I’ve been researching how to achieve a recording playback effect in iOS similar to the hands-free calling effect in the system’s phone app. How can this be implemented? I tried using the voice chat recording method, but found that the volume of the speaker output is too low. How should this issue be addressed? I couldn’t find a suitable API. Could you provide me with some documentation or sample code? Thank you.

Hi! I am writing a browser extension that allows you to control the playback of media content on a music service website. Unfortunately Safari does not support tracking changes to the audible property in an event tabs.onUpdated. Is there an alternative to this event? I'm looking for a way to track when the automatic inference engine interrupts playback on a music service website. That you.

3 I am working on an application to get when input audio device is being used. Basically I want to know the application using the microphone (built-in or external) This app runs on macOS. For Mac versions starting from Sonoma I can use this code: int getAudioProcessPID(AudioObjectID process) { pid_t pid; if (@available(macOS 14.0, *)) { constexpr AudioObjectPropertyAddress prop { kAudioProcessPropertyPID, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMain }; UInt32 dataSize = sizeof(pid); OSStatus error = AudioObjectGetPropertyData(process, &prop, 0, nullptr, &dataSize, &pid); if (error != noErr) { return -1; } } else { // Pre sonoma code goes here } return pid; } which works. However, kAudioProcessPropertyPID was added in macOS SDK 14.0. Does anyone know how to achieve the same functionality on previous versions?

I have an AUv3 that passes all validation and can be loaded into Logic Pro without issue. The UI for the plug in can be any aspect ratio but Logic insists on presenting it in a view with a fixed aspect ratio. That is when resizing, both the height and width are resized. I have never managed to work out what it is I need to do specify to Logic to allow the user to resize width or height independently of each other. Can anyone tell me what I need to specify in the AU code that will inform Logic that the view can be resized from any side of the window/panel?

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?

Hi, I'm working on an audio mixing app, that comes with bundled audio units that provide some of the app's core functionality. For the next release of that app, we are planning to make two changes: make the app sandboxed package the bundled audio units as .appex bundles instead as .component bundles, so we don't need to take care of the installation at the correct spot in the file system When trying this new approach, we run into problems where [[AVAudioUnitEffect alloc] initWithAudioComponentDescription:] crashes when trying to load our audio unit with the exception: AVAEInternal.h:109 [AUInterface.mm:468:AUInterfaceBaseV3: (AudioComponentInstanceNew(comp, &_auv2)): error -10863 Our audio unit has the `sandboxSafe flag enabled, and loads fine when the host app is not sandboxed, so I'm guessing I got the bundle id/code signing requirements for the .appex correct. It seems, that my .appex isn't even loaded, and the system rejects it because of its metadata. Maybe there something wrong the Info.plist generated by Juice? "BuildMachineOSBuild" => "23H222" "CFBundleDisplayName" => "elgato_sample_recorder" "CFBundleExecutable" => "ElgatoSampleRecorder" "CFBundleIdentifier" => "com.iwascoding.EffectLoader.samplerecorderAUv3" "CFBundleName" => "elgato_sample_recorder" "CFBundlePackageType" => "XPC!" "CFBundleShortVersionString" => "1.0.0.0" "CFBundleSignature" => "????" "CFBundleSupportedPlatforms" => [ 0 => "MacOSX" ] "CFBundleVersion" => "1.0.0.0" "DTCompiler" => "com.apple.compilers.llvm.clang.1_0" "DTPlatformBuild" => "24C94" "DTPlatformName" => "macosx" "DTPlatformVersion" => "15.2" "DTSDKBuild" => "24C94" "DTSDKName" => "macosx15.2" "DTXcode" => "1620" "DTXcodeBuild" => "16C5032a" "LSMinimumSystemVersion" => "10.13" "NSExtension" => { "NSExtensionAttributes" => { "AudioComponents" => [ 0 => { "description" => "Elgato Sample Recorder" "factoryFunction" => "elgato_sample_recorderAUFactoryAUv3" "manufacturer" => "Manu" "name" => "Elgato: Elgato Sample Recorder" "sandboxSafe" => 1 "subtype" => "Znyk" "tags" => [ 0 => "Effects" ] "type" => "aufx" "version" => 65536 } ] } "NSExtensionPointIdentifier" => "com.apple.AudioUnit-UI" "NSExtensionPrincipalClass" => "elgato_sample_recorderAUFactoryAUv3" } "NSHighResolutionCapable" => 1 } Any ideas what I am missing?

