Hi all, i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this?

Hello, I'm evaluating the Apple Music Feed dataset and I noticed that the total number of songs available in the feed is too small. As of today, the number of objects returned in each feed is: 51,198,712 albums 23,093,698 artists 173,235,315 songs This gives an average of 3.38 songs per album which is quite low. Also, iterating on the data I see that there are albums referencing songs that don't exist in the songs feed. I would like to know: Is the feed data incomplete? If so, in what situations an object may be missing from the feed? Thank you in advance!

My app Balletrax is a music player for people to use while they teach ballet. Used to be you could silence notifications during use, but now the customer seems to have to know how to use Focus mode, remember to turn it on and off, and have to check the notifications one does and doesn't want to use. Is there no way to silence all notifications when the app is in use?

Please consider adding the ability to programatically download Premium and Enhanced voices. At the moment it is extremely inconvenient for our users, as they have to navigate to settings themselves to download voices. Our app relies heavily on SpeechSynthesis integration, and it would greatly benefit from this feature. FB16307193

I wrote a Swift macOS app to control a PCI audio device. The code switches between the default output and input channels. As soon as I launch the Audio-Midi Setup utility, channel switching stops working. The driver properties allow switching, but the system doesn't respond. I have to delete the contents of /Library/Preferences/Audio and reset Core Audio. What am I missing? func setDefaultChannelsOutput() { guard let deviceID = getDeviceIDByName(deviceName: "PCI-424") else { return } let selectedIndex = DefaultChannelsOutput.indexOfSelectedItem if selectedIndex < 0 || selectedIndex >= 24 { return } let channel1 = UInt32(selectedIndex * 2 + 1) let channel2 = UInt32(selectedIndex * 2 + 2) var channels: [UInt32] = [channel1, channel2] var propertyAddress = AudioObjectPropertyAddress( mSelector: kAudioDevicePropertyPreferredChannelsForStereo, mScope: kAudioDevicePropertyScopeOutput, mElement: kAudioObjectPropertyElementWildcard ) let dataSize = UInt32(MemoryLayout<UInt32>.size * channels.count) let status = AudioObjectSetPropertyData(deviceID, &propertyAddress, 0, nil, dataSize, &channels) if status != noErr { print("Error setting default output channels: \(status)") } }

It sounds simple but searching for the name "Favorite Songs" is a non-starter because it's called different names in different countries, even if I specify "&l=en_us" on the query. So is there another property, relationship or combination thereof which I can use to tell me when I've found the right playlist? Properties I've looked at so far: canEdit: will always be false so narrows things down a little inFavorites: not helpful as it depends on whether the user has favourite the favourites playlist, so not relevant hasCatalog: seems always true so again may narrow things down a bit isPublic: doesn't help Adding the catalog relationship doesn't seem to show anything immediately useful either. Can anyone help? Ideally I'd like to see this as a "kind" or "type" as it has different properties to other playlists, but frankly I'll take anything at this point.

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?

Hi everyone, I'm running into an issue with AVAudioRecorder when handling interruptions such as phone calls or alarms. Problem: When the app is recording audio and an interruption occurs: I handle the interruption with audioRecorder?.pause() inside AVAudioSession.interruptionNotification (on .began). On .ended, I check for .shouldResume and call audioRecorder?.record() again. The recorder resumes successfully, but only the audio recorded after the interruption is saved. The audio recorded before the interruption is lost, even though I'm using the same file URL and not recreating the recorder. Repro: Start a recording with AVAudioRecorder Simulate a system interruption (e.g., incoming call) Resume recording after the interruption Stop and inspect the output audio file Expected: Full audio (before and after interruption) should be saved. Actual: Only the audio after interruption is saved; the earlier part is missing Notes: According to the documentation, calling .record() after .pause() should resume recording into the same file. I confirmed that the file URL does not change, and I do not recreate the recorder instance. No error is thrown by the system during this process. This behavior happens consistently when the app is interrupted and resumed. Question: Is this a known issue? Is there a recommended workaround for preserving the full recording when interruptions happen? Thanks in advance!

