Dive into the technical aspects of audio on your device, including codecs, format support, and customization options.

Audio Documentation

Posts under Audio subtopic

Post

Replies

Boosts

Views

Activity

Unable to play audio via MusicKit
Hey folks, I'm running into an odd issue suddenly with an app that had a working MusicKit integration before. I'm using ApplicationMusicPlayer to play Apple Music albums and songs. I'm testing on a physical device, signed in to Apple ID, and with a valid subscription. Apple Music via the first-party app works entirely fine on this device. Attempting to play back any content at all gives the log: <ICUserIdentityStoreACAccountBackend: 0x1070bf3e0> Failed to initialize primary apple 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} [ICUserIdentityStore] - initializing account histories with activeAccountDSID = nil, activeLockerAccountDSID = nil, timestamp = 14605951908 [ICUserIdentityStore] Failed to fetch local store account with error: Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store}. The album artwork, track names, etc, all appear in the control center playback controls, but the music doesn't play. Trying to trigger playback with control center just results in it skipping to the next track, which doesn't play either. This exact code used to work. I have the MusicKit service selected in Apple Connect. Since this isn't entitlement-based, I'm not sure how else to check that I'm set up correctly. I've tried deleting/reinstalling the app, restarting the device, cleaning/rebuilding, and deleting DerivedData, to no avail. Any help? Running Xcode 16.4 (16F6), testing on iOS 18.5 (22F76)
0
1
174
Jun ’25
AVAudioFile.read extremely slow after seeking in FLAC and MP3 files
I'm developing an audio player app that uses AVAudio​File to read PCM data from various formats. I'm experiencing severe performance issues when seeking in FLAC, while other compressed formats (M4A/AAC) work correctly. I don't intend to use them in my app, but I also tested mp3 files just by curiosity and they also have this issue. Environment: macOS 26 (Tahoe) Xcode 26.3 Apple Silicon (M1) The issue: After setting AVAudio​File​.frame​Position to a position mid-file, the subsequent call to AVAudio​File​.read(into​:frame​Count:) blocks for an unreasonable amount of time for FLAC and MP3 files. The delay scales linearly with the seek target, seeking near the beginning is fast, seeking toward the end is proportionally slower, which suggests the decoder is decoding linearly from the beginning of the file rather than using any seek index. (My app deals with “images” of Audio CDs ripped as a single long audio file.) The issue is particularly severe when reading files from an SMB network share (server on Ethernet, client on Wi-Fi with the access point ~2 meters away in line of sight). Quick Benchmark results: I tested with the same 75-minute audio content (16-bit/44.1 kHz stereo, 200,502,708 frames) encoded in five formats, seeking to the midpoint. Over SMB (Local Network, Server on Ethernet, Client on WiFi): Format | Seek + Read Time ----------|------------------ WAV | 0.007 s AIFF | 0.009 s Apple | 0.015 s Lossless | MP3 | 9.2 s FLAC | 30.2 s Locally (MacBook Air M1 SSD) : Format | Seek + Read Time ----------|------------------ WAV | 0.0005 s AIFF | 0.0004 s Apple | 0.0011 s Lossless | MP3 | 0.1958 s FLAC | 0.7528 s WAV, AIFF, and M4A all seek virtually instantly (< 15 ms). MP3 and FLAC exhibit linear-time behavior, with FLAC being the worst affected. Note that M4A (AAC) is also a compressed format that requires decoding after seeking, yet it completes in 15 ms. This rules out any inherent limitation of compressed formats, the MP4 container's packet index (stts/stco) is clearly being used for fast random access. Both MP3 (Xing/LAME TOC) and FLAC (SEEKTABLE metadata block) have their own seek mechanisms that should provide similar performance. Minimal CLI tool to reproduce: import Foundation guard CommandLine.arguments.count > 1 else { print("Usage: FLACSpeed <audio-file-path>") exit(1) } let path = CommandLine.arguments[1] let fileURL = URL(fileURLWithPath: path) do { let file = try AVAudioFile(forReading: fileURL) let format = file.processingFormat let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: 8192)! let totalFrames = file.length let seekTarget = totalFrames / 2 print("File: \(fileURL.lastPathComponent)") print("Format: \(format)") print("Total frames: \(totalFrames)") print("Seeking to frame: \(seekTarget)") file.framePosition = seekTarget let start = CFAbsoluteTimeGetCurrent() try file.read(into: buffer, frameCount: 8192) let elapsed = CFAbsoluteTimeGetCurrent() - start print("Read after seek took \(elapsed) seconds") } catch { print("Error: \(error.localizedDescription)") exit(1) } Expected behavior: AVAudio​File​.read(into​:frame​Count:) after setting frame​Position should use the available seek mechanisms in FLAC and MP3 files for fast random access, as it already does for M4A (AAC). Even accounting for the fact that seek tables provide approximate (not sample-precise) positioning, the "jump to nearest index point + decode forward" approach should complete in milliseconds, not seconds. Workaround: For FLAC, I've worked around this by using libFLAC directly, which provides instant seeking via FLAC__stream​_decoder​_seek​_absolute(). libFLAC Performance: For comparison, libFLAC's FLAC__stream​_decoder​_seek​_absolute() performs the same seek + read on the same FLAC file in around 0.015, using the FLAC seek table to jump to the nearest preceding seek point, then decoding forward a small number of frames to the exact target sample.
