I’m building a music app using Apple Music streaming via ApplicationMusicPlayer. My goal is to decrease the volume of the current song during the last 10 seconds, and when the next track begins, restore the volume to its normal level. I know that ApplicationMusicPlayer doesn’t expose a volume API, and I want to avoid triggering the system volume HUD. ✅ Using Apple Music streaming (not local files) ❓ Is it possible to implement per-track fade-out/fade-in logic with ApplicationMusicPlayer? Appreciate any clarification or official guidance!

We are currently working on a CarPlay navigation app and so far everything is working well except for speaking turn notifications. Our TTS implementation works fine on the phone and works fine on CarPlay if the voice is spoken over the speaker in the car. If users connect a BT headset to the car and listen through that headset, then the voice commands are chopped up / stutter. Why would users use BT headset? Well, we are working on a motorcycle app, and there are no speakers usually on a motorcycle. It sounds like the BT channel is opened and closed repeatedly for every character / word spoken. This happens on different CarPlay devices and different Bluetooth headsets, we have reports from multiple users that they find this behavior annoying and that other apps work fine. Is this a known issue? Are there possible workaround?

I’m building a music app using Apple Music streaming via ApplicationMusicPlayer. My goal is to decrease the volume of the current song during the last 10 seconds, and when the next track begins, restore the volume to its normal level. I know that ApplicationMusicPlayer doesn’t expose a volume API, and I want to avoid triggering the system volume HUD. ✅ Using Apple Music streaming (not local files) ❓ Is it possible to implement per-track fade-out/fade-in logic with ApplicationMusicPlayer? Appreciate any clarification or official guidance!

Dear Sirs, I’ve written a virtual audio driver based on AudioDriverKit and running as dext in my MacOS app. Sometimes when waking up from a sleep state the recording side of my driver extension seems to hang and I don’t see any calls to my io_operation callback. Then the recording app like a DAW seems to hang when trying to start a recording. This doesn’t happen after short sleep states or after a complete new start of my MacBook. I already opened a case in Feedback-Assistant on 5th of May (FB17503622) which also includes a sysdiagnose and a ktrace but I didn't get any feedback so far. Meanwhile some of our customers are getting angry and I'd like to know if there's anything I could do to fix this problem on my side. We’re not sure whether this worked in previous MacOS versions, we think we didn’t observe this before 15.3.1 but at least since 15.3.1. we’ve seen this problem. Best regards, Johannes

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!}

I use the Apple Music API to poll my listening history at regular intervals. Every morning between 5:30AM and 7:30AM, I observe a strange pattern in the API responses. During this window, one or more of the regular polling intervals returns a response that differs significantly from the prior history response, even though I had no listening activity at that time. I'm using this endpoint: https://api.music.apple.com/v1/me/recent/played/tracks?types=songs,library-songs&include[library-songs]=catalog&include[songs]=albums,artists Here’s a concrete example from this morning: Time: 5:45AM Fetch 1 Tracks (subset): 1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ... Time: 5:50AM Fetch 2 Tracks (subset): 1799261990, 1739657416, 1786317143, 1623924746, 1635185172, 1574004238, 1198763630, 1621299055, ... Time: 5:55AM Fetch 3 Tracks (subset): 1799261990, 1739657416, 1786317143, 1784288789, 1743250261, 1738681804, 1789325498, 1743036755, ... At 5:50, a materially different history is returned, then it returns back to the prior history at the next poll. I've listened to all of the tracks in each set, but the 5:50 history drops some tracks and returns some from further back in history. I've connected other accounts and the behavior is consistent and repeatable every day across them. It appears the API is temporarily returning a different (possibly outdated or cached?) view of the user's history during that early morning window. Has anyone seen this behavior before? Is this a known issue with the Apple Music API or MusicKit backend? I'd love any insights into what might cause this, or recommendations on how to work around it.

We are seeing logs were on iOS devices we see some keyframes request. but on safari browser don’t see any request like this. could you please explain what is it. /d8ceb9244ff889b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/d8ceb9244ff88e9b42b82eb807327531-c27dbcb10e0bbf3cde6c-1/keyframes/hls/.

