Many Apple users own both Bluetooth earphones (AirPods) and traditional wired earphones. While Bluetooth audio provides freedom of movement, some users still prefer wired earphones for comfort, sound profile, or personal preference. However, plugging wired earphones directly into an iPhone can feel restrictive and inconvenient during daily use. This proposal suggests a hybrid audio approach where wired earphones can be connected to a Bluetooth-enabled AirPods charging case (or a similar Apple-designed module), allowing users to enjoy wired earphones without a physical connection to the iPhone. #Problem Statement *Wired earphones offer consistent audio quality and zero latency *Bluetooth earphones provide freedom from cables *Users must currently choose one or the other *Plugging wired earphones into an iPhone limits movement and can feel intrusive in daily scenarios (walking, commuting, working) There is no native Apple solution that allows wired earphones to function wirelessly while maintaining Apple’s audio experience standards. #Proposed Solution Introduce a Wired-to-Wireless Audio Mode through the AirPods charging case or a dedicated Apple Bluetooth audio bridge. How it works: User plugs wired earphones into the AirPods case (or a future AirPods accessory port) The case acts as a Bluetooth audio transmitter Audio is streamed wirelessly from iPhone to the case The case outputs audio to the wired earphones #User experiences: No cable connected to the iPhone Familiar wired earphone sound Freedom of movement similar to Bluetooth earbuds User Experience (UX Flow) Plug wired earphones into the AirPods case iPhone automatically detects: “Wired Earphones via AirPods Case” Seamless pairing using existing AirPods framework Audio controls, volume, and switching handled through iOS No additional apps required #Key Benefits Combines wired sound reliability with wireless convenience Reduces physical cable disturbance during use Extends usefulness of existing wired earphones Minimal learning curve for users Fits naturally into Apple’s ecosystem and design philosophy #Privacy & Performance Considerations On-device audio processing only No cloud involvement Low-latency audio using Apple’s proprietary Bluetooth codecs Power-efficient usage leveraging AirPods case battery #Target Users Users who prefer wired earphones but want wireless freedom Commuters and walkers Developers and professionals who multitask Users sensitive to Bluetooth earbud fit or comfort #Ecosystem Fit Builds on existing AirPods pairing and audio stack Aligns with Apple’s focus on seamless UX Could be implemented via: New AirPods hardware Firmware update + accessory Dedicated Apple audio bridge

I need to implement a solution through an API or custom driver to completely block out the built-in speakers and microphone of Mac, because I need other apps to use specified external devices as audio input and output. Is there a way to achieve this requirement? What I mean is that even in system preferences, it should not be possible to choose the built-in microphone and speakers; only my external device can be used.

Hey everyone, I'm encountering an issue with audio sample rate conversion that I'm hoping someone can help with. Here's the breakdown: Issue Description: I've installed a tap on an input device to convert audio to an optimal sample rate. There's a converter node added on top of this setup. The problem arises when joining Zoom or FaceTime calls—the converter gets deallocated from memory, causing the program to crash. Symptoms: The converter node is being deallocated during video calls. The program crashes entirely when this happens. Traditional methods of monitoring sample rate changes (tracking nominal or actual sample rates) aren't working as expected. The Big Challenge: I can't figure out how to properly monitor sample rate changes. Listeners set up to track these changes don't trigger when the device joins a Zoom or FaceTime call. Please, if anyone has experience with this or knows a solution, I'd really appreciate your help. Thanks in advance! ⁠

Since iOS 18, the system setting “Allow Audio Playback” (enabled by default) allows third-party app audio to continue playing while the user is recording video with the Camera app. This has created a problem for the app I’m developing. ➡️ The problem: My app plays continuous audio in both foreground and background states. If the user starts recording video using the iOS Camera app, the app’s audio — still playing in the background — gets captured in the video — obviously an unintended behavior. Yes, the user could stop the app manually before starting the video recording, but that can’t be guaranteed. As a developer, I need a way to stop the app’s audio before the video recording begins. So far, I haven’t found a reliable way to detect when video recording starts if ‘Allow Audio Playback’ is ON. ➡️ What I’ve tried: — AVAudioSession.interruptionNotification → doesn’t fire — devicesChangedEventStream → not triggered I don’t want to request mic permission (app doesn’t use mic). also, disabling the app from playing audio in the background isn’t an option as it is a crucial part of the user experience ➡️ What I need: A reliable, supported way to detect when the Camera app begins video recording, without requiring mic access — so I can stop audio and avoid unintentional overlap with the user’s recordings. Any official guidance, workarounds, or AVFoundation techniques would be greatly appreciated. Thanks.

