Hello, I've discovered a buffer initialization bug in AVAudioUnitSampler that happens when loading presets with multiple zones referencing different regions in the same audio file (monolith/concatenated samples approach). Almost all zones output silence (i.e. zeros) at the beginning of playback instead of starting with actual audio data. The Problem Setup: Single audio file (monolith) containing multiple concatenated samples Multiple zones in an .aupreset, each with different sample start and sample end values pointing to different regions of the same file All zones load successfully without errors Expected Behavior: All zones should play their respective audio regions immediately from the first sample. Actual Behavior: Last zone in the zone list: Works perfectly - plays audio immediately All other zones: Output [0, 0, 0, 0, ..., _audio_data] instead of [real_audio_data] The number of zeros varies from event to event for each zone. It can be a couple of samples (<30) up to several buffers. After the initial zeros, the correct audio plays normally, so there is no shift in audio playback, just missing samples at the beginning. Minimal Reproduction 1. Create Test Monolith Audio File Create a single Wav file with 3 concatenated 1-second samples (44.1kHz): Sample 1: frames 0-44099 (constant amplitude 0.3) Sample 2: frames 44100-88199 (constant amplitude 0.6) Sample 3: frames 88200-132299 (constant amplitude 0.9) 2. Create Test Preset Create an .aupreset with 3 zones all referencing the same file: Pseudo code <Zone array> <zone 1> start : 0, end: 44099, note: 60, waveform: ref_to_monolith.wav; <zone 2> start sample: 44100, note: 62, end sample: 88199, waveform: ref_to_monolith.wav; <zone 3> start sample: 88200, note: 64, end sample: 132299, waveform: ref_to_monolith.wav; </Zone array> 3. Load and Test // Load preset into AVAudioUnitSampler let sampler = AVAudioUnitSampler() try sampler.loadAudioFiles(from: presetURL) // Play each zone (MIDI notes C4=60, D4=62, E4=64) sampler.startNote(60, withVelocity: 64, onChannel: 0) // Zone 1 sampler.startNote(62, withVelocity: 64, onChannel: 0) // Zone 2 sampler.startNote(64, withVelocity: 64, onChannel: 0) // Zone 3 4. Observed Result Zone 1 (C4): [0, 0, 0, ..., 0.3, 0.3, 0.3] ❌ Zeros at beginning Zone 2 (D4): [0, 0, 0, ..., 0.6, 0.6, 0.6] ❌ Zeros at beginning Zone 3 (E4): [0.9, 0.9, 0.9, ...] ✅ Works correctly (last zone) What I've Extensively Tested What DOES Work Separate files per zone: Each zone references its own individual audio file All zones play correctly without zeros Problem: Not viable for iOS apps with 500+ sample libraries due to file handle limitations What DOESN'T Work (All Tested) 1. Different Audio Formats: CAF (Float32 PCM, Int16 PCM, both interleaved and non-interleaved) M4A (AAC compressed) WAV (uncompressed) SF2 (SoundFont2) Bug persists across all formats 2. CAF Region Chunks: Created CAF files with embedded region chunks defining zone boundaries Set zones with no sampleStart/sampleEnd in preset (nil values) AVAudioUnitSampler completely ignores CAF region metadata Bug persists 3. Unique Waveform IDs: Gave each zone a unique waveform ID (268435456, 268435457, 268435458) Each ID has its own file reference entry (all pointing to same physical file) Hypothesized this might trigger separate buffer initialization Bug persists - no improvement 4. Different Sample Rates: Tested: 44.1kHz, 48kHz, 96kHz Bug occurs at all sample rates 5. Mono vs Stereo: Bug occurs with both mono and stereo files Environment macOS: Sonoma 14.x (tested across multiple minor versions) iOS: Tested on iOS 17.x with same results Xcode: 16.x Frameworks: AVFoundation, AudioToolbox Reproducibility: 100% reproducible with setup described above Impact & Use Case This bug severely impacts professional music applications that need: Small file sizes: Monolith files allow sharing compressed audio data (AAC/M4A) iOS file handle limits: Opening 400+ individual sample files is not viable on iOS Performance: Single file loading is much faster than hundreds of individual files Standard industry practice: Monolith/concatenated samples are used by EXS24, Kontakt, and most professional samplers Current Impact: Cannot use monolith files with AVAudioUnitSampler on iOS Forced to choose between: unusable audio (zeros at start) OR hitting iOS file limits No viable workaround exists Root Cause Hypothesis The bug appears to be in AVAudioUnitSampler's internal buffer initialization when: Multiple zones share the same source audio file Each zone specifies different sampleStart/sampleEnd offsets Key observation: The last zone in the zone array always works correctly. This is NOT related to: File permissions or security-scoped resources (separate files work fine) Audio codec issues (happens with uncompressed PCM too) Preset parsing (preset loads correctly, all zones are valid) Questions Is this a known issue? I couldn't find any documentation, bug reports, or discussions about this. Is there ANY workaround that allows monolith files to work with AVAudioUnitSampler? Alternative APIs? Is there a different API or approach for iOS that properly supports monolith sample files?

