i am working on an app that shows live generated content in full screen on my new macbook pro with monterey. it is meant to be projected in theatres etc. but because i use microphone input there is a visible orange dot in the upper right corner. how can i get rid of it?

I have a kind of trivial request, I need to seek sound playback. The problem is I don't have a local sound file, I got a pointer to the sound instead (as well as other params), from (my internal) native lib. There is a method that I use in order to convert UnsafeRawPointer to AVAudioPCMBuffer ... var byteCount: Int32 = 0       var buffer: UnsafeMutableRawPointer?               defer {         buffer?.deallocate()         buffer = nil       }               if audioReader?.getAudioByteData(byteCount: &byteCount, data: &buffer) ?? false && buffer != nil {         let audioFormat = AVAudioFormat(standardFormatWithSampleRate: Double(audioSampleRate), channels: AVAudioChannelCount(audioChannels))!         if let pcmBuf = AVAudioPCMBuffer(pcmFormat: audioFormat, frameCapacity: AVAudioFrameCount(byteCount)) {           let monoChannel = pcmBuf.floatChannelData![0]           pcmFloatData = [Float](repeating: 0.0, count: Int(byteCount))                       //>>> Convert UnsafeMutableRawPointer to [Int8] array           let int16Ptr: UnsafeMutablePointer<Int16> = buffer!.bindMemory(to: Int16.self, capacity: Int(byteCount))           let int16Buffer: UnsafeBufferPointer<Int16> = UnsafeBufferPointer(start: int16Ptr, count: Int(byteCount) / MemoryLayout<Int16>.size)           let int16Arr: [Int16] = Array(int16Buffer)           //<<<                       // Int16 ranges from -32768 to 32767 -- we want to convert and scale these to Float values between -1.0 and 1.0           var scale = Float(Int16.max) + 1.0           vDSP_vflt16(int16Arr, 1, &pcmFloatData[0], 1, vDSP_Length(int16Arr.count)) // Int16 to Float           vDSP_vsdiv(pcmFloatData, 1, &scale, &pcmFloatData[0], 1, vDSP_Length(int16Arr.count)) // divide by scale              memcpy(monoChannel, pcmFloatData, MemoryLayout<Float>.size * Int(int16Arr.count))           pcmBuf.frameLength = UInt32(int16Arr.count)                       usagePlayer.setupAudioEngine(with: audioFormat)           audioClip = pcmBuf         }       } ... So, at the end of the method, you can see this line audioClip = pcmBuf, where the prepared pcmBuf is pass to the local variable. Then, what I need is just to start it like this ... /*player is AVAudioPlayerNode*/ player.scheduleBuffer(buf, at: nil, options: .loops) player.play() ... and that is it, now I can hear the sound. But let's say I need to seek forward in 10 sec, in order to do this I need stop() the player node, set a new AVAudioPCMBuffer but this time with an offset of 10 sec. The problem is there is no method for offset, nor from the player node side nor AVAudioPCMBuffer. For example, if I would work with a file (instead of a buffer), I could use this method ... player.scheduleSegment(         file,         startingFrame: seekFrame,         frameCount: frameCount,         at: nil       ) ... There at least you can use startingFrame: seekFrame and frameCount: frameCount params. But in my case I don't use file I use buffer and the problem is that - there is no such params for buffer implementation. Looks like I can't implement seek logic if I use AVAudioPCMBuffer. What am I doing wrong?

