Hi, On macOS I used to open MP3 and MP4 files with ExtAudioFile. For a few years it doesn't work anymore. So I decided to try different macOS API using the AudioFileID of AudioToolbox framework. I decided to write a test: https://gist.github.com/joelkraehemann/7f5b241b52ca38c3a765c138fb647588 It fails right here: AudioFileOpenWithCallbacks() By telling OSStatus error 1954115647, which means kAudioFileUnsupportedFileTypeError. The filename was set to an MP4 file: ~/Music/test.mp4 Howto fix this? regards, Joël

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?

using iOS 26.2; Airpods 4 Long press stem to launch Siri Speak "Record Voice Memo" -> Recording starts Recording in progress... Long press stem to launch Siri -> Nothing happens. To stop recording need use phone. is this intended behaviour? i would like to be able to stop recording with Siri I am able to launch Siri from phone while recording, but point is to keep phone in pocket and start/stop recordings only via Airpods.

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).

There appears to be no method of going forward or backwards in Get Info in the Music application,

After investing more than a week into getting a bunch of audio unit projects converted into app + appex + framework, they all are now correctly loaded in-process in the demo host app that is part of Xcode's template. However, Logic Pro adamantly refuses to load them in-process. Does Logic Pro simply not do that ever, or is there some hint or configuration my plugins need to provide to enable that? If it is unsupported, will it be supported in some future version of Logic? The entire point of investing that week was performance, which is moot if it is impossible to test the impact of loading in-process in a real-world usage scenario.

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.

Hi, I'm working on a project that uses the AVSpeechSynthesizer and AVSpeechUtterance. I discovered by chance that the AVSpeechSynthesizer automatically completes some words instead of just outputting what it's supposed to. These are abbreviations for days of the week or months, but not all of them. I don't want either of them automatically completed, but only the specified text. The completion transcends languages. I have written a short example program for demonstration purposes. import SwiftUI import AVFoundation import Foundation let synthesizer: AVSpeechSynthesizer = AVSpeechSynthesizer() struct ContentView: View { var body: some View { VStack { Button { utter("mon") } label: { Text("mon") } .buttonStyle(.borderedProminent) Button { utter("tue") } label: { Text("tue") } .buttonStyle(.borderedProminent) Button { utter("thu") } label: { Text("thu") } .buttonStyle(.borderedProminent) Button { utter("feb") } label: { Text("feb") } .buttonStyle(.borderedProminent) Button { utter("feb", lang: "de-DE") } label: { Text("feb DE") } .buttonStyle(.borderedProminent) Button { utter("wed") } label: { Text("wed") } .buttonStyle(.borderedProminent) } .padding() } private func utter(_ text: String, lang: String = "en-US") { let utterance = AVSpeechUtterance(string: text) let voice = AVSpeechSynthesisVoice(language: lang) utterance.voice = voice synthesizer.speak(utterance) } } #Preview { ContentView() } Thank you Christian

Is it possible to play WebM audio on iOS? Either with AVPlayer, AVAudioEngine, or some other API? Safari has supported this for a few releases now, and I'm wondering if I missed something about how to do this. By default these APIs don't seem to work (nor does ExtAudioFileOpen). Our usecase is making it possible for iOS users to play back audio recorded in our webapp (desktop versions of Chrome & Firefox only support webm as a destination format for MediaRecorder)

