@siteed/expo-audio-stream 1.0.1 → 1.0.2

package/build/useAudioRecording.js.map

@@ -1 +1 @@
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{ Platform } from \"expo-modules-core\";\nimport { useCallback, useEffect, useReducer, useRef } from \"react\";\n\nimport { addAudioEventListener } from \".\";\nimport {\n  AudioEventPayload,\n  AudioStreamResult,\n  AudioStreamStatus,\n  RecordingConfig,\n  StartAudioStreamResult,\n} from \"./ExpoAudioStream.types\";\nimport ExpoAudioStreamModule from \"./ExpoAudioStreamModule\";\n\nexport interface AudioDataEvent {\n  data: string | Blob;\n  position: number;\n  fileUri: string;\n  eventDataSize: number;\n  totalSize: number;\n}\n\nexport interface UseAudioRecorderProps {\n  debug?: boolean;\n}\nexport interface UseAudioRecorderState {\n  startRecording: (_: RecordingConfig) => Promise<StartAudioStreamResult>;\n  stopRecording: () => Promise<AudioStreamResult | null>;\n  pauseRecording: () => void;\n  resumeRecording: () => void;\n  isRecording: boolean;\n  isPaused: boolean;\n  duration: number; // Duration of the recording\n  size: number; // Size in bytes of the recorded audio\n}\n\ninterface RecorderState {\n  isRecording: boolean;\n  isPaused: boolean;\n  duration: number;\n  size: number;\n}\n\ntype RecorderAction =\n  | { type: \"START\" | \"STOP\" | \"PAUSE\" | \"RESUME\" }\n  | { type: \"UPDATE_STATUS\"; payload: { duration: number; size: number } };\n\nfunction recorderReducer(\n  state: RecorderState,\n  action: RecorderAction,\n): RecorderState {\n  switch (action.type) {\n    case \"START\":\n      return {\n        ...state,\n        isRecording: true,\n        isPaused: false,\n        duration: 0,\n        size: 0,\n      };\n    case \"STOP\":\n      return { ...state, isRecording: false, isPaused: false };\n    case \"PAUSE\":\n      return { ...state, isPaused: true, isRecording: false };\n    case \"RESUME\":\n      return { ...state, isPaused: false, isRecording: true };\n    case \"UPDATE_STATUS\":\n      return {\n        ...state,\n        duration: action.payload.duration,\n        size: action.payload.size,\n      };\n    default:\n      return state;\n  }\n}\nconst TAG = \"[ useAudioRecorder ] \";\n\nexport function useAudioRecorder({\n  debug = false,\n}: UseAudioRecorderProps = {}): UseAudioRecorderState {\n  const [state, dispatch] = useReducer(recorderReducer, {\n    isRecording: false,\n    isPaused: false,\n    duration: 0,\n    size: 0,\n  });\n\n  const onAudioStreamRef = useRef<\n    ((_: AudioDataEvent) => Promise<void>) | null\n  >(null);\n\n  const logDebug = useCallback(\n    (message: string, data?: any) => {\n      if (debug) {\n        if (data) {\n          console.log(`${TAG} ${message}`, data);\n        } else {\n          console.log(`${TAG} ${message}`);\n        }\n      }\n    },\n    [debug],\n  );\n\n  const handleAudioEvent = useCallback(\n    async (eventData: AudioEventPayload) => {\n      const {\n        fileUri,\n        deltaSize,\n        totalSize,\n        lastEmittedSize,\n        position,\n        streamUuid,\n        encoded,\n        mimeType,\n        buffer,\n      } = eventData;\n      logDebug(`useAudioRecorder] Received audio event:`, {\n        fileUri,\n        deltaSize,\n        totalSize,\n        position,\n        mimeType,\n        lastEmittedSize,\n        streamUuid,\n        encodedLength: encoded?.length,\n      });\n      if (deltaSize === 0) {\n        // Ignore packet with no data\n        return;\n      }\n      try {\n        // Coming from native ( ios / android ) otherwise buffer is set\n        if (Platform.OS !== \"web\") {\n          // Read the audio file as a base64 string for comparison\n          if (!encoded) {\n            console.error(`${TAG} Encoded audio data is missing`);\n            throw new Error(\"Encoded audio data is missing\");\n          }\n          onAudioStreamRef.current?.({\n            data: encoded,\n            position,\n            fileUri,\n            eventDataSize: deltaSize,\n            totalSize,\n          });\n        } else if (buffer) {\n          // Coming from web\n          onAudioStreamRef.current?.({\n            data: buffer,\n            position,\n            fileUri,\n            eventDataSize: deltaSize,\n            totalSize,\n          });\n        }\n      } catch (error) {\n        console.error(`${TAG} Error processing audio event:`, error);\n      }\n    },\n    [logDebug],\n  );\n\n  const checkStatus = useCallback(async () => {\n    try {\n      if (!state.isRecording) {\n        logDebug(`${TAG} Not recording, exiting status check.