In my app I use AVAssetReaderTrackOutput to extract PCM audio from a user-provided video or audio file and display it as a waveform. Recently a user reported that the waveform is not in sync with his video, and after receiving the video I noticed that the waveform is in fact double as long as the video duration, i.e. it shows the audio in slow-motion, so to speak. Until now I was using CMFormatDescription.audioStreamBasicDescription.mSampleRate which for this particular user video returns 22'050. But in this case it seems that this value is wrong... because the audio file has two audio channels with different sample rates, as returned by CMFormatDescription.audioFormatList.map({ $0.mASBD.mSampleRate }) The first channel has a sample rate of 44'100, the second one 22'050. If I use the first sample rate, the waveform is perfectly in sync with the video. The problem is given by the fact that the ratio between the audio data length and the sample rate multiplied by the audio duration is 8, double the ratio for the first audio file (4). In the code below this ratio is given by Double(length) / (sampleRate * asset.duration.seconds) When commenting out the line with the sampleRate variable definition in the code below and uncommenting the following line, the ratios for both audio files are 4, which is the expected result. I would expect audioStreamBasicDescription to return the correct sample rate, i.e. the one used by AVAssetReaderTrackOutput, which (I think) somehow merges the stereo tracks. The documentation is sparse, and in particular it’s not documented whether the lower or higher sample rate is used; in this case, it seems like the higher one is used, but audioStreamBasicDescription for some reason returns the lower one. Does anybody know why this is the case or how I should extract the sample rate of the produced PCM audio data? Should I always take the higher one? I created FB19620455. let openPanel = NSOpenPanel() openPanel.allowedContentTypes = [.audiovisualContent] openPanel.runModal() let url = openPanel.urls[0] let asset = AVURLAsset(url: url) let assetTrack = asset.tracks(withMediaType: .audio)[0] let assetReader = try! AVAssetReader(asset: asset) let readerOutput = AVAssetReaderTrackOutput(track: assetTrack, outputSettings: [AVFormatIDKey: Int(kAudioFormatLinearPCM), AVLinearPCMBitDepthKey: 16, AVLinearPCMIsBigEndianKey: false, AVLinearPCMIsFloatKey: false, AVLinearPCMIsNonInterleaved: false]) readerOutput.alwaysCopiesSampleData = false assetReader.add(readerOutput) let formatDescriptions = assetTrack.formatDescriptions as! [CMFormatDescription] let sampleRate = formatDescriptions[0].audioStreamBasicDescription!.mSampleRate //let sampleRate = formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate }).max()! print(formatDescriptions[0].audioStreamBasicDescription!.mSampleRate) print(formatDescriptions[0].audioFormatList.map({ $0.mASBD.mSampleRate })) if !assetReader.startReading() { preconditionFailure() } var length = 0 while assetReader.status == .reading { guard let sampleBuffer = readerOutput.copyNextSampleBuffer(), let blockBuffer = sampleBuffer.dataBuffer else { break } length += blockBuffer.dataLength } print(Double(length) / (sampleRate * asset.duration.seconds))

According to the header file the outputVolume properties supported range is 0.0-1.0: /*! @property outputVolume @abstract The mixer's output volume. @discussion This accesses the mixer's output volume (0.0-1.0, inclusive). @property (nonatomic) float outputVolume; However when setting the volume to 2.0 the audio does indeed play louder. Is the header file out of date and if so, what is the supported range for outputVolume? Thanks

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 work on an audio app for iOS which is successfully using the MPRemoteCommandCenter for commands like next, back, skip forward, skip backward etc. I am trying to implement playback rate controls in my app (so that users can change the playback speed of audio to 0.5x or 2x for example). While the above commands work, the changePlaybackRateCommand does not seem to. I have enabled the command, given it a target/handler and set supported rates. With the other commands, this caused the UI to change on lock screen, in command center etc, by adding the control for the command (a next button for the next command for example). However, it does not seem to do anything for the playback rate command. I can implement my own "rate button" UI and rate change handling, but I'm wondering if this is a known bug within Apple? Looking online, it seems other people face the same issue and haven't been able to get this command to work. Why is this API provided if it doesn't seem to do anything? Is there something I'm missing? Kind regards.