Hi all, I've developed an audio DSP application in C++ using AudioToolbox and CoreAudio on MacOS 14.4.1 with Xcode 15. I use an AudioQueue for input and another for output. This works great. I'm now adding real-time audio analysis eg spectral analysis. I want this to run independently of my audio processing so it can not interfere with audio playback. Taps on AudioQueues seem to be a good way of doing this... Since the analytics won't modify the audio data, I am using a Siphon Tap by setting the AudioQueueProcessingTapFlags to kAudioQueueProcessingTap_PreEffects | kAudioQueueProcessingTap_Siphon; This works fine on my output queue. However, on my input queue the Tap callback is called once and then a EXC_BAD_ACCESS occurs - screen shot below. NB: I believe that a callback should only call AudioQueueProcessingTapGetSourceAudio when not using a Siphon, so I don't call it. Relevant code: AudioQueueProcessingTapCallback tap_callback) { // Makes an audio tap for a queue void * tap_data_ptr = NULL; AudioQueueProcessingTapFlags tap_flags = kAudioQueueProcessingTap_PostEffects | kAudioQueueProcessingTap_Siphon; uint32_t max_frames = 0; AudioStreamBasicDescription asbd; AudioQueueProcessingTapRef tap_ref; OSStatus status = AudioQueueProcessingTapNew(queue_ref, tap_callback, tap_data_ptr, tap_flags, &max_frames, &asbd, &tap_ref); if (status != noErr) printf("Error while making Tap\n"); else printf("Successfully made tap\n"); } void tapper(void * tap_data, AudioQueueProcessingTapRef tap_ref, uint32_t number_of_frames_in, AudioTimeStamp * ts_ptr, AudioQueueProcessingTapFlags * tap_flags_ptr, uint32_t * number_of_frames_out_ptr, AudioBufferList * buf_list) { // Callback function for audio queue tap printf("Tap callback"); }``` Image of exception stack provided by Xcode: ![]("https://developer.apple.com/forums/content/attachment/27479e8d-a118-459b-aa2d-7e30528910e3" "title=Screenshot 2025-06-14 at 1.29.14 PM.png;width=932;height=562") What have I missed? Appreciate any help you learned folks may be able to provide. Best, Geoff.

I'm writing a simple app for iOS and I'd like to be able to do some text to speech in it. I have a basic audio manager class with a "speak" function: import Foundation import AVFoundation class AudioManager { static let shared = AudioManager() var audioPlayer: AVAudioPlayer? var isPlaying: Bool { return audioPlayer?.isPlaying ?? false } var playbackPosition: TimeInterval = 0 func playSound(named name: String) { guard let url = Bundle.main.url(forResource: name, withExtension: "mp3") else { print("Sound file not found") return } do { if audioPlayer == nil || !isPlaying { audioPlayer = try AVAudioPlayer(contentsOf: url) audioPlayer?.currentTime = playbackPosition audioPlayer?.prepareToPlay() audioPlayer?.play() } else { print("Sound is already playing") } } catch { print("Error playing sound: \(error.localizedDescription)") } } func stopSound() { if let player = audioPlayer { playbackPosition = player.currentTime player.stop() } } func speak(text: String) { let synthesizer = AVSpeechSynthesizer() let utterance = AVSpeechUtterance(string: text) utterance.voice = AVSpeechSynthesisVoice(language: "en-GB") synthesizer.speak(utterance) } } And my app shows text in a ScrollView: ScrollView { Text(self.description) .padding() .foregroundColor(.black) .font(.headline) .background(Color.gray.opacity(0)) }.onAppear { AudioManager.shared.speak(text: self.description) } However, the text doesn't get read out (in the simulator). I see some output in the console: Error fetching voices: Swift.DecodingError.dataCorrupted(Swift.DecodingError.Context(codingPath: [], debugDescription: "Invalid container metadata for _UnkeyedDecodingContainer, found keyedGraphEncodingNodeID", underlyingError: nil)). Using fallback voices. I'm probably doing something wrong here, but not sure what.

A recent WWDC session "Learn about Apple Immersive Video technologies" showed a Apple Spatial Audio Format Panner plugin for Pro Tools. The presenter stated that it's available on a per-user license. Where can users access this?

