I ran 5.1 audio tests in both YouTube and Apple Music, and I noticed that when sound is supposed to play from the rear or front surround speakers, it’s also duplicated in the front left and right channels. I’m absolutely sure the issue is with the Apple TV, because I played the same video directly through my TV’s native system, and the channel separation was correct. Everything used to work perfectly before, so this must be a software issue. I’m currently on tvOS 26 Developer Beta 5, but I’m certain the problem also existed on the stable tvOS 18.5. I’ve already reset and updated my Apple TV, and I also tried switching the audio format to forced Dolby Atmos 5.1. On the forums, I mostly see complaints about Dolby Atmos not working at all — in my case, everything technically works, but not the way it’s supposed to.

Hi team, With regards to Call (Live) Translations on VOIP: Is it possible to invoke live translations within the app? (without going into the Call System UI) Is it possible to navigate users from app to Call System UI via an API? (and also invoking the new live translations directly) Will Apple support more languages apart from the current ones? (Currently I see 4 supported languages)

Using the official SwiftTranscriptionSampleApp from WWDC 2025, speech transcription takes 14+ seconds from audio input to first result, making it unusable for real-time applications. Environment iOS: 26.0 Beta Xcode: Beta 5 Device: iPhone 16 pro Sample App: Official Apple SwiftTranscriptionSampleApp from WWDC 2025 Configuration Tested Locale: en-US (properly allocated with AssetInventory.allocate(locale:)) and es-ES Setup: All optimizations applied (preheating, high priority, model retention) I started testing in my own app to replace SFSpeech API and include speech detection but after long fights with documentation (this part is quite terrible TBH) I tested the example (https://developer.apple.com/documentation/speech/bringing-advanced-speech-to-text-capabilities-to-your-app) and saw same results. I added some logs to check the specific time: 🎙️ [20:30:41.532] ✅ Analyzer started successfully - ready to receive audio! 🎙️ [20:30:41.532] Listening for transcription results... 🎙️ [20:30:56.342] 🚀 FIRST TRANSCRIPTION RESULT after 14.810s: 'Hello' (isFinal: false) Questions Is this expected performance for iOS 26 Beta, because old SFSpeech is far faster? Are there additional optimization steps for SpeechTranscriber? Should we expect significant performance improvements in later betas?

After updating to iOS 18.5, we’ve observed that outgoing audio from our app intermittently stops being transmitted during VoIP calls using AVAudioSession configured with .playAndRecord and .voiceChat. The session is set active without errors, and interruptions are handled correctly, yet audio capture suddenly ceases mid-call. This was not observed in earlier iOS versions (≤ 18.4). We’d like to confirm if there have been any recent changes in AVAudioSession, CallKit, or related media handling that could affect audio input behavior during long-running calls. func configureForVoIPCall() throws { try setCategory( .playAndRecord, mode: .voiceChat, options: [.allowBluetooth, .allowBluetoothA2DP, .defaultToSpeaker]) try setActive(true) }

I was testing audio playback from YouTube in Safari, and the sound was clipping heavily. At first, I thought it might be due to the poor quality of my small sound system. However, when I took a screenshot and the screenshot sound effect itself produced a loud clipping noise, it became clear that this is not a mechanical problem with my speakers, nor an issue specific to YouTube or Safari. This appears to be a system-wide audio issue in macOS Tahoe 26 - Beta 5.

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'm experiencing a significant limitation with MusicKit's Dolby Atmos implementation on macOS and would appreciate clarification on whether this is intended behavior or if there are solutions available. When streaming Dolby Atmos content through MusicKit's ApplicationMusicPlayer, the output is limited to 2-channel stereo, even when: Audio MIDI Setup is configured for 7.1.4 (12-channel) output The same tracks play in full multichannel through the native Apple Music app Dolby Atmos is set to "Automatic" in Apple Music preferences Please let me know if there is anyway to enable this. If not, is this documented anywhere? Thanks!

Hello, I’m new here. I'm developing an iOS app and I’d like to know whether it is possible to detect if a phone call is being recorded by another app running in the background. I’ve already reviewed the documentation for CallKit and AVAudioSession, but I couldn’t find anything related. My expectation was that iOS might provide some callback or API to indicate if a call is being recorded (third-party apps), but so far I haven’t found a way. My questions are: Does iOS expose any API to detect if a call is being recorded? If not, is there any indirect, Apple's policy compliant method (e.g., microphone usage events) that can be relied upon? Or is this something that iOS explicitly prevents for privacyreasons? Expecting solutions that align with Apple’s policies and would be accepted under the App Store Review Guidelines. Thanks in advance for any guidance.

