I have, for the past week, been trying to take audio from the microphone (on iOS), down sample it and write that to a '.aac' file.
I've finally gotten to point where it's almost working
let inputNode = audioEngine.inputNode
let inputFormat = inputNode.outputFormat(forBus: 0)
let bufferSize = UInt32(4096)
//let sampleRate = 44100.0
let sampleRate = 8000
let bitRate = sampleRate * 16
let fileUrl = url(appending: "NewRecording.aac")
print("Write to \(fileUrl)")
do {
outputFile = try AVAudioFile(forWriting: fileUrl,
settings: [
AVFormatIDKey: kAudioFormatMPEG4AAC,
AVSampleRateKey: sampleRate,
AVEncoderBitRateKey: bitRate,
AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue,
AVNumberOfChannelsKey: 1],
commonFormat: .pcmFormatFloat32,
interleaved: false)
} catch let error {
print("Failed to create audio file for \(fileUrl): \(error)")
return
}
recordButton.setImage(RecordingStyleKit.imageOfMicrophone(fill: .red), for: [])
// Down sample the audio to 8kHz
let fmt = AVAudioFormat(commonFormat: .pcmFormatFloat32, sampleRate: Double(sampleRate), channels: 1, interleaved: false)!
let converter = AVAudioConverter(from: inputFormat, to: fmt)!
inputNode.installTap(onBus: 0, bufferSize: AVAudioFrameCount(bufferSize), format: inputFormat) { (buffer, time) in
let inputCallback: AVAudioConverterInputBlock = { inNumPackets, outStatus in
outStatus.pointee = AVAudioConverterInputStatus.haveData
return buffer
}
let convertedBuffer = AVAudioPCMBuffer(pcmFormat: fmt,
frameCapacity: AVAudioFrameCount(fmt.sampleRate) * buffer.frameLength / AVAudioFrameCount(buffer.format.sampleRate))!
var error: NSError? = nil
let status = converter.convert(to: convertedBuffer, error: &error, withInputFrom: inputCallback)
assert(status != .error)
if let outputFile = self.outputFile {
do {
try outputFile.write(from: convertedBuffer)
}
catch let error {
print("Write failed: \(error)")
}
}
}
audioEngine.prepare()
do {
try audioEngine.start()
}
catch {
print(error.localizedDescription)
}
The problem is, the resulting file MUST be in MPEG ADTS, AAC, v4 LC, 8 kHz, monaural
format, but the code above only generates MPEG ADTS, AAC, v2 LC, 8 kHz, monaural
That is, it MUST be v4
, not v2
(I have no choice)
(This result is generated by using file {name}
on the command line to dump it's properties. I also use MediaInfo to provide additional information)
I've been trying to figure out if there is someway to provide a hint or setting to AVAudioFile
which will change the LC (Low Complexity) version from 2 to 4?
I've been scanning through the docs and examples but can't seem to find any suggestions