Hi,It seems like it's pretty easy to consume HTTP Live Streaming content in an iOS app. Unfortunately, I need to consume media from an RTSP server. It seems to me that this is a very similar thing, and that all of the underpinnings for doing it ought to be present in iOS, but I'm having a devil of a time figuring out how to make it work without doing a lot of programming.For starters, I know that there are web-based services that can consume an RTSP stream and rebroadcast it as an HTTP Live Stream that can be easily consumed by the media players in iOS. This won't work for me because my application needs to function in an environment where there is no internet access (it's on a private Wifi network where the only other thing on the network is the device that is serving the RTSP stream).Having read everything I can get my hands on and exploring third-party and open-source solutions, I've compiled the following list of ideas:1. Using an iOS build of the open-source ffmpeg library, which supports RTSP, I've come up with a test app that can receive the RTSP packets, decode them, create UIImages out of the frames, and display those frames on-screen. This provides a crude player, but performance is poor, most likely because ffmpeg can't take advantage of any hardware acceleration. It also doesn't provide me with any way to integrate the video stream into AVFoundation, so I'm on my own as far as saving the stream to a file, transcoding it, etc.2. I know that the AVURLAsset class doesn't directly support the RTSP scheme. Since I have access to the undecoded RTSP packets via ffmpeg, I've thought it should be possible to implement RTSP support myself via a custom NSURLProtocol, essentially fooling AVFoundation into reading those packets as if they originated in a file. I'm not sure if this would work, since the raw packets coming from the RTSP server might lack the headers that would otherwise be present in data being read from a file. I'm not even sure if AVFoundation would recognize my custom protocol.3. If a protocol doesn't work, I've considered that I might be able to implement my own local HTTP Live Streaming server that converts the RTSP packets into an HTTP stream that the media players can read. This sounds like a terribly convoluted solution to the problem, at best, and very difficult at worst.4. Going back to solution (1), if I could speed up the decoding by using some iOS CoreVideo function instead of ffmpeg, this solution might be okay. However, I can't find any documentation for CoreVideo on iOS (Apple only documents it for OS X).5. I'm certainly willing to license a third-party solution if it works well and provides good performance. Unfortunately, everything I've found so far is pretty crummy and mostly just leverages ffmpeg and/or VLC. What is most disappointing to me is that nobody seems to be able or willing to provide a solution that neatly integrates with AVFoundation. I really want to make my RTSP stream available as an AVAsset so I can use it with AVFoundation players and other classes -- I don't want to build an app that relies on custom third-party code for everything.Any ideas, tips, advice would be greatly appreciated.Thanks,Frank

Hi everyone, I am currently on MacOS Tahoe (26.1), and for some weird reason my mac is not connecting via HDMI. To be accurate: it is connecting and the LG TV shows up in the Displays settings, but no image shows up in it, I have no idea why. This used to work as I've tried this cable before with the same exact tv. The cable is a basic Amazon Basics HDMI one. Allow me just to advanced this question a little: usually terminal commands are more advanced recommendations, whereas basic questions like "have you connected it right" are just a waste of time

I'm wondering if someone happened issues with currentPlaybackRate in released version of iOS 26.0? There is no issue happened in case of iOS 18.5. When I changed currentPlaybackRate on iOS 26.0, it seems to unexpectedly change currentPlaybackRate to be 0 or 1.0 forcibly no matter DRM or non-DRM contents. And also, playing music will be abnormal behavior and unstable with noise if currentPlaybackRate is not 1.0. And changes stop state and play state frequently.

I've been wondering if there is a way to modify or even disable tones for indicating channel states. The behaviour regarding tones seems like a black box with little documentation. During migration to Apple's PT Framework we've noticed that there are few scenarios where a tone is played which doesn't match certain certifications. For example; moving from a channel to another produces a tone which would fail a test case. I understand the reasoning fully, as it marks that the channel is ready to transmit or receive, but this doesn't mirror the behaviour of TETRA which would be wanted in this case. I'm also wondering if there would be any way to directly communicate feedback regarding PT Framework?