Good day, ladies and gents. I have an application that reads audio from the microphone. I'd like it to also be able to read from the Mac's audio output stream. (A bonus would be if it could detect when the Mac is playing music.) I'd eventually be able to figure it out reading docs, but if someone can give a hint, I'd be very grateful, and would owe you the libation of your choice. Here's the code used to set up the AudioUnit: -(NSString*) configureAU { AudioComponent component = NULL; AudioComponentDescription description; OSStatus err = noErr; UInt32 param; AURenderCallbackStruct callback; if( audioUnit ) { AudioComponentInstanceDispose( audioUnit ); audioUnit = NULL; } // was CloseComponent // Open the AudioOutputUnit description.componentType = kAudioUnitType_Output; description.componentSubType = kAudioUnitSubType_HALOutput; description.componentManufacturer = kAudioUnitManufacturer_Apple; description.componentFlags = 0; description.componentFlagsMask = 0; if( component = AudioComponentFindNext( NULL, &description ) ) { err = AudioComponentInstanceNew( component, &audioUnit ); if( err != noErr ) { audioUnit = NULL; return [ NSString stringWithFormat: @"Couldn't open AudioUnit component (ID=%d)", err] ; } } // Configure the AudioOutputUnit: // You must enable the Audio Unit (AUHAL) for input and output for the same device. // When using AudioUnitSetProperty the 4th parameter in the method refers to an AudioUnitElement. // When using an AudioOutputUnit for input the element will be '1' and the output element will be '0'. param = 1; // Enable input on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32) ); chkerr("Couldn't set first EnableIO prop (enable inpjt) (ID=%d)"); param = 0; // Disable output on the AUHAL err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32) ); chkerr("Couldn't set second EnableIO property on the audio unit (disable ootpjt) (ID=%d)"); param = sizeof(AudioDeviceID); // Select the default input device AudioObjectPropertyAddress OutputAddr = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster }; err = AudioObjectGetPropertyData( kAudioObjectSystemObject, &OutputAddr, 0, NULL, &param, &inputDeviceID ); chkerr("Couldn't get default input device (ID=%d)"); // Set the current device to the default input unit err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDeviceID, sizeof(AudioDeviceID) ); chkerr("Failed to hook up input device to our AudioUnit (ID=%d)"); callback.inputProc = AudioInputProc; // Setup render callback, to be called when the AUHAL has input data callback.inputProcRefCon = self; err = AudioUnitSetProperty( audioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); param = sizeof(AudioStreamBasicDescription); // get hardware device format err = AudioUnitGetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &deviceFormat, &param ); chkerr("Could not install render callback on our AudioUnit (ID=%d)"); audioChannels = MAX( deviceFormat.mChannelsPerFrame, 2 ); // Twiddle the format to our liking actualOutputFormat.mChannelsPerFrame = audioChannels; actualOutputFormat.mSampleRate = deviceFormat.mSampleRate; actualOutputFormat.mFormatID = kAudioFormatLinearPCM; actualOutputFormat.mFormatFlags = kAudioFormatFlagIsFloat | kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonInterleaved; if( actualOutputFormat.mFormatID == kAudioFormatLinearPCM && audioChannels == 1 ) actualOutputFormat.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved; #if __BIG_ENDIAN__ actualOutputFormat.mFormatFlags |= kAudioFormatFlagIsBigEndian; #endif actualOutputFormat.mBitsPerChannel = sizeof(Float32) * 8; actualOutputFormat.mBytesPerFrame = actualOutputFormat.mBitsPerChannel / 8; actualOutputFormat.mFramesPerPacket = 1; actualOutputFormat.mBytesPerPacket = actualOutputFormat.mBytesPerFrame; // Set the AudioOutputUnit output data format err = AudioUnitSetProperty( audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &actualOutputFormat, sizeof(AudioStreamBasicDescription)); chkerr("Could not change the stream format of the output device (ID=%d)"); param = sizeof(UInt32); // Get the number of frames in the IO buffer(s) err = AudioUnitGetProperty( audioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &audioSamples, &param ); chkerr("Could not determine audio sample size (ID=%d)"); err = AudioUnitInitialize( audioUnit ); // Initialize the AU chkerr("Could not initialize the AudioUnit (ID=%d)"); // Allocate our audio buffers audioBuffer = [self allocateAudioBufferListWithNumChannels: actualOutputFormat.mChannelsPerFrame size: audioSamples * actualOutputFormat.mBytesPerFrame]; if( audioBuffer == NULL ) { [ self cleanUp ]; return [NSString stringWithFormat: @"Could not allocate buffers for recording (ID=%d)", err]; } return nil; } (...again, it would be nice to know if audio output is active and thereby choose the clean output stream over the noisy mic, but that would be a different chunk of code, and my main question may just be a quick edit to this chunk.) Thanks for your attention! ==Dave [p.s. if i get more than one useful answer, can i "Accept" more than one, to spread the credit around?] {pps: of course, the code lines up prettier in a monospaced font!}