Not able to record audio in AAC format with 96 kHz sample rate using AVAudioRecorder or Extended Audio File services with 96 kHz input audio from input device. The audio recording settings used are let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatMPEG4AAC), AVSampleRateKey: sampleRate AVNumberOfChannelsKey: 1 AVEncoderAudioQualityKey: AVAudioQuality.high.rawValue ] When tried using AVAudioEngine using AVAudioFile, AVAudioFile(forWriting: fileURL, // file extension .m4a settings: fileSettings, commonFormat: AVAudioCommonFormat.pcmFormatFloat32, interleaved: interleaved) else { return } got error CodecConverterFactory.cpp:977 unable to select compatible encoder sample rate AudioConverter.cpp:1017 Failed to create a new in process converter -> from 1 ch, 96000 Hz, Float32 to 1 ch, 96000 Hz, aac (0x00000000) 0 bits/channel, 0 bytes/packet, 0 frames/packet, 0 bytes/frame, with status 1718449215

I have a PCM audio buffer (AVAudioPCMFormatInt16). When I try to play it using AVPlayerNode / AVAudioEngine an exception is thrown: "[[busArray objectAtIndexedSubscript:(NSUInteger)element] setFormat:format error:&nsErr]: returned false, error Error Domain=NSOSStatusErrorDomain Code=-10868 (related thread https://forums.developer.apple.com/forums/thread/700497?answerId=780530022#780530022) If I convert the buffer to AVAudioPCMFormatFloat32 playback works. My questions are: Does AVAudioEngine / AVPlayerNode require AVAudioPCMBuffer to be in the Float32 format? Is there a way I can configure it to accept another format instead for my application? If 1 is YES is this documented anywhere? If 1 is YES is this required format subject to change at any point? Thanks! I was looking to watch the "AVAudioEngine in Practice" session video from WWDC 2014 but I can't find it anywhere (https://forums.developer.apple.com/forums/thread/747008).

I am building an app where I am using AudioRecordingIntent to start audio recording from shortcuts / Action button etc. Whenever I set that up, I notice that I get an error - Unknown NSError Live Activity start failed: The operation couldn’t be completed. Target is not foreground I explicitly try to start the live activity and then start the audio recording and that's when I see this error. How can I make this work? I am unable to find any examples.

Hello, I am wondering if it is possible to have audio from my AirPods be sent to my speech to text service and at the same time have the built in mic audio input be sent to recording a video? I ask because I want my users to be able to say "CAPTURE" and I start recording a video (with audio from the built in mic) and then when the user says "STOP" I stop the recording.

Hello, Has anyone else experienced variations in the accuracy of the playbackTime value? After a few seconds of playback, the reported time adjusts by a fraction of a second, making it difficult to calculate the actual playbackTime of the audio. This can be recreated by playing a song in MusicKit, recording the start time of the audio, playing for at least 10-20 seconds, and then comparing the playbackTime value to one calculated using the start time of the audio. In my experience this jump occurs after about 10 seconds of playback. Any help would be appreciated. Thanks!

The sysEx struct in the MIDIUniversalMessage struct has a channel member but the System Exclusive (7-Bit) Message doesn't have a channel field. The System Exclusive (7-Bit) Message has a # of bytes field but the sysEx struct doesn't have a nrOfBytes, byteCount or bytesUsed member. It looks like the channel member of the sysEx struct contains the number of used bytes. Is this a mistake in the header or did I misunderstand something?

FaceTime’s screen-share audio balance is insanely absurd right now. Whenever I share media, the system audio that gets sent through FaceTime is a tiny whisper even at full volume (or even when connected to my speaker or headphones). The moment anyone on the call makes any noise at all, the shared audio ducks so hard it disappears, while the voice (or rustling or air conditioning noise) spikes to painful levels. It’s impossible to watch or listen to anything together. Also, the feature where FaceTime would shrink to a square during screen-sharing has been completely removed. That was a good feature and I'm really confused why it's gone. Now, the FaceTime window stays as a long rectangle that covers part of the content I'm trying to share (unless I do full screen tile, but then I can't pull up any other windows during the call) and can't be made smaller than about a third of the screen. You can't resize the window or adjust its dimensions, so it ends up blocking the actual media you're trying to watch. Here are some feature requests/fixes that would greatly improve the FaceTime screen-share experience: Option to adjust the shared media volume independently of call audio. Disable/toggle the extreme automatic audio docking while screen-sharing Reintroduce the minimized “floating square” mode or allow full manual resizing and repositioning of the FaceTime window during screen-share sessions. Overall, this setup makes FaceTime screen-sharing basically unusable. The audio balance is so inconsistent that it’s easier to switch to Zoom or Google Meet, which both handle shared sound correctly and let you move the call window out of the way. Until these issues are fixed, there’s no practical reason to use FaceTime for shared viewing at all.

