如何用AudioKit获取录音数据并实现满512字节缓冲区TCP传输？

Hey there! Let's walk through how to adapt your AudioKit setup to capture 512-byte audio buffers and send them over TCP, instead of saving to a file. Here's a practical, step-by-step approach with code snippets to guide you:

First, you'll need to shift from file-saving logic to tapping into the raw audio stream. AudioKit's AKInputNode is ideal for this—it lets you attach a callback that triggers whenever new audio data becomes available.

Start by finalizing your session configuration (you already have a head start here):

AKAudioFile.cleanTempDirectory()
AKSettings.bufferLength = .medium

do {
    try AKSettings.setSession(category: .playAndRecord, with: .allowBluetoothA2DP)
} catch {
    AKLog("Session setup error: \(error)")
}

Next, grab the input node and install a tap. The critical part here is calculating the right frame count to hit your 512-byte target. For example, if you're using 16-bit (2 bytes per sample) mono audio, 512 bytes equals 256 frames. Adjust this math if you're working with stereo or a different bit depth.

You'll need a mutable buffer to collect audio samples until it reaches 512 bytes. A Data object works great for this, as it’s easy to append to and check length. Also, remember to handle thread safety: the audio tap runs on a dedicated audio thread, while TCP operations will use a separate queue.

Here's how to set this up:

import AudioKit
import Network

// Define your target buffer size (512 bytes)
let targetBufferSize = 512

// Mutable buffer to accumulate audio samples
var audioBuffer = Data()
// Dedicated queue for thread-safe buffer access
let bufferQueue = DispatchQueue(label: "com.yourapp.audioBufferQueue")

// Configure TCP connection (replace with your server's IP and port)
let tcpConnection = NWConnection(host: "your-server-ip", port: 1234, using: .tcp)

// Handle TCP connection state changes
tcpConnection.stateUpdateHandler = { state in
    switch state {
    case .ready:
        print("TCP connection ready to send data")
    case .failed(let error):
        print("TCP connection failed: \(error.localizedDescription)")
    case .cancelled:
        print("TCP connection cancelled")
    default:
        break
    }
}
tcpConnection.start(queue: .global())

// Get the input node and install the audio tap
let inputNode = AKManager.inputNode
let outputFormat = inputNode.outputFormat(forBus: 0)

// Calculate frame count to match 512 bytes
let bytesPerFrame = Int(outputFormat.streamDescription.pointee.mBytesPerFrame)
let targetFrameCount = targetBufferSize / bytesPerFrame

inputNode.installTap(onBus: 0, bufferSize: AVAudioFrameCount(targetFrameCount), format: outputFormat) { buffer, time in
    // Extract float PCM data from the buffer
    guard let floatChannelData = buffer.floatChannelData?[0] else { return }
    let sampleCount = Int(buffer.frameLength)
    
    // Convert float samples to 16-bit integers and append to our buffer
    bufferQueue.sync {
        for i in 0..<sampleCount {
            // Scale float values (-1.0 to 1.0) to 16-bit integer range (-32768 to 32767)
            let scaledSample = Int16(clamping: floatChannelData[i] * 32767)
            audioBuffer.append(Data(bytes: &scaledSample, count: MemoryLayout<Int16>.size))
            
            // Check if we've hit the 512-byte mark
            if audioBuffer.count >= targetBufferSize {
                self.sendOverTCP(audioBuffer)
                // Reset buffer for next batch
                audioBuffer.removeAll(keepingCapacity: true)
            }
        }
    }
}

// Start AudioKit
do {
    try AKManager.start()
} catch {
    AKLog("Failed to start AudioKit: \(error)")
}

Add this helper method to handle sending the completed buffer over TCP. It uses the Network framework's async send method to avoid blocking the audio thread:

func sendOverTCP(_ data: Data) {
    guard tcpConnection.state == .ready else {
        print("TCP connection not ready—dropping buffer")
        return
    }
    
    tcpConnection.send(content: data, completion: .contentProcessed { error in
        if let error = error {
            print("Failed to send buffer: \(error.localizedDescription)")
        } else {
            print("Successfully sent 512-byte audio buffer")
        }
    })
}

• Audio Format Alignment: Ensure the format (bit depth, channel count, endianness) matches what your server expects. The example uses 16-bit mono—adjust the sample conversion logic if you need stereo or 24-bit audio.
• Thread Safety: Always access audioBuffer via bufferQueue to prevent race conditions between the audio thread and TCP thread.
• Error Resilience: Expand the TCP state handler to handle reconnections if the connection drops, and add logic to buffer data temporarily if the server is unavailable.
• Buffer Adjustments: Some audio hardware may not support your exact requested frame count. If this happens, the tap will deliver smaller buffers—your accumulation logic will still work by collecting samples until it hits 512 bytes.

内容的提问来源于stack exchange，提问作者Sam.Calvin