I have an app (iPhone / iPad) that currently uses miniaudio and I'd to transition to a pure CoreAudio solution, and I cannot for the life of me get it to work. I want to set up a remoteIO with microphone and speaker callbacks, so that I get a callback when the microphone (USB microphone at 384kHz, mono) has samples for me, and I get a callback when the speakers (49kHz, stereo) need more samples. It should be insanely simple. It's Objective C as I have never got round to Swift and can't see it happening this late in life, but that shouldn't change things. So if anyone can tell me what I'm doing wrong here it would be GREATLY appreciated. My playbackCallback is never being fired, only the recordCallback. `-(void) launchRemoteIOandSleep { NSError *error; [ [AVAudioSession sharedInstance] setPreferredIOBufferDuration:(1024.0f / 48000.0f) error:&error ]; OSStatus status; AudioComponentInstance audioUnit; // set up callback structures - different sanity clauses to identify in breakpoints renderCallBackHandle sanityClause666; sanityClause666.remoteIO = audioUnit; sanityClause666.sanityCheck666 = 666; renderCallBackHandle sanityClause667; sanityClause667.remoteIO = audioUnit; sanityClause667.sanityCheck666 = 667; // set up audio formats AudioStreamBasicDescription audioFormatInput; FillOutASBDForLPCM(audioFormatInput,384000.0,1,16,16,0,0,0); AudioStreamBasicDescription audioFormatOutput; FillOutASBDForLPCM(audioFormatOutput,48000.0,2,16,16,0,0,0); // set up callback structs AURenderCallbackStruct callbackStructRender; callbackStructRender.inputProc = playbackCallback; callbackStructRender.inputProcRefCon = &sanityClause666; AURenderCallbackStruct callbackStructRecord; callbackStructRecord.inputProc = recordCallback; callbackStructRecord.inputProcRefCon = &sanityClause667; // grab remoteIO AudioComponentDescription desc; desc.componentType = kAudioUnitType_Output; desc.componentSubType = kAudioUnitSubType_RemoteIO; desc.componentManufacturer = kAudioUnitManufacturer_Apple; desc.componentFlags = 0; desc.componentFlagsMask = 0; AudioComponent component = AudioComponentFindNext(NULL, &desc); // Get audio unit status = AudioComponentInstanceNew(component, &audioUnit); checkStatus(status); // Enable IO for both recording and playback // this enables the OUTPUT side of the OUTPUT bus which is the speaker (I thnk ... ) UInt32 flag = 1; status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, kOutputBus, &flag, sizeof(flag)); checkStatus(status); // this enables the INPUT side of the INPUT bus which is the mic (I thnk ... ) flag = 1; status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, kInputBus, &flag, sizeof(flag)); checkStatus(status); // Apply format - INPUT bus of OUTPUT SCOPE which is my samples into remoteIO status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &audioFormatOutput, sizeof(audioFormatOutput)); checkStatus(status); // Apply format - OUTPUT bus of INPUT SCOPE which is where I pick up my samples from mic status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, kOutputBus, &audioFormatInput, sizeof(audioFormatInput)); checkStatus(status); // set output callback status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Output, kInputBus, &callbackStructRender, sizeof(callbackStructRender)); checkStatus(status); // Set input callback status = AudioUnitSetProperty(audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Input, kOutputBus, &callbackStructRecord, sizeof(callbackStructRecord)); checkStatus(status); // Disable buffer allocation for the recorder flag = 0; status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_ShouldAllocateBuffer, kAudioUnitScope_Input, kInputBus, &flag, sizeof(flag)); // Initialise status = AudioUnitInitialize(audioUnit); checkStatus(status); status = AudioOutputUnitStart(audioUnit); checkStatus(status); [ self waitForAudioStabilisation ]; while (1) sleep(2); }`

I am working on a VoIP based PTT app. Uses 'voip' apns notification type to get to know about new incoming PTT call. When my app receives a PTT call, the app plays audio. But the call audio is not heard. While checking the phone volume, the API [[AVAudioSession sharedInstance] outputVolume] returns 0. But clearly the phone volume is not zero. On checking the phone volume by pressing side volume button, the volume is above 50%. This behavior is observed in both app foreground and background scenario. Why does the API return zero volume level ? Is there any other reason why the app volume is not heard ?

