Hello everyone, I've written an audio unit plugin that needs to be aware of any upstream latency caused by heavy plugins before it on the channel. Is there any way to query this? I know that Logic applies PDC at the channel's output (summing point), but I need to know what the accumulated latency is at the point the audio enters my plugin. Thanks!

Environment→ ・Device: iPad 10th generation ・OS：**iOS18.3.2 I'm using AVAudioSession to record sound in my application. But I recently came to realize that when the app starts a recording session on a tablet, OS automatically sets the tablet volume to 50% and when after recording ends, it doesn't change back to the previous volume level before starting the recording. So I would like to know whether this is an OS default behavior or a bug? If it's a default behavior, I much appreciate if I can get a link to the documentation.

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

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

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.

Hello everyone, I am working on an app that allows you to review your own music using Apple Music. Currently I am running into an issue with the skipping forwards and backwards outside of the app. How it should work: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song on an album should play and the information should change to reflect that in the app. If you play a song in Apple Music, you can see a Now Playing view in the lock screen. When you skip forward or backwards, it will do either action and it would reflect that when you see a little frequency icon on artwork image of a song. What it's doing: When skipping forward or backwards on the lock or home screen of an iPhone, the next or previous song is reflected outside of the app, but not in the app. When skipping a song outside of the app, it works correctly to head to the next song. But when I return to the app, it is not reflected NOTE: I am not using MusicKit variables such as Track, Album to display the songs. Since I want to grab the songs and review them I need a rating so I created my own that grabs the MusicItemID, name, artist(s), etc. NOTE: I am using ApplicationMusicPlayer.shared Is there a way to get the song to reflect in my app? (If its easier, a simple example of it would be nice. No need to create an entire xprod file)

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]; }

I prefer to use the album fetched from the library instead of the catalog since this is faster. If doing so, how can I check if all tracks of an album are added to the library. In this case I'd like to fetch the catalog version or throw an error (for example when offline). Using .with(.tracks) on the library album fetches the tracks added to the library. The trackCount property is referring to the tracks that can be fetched from the library. The isComplete property is always nil when fetching from the library. One possible way is checking the trackNumber and discCount properties. However this only detects that not all tracks of an album are added to the library if there is a song not added ahead of one that is. I'd like to be able to handle this edge case as well. Is there currently a way to do this? I'd prefer to not rely on the apple music catalog for this since this is supposed to work offline as well. Fetching and storing all trackIDs when connected and later comparing against these would work, but this would potentially mean storing tens of thousands of track ids. Thank you

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?

In my app I use AVAssetReaderTrackOutput to extract PCM audio from a user-provided video or audio file and display it as a waveform. Recently a user reported that the waveform is not in sync with his video, and after receiving the video I noticed that the waveform is in fact double as long as the video duration, i.e. it shows the audio in slow-motion, so to speak. Until now I was using CMFormatDescription.audioStreamBasicDescription.mSampleRate which for this particular user video returns 22'050. But in this case it seems that this value is wrong... because the audio file has two audio channels with different sample rates, as returned by CMFormatDescription.audioFormatList.map({ $0.mASBD.mSampleRate }) The first channel has a sample rate of 44'100, the second one 22'050. If I use the first sample rate, the waveform is perfectly in sync with the video. The problem is given by the fact that the ratio between the audio data length and the sample rate multiplied by the audio duration is 8, double the ratio for the first audio file (4). In the code below this ratio is given by Double(length) / (sampleRate * asset.duration.seconds) When commenting out the line with the sampleRate variable definition in the code below and uncommenting the following line, the ratios for both audio files are 4, which is the expected result. I would expect audioStreamBasicDescription to return the correct sample rate, i.e. the one used by AVAssetReaderTrackOutput, which (I think) somehow merges the stereo tracks. The documentation is sparse, and in particular it’s not documented whether the lower or higher sample rate is used; in this case, it seems like the higher one is used, but audioStreamBasicDescription for some reason returns the lower one. Does anybody know why this is the case or how I should extract the sample rate of the produced PCM audio data? Should I always take the higher one? I created FB19620455. let openPanel = NSOpenPanel() openPanel.allowedContentTypes = [.audiovisualContent] openPanel.runModal() let url = openPanel.urls[0] let asset = AVURLAsset(url: url) let assetTrack = asset.tracks(withMediaType: .audio)[0] let assetReader = try! AVAssetReader(asset: asset) let readerOutput = AVAssetReaderTrackOutput(track: assetTrack, outputSettings: [AVFormatIDKey: Int(kAudioFormatLinearPCM), AVLinearPCMBitDepthKey: 16, AVLinearPCMIsBigEndianKey: false, AVLinearPCMIsFloatKey: false, AVLinearPCMIsNonInterleaved: false]) readerOutput.alwaysCopiesSampleData = false assetReader.add(readerOutput) let formatDescriptions = assetTrack.formatDescriptions as! [CMFormatDescription] let sampleRate = formatDescriptions[0].audioStreamBasicDescription!.mSampleRate //let sampleRate = formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate }).max()! print(formatDescriptions[0].audioStreamBasicDescription!.mSampleRate) print(formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate })) if !assetReader.startReading() { preconditionFailure() } var length = 0 while assetReader.status == .reading { guard let sampleBuffer = readerOutput.copyNextSampleBuffer(), let blockBuffer = sampleBuffer.dataBuffer else { break } length += blockBuffer.dataLength } print(Double(length) / (sampleRate * asset.duration.seconds))

