Using the PushToTalk library, call requestBeginTransmitting (channelUUID: UUID) on a Bluetooth device and then use the PTChannelManagerial Delegate proxy method channelManager:(PTChannelManager *)channelManager didActivateAudioSession:(AVAudioSession *)audioSession Start recording sound inside. Completed recording

Hi, I’m an iOS developer building an app with an use case that needs advanced playback on Apple Music subscription streams, specifically: • Real-time tempo change (BPM) during playback — i.e., time-stretch with key-lock, not just crossfade. • Beat-matched transitions between tracks. From what I can tell, this capability seems to exist only for approved partners and isn’t available through public MusicKit. Question: What’s the official request path to be evaluated for that restricted partner entitlement (application form, questionnaire, NDA, or internal team/BD contact)? If the entitlement identifier is internal, how can I get my account routed to the right Apple Music team? For reference, publicly announced partners include Algoriddim djay, Serato DJ Pro, rekordbox (AlphaTheta), and Engine DJ—all of which appear to implement mixing features that imply advanced playback (tempo/beat-matching) on Apple Music content. I’d prefer not to share product details publicly for the moment and can provide specifics privately if needed. Thanks in advance!

I'm writing a simple app for iOS and I'd like to be able to do some text to speech in it. I have a basic audio manager class with a "speak" function: import Foundation import AVFoundation class AudioManager { static let shared = AudioManager() var audioPlayer: AVAudioPlayer? var isPlaying: Bool { return audioPlayer?.isPlaying ?? false } var playbackPosition: TimeInterval = 0 func playSound(named name: String) { guard let url = Bundle.main.url(forResource: name, withExtension: "mp3") else { print("Sound file not found") return } do { if audioPlayer == nil || !isPlaying { audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.currentTime = playbackPosition audioPlayer?.prepareToPlay() audioPlayer?.play() } else { print("Sound is already playing") } } catch { print("Error playing sound: \(error.localizedDescription)") } } func stopSound() { if let player = audioPlayer { playbackPosition = player.currentTime player.stop() } } func speak(text: String) { let synthesizer = AVSpeechSynthesizer() let utterance = AVSpeechUtterance(string: text) utterance.voice = AVSpeechSynthesisVoice(language: "en-GB") synthesizer.speak(utterance) } } And my app shows text in a ScrollView: ScrollView { Text(self.description) .padding() .foregroundColor(.black) .font(.headline) .background(Color.gray.opacity(0)) }.onAppear { AudioManager.shared.speak(text: self.description) } However, the text doesn't get read out (in the simulator). I see some output in the console: Error fetching voices: Swift.DecodingError.dataCorrupted(Swift.DecodingError.Context(codingPath: [], debugDescription: "Invalid container metadata for _UnkeyedDecodingContainer, found keyedGraphEncodingNodeID", underlyingError: nil)). Using fallback voices. I'm probably doing something wrong here, but not sure what.

After upgrading to watchOS 26, users report that when playing music on Apple Watch, if a fitness reminder is received, the music automatically pauses and users need to manually tap the play button to resume music playback. This phenomenon occurs with multiple music and podcast apps. This issue did not exist before the upgrade. We would like to know if this is an Apple bug or if there are any special development configurations needed?"

New to iOS development and I've been trying to make heads or tails of the documentation. I know there is a difference between the data fields returned from songs from the user library and from the category, but whenever I search on the apple site I can't find a list of each. For example, Im trying to get the releaseDate of a song in my library, but it seems I'll have to cross-query either the catalog entry for the using song.catalogID or the song.irsc but when I try to use them I can't find a cross reference between the two. I'm totally turned around. Also trying to determine if a song in my library has been favorited or not? isFavorited (or something similar) doesn't seem to be a thing. Using this code and trying to find a way to display a solid star if the song has been favorited or an empty one if it's not. Seems like a basic request but I can't find anything on how to do it. I've searched docs, googled, tried. Does apple want us to query the user's Favorited Songs playlist or something? How do I know which playlist that is? I know isFavorited isn't a thing, just using it here so you can see what my intension is: HStack(spacing: 10) { Image(systemName: song.isFavorited ? "star.fill" : "star") .foregroundColor(song.isFavorited ? .yellow : .gray) Image(systemName: "magnifyingglass") }

The AVB AVnu MILAN Convention has a groweing Population. Many big companies (Cisco, Meyer Sound, d&b Audio, l‘acoustics, Presonus, digico etc.) implements the AVB AVnu Milan Standards. Is there a plan on the Apple side to also implement AVnu Milan on top of the AVB Protocol? The advantage for Apple Sound would be a great Integration in the professionell Audio market and a more stable intergration on top of the AVB protocol. The atdecc work, but Not that stable.

