As of today (27 Aug), Xcode 27 has not yet been released - you can see the latest versions listed at xcodereleases.com. Regarding Default Actor Isolation, it is indeed available as a build setting. From my experience with Xcode 26, it needs to be explicitly set to MainActor if you want compiler enforcement. That’s also consistent with what was mentioned in the WWDC session.

That clears it up - I’ll try DictationTranscriber with the availability check. Thanks again for your help, really appreciate it!

Thanks, Quinn, for taking the time to share your candid thoughts. The reminder about Default Actor Isolation is noted. And enjoy the summer rides! 🚴

Thanks for the confirmation! I’ve filed the bug as FB19881107.

I’ve now upgraded Xcode to the latest available build: Xcode 26.0 beta 5 (17A5295f). This should, in theory, resolve the version mismatch issue you highlighted. Please let me know if I should expect to wait for beta 6 specifically, but since this is the latest release, I assume it covers the ABI/runtime symbol alignment. Current behavior: macOS (Intel MacBook Pro): The app launches successfully, and the window is displayed. Pressing the record button (⭘) results in console logs: localeNotSupported could not record: invalidAudioDataType Playback works (I can hear the recorded sound), but there is no transcription displayed and no highlighting. Play button plays back my speech. iOS (iPhone 11, iOS 26.0 beta): Same as macOS above, except playback volume is extremely faint, even at maximum system volume. Question regarding Dictation: I also tried enabling Edit → Start Dictation from the menu, and accepted the prompt. After that: I don’t see any option to disable it within the app. Turning off Dictation in System Settings → Keyboard → Dictation → Off does not stop the behavior triggered via the app (console logs indication). Could you clarify: Are these console errors (localeNotSupported, invalidAudioDataType) expected in beta 5, or is there a configuration workaround? Is the low playback volume on iOS a known issue? What is the correct way to disable Dictation once it has been enabled from the app menu? Thanks again for your guidance!

Hey @ondeviceguy , Your platform looks impressive and fills an important gap for evaluating model and app performance across devices. It’s clear this requires good investment in hardware and expertise, and I can see how it serves as a valuable tool for developers. A few considerations that might enhance its utility: Ensuring diverse input payloads to guarantee balanced and representative analysis. Clear guidance for integrating performance insights into early-stage design decisions to optimize resource consumption effectively. Highlighting any accreditations or standards compliance to build additional trust in the results provided. Currently, I'm in the app design phase, considering how hardware resource consumption fits into the architecture, especially when working with streams of frame buffers. Once the design decisions are made and the app is implemented with the models in action, I plan to consult this platform to further evaluate performance across devices. This will help ensure the app is optimized to run effectively on various hardware configurations. Thanks for sharing—this looks like a much-needed resource for the developer community!

Thank you, Greg, for the clarification. It's clear now that Apple's frameworks focus on providing streamlined functionality without exposing the underlying models for direct access or fine-tuning. This approach is understandable for maintaining consistency and security. For those looking to fine-tune models, it seems leveraging platforms like AWS for training/fine-tuning PyTorch models, followed by conversion to Core ML using coremltools, is a viable path. This aligns with what Benoit Dupin highlighted at re:Invent [1] about Apple's usage of AWS for pretaining. Of course, the conversion process with coremltools remains a critical step, and ensuring compatibility can sometimes feel like a gating factor. Thanks again for your insights—this helps close the thread on a clear note! [1] https://aws.amazon.com/events/aws-reinvent-2024-recap/ Best regards

I wanted to provide an update on this. I’ve made progress using the Core ML Tools documentation, specifically the Introductory Quickstart Guide. This has helped clarify the steps for converting TensorFlow models. I still need to further my understanding of the best way to reference and adapt YAMNet for conversion, but I’m on the right track. Since there hasn’t been much activity on this thread, I’ll close it for now. Thanks!

Hi Greg, Thank you for taking the time to respond and clarifying the capabilities of the built-in sound classifier. I now understand that fine-tuning the built-in classification model isn’t supported, and I appreciate you confirming that. Looking back, I realize my query was unclear because I mistakenly referred to "SNSoundClassifier" instead of the built-in pre-trained sound classifier. I got mixed up with the terms from the Sound Analysis framework, such as SNRequest, SNClassifySoundRequest, SNAudioStreamAnalyzer, SNResultsObserving, SNClassificationResult. Thank you for interpreting my question correctly despite the confusion. I’ll mark your reply as accepted, as it fully answers the specific question about fine-tuning the sound classifier. However, I have a broader question regarding Apple’s ML ecosystem: Are there any other models within Vision, Natural Language, or Speech APIs that can be fine-tuned or extended for custom use cases? I’m exploring the general feasibility of designing custom ML solutions within Apple’s ecosystem and would appreciate any guidance or insights. Thank you again for your help!