The AVB AVnu MILAN Convention has a groweing Population. Many big companies (Cisco, Meyer Sound, d&b Audio, l‘acoustics, Presonus, digico etc.) implements the AVB AVnu Milan Standards. Is there a plan on the Apple side to also implement AVnu Milan on top of the AVB Protocol? The advantage for Apple Sound would be a great Integration in the professionell Audio market and a more stable intergration on top of the AVB protocol. The atdecc work, but Not that stable.

I might have misunderstood the docs, but is Call Translation going to be available for VOIP applications? Eg in an already connected VOIP call, would it be possible for Call Translations to be enabled on an iOS 26 and Apple Intelligence supported device? I have personally tried it and it doesn’t look like it supported VOIP but would love to confirm this. reference: https://developer.apple.com/documentation/callkit/cxsettranslatingcallaction/

HI Guys, I'm using Shazamkit in my IOS app and successfully capturing the currently playing track details, when using the devices (iPhone) built-in mic. When I test with AirPods though, my app cannot both send the output to through the AirPods and capture that same output with the AirPods mic, for Shazamkit recognition. I believe this must be possible, because the Shazamkit widget on IOS can do this. Is it restricted in some way for third party apps? If not, I'd appreciate some guidance on how to achieve this in Swift code. Thanks in advance.

My workout watch app supports audio playback during exercise sessions. When users carry both Apple Watch, iPhone, and AirPods, with AirPods connected to the iPhone, I want to route audio from Apple Watch to AirPods for playback. I've implemented this functionality using the following code. try? session.setCategory(.playback, mode: .default, policy: .longFormAudio, options: []) try await session.activate() When users are playing music on iPhone and trigger my code in the watch app, Apple Watch correctly guides users to select AirPods, pauses the iPhone's music, and plays my audio. However, when playback finishes and I end the session using the code below: try session.setActive(false, options:[.notifyOthersOnDeactivation]) the iPhone doesn't automatically resume the previously interrupted music playback—it requires manual intervention. Is this expected behavior, or am I missing other important steps in my code?

New to iOS development and I've been trying to make heads or tails of the documentation. I know there is a difference between the data fields returned from songs from the user library and from the category, but whenever I search on the apple site I can't find a list of each. For example, Im trying to get the releaseDate of a song in my library, but it seems I'll have to cross-query either the catalog entry for the using song.catalogID or the song.irsc but when I try to use them I can't find a cross reference between the two. I'm totally turned around. Also trying to determine if a song in my library has been favorited or not? isFavorited (or something similar) doesn't seem to be a thing. Using this code and trying to find a way to display a solid star if the song has been favorited or an empty one if it's not. Seems like a basic request but I can't find anything on how to do it. I've searched docs, googled, tried. Does apple want us to query the user's Favorited Songs playlist or something? How do I know which playlist that is? I know isFavorited isn't a thing, just using it here so you can see what my intension is: HStack(spacing: 10) { Image(systemName: song.isFavorited ? "star.fill" : "star") .foregroundColor(song.isFavorited ? .yellow : .gray) Image(systemName: "magnifyingglass") }

iOS 26.0 (23A5276f) – Bluetooth Call Audio Issue I’m experiencing a Bluetooth audio issue on iOS 26.0 (build 23A5276f). I cannot make or receive phone calls properly using Bluetooth devices — this affects both my car’s Bluetooth system and my AirPods Pro (2nd generation). Notably: Regular phone calls have no audio (either I can’t hear the other person, or they can’t hear me). WhatsApp and other VoIP apps work fine with the same Bluetooth devices. Media playback (music, video, etc.) works without issues over Bluetooth. It seems this bug is limited to the native Phone app or the system audio routing for regular cellular calls. Please advise if this is a known issue or if a fix is expected in upcoming beta releases.

Since the last update to IOS 26.0 (23A5276f) the AirPods connect to my IPhone and the Audio is still running through the phone. They are shown in the Bluetooth Icon that they’re paired.