AVAudioSessionCategoryOptionAllowBluetooth is marked as deprecated in iOS 8 in iOS 26 beta 5 when this option was not deprecated in iOS 18.6. I think this is a mistake and the deprecation is in iOS 26. Am I right? It seems that the substitute for this option is "AVAudioSessionCategoryOptionAllowBluetoothHFP". The documentation does not make clear if the behaviour is exactly the same or if any difference should be expected... Has anyone used this option in iOS 26? Should I expect any difference with the current behaviour of "AVAudioSessionCategoryOptionAllowBluetooth"? Thank you.

I have an app under development - demo here - https://youtu.be/VbAfUk_eYl0?si=s6EDBx-4G6P_QbZO - which is sort of an audio player for airdropped files - something useful to musicians who dump work in progress to their phone, make notes, revise and update. I've been testing my handling of audio session interruption notifications, but seems to be a lot of inconsistency in how, when and why iOS delivers them, and I'm wondering if there is some rhyme or reason to it that I'm just not detecting. For example, I am playing a song in my app. Switch to Apple Music and start playing a song there. My app gets an interruption began notification - this is consistent. Switch back to my app, and about half the time, I will get an interruption ended notification (coupled often with a blast of the tail of whatever audio buffer was partially played when the interruption started, even though the engine was stopped - and followed by call to my AVAudioPlayerNodeCompletionCallback - is there some way to avoid this?). Half the time I don't get an interruption ended notification; my app can (as expected) end the interruption by activating the AVAudioSession and playing something. I have not been able to determine any pattern to this behavior, other than that if my app started playing using AVAudioPlayerNode.scheduleSegment rather than scheduleFile I think the notification will be consistently delivered on app activation rather than when I activate the session programmatically. I would like my app to behave deterministically, and would appreciate any help in deciphering what causes the inconsistent behavior in notifications from iOS.

Hi all! I have been experiencing some issues when using the AVAudioEngine to play audio and record input while doing a voice chat (through the PTT Interface). I noticed if I connect any players to the AudioGraph OR call start that the audio session becomes active (this is on iOS). I don't see anything in the docs or the header files in the AVFoundation, but is it possible that calling the stop method on an engine deactivates the audio session too? In a normal app this behavior seems logical, but when using PTT all activation and deactivation of the audio session must go through the framework and its delegate methods. The issue I am debugging is that when the engine with the input node tapped gets stopped, and there is a gap between the input and when the server replies with inbound audio to be played and something seems to be getting the hardware/audio session into a jammed state. Thanks for any feedback and/or confirmation on this behavior!

I’m experiencing an unusual audio issue with AirPods on macOS Sequoia while developing VoIP applications like Zoom and FaceTime. When AirPods are connected, the other party’s voice sometimes sounds unnaturally stretched (approximately twice as long). This problem can be temporarily fixed by switching the sound output settings from AirPods to speakers and then back to AirPods. From our analysis, the issue appears to be related to the sample rate provided by AudioObjectGetPropertyData. Here’s what we’ve observed: When the issue occurs, the AudioStreamBasicDescription.sampleRate for AirPods is reported as 48000. Under normal conditions, it’s reported as 24000. It seems like the system is mistakenly returning a sample rate that doesn’t match the AirPods’ actual settings, perhaps defaulting to a system speaker value. Once the output setting is toggled, the correct sampleRate (24000) is retrieved. This discrepancy causes our application to transmit the audio stream at 48000, leading to the distorted playback. Has anyone encountered a similar issue or knows how to resolve it?

After upgrading to iOS 18.4, I'm no longer able to establish an AirPlay v1 connection to an audio system. The symptom is that the AirPlay route picker just spins when trying to connect to an audio system. It eventually gives up. I tested this on an iPhone 14, connecting to a HomePod, AirPort express, AppleTV and a Wiim Pro. If I try connecting with AirPlay v2, ex: using Apple Music, the connection succeeds and audio can be played. I'm the developer of an app that plays audio over AirPlay while also recording. My app has to use AirPlay v1 because AvAudioSession doesn't allow the policy .longFormAudio when the category is .playAndRecord. This issue is a real pain as it means my app is suddenly broken for many thousands of users. Is anyone else seeing this issue? Any suggestions for a workaround?

