLabels;\n } else {\n outputValues = Array.from(fftResult);\n frequencies = outputValues.map((_, i) => frequencyStep * i);\n labels = frequencies.map((f) => `${f.toFixed(2)} Hz`);\n }\n // We normalize to {0, 1}\n const normalizedOutput = outputValues.map((v) => {\n return Math.max(\n 0,\n Math.min((v - minDecibels) / (maxDecibels - minDecibels), 1),\n );\n });\n const values = new Float32Array(normalizedOutput);\n return {\n values,\n frequencies,\n labels,\n };\n }\n\n /**\n * Creates a new AudioAnalysis instance for an HTMLAudioElement\n * @param {HTMLAudioElement} audioElement\n * @param {AudioBuffer|null} [audioBuffer] If provided, will cache all frequency domain data from the buffer\n * @returns {AudioAnalysis}\n */\n constructor(audioElement, audioBuffer = null) {\n this.fftResults = [];\n if (audioBuffer) {\n /**\n * Modified from\n * https://stackoverflow.com/questions/75063715/using-the-web-audio-api-to-analyze-a-song-without-playing\n *\n * We do this to populate FFT values for the audio if provided an `audioBuffer`\n * The reason to do this is that Safari fails when using `createMediaElementSource`\n * This has a non-zero RAM cost so we only opt-in to run it on Safari, Chrome is better\n */\n const { length, sampleRate } = audioBuffer;\n const offlineAudioContext = new OfflineAudioContext({\n length,\n sampleRate,\n });\n const source = offlineAudioContext.createBufferSource();\n source.buffer = audioBuffer;\n const analyser = offlineAudioContext.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n source.connect(analyser);\n // limit is :: 128 / sampleRate;\n // but we just want 60fps - cuts ~1s from 6MB to 1MB of RAM\n const renderQuantumInSeconds = 1 / 60;\n const durationInSeconds = length / sampleRate;\n const analyze = (index) => {\n const suspendTime = renderQuantumInSeconds * index;\n if (suspendTime < durationInSeconds) {\n offlineAudioContext.suspend(suspendTime).then(() => {\n const fftResult = new Float32Array(analyser.frequencyBinCount);\n analyser.getFloatFrequencyData(fftResult);\n this.fftResults.push(fftResult);\n analyze(index + 1);\n });\n }\n if (index === 1) {\n offlineAudioContext.startRendering();\n } else {\n offlineAudioContext.resume();\n }\n };\n source.start(0);\n analyze(1);\n this.audio = audioElement;\n this.context = offlineAudioContext;\n this.analyser = analyser;\n this.sampleRate = sampleRate;\n this.audioBuffer = audioBuffer;\n } else {\n const audioContext = new AudioContext();\n const track = audioContext.createMediaElementSource(audioElement);\n const analyser = audioContext.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n track.connect(analyser);\n analyser.connect(audioContext.destination);\n this.audio = audioElement;\n this.context = audioContext;\n this.analyser = analyser;\n this.sampleRate = this.context.sampleRate;\n this.audioBuffer = null;\n }\n }\n\n /**\n * Gets the current frequency domain data from the playing audio track\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {AudioAnalysisOutputType}\n */\n getFrequencies(\n analysisType = 'frequency',\n minDecibels = -100,\n maxDecibels = -30,\n ) {\n let fftResult = null;\n if (this.audioBuffer && this.fftResults.length) {\n const pct = this.audio.currentTime / this.audio.duration;\n const index = Math.min(\n (pct * this.fftResults.length) | 0,\n this.fftResults.length - 1,\n );\n fftResult = this.fftResults[index];\n }\n return AudioAnalysis.getFrequencies(\n this.analyser,\n this.sampleRate,\n fftResult,\n analysisType,\n minDecibels,\n maxDecibels,\n );\n }\n\n /**\n * Resume the internal AudioContext if it was suspended due to the lack of\n * user interaction when the AudioAnalysis was instantiated.\n * @returns {Promise<true>}\n */\n async resumeIfSuspended() {\n if (this.context.state === 'suspended') {\n await this.context.resume();\n }\n return true;\n }\n}\n\nglobalThis.AudioAnalysis = AudioAnalysis;\n","export const StreamProcessorWorklet = `\nclass StreamProcessor extends AudioWorkletProcessor {\n constructor() {\n super();\n this.hasStarted = false;\n this.hasInterrupted = false;\n this.outputBuffers = [];\n this.bufferLength = 128;\n this.write = { buffer: new Float32Array(this.bufferLength), trackId: null };\n this.writeOffset = 0;\n this.trackSampleOffsets = {};\n this.port.onmessage = (event) => {\n if (event.data) {\n const payload = event.data;\n if (payload.event === 'write') {\n const int16Array = payload.buffer;\n const float32Array = new Float32Array(int16Array.length);\n for (let i = 0; i < int16Array.length; i++) {\n float32Array[i] = int16Array[i] / 0x8000; // Convert Int16 to Float32\n }\n this.writeData(float32Array, payload.trackId);\n } else if (\n payload.event === 'offset' ||\n payload.event === 'interrupt'\n ) {\n const requestId = payload.requestId;\n const trackId = this.write.trackId;\n const offset = this.trackSampleOffsets[trackId] || 0;\n this.port.postMessage({\n event: 'offset',\n requestId,\n trackId,\n offset,\n });\n if (payload.event === 'interrupt') {\n this.hasInterrupted = true;\n }\n } else {\n throw new Error(\\`Unhandled event \"\\${payload.event}\"\\`);\n }\n }\n };\n }\n\n writeData(float32Array, trackId = null) {\n let { buffer } = this.write;\n let offset = this.writeOffset;\n for (let i = 0; i < float32Array.length; i++) {\n buffer[offset++] = float32Array[i];\n if (offset >= buffer.length) {\n this.outputBuffers.push(this.write);\n this.write = { buffer: new Float32Array(this.bufferLength), trackId };\n buffer = this.write.buffer;\n offset = 0;\n }\n }\n this.writeOffset = offset;\n return true;\n }\n\n process(inputs, outputs, parameters) {\n const output = outputs[0];\n const outputChannelData = output[0];\n const outputBuffers = this.outputBuffers;\n if (this.hasInterrupted) {\n