I'm trying to change the audio input (microphone) between all the available devices from AVAudioSession.sharedInstance().availableInputs. I'm using AVAudioSession.routeChangeNotification to get automatic route changes when devices get connected/disconnected and change the preferred input with setPreferredInput, then I restart my audioEngine and it works fine. But when I try to change the preferred input programmatically It doesn't change the audio capture inputNode. But keeps the last connected device and capturing. Even the AVAudioSession.sharedInstance().currentRoute.inputs changes but the audioEngine?.inputNode doesn't change to setPreferredInput call. WhatsApp seems to have done that without any issues. Any suggestions or leads are highly appreciated. Thanks.

I'm currently developing an app that requires detecting Bluetooth connections and disconnections in cars. During testing, I've observed the following behavior: In certain vehicles, only a Bluetooth connection via the car's hands-free system is available. In these cases, the device initiates a call to itself, which is then displayed on the vehicle's infotainment system. In some of the tested vehicles, this self-call is brief and only occurs during the device's connection or disconnection process. However, in other vehicles, the self-call remains visible throughout the entire duration of the device's pairing with the car's Bluetooth system. This blocked call blocks the entire infotainment system and causes the connection/disconnection observers in my app to stop functioning as expected. I'm looking for a solution or preventative measures to address this issue. Any guidance would be greatly appreciated. Here is a snippet of my code: `func audioSessionSetup() { do { resetAudioSession() let audioOptions: AVAudioSession.CategoryOptions = [.duckOthers, .allowBluetooth, .defaultToSpeaker] try audioSession.setCategory(.playAndRecord, mode: .spokenAudio, options: audioOptions) registerNotifications() try audioSession.setActive(true) print("audioSession is active") } catch let error as NSError { print("Failed to set the audio audioSession category and mode: \(error.localizedDescription)") } } /// Reset the audio session to deactivate it. func resetAudioSession() { do { try audioSession.setActive(false, options: .notifyOthersOnDeactivation) } catch let error as NSError { print("Failed to reset the audio audioSession, error: \(error.localizedDescription)") } } @objc func handleRouteChange(_ notification: Notification) { guard let userInfo = notification.userInfo, let reasonValue = userInfo[AVAudioSessionRouteChangeReasonKey] as? UInt, let reason = AVAudioSession.RouteChangeReason(rawValue: reasonValue) else { return } switch reason { case .newDeviceAvailable: /// Handle new device connection print("New device connected.") checkConnectionForSelectedOutput(notification) case .oldDeviceUnavailable: /// Handle device disconnection print("Device disconnected.") handleLocationServices(state: false) default: print("break") handleCategoryChange(notification) break } } private func handleCategoryChange(_ notification: Notification) { if let connectedDeviceName = getConnectedBluetoothDeviceName() { if connectedDeviceName != connectedDevice && connectedDeviceName == BluetoothUtils.getBluetoothInfo().portName { connectedDevice = connectedDeviceName checkConnectionForSelectedOutput(notification) } } else { audioSessionSetup() checkConnectionForSelectedOutput(notification) print("handleRouteChange audio session is active") } }`

Hi, I am developing a POC music player app. I use AVAudioSession; I have implemented background music and integration with command center. I am focusing now on volume. I am able to receive volume changes, with systemVolumeDidChange. About setting the volume, I am able to set it using MPVolumeView, but not for remote wifi audio device (for example, HomePods). I have the following open points: the native Podcast app is able to control volume when connected to HomePods. How does it do? the native Podcast app has icons for AirPods, HomePods, even Car bluetooth. Are there icon propeties for audioSession.currentRoute.outputs? Or what should I use instead? Here an example of what I would like to achieve:

When configuring an AVAudioSession as playAndRecord, I have to select the CarPlay input as preferredInput to make sure that the output is also routed to the car - if I set the preferredInput to the built-in mic, the output is routed to the speakers instead. However, when I select the CarPlay input as preferredInput, AVAudioSession configures the output as mono: (lldb) po session.currentRoute.inputs.first! <AVAudioSessionPortDescription: 0x282fcec30, type = CarAudio; name = CarPlay; UID = 48:F0:7B:C6:21:A8-Audio-AudioMain-92004763965583; selectedDataSource = (null)> (lldb) po session.currentRoute.inputs.first!.channels ▿ Optional<Array<AVAudioSessionChannelDescription>> ▿ some : 1 element - 0 : <AVAudioSessionChannelDescription: 0x282fccc70, name = CarPlay; label = 0 (0x0); number = 1; port UID = 48:F0:7B:C6:21:A8-Audio-AudioMain-92004763965583> (lldb) po session.currentRoute.outputs ▿ 1 element - 0 : <AVAudioSessionPortDescription: 0x282fce9d0, type = CarAudio; name = CarPlay; UID = 48:F0:7B:C6:21:A8-Audio-AudioMain-92004763965583; selectedDataSource = (null)> (lldb) po session.currentRoute.outputs.first!.channels ▿ Optional<Array<AVAudioSessionChannelDescription>> ▿ some : 1 element - 0 : <AVAudioSessionChannelDescription: 0x282fd8590, name = CarPlay; label = 0 (0x0); number = 1; port UID = 48:F0:7B:C6:21:A8-Audio-AudioMain-92004763965583> When I configure the session only for playback, the output is stereo, as you'd expect from a car system. This is on iOS 17 beta1, and I'm afraid I can't check whether this is a new regression or has already existed before, but it's quite likely it has existed before. Any advice on how I can circumvent this issue?

