Hi, I'm working an a video editing software that lets you composite and export videos. I use a custom compositor to apply my effects etc. In my crash dashboard, I am seeing a report of an EXC_BAD_ACCESS crash from objc_msgSend. Below is the stacktrace. libobjc.A.dylib objc_msgSend libdispatch.dylib _dispatch_sync_invoke_and_complete_recurse libdispatch.dylib _dispatch_sync_f_slow [symbolication failed] libdispatch.dylib _dispatch_client_callout libdispatch.dylib _dispatch_lane_barrier_sync_invoke_and_complete AVFCore -[AVCustomVideoCompositorSession(AVCustomVideoCompositorSession_FigCallbackHandling) _customCompositorShouldCancelPendingFrames] AVFCore _customCompositorShouldCancelPendingFramesCallback MediaToolbox remoteVideoCompositor_HandleVideoCompositorClientMessage CoreMedia __figXPCConnection_CallClientMessageHandlers_block_invoke libdispatch.dylib _dispatch_call_block_and_release libdispatch.dylib _dispatch_client_callout libdispatch.dylib _dispatch_lane_serial_drain libdispatch.dylib _dispatch_lane_invoke libdispatch.dylib _dispatch_root_queue_drain_deferred_wlh libdispatch.dylib _dispatch_workloop_worker_thread libsystem_pthread.dylib _pthread_wqthread libsystem_pthread.dylib start_wqthread What stood out to me is that this is only being reported from IOS 26.0+ devices. A part of the stacktrace failed to be symbolicated [symbolication failed]. I'm 90% confident that this is Apple code, not my app's code. I cannot reproduce this locally. Is this a known issue? What are the possible root-causes, and how can I verify/eliminate them? Thanks,

I'm streaming mp3 audio data using URLSession/AudioFileStream/AVAudioConverter and getting occasional silent buffers and glitches (little bleeps and whoops as opposed to clicks). The issues are present in an offline test, so this isn't an issue of underruns. Doing some buffering on the input coming from the URLSession (URLSessionDataTask) reduces the glitches/silent buffers to rather infrequent, but they do still happen occasionally. var bufferedData = Data() func parseBytes(data: Data) { bufferedData.append(data) // XXX: this buffering reduces glitching // to rather infrequent. But why? if bufferedData.count > 32768 { bufferedData.withUnsafeBytes { (bytes: UnsafeRawBufferPointer) in guard let baseAddress = bytes.baseAddress else { return } let result = AudioFileStreamParseBytes(audioStream!, UInt32(bufferedData.count), baseAddress, []) if result != noErr { print("❌ error parsing stream: \(result)") } } bufferedData = Data() } } No errors are returned by AudioFileStream or AVAudioConverter. func handlePackets(data: Data, packetDescriptions: [AudioStreamPacketDescription]) { guard let audioConverter else { return } var maxPacketSize: UInt32 = 0 for packetDescription in packetDescriptions { maxPacketSize = max(maxPacketSize, packetDescription.mDataByteSize) if packetDescription.mDataByteSize == 0 { print("EMPTY PACKET") } if Int(packetDescription.mStartOffset) + Int(packetDescription.mDataByteSize) > data.count { print("❌ Invalid packet: offset \(packetDescription.mStartOffset) + size \(packetDescription.mDataByteSize) > data.count \(data.count)") } } let bufferIn = AVAudioCompressedBuffer(format: inFormat!, packetCapacity: AVAudioPacketCount(packetDescriptions.count), maximumPacketSize: Int(maxPacketSize)) bufferIn.byteLength = UInt32(data.count) for i in 0 ..< Int(packetDescriptions.count) { bufferIn.packetDescriptions![i] = packetDescriptions[i] } bufferIn.packetCount = AVAudioPacketCount(packetDescriptions.count) _ = data.withUnsafeBytes { ptr in memcpy(bufferIn.data, ptr.baseAddress, data.count) } if verbose { print("handlePackets: \(data.count) bytes") } // Setup input provider closure var inputProvided = false let inputBlock: AVAudioConverterInputBlock = { packetCount, statusPtr in if !inputProvided { inputProvided = true statusPtr.pointee = .haveData return bufferIn } else { statusPtr.pointee = .noDataNow return nil } } // Loop until converter runs dry or is done while true { let bufferOut = AVAudioPCMBuffer(pcmFormat: outFormat, frameCapacity: 4096)! bufferOut.frameLength = 0 var error: NSError? let status = audioConverter.convert(to: bufferOut, error: &error, withInputFrom: inputBlock) switch status { case .haveData: if verbose { print("✅ convert returned haveData: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(haveData) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } case .inputRanDry: if verbose { print("🔁 convert returned inputRanDry: \(bufferOut.frameLength) frames") } if bufferOut.frameLength > 0 { if bufferOut.isSilent { print("(inputRanDry) SILENT BUFFER at frame \(totalFrames), pending: \(pendingFrames), inputPackets=\(bufferIn.packetCount), outputFrames=\(bufferOut.frameLength)") } outBuffers.append(bufferOut) totalFrames += Int(bufferOut.frameLength) } return // wait for next handlePackets case .endOfStream: if verbose { print("✅ convert returned endOfStream") } return case .error: if verbose { print("❌ convert returned error") } if let error = error { print("error converting: \(error.localizedDescription)") } return @unknown default: fatalError() } } }

