AVAudioConverter with AVAudioConverterInputBlock stutters audio after processing

Question

I'm trying to convert audio buffers to a different format, and I'm using AVAudioConverter. AVAudioConverter does the job when you have the same sample rate and you don't need to use the AVAudioConverterInputBlock.

But if I'm dealing with the same sample rate, I'm getting a strange stutter in my audio data. I have a feeling I'm not handling the input block well. The output has words repeating two or three times. Below is the full method:

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()
    }
}

Any thoughts? I got this code from an Apple slide sample:

// 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 
    }  
}

Answer 1

So I believe I figured it out. The converted buffer frame capacity must be divided by a ratio of the sample rates being converted. So, the complete answer looks like this:

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()
    }
}

Answer 2

For anybody finding this, the actual underlying cause is the incorrect use of AVAudioConverterInputBlock. The destination buffer capacity doesn't matter as long as it is large enough, however the block will be called repeatedly until the destination buffer is filled.

If your source buffer contains ABC, it will fill up the destination with ABCABCABC.... Then, if you pipe it to realtime playback, the chunks are getting cut off randomly to fit the playback timing, resulting in this weird crackle.

The actual solution is to properly set AVAudioConverterInputStatus to .noDataNow once the buffer is submitted to the converter. Note that returning .endOfStream will lock up the converter object forever.

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
})