this.port.postMessage({ event: 'stop' });\n return false;\n } else if (outputBuffers.length) {\n this.hasStarted = true;\n const { buffer, trackId } = outputBuffers.shift();\n for (let i = 0; i < outputChannelData.length; i++) {\n outputChannelData[i] = buffer[i] || 0;\n }\n if (trackId) {\n this.trackSampleOffsets[trackId] =\n this.trackSampleOffsets[trackId] || 0;\n this.trackSampleOffsets[trackId] += buffer.length;\n }\n return true;\n } else if (this.hasStarted) {\n this.port.postMessage({ event: 'stop' });\n return false;\n } else {\n return true;\n }\n }\n}\n\nregisterProcessor('stream_processor', StreamProcessor);\n`;\n\nconst script = new Blob([StreamProcessorWorklet], {\n type: 'application/javascript',\n});\nconst src = URL.createObjectURL(script);\nexport const StreamProcessorSrc = src;\n","import { StreamProcessorSrc } from \"./worklets/stream_processor.js\";\nimport { AudioAnalysis } from \"./analysis/audio_analysis.js\";\n\n/**\n * Plays audio streams received in raw PCM16 chunks from the browser\n * @class\n */\nexport class WavStreamPlayer {\n /**\n * Creates a new WavStreamPlayer instance\n * @param {{sampleRate?: number}} options\n * @returns {WavStreamPlayer}\n */\n constructor({ finishedPlayingCallback = () => {}, sampleRate = 24000 } = {}) {\n this.scriptSrc = StreamProcessorSrc;\n this.sampleRate = sampleRate;\n this.context = null;\n this.stream = null;\n this.analyser = null;\n this.trackSampleOffsets = {};\n this.interruptedTrackIds = {};\n this.finishedPlayingCallback = finishedPlayingCallback;\n }\n\n /**\n * Connects the audio context and enables output to speakers\n * @returns {Promise<true>}\n */\n async connect() {\n this.context = new AudioContext({ sampleRate: this.sampleRate });\n if (this.context.state === \"suspended\") {\n await this.context.resume();\n }\n try {\n await this.context.audioWorklet.addModule(this.scriptSrc);\n } catch (e) {\n console.error(e);\n throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);\n }\n const analyser = this.context.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n this.analyser = analyser;\n return true;\n }\n\n /**\n * Gets the current frequency domain data from the playing track\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}\n */\n getFrequencies(analysisType = \"frequency\", minDecibels = -100, maxDecibels = -30) {\n if (!this.analyser) {\n throw new Error(\"Not connected, please call .connect() first\");\n }\n return AudioAnalysis.getFrequencies(this.analyser, this.sampleRate, null, analysisType, minDecibels, maxDecibels);\n }\n\n /**\n * Gets the real-time amplitude of the audio signal\n * @returns {number} Amplitude value between 0 and 1\n */\n getAmplitude() {\n if (!this.analyser) {\n throw new Error(\"AnalyserNode is not initialized. Please call connect() first.\");\n }\n\n const bufferLength = this.analyser.fftSize;\n const dataArray = new Uint8Array(bufferLength);\n this.analyser.getByteTimeDomainData(dataArray);\n\n // Calculate RMS (Root Mean Square) to get amplitude\n let sumSquares = 0;\n for (let i = 0; i < bufferLength; i++) {\n const normalized = (dataArray[i] - 128) / 128; // Normalize between -1 and 1\n sumSquares += normalized * normalized;\n }\n const rms = Math.sqrt(sumSquares / bufferLength);\n return rms;\n }\n\n /**\n * Starts amplitude monitoring\n * @param {function} callback - Function to call with amplitude value\n */\n startAmplitudeMonitoring(callback) {\n const monitor = () => {\n const amplitude = this.getAmplitude();\n callback(amplitude);\n requestAnimationFrame(monitor);\n };\n monitor();\n }\n\n /**\n * Starts audio streaming\n * @private\n * @returns {Promise<true>}\n */\n _start() {\n const streamNode = new AudioWorkletNode(this.context, \"stream_processor\");\n streamNode.connect(this.context.destination);\n streamNode.port.onmessage = (e) => {\n const { event } = e.data;\n if (event === \"stop\") {\n streamNode.disconnect();\n this.stream = null;\n this.finishedPlayingCallback();\n } else if (event === \"offset\") {\n const { requestId, trackId, offset } = e.data;\n const currentTime = offset / this.sampleRate;\n this.trackSampleOffsets[requestId] = { trackId, offset, currentTime };\n }\n };\n this.analyser.disconnect();\n streamNode.connect(this.analyser);\n this.stream = streamNode;\n return true;\n }\n\n /**\n * Adds 16BitPCM data to the currently playing audio stream\n * You can add chunks beyond the current play point and they will be queued for play\n * @param {ArrayBuffer|Int16Array} arrayBuffer\n * @param {string} [trackId]\n * @returns {Int16Array}\n */\n add16BitPCM(arrayBuffer, trackId = \"default\") {\n if (typeof trackId !== \"string\") {\n throw new Error(`trackId must be a string`);\n } else if (this.interruptedTrackIds[trackId]) {\n return;\n }\n if (!this.stream) {\n this._start();\n }\n let buffer;\n if (arrayBuffer instanceof Int16Array) {\n buffer = arrayBuffer;\n } else if (arrayBuffer instanceof ArrayBuffer) {\n buffer = new Int16Array(arrayBuffer);\n } else {\n throw new Error(`argument must be Int16Array or ArrayBuffer`);\n }\n this.stream.port.postMessage({ event: \"write\", buffer, trackId });\n return buffer;\n }\n\n /**\n * Gets the offset (sample count) of the currently playing stream\n * @param {boolean} [interrupt]\n * @returns {{trackId: string|null, offset: number, currentTime: number}}\n */\n async getTrackSampleOffset(interrupt = false) {\n if (!this.stream) {\n return null;\n }\n const requestId = crypto.randomUUID();\n this.stream.port.postMessage({\n event: interrupt ? \"interrupt\" : \"offset\",\n requestId,\n });\n let trackSampleOffset;\n while (!trackSampleOffset) {\n trackSampleOffset = this.trackSampleOffsets[requestId];\n await new Promise((r) => setTimeout(() => r(), 1));\n }\n const { trackId } = trackSampleOffset;\n if (interrupt && trackId) {\n this.interruptedTrackIds[trackId] = true;\n }\n return trackSampleOffset;\n }\n\n /**\n * Strips the current stream and returns the sample offset of the audio\n * @param {boolean} [interrupt]\n * @returns {{trackId: string|null, offset: number, currentTime: number}}\n */\n async interrupt() {\n return this.getTrackSampleOffset(true);\n }\n\n /**\n * Disconnects the audio context and cleans up resources\n * @returns {void}\n */\n disconnect() {\n if (this.stream) {\n this.stream.disconnect();\n this.stream = null;\n }\n\n if (this.analyser) {\n this.analyser.disconnect();\n }\n\n if (this.context) {\n this.context.close().catch((err) => console.error(\"Error closing audio context:\", err));\n }\n }\n}\n\nglobalThis.WavStreamPlayer = WavStreamPlayer;\n","const AudioProcessorWorklet = `\nclass AudioProcessor extends AudioWorkletProcessor {\n\n constructor() {\n super();\n this.port.onmessage = this.receive.bind(this);\n this.initialize();\n }\n\n initialize() {\n this.foundAudio = false;\n this.recording = false;\n this.chunks = [];\n }\n\n /**\n * Concatenates sampled chunks into channels\n * Format is chunk[Left[], Right[]]\n */\n readChannelData(chunks, channel = -1, maxChannels = 9) {\n let channelLimit;\n if (channel !== -1) {\n if (chunks[0] && chunks[0].length - 1 < channel) {\n throw new Error(\n \\`Channel \\${channel} out of range: max \\${chunks[0].length}\\`\n );\n }\n channelLimit = channel + 1;\n } else {\n channel = 0;\n channelLimit = Math.min(chunks[0] ? chunks[0].length : 1, maxChannels);\n }\n const channels = [];\n for (let n = channel; n < channelLimit; n++) {\n const length = chunks.reduce((sum, chunk) => {\n return sum + chunk[n].length;\n }, 0);\n const buffers = chunks.map((chunk) => chunk[n]);\n const result = new Float32Array(length);\n let offset = 0;\n for (let i = 0; i < buffers.length; i++) {\n result.set(buffers[i], offset);\n offset += buffers[i].length;\n }\n channels[n] = result;\n }\n return channels;\n }\n\n /**\n * Combines parallel audio data into correct format,\n * channels[Left[], Right[]] to float32Array[LRLRLRLR...]\n */\n formatAudioData(channels) {\n if (channels.length === 1) {\n // Simple case is only one channel\n const float32Array = channels[0].slice();\n const meanValues = channels[0].slice();\n return { float32Array, meanValues };\n } else {\n const float32Array = new Float32Array(\n channels[0].length * channels.length\n );\n const meanValues = new Float32Array(channels[0].length);\n for (let i = 0; i < channels[0].length; i++) {\n const offset = i * channels.length;\n let meanValue = 0;\n for (let n = 0; n < channels.length; n++) {\n float32Array[offset + n] = channels[n][i];\n meanValue += channels[n][i];\n }\n meanValues[i] = meanValue / channels.length;\n }\n return { float32Array, meanValues };\n }\n }\n\n /**\n * Converts 32-bit float data to 16-bit integers\n */\n floatTo16BitPCM(float32Array) {\n const buffer = new ArrayBuffer(float32Array.length * 2);\n const view = new DataView(buffer);\n let offset = 0;\n for (let i = 0; i < float32Array.length; i++, offset += 2) {\n let s = Math.max(-1, Math.min(1, float32Array[i]));\n view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);\n }\n return buffer;\n }\n\n /**\n * Retrieves the most recent amplitude values from the audio stream\n * @param {number} channel\n */\n getValues(channel = -1) {\n const channels = this.readChannelData(this.chunks, channel);\n const { meanValues } = this.formatAudioData(channels);\n return { meanValues, channels };\n }\n\n /**\n * Exports chunks as an audio/wav file\n */\n export() {\n const channels = this.readChannelData(this.chunks);\n const { float32Array, meanValues } = this.formatAudioData(channels);\n const audioData = this.floatTo16BitPCM(float32Array);\n return {\n meanValues: meanValues,\n audio: {\n bitsPerSample: 16,\n channels: channels,\n data: audioData,\n },\n };\n }\n\n receive(e) {\n const { event, id } = e.data;\n let receiptData = {};\n switch (event) {\n case 'start':\n this.recording = true;\n break;\n case 'stop':\n this.recording = false;\n break;\n case 'clear':\n this.initialize();\n break;\n case 'export':\n receiptData = this.export();\n break;\n case 'read':\n receiptData = this.getValues();\n break;\n default:\n break;\n }\n // Always send back receipt\n this.port.postMessage({ event: 'receipt', id, data: receiptData });\n }\n\n sendChunk(chunk) {\n const channels = this.readChannelData([chunk]);\n const { float32Array, meanValues } = this.formatAudioData(channels);\n const rawAudioData = this.floatTo16BitPCM(float32Array);\n const monoAudioData = this.floatTo16BitPCM(meanValues);\n this.port.postMessage({\n event: 'chunk',\n data: {\n mono: monoAudioData,\n raw: rawAudioData,\n },\n });\n }\n\n process(inputList, outputList, parameters) {\n // Copy input to output (e.g. speakers)\n // Note that this creates choppy sounds with Mac products\n const sourceLimit = Math.min(inputList.length, outputList.length);\n for (let inputNum = 0; inputNum < sourceLimit; inputNum++) {\n const input = inputList[inputNum];\n const output = outputList[inputNum];\n const channelCount = Math.min(input.length, output.length);\n for (let channelNum = 0; channelNum < channelCount; channelNum++) {\n input[channelNum].forEach((sample, i) => {\n output[channelNum][i] = sample;\n });\n }\n }\n const inputs = inputList[0];\n // There's latency at the beginning of a stream before recording starts\n // Make sure we actually receive audio data before we start storing chunks\n let sliceIndex = 0;\n if (!this.foundAudio) {\n for (const channel of inputs) {\n sliceIndex = 0; // reset for each channel\n if (this.foundAudio) {\n break;\n }\n if (channel) {\n for (const value of channel) {\n if (value !