Hi everyone, After updating my Apple TV HD (model A1625) to tvOS 26, I’ve noticed a significant spike in CPU usage—up to 3× higher than before the update. Go from around 40% to 120% Model: Apple TV HD (A1625) tvOS Version: 26 (stable release) and beta version of 26.1, App downgrade stream due to lack of cpu power If anyone else is experiencing this, please share your findings or workarounds. Would love to hear from Apple engineers or other developers if this is a known regression or if there’s a recommended fix. Thanks!

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?

The same H265 encrypted Fairplay content can be played in all Apple devices except A1625. The clear H265 content is played in A1625. The question is: will this model (A1625) support H265 Fairplay encrypted content? A ticket was created here: https://discussions.apple.com/thread/255658006?sortBy=best

Hi, I'm working an a video editing software that lets you composite and export videos. I use a custom compositor to apply my effects etc. In my crash dashboard, I am seeing a report of an EXC_BAD_ACCESS crash from objc_msgSend. Below is the stacktrace. libobjc.A.dylib objc_msgSend libdispatch.dylib _dispatch_sync_invoke_and_complete_recurse libdispatch.dylib _dispatch_sync_f_slow [symbolication failed] libdispatch.dylib _dispatch_client_callout libdispatch.dylib _dispatch_lane_barrier_sync_invoke_and_complete AVFCore -[AVCustomVideoCompositorSession(AVCustomVideoCompositorSession_FigCallbackHandling) _customCompositorShouldCancelPendingFrames] AVFCore _customCompositorShouldCancelPendingFramesCallback MediaToolbox remoteVideoCompositor_HandleVideoCompositorClientMessage CoreMedia __figXPCConnection_CallClientMessageHandlers_block_invoke libdispatch.dylib _dispatch_call_block_and_release libdispatch.dylib _dispatch_client_callout libdispatch.dylib _dispatch_lane_serial_drain libdispatch.dylib _dispatch_lane_invoke libdispatch.dylib _dispatch_root_queue_drain_deferred_wlh libdispatch.dylib _dispatch_workloop_worker_thread libsystem_pthread.dylib _pthread_wqthread libsystem_pthread.dylib start_wqthread What stood out to me is that this is only being reported from IOS 26.0+ devices. A part of the stacktrace failed to be symbolicated [symbolication failed]. I'm 90% confident that this is Apple code, not my app's code. I cannot reproduce this locally. Is this a known issue? What are the possible root-causes, and how can I verify/eliminate them? Thanks,

Hi, I'm trying to setup a AVAudioEngine for USB Audio recording and monitoring playthrough. As soon as I try to setup playthough I get an error in the console: AVAEInternal.h:83 required condition is false: [AVAudioEngineGraph.mm:1361:Initialize: (IsFormatSampleRateAndChannelCountValid(outputHWFormat))] Any ideas how to fix it? // Input-Device setzen try? setupInputDevice(deviceID: inputDevice) let input = audioEngine.inputNode // Stereo-Format erzwingen let inputHWFormat = input.inputFormat(forBus: 0) let stereoFormat = AVAudioFormat(commonFormat: inputHWFormat.commonFormat, sampleRate: inputHWFormat.sampleRate, channels: 2, interleaved: inputHWFormat.isInterleaved) guard let format = stereoFormat else { throw AudioError.deviceSetupFailed(-1) } print("Input format: \(inputHWFormat)") print("Forced stereo format: \(format)") audioEngine.attach(monitorMixer) audioEngine.connect(input, to: monitorMixer, format: format) // MonitorMixer -> MainMixer (Output) // Problem here, format: format also breaks. audioEngine.connect(monitorMixer, to: audioEngine.mainMixerNode, format: nil)