Hi, I've recently been working with the Apple Music API and have had success in loading all the playlists on my account, loading songs from each playlist, and adding songs to the ApplicationMusicPlayer.share.queue. The problem I'm running into is that not all songs from the playlist are being added to the queue, despite confirming all the songs are being based on the PlaybackView.swift I'm about to share with you. I would also like to answer other underlying questions if possible. I am also open to any other suggestions. In this scenario were also assuming isShuffled is true every time. How can I determine when a song has ended? How can I get the album title information? How can I get the current song title, album information, and artist information without pressing play? I can only seem to update the screen when I select my play meaning ApplicationMusicPlayer.shared.play() is being called. How do I get the endTime of the song? I believe it should be ApplicationMusicPlayer.shared.queue.currentEntry.endTime but this doesn't seem to work. // // PlayBackView.swift // // Created by Justin on 8/16/24. // import SwiftUI import MusicKit import Foundation enum PlayState { case play case pause } struct PlayBackView: View { @State var song: Track @Binding var songs: [Track]? @State var isShuffled = false @State private var playState: PlayState = .pause @State private var isFirstPlay = true private let player = ApplicationMusicPlayer.shared private var isPlaying: Bool { return (player.state.playbackStatus == .playing) } var body: some View { VStack { // Album Cover HStack(spacing: 20) { if let artwork = player.queue.currentEntry?.artwork { ArtworkImage(artwork, height: 100) } else { Image(systemName: "music.note") .resizable() .frame(width: 100, height: 100) } VStack(alignment: .leading) { // Song Title Text(player.queue.currentEntry?.title ?? "Song Title Not Found") .font(.title) .fixedSize(horizontal: false, vertical: true) // How do I get AlbumTitle from here?? // Artist Name Text(player.queue.currentEntry?.subtitle ?? "Artist Name Not Found") .font(.caption) } } .padding() Spacer() // Progress View // endTime doesn't work and not sure why. ProgressView(value: player.playbackTime, total: player.queue.currentEntry?.endTime ?? 1.00) .progressViewStyle(.linear) .tint(.red.opacity(0.5)) // Duration View HStack { Text(durationStr(from: player.playbackTime)) .font(.caption) Spacer() if let duration = player.queue.currentEntry?.endTime { Text(durationStr(from: duration)) .font(.caption) } } Spacer() Button { Task { do { try await player.skipToNextEntry() } catch { print(error.localizedDescription) } } } label: { Label("", systemImage: "forward.fill") .tint(.white) } Spacer() // Play/Pause Button Button(action: { handlePlayButton() isFirstPlay = false }, label: { Text(playState == .play ? "Pause" : isFirstPlay ? "Play" : "Resume") .frame(maxWidth: .infinity) }) .buttonStyle(.borderedProminent) .padding() .font(.largeTitle) .tint(.red) } .padding() .onAppear { if isShuffled { songs = songs?.shuffled() if let songs, let firstSong = songs.first { player.queue = .init(for: songs, startingAt: firstSong) player.state.shuffleMode = .songs } } } .onDisappear { player.stop() player.queue = [] player.playbackTime = .zero } } private func handlePlayButton() { Task { if isPlaying { player.pause() playState = .pause } else { playState = .play await playTrack() } } } @MainActor private func playTrack() async { do { try await player.play() } catch { print(error.localizedDescription) } } private func durationStr(from duration: TimeInterval) -> String { let seconds = Int(duration) let minutes = seconds / 60 let remainder = seconds % 60 // Format the string to ensure two digits for the remainder (seconds) return String(format: "%d:%02d", minutes, remainder) } } //#Preview { // PlayBackView() //}

Hello everyone, I'm implementing the new AVInputPickerInteraction API on iOS 26 to allow users to select their microphone from a custom settings menu before recording. The implementation seems correct, but I'm encountering a strange issue where the input selection immediately reverts to the previous device. The Situation: The picker is presented correctly via a manual call to .present(). I can see all available inputs (e.g., "iPhone Microphone" and "AirPods"). The current input is "iPhone Microphone". I tap on "AirPods". The UI updates to show "AirPods" as selected for a fraction of a second, then immediately jumps back to "iPhone Microphone". The same thing happens in reverse. It seems like the system is automatically reverting the audio route change requested by the picker. My Implementation: My setup follows the standard pattern discussed in the WWDC sessions. Setup Code: This setup is performed once before the user can trigger the picker. @available(iOS 26.0, *) var inputPickerInteraction: AVInputPickerInteraction? // Note: The AVAudioSession is configured to .playAndRecord // and set to active elsewhere in the code before this setup is called. if #available(iOS 26.0, *) { // Setup the picker let picker = AVInputPickerInteraction() self.inputPickerInteraction = picker self.view.addInteraction(picker) // Added to establish context } Presentation Code: When a user selects "Change Input" from my custom settings menu, I call .present() on the main thread. // In a delegate method from a custom menu if #available(iOS 26.0, *) { DispatchQueue.main.async { self.inputPickerInteraction?.present(animated: true) } } What I've already checked: The AVAudioSession is active and its category is .playAndRecord. The inputPickerInteraction object is not nil. The .present() method is being called on the main thread. The picker is added to a view using view.addInteraction() in the setup phase. I've reviewed my code to ensure there is no other logic that could be manually resetting the AVAudioSession's preferred input. Has anyone else experienced this behavior? I suspect this might be a bug in the new API, but I want to make sure I'm not missing a crucial step in managing the AVAudioSession state. Any insights or potential workarounds would be greatly appreciated. Thank you.