So I just got some airpods pro, and they wont connect to my phon. Im wondering if it is because of my phone running iOS 18, and i would like advice on how to fix this. (my phone is running smoothly with no noticeable issues, also my primary device)

In my iOS app, I've the functionality to record audio and video using the AVFoundation framework. While audio recording works smoothly on some devices, such as iPads and certain others, I'm encountering issues with newer models like iPhone 14, 14 Pro, and 15 series. Specifically, when attempting to initiate audio recording by tapping the microphone icon, the interface becomes unresponsive and remains static. This issue surfaced following an update to iOS 17.2.1. It seems to affect only a subset of devices, despite video recording functioning correctly across all devices.

Hi, I'm looking to implement PHASEStreamNode in Unity, but the current provided PHASE library for Unity doesn't contain this new typos of nodes yet. https://developer.apple.com/documentation/phase/phasestreamnode When you will be looking into releasing the beta of the Unity Plugins as well? This is very important for spatial audio in Unity to be consistent with Apple's standards. Best, Antonio

I'm trying to play an H265 video in opus audio format using iOS HLS, and a video slice file in fmp4 format, and I found that on iPhone13 ios 17.5.1, it plays fine, but on iPhone X ios 16.5.1, it doesn't play. Is it because some iOS versions do not support HLS in the opus audio format?

I am trying to use hls to play audio and video on iOS devices, the audio format is opus, sliced files use fmp4 format, on iPhone13 ios 17.5.1, it works fine, but on iPhoneX 16.5.1, it does not play, Is it because hls does not support the opus audio format on some versions?

Dear Sirs, I've written an audio driver based on IOUserAudioDevice. In my IOOperationHandler I can receive and send the audio samples as expected. Is there any way to configure the number of samples transferred in each call? Currently it seem to be around 512 samples per call, which relates to 10.7 millisecs when operating on 48 kHz samplerate. I'd like to achieve something like 48 or 96 samples per call. I did some experiments and tried calls to SetOutputLatency() etc. but so far I didn't find the right way to change the in_io_buffer_frame_size in the callback. I'd like to do this as smaller buffer sizes would allow lower latencies for the subsequent audio processing. Thanks and best regards, Johannes

I'm a newby at tvOS and want to know, if it is possible to override some system settings. Especially I want to override the output to audio devices. For the moment (using Apple TV 4K and tvOS 17) you can only select one device (TV, eARC, airPods) and I need a possibility to set a simultaneous output to airPods and TV or eARC. Is such a programming possible?

在我们App中，打开一个H5页面，使用webplayer播放H5中的视频。 然后再去播放App的播放器，播放视频、或音频文件， 都存在抢不到音频焦点问题，声音响一下就停了，播放器还在运行。 尝试在每次App播放都先调用setCategory、setActive也不生效。 这个问题，在beta1~beta3都存在。 请问，webkit的 player做了什么处理，会一直锁定着音频焦点，App要怎么处理才能把焦点拿过来？ In our App, open an H5 page and use webplayer to play the video in H5. Then go to the PlayApp player to play the video or audio file. There is a problem of not being able to grab the audio focus. The sound stops as soon as it sounds, but the player is still running. Trying to call setCategory and setActive every time in AppPlay does not work either. This problem exists in beta1~beta3. I would like to ask, what processing has been done by the webkit player to keep the audio focus locked? How can the app handle it so that it can take the focus?

AVAudioFormat has no Swift concurrency annotations but the documentation states "Instances of this class are immutable." This made me always assume it was safe to pass AVAudioFormat instances around. Is this the case? If so can it be marked as Sendable? Am I missing something?

I have an iPad Pro 12.9". I am looking to make an app which can take a simultaneous audio recording from two different microphones at the same time. I want to be able to specify which of the 5 built-in microphones each audio stream should use - ideally one should be from the microphone on the left side of the iPad, and the other should be from one of the mics at the top of the iPad. Is this possible to achieve with the API? The end goal here is to be able to use the two audio streams and do some DSP on the recordings to determine the approximate direction a particular sound comes from.

