I’m working on an iOS app for a client, and I have a question regarding a specific feature we're looking to implement. We want the app to respond to a user pressing the volume button three times while the app is in the background. The goal is to allow users to discreetly trigger a safety feature without drawing attention, particularly in situations where they may be in danger or at risk. This feature is critical for the app and would be a valuable addition, as it could potentially help protect users in emergency situations. However, I haven’t found much information on whether iOS allows background listening for volume button presses. Therefore, I would greatly appreciate your insights on the following: Is it possible to listen for volume button presses when the app is in the background, or are there system-level restrictions that prevent this? If it's not directly possible, are there any special provisions, APIs, or entitlements that can be requested from Apple to enable this functionality? In case this feature is not supported, are there alternative approaches to achieve a similar discreet activation mechanism? If this is something that requires special permission or a process, could you please guide me on how to proceed? I understand that maintaining user privacy and security is a priority for iOS, and I want to ensure that any implementation fully complies with Apple's guidelines. Thanks in advance for your help!

I have used AVQueuePlayer in my music app to play sequence of audios from a remote server, this how I have defined things my player in my ViewModel Variables private var cancellables = Set() private let audioSession = AVAudioSession.sharedInstance() private var avQueuePlayer: AVQueuePlayer? @Published var playbackSpeed: Float = 1.0 before starting playback, I am making sure that audio session is set properly, the code snippet used for that is do { try audioSession.setCategory(.playback, mode: .default, options: []) try audioSession.setActive(true, options: []) } catch { return } and this is the function I am using to update playback speed func updatePlaybackSpeed(_ newSpeed: Float){ if newSpeed > 0.0, newSpeed <= 2.0{ playbackSpeed = newSpeed avQueuePlayer?.rate = newSpeed print("requested speed is (newSpeed) and actual speed is (String(describing: avQueuePlayer?.rate))") } } sometimes whatever speed is set, player seems to play at the same speed as it was set, e.g. Once I got "requested speed is 1.5 and actual speed is 1.5", and player also seemed to play at the speed of 1.5 but another time I got "requested speed is 2.0 and actual speed is 2.0", but player still seemed to play at the speed of 1.0 to observe changes in rate, I used this **private func observeRateChanges() { guard let avQueuePlayer = self.avQueuePlayer else { return } NotificationCenter.default.publisher(for: AVQueuePlayer.rateDidChangeNotification, object: avQueuePlayer) .compactMap { $0.userInfo?[AVPlayer.rateDidChangeReasonKey] as? AVPlayer.RateDidChangeReason } .sink { reason in switch reason { case .appBackgrounded: print("The app transitioned to the background.") case .audioSessionInterrupted: print("The system interrupts the app’s audio session.") case .setRateCalled: print("The app set the player’s rate.") case .setRateFailed: print("An attempt to change the player’s rate failed.") default: break } } .store(in: &cancellables) }** when rate was set properly, I got this "The app set the player’s rate." from the above function, but when it wasn't, I got this "An attempt to change the player’s rate failed.," now I am not able to understand why rate is not being set, and if it gave "requested speed is 2.0 and actual speed is 2.0" from updatePlaybackSpeed function, why does the player seems to play with the speed of 1.0?

Currently we tested iOS AAC LC encoder using AudioToolbox framework, no matter we set mManufacturer to kAppleHardwareAudioCodecManufacturer or kAppleSoftwareAudioCodecManufacturer, it always run on CPU.

I’ve been researching how to achieve a recording playback effect in iOS similar to the hands-free calling effect in the system’s phone app. How can this be implemented? I tried using the voice chat recording method, but found that the volume of the speaker output is too low. How should this issue be addressed? I couldn’t find a suitable API. Could you provide me with some documentation or sample code? Thank you.

Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes

Hi. I am working on an audio app for iOS. I have added the CPNowPlayingPlaybackRateButton to my CPNowPlayingTemplate. When the button is clicked, my handler changes the rate in the AVPlayer and updates the MPNowPlayingInfoCenter to the new rate, for example, 2.0. Throughout, the Carplay button always displays "0x". I am wondering how to get this UI to accurately reflect the playback rate the user has selected, as always displaying 0x is a poor user experience. You may suggest MPChangePlaybackRateCommand is relevant here, but I have not been able to get that to work either, and judging by posts online, not many other people have either. I have made a post about that here: https://developer.apple.com/forums/thread/773099 Is this a known Apple bug? Is there a way to get the UI to accurately reflect the playback rate of my audio? Kind regards.

