My workout watch app supports audio playback during exercise sessions. When users carry both Apple Watch, iPhone, and AirPods, with AirPods connected to the iPhone, I want to route audio from Apple Watch to AirPods for playback. I've implemented this functionality using the following code. try? session.setCategory(.playback, mode: .default, policy: .longFormAudio, options: []) try await session.activate() When users are playing music on iPhone and trigger my code in the watch app, Apple Watch correctly guides users to select AirPods, pauses the iPhone's music, and plays my audio. However, when playback finishes and I end the session using the code below: try session.setActive(false, options:[.notifyOthersOnDeactivation]) the iPhone doesn't automatically resume the previously interrupted music playback—it requires manual intervention. Is this expected behavior, or am I missing other important steps in my code?

【溦N51888M】腾龙公司会员申请流程步骤【罔纸 211239.com 】输入官惘到浏览器打开联系24小时在线业务人员办理上下，打开公司官网. 二、点击主页右上角注册按钮. 三、填写账号信息. 四、输入手机号，验证码，密码. 五、勾选用户协议，完成注册协议，完成注册. 注意：若出现账号已存在」提示，需重新设置唯一账号名称

Is there any feasible way to get a Core Audio device's system effect status (Voice Isolation, Wide Spectrum)? AVCaptureDevice provides convenience properties for system effects for video devices. I need to get this status for Core Audio input devices.

hi all, as soon an audio is played in a whatever app, coreaudiod inserts a sleep prevent assertion for both, the system AND the display. can i somehow stop the insertion of the display sleep assertion? pid 223(coreaudiod): [0x00004e9e00058dc2] 00:03:18 PreventUserIdleDisplaySleep named: "com.apple.audio.AppleGFXHDAEngineOutputDP:10001:0:{B31A-08C6-00000000}.context.preventuseridledisplaysleep" Created for PID: 4145. where PID 4145 is spotify. but it doesn't matter which app is playing the audio. any help would be appreciated thanks

Is it possible to find IDR frame (CMSampleBuffer) in AVAsset h264 video file?

My app want Converting iphone12 HDR Video to SDR，to edit。 follow the doc Apple-HDR-Convert. My code setting the pixBuffAttributes        [pixBuffAttributes setObject:(id)(kCVImageBufferYCbCrMatrix_ITU_R_709_2) forKey:(id)kCVImageBufferYCbCrMatrixKey];       [pixBuffAttributes setObject:(id)(kCVImageBufferColorPrimaries_ITU_R_709_2) forKey:(id)kCVImageBufferColorPrimariesKey];       [pixBuffAttributes setObject:(id)kCVImageBufferTransferFunction_ITU_R_709_2 forKey:(id)kCVImageBufferTransferFunctionKey];       playerItemOutput = [[AVPlayerItemVideoOutput alloc] initWithPixelBufferAttributes:pixBuffAttributes]; but I get the playerItemOutput's output buffer   CFTypeRef colorAttachments = CVBufferGetAttachment(pixelBuffer, kCVImageBufferYCbCrMatrixKey, NULL);     CFTypeRef colorPrimaries = CVBufferGetAttachment(pixelBuffer, kCVImageBufferColorPrimariesKey, NULL);     CFTypeRef colorTransFunc = CVBufferGetAttachment(pixelBuffer, kCVImageBufferTransferFunctionKey, NULL);      NSLog(@"colorAttachments = %@", colorAttachments);     NSLog(@"colorPrimaries = %@", colorPrimaries);     NSLog(@"colorTransFunc = %@", colorTransFunc); log output: colorAttachments = ITU_R_2020 colorPrimaries = ITU_R_2020 colorTransFunc = ITU_R_2100_HLG pixBuffAttributes setting output format invalid，please help！

So experimenting with the new SpeechTranscriber, if I do: let transcriber = SpeechTranscriber( locale: locale, transcriptionOptions: [], reportingOptions: [.volatileResults], attributeOptions: [.audioTimeRange] ) only the final result has audio time ranges, not the volatile results. Is this a performance consideration? If there is no performance problem, it would be nice to have the option to also get speech time ranges for volatile responses. I'm not presenting the volatile text at all in the UI, I was just trying to keep statistics about the non-speech and the speech noise level, this way I can determine when the noise level falls under the noisefloor for a while. The goal here was to finalize the recording automatically, when the noise level indicate that the user has finished speaking.

I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?

I have a Catalyst app ('container') which hosts an embedded AUv3 Audio Unit extension ('plugin'). This used to work for years and has worked with this project until a few days ago. it still works on iOS as expected on MacOS the extension is never registered/installed and won't load extension won't show up with AUVal seems to have stopped working with the 26.1 XCode update I'm fairly certain the problem is not code related (i.e. likely build settings, project settings, entitlements, signing, etc.) I have compared all settings with another still-working project and can't find any meaningful difference (I can't request code-level support because even the minimal thing vastly exceeds the 250 lines of code limit.) How can I debug the issue? I literally don't know where to start to fix this problem, short of rebuilding the entire thing and hope that it magically starts working again.