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()

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

According to the header file the outputVolume properties supported range is 0.0-1.0: /*! @property outputVolume @abstract The mixer's output volume. @discussion This accesses the mixer's output volume (0.0-1.0, inclusive). @property (nonatomic) float outputVolume; However when setting the volume to 2.0 the audio does indeed play louder. Is the header file out of date and if so, what is the supported range for outputVolume? Thanks

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

Hello, i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this? Do i need to create a custom catalog?

Hi everyone, We’re encountering an issue where AudioQueueNewOutput blocks indefinitely and never returns, and we’re hoping to get some insight or confirmation if this is a known behavior/regression on newer iOS versions. Issue Description When triggering audio playback, we create an output AudioQueue using AudioQueueNewOutput. On some devices, the call hangs inside AudioQueueNewOutput and never returns, with no OSStatus error and no subsequent logs. This behavior is reproducible mainly on iOS 18.3. Earlier iOS versions do not show this issue under the same code path. if (audioDes) { mAudioDes.mSampleRate = audioDes->mSampleRate; mAudioDes.mBitsPerChannel = audioDes->mBitsPerChannel; mAudioDes.mChannelsPerFrame = audioDes->mChannelsPerFrame; mAudioDes.mFormatID = audioDes->mFormatID; mAudioDes.mFormatFlags = audioDes->mFormatFlags; mAudioDes.mFramesPerPacket = audioDes->mFramesPerPacket; mAudioDes.mBytesPerFrame = audioDes->mBytesPerFrame; mAudioDes.mBytesPerPacket = audioDes->mBytesPerFrame; mAudioDes.mReserved = 0; } // Create AudioQueue for output OSStatus status = AudioQueueNewOutput( &mAudioDes, AQOutputCallback, this, NULL, NULL, 0, &audioQueue ); code-block The thread blocks inside AudioQueueNewOutput, and execution never reaches the next line. Additional Notes / Observations ASBD is confirmed to be valid Standard PCM output Sample rate, channels, bytes per frame/packet all consistent Same ASBD works correctly on earlier iOS versions AudioQueue is created on a background thread Not on the main thread Not inside the AudioQueue callback On first creation, AVAudioSession may not yet be active setCategory and setActive:YES may be called shortly before creating the AudioQueue There may be a timing window where the session is still activating Issue is reported mainly on iOS 18.3 Multiple user reports point to iOS 18.3 devices Same code path works on iOS 17.x and earlier No OSStatus error is returned — the call simply never returns. Questions Is it expected that AudioQueueNewOutput can block indefinitely while waiting for AVAudioSession / audio route / HAL readiness? Have there been any behavior changes in iOS 18.3 regarding AudioQueue creation or AudioSession synchronization? Is it unsafe to call AudioQueueNewOutput before AVAudioSession is fully active on recent iOS versions? Are there recommended patterns (or delays / callbacks) to ensure AudioQueue creation does not hang? Any insight or confirmation would be greatly appreciated. Thanks in advance!

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?

I'm working on a project to support spatial audio editing, using this sample project as a reference: https://developer.apple.com/documentation/Cinematic/editing-spatial-audio-with-an-audio-mix This sample works well on an unedited capture, but does not work for a capture that has already been edited. The failure is occurring at "let audioInfo = try await CNAssetSpatialAudioInfo(asset: myAsset)", which is throwing "no eligible audio tracks in asset". I also find that for already edited captures, if i use CNAssetSpatialAudioInfo.assetContainsSpatialAudio, it returns false. What i mean by "already edited" is that if I take a spatial capture with my iPhone 16, and then edit that capture in the Photos app using the Cinematic effect, and then save the edited output (e.g. edited_capture.mov), I can't import that edited_capture.mov into my project as a spatial audio asset. Is this intentional behavior or a bug? If it's intentional, can you describe why?

Issue Description I'm implementing a system audio capture feature using AudioHardwareCreateProcessTap and AudioHardwareCreateAggregateDevice. The app successfully creates the tap and aggregate device, but when starting the IO procedure with AudioDeviceStart, it sometimes fails with OSStatus error 1852797029. (The operation couldn’t be completed. (OSStatus error 1852797029.)) The error occurs inconsistently, which makes it particularly difficult to debug and reproduce. Questions Has anyone encountered this intermittent "nope" error code (0x6e6f7065) when working with system audio capture? Are there specific conditions or system states that might trigger this error sporadically? Are there any known workarounds for handling this intermittent failure case? Any insights or guidance would be greatly appreciated. I'm wondering if anyone else has encountered this specific "nope" error code (0x6e6f7065) when working with system audio capture.