I have an AVMutableAudioMix and use MTAudioProcessingTap to process the audio data.But After I pass the buffer to AVAudioEngine and to render it with renderOffline,the audio has no any effects...How can I do it? Any idea? Here is the code for MTAudioProcessingTapProcessCallback var callback = MTAudioProcessingTapCallbacks(version: kMTAudioProcessingTapCallbacksVersion_0, clientInfo:UnsafeMutableRawPointer(Unmanaged.passUnretained(self.engine).toOpaque()), init: tapInit, finalize: tapFinalize, prepare: tapPrepare, unprepare: tapUnprepare) { tap, numberFrames, flags, bufferListInOut, numberFramesOut, flagsOut in                       guard MTAudioProcessingTapGetSourceAudio(tap, numberFrames, bufferListInOut, flagsOut, nil, numberFramesOut) == noErr else {         preconditionFailure()       }       let storage = MTAudioProcessingTapGetStorage(tap)       let engine = Unmanaged<Engine>.fromOpaque(storage).takeUnretainedValue()       // render the audio with effect       engine.render(bufferPtr: bufferListInOut,numberOfFrames: numberFrames)     } And here is the Engine code class Engine {   let engine = AVAudioEngine()       let player = AVAudioPlayerNode()   let pitchEffect = AVAudioUnitTimePitch()   let reverbEffect = AVAudioUnitReverb()   let rateEffect = AVAudioUnitVarispeed()   let volumeEffect = AVAudioUnitEQ()   let format = AVAudioFormat(commonFormat: .pcmFormatFloat32, sampleRate: 44100, channels: 2, interleaved: false)!   init() {     engine.attach(player)     engine.attach(pitchEffect)     engine.attach(reverbEffect)     engine.attach(rateEffect)     engine.attach(volumeEffect)           engine.connect(player, to: pitchEffect, format: format)     engine.connect(pitchEffect, to: reverbEffect, format: format)     engine.connect(reverbEffect, to: rateEffect, format: format)     engine.connect(rateEffect, to: volumeEffect, format: format)     engine.connect(volumeEffect, to: engine.mainMixerNode, format: format)           try! engine.enableManualRenderingMode(.offline, format: format, maximumFrameCount: 4096)           reverbEffect.loadFactoryPreset(AVAudioUnitReverbPreset.largeRoom2)     reverbEffect.wetDryMix = 100     pitchEffect.pitch = 2100           try! engine.start()     player.play()   }       func render(bufferPtr:UnsafeMutablePointer<AudioBufferList>,numberOfFrames:CMItemCount) {     let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: 4096)!     buffer.frameLength = AVAudioFrameCount(numberOfFrames)     buffer.mutableAudioBufferList.pointee = bufferPtr.pointee     self.player.scheduleBuffer(buffer) {       try! self.engine.renderOffline(AVAudioFrameCount(numberOfFrames), to: buffer)     }   } }

I have a simple audio graph, input, effect audioUnit, mixer and out. When I include my effect audioUnit, I lose my audio output. I believe the graph shows the problem, I just don't know how to interpret it. ________ GraphDescription ________ AVAudioEngineGraph 0x7fd2b7c08c90: initialized = 1, running = 1, number of nodes = 6 ******** output chain ******** node 0x6000016b3f00 {'auou' 'rioc' 'appl'}, 'I' inputs = 1 (bus0, en1) <- (bus0) 0x6000016b7580, {'aumx' 'mcmx' 'appl'}, [ 2 ch, 44100 Hz, 'lpcm' (0x00000029) 32-bit little-endian float, deinterleaved] node 0x6000016b7580 {'aumx' 'mcmx' 'appl'}, 'I' inputs = 1 (bus0, en1) <- (bus0) 0x600001680d80, {'aufx' 'vpio' 0x00000000}, [ 2 ch, 44100 Hz, 'lpcm' (0x00000029) 32-bit little-endian float, deinterleaved] outputs = 1 (bus0, en1) -> (bus0) 0x6000016b3f00, {'auou' 'rioc' 'appl'}, [ 2 ch, 44100 Hz, 'lpcm' (0x00000029) 32-bit little-endian float, deinterleaved] node 0x600001680d80 {'aufx' 'vpio' 0x00000000}, 'I' inputs = 1 (bus0, en1) <- (bus0) 0x600000abb330, {'aufc' 'conv' 'appl'}, [ 2 ch, 44100 Hz, 'lpcm' (0x00000029) 32-bit little-endian float, deinterleaved] outputs = 1 (bus0, en1) -> (bus0) 0x6000016b7580, {'aumx' 'mcmx' 'appl'}, [ 2 ch, 44100 Hz, 'lpcm' (0x00000029) 32-bit little-endian float, deinterleaved] node 0x600000abb330 {'aufc' 'conv' 'appl'}, 'I' outputs = 1 (bus0, en1) -> (bus0) 0x600001680d80, {'aufx' 'vpio' 0x00000000}, [ 2 ch, 44100 Hz, 'lpcm' (0x00000029) 32-bit little-endian float, deinterleaved] ******** input chain ******** node 0x600001681100 {'auou' 'rioc' 'appl'}, 'I' ******** other nodes ******** node 0x600001685080 {'aumx' 'mcmx' 'appl'}, 'U' I'm guessing these 0's are telling me something: node 0x600001680d80 {'aufx' 'vpio' 0x00000000}, 'I' inputs = 1 Any assistance is really appreciated. I encountered this, thought it would be simple to diagnose and fix, but I find myself spinning my wheels.