0
1
95
1d
MusicKit - ApplicationMusicPlayer fails to play certain Songs
[Note: this issue was happening on a main testing device, and after testing the same code on other devices, this issue is only happening on 1 out of 4 devices] We are successfully getting a MusicCatalogResourceResponse for every song ID where we make the MusicCatalogResourceRequest. We are able to display the song title, artist name, and album artwork for each Song in the response. However - when we go to play the song, there are some songs that play, and several songs that do not play. For the songs that don't play, the console shows “Failed to prepareToPlay error=<MPMusicPlayerControllerErrorDomain.6 "Failed to prepare to play" {}>” let musicPlayer = ApplicationMusicPlayer.shared func playSong(_ song: Song) { musicPlayer.queue = [song] Task { try await musicPlayer.prepareToPlay() try await musicPlayer.play() } Is there anything else we can investigate about what may be causing specific song IDs not to play on this specific device? Even if we remove line 6 musicPlayer.prepareToPlay() we still see the same console error when running playSong with the Songs that don't work. It is always the same song IDs that we can play and always the same song IDs that we cannot get to play, even trying them across different projects with different bundle identifiers. We can tap to play a song that works, and it starts playing immediately. Then tap a song that doesn't work, and nothing happens. Then back to a song that works. It's consistent which songs succeed and fail on this device. Perhaps there is an issue specific to this very iPad when it comes to certain specific songs, but we'd like to be confident that an app relying on MusicKit will be able to play songs that have been successfully loaded with a MusicCatalogResourceResponse. Thanks for any help or suggestions about what we may be able to investigate further on the device or what we should consider when launching an app that expects anyone with Apple Music to be able to listen to any of the songs loaded by the app. Specific iPad details: iPad Pro (12.9-inch) (6th generation) running iPadOS 26.4 Beta Two of the song IDs that won't play on this iPad (even though we can access and display their album artwork and all other information): 943204000 and 1441164805
2
1
199
3w
AVAudioSession : Audio issues when recording the screen in an app that changes IOBufferDuration on iOS 26.
Among Japanese end users, audio issues during screen recording—primarily in game applications—have become a topic of discussion. We have confirmed that the trigger for this issue is highly likely to be related to changes to IOBufferDuration. When using setPreferredIOBufferDuration and the IOBufferDuration is set to a value smaller than the default, audio problems occur in the recorded screen capture video. Audio playback is performed using AudioUnit (RemoteIO). https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc This issue was not observed on iOS 18, and it appears to have started occurring after upgrading to iOS 26. We provide an audio middleware solution, and we had incorporated changes to IOBufferDuration into our product to achieve low-latency audio playback. As a result, developers using our product as well as their end users are being affected by this issue. We kindly request that this issue be investigated and addressed in a future update. “This document has been translated by AI. The original text is included below for reference.” 日本のエンドユーザー間で主にゲームアプリケーションにおける画面収録時の音声の問題が話題になっています。 こちらの症状のトリガーが、IOBufferDurationの変更によるものである可能性が高いことを確認しました。 setPreferredIOBufferDurationを使用し、IOBufferDurationがデフォルトより小さい状態の時、画面収録された動画の音声に問題が発生することをしています。 音声の再生にはAudioUnit(RemoteIO)を使用しています。 https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc iOS 18ではこのような問題は確認されておらず、iOS26になってから問題が発生しているようです。 私たちはオーディオミドルウェアを提供しており、低遅延の再生のためにIOBufferDurationの変更を製品に組み込んでいました。 そのため、弊社製品をご利用いただいている開発者およびエンドユーザーの皆様がこの不具合の影響を受けています。 こちらの不具合の調査及び修正対応を検討いただけますでしょうか。
2
0
465
6d
Always audio from latest connected external USB mic
Hello! I've two mics connected to a USB-hub. The USB-hub is then connected to my iPad. Both mics are part of the audio session's list of available inputs. The problem is that regardless of which mic I select in my app (using setPreferredInput() on the audio session), the audio keeps coming from the mic that was last connected to the USB-hub. Anyone that knows if this is a limitation in iPadOS/iOS?
1
1
209
Jul ’25
Audio session activation occasionally fails from CarPlay
I'm working on adding CarPlay support to an audio app and am running into an issue. Occasionally, when a user opens the app from CarPlay while the main app scene is either not connected or is currently in the background, I will receive an error when attempting to activate the audio session. The code below mimics my setup: do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio) try AVAudioSession.sharedInstance().setActive(true) } catch { print(error) // NSOSStatusErrorDomain - 560557684: Session activation failed } That error code maps to AVAudioSession.ErrorCode.cannotInterruptOthers. Once in this state, all subsequent attempts to play different pieces of content will fail. However, things will start working normally if the user opens the app on their phone and tries again from CarPlay (while the app is in the foreground on their phone). I'm not sure why it would behave this way and want to note that I do have the audio background mode capability enabled. Has anyone else encountered this? Are there any workarounds or changes I could make to prevent this from happening?
