iOS核心音频生命周期-Avaudioionodeimpl.mm:365-要求的条件为false:hwformat



我正在开发一个iOS应用程序,它由2个主模块组成:核心音频上的音频分析模块基础,以及使用AudioKit的输出模块。

这是音频输入类:

import AVFoundation
typealias AudioInputCallback = (
    _ timeStamp: Double,
    _ numberOfFrames: Int,
    _ samples: [Float]
    ) -> Void
/// Sets up an audio input session and notifies when new buffer data is available.
class AudioInputUtility: NSObject {
    private(set) var audioUnit: AudioUnit!
    var audioSession : AVAudioSession = AVAudioSession.sharedInstance()
    var sampleRate: Float
    var numberOfChannels: Int
    /// When true, performs DC offset rejection on the incoming buffer before invoking the audioInputCallback.
    var shouldPerformDCOffsetRejection: Bool = false
    private let outputBus: UInt32 = 0
    private let inputBus: UInt32 = 1
    private var audioInputCallback: AudioInputCallback!
    /// Instantiate a AudioInput.
    /// - Parameter audioInputCallback: Invoked when audio data is available.
    /// - Parameter sampleRate: The sample rate to set up the audio session with.
    /// - Parameter numberOfChannels: The number of channels to set up the audio session with.
    init(audioInputCallback callback: @escaping AudioInputCallback, sampleRate: Float = 44100.0, numberOfChannels: Int = 1) { // default values if not specified
        self.sampleRate = sampleRate
        self.numberOfChannels = numberOfChannels
        audioInputCallback = callback
    }
    /// Start recording. Prompts for access to microphone if necessary.
    func startRecording() {
        do {
            if self.audioUnit == nil {
                setupAudioSession()
                setupAudioUnit()
            }
            try self.audioSession.setActive(true)
            var osErr: OSStatus = 0

            osErr =  AudioUnitInitialize(self.audioUnit)
            assert(osErr == noErr, "*** AudioUnitInitialize err (osErr)")
            osErr = AudioOutputUnitStart(self.audioUnit)
            assert(osErr == noErr, "*** AudioOutputUnitStart err (osErr)")
        } catch {
            print("*** startRecording error: (error)")
        }
    }
    /// Stop recording.
    func stopRecording() {
        do {
            var osErr: OSStatus = 0
            osErr = AudioOutputUnitStop(self.audioUnit)
            osErr = AudioUnitUninitialize(self.audioUnit)
            assert(osErr == noErr, "*** AudioUnitUninitialize err (osErr)")
            try self.audioSession.setActive(false)
        } catch {
            print("*** error: (error)")
        }
    }
    private let recordingCallback: AURenderCallback = { (inRefCon, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, ioData) -> OSStatus in
        let audioInput = unsafeBitCast(inRefCon, to: AudioInputUtility.self)
        var osErr: OSStatus = 0
        // We've asked CoreAudio to allocate buffers for us, so just set mData to nil and it will be populated on AudioUnitRender().
        var bufferList = AudioBufferList(
            mNumberBuffers: 1,
            mBuffers: AudioBuffer(
                mNumberChannels: UInt32(audioInput.numberOfChannels),
                mDataByteSize: 4,
                mData: nil))
        osErr = AudioUnitRender(audioInput.audioUnit,
                                ioActionFlags,
                                inTimeStamp,
                                inBusNumber,
                                inNumberFrames,
                                &bufferList)
        assert(osErr == noErr, "*** AudioUnitRender err (osErr)")