Hello, I hope this message finds you well. I am currently working on a Unity-based iOS application that requires continuous microphone input while also producing sound outputs. For this we need to use iOS echo cancellation, so some sounds need to be played via the iOS layer w/ echo cancellation, I am manually setting up the Audio Session after the app starts. Using the .playAndRecord mode of AVAudioSession. However, I am facing an issue where the volume of the sound output is inconsistent across different iOS devices and scenarios. The process is quite simple, for each AudioClip we are about to play via unity, we copy the buffer data to our iOS Swift layer, which then does all the processing then plays the audio via the native layer. Here are the specific issues I am encountering: The volume level for the game sound effects fluctuate between a normal audible volume and a very low volume. The sound output behaves differently depending on whether the app is launched with the device at full volume or on mute, and if the app is put into background and in foreground afterwards. The volume inconsistency affects my game negatively, as it is very hard to hear some audios, regardless of the device or its initial volume state. I have followed the basic setup for AVAudioSession as per the documentation, but the inconsistencies persist. I'm also aware that Unity uses FMOD to set up the audio routing in iOS, we configure our custom routing after that. We tried tweaking the output volume prior to playing an audio so there isn't much discrepancy, this seems to align the output volume, however there is still some places where the volume is super low, I've looked into the waveforms in Unity and they all seem consistent, there is no reason why the volume would take a dip. private var audioPlayer = AVAudioPlayerNode() @objc public func Play() { audioPlayer.volume = AVAudioSession.sharedInstance().outputVolume * 0.25 audioPlayer.play() } We also explored changing the audio session options to see if we had any luck but unfortunately nothing has changed. private func ConfigAudioSession() { let audioSession = AVAudioSession.sharedInstance(); do { try audioSession.setCategory(.playAndRecord, options: [.mixWithOthers, .allowBluetooth, .defaultToSpeaker]); try audioSession.setMode(.spokenAudio) try audioSession.setActive(true); } catch { //Treat error } } Could anyone provide guidance or suggest best practices to ensure a stable and consistent volume output in this scenario? Any advice on this issue would be greatly appreciated. Thank you in advance for your help!

Hello, I am building a new iOS app which uses AVSpeechSynthesizer and should be able to mix audio nicely with audio from other apps. AVSpeechSynthesizer seems to handle setting the AVAudioSession to active on it's own, but does not deactivate the audio session. This leads to issues, namely that other audio sources remain "ducked" after AVSpeechSynthesizer is done speaking. I have implemented deactivating the audio session myself, which "works", in that it allows other audio sources to become "un-ducked", but it throws this exception each time even though it appears successful. Error Domain=NSOSStatusErrorDomain Code=560030580 "Session deactivation failed" UserInfo={NSLocalizedDescription=Session deactivation failed} It appears to be a bug with how AVSpeechSynthesizer handles activating/deactivating the audio session. Below is a minimal example which illustrates the problem. It has two buttons, one which manually deactivates the audio sessions, which throws the exception, but otherwise works, and another button which leaves audio session management to the AVSpeechSynthesizer but does not "un-duck" other audio. If you play some audio from another app (ex: Music), you'll see the button which throws/catches an exception successfully ducks/un-ducks the audio, while the one without attempting to deactivate the session ducks but does not un-duck the audio. import AVFoundation struct ContentView: View { let workingSynthesizer = UnduckingSpeechSynthesizer() let brokenSynthesizer = BrokenSpeechSynthesizer() init() { let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playback, mode: .voicePrompt, options: [.duckOthers]) } catch { print("Setup error info: \(error)") } } var body: some View { VStack { Button("Works Correctly"){ workingSynthesizer.speak(text: "Hello planet") } Text("-------") Button("Does not work"){ brokenSynthesizer.speak(text: "Hello planet") } } .padding() } } class UnduckingSpeechSynthesizer: NSObject { var synth = AVSpeechSynthesizer() let audioSession = AVAudioSession.sharedInstance() override init(){ super.init() synth.delegate = self } func speak(text: String) { let utterance = AVSpeechUtterance(string: text) synth.speak(utterance) } } extension UnduckingSpeechSynthesizer: AVSpeechSynthesizerDelegate { func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { do { try audioSession.setActive(false, options: .notifyOthersOnDeactivation) } catch { // always throws an error // Error Domain=NSOSStatusErrorDomain Code=560030580 "Session deactivation failed" UserInfo={NSLocalizedDescription=Session deactivation failed} print("Deactivate error info: \(error)") } } } class BrokenSpeechSynthesizer { var synth = AVSpeechSynthesizer() let audioSession = AVAudioSession.sharedInstance() func speak(text: String) { let utterance = AVSpeechUtterance(string: text) synth.speak(utterance) } } (I have a separate issue where the first speech attempt takes a few seconds but I don't think it's related)