0
1
199
Apr ’25
MPNowPlayingInfoCenter nowPlayingInfo throttled
Hello, I have been running into issues with setting nowPlayingInfo information, specifically updating information for CarPlay and the CPNowPlayingTemplate. When I start playback for an item, I see lock screen information update as expected, along with the CarPlay now playing information. However, the playing items are books with collections of tracks. When I select a new track(chapter) within the book, I set the MPMediaItemPropertyTitle to the new chapter name. This change is reflected correctly on the lock screen, but almost never appears correctly on the CarPlay CPNowPlayingTemplate. The previous chapter title remains set and never updates. I see "Application exceeded audio metadata throttle limit." in the debug console fairly frequently. From that a I figured that I need to minimize updates to the nowPlayingInfo dictionary. What I did: I store the metadata dictionary in a local dictionary and only set values in the main nowPlayingInfo dictionary when they are different from the current value. I kick off the nowPlayingInfo update via a task that initially sleeps for around 2 seconds (not a final value, just for my current testing). If a previous Task is active, it gets cancelled, so that only one update can happen within that time window. Neither of these things have been sufficient. I can switch between different titles entirely and the information updates (including cover art). But when I switch chapters within a title, the MPMediaItemPropertyTitle continues to get dropped. I know the value is getting set, because it updates on the lock screen correctly. In total, I have 12 keys I update for info, though with the above changes, usually 2-4 of them actually get updated with high frequency. I am running out of ideas to satisfy the throttling thresholds to accurately display metadata. I could use some advice. Thanks.
4
1
222
May ’25
Improving Speech Analyzer Transcription for technical terms
I am developing an app with transcription and I am exploring ways to improve the transcription from the SpeechAnalyzer/Transcriber for technical terms. SFSpeech... recognition had the capability of being augmented by contextualStrings. Does something similar exist for SpeechAnalyzer/Transcriber? If so please point me towards the documentation and any sample code that may exist for this. If there are other options, please let me know.
1
1
307
Sep ’25
Indicate Packet Loss With AVAudioConverter for OPUS Decoding
I'm using an AVAudioConverter object to decode an OPUS stream for VoIP. The decoding itself works well, however, whenever the stream stalls (no more audio packet is available to decode because of network instability) this can be heard in crackling / abrupt stop in decoded audio. OPUS can mitigate this by indicating packet loss by passing a null pointer in the C-library to int opus_decode_float (OpusDecoder * st, const unsigned char * data, opus_int32 len, float * pcm, int frame_size, int decode_fec), see https://opus-codec.org/docs/opus_api-1.2/group__opus__decoder.html#ga9c554b8c0214e24733a299fe53bb3bd2. However, with AVAudioConverter using Swift I'm constructing an AVAudioCompressedBuffer like so:         let compressedBuffer = AVAudioCompressedBuffer(             format: VoiceEncoder.Constants.networkFormat,             packetCapacity: 1,             maximumPacketSize: data.count         )         compressedBuffer.byteLength = UInt32(data.count)         compressedBuffer.packetCount = 1   compressedBuffer.packetDescriptions! .pointee.mDataByteSize = UInt32(data.count)         data.copyBytes(             to: compressedBuffer.data .assumingMemoryBound(to: UInt8.self),             count: data.count         ) where data: Data contains the raw OPUS frame to be decoded. How can I specify data loss in this context and cause the AVAudioConverter to output PCM data whenever no more input data is available? More context: I'm specifying the audio format like this:         static let frameSize: UInt32 = 960         static let sampleRate: Float64 = 48000.0         static var networkFormatStreamDescription = AudioStreamBasicDescription(             mSampleRate: sampleRate,             mFormatID: kAudioFormatOpus,             mFormatFlags: 0,             mBytesPerPacket: 0,             mFramesPerPacket: frameSize,             mBytesPerFrame: 0,             mChannelsPerFrame: 1,             mBitsPerChannel: 0,             mReserved: 0         )         static let networkFormat = AVAudioFormat( streamDescription: &networkFormatStreamDescription )! I've tried 1) setting byteLength and packetCount to zero and 2) returning nil but setting .haveData in the AVAudioConverterInputBlock I'm using with no success.
1
1
922
May ’25
CMFormatDescription.audioStreamBasicDescription has wrong or unexpected sample rate for audio channels with different sample rates
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))
0
1
129
Aug ’25
Lock screen media controls for MusicKit/ ApplicationMusicPlayer
Hi, when using ApplicationMusicPlayer from MusicKit my app automatically gets the media controls on the lock screen: Play/ Pause, Skip Buttons, Playback Position etc. I would like to customize these. Tried a bunch of things, e.g. using MPRemoteCommandCenter. So far I haven't had any success. Does anyone know how I can customize the media controls of ApplicationMusicPlayer. Thank you.
2
0
529
Sep ’25
How can third-party iOS apps obtain real-time waveform / spectrogram data for Apple Music tracks (similar to djay & other DJ apps)?
Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.