== 0) {\n // find only one non-zero entry in any channel\n this.foundAudio = true;\n break;\n } else {\n sliceIndex++;\n }\n }\n }\n }\n }\n if (inputs && inputs[0] && this.foundAudio && this.recording) {\n // We need to copy the TypedArray, because the \\`process\\`\n // internals will reuse the same buffer to hold each input\n const chunk = inputs.map((input) => input.slice(sliceIndex));\n this.chunks.push(chunk);\n this.sendChunk(chunk);\n }\n return true;\n }\n}\n\nregisterProcessor('audio_processor', AudioProcessor);\n`;\n\nconst script = new Blob([AudioProcessorWorklet], {\n type: 'application/javascript',\n});\nconst src = URL.createObjectURL(script);\nexport const AudioProcessorSrc = src;\n","import { AudioProcessorSrc } from './worklets/audio_processor.js';\nimport { AudioAnalysis } from './analysis/audio_analysis.js';\nimport { WavPacker } from './wav_packer.js';\n\n/**\n * Decodes audio into a wav file\n * @typedef {Object} DecodedAudioType\n * @property {Blob} blob\n * @property {string} url\n * @property {Float32Array} values\n * @property {AudioBuffer} audioBuffer\n */\n\n/**\n * Records live stream of user audio as PCM16 \"audio/wav\" data\n * @class\n */\nexport class WavRecorder {\n /**\n * Create a new WavRecorder instance\n * @param {{sampleRate?: number, outputToSpeakers?: boolean, debug?: boolean}} [options]\n * @returns {WavRecorder}\n */\n constructor({\n sampleRate = 24000,\n outputToSpeakers = false,\n debug = false,\n } = {}) {\n // Script source\n this.scriptSrc = AudioProcessorSrc;\n // Config\n this.sampleRate = sampleRate;\n this.outputToSpeakers = outputToSpeakers;\n this.debug = !!debug;\n this._deviceChangeCallback = null;\n this._devices = [];\n // State variables\n this.stream = null;\n this.processor = null;\n this.source = null;\n this.node = null;\n this.recording = false;\n // Event handling with AudioWorklet\n this._lastEventId = 0;\n this.eventReceipts = {};\n this.eventTimeout = 5000;\n // Process chunks of audio\n this._chunkProcessor = () => {};\n this._chunkProcessorSize = void 0;\n this._chunkProcessorBuffer = {\n raw: new ArrayBuffer(0),\n mono: new ArrayBuffer(0),\n };\n }\n\n /**\n * Decodes audio data from multiple formats to a Blob, url, Float32Array and AudioBuffer\n * @param {Blob|Float32Array|Int16Array|ArrayBuffer|number[]} audioData\n * @param {number} sampleRate\n * @param {number} fromSampleRate\n * @returns {Promise<DecodedAudioType>}\n */\n static async decode(audioData, sampleRate = 24000, fromSampleRate = -1) {\n const context = new AudioContext({ sampleRate });\n let arrayBuffer;\n let blob;\n if (audioData instanceof Blob) {\n if (fromSampleRate !== -1) {\n throw new Error(\n `Can not specify \"fromSampleRate\" when reading from Blob`,\n );\n }\n blob = audioData;\n arrayBuffer = await blob.arrayBuffer();\n } else if (audioData instanceof ArrayBuffer) {\n if (fromSampleRate !== -1) {\n throw new Error(\n `Can not specify \"fromSampleRate\" when reading from ArrayBuffer`,\n );\n }\n arrayBuffer = audioData;\n blob = new Blob([arrayBuffer], { type: 'audio/wav' });\n } else {\n let float32Array;\n let data;\n if (audioData instanceof Int16Array) {\n data = audioData;\n float32Array = new Float32Array(audioData.length);\n for (let i = 0; i < audioData.length; i++) {\n float32Array[i] = audioData[i] / 0x8000;\n }\n } else if (audioData instanceof Float32Array) {\n float32Array = audioData;\n } else if (audioData instanceof Array) {\n float32Array = new Float32Array(audioData);\n } else {\n throw new Error(\n `\"audioData\" must be one of: Blob, Float32Arrray, Int16Array, ArrayBuffer, Array<number>`,\n );\n }\n if (fromSampleRate === -1) {\n throw new Error(\n `Must specify \"fromSampleRate\" when reading from Float32Array, In16Array or Array`,\n );\n } else if (fromSampleRate < 3000) {\n throw new Error(`Minimum \"fromSampleRate\" is 3000 (3kHz)`);\n }\n if (!data) {\n data = WavPacker.floatTo16BitPCM(float32Array);\n }\n const audio = {\n bitsPerSample: 16,\n channels: [float32Array],\n data,\n };\n const packer = new WavPacker();\n const result = packer.pack(fromSampleRate, audio);\n blob = result.blob;\n arrayBuffer = await blob.arrayBuffer();\n }\n const audioBuffer = await context.decodeAudioData(arrayBuffer);\n const values = audioBuffer.getChannelData(0);\n const url = URL.createObjectURL(blob);\n return {\n blob,\n url,\n values,\n audioBuffer,\n };\n }\n\n /**\n * Logs data in debug mode\n * @param {...any} arguments\n * @returns {true}\n */\n log() {\n if (this.debug) {\n this.log(...arguments);\n }\n return true;\n }\n\n /**\n * Retrieves the current sampleRate for the recorder\n * @returns {number}\n */\n getSampleRate() {\n return this.sampleRate;\n }\n\n /**\n * Retrieves the current status of the recording\n * @returns {\"ended\"|\"paused\"|\"recording\"}\n */\n getStatus() {\n if (!this.processor) {\n return 'ended';\n } else if (!this.recording) {\n return 'paused';\n } else {\n return 'recording';\n }\n }\n\n /**\n * Sends an event to the AudioWorklet\n * @private\n * @param {string} name\n * @param {{[key: string]: any}} data\n * @param {AudioWorkletNode} [_processor]\n * @returns {Promise<{[key: string]: any}>}\n */\n async _event(name, data = {}, _processor = null) {\n _processor = _processor || this.processor;\n if (!