Turn on address sanitizer on Xcode and use a real device and put a Test.mp3 file in the Xcode project. Then it will crash when you initialise a AVAudioPlayer with a mp3 file (with a wav file it works fine). I have made an entry in feedback assistent -> FB12425453. var player : AVAudioPlayer? func playSound() { if let url = Bundle.main.url(forResource: "Test", withExtension: "mp3") { self.player = try? AVAudioPlayer(contentsOf: url) // --> deallocation of non allocated memory problem --> with a "wav" file it works .... } }

I am analysing sounds by tapping the mic on the Mac. All is working well, but it disrupts other (what I assume) are low priority sounds e.g. dragging an item off the dock, sending a message is messages, speaking something in Shortcuts or Terminal. Other sounds like music.app playing, Siri speaking are not disrupted. The disruption sounds like the last part of the sound being repeated two extra times, very noticeable. This is the code: import Cocoa import AVFAudio class AudioHelper: NSObject { let audioEngine = AVAudioEngine() func start() async throws { audioEngine.inputNode.installTap(onBus: 0, bufferSize: 8192, format: nil) { buffer, time in } try audioEngine.start() } } I have tried increasing the buffer, changing the qos to utility (in the hope the sound analysis would become less important than the disrupted sounds),running on a non-main thread, but no luck. MacOS 13.4.1 Any assistance would be appreciated.

I have a timer in one of my apps. I now want to add audio that plays at the end of the timer. It's a workout app and the sound should remind the user that it is time for the next exercise. The audio should duck music playback (Apple Music / Spotify) and also work in background. Background audio is enabled for the app. I am not able to achieve everything at the same time. I set the audio session to category playback with options duckOthers. do { try AVAudioSession.sharedInstance().setCategory( .playback, options: .duckOthers ) } catch { print(error) } For playback I just use the AVAudioPlayer. When the user starts the timer, i schedule a timer in the future and play the sound. While this works perfectly in the foreground, the sound is not played back when going to background, as timers are not fired in the background, but rather when the user puts the app back in foreground. I have also tried using AVAudioEngine and AVAudioPlayerNode, as the latter can start playback delayed. The case from above works now, but the audio ducking begins immediately when initialising the AVAudioEngine, which is also not what i want. Is there any other approach that I am not aware of?

Hi everyone! We develop some fitness app and we met an issue with level of the sound in our videos. The flow is: the User is listening their music from streaming platform. Then he opens our app and decided to start a workout. While he is making their exercises they have an option to watch the video how to make it. And that's where smth weird happend. When the User clicks on video to play and once it's playing the music from streaming app is a little reducing, not so loud as it was before. When the User is reducing the sound lever with their side button on in our video player volume slider, the music volume is also reducing. Ans sometimes the music volume is reducing and doesn't go back up until the User kills the our app and restart the workout. So there's some conflict between our app and streaming app. Maybe someone now the solution how fix this issue and do not change the sound volume from streaming app while Users is watching videos? Thanks in advance, will be glad to receive your suggestions

I'd like to allow the speech synthesizer to play on the device speaker while simultaneously mixing with a phone call. I've worked with a number of different configurations but am unable to find a configuration that achieves the functionality I am trying to achieve - or allows mixing with a phone call at all. There is a flag: mixToTelephonyUplink that seems to suggest that at least some mixing with a phone call is possible using the speech synthesizer, but I'm currently unable to find almost any documentation about this flag besides basic API docs. I've had some some luck at least getting the synthesizer to always play to the speaker with the following audio session configuration - but the sound never is mixed with a phone call. Instead, it is ducked and muted while the phone call takes place. I've tried quite a few configuration combinations for the category and overrides, but nothings seems to work quite as I'd expect it to. synthesizer.mixToTelephonyUplink = true try? audioSession.setCategory(.playback, mode: .voicePrompt, options: [.mixWithOthers, .defaultToSpeaker]) try? audioSession.setActive(true, options: []) try? audioSession.overrideOutputAudioPort(.speaker) Is there some kind of documentation for this that's off the beaten path that I'm somehow missing? I'm going to continue with guess and check, but I'm starting to think this flag - and the functionality it implies, actually wasn't ever fully implemented.

