Change Sample rate with AudioConverter

问题

I am trying to re-sample the input audio 44.1 kHz to 48 kHz.

1. using AudioToolbox's AUAudioUnit.inputHandler
2. writing out the input 44.1 kHZ to a wav file (this is working perfectly)
3. converting the 44.1 kHz to 48 kHz and writing out this converted bytes to file. https://developer.apple.com/documentation/audiotoolbox/1503098-audioconverterfillcomplexbuffer

The problem is in the 3rd step. After writing out to a file the voice is very noisy. here is my code:

// convert to 48kHz
var audioConverterRef: AudioConverterRef?
CheckError(AudioConverterNew(&self.hardwareFormat,
                             &self.convertingFormat,
                             &audioConverterRef), "AudioConverterNew failed")

let outputBufferSize = inNumBytes
let outputBuffer = UnsafeMutablePointer<Int16>.allocate(capacity: MemoryLayout<Int16>.size * Int(outputBufferSize))
let convertedData = AudioBufferList.allocate(maximumBuffers: 1)
convertedData[0].mNumberChannels = self.hardwareFormat.mChannelsPerFrame
convertedData[0].mDataByteSize = outputBufferSize
convertedData[0].mData = UnsafeMutableRawPointer(outputBuffer)
var ioOutputDataPackets = UInt32(inNumPackets)

CheckError(AudioConverterFillComplexBuffer(audioConverterRef!,
                                           self.coverterCallback,
                                           &bufferList,
                                           &ioOutputDataPackets,
                                           convertedData.unsafeMutablePointer,
                                           nil), "AudioConverterFillComplexBuffer error")

let convertedmData = convertedData[0].mData!
let convertedmDataByteSize = convertedData[0].mDataByteSize

// Write converted packets to file -> audio_unit_int16_48.wav
CheckError(AudioFileWritePackets(self.outputFile48000!,
                                 false,
                                 convertedmDataByteSize,
                                 nil,
                                 recordPacket,
                                 &ioOutputDataPackets,
                                 convertedmData), "AudioFileWritePackets error")

and the conversion callback body is here:

let buffers = UnsafeMutableBufferPointer<AudioBuffer>(start: &bufferList.mBuffers, count: Int(bufferList.mNumberBuffers))

let dataPtr = UnsafeMutableAudioBufferListPointer(ioData)
dataPtr[0].mNumberChannels = 1
dataPtr[0].mData = buffers[0].mData
dataPtr[0].mDataByteSize = buffers[0].mDataByteSize
ioDataPacketCount.pointee = buffers[0].mDataByteSize / UInt32(MemoryLayout<Int16>.size)

the sample project is here: https://drive.google.com/file/d/1GvCJ5hEqf7PsBANwUpVTRE1L7S_zQxnL/view?usp=sharing

回答1:

If part of your chain is still AVAudioEngine, there's sample code from Apple for offline processing of AVAudioFiles.

Here's a modified version that includes the sampleRate change:

import Cocoa
import AVFoundation
import PlaygroundSupport


let outputSampleRate = 48_000.0
let outputAudioFormat = AVAudioFormat(standardFormatWithSampleRate: outputSampleRate, channels: 2)!

// file needs to be in ~/Documents/Shared Playground Data
let localURL = playgroundSharedDataDirectory.appendingPathComponent("inputFile_44.aiff")
let outputURL = playgroundSharedDataDirectory.appendingPathComponent("outputFile_48.aiff")

let sourceFile: AVAudioFile
let format: AVAudioFormat

do {
    sourceFile = try AVAudioFile(forReading: localURL)
    format = sourceFile.processingFormat
} catch {
    fatalError("Unable to load the source audio file: \(error.localizedDescription).")
}

let sourceSettings = sourceFile.fileFormat.settings
var outputSettings = sourceSettings
outputSettings[AVSampleRateKey] = outputSampleRate

let engine = AVAudioEngine()
let player = AVAudioPlayerNode()

engine.attach(player)

// Connect the nodes.
engine.connect(player, to: engine.mainMixerNode, format: format)

// Schedule the source file.
player.scheduleFile(sourceFile, at: nil)

do {
    // The maximum number of frames the engine renders in any single render call.
    let maxFrames: AVAudioFrameCount = 4096
    try engine.enableManualRenderingMode(.offline, format: outputAudioFormat,
                                         maximumFrameCount: maxFrames)
} catch {
    fatalError("Enabling manual rendering mode failed: \(error).")
}

do {
    try engine.start()
    player.play()
} catch {
    fatalError("Unable to start audio engine: \(error).")
}

let buffer = AVAudioPCMBuffer(pcmFormat: engine.manualRenderingFormat, frameCapacity: engine.manualRenderingMaximumFrameCount)!

var outputFile: AVAudioFile?
do {
    outputFile = try AVAudioFile(forWriting: outputURL, settings: outputSettings)
} catch {
    fatalError("Unable to open output audio file: \(error).")
}

let outputLengthD = Double(sourceFile.length) * outputSampleRate / sourceFile.fileFormat.sampleRate
let outputLength = Int64(ceil(outputLengthD)) // no sample left behind

while engine.manualRenderingSampleTime < outputLength {

    do {
        let frameCount = outputLength - engine.manualRenderingSampleTime
        let framesToRender = min(AVAudioFrameCount(frameCount), buffer.frameCapacity)

        let status = try engine.renderOffline(framesToRender, to: buffer)

        switch status {

        case .success:
            // The data rendered successfully. Write it to the output file.
            try outputFile?.write(from: buffer)

        case .insufficientDataFromInputNode:
            // Applicable only when using the input node as one of the sources.
            break

        case .cannotDoInCurrentContext:
            // The engine couldn't render in the current render call.
            // Retry in the next iteration.
            break

        case .error:
            // An error occurred while rendering the audio.
            fatalError("The manual rendering failed.")
        }
    } catch {
        fatalError("The manual rendering failed: \(error).")
    }
}

// Stop the player node and engine.
player.stop()
engine.stop()

outputFile = nil // AVAudioFile won't close until it goes out of scope, so we set output file back to nil here

来源：https://stackoverflow.com/questions/60711929/change-sample-rate-with-audioconverter