1
1
647
Dec ’25
Creating RTP-MIDI Sessions via MIDINetworkSession C API (dlopen/dlsym) on macOS 15?
I’m an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!
0
1
214
Jun ’25
AVAudioRecorder loses audio recorded before interruption
Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!
1
1
386
Dec ’25
On iOS 18, Mandarin is read aloud as Cantonese
Please include the line below in follow-up emails for this request. Case-ID: 11089799 When using AVSpeechUtterance and setting it to play in Mandarin, if Siri is set to Cantonese on iOS 18, it will be played in Cantonese. There is no such issue on iOS 17 and 16. 1.let utterance = AVSpeechUtterance(string: textView.text) let voice = AVSpeechSynthesisVoice(language: "zh-CN") utterance.voice = voice 2.In the phone settings, Siri is set to Cantonese
4
1
801
2w
Question about PT Framework channel tone behaviour
I've been wondering if there is a way to modify or even disable tones for indicating channel states. The behaviour regarding tones seems like a black box with little documentation. During migration to Apple's PT Framework we've noticed that there are few scenarios where a tone is played which doesn't match certain certifications. For example; moving from a channel to another produces a tone which would fail a test case. I understand the reasoning fully, as it marks that the channel is ready to transmit or receive, but this doesn't mirror the behaviour of TETRA which would be wanted in this case. I'm also wondering if there would be any way to directly communicate feedback regarding PT Framework?
3
0
413
Oct ’25
AVSpeechSynthesizer & Bluetooth Issues
Hello, I have a CarPlay Navigation app and utilize the AVSpeechSynthesizer to speak directions to a user. Everything works great on my CarPlay simulator as well as when plugged into my GMC truck. However, I found out yesterday that one of my users with a Ford truck the audio would cut in an out. After much troubleshooting, I was able to replicate this on my own truck when using Bluetooth to connect to CarPlay. My user was also utilizing Bluetooth. Has anyone else experienced this? Is there a fix to the problem? import SwiftUI import AVFoundation class TextToSpeechService: NSObject, ObservableObject, AVSpeechSynthesizerDelegate { private var speechSynthesizer = AVSpeechSynthesizer() static let shared = TextToSpeechService() override init() { super.init() speechSynthesizer.delegate = self } func configureAudioSession() { speechSynthesizer.delegate = self do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .voicePrompt, options: [.mixWithOthers, .allowBluetooth]) } catch { print("Failed to set audio session category: \(error.localizedDescription)") } } func speak(_ text: String) { Task(priority: .high) { let speechUtterance = AVSpeechUtterance(string: text) speechUtterance.voice = AVSpeechSynthesisVoice(language: AVSpeechSynthesisVoice.currentLanguageCode()) try AVAudioSession.sharedInstance().setActive(true, options: .notifyOthersOnDeactivation) speechSynthesizer.speak(speechUtterance) } } func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { Task { stopSpeech() try AVAudioSession.sharedInstance().setActive(false) } } func stopSpeech() { speechSynthesizer.stopSpeaking(at: .immediate) } }
1
1
777
Jan ’26
[AVPlayerItemVideoOutput initWithPixelBufferAttributes:] output attributes setting not work
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!
1
1
840
Nov ’25
iPad app on macOS not asking for microphone permission
Hello, I have an iOS app that is recording audio that is working fine on iPads/iPhones. It asks for microphone permission and after that recording works. I installed the same app on my M3 MacBook via TestFlight since iPad apps are supposed to work without a change that way. The app starts fine and everything, but it never asks for Microphone permission, so I can't record. Do I need to do something to make this happen (this is not macCatalyst, its running the arm64 iPhone binary on macOS) thanks
2
1
890
Mar ’25
Unable to play audio via MusicKit
Hey folks, I'm running into an odd issue suddenly with an app that had a working MusicKit integration before. I'm using ApplicationMusicPlayer to play Apple Music albums and songs. I'm testing on a physical device, signed in to Apple ID, and with a valid subscription. Apple Music via the first-party app works entirely fine on this device. Attempting to play back any content at all gives the log: <ICUserIdentityStoreACAccountBackend: 0x1070bf3e0> Failed to initialize primary apple 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} [ICUserIdentityStore] - initializing account histories with activeAccountDSID = nil, activeLockerAccountDSID = nil, timestamp = 14605951908 [ICUserIdentityStore] Failed to fetch local store account with error: Error Domain=ICError Code=-7013 "Client is not entitled to access account store" UserInfo={NSDebugDescription=Client is not entitled to access account store}. The album artwork, track names, etc, all appear in the control center playback controls, but the music doesn't play. Trying to trigger playback with control center just results in it skipping to the next track, which doesn't play either. This exact code used to work. I have the MusicKit service selected in Apple Connect. Since this isn't entitlement-based, I'm not sure how else to check that I'm set up correctly. I've tried deleting/reinstalling the app, restarting the device, cleaning/rebuilding, and deleting DerivedData, to no avail. Any help? Running Xcode 16.4 (16F6), testing on iOS 18.5 (22F76)
Replies
0
Boosts
1
Views
174
Activity
Jun ’25
AVAudioFile.read extremely slow after seeking in FLAC and MP3 files