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.

I'm developing an iOS app that requires continuous audio recording. Currently, when a phone call comes in, the AVAudioSession is interrupted and recording stops completely during the ringing phase. While I understand recording should stop if the call is answered, my app needs to continue recording while the phone is merely ringing. I've observed that Apple's Voice Memos app maintains recording during incoming call rings. This indicates the hardware and iOS are capable of supporting this functionality. Request Please advise on any available AVAudioSession configurations or APIs that would allow my app to: Continue recording during an incoming call ring Only stop recording if/when the call is actually answered Impact This interruption significantly impacts the user experience and core functionality of my app. Workarounds like asking users to enable airplane mode are impractical and create a poor user experience. Questions Is there an approved way to maintain microphone access during call rings? If not currently possible, could this capability be considered for addition to a future iOS SDK? Are there any interim solutions or best practices Apple recommends for this use case? Thank you for your help. SUPPORT INFORMATION Did someone from Apple ask you to submit a code-level support request? No Do you have a focused test project that demonstrates your issue? Yes, I have a focused test project to submit with my request What code level support issue are you having? Problems with an Apple framework API in my app

I'm experiencing audio issues while developing for visionOS when playing PCM data through AVAudioPlayerNode. Issue Description: Occasionally, the speaker produces loud popping sounds or distorted noise This occurs during PCM audio playback using AVAudioPlayerNode The issue is intermittent and doesn't happen every time Technical Details: Platform: visionOS Device: vision pro / simulator Audio Framework: AVFoundation Audio Node: AVAudioPlayerNode Audio Format: PCM I would appreciate any insights on: Common causes of audio distortion with AVAudioPlayerNode Recommended best practices for handling PCM playback in visionOS Potential configuration issues that might cause this behavior Has anyone encountered similar issues or found solutions? Any guidance would be greatly helpful. Thank you in advance!

On macOS Sequoia, I'm having the hardest time getting this basic audio output to work correctly. I'm compiling in XCode using C99, and when I run this, I get audio for a split second, and then nothing, indefinitely. Any ideas what could be going wrong? Here's a minimum code example to demonstrate: #include <AudioToolbox/AudioToolbox.h> #include <stdint.h> #define RENDER_BUFFER_COUNT 2 #define RENDER_FRAMES_PER_BUFFER 128 // mono linear PCM audio data at 48kHz #define RENDER_SAMPLE_RATE 48000 #define RENDER_CHANNEL_COUNT 1 #define RENDER_BUFFER_BYTE_COUNT (RENDER_FRAMES_PER_BUFFER * RENDER_CHANNEL_COUNT * sizeof(f32)) void RenderAudioSaw(float* outBuffer, uint32_t frameCount, uint32_t channelCount) { static bool isInverted = false; float scalar = isInverted ? -1.f : 1.f; for (uint32_t frame = 0; frame < frameCount; ++frame) { for (uint32_t channel = 0; channel < channelCount; ++channel) { // series of ramps, alternating up and down. outBuffer[frame * channelCount + channel] = 0.1f * scalar * ((float)frame / frameCount); } } isInverted = !isInverted; } AudioStreamBasicDescription coreAudioDesc = { 0 }; AudioQueueRef coreAudioQueue = NULL; AudioQueueBufferRef coreAudioBuffers[RENDER_BUFFER_COUNT] = { NULL }; void coreAudioCallback(void* unused, AudioQueueRef queue, AudioQueueBufferRef buffer) { // 0's here indicate no fancy packet magic AudioQueueEnqueueBuffer(queue, buffer, 0, 0); } int main(void) { const UInt32 BytesPerSample = sizeof(float); coreAudioDesc.mSampleRate = RENDER_SAMPLE_RATE; coreAudioDesc.mFormatID = kAudioFormatLinearPCM; coreAudioDesc.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked; coreAudioDesc.mBytesPerPacket = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mFramesPerPacket = 1; coreAudioDesc.mBytesPerFrame = RENDER_CHANNEL_COUNT * BytesPerSample; coreAudioDesc.mChannelsPerFrame = RENDER_CHANNEL_COUNT; coreAudioDesc.mBitsPerChannel = BytesPerSample * 8; coreAudioQueue = NULL; OSStatus result; // most of the 0 and NULL params here are for compressed sound formats etc. result = AudioQueueNewOutput(&coreAudioDesc, &coreAudioCallback, NULL, 0, 0, 0, &coreAudioQueue); if (result != noErr) { assert(false == "AudioQueueNewOutput failed!"); abort(); } for (int i = 0; i < RENDER_BUFFER_COUNT; ++i) { uint32_t bufferSize = coreAudioDesc.mBytesPerFrame * RENDER_FRAMES_PER_BUFFER; result = AudioQueueAllocateBuffer(coreAudioQueue, bufferSize, &(coreAudioBuffers[i])); if (result != noErr) { assert(false == "AudioQueueAllocateBuffer failed!"); abort(); } } for (int i = 0; i < RENDER_BUFFER_COUNT; ++i) { RenderAudioSaw(coreAudioBuffers[i]->mAudioData, RENDER_FRAMES_PER_BUFFER, RENDER_CHANNEL_COUNT); coreAudioBuffers[i]->mAudioDataByteSize = coreAudioBuffers[i]->mAudioDataBytesCapacity; AudioQueueEnqueueBuffer(coreAudioQueue, coreAudioBuffers[i], 0, 0); } AudioQueueStart(coreAudioQueue, NULL); sleep(10); // some time to hear the audio AudioQueueStop(coreAudioQueue, true); AudioQueueDispose(coreAudioQueue, true); return 0; }