I have some question about the TTS My device default language is zh-HK. (cantonese) my device is iPhone 16 Pro, IOS 18.6 I create a function speakMandarin I want the device to speak the zh-CN (putonghua) however the device only can speak zh-HK (cantonese). I already set the AVSpeechSynthesisVoice language as zh-CN func speakMandarin(text: String) { print("speakMandarin, \(text)") lastError = nil // Reset error // Stop any ongoing speech before starting new if synthesizer.isSpeaking { synthesizer.stopSpeaking(at: .immediate) } // Configure speech utterance let utterance = AVSpeechUtterance(ssmlRepresentation: text)! utterance.rate = 0.5 // Natural speaking speed utterance.pitchMultiplier = 1.0 utterance.volume = 1.0 utterance.voice = AVSpeechSynthesisVoice(language: "zh-CN") let preferredLanguages = [ "zh-CN" , "zh-TW"] var selectedVoice: AVSpeechSynthesisVoice? for lang in preferredLanguages { if let voice = AVSpeechSynthesisVoice(language: lang) { print(lang) selectedVoice = voice utterance.voice = voice break } } // If no Mandarin voice found, use system default if selectedVoice == nil { selectedVoice = AVSpeechSynthesisVoice(language: nil) lastError = "未偵測到普通話語音包，將使用系統預設語音" print (lastError) } utterance.voice = selectedVoice print(utterance) synthesizer.speak(utterance) } here is my log speakMandarin, <speak>你好！我們來聊聊喜歡的動物吧。你喜歡什麼動物呢？</speak> zh-CN [AVSpeechUtterance 0x1194efb80] String: 你好！我們來聊聊喜歡的動物吧。你喜歡什麼動物呢？ Voice: [AVSpeechSynthesisVoice 0x104ceff90] Language: zh-CN, Name: Tingting, Quality: Default [com.apple.voice.compact.zh-CN.Tingting] Rate: 0.50 Volume: 1.00 Pitch Multiplier: 1.00 Delays: Pre: 0.00(s) Post: 0.00(s)

I've got a setup using AVAudioEngine with several tone generator nodes, each with a chain of processing nodes, the chains then mixed into the main output. Generator ➡️ Effect ➡️... ➡️ .mainMixerNode ➡️ .outputNode). Generator ➡️ Effect ➡️... ⤴️ ... Generator ➡️ Effect ➡️... ⤴️ The user should be able to mute any chain individually. I've found several potential approaches to muting, but not terribly happy with any of them. Adjust the amplitudes directly in my tone generators. Issue: Consumes CPU even when completely muted. 4 generators adds ~15% cpu, even when all chains are muted. Detach/attach chains that are muted/unmuted. Issue: Causes loud clicking/popping sounds whenever muted/unmuted. Fade mixer output volume while detaching/attaching a chain (just cutting the volume immediately to 0 doesn't get rid of the clicking/popping). Issue: Causes all channels to fade during the transition, so not ideal. The rest of these ideas are variations on making volume control+detatch/attach work for individual chains, since approach #3 worked well. Add an AVAudioMixer to the end of each chain (just for volume control). Issue: Only the mixer on the final chain functions -- the others block all output. Not sure what's going on there. Use matrix mixer (for multi-input volume control). Plus detach/attach to reduce CPU if necessary. Not yet attempted, due to perceived complexity and reports of fragility in order of wiring in. A bunch of effort before I even know if it's going to work. Develop my own fader node to put on the end of each channel. Unlike the tone generator (simple AVSourceNode), developing an effect node seems complex and time consuming. Might not even fix CPU use. I'm not completely averse to the learning curve of either 5 or 6, but would rather get some guidance on best approach before diving in. They both seem likely to take more effort than I'd like for the simple behavior I'm trying to achieve.