Hi everyone! I’ve developed a location-based Audio AR app in Unity with FMOD &amp; Resonance Audio and AirPods Pro Head-Tracking to create a ubiquitous augmented soundscape experience. Think of it as an audio version of Pokémon Go, but with a more precise location requirement to ensure spatial audio is placed correctly. I want this experience to run in the background on iOS, but from what I’ve gathered, it seems Unity doesn’t support this well. So, I’m considering developing a Swift version instead. Since this is primarily for research purposes, privacy concerns are not a major issue in my case. However, I’ve come across some potential challenges: Real-time precise location updates – Can iOS provide fully instantaneous, high-accuracy location updates in the background? Continuous real-time data processing – Can an app continuously process spatial audio, head-tracking, and location data while running in the background? I’m not sure if newer iOS versions have improved in these areas or if there are workarounds to achieve this. Would this kind of experience be feasible to run in the background on iOS? Any insights or pointers would be greatly appreciated! I’m very new to iOS development, so apologies if this is a basic question. Thanks in advance!

Is there a way to destroy MIDIUMPMutableEndpoint again? In my app, the user has a setting to enable and disable MIDI 2.0. If MIDI 2.0 should not be supported (or if iOS version < 18), it creates a virtual destination and a virtual source. And if MIDI 2.0 should be enabled, it instead creates a MIDIUMPMutableEndpoint, which itself creates the virtual destination and source automatically. So here is my problem: I didn't find any way to destroy the MIDIUMPMutableEndpoint again. There is a method to disable it (setEnabled:NO), but that doesn't destroy or hide the virtual destination and source. So when the user turns MIDI 2.0 support off, I will have two virtual destinations and sources, and cannot get rid of the 2.0 ones. What is the correct way to get rid of the MIDIUMPMutableEndpoint once it is created?

We’ve encountered a reproducible issue where the iPhone fails to reconnect to a Wi-Fi access point under the following conditions: The device is connected to a 2.4GHz Wi-Fi network. A Bluetooth audio accessory is connected (e.g. headset). AVAudioSession is active (such as during a voice call or when using the Voice Memos app). The user moves away from the access point, causing a disconnect. Upon returning within range, the access point is no longer recognized or reconnected while AVAudioSession remains active. However, if the Bluetooth device is disconnected or the AVAudioSession is deactivated, the Wi-Fi access point is immediately recognized again. We confirmed this behavior not only in my app but also using Apple's built-in Voice Memos app, suggesting this is not specific to our implementation. It appears that the Wi-Fi system deprioritizes reconnection while AVAudioSession is engaged. Could this be by design? Or is this a known issue or limitation with Wi-Fi and AVAudioSession interaction? Test Environment: Device: iPhone 13 mini iOS: 17.5.1 Wi-Fi: 2.4GHz band Accessories: Bluetooth headset We’d appreciate clarification on whether this is expected behavior or a bug. Thank you!