Hello, I'm working on a Flutter app targeting both Android and iOS, where I implemented ShazamKit. In order to achieve that, I first tried with the flutter_shazam_kit package, but since it's not maintained anymore, I forked it here, and tried to update it to meet the Google Play Store requirements, as you can see here: https://github.com/mregnauld/flutter_shazam_kit/tree/fix-16k Unfortunately, after trying everything, my app still doesn't meet the (not so) new 16 KB native library alignment. Also, I'm 100% sure it comes from that because the error message disappears if I remove that package from my app. So after investigating, it seems that the problem comes from the ShazamKit for Android (that you can find here: https://developer.apple.com/download/all/?q=Android%20ShazamKit), and especially the .so files in the .aar file. Is there anything I can do to fix that, or should I wait before the ShazamKit team fix that? I'm totally stuck with that so any help is highly appreciated. Thanks.

Hey there, I just upgraded to Mac OS Tahoe ,son an apple MacBook Pro 2019 16inch. am using IntellijIDEA and Flutter to develop a mobile app which I test on the simulator app running iOS 18.4 . the issue: when I start the simulator app. ( while in the loading phase and in the operation phase as well ), the audio from an already open YouTube tab on safari (this happens on chrome browser as well). the sound glitches and becomes Noise. a fix I found online is to kill the audio deamon on Mac OS, This works using the command: "sudo killall coreaudiod" this kills the audio process, (while the emulator is operational), then the macOS restarts the audio deamon then the audio works fine alongside with the simulator being open. I just want to ask is there a permanent fix for this? is Apple working on a fix for this in the upcoming update?

I am trying to use SpeechTranscriber from Speech framework. Is it possible to use it on Simulator of iOS 26 (Mac OS Tahoe)? Function "supportedLocales" returns an empty array.

I am using https://developer.apple.com/documentation/applemusicapi/add-tracks-to-a-library-playlist to add tracks to playlists. This endpoint works fine for all playlists except for collaborative playlists. For collaborative playlist I get the following 500 error as a response: "errors": [ { "id": "<some id>", "title": "Upstream Service Error", "detail": "Unable to update tracks", "status": "500", "code": "50001" } ] } Steps to reproduce: Create a playlist in your library. Use the api to add a song. Confirm that it works. Make that same playlist collaborative. Update the playlist ID in your api request (as making a playlist collaborative changes its id) Confirm that you get the 500 error.

Just downloaded iOS 26.1 and my phone keeps ringing after the call has been answered. Any fixes for this?

We build mobile apps for creators to edit their videos. Post editing the video, the creator has to export the video so that it can be uploaded to Youtube. The export is a time consuming and GPU intensive process. The creator can exit the app due to various reasons like receiving the call, putting the app in background etc. This causes the export to fail :( Keeping this limitation in mind there was an announcement from Apple that with the IOS 26 launch would start to support background GPU access. Here is the official documentation: https://developer.apple.com/documentation/BundleResources/Entitlements/com.apple.developer.background-tasks.continued-processing.gpu When we tried using this feature, we were not able to get it to work on IOS 26. We stumbled upon this ticket(https://developer.apple.com/forums/thread/797538?answerId=854825022#854825022) in the Apple Developer forum, in which possibly an Apple engineer claims it is supported ONLY for iPadOS 26. This is a very big bummer for us. 96% of the users are on iPhone(compared to iPad), and if we refer to the official documentation above, it claims that this feature should work on IOS 26. This feature is extremely important for having the best user experience and reducing user frustration and will be useful for other video editing apps. Looking forward to a resolution.