I'm developing an audio player app that uses AVAudio​File to read PCM data from various formats. I'm experiencing severe performance issues when seeking in FLAC, while other compressed formats (M4A/AAC) work correctly. I don't intend to use them in my app, but I also tested mp3 files just by curiosity and they also have this issue. Environment: macOS 26 (Tahoe) Xcode 26.3 Apple Silicon (M1) The issue: After setting AVAudio​File​.frame​Position to a position mid-file, the subsequent call to AVAudio​File​.read(into​:frame​Count:) blocks for an unreasonable amount of time for FLAC and MP3 files. The delay scales linearly with the seek target, seeking near the beginning is fast, seeking toward the end is proportionally slower, which suggests the decoder is decoding linearly from the beginning of the file rather than using any seek index. (My app deals with “images” of Audio CDs ripped as a single long audio file.) The issue is particularly severe when reading files from an SMB network share (server on Ethernet, client on Wi-Fi with the access point ~2 meters away in line of sight). Quick Benchmark results: I tested with the same 75-minute audio content (16-bit/44.1 kHz stereo, 200,502,708 frames) encoded in five formats, seeking to the midpoint. Over SMB (Local Network, Server on Ethernet, Client on WiFi): Format | Seek + Read Time ----------|------------------ WAV | 0.007 s AIFF | 0.009 s Apple | 0.015 s Lossless | MP3 | 9.2 s FLAC | 30.2 s Locally (MacBook Air M1 SSD) : Format | Seek + Read Time ----------|------------------ WAV | 0.0005 s AIFF | 0.0004 s Apple | 0.0011 s Lossless | MP3 | 0.1958 s FLAC | 0.7528 s WAV, AIFF, and M4A all seek virtually instantly (< 15 ms). MP3 and FLAC exhibit linear-time behavior, with FLAC being the worst affected. Note that M4A (AAC) is also a compressed format that requires decoding after seeking, yet it completes in 15 ms. This rules out any inherent limitation of compressed formats, the MP4 container's packet index (stts/stco) is clearly being used for fast random access. Both MP3 (Xing/LAME TOC) and FLAC (SEEKTABLE metadata block) have their own seek mechanisms that should provide similar performance. Minimal CLI tool to reproduce: import Foundation guard CommandLine.arguments.count > 1 else { print("Usage: FLACSpeed <audio-file-path>") exit(1) } let path = CommandLine.arguments[1] let fileURL = URL(fileURLWithPath: path) do { let file = try AVAudioFile(forReading: fileURL) let format = file.processingFormat let buffer = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: 8192)! let totalFrames = file.length let seekTarget = totalFrames / 2 print("File: \(fileURL.lastPathComponent)") print("Format: \(format)") print("Total frames: \(totalFrames)") print("Seeking to frame: \(seekTarget)") file.framePosition = seekTarget let start = CFAbsoluteTimeGetCurrent() try file.read(into: buffer, frameCount: 8192) let elapsed = CFAbsoluteTimeGetCurrent() - start print("Read after seek took \(elapsed) seconds") } catch { print("Error: \(error.localizedDescription)") exit(1) } Expected behavior: AVAudio​File​.read(into​:frame​Count:) after setting frame​Position should use the available seek mechanisms in FLAC and MP3 files for fast random access, as it already does for M4A (AAC). Even accounting for the fact that seek tables provide approximate (not sample-precise) positioning, the "jump to nearest index point + decode forward" approach should complete in milliseconds, not seconds. Workaround: For FLAC, I've worked around this by using libFLAC directly, which provides instant seeking via FLAC__stream​_decoder​_seek​_absolute(). libFLAC Performance: For comparison, libFLAC's FLAC__stream​_decoder​_seek​_absolute() performs the same seek + read on the same FLAC file in around 0.015, using the FLAC seek table to jump to the nearest preceding seek point, then decoding forward a small number of frames to the exact target sample.
Replies
0
Boosts
1
Views
95
Activity
1d
MusicKit - ApplicationMusicPlayer fails to play certain Songs
[Note: this issue was happening on a main testing device, and after testing the same code on other devices, this issue is only happening on 1 out of 4 devices] We are successfully getting a MusicCatalogResourceResponse for every song ID where we make the MusicCatalogResourceRequest. We are able to display the song title, artist name, and album artwork for each Song in the response. However - when we go to play the song, there are some songs that play, and several songs that do not play. For the songs that don't play, the console shows “Failed to prepareToPlay error=<MPMusicPlayerControllerErrorDomain.6 "Failed to prepare to play" {}>” let musicPlayer = ApplicationMusicPlayer.shared func playSong(_ song: Song) { musicPlayer.queue = [song] Task { try await musicPlayer.prepareToPlay() try await musicPlayer.play() } Is there anything else we can investigate about what may be causing specific song IDs not to play on this specific device? Even if we remove line 6 musicPlayer.prepareToPlay() we still see the same console error when running playSong with the Songs that don't work. It is always the same song IDs that we can play and always the same song IDs that we cannot get to play, even trying them across different projects with different bundle identifiers. We can tap to play a song that works, and it starts playing immediately. Then tap a song that doesn't work, and nothing happens. Then back to a song that works. It's consistent which songs succeed and fail on this device. Perhaps there is an issue specific to this very iPad when it comes to certain specific songs, but we'd like to be confident that an app relying on MusicKit will be able to play songs that have been successfully loaded with a MusicCatalogResourceResponse. Thanks for any help or suggestions about what we may be able to investigate further on the device or what we should consider when launching an app that expects anyone with Apple Music to be able to listen to any of the songs loaded by the app. Specific iPad details: iPad Pro (12.9-inch) (6th generation) running iPadOS 26.4 Beta Two of the song IDs that won't play on this iPad (even though we can access and display their album artwork and all other information): 943204000 and 1441164805
Replies
2
Boosts
1
Views
199
Activity
3w
AVAudioSession : Audio issues when recording the screen in an app that changes IOBufferDuration on iOS 26.