I am developing a VOD playback app, but when I stream video to an external monitor connected via HDMI with Lightning on iOS 18 or later, the screen goes dark and I cannot confirm playback. The app I am developing does not detect the HDMI and display the Player separately, but simply mirrors the video. We have confirmed that the same phenomenon occurs with other services, but we were able to confirm playback with some services such as Apple TV. Please let us know if there are any other necessary settings such as video certificates required for video playback. We would also like to know if the problem occurs with iOS 18 or later.

Among the millions of users of our online product, we have identified through data metrics that the silent audio data capture rate on iPadOS 18.4.1 or 18.5 has increased abnormally. However, we are unable to reproduce the issue. Has anyone encountered a similar issue? The parameters we used are as follows: AudioSession: category:AVAudioSessionCategoryPlayAndRecord mode:AVAudioSessionModeDefault option:77 preferredSampleRate:48000.000000 preferredIOBufferDuration:0.010000 AudioUnit format.mFormatID = kAudioFormatLinearPCM; format.mSampleRate = 48000.0; format.mChannelsPerFrame = 2; format.mBitsPerChannel = 16; format.mFramesPerPacket = 1; format.mBytesPerFrame = format.mChannelsPerFrame * 16 / 8; format.mBytesPerPacket = format.mBytesPerFrame * format.mFramesPerPacket; format.mFormatFlags = kAudioFormatFlagsNativeEndian | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsSignedInteger; component.componentType = kAudioUnitType_Output; component.componentSubType = kAudioUnitSubType_RemoteIO; component.componentManufacturer = kAudioUnitManufacturer_Apple; component.componentFlags = 0; component.componentFlagsMask = 0;

Hello. My app uses AVAudioRecorder to generate recording files, which are consistently only 4kb in size. Most users generate audio files normally, with only a few users experiencing this phenomenon occasionally. After uninstalling and installing the app, it will work normally, but it will reappear after a period of time. I have compared that the problematic audio files generated each time are fixed and cannot be played. Added the audioRecorderDidFinishRecording proxy method, which shows that the recording was completed normally. The user also reported that the recording is normal, but there is a problem with the generated file. How should I handle this issue? Look forward to your reply. - (void)startRecordWithOrderID:(NSString *)orderID { AVAudioSession *audioSession = [AVAudioSession sharedInstance]; [audioSession setCategory:AVAudioSessionCategoryRecord error:nil]; [audioSession setActive:YES error:nil]; NSMutableDictionary *settings = [[NSMutableDictionary alloc] init]; [settings setObject:[NSNumber numberWithFloat: 8000.0] forKey:AVSampleRateKey]; [settings setObject:[NSNumber numberWithInt: kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [settings setObject:[NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [settings setObject:[NSNumber numberWithInt: 1] forKey:AVNumberOfChannelsKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [settings setObject:[NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; NSString *path = [WDUtility createDirInDocument:@"audios" withOrderID:orderID withPathExtension:@"wav"]; NSURL *tmpFile = [NSURL fileURLWithPath:path]; recorder = [[AVAudioRecorder alloc] initWithURL:tmpFile settings:settings error:nil]; [recorder setDelegate:self]; [recorder prepareToRecord]; [recorder record]; }

According to the documentation (https://developer.apple.com/documentation/avfoundation/avplayeritem/externalmetadata), AVPlayerItem should have an externalMetadata property. However it does not appear to be visible to my app. When I try, I get: Value of type 'AVPlayerItem' has no member 'externalMetadata' Documentation states iOS 12.2+; I am building with a minimum deployment target of iOS 18. Code snippet: import Foundation import AVFoundation /// ... in function ... // create metadata as described in https://developer.apple.com/videos/play/wwdc2022/110338 var title = AVMutableMetadataItem() title.identifier = .commonIdentifierAlbumName title.value = "My Title" as NSString? title.extendedLanguageTag = "und" var playerItem = await AVPlayerItem(asset: composition) playerItem.externalMetadata = [ title ]

I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()

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.