Issue Description When playing certain MIDI files using AVMIDIPlayer, the initial volume settings for individual tracks are being ignored during the first playback. This results in all tracks playing at the same volume level, regardless of their specified volume settings in the MIDI file. Steps to Reproduce Load a MIDI file that contains different volume settings for multiple tracks Start playback using AVMIDIPlayer Observe that all tracks play at the same volume level, ignoring their individual volume settings Current Behavior All tracks play at the same volume level during initial playback Track volume settings specified in the MIDI file are not being respected This behavior consistently occurs on first playback of affected MIDI files Expected Behavior Each track should play at its specified volume level from the beginning Volume settings in the MIDI file should be respected from the first playback Workaround I discovered that the correct volume settings can be restored by: Starting playback of the MIDI file Setting the currentPosition property to (current time - 1 second) After this operation, all tracks play at their intended volume levels However, this is not an ideal solution as it requires manual intervention and may affect the playback experience. Questions Is there a way to ensure the track volume settings are respected during the initial playback? Is this a known issue with AVMIDIPlayer? Are there any configuration settings or alternative approaches that could resolve this issue? Technical Details iOS Version: 18.1.1 (22B91) Xcode Version: 16.1 (16B40)

For an upcoming update of one of my apps, I’m facing an issue: The .rate parameter of a AVAudioUnitTimePitch allows me to slow down an audio track without any issues: setting .rate to 0.7 or 0.8 results in an almost perfect playback without changing pitch. However, whenever the .rate parameter is greater than 1 (e.g. 1.1 or 1.15), I’m starting to hear audio artifacts (“flattering”) in the audio output which is not so nice (even at .overlap = 32). Intuitively, I’d’ve thought that speeding up the file should contain less artifacts than slowing it down?? I’ve tried different sample rates (44.1 kHz and 48 kHz), but same result. Grateful for any input on this 🙏

I bought two "Apple USB-C to Headphone Jack Adapters". Upon closer inspection, they seems to be of different generations: The one with product ID 0x110a on top is working fine. The one with product ID 0x110b has two issues: There is a short but loud click noise on the headphone when I connect it to the iPad. When I play audio using AVAudioPlayer the first half of a second or so is cut off. Here's how I'm playing the audio: audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.delegate = self audioPlayer?.prepareToPlay() audioPlayer?.play() Is this a known issue? Am I doing something wrong?

Hi everyone, I just upgraded my iPhone to 18.1.1. I noticed the absence of the normal mode on my AirPods. I can't stand the noise cancellation mode for too long and the transparency one is overwhelming everytime a hair is brushing near by my AirPods. Why is that? I don't really see the progression here. Can you this be fixed in the next version? I don't want to buy another brand, I use them everyday for many hours.

I'm using a 4 channel USB Audio interface, with 4 microphones, and want to process them through 4 independent effect chains. However the output from AVAudioInputNode is a single 4 channel bus. How can I split this into 4 mono busses? The following code splits the input into 4 copies, and routes them through the effects, but each bus contains all four channels. How can I remap the channels to remove the unwanted channels from the bus? I tried using channelMap on the mixer node but that had no effect. I'm currently using this code primarily on iOS but it should be portable between iOS and MacOS. It would be possible to do this through a Matrix Mixer Node, but that seems completely overkill, for such a basic operation. I'm already using a Matrix Mixer to combine the inputs, and it's not well supported in AVAudioEngine. AVAudioInputNode *inputNode=[engine inputNode]; [inputNode setVoiceProcessingEnabled:NO error:nil]; NSMutableArray *micDestinations=[NSMutableArray arrayWithCapacity:trackCount]; for(i=0;i<trackCount;i++) { fixMicFormat[i]=[AVAudioMixerNode new]; [engine attachNode:fixMicFormat[i]]; // And create reverb/compressor and eq the same way... [engine connect:reverb[i] to:matrixMixerNode fromBus:0 toBus:i format:nil]; [engine connect:eq[i] to:reverb[i] fromBus:0 toBus:0 format:nil]; [engine connect:compressor[i] to:eq[i] fromBus:0 toBus:0 format:nil]; [engine connect:fixMicFormat[i] to:compressor[i] fromBus:0 toBus:0 format:nil]; [micDestinations addObject:[[AVAudioConnectionPoint alloc] initWithNode:fixMicFormat[i] bus:0] ]; } AVAudioFormat *inputFormat = [inputNode outputFormatForBus: 1]; [engine connect:inputNode toConnectionPoints:micDestinations fromBus:1 format:inputFormat];

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) } }

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?