Currently, I have successfully used ChannelMap to map hardware input channels and obtained audio data from the hardware device's MIC and OTG inputs. Additionally, I have used ChannelMap to map output channels to freely feed data for playback to each output channel. However, I now have a problem. I have a hardware device that only has output channels (no input channels), and the system has set this hardware device as the default playback device. In this case, how can I obtain the audio data being played to the output channels for modification?

Two issues: No matter what I set in try audioSession.setPreferredSampleRate(x) the sample rate on both iOS and macOS is always 48000 when the output goes through the speaker, and 24000 when my Airpods connect to an iPhone/iPad. Now, I'm checking the current output loudness to animate a 3D character, using mixerNode.installTap(onBus: 0, bufferSize: y, format: nil) { [weak self] buffer, time in Task { @MainActor in // calculate rms and animate character accordingly but any buffer size under 4800 is just ignored and the buffers I get are 4800 sized. This is ok, when the sampleRate is currently 48000, as 10 samples per second lead to decent visual results. But when AirPods connect, the samplerate is 24000, which means only 5 samples per second, so the character animation looks lame. My AVAudioEngine setup is the following: audioEngine.connect(playerNode, to: pitchShiftEffect, format: format) audioEngine.connect(pitchShiftEffect, to: mixerNode, format: format) audioEngine.connect(mixerNode, to: audioEngine.outputNode, format: nil) Now, I'd be fine if the outputNode runs at whatever if it needs, as long as my tap would get at least 10 samples per second. PS: Specifying my favorite format in the let format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channels: 2)! mixerNode.installTap(onBus: 0, bufferSize: y, format: format) doesn't change anything either

I have a flutter iOS app that has some simple sound FX for button clicks, swipes, etc. In simulator and on real device the sound works fine, but when i upload the app to testflight (and App store) the sound FX don't play. When I upload the app to my phone via xcode I am using the release profile so I don't see what the difference could be. I have also gone through the archive that i uploaded and verified that the sound files are indeed there. I have other flutter apps that use sound but non since the iOS 26 update. I've tried 3 different flutter sound libraries and all face the same issue. Wondering if anyone else is seeing this issue or if I'm missing a simple permission or something that has changed recently? Thanks in advanced

Hello, I'm observing an intermittent memory leak being reported in the iOS Simulator when initializing and starting an AVAudioEngine. Even with minimal setup—just attaching a single AVAudioPlayerNode and connecting it to the mainMixerNode—Xcode's memory diagnostics and Instruments sometimes flag a leak. Here is a simplified version of the code I'm using: // This function is called when the user taps a button in the view controller: #import "ViewController.h" @interface ViewController () @end @implementation ViewController - (void)viewDidLoad { [super viewDidLoad]; } - (IBAction)myButtonAction:(id)sender { NSLog(@"Test"); soundCreate(); } @end // media.m static AVAudioEngine *audioEngine = nil; void soundCreate(void) { if (audioEngine != nil) return; [[AVAudioSession sharedInstance] setCategory:AVAudioSessionCategoryAmbient error:nil]; [[AVAudioSession sharedInstance] setActive:YES error:nil]; audioEngine = [[AVAudioEngine alloc] init]; AVAudioPlayerNode* playerNode = [[AVAudioPlayerNode alloc] init]; [audioEngine attachNode:playerNode]; [audioEngine connect:playerNode to:(AVAudioNode *)[audioEngine mainMixerNode] format:nil]; [audioEngine startAndReturnError:nil]; } In the memory leak report, the following call stack is repeated, seemingly in a loop: ListenerMap::InsertEvent(XAudioUnitEvent const&, ListenerBinding*) AudioToolboxCore ListenerMap::AddParameter(AUListener*, void*, XAudioUnitEvent const&) AudioToolboxCore AUListenerAddParameter AudioToolboxCore addOrRemoveParameterListeners(OpaqueAudioComponentInstance*, AUListenerBase*, AUParameterTree*, bool) AudioToolboxCore 0x180178ddf

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; }

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.