I’ve been using Apple Music API for quite a while now and a recent change must have happened which is quite disruptive. On many occasions, artists release singles from an album as part of promoting this album. For recent examples, Harry Styles released “Aperture” (a single) to promote his upcoming album “Kiss All The Time. Disco, Occasionally“. Similarly, Bruno Mars released “I Just Might”, a single from the upcoming album “The Romantic”. Previously, those would return at the endpoint ”artists/{artist_id}/albums” with a “- Single” suffix. But it seems a recent change happens where they only appear as playable tracks inside the album. This behavior is also evident in the Apple Music app itself. Those singles no longer appear under “Singles & EPs”. Instead, they would only be visible if the single becomes popular enough to be shown on “Top Songs“. Otherwise one would have to know to tap on the (future) album to discover if there are released singles. Meanwhile Spotify’s API returns those as singles properly, just like Apple Music API used to. This change must be recent but the question is if it’s intentional, and if so, how can the API be used from here on out to “extract” those singles and represent them?

How can media resources in my app be recommended to the system media control center, just like TikTok in the picture

I have implemented fetching Apple Music preview songs using a Swift framework integrated into a Unity app. My requirement is to fetch full tracks from a user’s Apple Music library and play them inside Unity. To do this, I understand that I need to handle authentication, generate a Developer Token, and then obtain a Music User Token to access the user’s Apple Music content. Currently, I have an Individual Apple Developer account (not Organization). Based on my research, it seems that: With an Individual account, I can implement this functionality and even upload builds to TestFlight for internal testing. However, when releasing the app publicly on the App Store, full-track playback may be restricted for Individual accounts and allowed only for Organization accounts. 👉 Can you confirm if this understanding is correct? 👉 Specifically, is it possible for an Individual account to fetch and play full-length tracks from a subscribed Apple Music user’s library (at least for internal/TestFlight testing)?

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?

Hello, I am trying to access the Apple Music Feed API, but I am recieving a 401 Unauthorized error message whenever I try to access it. I have tried using my own code to generate a JWT and directly call the API (which can call the standard Apple Music API successfully). > GET /v1/feed/song/latest HTTP/2 > Host: api.media.apple.com > user-agent: insomnia/2023.5.8 > authorization: Bearer [REDACTED] > accept: */* < HTTP/2 401 < content-type: application/json; charset=utf-8 < content-length: 0 < x-apple-jingle-correlation-key: AV5IOHBNM2UUJVOFQ4HZ2TGF6Q < x-daiquiri-instance: daiquiri:10001:daiquiri-all-shared-ext-7bb7c9b9bb-r459v:7987:25RELEASE91:daiquiri-amp-kubernetes-shared-ext-ak8s-prod-pv4-amp-daiquiri-ingress-prod and also the Apple provided Python example code, which gives me authentication errors too. $ python3 ./apple_music_feed_example.py --key-id NMBH[...] --team-id 3TNZ[...] --secret-key-file-path "/Users/foxt/Documents/am-feed/NMBH[...].p8" --out-dir . running.... INFO:__main__:Sending requests to https://api.media.apple.com INFO:__main__:Getting the latest export for feed artist Exception: Authentication Failed. Did you provide the correct team id, key id, and p8 file? Does this API need to be enabled on my account separately from the main Apple Music API? The documentation reads to me as if anyone with an Apple Developer Programme membership can use this API and I did not see any information regarding any other requirements

