我试图将音频缓冲区转换为其他格式,我正在使用AVAudioConverter。当您拥有相同的采样率并且不需要使用AVAudioConverterInputBlock时,AVAudioConverter可以完成这项工作。
但如果我处理相同的采样率,我的音频数据会出现奇怪的口吃。我有一种感觉,我没有很好地处理输入块。输出的单词重复两次或三次。以下是完整的方法:
func sendAudio(audioFile: URL, completionHandler: @escaping (Bool, Bool, Data?)->Void) {
createSession(){ sessionUrl, observeURL, session in
let file = try! AVAudioFile(forReading: audioFile)
let formatOfAudio = file.processingFormat
self.engine = AVAudioEngine()
guard let input = self.engine.inputNode else {
print("no input")
return
}
//The audio in format in this case is: <AVAudioFormat 0x61800009d010: 2 ch, 44100 Hz, Float32, non-inter>
let formatIn = formatOfAudio
let formatOut = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 16000, channels: 1, interleaved: true)
let mixer = AVAudioMixerNode()
self.engine.attach(mixer)
mixer.volume = 0.0
self.engine.attach(self.audioPlayerNode)
self.engine.connect(self.audioPlayerNode, to: mixer, format: formatIn)
self.engine.connect(input, to: mixer, format: input.outputFormat(forBus: 0))
self.engine.connect(mixer, to: self.engine.mainMixerNode, format: formatIn)
let audioConverter = AVAudioConverter(from: formatIn, to: formatOut)
mixer.installTap(onBus: 0, bufferSize: 32000, format: formatIn, block: {
(buffer: AVAudioPCMBuffer!, time: AVAudioTime!) -> Void in
let convertedBuffer = AVAudioPCMBuffer(pcmFormat: formatOut, frameCapacity: buffer.frameCapacity)
let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in
outStatus.pointee = AVAudioConverterInputStatus.haveData
return buffer
}
var error: NSError? = nil
let status = audioConverter.convert(to: convertedBuffer, error: &error, withInputFrom: inputBlock)
let myData = convertedBuffer.toData()
completionHandler(true, false, myData)
})
self.audioPlayerNode.scheduleFile(file, at: nil){
self.delayWithSeconds(3.0){
self.engine.stop()
mixer.removeTap(onBus: 0)
completionHandler(true, true, nil)
}
}
do {
try self.engine.start()
} catch {
print(error)
}
self.audioPlayerNode.play()
}
}
有什么想法?我从Apple slide sample获得了此代码:
// Create an input block that’s called when converter needs input
let inputBlock : AVAudioConverterInputBlock = {inNumPackets, outStatus in
if (<no_data_available>) {
outStatus.memory = AVAudioConverterInputStatus.NoDataNow;
return nil;
} else if (<end_of_stream>) {
outStatus.memory = AVAudioConverterInputStatus.EndOfStream;
return nil;
} else {
..outStatus.memory = AVAudioConverterInputStatus.HaveData;
return inBuffer; // fill and return input buffer
}
}
答案 0 :(得分:3)
对于发现此问题的任何人,真正的根本原因是对AVAudioConverterInputBlock
的错误使用。只要足够大,目标缓冲区的容量就无关紧要,但是将重复调用该块,直到目标缓冲区被填满。
如果源缓冲区包含ABC
,它将用ABCABCABC...
填充目标。然后,如果将其通过管道传输到实时回放,则会随机切断这些块以适应回放时间,从而产生怪异的裂纹。
实际的解决方案是将缓冲区提交到转换器后,将AVAudioConverterInputStatus
设置为.noDataNow
。请注意,返回.endOfStream
将永远锁定转换器对象。
var gotData = false
self.converter.convert(to: convertedBuffer, error: nil, withInputFrom: { (_, outStatus) -> AVAudioBuffer? in
if gotData {
outStatus.pointee = .noDataNow
return nil
}
gotData = true
outStatus.pointee = .haveData
return inputBuffer
})
答案 1 :(得分:2)
所以我相信我弄明白了。转换后的缓冲帧容量必须除以被转换的采样率的比率。所以,完整的答案如下:
func sendAudio(audioFile: URL, completionHandler: @escaping (Bool, Bool, Data?)->Void) {
createSession(){ sessionUrl, observeURL, session in
let file = try! AVAudioFile(forReading: audioFile)
let formatOfAudio = file.processingFormat
self.engine = AVAudioEngine()
guard let input = self.engine.inputNode else {
print("no input")
return
}
//The audio in format in this case is: <AVAudioFormat 0x61800009d010: 2 ch, 44100 Hz, Float32, non-inter>
let formatIn = formatOfAudio
let formatOut = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: 16000, channels: 1, interleaved: true)
let mixer = AVAudioMixerNode()
self.engine.attach(mixer)
mixer.volume = 0.0
self.engine.attach(self.audioPlayerNode)
self.engine.connect(self.audioPlayerNode, to: mixer, format: formatIn)
self.engine.connect(input, to: mixer, format: input.outputFormat(forBus: 0))
self.engine.connect(mixer, to: self.engine.mainMixerNode, format: formatIn)
let audioConverter = AVAudioConverter(from: formatIn, to: formatOut)
//Here is where I adjusted for the sample rate. It's hard coded here, but you would want to adjust so that you're dividing the input sample rate by your chosen sample rate.
let sampleRateConversionRatio: Float = 44100.0/16000.0
mixer.installTap(onBus: 0, bufferSize: 32000, format: formatIn, block: {
(buffer: AVAudioPCMBuffer!, time: AVAudioTime!) -> Void in
//And this is where you set the appropriate capacity!
let capacity = UInt32(Float(buffer.frameCapacity)/ratio)
let convertedBuffer = AVAudioPCMBuffer(pcmFormat: formatOut, frameCapacity: capacity)
let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in
outStatus.pointee = AVAudioConverterInputStatus.haveData
return buffer
}
var error: NSError? = nil
let status = audioConverter.convert(to: convertedBuffer, error: &error, withInputFrom: inputBlock)
let myData = convertedBuffer.toData()
completionHandler(true, false, myData)
})
self.audioPlayerNode.scheduleFile(file, at: nil){
self.delayWithSeconds(3.0){
self.engine.stop()
mixer.removeTap(onBus: 0)
completionHandler(true, true, nil)
}
}
do {
try self.engine.start()
} catch {
print(error)
}
self.audioPlayerNode.play()
}
}