_processor) {\n throw new Error('Can not send events without recording first');\n }\n const message = {\n event: name,\n id: this._lastEventId++,\n data,\n };\n _processor.port.postMessage(message);\n const t0 = new Date().valueOf();\n while (!this.eventReceipts[message.id]) {\n if (new Date().valueOf() - t0 > this.eventTimeout) {\n throw new Error(`Timeout waiting for \"${name}\" event`);\n }\n await new Promise((res) => setTimeout(() => res(true), 1));\n }\n const payload = this.eventReceipts[message.id];\n delete this.eventReceipts[message.id];\n return payload;\n }\n\n /**\n * Sets device change callback, remove if callback provided is `null`\n * @param {(Array<MediaDeviceInfo & {default: boolean}>): void|null} callback\n * @returns {true}\n */\n listenForDeviceChange(callback) {\n if (callback === null && this._deviceChangeCallback) {\n navigator.mediaDevices.removeEventListener(\n 'devicechange',\n this._deviceChangeCallback,\n );\n this._deviceChangeCallback = null;\n } else if (callback !== null) {\n // Basically a debounce; we only want this called once when devices change\n // And we only want the most recent callback() to be executed\n // if a few are operating at the same time\n let lastId = 0;\n let lastDevices = [];\n const serializeDevices = (devices) =>\n devices\n .map((d) => d.deviceId)\n .sort()\n .join(',');\n const cb = async () => {\n let id = ++lastId;\n const devices = await this.listDevices();\n if (id === lastId) {\n if (serializeDevices(lastDevices) !== serializeDevices(devices)) {\n lastDevices = devices;\n callback(devices.slice());\n }\n }\n };\n navigator.mediaDevices.addEventListener('devicechange', cb);\n cb();\n this._deviceChangeCallback = cb;\n }\n return true;\n }\n\n /**\n * Manually request permission to use the microphone\n * @returns {Promise<true>}\n */\n async requestPermission() {\n const permissionStatus = await navigator.permissions.query({\n name: 'microphone',\n });\n if (permissionStatus.state === 'denied') {\n window.alert('You must grant microphone access to use this feature.');\n } else if (permissionStatus.state === 'prompt') {\n try {\n const stream = await navigator.mediaDevices.getUserMedia({\n audio: true,\n });\n const tracks = stream.getTracks();\n tracks.forEach((track) => track.stop());\n } catch (e) {\n window.alert('You must grant microphone access to use this feature.');\n }\n }\n return true;\n }\n\n /**\n * List all eligible devices for recording, will request permission to use microphone\n * @returns {Promise<Array<MediaDeviceInfo & {default: boolean}>>}\n */\n async listDevices() {\n if (\n !navigator.mediaDevices ||\n !('enumerateDevices' in navigator.mediaDevices)\n ) {\n throw new Error('Could not request user devices');\n }\n await this.requestPermission();\n const devices = await navigator.mediaDevices.enumerateDevices();\n const audioDevices = devices.filter(\n (device) => device.kind === 'audioinput',\n );\n const defaultDeviceIndex = audioDevices.findIndex(\n (device) => device.deviceId === 'default',\n );\n const deviceList = [];\n if (defaultDeviceIndex !== -1) {\n let defaultDevice = audioDevices.splice(defaultDeviceIndex, 1)[0];\n let existingIndex = audioDevices.findIndex(\n (device) => device.groupId === defaultDevice.groupId,\n );\n if (existingIndex !== -1) {\n defaultDevice = audioDevices.splice(existingIndex, 1)[0];\n }\n defaultDevice.default = true;\n deviceList.push(defaultDevice);\n }\n return deviceList.concat(audioDevices);\n }\n\n /**\n * Begins a recording session and requests microphone permissions if not already granted\n * Microphone recording indicator will appear on browser tab but status will be \"paused\"\n * @param {string} [deviceId] if no device provided, default device will be used\n * @returns {Promise<true>}\n */\n async begin(deviceId) {\n if (this.processor) {\n throw new Error(\n `Already connected: please call .end() to start a new session`,\n );\n }\n\n if (\n !navigator.mediaDevices ||\n !('getUserMedia' in navigator.mediaDevices)\n ) {\n throw new Error('Could not request user media');\n }\n try {\n const config = { audio: true };\n if (deviceId) {\n config.audio = { deviceId: { exact: deviceId } };\n }\n this.stream = await navigator.mediaDevices.getUserMedia(config);\n } catch (err) {\n throw new Error('Could not start media stream');\n }\n\n const context = new AudioContext({ sampleRate: this.sampleRate });\n const source = context.createMediaStreamSource(this.stream);\n // Load and execute the module script.\n try {\n await context.audioWorklet.addModule(this.scriptSrc);\n } catch (e) {\n console.error(e);\n throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);\n }\n const processor = new AudioWorkletNode(context, 'audio_processor');\n processor.port.onmessage = (e) => {\n const { event, id, data } = e.data;\n if (event === 'receipt') {\n this.eventReceipts[id] = data;\n } else if (event === 'chunk') {\n if (this._chunkProcessorSize) {\n const buffer = this._chunkProcessorBuffer;\n this._chunkProcessorBuffer = {\n raw: WavPacker.mergeBuffers(buffer.raw, data.raw),\n mono: WavPacker.mergeBuffers(buffer.mono, data.mono),\n };\n if (\n this._chunkProcessorBuffer.mono.byteLength >=\n this._chunkProcessorSize\n ) {\n this._chunkProcessor(this._chunkProcessorBuffer);\n this._chunkProcessorBuffer = {\n raw: new ArrayBuffer(0),\n mono: new ArrayBuffer(0),\n };\n }\n } else {\n this._chunkProcessor(data);\n }\n }\n };\n\n const node = source.connect(processor);\n const analyser = context.createAnalyser();\n analyser.fftSize = 8192;\n analyser.smoothingTimeConstant = 0.1;\n node.connect(analyser);\n if (this.outputToSpeakers) {\n // eslint-disable-next-line no-console\n console.warn(\n 'Warning: Output to speakers may affect sound quality,\\n' +\n 'especially due to system audio feedback preventative measures.