`);\n        return;\n      }\n\n      const status: AudioStreamStatus = ExpoAudioStreamModule.status();\n      if (debug) {\n        logDebug(`${TAG} Status:`, status);\n      }\n\n      if (!status.isRecording) {\n        dispatch({ type: \"STOP\" });\n      } else {\n        dispatch({\n          type: \"UPDATE_STATUS\",\n          payload: { duration: status.duration, size: status.size },\n        });\n      }\n    } catch (error) {\n      console.error(`${TAG} Error getting status:`, error);\n    }\n  }, [state.isRecording, logDebug]);\n\n  useEffect(() => {\n    let interval: number;\n    if (state.isRecording) {\n      interval = setInterval(checkStatus, 1000);\n    }\n    return () => {\n      if (interval) {\n        clearInterval(interval);\n      }\n    };\n  }, [checkStatus, state.isRecording]);\n\n  useEffect(() => {\n    logDebug(`${TAG} Registering audio event listener`);\n    const subscribe = addAudioEventListener(handleAudioEvent);\n    logDebug(`${TAG} Subscribed to audio event listener`, subscribe);\n\n    return () => {\n      logDebug(`${TAG} Removing audio event listener`);\n      subscribe.remove();\n    };\n  }, [handleAudioEvent, logDebug]);\n\n  const startRecording = useCallback(\n    async (recordingOptions: RecordingConfig) => {\n      if (debug) {\n        logDebug(`${TAG} start recoding`, recordingOptions);\n      }\n      // remove onAudioStream from recordingOptions\n      const { onAudioStream, ...options } = recordingOptions;\n      if (typeof onAudioStream === \"function\") {\n        onAudioStreamRef.current = onAudioStream;\n      } else {\n        console.warn(`${TAG} onAudioStream is not a function`, onAudioStream);\n        onAudioStreamRef.current = null;\n      }\n      const startResult: StartAudioStreamResult =\n        await ExpoAudioStreamModule.startRecording(options);\n      dispatch({ type: \"START\" });\n\n      return startResult;\n    },\n    [logDebug],\n  );\n\n  const stopRecording = useCallback(async () => {\n    logDebug(`${TAG} stoping recording`);\n    const stopResult: AudioStreamResult =\n      await ExpoAudioStreamModule.stopRecording();\n    onAudioStreamRef.current = null;\n    logDebug(`${TAG} recording stopped`, stopResult);\n    dispatch({ type: \"STOP\" });\n    return stopResult;\n  }, [logDebug]);\n\n  const pauseRecording = useCallback(async () => {\n    logDebug(`${TAG} pause recording`);\n    const pauseResult = await ExpoAudioStreamModule.pauseRecording();\n    dispatch({ type: \"PAUSE\" });\n    return pauseResult;\n  }, [logDebug]);\n\n  const resumeRecording = useCallback(async () => {\n    logDebug(`${TAG} resume recording`);\n    const resumeResult = await ExpoAudioStreamModule.resumeRecording();\n    dispatch({ type: \"RESUME\" });\n    return resumeResult;\n  }, [logDebug]);\n\n  return {\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    isPaused: state.isPaused,\n    isRecording: state.isRecording,\n    duration: state.duration,\n    size: state.size,\n  };\n}\n"]}
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src/useAudioRecording.ts\nimport { Platform, Subscription } from \"expo-modules-core\";\nimport { useCallback, useEffect, useReducer, useRef } from \"react\";\n\nimport { addAudioAnalysisListener, addAudioEventListener } from \".\";\nimport {\n  AudioAnalysisData,\n  AudioDataEvent,\n  AudioEventPayload,\n  AudioFeaturesOptions,\n  AudioStreamResult,\n  AudioStreamStatus,\n  RecordingConfig,\n  StartAudioStreamResult,\n} from \"./ExpoAudioStream.types\";\nimport ExpoAudioStreamModule from \"./ExpoAudioStreamModule\";\nimport { WavFileInfo } from \"./utils\";\n\nexport interface ExtractMetadataProps {\n  fileUri?: string; // should provide either fileUri or arrayBuffer\n  wavMetadata?: WavFileInfo;\n  arrayBuffer?: ArrayBuffer;\n  bitDepth?: number;\n  skipWavHeader?: boolean;\n  durationMs?: number;\n  sampleRate?: number;\n  numberOfChannels?: number;\n  algorithm?: \"peak\" | \"rms\";\n  position?: number; // Optional number of bytes to skip. Default is 0\n  length?: number; // Optional number of bytes to read.\n  pointsPerSecond?: number; // Optional number of points per second. Use to reduce the number of points and compute the number of datapoints to return.\n  features?: AudioFeaturesOptions;\n  featuresExtratorUrl?: string;\n}\n\nexport interface UseAudioRecorderProps {\n  debug?: boolean;\n  audioWorkletUrl?: string;\n  featuresExtratorUrl?: string;\n}\n\nexport interface UseAudioRecorderState {\n  startRecording: (_: RecordingConfig) => Promise<StartAudioStreamResult>;\n  stopRecording: () => Promise<AudioStreamResult | null>;\n  pauseRecording: () => void;\n  resumeRecording: () => void;\n  isRecording: boolean;\n  isPaused: boolean;\n  durationMs: number; // Duration of the recording\n  size: number; // Size in bytes of the recorded audio\n  analysisData?: AudioAnalysisData;\n  audioWorkletUrl?: string;\n  featuresExtratorUrl?: string;\n}\n\ninterface RecorderState {\n  isRecording: boolean;\n  isPaused: boolean;\n  durationMs: number;\n  size: number;\n  analysisData?: AudioAnalysisData;\n}\n\ntype RecorderAction =\n  | { type: \"START\" | \"STOP\" | \"PAUSE\" | \"RESUME\" }\n  | { type: \"UPDATE_STATUS\"; payload: { durationMs: number; size: number } }\n  | { type: \"UPDATE_ANALYSIS\"; payload: AudioAnalysisData };\n\nconst defaultAnalysis: AudioAnalysisData = {\n  pointsPerSecond: 20,\n  bitDepth: 32,\n  numberOfChannels: 1,\n  durationMs: 0,\n  sampleRate: 44100,\n  samples: 0,\n  dataPoints: [],\n  amplitudeRange: {\n    min: Number.POSITIVE_INFINITY,\n    max: Number.NEGATIVE_INFINITY,\n  },\n};\n\nfunction recorderReducer(\n  state: RecorderState,\n  action: RecorderAction,\n): RecorderState {\n  switch (action.type) {\n    case \"START\":\n      return {\n        ...state,\n        isRecording: true,\n        isPaused: false,\n        durationMs: 0,\n        size: 0,\n        analysisData: defaultAnalysis, // Reset analysis data\n      };\n    case \"STOP\":\n      return { ...state, isRecording: false, isPaused: false };\n    case \"PAUSE\":\n      return { ...state, isPaused: true, isRecording: false };\n    case \"RESUME\":\n      return { ...state, isPaused: false, isRecording: true };\n    case \"UPDATE_STATUS\":\n      return {\n        ...state,\n        durationMs: action.payload.durationMs,\n        size: action.payload.size,\n      };\n    case \"UPDATE_ANALYSIS\":\n      return {\n        ...state,\n        analysisData: action.payload,\n      };\n    default:\n      return state;\n  }\n}\nconst TAG = \"[ useAudioRecorder ] \";\n\nexport function useAudioRecorder({\n  debug = false,\n  audioWorkletUrl,\n  featuresExtratorUrl,\n}: UseAudioRecorderProps = {}): UseAudioRecorderState {\n  const [state, dispatch] = useReducer(recorderReducer, {\n    isRecording: false,\n    isPaused: false,\n    durationMs: 0,\n    size: 0,\n    analysisData: undefined,\n  });\n\n  const analysisListenerRef = useRef<Subscription | null>(null);\n  const analysisRef = useRef<AudioAnalysisData>({ ...defaultAnalysis });\n\n  // Instantiate the module for web with URLs\n  const ExpoAudioStream =\n    Platform.OS === \"web\"\n      ? ExpoAudioStreamModule({ audioWorkletUrl, featuresExtratorUrl })\n      : ExpoAudioStreamModule;\n\n  const onAudioStreamRef = useRef<\n    ((_: AudioDataEvent) => Promise<void>) | null\n  >(null);\n\n  const logDebug = useCallback(\n    (message: string, data?: any) => {\n      if (debug) {\n        if (data) {\n          console.log(`${TAG} ${message}`, data);\n        } else {\n          console.log(`${TAG} ${message}`);\n        }\n      }\n    },\n    [debug],\n  );\n\n  const handleAudioAnalysis = useCallback(\n    async (analysis: AudioAnalysisData, visualizationDuration: number) => {\n      const savedAnalysisData = analysisRef.current || { ...defaultAnalysis };\n\n      const maxDuration = visualizationDuration;\n\n      logDebug(\n        `[handleAudioAnalysis] Received audio analysis: maxDuration=${maxDuration} analysis.dataPoints=${analysis.dataPoints.length} analysisData.dataPoints=${savedAnalysisData.dataPoints.length}`,\n        analysis,\n      );\n\n      // Combine data points\n      const combinedDataPoints = [\n        ...savedAnalysisData.dataPoints,\n        ...analysis.dataPoints,\n      ];\n\n      // Calculate the new duration\n      const pointsPerSecond =\n        analysis.pointsPerSecond || savedAnalysisData.pointsPerSecond;\n      const maxDataPoints = (pointsPerSecond * visualizationDuration) / 1000;\n\n      logDebug(\n        `[handleAudioAnalysis] Combined data points before trimming: pointsPerSecond=${pointsPerSecond} visualizationDuration=${visualizationDuration} combinedDataPointsLength=${combinedDataPoints.length} vs maxDataPoints=${maxDataPoints}`,\n      );\n\n      // Trim data points to keep within the maximum number of data points\n      if (combinedDataPoints.length > maxDataPoints) {\n        combinedDataPoints.splice(0, combinedDataPoints.length - maxDataPoints);\n      }\n\n      savedAnalysisData.dataPoints = combinedDataPoints;\n      savedAnalysisData.bitDepth =\n        analysis.bitDepth || savedAnalysisData.bitDepth;\n      savedAnalysisData.durationMs =\n        combinedDataPoints.length * (1000 / pointsPerSecond);\n\n      // Update amplitude range\n      const newMin = Math.min(\n        savedAnalysisData.amplitudeRange.min,\n        analysis.amplitudeRange.min,\n      );\n      const newMax = Math.max(\n        savedAnalysisData.amplitudeRange.max,\n        analysis.amplitudeRange.max,\n      );\n\n      savedAnalysisData.amplitudeRange = {\n        min: newMin,\n        max: newMax,\n      };\n\n      logDebug(\n        `[handleAudioAnalysis] Updated analysis data: durationMs=${savedAnalysisData.durationMs}`,\n        savedAnalysisData,\n      );\n\n      // Update the ref\n      analysisRef.current = savedAnalysisData;\n\n      // Dispatch the updated analysis data to state to trigger re-render\n      // need to use spread operator otherwise it doesnt trigger update.\n      dispatch({ type: \"UPDATE_ANALYSIS\", payload: { ...savedAnalysisData } });\n    },\n    [logDebug],\n  );\n\n  const handleAudioEvent = useCallback(\n    async (eventData: AudioEventPayload) => {\n      const {\n        fileUri,\n        deltaSize,\n        totalSize,\n        lastEmittedSize,\n        position,\n        streamUuid,\n        encoded,\n        mimeType,\n        buffer,\n      } = eventData;\n      logDebug(`[handleAudioEvent] Received audio event:`, {\n        fileUri,\n        deltaSize,\n        totalSize,\n        position,\n        mimeType,\n        lastEmittedSize,\n        streamUuid,\n        encodedLength: encoded?.length,\n      });\n      if (deltaSize === 0) {\n        // Ignore packet with no data\n        return;\n      }\n      try {\n        // Coming from native ( ios / android ) otherwise buffer is set\n        if (Platform.OS !