We are developing an apple music app on phone, the developed web works fine on chrome, but when i load it on webivew on my phone, i can't play the first song, We doubt that the drm init, key exchange, session creation was on the music.play() function, while we trigger the play, the drm or session was not ok for play a real song, so it got an error so we may wanna know: what about the realative process of drm, key, session, etc in the play() function? are there some state detect function to show weather the drm is ok？

Hi everyone, I’m working on an iOS MusicKit app that overlays a metronome on top of Apple Music playback, using ApplicationMusicPlayer. To line the clicks up perfectly I’d like access to low-level audio analysis data—ideally a waveform / spectrogram or beat grid—while the track is playing. I’ve noticed that several approved DJ apps (e.g. djay, Serato, rekordbox) can already: • Display detailed scrolling waveforms of Apple Music songs • Scratch, loop or time-stretch those tracks in real time That implies they receive decoded PCM frames or at least high-resolution analysis data from Apple Music under a special entitlement. My questions: Does MusicKit (or any public framework) expose real-time audio buffers, FFT bins, or beat markers for streaming Apple Music content? If not, is there an Apple program or entitlement that developers can apply for—similar to the “DJ with Apple Music” initiative—to gain that deeper access? Where can I find official documentation or a point of contact for this kind of request? I’ve searched the docs and forums but only see standard MusicKit playback APIs, which don’t appear to expose raw audio for DRM-protected songs. Any guidance, links or insider tips on the proper application process would be hugely appreciated! Thanks in advance.

I have an app that records a health provider’s conversation with a patient. I am using Audio Queue Services for this. If a phone call comes in while recording, the doctor wants to be able to ignore the call and continue the conversation without touching the phone. If the doctor answers the call, that’s fine – I will stop the recording. I can detect when the call comes in and ends using CXCallObserver and AVAudioSession.interruptionNotification. Unfortunately, when a call comes in and before it is answered or dismissed, the audio is suppressed. After the call is dismissed, the audio continues to be suppressed. How can I continue to get audio from the mic as long as the user does not answer the phone call?

Hi everyone, I’m testing audio recording on an iPhone 15 Plus using AVFoundation. Here’s a simplified version of my setup: let settings: [String: Any] = [ AVFormatIDKey: Int(kAudioFormatLinearPCM), AVSampleRateKey: 8000, AVNumberOfChannelsKey: 1, AVLinearPCMBitDepthKey: 16, AVLinearPCMIsFloatKey: false ] audioRecorder = try AVAudioRecorder(url: fileURL, settings: settings) audioRecorder?.record() When I check the recorded file’s sample rate, it logs: Actual sample rate: 8000.0 However, when I inspect the hardware sample rate: try session.setCategory(.playAndRecord, mode: .default) try session.setActive(true) print("Hardware sample rate:", session.sampleRate) I consistently get: `Hardware sample rate: 48000.0 My questions are: Is the iPhone mic actually capturing at 8 kHz, or is it recording at 48 kHz and then downsampling to 8 kHz internally? Is there any way to force the hardware to record natively at 8 kHz? If not, what’s the recommended approach for telephony-quality audio (true 8 kHz) on iOS devices? Thanks in advance for your guidance!

I'm working in Swift/SwiftUI, running XCode 16.3 on macOS 15.4 and I've seen this when running in the iOS simulator and in a macOS app run from XCode. I've also seen this behaviour with 3 different audio files. Nothing in the documentation says that the speechRecognitionMetadata property on an SFSpeechRecognitionResult will be nil until isFinal, but that's the behaviour I'm seeing. I've stripped my class down to the following: private var isAuthed = false // I call this in a .task {} in my SwiftUI View public func requestSpeechRecognizerPermission() { SFSpeechRecognizer.requestAuthorization { authStatus in Task { self.isAuthed = authStatus == .authorized } } } public func transcribe(from url: URL) { guard isAuthed else { return } let locale = Locale(identifier: "en-US") let recognizer = SFSpeechRecognizer(locale: locale) let recognitionRequest = SFSpeechURLRecognitionRequest(url: url) // the behaviour occurs whether I set this to true or not, I recently set // it to true to see if it made a difference recognizer?.supportsOnDeviceRecognition = true recognitionRequest.shouldReportPartialResults = true recognitionRequest.addsPunctuation = true recognizer?.recognitionTask(with: recognitionRequest) { (result, error) in guard result != nil else { return } if result!.isFinal { //speechRecognitionMetadata is not nil } else { //speechRecognitionMetadata is nil } } } } Further, and this isn't documented either, the SFTranscriptionSegment values don't have correct timestamp and duration values until isFinal. The values aren't all zero, but they don't align with the timing in the audio and they change to accurate values when isFinal is true. The transcription otherwise "works", in that I get transcription text before isFinal and if I wait for isFinal the segments are correct and speechRecognitionMetadata is filled with values. The context here is I'm trying to generate a transcription that I can then highlight the spoken sections of as audio plays and I'm thinking I must be just trying to use the Speech framework in a way it does not work. I got my concept working if I pre-process the audio (i.e. run it through until isFinal and save the results I need to json), but being able to do even a rougher version of it 'on the fly' - which requires segments to have the right timestamp/duration before isFinal - is perhaps impossible?