I’m developing a macOS audio monitoring app using AVAudioEngine, and I’ve run into a critical issue on macOS 26 beta where AVFoundation fails to detect any input devices, and AVAudioEngine.start() throws the familiar error 10877. FB#: FB19024508 Strange Behavior: AVAudioEngine.inputNode shows no channels or input format on bus 0. AVAudioEngine.start() fails with -10877 (AudioUnit connection error). AVCaptureDevice.DiscoverySession returns zero audio devices. Microphone permission is granted (authorized), and the app is properly signed and sandboxed with com.apple.security.device.audio-input. However, CoreAudio HAL does detect all input/output devices: Using AudioObjectGetPropertyDataSize and AudioObjectGetPropertyData with kAudioHardwarePropertyDevices, I can enumerate 14+ devices, including AirPods, USB DACs, and BlackHole. This suggests the lower-level audio stack is functional. I have tried: Resetting CoreAudio with sudo killall coreaudiod Rebuilding and re-signing the app Clearing TCC with tccutil reset Microphone Running on Apple Silicon and testing Rosetta/native detection via sysctl.proc_translated Using a fallback mechanism that logs device info from HAL and rotates logs for submission via Feedback Assistant I have submitted logs and a reproducible test case via Feedback Assitant : FB#: FB19024508]

Recently, after the update of 26.3 Mac OS (Tahoe), the ordering of my music app has been horrible at best - music disappearing, tracks not aligning with albums (even if the albums are different years). It's created quite a problem, because the disappearing tracks issue seems to be replicating to iOS devices as well (although track numbering and album association seem to be stable). Has anyone else heard of this issue?

I'm encountering errors while using AVAudioEngine with voice processing enabled (setVoiceProcessingEnabled(true)) in scenarios where the input and output audio devices are not the same. This issue arises specifically with mismatched devices, preventing the application from functioning as expected. Works: Paired devices (e.g., MacBook Pro mic → MacBook Pro speakers) Fails: Mismatched devices (e.g., AirPods mic → MacBook Pro speakers) When using paired input and output devices: The setup works as expected. Example: MacBook Pro microphone → MacBook Pro speakers. When using mismatched devices: AVAudioEngine setup fails during aggregate device construction. Example: AirPods microphone → MacBook Pro speakers. Error logs indicate a channel count mismatch. Here are the partial logs. Due to the content limit, I cannot post the entire logs. AUVPAggregate.cpp:1000 client-side input and output formats do not match (err=-10875) AUVPAggregate.cpp:1036 err=-10875 AVAEInternal.h:109 [AVAudioEngineGraph.mm:1344:Initialize: (err = PerformCommand(*outputNode, kAUInitialize, NULL, 0)): error -10875 AggregateDevice.mm:329 Failed expectation of constructed aggregate (312): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (312): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (336): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (336): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AudioHardware-mac-imp.cpp:3484 AudioDeviceSetProperty: no device with given ID AUHAL.cpp:1782 ca_verify_noerr: [AudioDeviceSetProperty(mDeviceID, NULL, 0, isInput, kAudioDevicePropertyIOProcStreamUsage, theSize, theStreamUsage), 560227702] AggregateDevice.mm:182 error fetching default pair AggregateDevice.mm:329 Failed expectation of constructed aggregate (348): mInput.streamChannelCounts == inputStreamChannelCounts AggregateDevice.mm:331 Failed expectation of constructed aggregate (348): mInput.totalChannelCount == std::accumulate(inputStreamChannelCounts.begin(), inputStreamChannelCounts.end(), 0U) Is it possible to use voice processing with different input/output devices? If yes, are there any specific configurations required to handle mismatched devices? How can we resolve channel count mismatch errors during aggregate device construction? Are there settings or API adjustments to enforce compatibility between input/output devices? Are there any workarounds or alternative approaches to achieve voice processing functionality with mismatched devices? For instance, can we force an intermediate channel configuration or downmix input/output formats?

Hello, I want to know if there are any restrictions with MusicKit to be used in a mobile app to be able to manipulate audio with an EQ on tracks coming from Apple Music, without modifying the actual track structure/data of course, just the audio output.