Hi all, with my app ScreenFloat, you can record your screen, along with system- and microphone audio. Those two audio feeds are recorded into separate audio tracks in order to individually remove or edit them later on. Now, these recordings you create with ScreenFloat can be drag-and-dropped to other apps instantly. So far, so good, but some apps, like Slack, or VLC, or even websites like YouTube, do not play back multiple audio tracks, just one. So what I'm trying to do is, on dragging the video recording file out of ScreenFloat, instantly baking together the two individual audio tracks into one, and offering that new file as the drag and drop file, so that all audio is played in the target app. But it's slow. I mean, it's actually quite fast, but for drag and drop, it's slow. My approach is this: "Bake together" the two audio tracks into a one-track m4a audio file using AVMutableAudioMix and AVAssetExportSession Take the video track, add the new audio file as an audio track to it, and render that out using AVAssetExportSession For a quick benchmark, a 3'40'' movie, step 1 takes ~1.7 seconds, and step two adds another ~1.5 seconds, so we're at ~3.2 seconds. That's an eternity for a drag and drop, where the user might cancel if there's no immediate feedback. I could also do it in one step, but then I couldn't use the AV*Passthrough preset, and that makes it take around 32 seconds then, because I assume it touches the video data (which is unnecessary in this case, so I think the two-step approach here is the fastest). So, my question is, is there a faster way? The best idea I can come up with right now is, when initially recording the screen with system- and microphone audio as separate tracks, to also record both of them into a third, muted, "hidden" track I could use later on, basically eliminating the need for step one and just ripping the two single audio tracks out of the movie and only have the video and the "hidden" track (then unmuted), but I'd still have a ~1.5 second delay there. Also, there's the processing and data overhead (basically doubling the movie's audio data). All this would be great for an export operation (where one expects it to take a little time), but for a drag-and-drop operation, it's not ideal. I've discarded the idea of doing a promise file drag, because many apps do not accept those, and I want to keep wide compatibility with all sorts of apps. I'd appreciate any ideas or pointers. Thank you kindly, Matthias

I have tried everything. The songs load unto the playlists and on searches, but when prompted to play, they just won't play. I have a wrapper since my main player (which carries the buttons for play/rewind/forward/etc.), is in Objc. // // ApplePlayerWrapper.swift // UniversallyMac // // Created by Dorian Mattar on 11/10/24. // import Foundation import MusicKit import MediaPlayer @objc public class MusicKitWrapper: NSObject { @objc public static let shared = MusicKitWrapper() private let player = ApplicationMusicPlayer.shared // Play the current track @objc public func play() { guard !player.queue.entries.isEmpty else { print("Queue is empty. Cannot start playback.") return } logPlayerState(message: "Before play") Task { do { try await player.prepareToPlay() try await player.play() print("Playback started successfully.") } catch { if let nsError = error as NSError? { print("NSError Code: \(nsError.code), Domain: \(nsError.domain)") } } logPlayerState(message: "After play") } } // Log the current player state @objc public func logPlayerState(message: String = "") { print("Player State - \(message):") print("Playback Status: \(player.state.playbackStatus)") print("Queue Count: \(player.queue.entries.count)") // Only log current track details if the player is playing if player.state.playbackStatus == .playing { if let currentEntry = player.queue.currentEntry { print("Current Track: \(currentEntry.title)") print("Current Position: \(player.playbackTime) seconds") print("Track Length: \(currentEntry.endTime ?? 0.0) seconds") } else { print("No current track.") } } else { print("No track is playing.") } print("----------") } // Debug the queue @objc public func debugQueue() { print("Debugging Queue:") for (index, entry) in player.queue.entries.enumerated() { print("\(index): \(entry.title)") } } // Ensure track availability in the queue public func queueTracks(_ tracks: [Track]) { Task { do { for track in tracks { // Validate Play Parameters guard let playParameters = track.playParameters else { print("Track \(track.title) has no Play Parameters.") continue } // Log the Play Parameters print("Track Title: \(track.title)") print("Play Parameters: \(playParameters)") print("Raw Values: \(track.id.rawValue)") // Ensure the ID is valid if track.id.rawValue.isEmpty { print("Track \(track.title) has an invalid or empty ID in Play Parameters.") continue } // Queue the track try await player.queue.insert(track, position: .afterCurrentEntry) print("Queued track: \(track.title)") } print("Tracks successfully added to the queue.") } catch { print("Error queuing tracks: \(error)") } debugQueue() } } // Clear the current queue @objc public func resetMusicPlayer() { Task { player.stop() player.queue.entries.removeAll() print("Queue cleared.") print("Apple Music player reset successfully.") } } } I opened an Apple Dev. ticket, but I'm trying here as well. Thanks!