Hello Apple Community, I am developing an iOS app and would like to add a feature that allows users to play and organize Audible.com files within the app. Does Audible or the App Store provide any API or SDK for third-party apps to access and manage Audible content? If so, could you please provide some guidance on how to integrate it into my app? Thank you for your assistance! Best regards, Yes it labs

Music app stops playing when switching to the background In apps that play music or music files, if you move to the home screen or run another app while the app is running, the music playback stops. Our app does not have the code to stop playing when switching to the background. We are guessing that some people experience this and others do not. We usually guide users to reboot their devices and try again. How can this phenomenon be improved in the code? Or is this a bug or error in the OS?

MPMusicPlayerControllers nowPlayingItem no longer seems to be able to change a song. The code use to work but seems to be broken on iOS 16, 17 and now the iOS 18 beta. When newSong is triggered, the song restarts but it does not change songs. Instead I get the following error: Failed to set now playing item error=<MPMusicPlayerControllerErrorDomain.5 "Unable to play item <MPConcreteMediaItem: 0x9e9f0ef70> 206357861099970620" {}>. The documentation seems to indicate I’m doing things correctly. class MusicPlayer { var songTwo: MPMediaItem? let player = MPMusicPlayerController.applicationMusicPlayer func start() async { await MPMediaLibrary.requestAuthorization() let myPlaylistsQuery = MPMediaQuery.playlists() let playlists = myPlaylistsQuery.collections!.filter { $0.items.count > 2} let playlist = playlists.first! let songOne = playlist.items.first! songTwo = playlist.items[1] player.setQueue(with: playlist) play(songOne) } func newSong() { guard let songTwo else { return } play(songTwo) } private func play(_ song: MPMediaItem) { player.stop() player.nowPlayingItem = song player.prepareToPlay() player.play() } }

HI, I'm developing an iOS app that accepts an audio signal as input with the goal of analyzing the signal. For my experiment I purchased a cheap ADC-DAC produced by Sabrent. It works well but the sampling rate is 44.1 khz but I need at least something more (96 khz). I'm looking around but I find many DACs used to connect headphones. Can any of you suggest me an ADC-DAC, preferably not too expensive with a sampling rate of at least 96khz, working with iphones?

I have a user who is reporting an error and has been kind enough to share screen recordings to help diagnose. I am not experiencing this error, nor am I able to replicate on other devices I've tried, so I'm stuck trying to fix. His & other devices tested were all running iOS 17.5.1. Any details on the cause of this error or potential workarounds I could use to resolve would be greatly appreciated. try await ApplicationMusicPlayer.shared.play() throws: The operation couldn't be completed (MPMusicPlayerControllerErrorDomain error 6.) MusicAuthorization.currentStatus is .authorized ApplicationMusicPlayer.shared.isPreparedToPlay is false ApplicationMusicPlayer.shared.queue.currentEntry is nil (I've noticed this to be the case even when I am able to successfully play as well) Queue was loaded using ApplicationMusicPlayer.shared.queue = [album] but I also tried ApplicationMusicPlayer.shared.queue = ApplicationMusicPlayer.Queue(album:startingAt:) and it made no difference. album.playParameters are correct. He experiences the error when attempting to play any album. Any and all help is truly appreciated. Feedback Assistant filed has gone unanswered.

I am running the code sample here https://developer.apple.com/documentation/avfoundation/speech_synthesis/ in a REPL on Mac OS Ventura import AVFoundation // Create an utterance. let utterance = AVSpeechUtterance(string: "The quick brown fox jumped over the lazy dog.") // Configure the utterance. utterance.rate = 0.57 utterance.pitchMultiplier = 0.8 utterance.postUtteranceDelay = 0.2 utterance.volume = 0.8 // Retrieve the British English voice. let voice = AVSpeechSynthesisVoice(language: "en-GB") // Assign the voice to the utterance. utterance.voice = voice // Create a speech synthesizer. let synthesizer = AVSpeechSynthesizer() // Tell the synthesizer to speak the utterance. synthesizer.speak(utterance) It runs without errors but I don't hear any sound and the call to synthesizer.speak returns immediately. How can I fix this? Note I am running in REPL so synthesizer is not going out of scope and getting garbage collected.