I've tried SpeechTranscriber with a lot of my devices (from iPhone 12 series ~ iPhone 17 series) without issues. However, SpeechTranscriber.isAvailable value is false for my iPhone 11 Pro. https://developer.apple.com/documentation/speech/speechtranscriber/isavailable I'am curious why the iPhone 11 Pro device is not supported. Are all iPhone 11 series not supported intentionally? Or is there any problem with my specific device? I've also checked the supportedLocales, and the value is an empty array. https://developer.apple.com/documentation/speech/speechtranscriber/supportedlocales

I have the new iOS 26 SpeechTranscriber working in my application. The issue I am facing is how to determine if the device I am running on supports SpeechTranscriber. I was able to create code that tests if the device supports transcription but it takes a bit of time to run and thus the results are not available when the app launches. What I am looking for is a list of what iOS 26 devices it doesn't run on. I think its safe to assume any new devices will support it so if we can just have a list of what devices that can run iOS 26 and not able to do transcription it would be much faster for the app. I have determined it doesn't work on a SE 2nd Gen, it works on iPhone 12, SE 3rd Gen, iPhone 14 Pro, 15 Pro. As the SpeechTranscriber doesn't work in the simulator I can't determine that way. I have checked the docs and it doesn't list the devices it doesn't work on.

Hi, everyone, I downloaded the source code EditingSpatialAudioWithAnAudioMix.zip from https://developer.apple.com/documentation/Cinematic/editing-spatial-audio-with-an-audio-mix, when I carried out one of the actions named "process" in command line the program crashed!! Form the source code, I found that the value of componentType is set to kAudioUnitType_FormatConverter: // The actual `AudioUnit`. public var auAudioMix = AVAudioUnitEffect() init() { // Generate a component description for the audio unit. let componentDescription = AudioComponentDescription( componentType: kAudioUnitType_FormatConverter, componentSubType: kAudioUnitSubType_AUAudioMix, componentManufacturer: kAudioUnitManufacturer_Apple, componentFlags: 0, componentFlagsMask: 0) auAudioMix=AVAudioUnitEffect(audioComponentDescription: componentDescription) } But in the document from https://developer.apple.com/documentation/avfaudio/avaudiouniteffect/init(audiocomponentdescription:), it seems that componentType can not be set to kAudioUnitType_FormatConverter and : Has everyone encountered this problem?