Issue Description I'm implementing a system audio capture feature using AudioHardwareCreateProcessTap and AudioHardwareCreateAggregateDevice. The app successfully creates the tap and aggregate device, but when starting the IO procedure with AudioDeviceStart, it sometimes fails with OSStatus error 1852797029. (The operation couldn’t be completed. (OSStatus error 1852797029.)) The error occurs inconsistently, which makes it particularly difficult to debug and reproduce. Questions Has anyone encountered this intermittent "nope" error code (0x6e6f7065) when working with system audio capture? Are there specific conditions or system states that might trigger this error sporadically? Are there any known workarounds for handling this intermittent failure case? Any insights or guidance would be greatly appreciated. I'm wondering if anyone else has encountered this specific "nope" error code (0x6e6f7065) when working with system audio capture.

Recently, after the update of 26.3 Mac OS (Tahoe), the ordering of my music app has been horrible at best - music disappearing, tracks not aligning with albums (even if the albums are different years). It's created quite a problem, because the disappearing tracks issue seems to be replicating to iOS devices as well (although track numbering and album association seem to be stable). Has anyone else heard of this issue?

My current app implements a custom video player, based on a AVSampleBufferRenderSynchronizer synchronising two renderers: an AVSampleBufferDisplayLayer receiving decoded CVPixelBuffer-based video CMSampleBuffers, and an AVSampleBufferAudioRenderer receiving decoded lpcm-based audio CMSampleBuffers. The AVSampleBufferRenderSynchronizer is started when the first image (in presentation order) is decoded and enqueued, using avSynchronizer.setRate(_ rate: Float, time: CMTime), with rate = 1 and time the presentation timestamp of the first decoded image. Presentation timestamps of video and audio sample buffers are consistent, and on most streams, the audio and video are correctly synchronized. However on some network streams, on iOS, the audio and video aren't synchronized, with a time difference that seems to increase with time. On the other hand, with the same player code and network streams on macOS, the synchronization always works fine. This reminds me of something I've read, about cases where an AVSampleBufferRenderSynchronizer could not synchronize audio and video, causing them to run with independent and potentially drifting clocks, but I cannot find it again. So, any help / hints on this sync problem will be greatly appreciated! :)

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?

When using the Apple Devices to sync Apple Music to iPhone where is the Apple Devices backup being written to? Apple Devices->music->sync. Not trying to backup the iPhone via Apple Devices app.

Using an iPhone Pro 12 running iOS 26.0.1, with AirPods Pro 3. Camera app does capture video with what seems to be "Studio Quality Recording". Am trying to replicate that SQR with my own Camera like app, and while I can pull audio in from the APP3 mic, and my video capture app is recording a 48,000Hz high-bitrate video, the audio still sounds non-SQR. I'm seeing bluetoothA2DP , bluetoothLE , bluetoothHFP as portType, and not sure if SQR depends on one of those? Is there sample code demonstrating a SQR capture? Nevermind video and camera, just audio even? Also, I don't understand what SQR is doing between the APP3 and the iPhone. What codec is that? What bitrate is that? If I capture video using Capture and inspect the audio stream I see mono 74.14 kbit/s MPEG-4 AAC, 48000 Hz. But I assume that's been recompressed and not really giving me any insight into the APP3 H2 transmission?