== \"web\") {\n          // Read the audio file as a base64 string for comparison\n          if (!encoded) {\n            console.error(`${TAG} Encoded audio data is missing`);\n            throw new Error(\"Encoded audio data is missing\");\n          }\n          onAudioStreamRef.current?.({\n            data: encoded,\n            position,\n            fileUri,\n            eventDataSize: deltaSize,\n            totalSize,\n          });\n        } else if (buffer) {\n          // Coming from web\n          onAudioStreamRef.current?.({\n            data: buffer,\n            position,\n            fileUri,\n            eventDataSize: deltaSize,\n            totalSize,\n          });\n        }\n      } catch (error) {\n        console.error(`${TAG} Error processing audio event:`, error);\n      }\n    },\n    [logDebug],\n  );\n\n  const checkStatus = useCallback(async () => {\n    try {\n      if (!state.isRecording) {\n        logDebug(`${TAG} Not recording, exiting status check.`);\n        return;\n      }\n\n      const status: AudioStreamStatus = ExpoAudioStream.status();\n      if (debug) {\n        logDebug(`${TAG} Status:`, status);\n      }\n\n      dispatch({\n        type: \"UPDATE_STATUS\",\n        payload: { durationMs: status.durationMs, size: status.size },\n      });\n    } catch (error) {\n      console.error(`${TAG} Error getting status:`, error);\n    }\n  }, [state.isRecording, logDebug]);\n\n  useEffect(() => {\n    let interval: ReturnType<typeof setTimeout>;\n    if (state.isRecording) {\n      interval = setInterval(checkStatus, 1000);\n    }\n    return () => {\n      if (interval) {\n        clearInterval(interval);\n      }\n    };\n  }, [checkStatus, state.isRecording]);\n\n  useEffect(() => {\n    logDebug(`Registering audio event listener`);\n    const subscribeAudio = addAudioEventListener(handleAudioEvent);\n\n    logDebug(`Subscribed to audio event listener and analysis listener`, {\n      subscribeAudio,\n    });\n\n    return () => {\n      logDebug(`Removing audio event listener`);\n      subscribeAudio.remove();\n    };\n  }, [handleAudioEvent, handleAudioAnalysis, logDebug]);\n\n  const startRecording = useCallback(\n    async (recordingOptions: RecordingConfig) => {\n      if (debug) {\n        logDebug(`start recoding`, recordingOptions);\n      }\n\n      analysisRef.current = { ...defaultAnalysis }; // Reset analysis data\n\n      const { onAudioStream, ...options } = recordingOptions;\n      const { maxRecentDataDuration = 10000, enableProcessing } = options;\n      if (typeof onAudioStream === \"function\") {\n        onAudioStreamRef.current = onAudioStream;\n      } else {\n        console.warn(`${TAG} onAudioStream is not a function`, onAudioStream);\n        onAudioStreamRef.current = null;\n      }\n      const startResult: StartAudioStreamResult =\n        await ExpoAudioStream.startRecording(options);\n      dispatch({ type: \"START\" });\n\n      if (enableProcessing) {\n        logDebug(`Enabling audio analysis listener`);\n        const listener = addAudioAnalysisListener(async (analysisData) => {\n          try {\n            await handleAudioAnalysis(analysisData, maxRecentDataDuration);\n          } catch (error) {\n            console.warn(`${TAG} Error processing audio analysis:`, error);\n          }\n        });\n\n        analysisListenerRef.current = listener;\n      }\n\n      return startResult;\n    },\n    [logDebug],\n  );\n\n  const stopRecording = useCallback(async () => {\n    logDebug(`${TAG} stoping recording`);\n\n    if (analysisListenerRef.current) {\n      analysisListenerRef.current.remove();\n      analysisListenerRef.current = null;\n    }\n\n    const stopResult: AudioStreamResult = await ExpoAudioStream.stopRecording();\n    onAudioStreamRef.current = null;\n    logDebug(`${TAG} recording stopped`, stopResult);\n    dispatch({ type: \"STOP\" });\n    return stopResult;\n  }, [logDebug]);\n\n  const pauseRecording = useCallback(async () => {\n    logDebug(`${TAG} pause recording`);\n    const pauseResult = await ExpoAudioStream.pauseRecording();\n    dispatch({ type: \"PAUSE\" });\n    return pauseResult;\n  }, [logDebug]);\n\n  const resumeRecording = useCallback(async () => {\n    logDebug(`${TAG} resume recording`);\n    const resumeResult = await ExpoAudioStream.resumeRecording();\n    dispatch({ type: \"RESUME\" });\n    return resumeResult;\n  }, [logDebug]);\n\n  return {\n    startRecording,\n    stopRecording,\n    pauseRecording,\n    resumeRecording,\n    isPaused: state.isPaused,\n    isRecording: state.isRecording,\n    durationMs: state.durationMs,\n    size: state.size,\n    analysisData: state.analysisData,\n  };\n}\n"]}