Among Japanese end users, audio issues during screen recording—primarily in game applications—have become a topic of discussion. We have confirmed that the trigger for this issue is highly likely to be related to changes to IOBufferDuration. When using setPreferredIOBufferDuration and the IOBufferDuration is set to a value smaller than the default, audio problems occur in the recorded screen capture video. Audio playback is performed using AudioUnit (RemoteIO). https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc This issue was not observed on iOS 18, and it appears to have started occurring after upgrading to iOS 26. We provide an audio middleware solution, and we had incorporated changes to IOBufferDuration into our product to achieve low-latency audio playback. As a result, developers using our product as well as their end users are being affected by this issue. We kindly request that this issue be investigated and addressed in a future update. “This document has been translated by AI. The original text is included below for reference.” 日本のエンドユーザー間で主にゲームアプリケーションにおける画面収録時の音声の問題が話題になっています。 こちらの症状のトリガーが、IOBufferDurationの変更によるものである可能性が高いことを確認しました。 setPreferredIOBufferDurationを使用し、IOBufferDurationがデフォルトより小さい状態の時、画面収録された動画の音声に問題が発生することをしています。 音声の再生にはAudioUnit(RemoteIO)を使用しています。 https://developer.apple.com/documentation/avfaudio/avaudiosession/setpreferrediobufferduration(_:)?language=objc iOS 18ではこのような問題は確認されておらず、iOS26になってから問題が発生しているようです。 私たちはオーディオミドルウェアを提供しており、低遅延の再生のためにIOBufferDurationの変更を製品に組み込んでいました。 そのため、弊社製品をご利用いただいている開発者およびエンドユーザーの皆様がこの不具合の影響を受けています。 こちらの不具合の調査及び修正対応を検討いただけますでしょうか。
Replies
2
Boosts
0
Views
465
Activity
6d
Always audio from latest connected external USB mic
Hello! I've two mics connected to a USB-hub. The USB-hub is then connected to my iPad. Both mics are part of the audio session's list of available inputs. The problem is that regardless of which mic I select in my app (using setPreferredInput() on the audio session), the audio keeps coming from the mic that was last connected to the USB-hub. Anyone that knows if this is a limitation in iPadOS/iOS?
Replies
1
Boosts
1
Views
209
Activity
Jul ’25
Audio session activation occasionally fails from CarPlay
I'm working on adding CarPlay support to an audio app and am running into an issue. Occasionally, when a user opens the app from CarPlay while the main app scene is either not connected or is currently in the background, I will receive an error when attempting to activate the audio session. The code below mimics my setup: do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio) try AVAudioSession.sharedInstance().setActive(true) } catch { print(error) // NSOSStatusErrorDomain - 560557684: Session activation failed } That error code maps to AVAudioSession.ErrorCode.cannotInterruptOthers. Once in this state, all subsequent attempts to play different pieces of content will fail. However, things will start working normally if the user opens the app on their phone and tries again from CarPlay (while the app is in the foreground on their phone). I'm not sure why it would behave this way and want to note that I do have the audio background mode capability enabled. Has anyone else encountered this? Are there any workarounds or changes I could make to prevent this from happening?
Replies
0
Boosts
1
Views
199
Activity
Apr ’25
MPNowPlayingInfoCenter nowPlayingInfo throttled
Hello, I have been running into issues with setting nowPlayingInfo information, specifically updating information for CarPlay and the CPNowPlayingTemplate. When I start playback for an item, I see lock screen information update as expected, along with the CarPlay now playing information. However, the playing items are books with collections of tracks. When I select a new track(chapter) within the book, I set the MPMediaItemPropertyTitle to the new chapter name. This change is reflected correctly on the lock screen, but almost never appears correctly on the CarPlay CPNowPlayingTemplate. The previous chapter title remains set and never updates. I see "Application exceeded audio metadata throttle limit." in the debug console fairly frequently. From that a I figured that I need to minimize updates to the nowPlayingInfo dictionary. What I did: I store the metadata dictionary in a local dictionary and only set values in the main nowPlayingInfo dictionary when they are different from the current value. I kick off the nowPlayingInfo update via a task that initially sleeps for around 2 seconds (not a final value, just for my current testing). If a previous Task is active, it gets cancelled, so that only one update can happen within that time window. Neither of these things have been sufficient. I can switch between different titles entirely and the information updates (including cover art). But when I switch chapters within a title, the MPMediaItemPropertyTitle continues to get dropped. I know the value is getting set, because it updates on the lock screen correctly. In total, I have 12 keys I update for info, though with the above changes, usually 2-4 of them actually get updated with high frequency. I am running out of ideas to satisfy the throttling thresholds to accurately display metadata. I could use some advice. Thanks.