I am wondering if it's possible to obtain audio focus when the app is in the background without using the duckOthers option. I tested the Amazon Echo buds with the Alexa app and found that it can obtain audio focus. I am curious about how this is accomplished. I have a BLE device that can connect with my app. After connecting the device and my app, I put my app in the background and play a song from the Spotify app. Then, when I press a button on my BLE device, it sends a BLE command to my app to play music. However, my app cannot obtain audio focus, so the music cannot be played. The only way to make it work is to configure duckOthers. Compare with Echo buds, if we do the same steps, it can get audio focus. Is it because it has the MFI? do { let options: AVAudioSession.CategoryOptions = [.allowBluetoothA2DP, .defaultToSpeaker, .duckOthers] try audioSession.setCategory(.playAndRecord, mode: .spokenAudio, options: options) DDLogDebug("\(LOG_TAG) \(#function) setting category: \(audioSession.category.rawValue), " + "options: \(audioSession.categoryOptions.rawValue)") } catch { DDLogWarn("\(LOG_TAG) \(#function) Failed to configure audio session: \(error.localizedDescription)") } }

I was play a pcm(24kHz 16bit) file with AudioQueue, and it mix with other sound( 192kHz 24bit) named sound2. Setting for AVAudioSession as: category (AVAudioSessionCategoryPlayback), options (AVAudioSessionCategoryOptionMixWithOthers|AVAudioSessionCategoryOptionDuckOthers) when playing pcm the sound2 should pushed volume lower as setting. BUT, there has a absolutly mute keep 0.5 second when the sound2 become low, and after a 0.5s mute the pushed lower sound came out. It only become in sound2 situation(192kHz, 24bit). if the sound2's quality lower everything is OK.

The loop plays smoothly in audacity but when I run it in the device or simulator it clicks each loop at different intensities. I config the session at App level: let audioSession = AVAudioSession.sharedInstance() do { try audioSession.setCategory(.playback, mode: .default, options: [.mixWithOthers]) try audioSession.setActive(true) } catch { print("Setting category session for AVAudioSession Failed") } And then I made my method on my class: func playSound(soundId: Int) { let sound = ModelData.shared.sounds[soundId] if let bundle = Bundle.main.path(forResource: sound.filename, ofType: "flac") { let backgroundMusic = NSURL(fileURLWithPath: bundle) do { audioPlayer = try AVAudioPlayer(contentsOf:backgroundMusic as URL) audioPlayer?.prepareToPlay() audioPlayer?.numberOfLoops = -1 // for infinite times audioPlayer?.play() isPlayingSounds = true } catch { print(error) } } } Does anyone have any clue? Thanks! PS: If I use AVQueuePlayer and repeat the item the click noise disappear (but its no use, because I would need to repeat it indefinitely without wasting memory), if I use AVLooper I get a silence between loops. All with the same sound. Idk :/ PS2: The same happens with ALAC files.

I have the Flutter mobile app and I'm using the record flutter package for recording audio. So I'm facing an issue while recording the audio while the phone is locked. App Behavior: First we start the app and connect it to a Bluetooth device Then the app starts looking for the trigger of 1 from the device connected with it. On receiving the trigger from device it start recording. while mobile locked and app is running in background. AVAudioSession_iOS.mm:2367 Failed to set category, error: '!int' Failed to set up audio session: Error Domain=NSOSStatusErrorDomain Code=560557684 "(null)" I'm getting this error when AVAudioSession setting the category. My is for Users security purpose so it need to record background let me know how can I achive this functionality

Our app is a game written in Unity where we have most of our audio playback handled by Unity. However, one of our game experiences utilized microphone input for speech recognition, and so in order for us to perform echo cancellation (while the game has audio playback), we setup an audio stream from Unity to native Swift code that performs the mixing of the input/output nodes. We however found that by streaming the audio buffer to our AVAudioSession: The volume of the audio playback appears to output differently When capturing a screen recording of the app, the audio playback being played from AVAudioSession does not get captured at all. Looking to figure out what could be causing the discrepency in playback as well as capture behaviour during screen recordings. We setup the AVAudioSession with this configuration: AVAudioSession.sharedInstance().setCategory(AVAudioSession.Category.playAndRecord, options: .mixWithOthers) with inputNode.setVoiceProcessingEnabled(true) after connecting our IO and mixer nodes. Any suggestions or ideas on what to look out for would be appreciated!