I’m facing a problem while trying to achieve spatial audio effects in my iOS 18 app. I have tried several approaches to get good 3D audio, but the effect never felt good enough or it didn’t work at all. Also what mostly troubles me is I noticed that AirPods I have doesn’t recognize my app as one having spatial audio (in audio settings it shows "Spatial Audio Not Playing"). So i guess my app doesn't use spatial audio potential. First approach uses AVAudioEnviromentNode with AVAudioEngine. Chaining position of player as well as changing listener’s doesn’t seem to change anything in how audio plays. Here's simple how i initialize AVAudioEngine import Foundation import AVFoundation class AudioManager: ObservableObject { // important class variables var audioEngine: AVAudioEngine! var environmentNode: AVAudioEnvironmentNode! var playerNode: AVAudioPlayerNode! var audioFile: AVAudioFile? ... //Sound set up func setupAudio() { do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } audioEngine = AVAudioEngine() environmentNode = AVAudioEnvironmentNode() playerNode = AVAudioPlayerNode() audioEngine.attach(environmentNode) audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: environmentNode, format: nil) audioEngine.connect(environmentNode, to: audioEngine.mainMixerNode, format: nil) environmentNode.listenerPosition = AVAudio3DPoint(x: 0, y: 0, z: 0) environmentNode.listenerAngularOrientation = AVAudio3DAngularOrientation(yaw: 0, pitch: 0, roll: 0) environmentNode.distanceAttenuationParameters.referenceDistance = 1.0 environmentNode.distanceAttenuationParameters.maximumDistance = 100.0 environmentNode.distanceAttenuationParameters.rolloffFactor = 2.0 // example.mp3 is mono sound guard let audioURL = Bundle.main.url(forResource: "example", withExtension: "mp3") else { print("Audio file not found") return } do { audioFile = try AVAudioFile(forReading: audioURL) } catch { print("Failed to load audio file: \(error)") } } ... //Playing sound func playSpatialAudio(pan: Float ) { guard let audioFile = audioFile else { return } // left side playerNode.position = AVAudio3DPoint(x: pan, y: 0, z: 0) playerNode.scheduleFile(audioFile, at: nil, completionHandler: nil) do { try audioEngine.start() playerNode.play() } catch { print("Failed to start audio engine: \(error)") } ... } Second more complex approach using PHASE did better. I’ve made an exemplary app that allows players to move audio player in 3D space. I have added reverb, and sliders changing audio position up to 10 meters each direction from listener but audio seems to only really change left to right (x axis) - again I think it might be trouble with the app not being recognized as spatial. //Crucial class Variables: class PHASEAudioController: ObservableObject{ private var soundSourcePosition: simd_float4x4 = matrix_identity_float4x4 private var audioAsset: PHASESoundAsset! private let phaseEngine: PHASEEngine private let params = PHASEMixerParameters() private var soundSource: PHASESource private var phaseListener: PHASEListener! private var soundEventAsset: PHASESoundEventNodeAsset? // Initialization of PHASE init{ do { let session = AVAudioSession.sharedInstance() try session.setCategory(.playback, mode: .default, options: []) try session.setActive(true) } catch { print("Failed to configure AVAudioSession: \(error.localizedDescription)") } // Init PHASE Engine phaseEngine = PHASEEngine(updateMode: .automatic) phaseEngine.defaultReverbPreset = .mediumHall phaseEngine.outputSpatializationMode = .automatic //nothing helps // Set listener position to (0,0,0) in World space let origin: simd_float4x4 = matrix_identity_float4x4 phaseListener = PHASEListener(engine: phaseEngine) phaseListener.transform = origin phaseListener.automaticHeadTrackingFlags = .orientation try! self.phaseEngine.rootObject.addChild(self.phaseListener) do{ try self.phaseEngine.start(); } catch { print("Could not start PHASE engine") } audioAsset = loadAudioAsset() // Create sound Source // Sphere soundSourcePosition.translate(z:3.0) let sphere = MDLMesh.newEllipsoid(withRadii: vector_float3(0.1,0.1,0.1), radialSegments: 14, verticalSegments: 14, geometryType: MDLGeometryType.triangles, inwardNormals: false, hemisphere: false, allocator: nil) let shape = PHASEShape(engine: phaseEngine, mesh: sphere) soundSource = PHASESource(engine: phaseEngine, shapes: [shape]) soundSource.transform = soundSourcePosition print(soundSourcePosition) do { try phaseEngine.rootObject.addChild(soundSource) } catch { print ("Failed to add a child object to the scene.") } let simpleModel = PHASEGeometricSpreadingDistanceModelParameters() simpleModel.rolloffFactor = rolloffFactor soundPipeline.distanceModelParameters = simpleModel let samplerNode = PHASESamplerNodeDefinition( soundAssetIdentifier: audioAsset.identifier, mixerDefinition: soundPipeline, identifier: audioAsset.identifier + "_SamplerNode") samplerNode.playbackMode = .looping do {soundEventAsset = try phaseEngine.assetRegistry.registerSoundEventAsset( rootNode: samplerNode, identifier: audioAsset.identifier + "_SoundEventAsset") } catch { print("Failed to register a sound event asset.") soundEventAsset = nil } } //Playing sound func playSound(){ // Fire new sound event with currently set properties guard let soundEventAsset else { return } params.addSpatialMixerParameters( identifier: soundPipeline.identifier, source: soundSource, listener: phaseListener) let soundEvent = try! PHASESoundEvent(engine: phaseEngine, assetIdentifier: soundEventAsset.identifier, mixerParameters: params) soundEvent.start(completion: nil) } ... } Also worth mentioning might be that I only own personal team account

Not able to record audio in AAC format with 96 kHz sample rate using AVAudioRecorder or Extended Audio File services with 96 kHz input audio from input device. The audio recording settings used are let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatMPEG4AAC), AVSampleRateKey: sampleRate AVNumberOfChannelsKey: 1 AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] When tried using AVAudioEngine using AVAudioFile, AVAudioFile(forWriting: fileURL, // file extension .m4a settings: fileSettings, commonFormat: AVAudioCommonFormat.pcmFormatFloat32, interleaved: interleaved) else { return } got error CodecConverterFactory.cpp:977 unable to select compatible encoder sample rate AudioConverter.cpp:1017 Failed to create a new in process converter -> from 1 ch, 96000 Hz, Float32 to 1 ch, 96000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame, with status 1718449215

I am using the sample app from: https://developer.apple.com/videos/play/wwdc2025/277/?time=763 I installed this on an Iphone 15 Pro with iOS 26 beta 1. I was able to get good transcription with it. The app did crash sometimes when transcribing and I was going to post here with the details. I then installed iOS beta 2 and uninstalled the sample app. Now every time I try to run the sample app on the 15 Pro I get this message: SpeechAnalyzer: Input loop ending with error: Error Domain=SFSpeechErrorDomain Code=10 "Cannot use modules with unallocated locales [en_US (fixed en_US)]" UserInfo={NSLocalizedDescription=Cannot use modules with unallocated locales [en_US (fixed en_US)]} I can't continue our our work towards using SpeechAnalyzer now with this error. I have set breakpoints on all the catch handlers and it doesn't catch this error. My phone region is "United States"

I am trying to use SpeechTranscriber from Speech framework. Is it possible to use it on Simulator of iOS 26 (Mac OS Tahoe)? Function "supportedLocales" returns an empty array.