Hi, I believe I've found a potential error in the sample code on the documentation page for creating and using a process tap with an aggregate device. The issue is in the section explaining how to add a tap to the aggregate device. I have already filed a Feedback Assistant ticket on this (ID: FB17411663) but haven't heard back for months. Capturing system audio with Core Audio taps The sample code for modifying the kAudioAggregateDevicePropertyTapList incorrectly uses the tapID as the target AudioObjectID when calling AudioObjectSetPropertyData. // (Code to get the list and potentially modify listAsArray) if var listAsArray = list as? [CFString] { // ... (modification logic) ... // Set the list back on the aggregate device. <--- The comment is correct list = listAsArray as CFArray _ = withUnsafeMutablePointer(to: &list) { list in // INCORRECT: This call uses tapID as the target object. AudioObjectSetPropertyData(tapID, &propertyAddress, 0, nil, propertySize, list) } } The kAudioAggregateDevicePropertyTapList is a property that belongs to the aggregate device, not the individual tap. Therefore, to set this property, the AudioObjectSetPropertyData function must target the AudioObjectID of the aggregate device itself. Using tapID as the first argument is logically incorrect for this operation and will not update the aggregate device as intended. Furthermore, the preceding AudioObjectGetPropertyData call to fetch the list also appears to use the incorrect tapID as its target in the sample. The AudioObjectID for both getting and setting this property should be the ID of the aggregate device. _ = AudioObjectGetPropertyData(aggregateDeviceID, &propertyAddress, 0, nil, &propertySize, &list) _ = AudioObjectSetPropertyData(aggregateDeviceID, &propertyAddress, 0, nil, propertySize, newList) Thank you!

I’m spot-checking some of the data I’m extracting from the Apple Music Feed parquet files and finding numerous issues of invalid album IDs. For example, just looking at any album with a primary artist id of 163043, I see a few albums that are not available at music.apple.com/us/album/NNN. These include: 981094158 - Time and the River 1803443737 - Celebration (Live New York ’80) 1525426873 - Anything You Want: The Warner-Reprise-Elektra Years I notice that neither of these album ids are returned using the general Music API, so I’m a bit confused why they would exist in the parquet files at all. Thanks.

We have a React website build to scan qr codes. The website is properly working for Android devices but for Iphone we see a camera glitch causing delay in scan which is unexpected. Website URL : https://react-qr-code-scanner-app.vercel.app/

Hi, I'm trying to plan out development of an app and am wondering if it is possible to have user generated content automatically populate into a custom shazamkit catalogue and be able to query this catalogue non-locally? Storing all the submissions locally would obviously not scale.

When making a call to https://api.music.apple.com/v1/me/library/artists to get a user's library artists, it returns the following (as an example): [ { id: 'r.FCwruQb', type: 'library-artists', href: '/v1/me/library/artists/r.FCwruQb?l=en-US', attributes: { name: 'A Great Big World' } }, { id: 'r.7VSWOgj', type: 'library-artists', href: '/v1/me/library/artists/r.7VSWOgj?l=en-US', attributes: { name: 'Aaliyah' } }, ... ] If I try and use an artist id from that retuned data to look up additional information about the artist by calling https://api.music.apple.com/v1/catalog/us/artists/{id}, it fails. User Library Artists don't seem to equal Catalog Artists. It'd be great if there was a way to use these interchangeably. Am I missing something?

We are facing a strange issue where a small portion of our large userbase can not start the capture session in our app, as it gets interrupted with the following reason: AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableWithMultipleForegroundApps Our users are all from iPhones, no one is using an iPad. Just to be sure we have set session.isMultitaskingCameraAccessEnabled = true but it does not seem to make any difference. Another weird interruption we are seeing