\\n' +\n 'use only for debugging',\n );\n analyser.connect(context.destination);\n }\n\n this.source = source;\n this.node = node;\n this.analyser = analyser;\n this.processor = processor;\n return true;\n }\n\n /**\n * Gets the current frequency domain data from the recording track\n * @param {\"frequency\"|\"music\"|\"voice\"} [analysisType]\n * @param {number} [minDecibels] default -100\n * @param {number} [maxDecibels] default -30\n * @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}\n */\n getFrequencies(\n analysisType = 'frequency',\n minDecibels = -100,\n maxDecibels = -30,\n ) {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n return AudioAnalysis.getFrequencies(\n this.analyser,\n this.sampleRate,\n null,\n analysisType,\n minDecibels,\n maxDecibels,\n );\n }\n\n\n /**\n * Gets the real-time amplitude of the audio signal\n * @returns {number} Amplitude value between 0 and 1\n */\n getAmplitude() {\n if (!this.analyser) {\n throw new Error('AnalyserNode is not initialized. Please call connect() first.');\n }\n\n const bufferLength = this.analyser.fftSize;\n const dataArray = new Uint8Array(bufferLength);\n this.analyser.getByteTimeDomainData(dataArray);\n\n // Calculate RMS (Root Mean Square) to get amplitude\n let sumSquares = 0;\n for (let i = 0; i < bufferLength; i++) {\n const normalized = (dataArray[i] - 128) / 128; // Normalize between -1 and 1\n sumSquares += normalized * normalized;\n }\n const rms = Math.sqrt(sumSquares / bufferLength);\n return rms;\n }\n\n /**\n * Starts amplitude monitoring\n * @param {function} callback - Function to call with amplitude value\n */\n startAmplitudeMonitoring(callback) {\n const monitor = () => {\n const amplitude = this.getAmplitude();\n callback(amplitude);\n requestAnimationFrame(monitor);\n };\n monitor();\n } \n\n /**\n * Pauses the recording\n * Keeps microphone stream open but halts storage of audio\n * @returns {Promise<true>}\n */\n async pause() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n } else if (!this.recording) {\n throw new Error('Already paused: please call .record() first');\n }\n if (this._chunkProcessorBuffer.raw.byteLength) {\n this._chunkProcessor(this._chunkProcessorBuffer);\n }\n this.log('Pausing ...');\n await this._event('stop');\n this.recording = false;\n return true;\n }\n\n /**\n * Start recording stream and storing to memory from the connected audio source\n * @param {(data: { mono: Int16Array; raw: Int16Array }) => any} [chunkProcessor]\n * @param {number} [chunkSize] chunkProcessor will not be triggered until this size threshold met in mono audio\n * @returns {Promise<true>}\n */\n async record(chunkProcessor = () => {}, chunkSize = 8192) {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n } else if (this.recording) {\n throw new Error('Already recording: please call .pause() first');\n } else if (typeof chunkProcessor !== 'function') {\n throw new Error(`chunkProcessor must be a function`);\n }\n this._chunkProcessor = chunkProcessor;\n this._chunkProcessorSize = chunkSize;\n this._chunkProcessorBuffer = {\n raw: new ArrayBuffer(0),\n mono: new ArrayBuffer(0),\n };\n this.log('Recording ...');\n await this._event('start');\n this.recording = true;\n return true;\n }\n\n /**\n * Clears the audio buffer, empties stored recording\n * @returns {Promise<true>}\n */\n async clear() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n await this._event('clear');\n return true;\n }\n\n /**\n * Reads the current audio stream data\n * @returns {Promise<{meanValues: Float32Array, channels: Array<Float32Array>}>}\n */\n async read() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n this.log('Reading ...');\n const result = await this._event('read');\n return result;\n }\n\n /**\n * Saves the current audio stream to a file\n * @param {boolean} [force] Force saving while still recording\n * @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}\n */\n async save(force = false) {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n if (!force && this.recording) {\n throw new Error(\n 'Currently recording: please call .pause() first, or call .save(true) to force',\n );\n }\n this.log('Exporting ...');\n const exportData = await this._event('export');\n const packer = new WavPacker();\n const result = packer.pack(this.sampleRate, exportData.audio);\n return result;\n }\n\n /**\n * Ends the current recording session and saves the result\n * @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}\n */\n async end() {\n if (!this.processor) {\n throw new Error('Session ended: please call .begin() first');\n }\n\n const _processor = this.processor;\n\n this.log('Stopping ...');\n await this._event('stop');\n this.recording = false;\n const tracks = this.stream.getTracks();\n tracks.forEach((track) => track.stop());\n\n this.log('Exporting ...');\n const exportData = await this._event('export', {}, _processor);\n\n this.processor.disconnect();\n this.source.disconnect();\n this.node.disconnect();\n this.analyser.disconnect();\n this.stream = null;\n this.processor = null;\n this.source = null;\n this.node = null;\n\n const packer = new WavPacker();\n const result = packer.pack(this.sampleRate, exportData.audio);\n return result;\n }\n\n /**\n * Performs a full cleanup of WavRecorder instance\n * Stops actively listening via microphone and removes existing listeners\n * @returns {Promise<true>}\n */\n async quit() {\n this.listenForDeviceChange(null);\n if (this.processor) {\n await this.end();\n }\n return true;\n }\n}\n\nglobalThis.WavRecorder = WavRecorder;\n","/**\n * Converts a base64 string to an ArrayBuffer.\n * @param {string} base64 - The base64 string to convert.\n * @returns {ArrayBuffer} The resulting ArrayBuffer.\n */\nexport function base64ToArrayBuffer(base64) {\n const binaryString = atob(base64);\n const len = binaryString.length;\n const bytes = new Uint8Array(len);\n for (let i = 0; i < len; i++) {\n bytes[i] = binaryString.charCodeAt(i);\n }\n return bytes.buffer;\n}\n\n/**\n * Converts an ArrayBuffer to a base64 string.\n * @param {ArrayBuffer|Float32Array|Int16Array} arrayBuffer - The ArrayBuffer to convert.