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

Hello, i can successfully match music using shazamkit on Apple using SwiftUI, a simple app that let user to load an audio file and exctracts the relative match, while i am unable to match music using shamzamkit on Android. I am trying to make the same simple app but i cannot match music as i get MATCH_ATTEMPT_FAILED every time i try to. I don't know what i am doing wrong but the shazam part in the kotlin Android code is in this method : suspend fun processAudioFileInBackground( filePath: String, developerTokenProvider: DeveloperTokenProvider ) = withContext(Dispatchers.IO) { val bufferSize = 1024 * 1024 val audioFile = FileInputStream(filePath) val byteBuffer = ByteBuffer.allocate(bufferSize) byteBuffer.order(ByteOrder.LITTLE_ENDIAN) var bytesRead: Int while (audioFile.read(byteBuffer.array()).also { bytesRead = it } != -1) { val signatureGenerator = (ShazamKit.createSignatureGenerator(AudioSampleRateInHz.SAMPLE_RATE_44100) as ShazamKitResult.Success).data signatureGenerator.append(byteBuffer.array(), bytesRead, System.currentTimeMillis()) val signature = signatureGenerator.generateSignature() println("Signature: ${signature.durationInMs}") val catalog = ShazamKit.createShazamCatalog(developerTokenProvider, Locale.ENGLISH) val session = (ShazamKit.createSession(catalog) as ShazamKitResult.Success).data val matchResult = session.match(signature) println("MatchResult : $matchResult") setMatchResult(matchResult) byteBuffer.clear() } audioFile.close() } I noticed that changing Locale in catalog creation results in different result as i get NoMatch without exception. Can you please help me with this? Do i need to create a custom catalog?

Hi. I am working on an audio app for iOS. I have implemented UI and handling which allows the user to change playback rate of audio. When the user selects a different rate, I update the rate property on my AVQueuePlayer. This is working well on device. When I use Airplay, it works for some devices and not for others. Some devices won't change playback rate and will always play at 1x speed. Is this possibly a limitation of those 3rd-party devices? Or is there something I'm missing/should check? Would love to get playback rate changes working across all Airplay devices with our app. Kind regards.

Hello! I'm experiencing an issue with iOS's audio routing system when trying to use Bluetooth headphones for audio output while also recording environmental audio from the built-in microphone. Desired behavior: Play audio through Bluetooth headset (AirPods) Record unprocessed environmental audio from the iPhone's built-in microphone Actual behavior: When explicitly selecting the built-in microphone, iOS reports it's using it (in currentRoute.inputs) However, the actual audio data received is clearly still coming from the AirPods microphone The audio is heavily processed with voice isolation/noise cancellation, removing environmental sounds Environment Details Device: iPhone 12 Pro Max iOS Version: 18.4.1 Hardware: AirPods Audio Framework: AVAudioEngine (also tried AudioQueue) Code Attempted I've tried multiple approaches to force the correct routing: func configureAudioSession() { let session = AVAudioSession.sharedInstance() // Configure to allow Bluetooth output but use built-in mic try? session.setCategory(.playAndRecord, options: [.allowBluetoothA2DP, .defaultToSpeaker]) try? session.setActive(true) // Explicitly select built-in microphone if let inputs = session.availableInputs, let builtInMic = inputs.first(where: { $0.portType == .builtInMic }) { try? session.setPreferredInput(builtInMic) print("Selected input: \(builtInMic.portName)") } // Log the current route let route = session.currentRoute print("Current input: \(route.inputs.first?.portName ?? "None")") // Configure audio engine with native format let inputNode = audioEngine.inputNode let nativeFormat = inputNode.inputFormat(forBus: 0) inputNode.installTap(onBus: 0, bufferSize: 1024, format: nativeFormat) { buffer, time in // Process audio buffer // Despite showing "Built-in Microphone" in route, audio appears to be // coming from AirPods with voice isolation applied - welp! } try? audioEngine.start() } I've also tried various combinations of: Different audio session modes (.default, .measurement, .voiceChat) Different option combinations (with/without .allowBluetooth, .allowBluetoothA2DP) Setting session.setPreferredInput() both before and after activation Diagnostic Observations When AirPods are connected: AVAudioSession.currentRoute.inputs correctly shows "Built-in Microphone" after setPreferredInput() The actual audio data received shows clear signs of AirPods' voice isolation processing Background/environmental sounds are actively filtered out... When recording a test audio played near the phone (not through the app), the recording is nearly silent. Only headset voice goes through. Questions Is there a workaround to force iOS to actually use the built-in microphone while maintaining Bluetooth output? Are there any lower-level configurations that might resolve this issue? Any insights, workarounds, or suggestions would be greatly appreciated. This is blocking a critical feature in my application that requires environmental audio recording while providing audio feedback through headphones 😅

