Hello there! Is there any list of voices that are always available on iOS/iPadOS devices? It seems that AVSpeechSynthesisVoice(identifier: "com.apple.voice.compact.en-US.Samantha") is always available on all devices. I thought that AVSpeechSynthesisVoice(identifier: "com.apple.ttsbundle.siri_Nicky_en-US_compact") and AVSpeechSynthesisVoice(identifier: "com.apple.ttsbundle.siri_Aaron_en-US_compact") were available by default on certain newer devices. Is this true? I also noticed that on the same iPad where I was using those 2 voices (Nicky and Aaron) - when I updated to the iPadOS 26 beta, those voices were no longer available. Any information you can share about which voices should be reliably available on which devices would be extremely helpful for our development. Thanks so much!

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

Is it possible to play WebM audio on iOS? Either with AVPlayer, AVAudioEngine, or some other API? Safari has supported this for a few releases now, and I'm wondering if I missed something about how to do this. By default these APIs don't seem to work (nor does ExtAudioFileOpen). Our usecase is making it possible for iOS users to play back audio recorded in our webapp (desktop versions of Chrome & Firefox only support webm as a destination format for MediaRecorder)

Hi all, I've developed an audio DSP application in C++ using AudioToolbox and CoreAudio on MacOS 14.4.1 with Xcode 15. I use an AudioQueue for input and another for output. This works great. I'm now adding real-time audio analysis eg spectral analysis. I want this to run independently of my audio processing so it can not interfere with audio playback. Taps on AudioQueues seem to be a good way of doing this... Since the analytics won't modify the audio data, I am using a Siphon Tap by setting the AudioQueueProcessingTapFlags to kAudioQueueProcessingTap_PreEffects | kAudioQueueProcessingTap_Siphon; This works fine on my output queue. However, on my input queue the Tap callback is called once and then a EXC_BAD_ACCESS occurs - screen shot below. NB: I believe that a callback should only call AudioQueueProcessingTapGetSourceAudio when not using a Siphon, so I don't call it. Relevant code: AudioQueueProcessingTapCallback tap_callback) { // Makes an audio tap for a queue void * tap_data_ptr = NULL; AudioQueueProcessingTapFlags tap_flags = kAudioQueueProcessingTap_PostEffects | kAudioQueueProcessingTap_Siphon; uint32_t max_frames = 0; AudioStreamBasicDescription asbd; AudioQueueProcessingTapRef tap_ref; OSStatus status = AudioQueueProcessingTapNew(queue_ref, tap_callback, tap_data_ptr, tap_flags, &max_frames, &asbd, &tap_ref); if (status != noErr) printf("Error while making Tap\n"); else printf("Successfully made tap\n"); } void tapper(void * tap_data, AudioQueueProcessingTapRef tap_ref, uint32_t number_of_frames_in, AudioTimeStamp * ts_ptr, AudioQueueProcessingTapFlags * tap_flags_ptr, uint32_t * number_of_frames_out_ptr, AudioBufferList * buf_list) { // Callback function for audio queue tap printf("Tap callback"); }``` Image of exception stack provided by Xcode: ![]("https://developer.apple.com/forums/content/attachment/27479e8d-a118-459b-aa2d-7e30528910e3" "title=Screenshot 2025-06-14 at 1.29.14 PM.png;width=932;height=562") What have I missed? Appreciate any help you learned folks may be able to provide. Best, Geoff.

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

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.