\n * @returns {string} The resulting base64 string.\n */\nexport function arrayBufferToBase64(arrayBuffer) {\n if (arrayBuffer instanceof Float32Array) {\n arrayBuffer = this.floatTo16BitPCM(arrayBuffer);\n } else if (arrayBuffer instanceof Int16Array) {\n arrayBuffer = arrayBuffer.buffer;\n }\n let binary = '';\n let bytes = new Uint8Array(arrayBuffer);\n const chunkSize = 0x8000; // 32KB chunk size\n for (let i = 0; i < bytes.length; i += chunkSize) {\n let chunk = bytes.subarray(i, i + chunkSize);\n binary += String.fromCharCode.apply(null, chunk);\n }\n return btoa(binary);\n}\n","/* eslint-env browser */\nimport { WavRecorder, WavStreamPlayer } from './wavtools/index.js';\nimport { base64ToArrayBuffer, arrayBufferToBase64 } from './utils.js';\n\n/**\n * Interface for LayercodeClient constructor options\n */\ninterface LayercodeClientOptions {\n /** The ID of the Layercode pipeline to connect to */\n pipelineId: string;\n /** The ID of the session to connect to */\n sessionId?: string | null;\n /** The endpoint URL for the audio agent API */\n authorizeSessionEndpoint: string;\n /** Metadata to send with webhooks */\n metadata?: Record<string, any>;\n /** Callback when connection is established */\n onConnect?: ({ sessionId }: { sessionId: string | null }) => void;\n /** Callback when connection is closed */\n onDisconnect?: () => void;\n /** Callback when an error occurs */\n onError?: (error: Error) => void;\n /** Callback for data messages */\n onDataMessage?: (message: any) => void;\n /** Callback for user audio amplitude changes */\n onUserAmplitudeChange?: (amplitude: number) => void;\n /** Callback for agent audio amplitude changes */\n onAgentAmplitudeChange?: (amplitude: number) => void;\n /** Callback when connection status changes */\n onStatusChange?: (status: string) => void;\n}\n\n/**\n * @class LayercodeClient\n * @classdesc Core client for Layercode audio pipeline that manages audio recording, WebSocket communication, and speech processing.\n */\nclass LayercodeClient {\n private options: Required<LayercodeClientOptions>;\n private wavRecorder: WavRecorder;\n private wavPlayer: WavStreamPlayer;\n private ws: WebSocket | null;\n private AMPLITUDE_MONITORING_SAMPLE_RATE: number;\n private pushToTalkActive: boolean;\n _websocketUrl: string;\n status: string;\n userAudioAmplitude: number;\n agentAudioAmplitude: number;\n sessionId: string | null;\n\n /**\n * Creates an instance of LayercodeClient.\n * @param {Object} options - Configuration options\n */\n constructor(options: LayercodeClientOptions) {\n this.options = {\n pipelineId: options.pipelineId,\n sessionId: options.sessionId || null,\n authorizeSessionEndpoint: options.authorizeSessionEndpoint,\n metadata: options.metadata || {},\n onConnect: options.onConnect || (() => {}),\n onDisconnect: options.onDisconnect || (() => {}),\n onError: options.onError || (() => {}),\n onDataMessage: options.onDataMessage || (() => {}),\n onUserAmplitudeChange: options.onUserAmplitudeChange || (() => {}),\n onAgentAmplitudeChange: options.onAgentAmplitudeChange || (() => {}),\n onStatusChange: options.onStatusChange || (() => {}),\n };\n\n this.AMPLITUDE_MONITORING_SAMPLE_RATE = 10;\n this._websocketUrl = 'wss://api.layercode.com/v1/pipelines/websocket';\n\n this.wavRecorder = new WavRecorder({ sampleRate: 8000 }); // TODO should be set my fetched pipeline config\n this.wavPlayer = new WavStreamPlayer({\n finishedPlayingCallback: this._clientResponseAudioReplayFinished.bind(this),\n sampleRate: 16000, // TODO should be set my fetched pipeline config\n });\n\n this.ws = null;\n this.status = 'disconnected';\n this.userAudioAmplitude = 0;\n this.agentAudioAmplitude = 0;\n this.sessionId = options.sessionId || null;\n this.pushToTalkActive = false;\n\n // Bind event handlers\n this._handleWebSocketMessage = this._handleWebSocketMessage.bind(this);\n this._handleDataAvailable = this._handleDataAvailable.bind(this);\n }\n\n /**\n * Updates the connection status and triggers the callback\n * @param {string} status - New status value\n * @private\n */\n private _setStatus(status: string): void {\n this.status = status;\n this.options.onStatusChange(status);\n }\n\n /**\n * Handles when agent audio finishes playing\n * @private\n */\n private _clientResponseAudioReplayFinished(): void {\n console.log('clientResponseAudioReplayFinished');\n // NOTE: Not currently used in voice pipelines\n // this._wsSend({\n // type: 'trigger.response.audio.replay_finished',\n // reason: 'completed',\n // });\n }\n\n private async _clientInterruptAssistantReplay(): Promise<void> {\n await this.wavPlayer.interrupt();\n // TODO: Use in voice pipeline to know how much of the audio has been played and how much to truncate transcript\n // this._wsSend({\n // type: 'trigger.response.audio.replay_finished',\n // reason: 'interrupted',\n // delta_id: 'TODO'\n // });\n }\n\n async triggerUserTurnStarted(): Promise<void> {\n if (!this.pushToTalkActive) {\n this.pushToTalkActive = true;\n this._wsSend({ type: 'trigger.turn.start', role: 'user' });\n await this._clientInterruptAssistantReplay();\n }\n }\n\n async triggerUserTurnFinished(): Promise<void> {\n if (this.pushToTalkActive) {\n this.pushToTalkActive = false;\n this._wsSend({ type: 'trigger.turn.end', role: 'user' });\n }\n }\n\n /**\n * Handles incoming WebSocket messages\n * @param {MessageEvent} event - The WebSocket message event\n * @private\n */\n private async _handleWebSocketMessage(event: MessageEvent): Promise<void> {\n try {\n const message = JSON.parse(event.data);\n if (message.type !