package/build/utils.d.ts

@@ -0,0 +1,31 @@
+import { EncodingType } from "./ExpoAudioStream.types";
+export declare const WAV_HEADER_SIZE = 44;
+export declare const convertPCMToFloat32: ({ bitDepth, buffer, skipWavHeader, }: {
+    buffer: ArrayBuffer;
+    bitDepth: number;
+    skipWavHeader?: boolean;
+}) => {
+    pcmValues: Float32Array;
+    min: number;
+    max: number;
+};
+interface WavHeaderOptions {
+    buffer: ArrayBuffer;
+    sampleRate: number;
+    numChannels: number;
+    bitDepth: number;
+}
+export declare const writeWavHeader: ({ buffer, sampleRate, numChannels, bitDepth, }: WavHeaderOptions) => ArrayBuffer;
+export interface WavFileInfo {
+    sampleRate: number;
+    numChannels: number;
+    bitDepth: number;
+    size: number;
+    durationMs: number;
+}
+export declare const getWavFileInfo: (arrayBuffer: ArrayBuffer) => Promise<WavFileInfo>;
+export declare const encodingToBitDepth: ({ encoding, }: {
+    encoding: EncodingType;
+}) => number;
+export {};
+//# sourceMappingURL=utils.d.ts.map

package/build/utils.d.ts.map

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package/build/utils.js

@@ -0,0 +1,143 @@
+export const WAV_HEADER_SIZE = 44;
+export const convertPCMToFloat32 = ({ bitDepth, buffer, skipWavHeader = false, }) => {
+    const dataView = new DataView(buffer);
+    const headerOffset = skipWavHeader ? WAV_HEADER_SIZE : 0;
+    const dataLength = buffer.byteLength - headerOffset;
+    const sampleLength = dataLength / (bitDepth / 8);
+    const float32Array = new Float32Array(sampleLength);
+    let min = Infinity;
+    let max = -Infinity;
+    for (let i = 0; i < sampleLength; i++) {
+        let value = 0;
+        const offset = headerOffset + i * (bitDepth / 8);
+        switch (bitDepth) {
+            case 8:
+                value = dataView.getUint8(offset) / 128;
+                break;
+            case 16:
+                value = dataView.getInt16(offset, true) / 32768;
+                break;
+            case 24:
+                value =
+                    (dataView.getUint8(offset) +
+                        (dataView.getUint8(offset + 1) << 8) +
+                        (dataView.getUint8(offset + 2) << 16)) /
+                        8388608;
+                break;
+            case 32:
+                value = dataView.getFloat32(offset, true);
+                break;
+            default:
+                throw new Error(`Unsupported bit depth: ${bitDepth}`);
+        }
+        if (value < min)
+            min = value;
+        if (value > max)
+            max = value;
+        float32Array[i] = value;
+    }
+    return { pcmValues: float32Array, min, max };
+};
+export const writeWavHeader = ({ buffer, sampleRate, numChannels, bitDepth, }) => {
+    const bytesPerSample = bitDepth / 8;
+    const numSamples = buffer.byteLength / (numChannels * bytesPerSample);
+    const view = new DataView(buffer);
+    const blockAlign = numChannels * bytesPerSample;
+    const byteRate = sampleRate * blockAlign;
+    // Function to write a string to the DataView
+    const writeString = (view, offset, string) => {
+        for (let i = 0; i < string.length; i++) {
+            view.setUint8(offset + i, string.charCodeAt(i));
+        }
+    };
+    // Check if the buffer already has a WAV header by looking for "RIFF" at the start
+    const existingHeader = view.getUint32(0, false) === 0x52494646; // "RIFF" in ASCII
+    if (!existingHeader) {
+        // Write the WAV header
+        writeString(view, 0, "RIFF"); // ChunkID