I have tried everything. The songs load unto the playlists and on searches, but when prompted to play, they just won't play. I have a wrapper since my main player (which carries the buttons for play/rewind/forward/etc.), is in Objc. // // ApplePlayerWrapper.swift // UniversallyMac // // Created by Dorian Mattar on 11/10/24. // import Foundation import MusicKit import MediaPlayer @objc public class MusicKitWrapper: NSObject { @objc public static let shared = MusicKitWrapper() private let player = ApplicationMusicPlayer.shared // Play the current track @objc public func play() { guard !player.queue.entries.isEmpty else { print("Queue is empty. Cannot start playback.") return } logPlayerState(message: "Before play") Task { do { try await player.prepareToPlay() try await player.play() print("Playback started successfully.") } catch { if let nsError = error as NSError? { print("NSError Code: \(nsError.code), Domain: \(nsError.domain)") } } logPlayerState(message: "After play") } } // Log the current player state @objc public func logPlayerState(message: String = "") { print("Player State - \(message):") print("Playback Status: \(player.state.playbackStatus)") print("Queue Count: \(player.queue.entries.count)") // Only log current track details if the player is playing if player.state.playbackStatus == .playing { if let currentEntry = player.queue.currentEntry { print("Current Track: \(currentEntry.title)") print("Current Position: \(player.playbackTime) seconds") print("Track Length: \(currentEntry.endTime ?? 0.0) seconds") } else { print("No current track.") } } else { print("No track is playing.") } print("----------") } // Debug the queue @objc public func debugQueue() { print("Debugging Queue:") for (index, entry) in player.queue.entries.enumerated() { print("\(index): \(entry.title)") } } // Ensure track availability in the queue public func queueTracks(_ tracks: [Track]) { Task { do { for track in tracks { // Validate Play Parameters guard let playParameters = track.playParameters else { print("Track \(track.title) has no Play Parameters.") continue } // Log the Play Parameters print("Track Title: \(track.title)") print("Play Parameters: \(playParameters)") print("Raw Values: \(track.id.rawValue)") // Ensure the ID is valid if track.id.rawValue.isEmpty { print("Track \(track.title) has an invalid or empty ID in Play Parameters.") continue } // Queue the track try await player.queue.insert(track, position: .afterCurrentEntry) print("Queued track: \(track.title)") } print("Tracks successfully added to the queue.") } catch { print("Error queuing tracks: \(error)") } debugQueue() } } // Clear the current queue @objc public func resetMusicPlayer() { Task { player.stop() player.queue.entries.removeAll() print("Queue cleared.") print("Apple Music player reset successfully.") } } } I opened an Apple Dev. ticket, but I'm trying here as well. Thanks!

Hi guys, I am having issue in live-streaming audio from Bluetooth headset and playing it live on the iPhone speaker. I am able to redirect audio back to the headset but this is not what I want. The issue happens when I am trying to override output - the iPhone switches to speaker but also switches a microphone. This is example of the code: import AVFoundation class AudioRecorder { let player: AVAudioPlayerNode let engine:AVAudioEngine let audioSession:AVAudioSession let audioSessionOutput:AVAudioSession init() { self.player = AVAudioPlayerNode() self.engine = AVAudioEngine() self.audioSession = AVAudioSession.sharedInstance() self.audioSessionOutput = AVAudioSession() do { try self.audioSession.setCategory(AVAudioSession.Category.playAndRecord, options: [.defaultToSpeaker]) try self.audioSessionOutput.setCategory(AVAudioSession.Category.playAndRecord, options: [.allowBluetooth]) // enables Bluetooth HFP profile try self.audioSession.setMode(AVAudioSession.Mode.default) try self.audioSession.setActive(true) // try self.audioSession.overrideOutputAudioPort(.speaker) // doens't work } catch { print(error) } let input = self.engine.inputNode self.engine.attach(self.player) let bus = 0 let inputFormat = input.inputFormat(forBus: bus) self.engine.connect(self.player, to: engine.mainMixerNode, format: inputFormat) input.installTap(onBus: bus, bufferSize: 512, format: inputFormat) { (buffer, time) -> Void in self.player.scheduleBuffer(buffer) print(buffer) } } public func start() { try! self.engine.start() self.player.play() } public func stop() { self.player.stop() self.engine.stop() } } I am not sure if this is a bug or not. Can somebody point me into the right direction? I there a way to design a custom audio routing? I would also appreciate some good documentation besides AVFoundation docs.