== 'response.audio') {\n console.log('received ws msg:', message);\n }\n\n switch (message.type) {\n case 'turn.start':\n // Sent from the server to this client when a new user turn is detected\n console.log('received turn.start from server');\n if (message.role === 'user' && !this.pushToTalkActive) {\n // Interrupt any playing assistant audio if this is a turn trigged by the server (and not push to talk, which will have already called interrupt)\n console.log('interrupting assistant audio, as user turn has started and pushToTalkActive is false');\n await this._clientInterruptAssistantReplay();\n }\n break;\n\n case 'response.audio':\n const audioBuffer = base64ToArrayBuffer(message.content);\n this.wavPlayer.add16BitPCM(audioBuffer, message.turn_id);\n break;\n\n case 'response.end':\n console.log('received response.end');\n break;\n\n case 'response.data':\n console.log('received response.data', message);\n this.options.onDataMessage(message);\n break;\n\n default:\n console.error('Unknown message type received:', message);\n break;\n }\n } catch (error) {\n console.error('Error processing WebSocket message:', error);\n this.options.onError(error instanceof Error ? error : new Error(String(error)));\n }\n }\n\n /**\n * Handles available client browser microphone audio data and sends it over the WebSocket\n * @param {ArrayBuffer} data - The audio data buffer\n * @private\n */\n private _handleDataAvailable(data: { mono: ArrayBuffer }): void {\n try {\n const base64 = arrayBufferToBase64(data.mono);\n this._wsSend({ type: 'client.audio', content: base64 });\n } catch (error) {\n console.error('Error processing audio:', error);\n this.options.onError(error instanceof Error ? error : new Error(String(error)));\n }\n }\n\n private _wsSend(message: any): void {\n if (message.type !== 'client.audio') {\n console.log('sent ws msg:', message);\n }\n const messageString = JSON.stringify(message);\n if (this.ws?.readyState === WebSocket.OPEN) {\n this.ws.send(messageString);\n } else {\n console.error('WebSocket is not open. Did not send message:', messageString);\n }\n }\n\n /**\n * Sets up amplitude monitoring for a given audio source.\n * @param {WavRecorder | WavStreamPlayer} source - The audio source (recorder or player).\n * @param {(amplitude: number) => void} callback - The callback function to invoke on amplitude change.\n * @param {(amplitude: number) => void} updateInternalState - Function to update the internal amplitude state.\n * @private\n */\n private _setupAmplitudeMonitoring(source: WavRecorder | WavStreamPlayer, callback: (amplitude: number) => void, updateInternalState: (amplitude: number) => void): void {\n // Set up amplitude monitoring only if a callback is provided\n // Check against the default no-op function defined in the constructor options\n if (callback !== (() => {})) {\n let updateCounter = 0;\n source.startAmplitudeMonitoring((amplitude: number) => {\n // Only update and call callback at the specified sample rate\n if (updateCounter >= this.AMPLITUDE_MONITORING_SAMPLE_RATE) {\n updateInternalState(amplitude);\n callback(amplitude);\n updateCounter = 0; // Reset counter after sampling\n }\n updateCounter++;\n });\n }\n }\n\n /**\n * Connects to the Layercode pipeline and starts the audio session\n * @async\n * @returns {Promise<void>}\n */\n async connect(): Promise<void> {\n try {\n this._setStatus('connecting');\n\n // Get session key from server\n let authorizeSessionRequestBody = {\n pipeline_id: this.options.pipelineId,\n metadata: this.options.metadata,\n } as { pipeline_id: string; metadata: Record<string, any>; session_id?: string };\n // 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.\n if (this.options.sessionId) {\n authorizeSessionRequestBody.session_id = this.options.sessionId;\n }\n const authorizeSessionResponse = await fetch(this.options.authorizeSessionEndpoint, {\n method: 'POST',\n headers: {\n 'Content-Type': 'application/json',\n },\n body: JSON.stringify(authorizeSessionRequestBody),\n });\n if (!authorizeSessionResponse.ok) {\n throw new Error(`Failed to authorize session: ${authorizeSessionResponse.statusText}`);\n }\n const authorizeSessionResponseBody = await authorizeSessionResponse.json();\n this.sessionId = authorizeSessionResponseBody.session_id; // Save the session_id for use in future reconnects\n\n // Connect WebSocket\n this.ws = new WebSocket(\n `${this._websocketUrl}?${new URLSearchParams({\n client_session_key: authorizeSessionResponseBody.client_session_key,\n })}`\n );\n // Bind the websocket message callbacks\n this.ws.onmessage = this._handleWebSocketMessage;\n this.ws.onopen = () => {\n console.log('WebSocket connection established');\n this._setStatus('connected');\n this.options.onConnect({ sessionId: this.sessionId });\n };\n this.ws.onclose = () => {\n console.log('WebSocket connection closed');\n this._setStatus('disconnected');\n this.options.onDisconnect();\n };\n this.ws.onerror = (error: Event) => {\n console.error('WebSocket error:', error);\n this._setStatus('error');\n this.options.onError(new Error('WebSocket connection error'));\n };\n\n // Initialize microphone audio capture\n await this.wavRecorder.begin();\n await this.wavRecorder.record(this._handleDataAvailable);\n // Set up microphone amplitude monitoring\n this._setupAmplitudeMonitoring(this.wavRecorder, this.options.onUserAmplitudeChange, (amp) => (this.userAudioAmplitude = amp));\n\n // Initialize audio player\n await this.wavPlayer.connect();\n // Set up audio player amplitude monitoring\n this._setupAmplitudeMonitoring(this.wavPlayer, this.options.onAgentAmplitudeChange, (amp) => (this.agentAudioAmplitude = amp));\n } catch (error) {\n console.error('Error connecting to Layercode pipeline:', error);\n this._setStatus('error');\n this.options.onError(error instanceof Error ? error : new Error(String(error)));\n throw error;\n }\n }\n}\n\nexport default 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