Hi all, I have been quite stumped on this behavior for a little bit now, so thought it best to share here and see if someone more experience with AVAudioEngine / AVAudioSession can weigh in. Right now I have a AVAudioEngine that I am using to perform some voice chat with and give buffers to play. This works perfectly until route changes start to occur, which causes the AVAudioEngine to reset itself, which then causes all players attached to this engine to be stopped. Once a AVPlayerNode gets stopped due to this (but also any other time), all samples that were scheduled to be played then get purged. Where this becomes confusing for me is the completion handler gets called every time regardless of the sound actually being played. Is there a reliable way to know if a sample needs to be rescheduled after a player has been reset? I am not quite sure in my case what my observer of AVAudioEngineConfigurationChange needs to be doing, as this engine only handles output. All input is through a separate engine for simplicity. Currently I am storing a queue of samples as they get sent to the AVPlayerNode for playback, and after that completion checking if the player isPlaying or not. If it's playing I assume that the sound actually was played- and if not then I leave it in the queue and assume that an observer on the route change or the configuration change will realize there are samples in the queue and reset them Thanks for any feedback!

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 want the audio session to always use the built-in microphone. However, when using the setPreferredInput() method like in this example private func enableBuiltInMic() { // Get the shared audio session. let session = AVAudioSession.sharedInstance() // Find the built-in microphone input. guard let availableInputs = session.availableInputs, let builtInMicInput = availableInputs.first(where: { $0.portType == .builtInMic }) else { print("The device must have a built-in microphone.") return } // Make the built-in microphone input the preferred input. do { try session.setPreferredInput(builtInMicInput) } catch { print("Unable to set the built-in mic as the preferred input.") } } and calling that function once in the initializer, the audio session still switches to the external microphone once one is plugged in. The session's preferredInput is nil again at that point, even if the built-in microphone is still listed in availableInputs. So, why is the preferredInput suddenly reset? when would be the appropriate time to set the preferredInput again? Observing the session’s availableInputs did not work and setting the preferredInput again in the routeChangeNotification handler seems a bad choice as it’s already a bit too late then.

I have an iPadOS M-processor application with two different running configurations. In config1, the shared AVAudioSession is configured for .videoChat mode using the built-in microphone. The input/output nodes of the AVAudioEngine are configured with voice processing enabled. The built-in mic is formatted for 1 channel at 48KHz. In config2, the shared AVAudioSession is configured for .measurement mode using an external USB microphone. The input/output nodes of the AVAudioEngine are configured with voice processing disabled. The external mic is formatted for 2 channels at 44.1KHz I've written a configuration manager designed to safely switch between these two configurations. It works by stopping AVAudioEngine and detaching all but the input and output nodes, updating the shared audio session for the desired mic and sample-rates, and setting the appropriate state for voice processing to either true or false as required by the configuration. Finally the new audio graph is constructed by attaching appropriate nodes, connecting them, and re-starting AVAudioEngine I'm experiencing what I believe is a race-condition between switching voice processing on or off and then trying to re-build and start the new audio graph. Even though notifications, which are dumped to the console indicate that my requested input and sample-rate settings are in place, I crash when trying to start the audio engine because the sample-rate is wrong. Investigating further it looks like the switch from remote I/O to voice-processing I/O or vice-versa has not yet actually completed. I introduced a 100ms second delay and that seems to help but is obviously not a reliable way to build software that must work consistently. How can I make sure that what are apparently asynchronous configuration changes to the shared audio session and the input/output nodes have completed before I go on? I tried using route change notifications from the shared AVAudioSession but these lie. They say my preferred mic input and sample-rate setting is in place but when I dump the AVAudioEngine graph to the debugger console, I still see the wrong sample rate assigned to the input/output nodes. Also these are the wrong AU nodes. That is, VPIO is still in place when RIO should be, or vice-versa. How can I make the switch reliable without arbitrary time delays? Is my configuration manager approach appropriate (question for Apple engineers)?

My app has been using the iTunes Search API (itunes.apple.com/search) for a few years now, but at some point over the last week or so (late Sept. 2025) it is no longer returning track results with explicit content, regardless of whether I provide "explicit=Yes" (which is the default anyway, according to the API documentation - https://performance-partners.apple.com/search-api). Has anyone else experienced this with this API and have you figured out a workaround? FYI, I do also use the more robust Apple Music API in another part of my app, which isn't going through this issue, so I know it's technically an alternative. I just need to stick with iTunes Search API in this particular case. Thanks.