Good day, ladies and gents. I have an application that reads audio from the microphone. I'd like it to also be able to read from the Mac's audio output stream. (A bonus would be if it could detect when the Mac is playing music.) I'd eventually be able to figure it out reading docs, but if someone can give a hint, I'd be very grateful, and would owe you the libation of your choice. Here's the code used to set up the AudioUnit: -(NSString*) configureAU { AudioComponent component = NULL; AudioComponentDescription description; OSStatus err = noErr; UInt32 param; AURenderCallbackStruct callback; if( audioUnit ) { AudioComponentInstanceDispose( audioUnit ); audioUnit = NULL; } // was CloseComponent // Open the AudioOutputUnit description.componentType = kAudioUnitType_Output; description.componentSubType = kAudioUnitSubType_HALOutput; description.componentManufacturer = kAudioUnitManufacturer_Apple; description.componentFlags = 0; description.componentFlagsMask = 0; if( component = AudioComponentFindNext( NULL, &description ) ) { err = AudioComponentInstanceNew( component, &audioUnit ); if( err != noErr ) { audioUnit = NULL; return [ NSString stringWithFormat: @"Couldn't open AudioUnit component (ID=%d)", err] ; } } // Configure the AudioOutputUnit: // You must enable the Audio Unit (AUHAL) for input and output for the same device. // When using AudioUnitSetProperty the 4th parameter in the method refers to an AudioUnitElement. // When using an AudioOutputUnit for input the element will be '1' and the output element will be '0'. param = 1; // Enable input on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32) ); chkerr("Couldn't set first EnableIO prop (enable inpjt) (ID=%d)"); param = 0; // Disable output on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32) ); chkerr("Couldn't set second EnableIO property on the audio unit (disable ootpjt) (ID=%d)"); param = sizeof(AudioDeviceID); // Select the default input device AudioObjectPropertyAddress OutputAddr = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; err = AudioObjectGetPropertyData( kAudioObjectSystemObject, &OutputAddr, 0, NULL, &param, &inputDeviceID ); chkerr("Couldn't get default input device (ID=%d)"); // Set the current device to the default input unit err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDeviceID, sizeof(AudioDeviceID) ); chkerr("Failed to hook up input device to our AudioUnit (ID=%d)"); callback.inputProc = AudioInputProc; // Setup render callback, to be called when the AUHAL has input data callback.inputProcRefCon = self; err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); param = sizeof(AudioStreamBasicDescription); // get hardware device format err = AudioUnitGetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &deviceFormat, &param ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); audioChannels = MAX( deviceFormat.mChannelsPerFrame, 2 ); // Twiddle the format to our liking actualOutputFormat.mChannelsPerFrame = audioChannels; actualOutputFormat.mSampleRate = deviceFormat.mSampleRate; actualOutputFormat.mFormatID = kAudioFormatLinearPCM; actualOutputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; if( actualOutputFormat.mFormatID == kAudioFormatLinearPCM && audioChannels == 1 ) actualOutputFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; #if __BIG_ENDIAN__ actualOutputFormat.mFormatFlags |= kAudioFormatFlagIsBigEndian; #endif actualOutputFormat.mBitsPerChannel = sizeof(Float32) * 8; actualOutputFormat.mBytesPerFrame = actualOutputFormat.mBitsPerChannel / 8; actualOutputFormat.mFramesPerPacket = 1; actualOutputFormat.mBytesPerPacket = actualOutputFormat.mBytesPerFrame; // Set the AudioOutputUnit output data format err = AudioUnitSetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &actualOutputFormat, sizeof(AudioStreamBasicDescription)); chkerr("Could not change the stream format of the output device (ID=%d)"); param = sizeof(UInt32); // Get the number of frames in the IO buffer(s) err = AudioUnitGetProperty( audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &audioSamples, &param ); chkerr("Could not determine audio sample size (ID=%d)"); err = AudioUnitInitialize( audioUnit ); // Initialize the AU chkerr("Could not initialize the AudioUnit (ID=%d)"); // Allocate our audio buffers audioBuffer = [self allocateAudioBufferListWithNumChannels: actualOutputFormat.mChannelsPerFrame size: audioSamples * actualOutputFormat.mBytesPerFrame]; if( audioBuffer == NULL ) { [ self cleanUp ]; return [NSString stringWithFormat: @"Could not allocate buffers for recording (ID=%d)", err]; } return nil; } (...again, it would be nice to know if audio output is active and thereby choose the clean output stream over the noisy mic, but that would be a different chunk of code, and my main question may just be a quick edit to this chunk.) Thanks for your attention! ==Dave [p.s. if i get more than one useful answer, can i "Accept" more than one, to spread the credit around?] {pps: of course, the code lines up prettier in a monospaced font!}