Replies
4
Boosts
1
Views
222
Activity
May ’25
Improving Speech Analyzer Transcription for technical terms
I am developing an app with transcription and I am exploring ways to improve the transcription from the SpeechAnalyzer/Transcriber for technical terms. SFSpeech... recognition had the capability of being augmented by contextualStrings. Does something similar exist for SpeechAnalyzer/Transcriber? If so please point me towards the documentation and any sample code that may exist for this. If there are other options, please let me know.
Replies
1
Boosts
1
Views
307
Activity
Sep ’25
Indicate Packet Loss With AVAudioConverter for OPUS Decoding
I'm using an AVAudioConverter object to decode an OPUS stream for VoIP. The decoding itself works well, however, whenever the stream stalls (no more audio packet is available to decode because of network instability) this can be heard in crackling / abrupt stop in decoded audio. OPUS can mitigate this by indicating packet loss by passing a null pointer in the C-library to int opus_decode_float (OpusDecoder * st, const unsigned char * data, opus_int32 len, float * pcm, int frame_size, int decode_fec), see https://opus-codec.org/docs/opus_api-1.2/group__opus__decoder.html#ga9c554b8c0214e24733a299fe53bb3bd2. However, with AVAudioConverter using Swift I'm constructing an AVAudioCompressedBuffer like so:         let compressedBuffer = AVAudioCompressedBuffer(             format: VoiceEncoder.Constants.networkFormat,             packetCapacity: 1,             maximumPacketSize: data.count         )         compressedBuffer.byteLength = UInt32(data.count)         compressedBuffer.packetCount = 1   compressedBuffer.packetDescriptions! .pointee.mDataByteSize = UInt32(data.count)         data.copyBytes(             to: compressedBuffer.data .assumingMemoryBound(to: UInt8.self),             count: data.count         ) where data: Data contains the raw OPUS frame to be decoded. How can I specify data loss in this context and cause the AVAudioConverter to output PCM data whenever no more input data is available? More context: I'm specifying the audio format like this:         static let frameSize: UInt32 = 960         static let sampleRate: Float64 = 48000.0         static var networkFormatStreamDescription = AudioStreamBasicDescription(             mSampleRate: sampleRate,             mFormatID: kAudioFormatOpus,             mFormatFlags: 0,             mBytesPerPacket: 0,             mFramesPerPacket: frameSize,             mBytesPerFrame: 0,             mChannelsPerFrame: 1,             mBitsPerChannel: 0,             mReserved: 0         )         static let networkFormat = AVAudioFormat( streamDescription: &networkFormatStreamDescription )! I've tried 1) setting byteLength and packetCount to zero and 2) returning nil but setting .haveData in the AVAudioConverterInputBlock I'm using with no success.
Replies
1
Boosts
1
Views
922
Activity
May ’25
CMFormatDescription.audioStreamBasicDescription has wrong or unexpected sample rate for audio channels with different sample rates
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))
Replies
0
Boosts
1
Views
129
Activity
Aug ’25
Lock screen media controls for MusicKit/ ApplicationMusicPlayer
Hi, when using ApplicationMusicPlayer from MusicKit my app automatically gets the media controls on the lock screen: Play/ Pause, Skip Buttons, Playback Position etc. I would like to customize these. Tried a bunch of things, e.g. using MPRemoteCommandCenter. So far I haven't had any success. Does anyone know how I can customize the media controls of ApplicationMusicPlayer. Thank you.
Replies
2
Boosts
0
Views
529
Activity
Sep ’25
How can third-party iOS apps obtain real-time waveform / spectrogram data for Apple Music tracks (similar to djay & other DJ apps)?
Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: 1. Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? 2. If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? 3. Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.
Replies
1
Boosts
1
Views
647
Activity
Dec ’25
Creating RTP-MIDI Sessions via MIDINetworkSession C API (dlopen/dlsym) on macOS 15?