        // Move samples from mData into our native [Float] format.
        var monoSamples = [Float]()
        let ptr = bufferList.mBuffers.mData?.assumingMemoryBound(to: Float.self)
        monoSamples.append(contentsOf: UnsafeBufferPointer(start: ptr, count: Int(inNumberFrames)))
        if audioInput.shouldPerformDCOffsetRejection {
            DCRejectionFilterProcessInPlace(&monoSamples, count: Int(inNumberFrames))
        }
        // Not compatible with Obj-C...
        audioInput.audioInputCallback(inTimeStamp.pointee.mSampleTime / Double(audioInput.sampleRate),
                                      Int(inNumberFrames),
                                      monoSamples)
        return 0
    }
    private func setupAudioSession() {
        if !audioSession.availableCategories.contains(AVAudioSessionCategoryRecord) {
            print("can't record! bailing.")
            return
        }
        do {
            //https://developer.apple.com/reference/avfoundation/avaudiosession/1669963-audio_session_categories
            try audioSession.setCategory(AVAudioSessionCategoryRecord)
            // "Appropriate for applications that wish to minimize the effect of system-supplied signal processing for input and/or output audio signals."
            // NB: This turns off the high-pass filter that CoreAudio normally applies.

            try audioSession.setMode(AVAudioSessionModeMeasurement)
            try audioSession.setPreferredSampleRate(Double(sampleRate))
            // NB: This is considered a 'hint' and more often than not is just ignored.
            // number of seconds to record -> voglio 1024 samples
            try audioSession.setPreferredIOBufferDuration(0.05)
            audioSession.requestRecordPermission { (granted) -> Void in
                if !granted {
                    print("*** record permission denied")
                }
            }
        } catch {
            print("*** audioSession error: (error)")
        }
    }
    private func setupAudioUnit() {
        var componentDesc:AudioComponentDescription = AudioComponentDescription(
            componentType: OSType(kAudioUnitType_Output),
            componentSubType: OSType(kAudioUnitSubType_RemoteIO), // Always this for iOS.
            componentManufacturer: OSType(kAudioUnitManufacturer_Apple),
            componentFlags: 0,
            componentFlagsMask: 0)
        var osErr: OSStatus = 0
        // Get an audio component matching our description.
        let component: AudioComponent! = AudioComponentFindNext(nil, &componentDesc)
        assert(component != nil, "Couldn't find a default component")
        // Create an instance of the AudioUnit
        var tempAudioUnit: AudioUnit?
        osErr = AudioComponentInstanceNew(component, &tempAudioUnit)
        self.audioUnit = tempAudioUnit
        assert(osErr == noErr, "*** AudioComponentInstanceNew err (osErr)")
        // Enable I/O for input.
        var one:UInt32 = 1
        osErr = AudioUnitSetProperty(audioUnit,
                                     kAudioOutputUnitProperty_EnableIO,
                                     kAudioUnitScope_Input,
                                     inputBus,
                                     &one,
                                     UInt32(MemoryLayout<UInt32>.size))
        assert(osErr == noErr, "*** AudioUnitSetProperty err (osErr)")

        osErr = AudioUnitSetProperty(audioUnit,
                                     kAudioOutputUnitProperty_EnableIO,
                                     kAudioUnitScope_Output,
                                     outputBus,
                                     &one,
                                     UInt32(MemoryLayout<UInt32>.size))
        assert(osErr == noErr, "*** AudioUnitSetProperty err (osErr)")

        // Set format to 32 bit, floating point, linear PCM
        var streamFormatDesc:AudioStreamBasicDescription = AudioStreamBasicDescription(
            mSampleRate:        Double(sampleRate),
            mFormatID:          kAudioFormatLinearPCM,
            mFormatFlags:       kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved, // floating point data - docs say this is fastest
            mBytesPerPacket:    4,
            mFramesPerPacket:   1,
            mBytesPerFrame:     4,
            mChannelsPerFrame:  UInt32(self.numberOfChannels),
            mBitsPerChannel:    4 * 8,
            mReserved: 0
        )
        // Set format for input and output busses
        osErr = AudioUnitSetProperty(audioUnit,
                                     kAudioUnitProperty_StreamFormat,
                                     kAudioUnitScope_Input, outputBus,
                                     &streamFormatDesc,
                                     UInt32(MemoryLayout<AudioStreamBasicDescription>.size))
        assert(osErr == noErr, "*** AudioUnitSetProperty err (osErr)")