+        view.setUint32(4, 36 + numSamples * blockAlign, true); // ChunkSize
+        writeString(view, 8, "WAVE"); // Format
+        writeString(view, 12, "fmt "); // Subchunk1ID
+        view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM)
+        view.setUint16(20, bitDepth === 32 ? 3 : 1, true); // AudioFormat (3 for float, 1 for PCM)
+        view.setUint16(22, numChannels, true); // NumChannels
+        view.setUint32(24, sampleRate, true); // SampleRate
+        view.setUint32(28, byteRate, true); // ByteRate
+        view.setUint16(32, blockAlign, true); // BlockAlign
+        view.setUint16(34, bitDepth, true); // BitsPerSample
+        writeString(view, 36, "data"); // Subchunk2ID
+        view.setUint32(40, numSamples * blockAlign, true); // Subchunk2Size
+    }
+    else {
+        // Update the existing WAV header if necessary
+        view.setUint32(4, 36 + numSamples * blockAlign, true); // Update ChunkSize
+        view.setUint32(24, sampleRate, true); // Update SampleRate
+        view.setUint32(28, byteRate, true); // Update ByteRate
+        view.setUint16(32, blockAlign, true); // Update BlockAlign
+        view.setUint32(40, numSamples * blockAlign, true); // Update Subchunk2Size
+    }
+    return buffer;
+};
+export const getWavFileInfo = async (arrayBuffer) => {
+    const view = new DataView(arrayBuffer);
+    // Check if the file is a valid RIFF/WAVE file
+    const riffHeader = view.getUint32(0, false); // "RIFF"
+    const waveHeader = view.getUint32(8, false); // "WAVE"
+    if (riffHeader !== 0x52494646 || waveHeader !== 0x57415645) {
+        throw new Error("Invalid WAV file");
+    }
+    // Locate the "fmt " chunk
+    let fmtChunkOffset = 12;
+    let sampleRate = 0;
+    let numChannels = 0;
+    let bitDepth = 0;
+    let dataChunkSize = 0;
+    let audioFormat = 0;
+    while (fmtChunkOffset < view.byteLength) {
+        const chunkId = view.getUint32(fmtChunkOffset, false);
+        const chunkSize = view.getUint32(fmtChunkOffset + 4, true);
+        if (chunkId === 0x666d7420) {
+            // "fmt "
+            audioFormat = view.getUint16(fmtChunkOffset + 8, true);
+            if (audioFormat !== 1 && audioFormat !== 3) {
+                throw new Error("Unsupported WAV file format");
+            }
+            numChannels = view.getUint16(fmtChunkOffset + 10, true);
+            sampleRate = view.getUint32(fmtChunkOffset + 12, true);
+            bitDepth = view.getUint16(fmtChunkOffset + 22, true);
+        }
+        else if (chunkId === 0x64617461) {
+            // "data"
+            dataChunkSize = chunkSize;
+            break;
+        }
+        fmtChunkOffset += 8 + chunkSize;
+    }
+    if (!sampleRate || !numChannels || !bitDepth || !dataChunkSize) {
+        throw new Error("Incomplete WAV file information");
+    }
+    // Calculate duration
+    const bytesPerSample = bitDepth / 8;
+    const numSamples = dataChunkSize / (numChannels * bytesPerSample);
+    const durationMs = (numSamples / sampleRate) * 1000;
+    return {
+        sampleRate,
+        numChannels,
+        bitDepth,
+        size: arrayBuffer.byteLength,
+        durationMs,
+    };
+};
+export const encodingToBitDepth = ({ encoding, }) => {
+    switch (encoding) {
+        case "pcm_32bit":
+            return 32;
+        case "pcm_16bit":
+            return 16;
+        case "pcm_8bit":
+            return 8;
+        default:
+            throw new Error(`Unsupported encoding type: ${encoding}`);
+    }
+};
+//# sourceMappingURL=utils.js.map