AVSpeechSynthesizer was not working. it was working perfect before. below is my code objective - c. -(void)playVoiceMemoforMessageEVO:(NSString*)msg { [[AVAudioSession sharedInstance] overrideOutputAudioPort:AVAudioSessionPortOverrideSpeaker error:nil]; AVSpeechSynthesizer *synthesizer = [[AVSpeechSynthesizer alloc]init]; AVSpeechUtterance *speechutt = [AVSpeechUtterance speechUtteranceWithString:msg]; speechutt.volume=90.0f; speechutt.rate=0.50f; speechutt.pitchMultiplier=0.80f; [speechutt setRate:0.3f]; speechutt.voice = [AVSpeechSynthesisVoice voiceWithLanguage:@"en-us"]; [synthesizer speakUtterance:speechutt]; } please help me to solve this issue.

When recording audio over bluetooth from AirPods to iPhone using the AVAudioRecorder the Bluetooth audio codec used is always AAC-ELD independent of the codec to store which is selected in the AVAudioRecorder instance. As far as I know must every Bluetooth device support SBC, hence, it should be possible for the AirPods to transmit the recorded audio using the SBC codec instead of AAC-ELD. However, I could not find any resource on how the request this codec using the AVAudioRecorder or AVAudioEngine. Is it possible to request SBC at all and if yes how?

AVSpeechSynthesisVoice.speechVoices() returns voices that are no longer available after upgrading from iOS 16 to iOS 17 (although this has been an issue for a long time, I think). To reproduce: On iOS 16 download 1 or more enhanced voices under “Accessibility > Spoken Content > Voices”. Upgrade to iOS 17 Call AVSpeechSynthesisVoice.speechVoices() and note that the voices installed in step (1) are still present, yet they are no longer downloaded, therefore they don’t work. And there is no property on AVSpeechSynthesisVoice to indicate if the voice is still available or not. This is a problem for apps that allow users to choose among the available system voices. I receive many support emails surrounding iOS upgrades about this issue. I have to tell them to re-download the voices which is not obvious to them. I've created a feedback item for this as well (FB12994908).

Hi! I am working on an audio application on iOS. This is how I retreive the workgroup from the remoteIO audiounit (ioUnit). The unit is initialized and is working fine (meaning that it is regularly called by the system). os_workgroup_t os_workgroup{nullptr}; uint32_t os_workgroup_index_size; if (status = AudioUnitGetProperty(ioUnit, kAudioOutputUnitProperty_OSWorkgroup, kAudioUnitScope_Global, 0, &os_workgroup, &os_workgroup_index_size); status != noErr) { throw runtime_error("AudioUnitSetProperty kAudioOutputUnitProperty_OSWorkgroup - Failed with OSStatus: " + to_string(status)); } However the resulting os_workgroup's value is 0x40. Which seems not correct. No wonder I cannot join any other realtime threads to the workgroup as well. The returned status however is a solid 0. Can anyone help?

From an app that reads audio from the built-in microphone, I'm receiving many crash logs where the AVAudioEngine fails to start again after the app was suspended. Basically, I'm calling these two methods in the app delegate's applicationDidBecomeActive and applicationDidEnterBackground methods respectively: let audioSession = AVAudioSession.sharedInstance() func startAudio() throws { self.audioEngine = AVAudioEngine() try self.audioSession.setCategory(.record, mode: .measurement)} try audioSession.setActive(true) self.audioEngine!.inputNode.installTap(onBus: 0, bufferSize: 4096, format: nil, block: { ... }) self.audioEngine!.prepare() try self.audioEngine!.start() } func stopAudio() throws { self.audioEngine?.stop() self.audioEngine?.inputNode.removeTap(onBus: 0) self.audioEngine = nil try self.audioSession.setActive(false, options: [.notifyOthersOnDeactivation]) } In the crash logs (iOS 16.6) I'm seeing that this works fine several times as the app is opened and closed, but suddenly the audioEngine.start() call fails with the error Error Domain=com.apple.coreaudio.avfaudio Code=-10851 "(null)" UserInfo={failed call=err = AUGraphParser::InitializeActiveNodesInInputChain(ThisGraph, *GetInputNode())} and the audioEngine!.inputNode.outputFormat(forBus: 0) is something like <AVAudioFormat 0x282301c70: 2 ch, 0 Hz, Float32, deinterleaved> . Also, right before installing the tap, audioSession.availableInputs contains an entry of type MicrophoneBuiltIn but audioSession.currentRoute lists no inputs at all. I was not able to reproduce this situation on my own devices yet. Does anyone have an idea why this is happening?

when i found other sounds by using '[AVAudioSession sharedInstance].otherAudioPlaying', if it return true, i will get currentRoute. but i meet some issues, it's outputs & inputs number all > 0 . i want to judge the sound is by inputs devices or output devices? please help!!!!!!!