How would I insert audio file to button from iCloud and have that file be playable?

I’m trying to do something really complex with audio streams. I.e. process the stream live edit it and then save it in snippets, all while the user is still speaking. I’m a book person, and reading hardcopy documentation is much easier for me.

I recently released my first ShazamKit app, but there is one thing that still bothers me. When I started I followed the steps as documented by Apple right here : https://developer.apple.com/documentation/shazamkit/shsession/matching_audio_using_the_built-in_microphone however when I was running this on iPad I receive a lot of high pitched feedback noise when I ran my app with this configuration. I got it to work by commenting out the output node and format and only use the input. But now I want to be able to recognise the song that’s playing from the device that has my app open and was wondering if I need the output nodes for that or if I can do something else to prevent the Mic. Feedback from happening. In short: What can I do to prevent feedback from happening Can I use the output of a device to recognise songs or do I just need to make sure that the microphone can run at the same time as playing music? Other than that I really love the ShazamKit API and can highly recommend to have a go with it! This is the code as documented in the above link (I just added the comments of what broke it for me) func configureAudioEngine() { // Get the native audio format of the engine's input bus. let inputFormat = audioEngine.inputNode.inputFormat(forBus: 0) // THIS CREATES FEEDBACK ON IPAD PRO let outputFormat = AVAudioFormat(standardFormatWithSampleRate: 48000, channels: 1) // Create a mixer node to convert the input. audioEngine.attach(mixerNode) // Attach the mixer to the microphone input and the output of the audio engine. audioEngine.connect(audioEngine.inputNode, to: mixerNode, format: inputFormat) // THIS CREATES FEEDBACK ON IPAD PRO audioEngine.connect(mixerNode, to: audioEngine.outputNode, format: outputFormat) // Install a tap on the mixer node to capture the microphone audio. mixerNode.installTap(onBus: 0, bufferSize: 8192, format: outputFormat) { buffer, audioTime in // Add captured audio to the buffer used for making a match. self.addAudio(buffer: buffer, audioTime: audioTime) } }

I understand that both the AirPods Pro and AirPods Max have an outer microphone whose purpose is to record the environment and replay processed audio under "transparency mode." I am wishing to access the samples recorded by this microphone and run them through processing via the AVAudioEngine. Is the audio from this microphone accessible via Xcode or is it only internally wired? If the former, how does one access theses samples?

Am trying to go from the installTap straight to AVAudioFile(forWriting: I call: let recordingFormat = node.outputFormat(forBus: 0) and I get back : <AVAudioFormat 0x60000278f750:  1 ch,  48000 Hz, Float32> But AVAudioFile has a settings parameter of [String : Any] and am curious of how to place those values into recording the required format. Hopefully these are the values I need?

