我正在开发一个应用程序,该应用程序应该记录用户的语音并通过MQTT协议将其传输到自定义设备。自定义设备的音频规范:小终端,无签名,16位LPCM,8khz采样率.每个数据包应该是1000个字节。
我不熟悉AudioEngine,我发现了这个代码示例,我认为它适合我的情况:
func startRecord() {
audioEngine = AVAudioEngine()
let bus = 0
let inputNode = audioEngine.inputNode
let inputFormat = inputNode.outputFormat(forBus: bus)
var streamDescription = AudioStreamBasicDescription()
streamDescription.mFormatID = kAudioFormatLinearPCM.littleEndian
streamDescription.mSampleRate = 8000.0
streamDescription.mChannelsPerFrame = 1
streamDescription.mBitsPerChannel = 16
streamDescription.mBytesPerPacket = 1000
let outputFormat = AVAudioFormat(streamDescription: &streamDescription)!
guard let converter: AVAudioConverter = AVAudioConverter(from: inputFormat, to: outputFormat) else {
print("Can't convert in to this format")
return
}
inputNode.installTap(onBus: 0, bufferSize: 1024, format: inputFormat) { (buffer, time) in
print("Buffer format: \(buffer.format)")
var newBufferAvailable = true
let inputCallback: AVAudioConverterInputBlock = { inNumPackets, outStatus in
if newBufferAvailable {
outStatus.pointee = .haveData
newBufferAvailable = false
return buffer
} else {
outStatus.pointee = .noDataNow
return nil
}
}
let convertedBuffer = AVAudioPCMBuffer(pcmFormat: outputFormat, frameCapacity: AVAudioFrameCount(outputFormat.sampleRate) * buffer.frameLength / AVAudioFrameCount(buffer.format.sampleRate))!
var error: NSError?
let status = converter.convert(to: convertedBuffer, error: &error, withInputFrom: inputCallback)
assert(status != .error)
print("Converted buffer format:", convertedBuffer.format)
}
audioEngine.prepare()
do {
try audioEngine.start()
} catch {
print("Can't start the engine: \(error)")
}
}但是目前,转换器无法将输入格式转换为我的输出格式,我不明白为什么。如果我将输出格式更改为类似的内容:
let outputFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 8000.0, channels: 1, interleaved: false)!那就成功了。
发布于 2020-12-11 23:45:04
您的streamDescription是错误的,您没有填写所有字段,而mBytesPerPacket则是错误的--这与您的协议要求的数据包不同。对于未压缩音频(如LPCM),AudioStreamBasicDescription要求此字段为1。如果您的协议要求样本在1000个组中,那么您必须这样做。
尝尝这个
var streamDescription = AudioStreamBasicDescription()
streamDescription.mSampleRate = 8000.0
streamDescription.mFormatID = kAudioFormatLinearPCM
streamDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger // no endian flag means little endian
streamDescription.mBytesPerPacket = 2
streamDescription.mFramesPerPacket = 1
streamDescription.mBytesPerFrame = 2
streamDescription.mChannelsPerFrame = 1
streamDescription.mBitsPerChannel = 16
streamDescription.mReserved = 0https://stackoverflow.com/questions/65259492
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