Use case: When SharePlay -ing a fully immersive 3D scene (e.g. a virtual stage), I would like to shine lights on specific Personas, so they show up brighter when someone in the scene is recording the feed (think a camera person in the scene wearing Vision Pro). Note: This spotlight effect only needs to render in the camera person's headset and does NOT need to be journaled or shared. Before I dive into this, my technical question: Can environmental and/or scene lighting affect Persona brightness in a SharePlay? If not, is there a way to programmatically make Personas "brighter" when recording? My screen recordings always seem to turn out darker than what's rendered in environment, and manually adjusting the contrast tends to blow out the details in a Persona's face (especially in visionOS 26).

Hi Apple Music API / MusicKit / MediaPlayer Team, Similar to the currentPlaybackRate keeps the same pitch, it would be great to have a currentPlaybackPitch parameter as well. Alternatively, adding a preservesPitch parameter would also work. I see that iOS 26 AutoMix on Apple Music currently does pitch shifting during music transitions, so maybe this is something that could be exposed on the later betas of iOS 26? Main feature request we get is to have simple pitch changes to Apple Music we play through our app. Is this being considered?

I need to implement a solution through an API or custom driver to completely block out the built-in speakers and microphone of Mac, because I need other apps to use specified external devices as audio input and output. Is there a way to achieve this requirement? What I mean is that even in system preferences, it should not be possible to choose the built-in microphone and speakers; only my external device can be used.

Hello everyone, I'm working on implementing a screen sharing feature using RPSystemBroadcastPickerView and a Broadcast Upload Extension to share the entire app screen in an iOS application. The Broadcast Upload Extension is set up following Apple's ReplayKit guidelines. However, I’m encountering an issue during the broadcast startup sequence: ❗ Problem Description The Screen Broadcast UI appears as expected I tap “Start Broadcast” The countdown (3 → 2 → 1) completes Then it immediately reverts to the "Start Broadcast" screen, and screen sharing does not begin No error messages are displayed None of the extension lifecycle methods (broadcastStarted(withSetupInfo:), processSampleBuffer, etc.) are called There are no logs or crash reports, neither in the main app nor in the extension ✅ What Has Been Verified Info.plist of the Broadcast Upload Extension includes: NSExtensionPointIdentifier = com.apple.broadcast-services-upload NSExtensionPrincipalClass set correctly RPBroadcastProcessMode = RPBroadcastProcessModeSampleBuffer preferredExtension is set properly to the extension’s bundle identifier Extension is listed in the main app's build settings under "Frameworks, Libraries, and Embedded Content" ⚠️ Additional Concern We noticed that in Xcode (latest version), the Broadcast Upload Extension is listed under "Embedded Frameworks" with the setting "Embed Without Signing", and there is no option to change it to "Embed & Sign". We're wondering if this could be the reason the extension fails to launch correctly at runtime, despite being detected by the broadcast picker. ❓ Questions Has anyone faced similar issues where the broadcast never starts despite correct setup? Could the "Embed Without Signing" be causing the system to silently cancel or ignore the extension at runtime? Are there any provisioning profile or entitlement requirements specific to Broadcast Upload Extensions that might trigger this behavior silently? Any insights, suggestions, or workarounds would be greatly appreciated. Thank you in advance!

If a Tap or AggregateDevice with the Private property set is created, does it automatically disappear when the process ends? If not, how can I remove the Tap or AggregateDevice before the main process terminates?

Howdy. I'm trying to access media from a users song library and receive: <ICUserIdentityStoreACAccountBackend: 0x148f8af30> Failed to initialize active account, error=Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store} I'm told I need to add a Media Library Access Capability. Nothing like this shows up in Xcode under Signing & Capabilities > +Capabilities. Also I can't find anything like this in my account in dev.apple.com. How do I enable myself and a test user using another iPhone device to access my music and their music respectively? Thanks!

ApplicationMusicPlayer is not available on watchOS but all other platforms. Is there a technical reason for that like battery life? Same goes for SystemMusicPlayer and MPMusicPlayerController. I already filed feedbacks for that.