Hello, Am starting to work with/learn the AVAudioEngine. Currently am at the point where I would like to be able read an audio file of a speech and determine if there are any moments of silence in the speech. Does this framework provide any such properties, such as power lever, decibels, etc. that I can use in finding long enough moments of silence?

Hello, I would like to create a VoIP App I want to build a sound visualizer for the voice of the other party. example: https://medium.com/swlh/swiftui-create-a-sound-visualizer-cadee0b6ad37 The call function was implemented using the call kit. (Based on twilio's quick-start) https://jp.twilio.com/docs/voice/sdks/ios/get-started https://github.com/twilio/voice-quickstart-ios After importing AVFoundation, have to need routing the AVAudioengine mixer. I have no idea how to put voice in the mixer. Any guidance is appreciated! Thank you!

Hello, I am trying to use AVAudioFile to save audio buffer to .wav file. The buffer is of type [Float]. Currently I am able to successfully create the .wav files and even play them, but they are blank - I cannot hear any sound. private func saveAudioFile(using buffer: [Float]) { let fileUrl = FileManager.default.urls(for: .documentDirectory, in: .userDomainMask).first!.appendingPathComponent("\(UUID().uuidString).wav") let fileSettings = [ AVFormatIDKey: Int(kAudioFormatLinearPCM), AVSampleRateKey: 15600, AVNumberOfChannelsKey: 1, AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] guard let file = try? AVAudioFile(forWriting: fileUrl, settings: fileSettings, commonFormat: .pcmFormatInt16, interleaved: true) else { print("Cannot create AudioFile") return } guard let bufferFormat = AVAudioFormat(settings: settings) else { print("Cannot create buffer format") return } guard let outputBuffer = AVAudioPCMBuffer(pcmFormat: bufferFormat, frameCapacity: AVAudioFrameCount(buffer.count)) else { print("Cannot create output buffer") return } for i in 0..<buffer.count { outputBuffer.int16ChannelData!.pointee[i] = Int16(buffer[i]) } outputBuffer.frameLength = AVAudioFrameCount(buffer.count) do { try file.write(from: outputBuffer) } catch { print(error.localizedDescription) print("Write to file failed") } } Where should I be looking first for the problem? Is it format issue? I am getting the data from the microphone with the AVAudioEngine. Its format is created like this: let outputFormat = AVAudioFormat(commonFormat: .pcmFormatInt16, sampleRate: Double(15600), channels: 1, interleaved: true)! And here is the installTap implementation with the buffer callback: input.installTap(onBus: 0, bufferSize: AVAudioFrameCount(sampleRate*2), format: inputFormat) { (incomingBuffer, time) in DispatchQueue.global(qos: .background).async { let pcmBuffer = AVAudioPCMBuffer(pcmFormat: outputFormat, frameCapacity: AVAudioFrameCount(outputFormat.sampleRate * 2.0)) var error: NSError? = nil let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in outStatus.pointee = AVAudioConverterInputStatus.haveData return incomingBuffer } formatConverter.convert(to: pcmBuffer!, error: &error, withInputFrom: inputBlock) if error != nil { print(error!.localizedDescription) } else if let pcmBuffer = pcmBuffer, let channelData = pcmBuffer.int16ChannelData { let channelDataPointer = channelData.pointee self.buffer = stride(from: 0, to: self.windowLengthSamples, by: 1).map { Float(channelDataPointer[$0]) / 32768.0 } onBufferUpdated(self.buffer) } } } The onBufferUpdated is the block that provides [Float] for the saveAudioFile method above. I have tried some experiements with different output formats, but that ended up with unplayable audio files.

I have my Swift app that records audio in chunks of multiple files, each M4A file is approx 1 minute long. I would like to go through those files and detect silence, or the lowest level. While I am able to read the file into a buffer, my problem is deciphering it. Even with Google, all it comes up with is "audio players" instead of sites that describe the header and the data. Where can I find what to look for? Or even if I should be reading it into a WAV file? But even then I cannot seem to find a tool, or a site, that tells me how to decipher what I am reading. Obviously it exists, since Siri knows when you've stopped speaking. Just trying to find the key.