In our app we have implemented a AVAssetResourceLoaderDelegate to handle encrypted downloaded files. We have it working on all iOS versions but we are seeing issues on iOS 15 (15.8.3) with large files (> 1 Gb). We have so far seen two cases where either the load method on the AVURLAsset fails early and throws an unknown error code or starts requesting more data than the device has available RAM. The CPU usage is almost always over 100%, even after pausing playback. The memory issue can happen even though the player has successfully started playback. When running this on devices running iOS 16 and above we set the isEntireLengthAvailableOnDemand to true on the AVAssetResourceLoadingContentInformationRequest. This seems to be key to solving the issue those devices that support it. If we set the property to false we see the same memory issue as on iOS 15. So we have a solution for iOS 16 and upwards but are at a loss for how to handle iOS 15. Is there something we have overlooked or is it in fact an issue with that iOS version?

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.

I developed an educational app that implements audio-video communication through RTC, while using WebView to display course materials during classes. However, some users are experiencing an issue where the audio playback from WebView is very quiet. I've checked that the AVAudioSessionCategory is set by RTC to AVAudioSessionCategoryPlayAndRecord, and the AVAudioSessionCategoryOption also includes AVAudioSessionCategoryOptionMixWithOthers. What could be causing the WebView audio to be suppressed, and how can this be resolved?

I'm encountering numerous crashes involving the com.apple.coreaudio.AQClient thread on our application. The crash details are as follows: #10 com.apple.coreaudio.AQClient SIGSEGV SEGV_ACCERR 0 libobjc.A.dylib _objc_msgSend + 44 1 AudioToolbox ClientMessageHandler::PropertyChanged(unsigned int) + 872 2 AudioToolbox ClientAudioQueue::FetchAndDeliverPendingCallbacks(unsigned int) + 924 3 AudioToolbox __XCallbackNotificationsAvailable + 212 4 libAudioToolboxUtility.dylib _mshMIGPerform + 260 5 CoreFoundation ___CFRUNLOOP_IS_CALLING_OUT_TO_A_SOURCE1_PERFORM_FUNCTION__ + 56 6 CoreFoundation ___CFRunLoopDoSource1 + 596 7 CoreFoundation ___CFRunLoopRun + 2392 8 CoreFoundation _CFRunLoopRunSpecific + 572 9 AudioToolbox CADeprecated::GenericRunLoopThread::Entry(void*) + 156 10 libAudioToolboxUtility.dylib CADeprecated::CAPThread::Entry(CADeprecated::CAPThread*) + 88 11 libsystem_pthread.dylib __pthread_start + 116 All these crashes occur on system versions below iOS/iPadOS 17, primarily when the device's available RAM is low. What steps can I take to resolve this issue? Any insights would be greatly appreciated!