We’ve encountered a reproducible issue where the iPhone fails to reconnect to a Wi-Fi access point under the following conditions: The device is connected to a 2.4GHz Wi-Fi network. A Bluetooth audio accessory is connected (e.g. headset). AVAudioSession is active (such as during a voice call or when using the Voice Memos app). The user moves away from the access point, causing a disconnect. Upon returning within range, the access point is no longer recognized or reconnected while AVAudioSession remains active. However, if the Bluetooth device is disconnected or the AVAudioSession is deactivated, the Wi-Fi access point is immediately recognized again. We confirmed this behavior not only in my app but also using Apple's built-in Voice Memos app, suggesting this is not specific to our implementation. It appears that the Wi-Fi system deprioritizes reconnection while AVAudioSession is engaged. Could this be by design? Or is this a known issue or limitation with Wi-Fi and AVAudioSession interaction? Test Environment: Device: iPhone 13 mini iOS: 17.5.1 Wi-Fi: 2.4GHz band Accessories: Bluetooth headset We’d appreciate clarification on whether this is expected behavior or a bug. Thank you!

Hello, I'm trying to determine the best/recommended AVAudioSession configuration (i.e category, mode, and options) for the following use-case. Essentially, I'd like to switch between periods of playing an audio file and then recognizing speech. The audio file is typically speech and I don't intend for playback and speech recognition to occur simultaneously. I'd like for the user to sill be able to interact with Siri and I'd like for it to work with CarPlay where navigation prompts can occur. I would assume the category to use is 'playAndRecord', but I'm not sure if it's better to just set that once for the entire lifecycle, or set to 'playback' for audio file playback and then switch to 'playAndRecord' for speech recognition . I'm also not sure on the best 'mode' and 'options' to set. Any suggestions would be appreciated. Thanks.

I recently got some plugins from Universal Audio, and have licensed them properly through both UA and iLok manager. Whenever I try to load up the plugins (specifically from UA) in GarageBand, it first says that "NSCreateObjectFileImageFromMemory-p47UEwps” because the developper can not be verified. After clicking either 'show in finder' or 'okay', it opens the plugin in a form without its GUI and showing that it is not licensed (even though it is). It also displays error code 100001. I have tried only some basic stuff to troubleshoot like restarting the DAW/my computer and reinstalling/relicensing the softwares. I don't know if the macOS version has anything to do with it but for some reason I just can't get it to work.

Is it possible to find IDR frame (CMSampleBuffer) in AVAsset h264 video file?

Hello everyone, I’m new to Swift development and have been working on an audio module that plays a specific sound at regular intervals - similar to a workout timer that signals switching exercises every few minutes. Following AVFoundation documentation, I’m configuring my audio session like this: let session = AVAudioSession.sharedInstance() try session.setCategory( .playback, mode: .default, options: [.interruptSpokenAudioAndMixWithOthers, .duckOthers] ) self.engine.attach(self.player) self.engine.connect(self.player, to: self.engine.outputNode, format: self.audioFormat) try? session.setActive(true) When it’s time to play cues, I schedule playback on a DispatchQueue: // scheduleAudio uses DispatchQueue self.scheduleAudio(at: interval.start) { do { try audio.engine.start() audio.node.play() for sample in interval.samples { audio.node.scheduleBuffer(sample.buffer, at: AVAudioTime(hostTime: sample.hostTime)) } } catch { print("Audio activation failed: \(error)") } } This works perfectly in the foreground. But once the app goes into the background, the scheduled callback runs, yet the audio engine fails to start, resulting in an error with code 561015905. Interestingly, if the app is already playing audio before going to the background, the scheduled sounds continue to play as expected. I have added the required background audio mode to my Info plist file by including the key UIBackgroundModes with the value audio. Is there anything else I should configure? What is the best practice to play periodic audio when the app runs in the background? How do apps like turn-by-turn navigation handle continuous audio playback in the background? Any advice or pointers would be greatly appreciated!

I develop a application with an uvc camera, this camera is a webcam, I use the AVFoundation library ,but when I run the code "[self.mCaptureSession startRunning]" ,I can not get the buffer, I already set the delegate, any answer will help.