We require assistance in resolving a critical audio design conflict within our Push-to-Talk (PTT) application. Our current volume amplification strategy—which relies on applying a GAIN factor to PCM samples in conjunction with setting the AVAudioSession category to Playback—is working successfully when PTT is used independently. However, upon integrating and reporting the same PTT call through the CallKit framework, this amplification effect is lost. The CallKit integration appears to be forcing a different, non-amplifying audio session category or configuration, negatively impacting the user's perceived call volume. We need guidance on how to maintain the AVAudioSessionCategoryPlayback setting, or an equivalent high-volume configuration, while operating under the control of CallKit.

I'm using a 4 channel USB Audio interface, with 4 microphones, and want to process them through 4 independent effect chains. However the output from AVAudioInputNode is a single 4 channel bus. How can I split this into 4 mono busses? The following code splits the input into 4 copies, and routes them through the effects, but each bus contains all four channels. How can I remap the channels to remove the unwanted channels from the bus? I tried using channelMap on the mixer node but that had no effect. I'm currently using this code primarily on iOS but it should be portable between iOS and MacOS. It would be possible to do this through a Matrix Mixer Node, but that seems completely overkill, for such a basic operation. I'm already using a Matrix Mixer to combine the inputs, and it's not well supported in AVAudioEngine. AVAudioInputNode *inputNode=[engine inputNode]; [inputNode setVoiceProcessingEnabled:NO error:nil]; NSMutableArray *micDestinations=[NSMutableArray arrayWithCapacity:trackCount]; for(i=0;i<trackCount;i++) { fixMicFormat[i]=[AVAudioMixerNode new]; [engine attachNode:fixMicFormat[i]]; // And create reverb/compressor and eq the same way... [engine connect:reverb[i] to:matrixMixerNode fromBus:0 toBus:i format:nil]; [engine connect:eq[i] to:reverb[i] fromBus:0 toBus:0 format:nil]; [engine connect:compressor[i] to:eq[i] fromBus:0 toBus:0 format:nil]; [engine connect:fixMicFormat[i] to:compressor[i] fromBus:0 toBus:0 format:nil]; [micDestinations addObject:[[AVAudioConnectionPoint alloc] initWithNode:fixMicFormat[i] bus:0] ]; } AVAudioFormat *inputFormat = [inputNode outputFormatForBus: 1]; [engine connect:inputNode toConnectionPoints:micDestinations fromBus:1 format:inputFormat];

Hi, I'm still stuck getting a basic record-with-playthrouh pipeline to work. Has anyone a sample of setting up a AVAudioEngine pipeline for recording with playthrough? Plkaythrough works with AVPlayerNode as input but not with any microphone input. The docs mention the "enabled state" of the outputNode of the engine without explaining the concept, i.e. how to enable an output. When the engine renders to and from an audio device, the AVAudioSession category and the availability of hardware determines whether an app performs output. Check the output node’s output format (specifically, the hardware format) for a nonzero sample rate and channel count to see if output is in an enabled state. Well, in my setup the output is NOT enabled, and any attempt to switch (e.g. audioEngine.outputNode.auAudioUnit.setDeviceID(deviceID) )/ attach a dedicated device / ... results in exceptions / errors

Hello, Starting in iOS 17, our application started having some issue publishing to our video session. More specifically the video capture seems to be broken in some, but not all sessions. What's troubling is that we're seeing that it fails consistently every 4 sessions. It also fails silently, without reporting any problems to the app. We only notice that there are no frames being rendered or sent to the remote devices. Here's what shows-up in the console: <<<< FigCaptureSourceRemote >>>> Fig assert: "! storage->connectionDied" at bail (FigCaptureSourceRemote.m:235) - (err=0) <<<< FigCaptureSourceRemote >>>> Fig assert: "err == 0 " at bail (FigCaptureSourceRemote.m:253) - (err=-16453) Anyone seeing this? Any idea what could be the cause? Our sessions work perfectly on iOS16 and below. Thanks

Hello! We stumbled upon a problem with our karaoke app where user on iPhone 16e/iOS 18.5 has problem with mic capture, other users cannot hear him. The mic capture is working fine on 17.5, 16.8. Maybe there is something else we need when configuring AVAudioSession for iOS 18.5? Currently it's set up like this: override func viewDidLoad() { super.viewDidLoad() UIApplication.shared.isIdleTimerDisabled = true mRoomId = appDelegate.getRoomId() let audioSession = AVAudioSession.sharedInstance() try! audioSession.setCategory(.playAndRecord, mode: .voiceChat, options: [.defaultToSpeaker]) try! audioSession.setPreferredSampleRate(48000) try! audioSession.setActive(true, options: []) }

