javascript - AudioWorklet - 将输出设置为 Float32Array 以流式传输实时音频？

Question

我有从服务器到客户端的音频数据流。它以 Node.js 缓冲区(Uint8Array)开始，然后通过 port.postMessage() 发送到 AudioWorkletProcessor，在那里它被转换为 Float32Array 并存储在 this.data 中。我花了几个小时试图将输出设置为包含在 Float32Array 中的音频数据。记录 Float32Array 预处理显示准确的数据，但在处理过程中记录它表明它在发布新消息时没有变化。这可能是我低级音频编程知识的差距。

当数据到达客户端时，调用以下函数:

  process = (data) => {
        this.node.port.postMessage(data)
  }

顺便说一句，(你可以告诉我)也许我应该使用参数描述符而不是 postMessage？无论如何，这是我的 AudioWorkletProcessor:

class BypassProcessor extends AudioWorkletProcessor {

  constructor() {
    super();
    this.isPlaying = true;
    this.port.onmessage = this.onmessage.bind(this)
  }

  static get parameterDescriptors() {
    return [{ // Maybe we should use parameters. This is not utilized at present.
      name: 'stream',
      defaultValue: 0.707
    }];
  }

  convertBlock = (incomingData) => { // incoming data is a UInt8Array
    let i, l = incomingData.length;
    let outputData = new Float32Array(incomingData.length);
    for (i = 0; i < l; i++) {
      outputData[i] = (incomingData[i] - 128) / 128.0;
    }
    return outputData;
  }

  onmessage(event) {
    const { data } = event;
    let ui8 = new Uint8Array(data);
    this.data = this.convertBlock(ui8)
  }

  process(inputs, outputs) {
    const input = inputs[0];
    const output = outputs[0];
    if (this.data) {
      for (let channel = 0; channel < output.length; ++channel) {
        const inputChannel = input[channel]
        const outputChannel = output[channel]
        for (let i = 0; i < inputChannel.length; ++i) {
          outputChannel[i] = this.data[i]
        }
      }
    }

    return true;

  }
}

registerProcessor('bypass-processor', BypassProcessor);

我怎样才能简单地将 AudioWorkletProcessor 的输出设置为通过的数据？

Answer 1

AudioWorkletProcessor 只处理每 128 个字节，因此您需要管理自己的缓冲区以确保 AudioWorklet 也是如此，可能通过添加 FIFO 来实现。我使用在 WebAssembly 中实现的 RingBuffer(FIFO) 解决了类似的问题，在我的例子中，我收到了一个 160 字节的缓冲区。

看看我的 AudioWorkletProcessor 实现

import Module from './buffer-kernel.wasmodule.js';
import { HeapAudioBuffer, RingBuffer, ALAW_TO_LINEAR } from './audio-helper.js';

class SpeakerWorkletProcessor extends AudioWorkletProcessor {
  constructor(options) {
    super();
    this.payload = null;
    this.bufferSize = options.processorOptions.bufferSize; // Getting buffer size from options
    this.channelCount = options.processorOptions.channelCount;
    this.inputRingBuffer = new RingBuffer(this.bufferSize, this.channelCount);
    this.outputRingBuffer = new RingBuffer(this.bufferSize, this.channelCount);
    this.heapInputBuffer = new HeapAudioBuffer(Module, this.bufferSize, this.channelCount);
    this.heapOutputBuffer = new HeapAudioBuffer(Module, this.bufferSize, this.channelCount);
    this.kernel = new Module.VariableBufferKernel(this.bufferSize);
    this.port.onmessage = this.onmessage.bind(this);
  }

  alawToLinear(incomingData) {
    const outputData = new Float32Array(incomingData.length);
    for (let i = 0; i < incomingData.length; i++) {
      outputData[i] = (ALAW_TO_LINEAR[incomingData[i]] * 1.0) / 32768;
    }
    return outputData;
  }

  onmessage(event) {
    const { data } = event;
    if (data) {
      this.payload = this.alawToLinear(new Uint8Array(data)); //Receiving data from my Socket listener and in my case converting PCM alaw to linear
    } else {
      this.payload = null;
    }
  }

  process(inputs, outputs) {
    const output = outputs[0];
    if (this.payload) {
      this.inputRingBuffer.push([this.payload]); // Pushing data from my Socket

      if (this.inputRingBuffer.framesAvailable >= this.bufferSize) { // if the input data size hits the buffer size, so I can "outputted"  
        this.inputRingBuffer.pull(this.heapInputBuffer.getChannelData());
        this.kernel.process(
          this.heapInputBuffer.getHeapAddress(),
          this.heapOutputBuffer.getHeapAddress(),
          this.channelCount,
        );
        this.outputRingBuffer.push(this.heapOutputBuffer.getChannelData());
      }
      this.outputRingBuffer.pull(output); // Retriving data from FIFO and putting our output
    }
    return true;
  }
}

registerProcessor(`speaker-worklet-processor`, SpeakerWorkletProcessor);

查看 AudioContext 和 AudioWorklet 实例

 this.audioContext = new AudioContext({
      latencyHint: 'interactive',
      sampleRate: this.sampleRate,
      sinkId: audioinput || "default"
    });

    this.audioBuffer = this.audioContext.createBuffer(1, this.audioSize, this.sampleRate);
    this.audioSource = this.audioContext.createBufferSource();
    this.audioSource.buffer = this.audioBuffer;
    this.audioSource.loop = true;
    this.audioContext.audioWorklet
    .addModule('workers/speaker-worklet-processor.js')
    .then(() => {
      this.speakerWorklet = new AudioWorkletNode(
        this.audioContext,
        'speaker-worklet-processor',
        {
          channelCount: 1,
          processorOptions: {
            bufferSize: 160, //Here I'm passing the size of my output, I'm just saying to RingBuffer what size I need 
            channelCount: 1,
          },
        },
      );
      this.audioSource.connect(this.speakerWorklet).connect(this.audioContext.destination);
    }).catch((err)=>{
      console.log("Receiver ", err);
    })

看看我如何从 Socket 接收数据并将数据发送到 audioWorklet

  protected onMessage(e: any): void { //My Socket message listener
    const { data:serverData } = e;
    const socketId = e.socketId;
    if (this.audioWalking && this.ws && !this.ws.isPaused() && this.ws.info.socketId === socketId) {
      const buffer = arrayBufferToBuffer(serverData);
      const rtp = RTPParser.parseRtpPacket(buffer);
      const sharedPayload = new Uint8Array(new SharedArrayBuffer(rtp.payload.length)); //sharing javascript buffer memory between main thread and worklet thread
      sharedPayload.set(rtp.payload, 0);
      this.speakerWorklet.port.postMessage(sharedPayload); //Sending data to worklet
    }
  } 

为了帮助人们，我把这个解决方案的重要部分放在了 Github 上

• > audio-worklet-processor-wasm-example

我按照这个例子，它解释了 RingBuffer 是如何工作的

• > wasm-ring-buffer