        osErr = AudioUnitSetProperty(audioUnit,
                                     kAudioUnitProperty_StreamFormat,
                                     kAudioUnitScope_Output,
                                     inputBus,
                                     &streamFormatDesc,
                                     UInt32(MemoryLayout<AudioStreamBasicDescription>.size))
        assert(osErr == noErr, "*** AudioUnitSetProperty err (osErr)")
        // Set up our callback.
        var inputCallbackStruct = AURenderCallbackStruct(inputProc: recordingCallback, inputProcRefCon: UnsafeMutableRawPointer(Unmanaged.passUnretained(self).toOpaque()))
        osErr = AudioUnitSetProperty(audioUnit,
                                     AudioUnitPropertyID(kAudioOutputUnitProperty_SetInputCallback),
                                     AudioUnitScope(kAudioUnitScope_Global),
                                     inputBus,
                                     &inputCallbackStruct,
                                     UInt32(MemoryLayout<AURenderCallbackStruct>.size))
        assert(osErr == noErr, "*** AudioUnitSetProperty err (osErr)")
        // Ask CoreAudio to allocate buffers for us on render. (This is true by default but just to be explicit about it...)
        osErr = AudioUnitSetProperty(audioUnit,
                                     AudioUnitPropertyID(kAudioUnitProperty_ShouldAllocateBuffer),
                                     AudioUnitScope(kAudioUnitScope_Output),
                                     inputBus,
                                     &one,
                                     UInt32(MemoryLayout<UInt32>.size))
        assert(osErr == noErr, "*** AudioUnitSetProperty err (osErr)")
    }
}
private func DCRejectionFilterProcessInPlace(_ audioData: inout [Float], count: Int) {
    let defaultPoleDist: Float = 0.975
    var mX1: Float = 0
    var mY1: Float = 0
    for i in 0..<count {
        let xCurr: Float = audioData[i]
        audioData[i] = audioData[i] - mX1 + (defaultPoleDist * mY1)
        mX1 = xCurr
        mY1 = audioData[i]
    }
}

这是输出类:

private func initPlayer(){
        do{
            /*
            let audioSession : AVAudioSession = AVAudioSession.sharedInstance()
            //try audioSession.setActive(false)
            try audioSession.setCategory(AVAudioSessionCategoryPlayback)
*/            
            // http://audiokit.io/playgrounds/Playback/Reading%20and%20Writing%20Audio%20Files/
            let file = try AKAudioFile(readFileName: self.soundPath,
                                       baseDir: .resources)
            self.player = try AKAudioPlayer(file: file)
            //player options
            self.player!.looping = true


            AKSettings.playbackWhileMuted = true
            try AKSettings.setSession(category: .playback)
AudioKit.output = self.player

        }catch{
            print("Unresolved error (error)")
        }
    }

public func stopMaskingSound(){
            if(player!.isPlaying){
                self.player!.stop()
            }
            if audioKitIsStarted == true{
                AudioKit.stop()            
                self.audioKitIsStarted = false
            }

        }

您可以看到音频输入和输出由2个不同的类管理。

我遇到的问题是,如果我执行此步骤:1)初始玩家和记录 ->停止它2)播放输出 ->停止3)Reinit Player

在第三步,我有以下例外:

[central] 54:   ERROR:    [0x16dfc3000] >avae> AVAudioIONodeImpl.mm:365: _GetHWFormat: required condition is false: hwFormat
*** Terminating app due to uncaught exception 'com.apple.coreaudio.avfaudio', reason: 'required condition is false: hwFormat'

有人知道它与什么相关吗?Audiokit&lt; -> Core Audio是否有生命周期问题?

停止并重新启动音频单元可能会出现问题,因为音频过程的某些部分确实在另一个线程或线程中停止。周围的一项可能的工作可能是在停止和重新启动之间延迟约1秒钟,以便在尝试重新启动之前,将远程io异步滑到停止。

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