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 everyone, I’m new to Swift development and have been working on an audio module that plays a specific sound at regular intervals - similar to a workout timer that signals switching exercises every few minutes. Following AVFoundation documentation, I’m configuring my audio session like this: let session = AVAudioSession.sharedInstance() try session.setCategory( .playback, mode: .default, options: [.interruptSpokenAudioAndMixWithOthers, .duckOthers] ) self.engine.attach(self.player) self.engine.connect(self.player, to: self.engine.outputNode, format: self.audioFormat) try? session.setActive(true) When it’s time to play cues, I schedule playback on a DispatchQueue: // scheduleAudio uses DispatchQueue self.scheduleAudio(at: interval.start) { do { try audio.engine.start() audio.node.play() for sample in interval.samples { audio.node.scheduleBuffer(sample.buffer, at: AVAudioTime(hostTime: sample.hostTime)) } } catch { print("Audio activation failed: \(error)") } } This works perfectly in the foreground. But once the app goes into the background, the scheduled callback runs, yet the audio engine fails to start, resulting in an error with code 561015905. Interestingly, if the app is already playing audio before going to the background, the scheduled sounds continue to play as expected. I have added the required background audio mode to my Info plist file by including the key UIBackgroundModes with the value audio. Is there anything else I should configure? What is the best practice to play periodic audio when the app runs in the background? How do apps like turn-by-turn navigation handle continuous audio playback in the background? Any advice or pointers would be greatly appreciated!

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?

Hello! I'm experiencing an issue with iOS's audio routing system when trying to use Bluetooth headphones for audio output while also recording environmental audio from the built-in microphone. Desired behavior: Play audio through Bluetooth headset (AirPods) Record unprocessed environmental audio from the iPhone's built-in microphone Actual behavior: When explicitly selecting the built-in microphone, iOS reports it's using it (in currentRoute.inputs) However, the actual audio data received is clearly still coming from the AirPods microphone The audio is heavily processed with voice isolation/noise cancellation, removing environmental sounds Environment Details Device: iPhone 12 Pro Max iOS Version: 18.4.1 Hardware: AirPods Audio Framework: AVAudioEngine (also tried AudioQueue) Code Attempted I've tried multiple approaches to force the correct routing: func configureAudioSession() { let session = AVAudioSession.sharedInstance() // Configure to allow Bluetooth output but use built-in mic try? session.setCategory(.playAndRecord, options: [.allowBluetoothA2DP, .defaultToSpeaker]) try? session.setActive(true) // Explicitly select built-in microphone if let inputs = session.availableInputs, let builtInMic = inputs.first(where: { $0.portType == .builtInMic }) { try? session.setPreferredInput(builtInMic) print("Selected input: \(builtInMic.portName)") } // Log the current route let route = session.currentRoute print("Current input: \(route.inputs.first?.portName ?? "None")") // Configure audio engine with native format let inputNode = audioEngine.inputNode let nativeFormat = inputNode.inputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: nativeFormat) { buffer, time in // Process audio buffer // Despite showing "Built-in Microphone" in route, audio appears to be // coming from AirPods with voice isolation applied - welp! } try? audioEngine.start() } I've also tried various combinations of: Different audio session modes (.default, .measurement, .voiceChat) Different option combinations (with/without .allowBluetooth, .allowBluetoothA2DP) Setting session.setPreferredInput() both before and after activation Diagnostic Observations When AirPods are connected: AVAudioSession.currentRoute.inputs correctly shows "Built-in Microphone" after setPreferredInput() The actual audio data received shows clear signs of AirPods' voice isolation processing Background/environmental sounds are actively filtered out... When recording a test audio played near the phone (not through the app), the recording is nearly silent. Only headset voice goes through. Questions Is there a workaround to force iOS to actually use the built-in microphone while maintaining Bluetooth output? Are there any lower-level configurations that might resolve this issue? Any insights, workarounds, or suggestions would be greatly appreciated. This is blocking a critical feature in my application that requires environmental audio recording while providing audio feedback through headphones 😅

The device is connected to Bluetooth A and Bluetooth B, currently the audio is played through Bluetooth A, click the interface button, how to realize the code to switch to Bluetooth B？

Hi, May I ask if there is any iOS API or similar way to switch between the transparency and ANC modes of AirPods Pro 2? I know there is one way to configure and activate the shortcut in the APP, which requires an inconvenient manual setting. May I ask for any other advice? Thx in advance!

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