package/build/utils.js.map

@@ -0,0 +1 @@
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{ EncodingType } from \"./ExpoAudioStream.types\";\n\nexport const WAV_HEADER_SIZE = 44;\nexport const convertPCMToFloat32 = ({\n  bitDepth,\n  buffer,\n  skipWavHeader = false,\n}: {\n  buffer: ArrayBuffer;\n  bitDepth: number;\n  skipWavHeader?: boolean;\n}): { pcmValues: Float32Array; min: number; max: number } => {\n  const dataView = new DataView(buffer);\n  const headerOffset = skipWavHeader ? WAV_HEADER_SIZE : 0;\n  const dataLength = buffer.byteLength - headerOffset;\n  const sampleLength = dataLength / (bitDepth / 8);\n  const float32Array = new Float32Array(sampleLength);\n  let min = Infinity;\n  let max = -Infinity;\n\n  for (let i = 0; i < sampleLength; i++) {\n    let value = 0;\n    const offset = headerOffset + i * (bitDepth / 8);\n    switch (bitDepth) {\n      case 8:\n        value = dataView.getUint8(offset) / 128;\n        break;\n      case 16:\n        value = dataView.getInt16(offset, true) / 32768;\n        break;\n      case 24:\n        value =\n          (dataView.getUint8(offset) +\n            (dataView.getUint8(offset + 1) << 8) +\n            (dataView.getUint8(offset + 2) << 16)) /\n          8388608;\n        break;\n      case 32:\n        value = dataView.getFloat32(offset, true);\n        break;\n      default:\n        throw new Error(`Unsupported bit depth: ${bitDepth}`);\n    }\n    if (value < min) min = value;\n    if (value > max) max = value;\n    float32Array[i] = value;\n  }\n\n  return { pcmValues: float32Array, min, max };\n};\n\ninterface WavHeaderOptions {\n  buffer: ArrayBuffer;\n  sampleRate: number;\n  numChannels: number;\n  bitDepth: number;\n}\n\nexport const writeWavHeader = ({\n  buffer,\n  sampleRate,\n  numChannels,\n  bitDepth,\n}: WavHeaderOptions): ArrayBuffer => {\n  const bytesPerSample = bitDepth / 8;\n  const numSamples = buffer.byteLength / (numChannels * bytesPerSample);\n  const view = new DataView(buffer);\n  const blockAlign = numChannels * bytesPerSample;\n  const byteRate = sampleRate * blockAlign;\n\n  // Function to write a string to the DataView\n  const writeString = (view: DataView, offset: number, string: string) => {\n    for (let i = 0; i < string.length; i++) {\n      view.setUint8(offset + i, string.charCodeAt(i));\n    }\n  };\n\n  // Check if the buffer already has a WAV header by looking for \"RIFF\" at the start\n  const existingHeader = view.getUint32(0, false) === 0x52494646; // \"RIFF\" in ASCII\n\n  if (!existingHeader) {\n    // Write the WAV header\n    writeString(view, 0, \"RIFF\"); // ChunkID\n    view.setUint32(4, 36 + numSamples * blockAlign, true); // ChunkSize\n    writeString(view, 8, \"WAVE\"); // Format\n    writeString(view, 12, \"fmt \"); // Subchunk1ID\n    view.setUint32(16, 16, true); // Subchunk1Size (16 for PCM)\n    view.setUint16(20, bitDepth === 32 ? 3 : 1, true); // AudioFormat (3 for float, 1 for PCM)\n    view.setUint16(22, numChannels, true); // NumChannels\n    view.setUint32(24, sampleRate, true); // SampleRate\n    view.setUint32(28, byteRate, true); // ByteRate\n    view.setUint16(32, blockAlign, true); // BlockAlign\n    view.setUint16(34, bitDepth, true); // BitsPerSample\n    writeString(view, 36, \"data\"); // Subchunk2ID\n    view.setUint32(40, numSamples * blockAlign, true); // Subchunk2Size\n  } else {\n    // Update the existing WAV header if necessary\n    view.setUint32(4, 36 + numSamples * blockAlign, true); // Update ChunkSize\n    view.setUint32(24, sampleRate, true); // Update SampleRate\n    view.setUint32(28, byteRate, true); // Update ByteRate\n    view.setUint16(32, blockAlign, true); // Update BlockAlign\n    view.setUint32(40, numSamples * blockAlign, true); // Update Subchunk2Size\n  }\n\n  return buffer;\n};\n\nexport interface WavFileInfo {\n  sampleRate: number;\n  numChannels: number;\n  bitDepth: number;\n  size: number; // in bytes\n  durationMs: number; // in seconds\n}\n\nexport const getWavFileInfo = async (\n  arrayBuffer: ArrayBuffer,\n): Promise<WavFileInfo> => {\n  const view = new DataView(arrayBuffer);\n\n  // Check if the file is a valid RIFF/WAVE file\n  const riffHeader = view.getUint32(0, false); // \"RIFF\"\n  const waveHeader = view.getUint32(8, false); // \"WAVE\"\n  if (riffHeader !== 0x52494646 || waveHeader !== 0x57415645) {\n    throw new Error(\"Invalid WAV file\");\n  }\n\n  // Locate the \"fmt \" chunk\n  let fmtChunkOffset = 12;\n  let sampleRate = 0;\n  let numChannels = 0;\n  let bitDepth = 0;\n  let dataChunkSize = 0;\n  let audioFormat = 0;\n\n  while (fmtChunkOffset < view.byteLength) {\n    const chunkId = view.getUint32(fmtChunkOffset, false);\n    const chunkSize = view.getUint32(fmtChunkOffset + 4, true);\n    if (chunkId === 0x666d7420) {\n      // \"fmt \"\n      audioFormat = view.getUint16(fmtChunkOffset + 8, true);\n      if (audioFormat !== 1 && audioFormat !== 3) {\n        throw new Error(\"Unsupported WAV file format\");\n      }\n      numChannels = view.getUint16(fmtChunkOffset + 10, true);\n      sampleRate = view.getUint32(fmtChunkOffset + 12, true);\n      bitDepth = view.getUint16(fmtChunkOffset + 22, true);\n    } else if (chunkId === 0x64617461) {\n      // \"data\"\n      dataChunkSize = chunkSize;\n      break;\n    }\n    fmtChunkOffset += 8 + chunkSize;\n  }\n\n  if (!sampleRate || !numChannels || !bitDepth || !dataChunkSize) {\n    throw new Error(\"Incomplete WAV file information\");\n  }\n\n  // Calculate duration\n  const bytesPerSample = bitDepth / 8;\n  const numSamples = dataChunkSize / (numChannels * bytesPerSample);\n  const durationMs = (numSamples / sampleRate) * 1000;\n\n  return {\n    sampleRate,\n    numChannels,\n    bitDepth,\n    size: arrayBuffer.byteLength,\n    durationMs,\n  };\n};\n\nexport const encodingToBitDepth = ({\n  encoding,\n}: {\n  encoding: EncodingType;\n}): number => {\n  switch (encoding) {\n    case \"pcm_32bit\":\n      return 32;\n    case \"pcm_16bit\":\n      return 16;\n    case \"pcm_8bit\":\n      return 8;\n    default:\n      throw new Error(`Unsupported encoding type: ${encoding}`);\n  }\n};\n"]}