I am trying to debug the AAX version of my plugin (MIDI effect) on Pro Tools, but I am getting the following error (Mac console) when attempting to load it: dlsym cannot find symbol g_dwILResult in CFBundle etc.. I used Xcode 16.4 to build the plugin. Has anybody come across the same or a similar message? Best, Achillefs Axart Labs

Hi! I have a music app using AVAudioEngine. Right now, I have set it up to play multi channel tracks and show "Multichannel" in the volume controls. However, I am unable to figure out how to get it to use Dolby Atmos. Is there something that needs to be enabled? Is it even possible for AVAudioEngine? I saw some apps that are able of playing with Dolby Atmos, but they do not have EQ feature, so I'm guessing that they are not using AVAudioEngine.

Hi! I have an AVAudioSequencer with some AVMusicTracks that are filled with AVParameterEvents. If I toggle the isMuted property of a track, it will instantly mute when changed to true. However, after turning the muting to false, the events will only triggers on the next round of a loop and not instantly. Is this intended behaviour, and is there some way to get the events to trigger immediately after toggling the isMuted to be false?

In MusicKit Web the playback states are provided as numbers. For example the playbackStateDidChange event listener will return: {oldState: 2, state: 3, item:...} When the state changes from playing (2) to paused (3). Those are pretty easy to guess, but I'm having a hard time with some of the others: completed, ended, loading, none, paused, playing, seeking, stalled, stopped, waiting. I cannot find a mapping of states to numbers documented anywhere. I got the above states from an enum in a d.ts file that is often incorrect/incomplete. Can someone help out pointing to the docs or provide a mapping? Thanks.

Hello, Starting in iOS 17, our application started having some issue publishing to our video session. More specifically the video capture seems to be broken in some, but not all sessions. What's troubling is that we're seeing that it fails consistently every 4 sessions. It also fails silently, without reporting any problems to the app. We only notice that there are no frames being rendered or sent to the remote devices. Here's what shows-up in the console: <<<< FigCaptureSourceRemote >>>> Fig assert: "! storage->connectionDied" at bail (FigCaptureSourceRemote.m:235) - (err=0) <<<< FigCaptureSourceRemote >>>> Fig assert: "err == 0 " at bail (FigCaptureSourceRemote.m:253) - (err=-16453) Anyone seeing this? Any idea what could be the cause? Our sessions work perfectly on iOS16 and below. Thanks

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?

I’m using the shared instance of AVAudioSession. After activating it with .setActive(true), I observe the outputVolume, and it correctly reports the device’s volume. However, after deactivating the session using .setActive(false), changing the volume, and then reactivating it again, the outputVolume returns the previous volume (before deactivation), not the current device volume. The correct volume is only reported after the user manually changes it again using physical buttons or Control Center, which triggers the observer. What I need is a way to retrieve the actual current device volume immediately after reactivating the audio session, even on the second and subsequent activations. Disabling and re-enabling the audio session is essential to how my application functions. I’ve tested this behavior with my colleagues, and the issue is consistently reproducible on iOS 18.0.1, iOS 18.1, iOS 18.3, iOS 18.5 and iOS 18.6.2. On devices running iOS 17.6.1 and iOS 16.0.3, outputVolume correctly reflects the current volume immediately after calling .setActive(true) multiple times.

Hi, I'm facing an issuer with audio worklet in safari. This issue is clearly an iOS bug (it doesn't occur on iPad or Mac) Here's the minimal reproduction: Go to https://googlechromelabs.github.io/web-audio-samples/audio-worklet/basic/hello-audio-worklet/ Press start Audio will not be playing Open YouTube on another tab and start any video Audio from the worklet will start playing Is this a known issue? Any plans to address that? Any workaround available?

I donate some INPlayMediaIntent to system, and I find them in Control center, when I click one of them to play media background, the handler don't execute resolve method, I wanna resolve some mediaItems for suggestion playlist