Is there any possible way to produce a system for my macOS program that will: -Allow the user to pick whether my program will output its audio either to the system output or to an AirPlay destination? -While doing so, offer the ability to control the volume of the asset currently playing? (AVPlayer 'volume' setter stops responding when connected to an AirPlay endpoint) -Also allow me to attach a 10-band EQ to the output? I tried to do this five years ago, in 2017, and expected the ecosystem to have improved by now. While AVRoutePickerView and AVPlayer are user-friendly and convenient, the fact that basic functionality like the volume control ceases functioning over AirPlay is quite frustrating. AVAudioPlayer seems like it might offer this functionality, but only on iOS and not on macOS! I am basically only trying to offer the same AirPlay controls that Music.app does. Is this really so difficult?

I am working on music application where multiple wav files are scheduling within time frame. Everything is working perfect except one scenario where there is small beef is coming while scheduling player node again. For example - one.wav is playing on PlayerNode1 and now I am rescheduling to second.wav after 2 second then there is small beep is coming. I have tried to stop node by checking isPlaying condition. Still it is not working. Am I doing anything wrong here. if playerNode.isPlaying { playerNode.stop() } playerNode.scheduleFile(audioFile, at: nil, completionHandler: nil) playerNode.play() I am using same player node for performance as there are 24 wav files that needs to be played in 1 minute so there is no point to keeping all player nodes. How would stop beef while rescheduling new audio file for same player node? I have shared link below to check issue. https://drive.google.com/file/d/1FjZtLUj_wUp0LQPyjIwfJNy67HWUlt0I/view?usp=sharing Expected result should be song continuity

I'm using an AVAudioConverter object to decode an OPUS stream for VoIP. The decoding itself works well, however, whenever the stream stalls (no more audio packet is available to decode because of network instability) this can be heard in crackling / abrupt stop in decoded audio. OPUS can mitigate this by indicating packet loss by passing a null pointer in the C-library to int opus_decode_float (OpusDecoder * st, const unsigned char * data, opus_int32 len, float * pcm, int frame_size, int decode_fec), see https://opus-codec.org/docs/opus_api-1.2/group__opus__decoder.html#ga9c554b8c0214e24733a299fe53bb3bd2. However, with AVAudioConverter using Swift I'm constructing an AVAudioCompressedBuffer like so:         let compressedBuffer = AVAudioCompressedBuffer(             format: VoiceEncoder.Constants.networkFormat,             packetCapacity: 1,             maximumPacketSize: data.count         )         compressedBuffer.byteLength = UInt32(data.count)         compressedBuffer.packetCount = 1   compressedBuffer.packetDescriptions! .pointee.mDataByteSize = UInt32(data.count)         data.copyBytes(             to: compressedBuffer.data .assumingMemoryBound(to: UInt8.self),             count: data.count         ) where data: Data contains the raw OPUS frame to be decoded. How can I specify data loss in this context and cause the AVAudioConverter to output PCM data whenever no more input data is available? More context: I'm specifying the audio format like this:         static let frameSize: UInt32 = 960         static let sampleRate: Float64 = 48000.0         static var networkFormatStreamDescription = AudioStreamBasicDescription(             mSampleRate: sampleRate,             mFormatID: kAudioFormatOpus,             mFormatFlags: 0,             mBytesPerPacket: 0,             mFramesPerPacket: frameSize,             mBytesPerFrame: 0,             mChannelsPerFrame: 1,             mBitsPerChannel: 0,             mReserved: 0         )         static let networkFormat = AVAudioFormat( streamDescription: &networkFormatStreamDescription )! I've tried 1) setting byteLength and packetCount to zero and 2) returning nil but setting .haveData in the AVAudioConverterInputBlock I'm using with no success.