We encounter issue with avplayer in case of EXT-X-DISCONTINUITY misalignment between audio and video produced after insertion of gaps. The initial objective is to introduce an EXT-X-DISCONTINUITY in audio playlist after some missing segments (EXT-X-GAP) which durations are aligned to video segments durations, to handle irregular audio durations. Please find below an example of corresponding video and audio playlists: video: #EXTM3U #EXT-X-VERSION:7 #EXT-X-MEDIA-SEQUENCE:872524632 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-TARGETDURATION:2 #USP-X-TIMESTAMP-MAP:MPEGTS=7096045027,LOCAL=2025-05-09T12:38:32.369100Z #EXT-X-MAP:URI="hls/StreamingBasic-video=979200.m4s" #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.369111Z #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524632.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524633.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524634.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524635.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524636.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524637.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524638.m4s #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.383111Z #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524639.m4s #EXTINF:2.002, no desc hls/StreamingBasic-video=979200-872524640.m4s audio: EXTM3U #EXT-X-VERSION:7 #EXT-X-MEDIA-SEQUENCE:872524632 #EXT-X-INDEPENDENT-SEGMENTS #EXT-X-TARGETDURATION:2 #USP-X-TIMESTAMP-MAP:MPEGTS=7096045867,LOCAL=2025-05-09T12:38:32.378400Z #EXT-X-MAP:URI="hls/StreamingBasic-audio_99500_eng=98800.m4s" #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:32.378444Z #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524632.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524633.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524634.m4s #EXTINF:1.984, no desc hls/StreamingBasic-audio_99500_eng=98800-872524635.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524636.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-audio_99500_eng=98800-872524637.m4s ## Media sequence discontinuity #EXT-X-GAP #EXTINF:2.002, no desc hls/StreamingBasic-audio_99500_eng=98800-872524638.m4s #EXT-X-DISCONTINUITY #EXT-X-PROGRAM-DATE-TIME:2025-05-09T12:38:46.778444Z #EXTINF:1.6213, no desc hls/StreamingBasic-audio_99500_eng=98800-872524639.m4s #EXTINF:2.0053, no desc hls/StreamingBasic-audio_99500_eng=98800-872524640.m4s In this case playback is broken with avplayer. Is it conformed to Http Live Streaming? Is it an avplayer bug? What are the guidelines to handle such gaps?

Hey, Quick question. I noticed that Adobe's new app, Project Indigo, allows you to open the app using the Camera Control button. However, when your device is locked it just shows this screen: Would this normally be approved by the Appstore approval process? I ask because I would like to do something similar with my camera app. I know that this is not the best user experience, but my apps UI is not built in Swift and I don't have the resources to build the UI again. At least this way the user experience would be improved from what it is now, where users cannot even launch the app. I get many requests per week about this feature and would love to improve the UX for my users, even if it's not the best possible. Thanks, Alex

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

Hi everyone, I’m trying to use AVAssetResourceLoaderDelegate to handle a live radio stream (e.g. Icecast/HTTP stream). My goal is to have access to the last 30 seconds of audio data during playback, so I can analyze it for specific audio patterns in near-real-time. I’ve implemented a custom resource loader that works fine for podcasts and static files, where the file size and content length are known. However, for infinite live streams, my current implementation stops receiving new loading requests after the first one is served. As a result, the playback either stalls or fails to continue. Has anyone successfully used AVAssetResourceLoaderDelegate with a continuous radio stream? Or maybe you can suggest betterapproach for buffering and analyzing live audio? Any tips, examples, or advice would be appreciated. Thanks!

On Apple TV 4K 3rd generation, with tvOS 26 beta 2, when two HomePod 2 are paired to the device, music and movie sources with Dolby Atmos can only be listened to in stereo. dolby atmos not supported

Is there a way to permanently disable PHASE SDK logging? It seems to be a lot chattier than Apple's other SDKs. While developing a RealityKit app that uses AudioPlaybackController, I must manually hide the PHASE SDK log output several times each day so I can see my app's log messages. Thank you.

AVCaptureVideoDataOutput.preparesCellularRadioForNetworkConnection requires com.apple.developer.avfoundation.video-data-output-prepares-cellular-radio-for-machine-readable-code-scanning. But I cannot acquire its entitlement. I can't find its entitlement on 'Certificates, Identifiers &amp; Profiles'. Any solutions? Provisioning profile "iOS Team Provisioning Profile: ......" doesn't include the com.apple.developer.avfoundation.video-data-output-prepares-cellular-radio-for-machine-readable-code-scanning entitlement.