Please consider adding the ability to programatically download Premium and Enhanced voices. At the moment it is extremely inconvenient for our users, as they have to navigate to settings themselves to download voices. Our app relies heavily on SpeechSynthesis integration, and it would greatly benefit from this feature. FB16307193

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.

I'm seeing crashes in _MPRemoteCommandEventDispatch on iOS 26.x devices in 3 apps. According to Bugsnag logs they are: NSInternalInconsistencyException: event dispatch <_MPRemoteCommandEventDispatch: <MPRemoteCommandEvent: 0x11c049500 commandID=THV0 command=<MPRemoteCommand: 0x109ad1ea0 type=Play (0) enabled=YES handlers=[0x109b6a310]> sourceID=(null) ([HostedRoutingSessionDataSource] handleControlSendingCommand<2W5E>)> state:201> deallocated without calling continuation I attached a log from Xcode organizer matching Bugsnag crash. mpr_remote_command_event.crash When I set the brakpoint on the -[_MPRemoteCommandEventDispatch dealloc] I can see it it's hit every time I tap play or pause on locked screen play button. Thread 0 Crashed: 0 libsystem_kernel.dylib 0x00000002370420cc __pthread_kill + 8 (:-1) 1 libsystem_pthread.dylib 0x00000001e975c810 pthread_kill + 268 (pthread.c:1721) 2 libsystem_c.dylib 0x0000000198f8ff64 abort + 124 (abort.c:122) 3 libc++abi.dylib 0x000000018a7cf808 __abort_message + 132 (abort_message.cpp:66) 4 libc++abi.dylib 0x000000018a7be484 demangling_terminate_handler() + 304 (cxa_default_handlers.cpp:76) 5 libobjc.A.dylib 0x000000018a6cff78 _objc_terminate() + 156 (objc-exception.mm:496) 6 xxxxxxxxxxxxxx 0x00000001003a7db8 CPPExceptionTerminate() + 416 (BSG_KSCrashSentry_CPPException.mm:156) 7 libc++abi.dylib 0x000000018a7cebdc std::__terminate(void (*)()) + 16 (cxa_handlers.cpp:59) 8 libc++abi.dylib 0x000000018a7ceb80 std::terminate() + 108 (cxa_handlers.cpp:88) 9 CoreFoundation 0x000000018d7341c4 __CFRunLoopPerCalloutARPEnd + 256 (CFRunLoop.c:769) 10 CoreFoundation 0x000000018d70bb5c __CFRunLoopRun + 1976 (CFRunLoop.c:3179) 11 CoreFoundation 0x000000018d70aa6c _CFRunLoopRunSpecificWithOptions + 532 (CFRunLoop.c:3462) 12 GraphicsServices 0x000000022e31c498 GSEventRunModal + 120 (GSEvent.c:2049) 13 UIKitCore 0x00000001930ceba4 -[UIApplication _run] + 792 (UIApplication.m:3902) 14 UIKitCore 0x0000000193077a78 UIApplicationMain + 336 (UIApplication.m:5577) 15 xxxxxxxxxxxxxx 0x00000001000c0134 main + 308 (main.swift:15) 16 dyld 0x000000018a722e28 start + 7116 (dyldMain.cpp:1477) Is the crash happening when the app is being terminated? Thank you!