My current app implements a custom video player, based on a AVSampleBufferRenderSynchronizer synchronising two renderers: an AVSampleBufferDisplayLayer receiving decoded CVPixelBuffer-based video CMSampleBuffers, and an AVSampleBufferAudioRenderer receiving decoded lpcm-based audio CMSampleBuffers. The AVSampleBufferRenderSynchronizer is started when the first image (in presentation order) is decoded and enqueued, using avSynchronizer.setRate(_ rate: Float, time: CMTime), with rate = 1 and time the presentation timestamp of the first decoded image. Presentation timestamps of video and audio sample buffers are consistent, and on most streams, the audio and video are correctly synchronized. However on some network streams, on iOS, the audio and video aren't synchronized, with a time difference that seems to increase with time. On the other hand, with the same player code and network streams on macOS, the synchronization always works fine. This reminds me of something I've read, about cases where an AVSampleBufferRenderSynchronizer could not synchronize audio and video, causing them to run with independent and potentially drifting clocks, but I cannot find it again. So, any help / hints on this sync problem will be greatly appreciated! :)

We have the necessary background recording entitlements, and for many users... do not run into any issues. However, there is a subset of users that routinely get recordings ending.. we have narrowed this down and believe it to be the work of the watch dog. First we removed the entire view hierarchy when app is backgrounded. There is just 'Text("Recording")' This got the CPU usage in profiler down to 0%. We saw massive improvements to recording success rate. We walked away assuming that was enough. However we are still seeing the same sort of crashes. All in the background. We're using Observation to drive audio state changes to a Live Activity. Are those Observations causing the problem? Why doesn't apple provide a better API to background audio? The internet is full of weird issues https://stackoverflow.com/questions/76010213/why-is-my-react-native-app-sometimes-terminated-in-the-background-while-tracking https://stackoverflow.com/questions/71656047/why-is-my-react-native-app-terminating-in-the-background-while-recording-ios-r https://github.com/expo/expo/issues/16807 This is such a terrible user experience. And we have very little visibility into what is happening and why. No where in apple documentation states that in order for background recording to work, the app can only be 'Text("Recording")' It does not outline a CPU or memory threshold. It just kills us.

Is it possible to find IDR frame (CMSampleBuffer) in AVAsset h264 video file?

I am using an AVAudioPlayer to play a "tick" sound once per second in a SwiftUI app. When running the app on an iPhone 16 (18.2.1) the tick sounds increase in volume after a few seconds. This does not happen in the simulator nor on an iPhone SE 2020 (18.1.1).

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, I'm trying to plan out development of an app and am wondering if it is possible to have user generated content automatically populate into a custom shazamkit catalogue and be able to query this catalogue non-locally? Storing all the submissions locally would obviously not scale.

Mobile app - Ellie's Gift https://apps.apple.com/gb/app/ellies-gift/id1617597875 Using AVFoundation to play audio tracks within the app. Has always been working fine across apple and android, but iphone 14 and newer devices are unable to play audio. Any idea's or suggestions?