package/expo-module.config.json

@@ -1,9 +1,18 @@
 {
-  "platforms": ["ios", "tvos", "android", "web"],
+  "platforms": [
+    "ios",
+    "tvos",
+    "android",
+    "web"
+  ],
   "ios": {
-    "modules": ["ExpoAudioStreamModule"]
+    "modules": [
+      "ExpoAudioStreamModule"
+    ]
   },
   "android": {
-    "modules": ["net.siteed.audiostream.ExpoAudioStreamModule"]
+    "modules": [
+      "net.siteed.audiostream.ExpoAudioStreamModule"
+    ]
   }
-}
+}

package/ios/AudioAnalysisData.swift

@@ -0,0 +1,39 @@
+//
+//  AudioAnalysisData.swift
+//  ExpoAudioStream
+//
+//  Created by Arthur Breton on 23/6/2024.
+//
+
+import Foundation
+
+public struct AudioAnalysisData {
+    public var pointsPerSecond: Int
+    public var durationMs: Float
+    public var bitDepth: Int
+    public var numberOfChannels: Int
+    public var sampleRate: Float
+    public var samples: Int
+    public var dataPoints: [DataPoint]
+    public var amplitudeRange: (min: Float, max: Float)
+    public var speakerChanges: [(timestamp: Float, speaker: Int)]?
+    public var extractionTimeMs: Float
+}
+
+extension AudioAnalysisData {
+    func toDictionary() -> [String: Any] {
+        let dataPointsArray = dataPoints.map { $0.toDictionary() }
+        return [
+            "pointsPerSecond": pointsPerSecond,
+            "durationMs": durationMs,
+            "bitDepth": bitDepth,
+            "numberOfChannels": numberOfChannels,
+            "sampleRate": sampleRate,
+            "samples": samples,
+            "dataPoints": dataPointsArray,
+            "amplitudeRange": ["min": amplitudeRange.min, "max": amplitudeRange.max],
+            "speakerChanges": speakerChanges?.map { ["timestamp": $0.timestamp, "speaker": $0.speaker] } ?? [],
+            "extractionTimeMs": extractionTimeMs
+        ]
+    }
+}

package/ios/AudioProcessingHelpers.swift

@@ -0,0 +1,59 @@
+// AudioProcessingHelpers.swift
+
+import Accelerate
+
+func extractMFCC(from segment: [Float], sampleRate: Float) -> [Float] {
+    // Placeholder for MFCC extraction logic
+    return []
+}
+
+func extractSpectralCentroid(from segment: [Float], sampleRate: Float) -> Float {
+    Logger.debug("Extracting Spectral Centroid from segment of length \(segment.count)")
+    return 0.0 // TODO: Implement spectral centroid extraction logic
+}
+
+func extractSpectralFlatness(from segment: [Float]) -> Float {
+    Logger.debug("Extracting Spectral Flatness from segment of length \(segment.count)")
+    
+    var mean: Float = 0.0
+    var geometricMean: Float = 1.0
+    let count = vDSP_Length(segment.count)
+    
+    vDSP_meamgv(segment, 1, &mean, count)
+    
+    var sumLogValues: Float = 0.0
+    for value in segment {
+        let adjustedValue = max(value, 1e-10)
+        sumLogValues += log(adjustedValue)
+    }
+    geometricMean = exp(sumLogValues / Float(count))
+    
+    let spectralFlatness = mean > 0 ? geometricMean / mean : 0.0
+    Logger.debug("Spectral Flatness: \(spectralFlatness)")
+    return spectralFlatness
+}
+
+func extractSpectralRollOff(from segment: [Float], sampleRate: Float) -> Float {
+    // Implement spectral roll-off extraction logic
+    return 0.0
+}
+
+func extractSpectralBandwidth(from segment: [Float], sampleRate: Float) -> Float {
+    // Implement spectral bandwidth extraction logic
+    return 0.0
+}
+
+func extractChromagram(from segment: [Float], sampleRate: Float) -> [Float] {
+    // Implement chromagram extraction logic
+    return []
+}
+
+func extractTempo(from segment: [Float], sampleRate: Float) -> Float {
+    // Implement tempo extraction logic
+    return 0.0
+}
+
+func extractHNR(from segment: [Float]) -> Float {
+    // Implement harmonic-to-noise ratio extraction logic
+    return 0.0
+}