I’m an amateur developer working on a free utility for composers/producers, for which the macOS release needs to create and name RTP-MIDI sessions in Audio MIDI Setup from the command line (so I can ship a small C helper instead of telling users to click through the UI). Here’s what I’ve tried so far, without luck: • Plist hacks: Injecting entries into ~/Library/Audio/MIDI Configurations/*.mcfg works when AMS is closed, but AMS immediately locks and reverts my changes when it’s open. • CoreMIDI C API: I can create virtual ports with MIDISourceCreate, but attempting MIDIObjectGetDataProperty on the apple.midirtp.session plugin always returns err –10836. • Obj-C & Swift: Loading MIDINetworkSession and calling defaultSession, init, setNetworkName: and setting enabled = YES doesn’t produce a new session object in the Network panel. • dlopen/dlsym: I extracted the real CoreMIDI binary out of the dyld shared cache and tried binding _MIDINetworkSessionCreate, _SetName, _SetEnabled, etc., but all the symbols come back null or my tool segfaults. • Plugin registration: I’ve pulled the factory UUID (70C9C5EA-7C65-11D8-B317-000393A34B5A) from /System/Library/Extensions/AppleMIDIRTPDriver.plugin/Contents/Info.plist and called CFPlugInRegisterFactories, but it still never exposes the session-creation calls. At this point I’m convinced I’m either loading the wrong binary or missing one critical step in registering the RTP-MIDI plugin’s private API. Can anyone point me to: The exact path of the dylib or bundle that actually exports the MIDINetworkSessionCreate/MIDINetworkSessionSetName/MIDINetworkSessionSetEnabled symbols? A minimal working snippet (C or Obj-C) that reliably creates and names a Network-MIDI session? Any pointers, sample code, or even ideas about where Apple hides this functionality on macOS 15 would be hugely appreciated. Thanks!
Replies
0
Boosts
1
Views
214
Activity
Jun ’25
AVAudioRecorder loses audio recorded before interruption
Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!
Replies
1
Boosts
1
Views
386
Activity
Dec ’25
Get device Voice Isolation status via Core Audio?
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.
Replies
1
Boosts
1
Views
886
Activity
Nov ’25
On iOS 18, Mandarin is read aloud as Cantonese
Please include the line below in follow-up emails for this request. Case-ID: 11089799 When using AVSpeechUtterance and setting it to play in Mandarin, if Siri is set to Cantonese on iOS 18, it will be played in Cantonese. There is no such issue on iOS 17 and 16. 1.let utterance = AVSpeechUtterance(string: textView.text) let voice = AVSpeechSynthesisVoice(language: "zh-CN") utterance.voice = voice 2.In the phone settings, Siri is set to Cantonese
Replies
4
Boosts
1
Views
801
Activity
2w
Question about PT Framework channel tone behaviour
I've been wondering if there is a way to modify or even disable tones for indicating channel states. The behaviour regarding tones seems like a black box with little documentation. During migration to Apple's PT Framework we've noticed that there are few scenarios where a tone is played which doesn't match certain certifications. For example; moving from a channel to another produces a tone which would fail a test case. I understand the reasoning fully, as it marks that the channel is ready to transmit or receive, but this doesn't mirror the behaviour of TETRA which would be wanted in this case. I'm also wondering if there would be any way to directly communicate feedback regarding PT Framework?
Replies
3
Boosts
0
Views
413
Activity
Oct ’25
AVSpeechSynthesizer & Bluetooth Issues
Hello, I have a CarPlay Navigation app and utilize the AVSpeechSynthesizer to speak directions to a user. Everything works great on my CarPlay simulator as well as when plugged into my GMC truck. However, I found out yesterday that one of my users with a Ford truck the audio would cut in an out. After much troubleshooting, I was able to replicate this on my own truck when using Bluetooth to connect to CarPlay. My user was also utilizing Bluetooth. Has anyone else experienced this? Is there a fix to the problem? import SwiftUI import AVFoundation class TextToSpeechService: NSObject, ObservableObject, AVSpeechSynthesizerDelegate { private var speechSynthesizer = AVSpeechSynthesizer() static let shared = TextToSpeechService() override init() { super.init() speechSynthesizer.delegate = self } func configureAudioSession() { speechSynthesizer.delegate = self do { try AVAudioSession.sharedInstance().setCategory(.playback, mode: .voicePrompt, options: [.mixWithOthers, .allowBluetooth]) } catch { print("Failed to set audio session category: \(error.localizedDescription)") } } func speak(_ text: String) { Task(priority: .high) { let speechUtterance = AVSpeechUtterance(string: text) speechUtterance.voice = AVSpeechSynthesisVoice(language: AVSpeechSynthesisVoice.currentLanguageCode()) try AVAudioSession.sharedInstance().setActive(true, options: .notifyOthersOnDeactivation) speechSynthesizer.speak(speechUtterance) } } func speechSynthesizer(_ synthesizer: AVSpeechSynthesizer, didFinish utterance: AVSpeechUtterance) { Task { stopSpeech() try AVAudioSession.sharedInstance().setActive(false) } } func stopSpeech() { speechSynthesizer.stopSpeaking(at: .immediate) } }
Replies
1
Boosts
1
Views
777
Activity
Jan ’26
[AVPlayerItemVideoOutput initWithPixelBufferAttributes:] output attributes setting not work
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!
Replies
1
Boosts
1
Views
840
Activity
Nov ’25
iPad app on macOS not asking for microphone permission
Hello, I have an iOS app that is recording audio that is working fine on iPads/iPhones. It asks for microphone permission and after that recording works. I installed the same app on my M3 MacBook via TestFlight since iPad apps are supposed to work without a change that way. The app starts fine and everything, but it never asks for Microphone permission, so I can't record. Do I need to do something to make this happen (this is not macCatalyst, its running the arm64 iPhone binary on macOS) thanks
Replies
2
Boosts
1
Views
890
Activity
Mar ’25