{ "aps": { "content-available": 1 }, "audio_file_name": "ding.caf", "audio_url": "https://example.com/audio.mp3" } When the app is in the background or killed, it receives a remote APNs push. The data format is roughly as shown above. How can I play the MP3 audio file at the specified "audio_url"? The user does not need to interact with the device when receiving the APNs. How can I play the audio file immediately after receiving it?

Session player regions populate blank, with no sound media when tracks or regions are created.

I have both apple devices, AirPods Pro 3 is up to date and Ultra 3 is on watch os 26.1 latest public beta. Each morning when I would go on my mindfulness app and start a meditation or listen to Apple Music on my watch and AirPods Pro 3, it will play for a few seconds then disconnects. My bluetooth settings on my watch says my AirPods is connected to my watch. I also have removed the tick about connecting automatically to iPhone on the AirPods setting in my iPhone. To fix this I invariably turn off my Apple Watch Ultra 3 and turn it on again. Then the connection becomes stable. I am not sure why I have to do this each morning. It is frustrating. I am not sure why this fix does not last long? Is there something wrong with my AirPods? Has anyone encountered this before?

I am developing an app that uses MusicKit to play music and then I need to have spoken words played to the user, while ducking the audio coming from MusicKit (application music player) the built in Siri voices are not off sufficient quality so I am using an external service to create an mp3 file and then play this back using AVAudioSession Sample code below the problem I am having is that .duckOthers is not ducking the Application Music Player output Is this a bug or am I doing this wrong? // Configure audio session for system-wide ducking try AVAudioSession.sharedInstance().setCategory(.playback, mode: .spokenAudio, options: [.duckOthers, .mixWithOthers]) try AVAudioSession.sharedInstance().setActive(true) // Set the ducking level to maximum try AVAudioSession.sharedInstance().setPreferredIOBufferDuration(0.005) // Create and configure audio player self.audioPlayer = try AVAudioPlayer(data: audioData) self.audioPlayer?.delegate = self self.audioPlayer?.volume = 1.0 // Ensure full volume for speech self.audioPlayer?.prepareToPlay() // Set the audio player's settings for maximum clarity self.audioPlayer?.enableRate = false self.audioPlayer?.pan = 0.0 // Center the audio self.audioPlayer?.play()

Hi everyone 👋 I’m building an iOS app in Swift where I want to do the following: Record the user’s voice Transcribe the spoken sentence (speech-to-text) Auto-detect the spoken language Translate it to another language selected by the user (e.g., English → Spanish or Hindi → English) Speak back (text-to-speech) the translated text on the same device Is this possible to record via phone mic and play the transcribe voice into headphone's audio?

Please Update Andorid MusicKit，the version 1.1.2 will complied fail。the error msg：•SDKUriHandlerActivity>. Apps targeting Android 12 and higher are required to specify an explicit value for android:exported when the corres

Hi folks - I'm having trouble finding specific documentation about Audio Unit MIDI plugins - as in MIDI -only. Any suggestions welcome as searches aren't returning much. (too niche? user error?)

Hi, In my project I am using AVFoundation for recording the audio. We are using AVAudioMixerNode class below method to record the audio packet. **func installTap( onBus bus: AVAudioNodeBus, bufferSize: AVAudioFrameCount, format: AVAudioFormat?, block tapBlock: @escaping AVAudioNodeTapBlock ) ** It works perfectly fine. But in production env some small percentage of the user we are facing issue like after recording few packets it stops automatically without stopping the audio engine. Can anyone help here that why this happens? I have also observed for mediaServicesWereResetNotification and added log on receiving this notification but when this issue happens I don't see any occurence of this log. Also is there any callback when the engine stops?

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?

Does anyone know how to pronounce the sound of a specific instrument when you tap a button on the screen on your iPhone or iPad? Now, in the middle of creating a music learning app, I'm thinking of assigning monotones or chords to the button-like frames on the keyboard and fingerboard on the screen. Can it be achieved with SwiftUI chords alone? Once upon a time, MIDI level 1 I remember that there was a pronunciation function of the instrument, but I don't think about implementing the same function in the current OS. Please lend me your wisdom.