I have both apple devices, AirPods Pro 3 is up to date and Ultra 3 is on watch os 26.1 latest public beta. Each morning when I would go on my mindfulness app and start a meditation or listen to Apple Music on my watch and AirPods Pro 3, it will play for a few seconds then disconnects. My bluetooth settings on my watch says my AirPods is connected to my watch. I also have removed the tick about connecting automatically to iPhone on the AirPods setting in my iPhone. To fix this I invariably turn off my Apple Watch Ultra 3 and turn it on again. Then the connection becomes stable. I am not sure why I have to do this each morning. It is frustrating. I am not sure why this fix does not last long? Is there something wrong with my AirPods? Has anyone encountered this before?

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

{ "aps": { "content-available": 1 }, "audio_file_name": "ding.caf", "audio_url": "https://example.com/audio.mp3" } When the app is in the background or killed, it receives a remote APNs push. The data format is roughly as shown above. How can I play the MP3 audio file at the specified "audio_url"? The user does not need to interact with the device when receiving the APNs. How can I play the audio file immediately after receiving it?

I have some tried-and-tested code that records and plays back audio via AUHAL which breaks on Tahoe on Intel. The same code works fine on Sequioa and also works on Tahoe on Apple Silicon. To start with something simple, the following code to request access to the Microphone doesn't work as it should: bool RequestMicrophoneAccess () { __block AVAuthorizationStatus status = [AVCaptureDevice authorizationStatusForMediaType: AVMediaTypeAudio]; if (status == AVAuthorizationStatusAuthorized) return true; __block bool done = false; [AVCaptureDevice requestAccessForMediaType: AVMediaTypeAudio completionHandler: ^ (BOOL granted) { status = (granted) ? AVAuthorizationStatusAuthorized : AVAuthorizationStatusDenied; done = true; }]; while (!done) CFRunLoopRunInMode (kCFRunLoopDefaultMode, 2.0, true); return status == AVAuthorizationStatusAuthorized; } On Tahoe on Intel, the code runs to completion but granted is always returned as NO. Tellingly, the popup to ask the user to grant microphone access is never displayed, even though the app is not present in the Privacy pane and never appears there. On Apple Silicon, everything works fine. There are some other problems, but I'm hoping they have a common underlying cause and that the Apple guys can figure out what's wrong from the information in this post. I'd be happy to test any potential fix. Thanks.

I've been wondering if there is a way to modify or even disable tones for indicating channel states. The behaviour regarding tones seems like a black box with little documentation. During migration to Apple's PT Framework we've noticed that there are few scenarios where a tone is played which doesn't match certain certifications. For example; moving from a channel to another produces a tone which would fail a test case. I understand the reasoning fully, as it marks that the channel is ready to transmit or receive, but this doesn't mirror the behaviour of TETRA which would be wanted in this case. I'm also wondering if there would be any way to directly communicate feedback regarding PT Framework?

Hi, I try to record audio on the iPhone with the AVAudioRecorder and Xcode 26.0.1. Maybe the problem is that I can not record audio with the simulator. But there's a menu for audio. In the plist I added 'Privacy - Microphone Usage Description' and I ask for permission before recording. if await AVAudioApplication.requestRecordPermission() { print("permission granted") recordPermission = true } else { print("permission denied") } Permission is granted. let settings: [String : Any] = [ AVFormatIDKey: kAudioFormatMPEG4AAC, AVSampleRateKey: 12000, AVNumberOfChannelsKey: 1, AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] recorder = try AVAudioRecorder(url: filename, settings: settings) let prepared = recorder.prepareToRecord() print("prepared started: \(prepared)") let started = recorder.record() print("recording started: \(started)") started is always false and I tried many settings. Error messages AddInstanceForFactory: No factory registered for id <CFUUID 0x600000211480> F8BB1C28-BAE8-11D6-9C31-00039315CD46 AudioConverter.cpp:1052 Failed to create a new in process converter -> from 0 ch, 12000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to 1 ch, 12000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame, with status -50 AudioQueueObject.cpp:1892 BuildConverter: AudioConverterNew returned -50 from: 0 ch, 12000 Hz, .... (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame to: 1 ch, 12000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 1024 frames/packet, 0 bytes/frame prepared started: true AudioQueueObject.cpp:7581 ConvertInput: aq@0x10381be00: AudioConverterFillComplexBuffer returned -50, packetCount 5 recording started: false All examples I find are the same, but apparently there must be something different.

After update,WeChat voice chatting no sounds, please help