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!

Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

Hi All, I am currently working on an app that has some navigation functionality, and since my minimum iOS is 18 wanted to incorporate the new APIs that yield a AsyncStream of locations. I have watched both WWDC sessions, the one where the new API is introduced to retrieve the location points, and also the other video where the new authorization process for location is simplified as well. I have an app currently working in its current state, but am noticing some weird quirks when using the CLBackgroundActivitySession to get the elevated background permission. What I am doing here is to create this stream and the background object is below: return AsyncThrowingStream { continuation in let task = Task { do { for try await update in CLLocationUpdate.liveUpdates(updateType) { if shouldStopUpdate { continuation.finish() break } continuation.yield(update) } } catch { continuation.finish(throwing: error) } } state = .started(locationTask: task, background: CLBackgroundActivitySession()) } When I have an active navigation session going and am strongly holding this object and the user force quits the app (or I stop the target through Xcode) the navigation activity indicator in the status bar (or dynamic island) remains present. Even if I relaunch the app, start navigation again, and then call the invalidate method on the CLBackgroundActivitySession I then am seeing that navigation indicator even if I delete my app, and often need to do a full restart to get out of this state. Is there a step I am missing, or do I not understand the way the new API works to run in the background?

Hi all, I have spent a lot of time reading the tech note and watching the WDDC video that introduce the PTTFramework on iOS. I currently have a custom setup where I am using AVAudioEngine to schedule and play buffers that are being streamed through a call. I am looking to use the PTTFramework to allow a user to trigger this push to talk behavior from the lock screen and the various places with the system UI it provides. However I am unsure what the correct behavior is regarding the handling of the audio session. Right now I am using .playback when there is no active voice transmission so that devices such as AirPods can be in AD2P mode where applicable, and then transitioning to .playbackAndRecord category only when the mic input should become active. Following this change in my AVAudioEngine manager I am then manually activating and deactivating the audio session manually when the engine is either playing/recording or idle. In the documentation it states that you should not attempt to activate or deactivate your audio session directly, but allow the framework to handle it. Does that mean that I need to either call the request to transmit delegate function or set an active participant on the channel manager first, and then wait for the didBecomeActive delegate method to trigger before I actually attempt to play or record any audio? (I am using the fullDuplex mode currently.) I noticed that that delegate method will only trigger if the audio session wasn't active before doing one of the above (setting active participant, requesting transmit). Lastly, when using the PTTFramework it also mentions that we get support for PTT devices and I notice on the didBeginTransmittingFrom property we have a handsfreeButton case. Is there any documentation or resources for what is actually supported out of the box for this? I am currently working on handling a lot of the push to talk through bluetooth LE, and wanted to make sure there wasn't overlap with what the system provides. Thank you!

Hi all, I am working on an app where I have live prompts playing, in addition to a voice channel that sometimes becomes active. Right now I am using two different AVAudioSession Configurations so what we only switch to a mic enabled mode when we actually need input from the mic. These are defined below. When just using the device hardware, everything works as expected and the modes change and the playback continues as needed. However when using bluetooth devices such as AirPods where the switch from AD2P to HFP is needed, I am getting a AVAudioEngineConfigurationChange notification. In response I am tearing down the engine and creating a new one with the same 2 player nodes. This does work fine and there are no crashes, except all the audio I have scheduled on a player node has now been cleared. All the completion blocks marked with ".dataPlayedBack" return the second this event happens, and leaves me in a state where I now have a valid engine setup again but have no idea what actually played, or was errantly marked as such. Is this the expected behavior when getting a configuration change notification? Adding some information below to my audio graph for context: All my parts of the graph, I disconnect when getting this event and do the same to the new engine private var inputEngine: AVAudioEngine private var audioEngine: AVAudioEngine private let voicePlayerNode: AVAudioPlayerNode private let promptPlayerNode: AVAudioPlayerNode audioEngine.attach(voicePlayerNode) audioEngine.attach(promptPlayerNode) audioEngine.connect( voicePlayerNode, to: audioEngine.mainMixerNode, format: voiceNodeFormat ) audioEngine.connect( promptPlayerNode, to: audioEngine.mainMixerNode, format: nil ) An example of how I am scheduling playback, and where that completion is firing even if it didn't actually play. private func scheduleVoicePlayback(_ id: AudioPlaybackSample.Id, buffer: AVAudioPCMBuffer) async throws { guard !voicePlayerQueue.samples.contains(where: { $0 == id }) else { return } seprateQueue.append(buffer) if !isVoicePlaying { activateAudioSession() } voicePlayerQueue.samples.append(id) if !voicePlayerNode.isPlaying { voicePlayerNode.play() } if let convertedBuffer = buffer.convert(to: voiceNodeFormat) { await voicePlayerNode.scheduleBuffer(convertedBuffer, completionCallbackType: .dataPlayedBack) } else { throw AudioPlaybackError.failedToConvert } voiceSampleHasBeenPlayed(id) } And lastly my audio session configuration if its useful. extension AVAudioSession { static func setDefaultCategory() { do { try sharedInstance().setCategory( .playback, options: [ .duckOthers, .interruptSpokenAudioAndMixWithOthers ] ) } catch { print("Failed to set default category? \(error.localizedDescription)") } } static func setVoiceChatCategory() { do { try sharedInstance().setCategory( .playAndRecord, options: [ .defaultToSpeaker, .allowBluetooth, .allowBluetoothA2DP, .duckOthers, .interruptSpokenAudioAndMixWithOthers ] ) } catch { print("Failed to set category? \(error.localizedDescription)") } } }