Hi there, I recently launched a dj app to the mac app store, and was wondering how I could access songs for mixing purposes via Apple Music just like how serato, rekordbox, djay, and other DJ apps do? Thanks, Gunek

I’m running the iOS 26.2 Public Beta update and my album artwork is missing from the music app (I’m not using Apple Music). I use google to get my album artwork. Do I need to wait for a new update?

I have an AUv3 plugin which uses an FFT - which requires n samples before it can produce any output - so, depending on the relation between the host's buffer size and the FFT window size, it may receive a several buffers of samples, producing no output, and then dumping out what it has once a sufficient number of samples have been received. This means that output is produced in fits and starts, in batches that match the FFT size (modulo oversampling) - e.g. if being fed buffers of 256 samples with an fft size of 1024, the output buffer sizes will be 0 for the first 3 buffers, and upon the fourth, the first 256 processed samples are returned and the remaining 768 cached; the next three buffers will return the remaining cached samples while processing and buffering subsequent ones, and so forth. The internal mechanics of that I have solved, caching output if the current output buffer is too small, and so forth - so it all works as advertised, and the plugin reports its latency correctly. And when run as an app in demo-mode, playback works as expected. In the plugin's render block, it captures the number of frames written, and if it is less than the number of frames passed in, adjusts the mDataByteSize of the output buffers to match the actual quantity of data being returned: unsigned int framesWritten = (unsigned int) processHelper->processWithEvents(inAudioBufferList, outAudioBufferList, timestamp, frameCount, realtimeEventListHead); if (framesWritten < frameCount) { for (UInt32 i = 0; i < outAudioBufferList->mNumberBuffers; ++i) { outAudioBufferList->mBuffers[i].mDataByteSize = framesWritten * 4; // assume 4 byte floats } } However, there are a couple of serious issues: auval -v fails it with - Render Test at 64 frames, sample rate: 22050 Hz ERROR: Output Buffer Size does not match requested When connected to Logic Pro, it appears that mDataByteSize is ignored, and the entire allocated buffer is read - audio has sections of silence snipped into it which corresponds the number of empty buffers being returned If I set Logic's buffer size to 1024 and use a 1024 sample FFT window, the plugin works correctly - but of course a plugin cannot dictate buffer size, and `1024 is too small a window size to be useful for anything but filtering very high frequencies This seems like it has to be a solvable problem, and most likely the issue is in how my code reports the number of usable samples in the returned buffer. So, what is the correct way for a plugin to report that it has no samples to return, but will, uh, real soon now? I know I could convert this plugin to be one that does offline rendering of the entire input, but this is real-time processing, just with a fixed amount of latency, so that should not be necessary.

I'm encountering numerous crashes involving the com.apple.coreaudio.AQClient thread on our application. The crash details are as follows: #10 com.apple.coreaudio.AQClient SIGSEGV SEGV_ACCERR 0 libobjc.A.dylib _objc_msgSend + 44 1 AudioToolbox ClientMessageHandler::PropertyChanged(unsigned int) + 872 2 AudioToolbox ClientAudioQueue::FetchAndDeliverPendingCallbacks(unsigned int) + 924 3 AudioToolbox __XCallbackNotificationsAvailable + 212 4 libAudioToolboxUtility.dylib _mshMIGPerform + 260 5 CoreFoundation ___CFRUNLOOP_IS_CALLING_OUT_TO_A_SOURCE1_PERFORM_FUNCTION__ + 56 6 CoreFoundation ___CFRunLoopDoSource1 + 596 7 CoreFoundation ___CFRunLoopRun + 2392 8 CoreFoundation _CFRunLoopRunSpecific + 572 9 AudioToolbox CADeprecated::GenericRunLoopThread::Entry(void*) + 156 10 libAudioToolboxUtility.dylib CADeprecated::CAPThread::Entry(CADeprecated::CAPThread*) + 88 11 libsystem_pthread.dylib __pthread_start + 116 All these crashes occur on system versions below iOS/iPadOS 17, primarily when the device's available RAM is low. What steps can I take to resolve this issue? Any insights would be greatly appreciated!