I'm trying to implement Ambisonic B-Format audio playback on Vision Pro with head tracking. So far audio plays, head tracking works, and the sound appears to be stereo. The problem is that it is not a proper binaural playback when compared to playing back the audiofile with a DAW. Has anyone successfully implemented B-Format playback on Vision Pro? Any suggestions on my current implementation: func playAmbiAudioForum() async { do { try AVAudioSession.sharedInstance().setCategory(.playback) try AVAudioSession.sharedInstance().setActive(true) // AudioFile laoding/preperation guard let testFileURL = Bundle.main.url(forResource: "audiofile", withExtension: "wav") else { print("Test file not found") return } let audioFile = try AVAudioFile(forReading: testFileURL) let audioFileFormat = audioFile.fileFormat // create AVAudioFormat with Ambisonics B Format guard let layout = AVAudioChannelLayout(layoutTag: kAudioChannelLayoutTag_Ambisonic_B_Format) else { print("layout failed") return } let format = AVAudioFormat( commonFormat: audioFile.processingFormat.commonFormat, sampleRate: audioFile.fileFormat.sampleRate, interleaved: false, channelLayout: layout ) // write audiofile to buffer guard let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: UInt32(audioFile.length)) else { print("buffer failed") return } try audioFile.read(into: buffer) playerNode.renderingAlgorithm = .HRTF // connecting nodes audioEngine.attach(playerNode) audioEngine.connect(playerNode, to: audioEngine.outputNode, format: format) audioEngine.prepare() playerNode.scheduleBuffer(buffer, at: nil) { print("File finished playing") } try audioEngine.start() playerNode.play() } catch { print("Setup error:", error) } }

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?

Hi, I'm trying to setup a AVAudioEngine for USB Audio recording and monitoring playthrough. As soon as I try to setup playthough I get an error in the console: AVAEInternal.h:83 required condition is false: [AVAudioEngineGraph.mm:1361:Initialize: (IsFormatSampleRateAndChannelCountValid(outputHWFormat))] Any ideas how to fix it? // Input-Device setzen try? setupInputDevice(deviceID: inputDevice) let input = audioEngine.inputNode // Stereo-Format erzwingen let inputHWFormat = input.inputFormat(forBus: 0) let stereoFormat = AVAudioFormat(commonFormat: inputHWFormat.commonFormat, sampleRate: inputHWFormat.sampleRate, channels: 2, interleaved: inputHWFormat.isInterleaved) guard let format = stereoFormat else { throw AudioError.deviceSetupFailed(-1) } print("Input format: \(inputHWFormat)") print("Forced stereo format: \(format)") audioEngine.attach(monitorMixer) audioEngine.connect(input, to: monitorMixer, format: format) // MonitorMixer -> MainMixer (Output) // Problem here, format: format also breaks. audioEngine.connect(monitorMixer, to: audioEngine.mainMixerNode, format: nil)

Hello. My team and I think we have an issue where our app is asked to gracefully shutdown with a following SIGTERM. As we’ve learned, this is normally not an issue. However, it seems to also be happening while our app (an audio streamer) is actively playing in the background. From our perspective, starting playback is indicating strong user intent. We understand that there can be extreme circumstances where the background audio needs to be killed, but should it be considered part of normal operation? We hope that’s not the case. All we see in the logs is the graceful shutdown request. We can say with high certainty that it’s happening though, as we know that playback is running within 0.5 seconds of the crash, without any other tracked user interaction. Can you verify if this is intended behavior, and if there’s something we can do about it from our end. From our logs it doesn’t look to be related to either memory usage within the app, or the system as a whole. Best, John

Hi. I work on an audio app for iOS which is successfully using the MPRemoteCommandCenter for commands like next, back, skip forward, skip backward etc. I am trying to implement playback rate controls in my app (so that users can change the playback speed of audio to 0.5x or 2x for example). While the above commands work, the changePlaybackRateCommand does not seem to. I have enabled the command, given it a target/handler and set supported rates. With the other commands, this caused the UI to change on lock screen, in command center etc, by adding the control for the command (a next button for the next command for example). However, it does not seem to do anything for the playback rate command. I can implement my own "rate button" UI and rate change handling, but I'm wondering if this is a known bug within Apple? Looking online, it seems other people face the same issue and haven't been able to get this command to work. Why is this API provided if it doesn't seem to do anything? Is there something I'm missing? Kind regards.

On Apple TV 4K 3rd generation, with tvOS 26 beta 2, when two HomePod 2 are paired to the device, music and movie sources with Dolby Atmos can only be listened to in stereo. dolby atmos not supported

Session player regions populate blank, with no sound media when tracks or regions are created.