I've been trying to use AVMIDIControlChangeEvent with a bankSelect message type to change the instrument the sequencer uses on a AVMusicTrack with no luck. I started with the Apple AVAEMixerSample, converting the initial setup/loading and portions dealing with the sequencer to Swift. I got that working and playing the "bluesyRiff" and then modified it to play individual notes. So my createAndSetupSequencer looked like func createAndSetupSequencer() { sequencer = AVAudioSequencer(audioEngine: engine) // guard let midiFileURL = Bundle.main.url(forResource: "bluesyRiff", withExtension: "mid") else { // print (" failed guard trying to get URL for bluesyRiff") // return // } let track = sequencer.createAndAppendTrack() var currTime = 1.0 for i: UInt32 in 0...8 { let newNoteEvent = AVMIDINoteEvent(channel: 0, key: 60+i, velocity: 64, duration: 2.0) track.addEvent(newNoteEvent, at: AVMusicTimeStamp(currTime)) currTime += 2.0 } The notes played, so then I also replaced the gs_instruments sound bank with GeneralUser GS MuseScore v1.442 first by trying guard let soundBankURL = Bundle.main.url(forResource: "GeneralUser GS MuseScore v1.442", withExtension: "sf2") else { return} do { try sampler.loadSoundBankInstrument(at: soundBankURL, program: 0x001C, bankMSB: 0x79, bankLSB: 0x08) } catch{.... } This appears to work, the instrument (8 which is "Funk Guitar") plays. If I change to bankLSB: 0x00 I get the "Palm Muted guitar". So I know that the soundfont has these instruments Stuff goes off the rails when I try to change the instruments in createAndSetupSequencer. Putting let programChange = AVMIDIProgramChangeEvent(channel: 0, programNumber: 0x001C) let bankChange = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.bankSelect, value: 0x00) track.addEvent(programChange, at: AVMusicTimeStamp(1.0)) track.addEvent(bankChange, at: AVMusicTimeStamp(1.0)) just before my add note loop doesn't produce any change. Loading bankLSB 8 (Funk) in sampler.loadSoundBankInstrument and trying to change with bankSelect 0 (Palm muted) in createAndSetupSequencer results in instrument 8 (Funk) playing not Palm Muted. Loading bankLSB 0 (Palm muted) and trying to change with bankSelect 8 (Funk) doesn't work, 0 (Palm muted) plays I also tried sampler.loadInstrument(at: soundBankURL) and then I always get the first instrument in the sound font file (piano)no matter what values I put in my programChange/bankChange I've also changed the time in the track.addEvent to be 0, 1.0, 3.0 etc to no success The sampler.loadSoundBankInstrument specifies two UInt8 parameters, bankMSB and BankLSB while the AVMIDIControlChangeEvent bankSelect value is UInt32 suggesting it might be some combination of bankMSB and BankLSB. But the documentation makes no mention of what this should look like. I tried various combinations of 0x7908, 0X0879 etc to no avail I will also point out that I am able to successfully execute other control change events For example adding if i == 1 { let portamentoOnEvent = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.portamento, value: 0xFF) track.addEvent(portamentoOnEvent, at: AVMusicTimeStamp(currTime)) let portamentoRateEvent = AVMIDIControlChangeEvent(channel: 0, messageType: AVMIDIControlChangeEvent.MessageType.portamentoTime, value: 64) track.addEvent(portamentoRateEvent, at: AVMusicTimeStamp(currTime)) } does produce a change in the sound. (As an aside, a definition of what portamento time is, other than "the rate of portamento" would be welcome. is it notes/seconds? freq/minute? beats/hour?) I was able to get the instrument to change in a different program using MusicPlayer and a series of MusicTrackNewMIDIChannelEvent on a track but these operate on a MusicTrack not the AVMusicTrack which the sequencer uses. Has anyone been successful in switching instruments through an AVMIDIControlChangeEvent or have any feedback on how to do this?

Since the last update to IOS 26.0 (23A5276f) the AirPods connect to my IPhone and the Audio is still running through the phone. They are shown in the Bluetooth Icon that they’re paired.

Hi, Not sure if this is the right forum to ask this question in, but could you please advise if I can use Apple Digital Masters logo (badge) in my iOS app that is playing music from Apple Music service?

Is there a way to permanently disable PHASE SDK logging? It seems to be a lot chattier than Apple's other SDKs. While developing a RealityKit app that uses AudioPlaybackController, I must manually hide the PHASE SDK log output several times each day so I can see my app's log messages. Thank you.