use ring buffer to avoid memory leak
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@@ -2,17 +2,48 @@ import CoreAudio
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import Foundation
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public class AudioBuffer {
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private var buffer = Data()
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private let targetChunkDuration: Double
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private let streamFormat: AudioStreamBasicDescription
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private var buffer: [UInt8]
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private var writeIndex: Int = 0
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private var readIndex: Int = 0
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private var availableBytes: Int = 0
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private let maxBufferSize: Int
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// Pre-calculated values for efficiency
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private let bytesPerChunk: Int
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private let chunkDuration: Double
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public init(format: AudioStreamBasicDescription, chunkDuration: Double = 0.2) {
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self.streamFormat = format
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self.targetChunkDuration = chunkDuration
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// Pre-calculate chunk parameters
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let bytesPerFrame = Int(format.mBytesPerFrame)
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let samplesPerChunk = Int(format.mSampleRate * chunkDuration)
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self.bytesPerChunk = samplesPerChunk * bytesPerFrame
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self.chunkDuration = Double(samplesPerChunk) / format.mSampleRate
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// Calculate max buffer size to hold ~10 seconds of audio, way more than the maximum we allow
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let bytesPerSecond = Int(format.mSampleRate) * bytesPerFrame
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self.maxBufferSize = bytesPerSecond * 10
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// Pre-allocate ring buffer
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self.buffer = Array(repeating: 0, count: maxBufferSize)
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}
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public func append(_ data: Data) {
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buffer.append(data)
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guard availableBytes + data.count <= maxBufferSize else {
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Logger.error("Audio buffer overflow", context: [
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"requested": String(data.count),
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"available": String(maxBufferSize - availableBytes)
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])
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return
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}
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// Simple, clean, fast enough
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for byte in data {
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buffer[writeIndex] = byte
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writeIndex = (writeIndex + 1) % maxBufferSize
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}
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availableBytes += data.count
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}
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public func processChunks() -> [AudioPacket] {
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@@ -26,36 +57,48 @@ public class AudioBuffer {
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}
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public func flushRemaining() -> AudioPacket? {
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guard !buffer.isEmpty else { return nil }
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guard availableBytes > 0 else { return nil }
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// Create Data from remaining bytes
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var remainingData = Data(capacity: availableBytes)
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for _ in 0..<availableBytes {
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remainingData.append(buffer[readIndex])
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readIndex = (readIndex + 1) % maxBufferSize
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}
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availableBytes = 0
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let packet = AudioPacket(
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timestamp: Date(),
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duration: 0.0, // Unknown duration for final chunk
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peakAmplitude: 0.0,
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rawAudioData: buffer
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rawAudioData: remainingData
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)
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buffer.removeAll()
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return packet
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}
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private func nextChunk() -> AudioPacket? {
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let bytesPerFrame = Int(streamFormat.mBytesPerFrame)
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let samplesPerChunk = Int(streamFormat.mSampleRate * targetChunkDuration)
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let bytesPerChunk = samplesPerChunk * bytesPerFrame
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// Check if we have enough data for a complete chunk
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guard availableBytes >= bytesPerChunk else { return nil }
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guard buffer.count >= bytesPerChunk else { return nil }
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let chunkData = buffer.prefix(bytesPerChunk)
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// Extract chunk data - bounds-checked but still efficient
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var chunkData = Data(capacity: bytesPerChunk)
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for _ in 0..<bytesPerChunk {
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chunkData.append(buffer[readIndex])
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readIndex = (readIndex + 1) % maxBufferSize
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}
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availableBytes -= bytesPerChunk
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let packet = AudioPacket(
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timestamp: Date(),
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duration: Double(samplesPerChunk) / streamFormat.mSampleRate,
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duration: chunkDuration,
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peakAmplitude: 0.0, // No analysis in raw mode
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rawAudioData: Data(chunkData)
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rawAudioData: chunkData
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)
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buffer.removeFirst(bytesPerChunk)
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return packet
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}
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}
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@@ -75,6 +75,7 @@ public class AudioRecorder {
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}
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// FIXME: note to self, what about installTap? Would require audio engine and a node?
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// No; AudioEngine.installTap() can only fire as often as 100ms. too slow for us
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private func setupAndStartIOProc() {
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Logger.debug("Creating IO proc")
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var status = AudioDeviceCreateIOProcID(
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@@ -8,6 +8,7 @@ public enum AudioTeeError: Error {
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case aggregateDeviceCreationFailed(OSStatus)
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case tapAssignmentFailed(OSStatus)
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case pidTranslationFailed([Int32])
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case bufferOverflow(requested: Int, available: Int)
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}
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// MARK: - Audio Format Conversion Errors
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