Highlighted difference indicating code used from the sample. --- > 68,69d67 < private let videoDataOutputQueue = DispatchQueue(label: "VideoDataOutput", qos: .userInitiated, attributes: [], autoreleaseFrequency: .workItem) < var bufferSize: CGSize = .zero --- > 102d99 < print("captureOutput error") 114,142d110 < < private func setForVideoClassify() { < print("setForVideoClassify()") < // Add the video data output for classification if it's not already added < let videoDataOutput = AVCaptureVideoDataOutput() < videoDataOutput.alwaysDiscardsLateVideoFrames = true < videoDataOutput.videoSettings = [ < kCVPixelBufferPixelFormatTypeKey as String: Int(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange) < ] < videoDataOutput.setSampleBufferDelegate(self, queue: videoDataOutputQueue) < < if captureSession.canAddOutput(videoDataOutput) { < captureSession.addOutput(videoDataOutput) < } else { < print("Could not add video data output") < } < let captureConnection = videoDataOutput.connection(with: .video) < // Always process the frames < captureConnection?.isEnabled = true < do { < let videoDevice = currentDevice < try videoDevice.lockForConfiguration() < let dimensions = CMVideoFormatDescriptionGetDimensions((videoDevice.activeFormat.formatDescription)) < bufferSize.width = CGFloat(dimensions.width) < bufferSize.height = CGFloat(dimensions.height) < videoDevice.unlockForConfiguration() < } catch { < print("setForVideoClassify error: \(error)") < } 144,145d111 < } < 262,265c228 < setupVision() < setForVideoClassify() < captureSession.sessionPreset = .vga640x480 < //captureSession.sessionPreset = .high --- > captureSession.sessionPreset = .high 526a490 > setForVideoClassify() 529a494,510 > private func setForVideoClassify() { > print("setForVideoClassify()") > // Add the video data output for classification if it's not already added > let videoDataOutput = AVCaptureVideoDataOutput() > videoDataOutput.alwaysDiscardsLateVideoFrames = true > videoDataOutput.videoSettings = [ > kCVPixelBufferPixelFormatTypeKey as String: Int(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange) > ] > videoDataOutput.setSampleBufferDelegate(self, queue: DispatchQueue(label: "VideoDataOutputQueue")) > > if captureSession.canAddOutput(videoDataOutput) { > captureSession.addOutput(videoDataOutput) > } else { > print("Could not add video data output") > } > } > 567a549 >

In WWDC 2024 session 10223, Apple demonstrated the availability of multiple Core ML modules, including Vision, Natural Language, Sound, Speech, and Translation. These modules can be leveraged concurrently within an app, allowing tasks to run in parallel. Could you provide guidance on best practices for performing multiple Core ML tasks simultaneously on iPhone and MacBook hardware? Specifically, I’d like to know about the feasibility and efficiency of running three parallel tasks on different Apple chipsets (A-series for iPhone and M-series for MacBook). Are there particular recommendations for optimizing parallel tasks on these chips to maintain performance and battery efficiency? Thank you!

Conversion Script: import tensorflow as tf import tensorflow_hub as hub import coremltools as ct import numpy as np # Load YAMNet model from TensorFlow Hub yamnet_layer = hub.KerasLayer("https://tfhub.dev/google/yamnet/1", trainable=False, dtype=tf.float32) # Define input shape and input layer with explicit dtype input_shape = (15600,) inputs = tf.keras.Input(shape=input_shape, name="audio_waveform", dtype=tf.float32) outputs = yamnet_layer(tf.reshape(inputs, [-1])) # Ensure correct reshaping model = tf.keras.Model(inputs, outputs) # Save the model to a .saved_model format model.save("yamnet_saved_model") # Convert the saved model to Core ML format mlmodel = ct.convert( "yamnet_saved_model", source="tensorflow", inputs=[ct.TensorType(shape=(1, 15600), name="audio_waveform", dtype=np.float32)] ) # Save the converted model as a .mlmodel file mlmodel.save("YAMNet.mlmodel") print("Conversion successful! Model saved as YAMNet.mlmodel") Execution Error: √ yamnet-tensorflow2-yamnet-v1 % python convertItScript.py 2024-11-09 15:41:35.807906: I tensorflow/core/platform/cpu_feature_guard.cc:193] This TensorFlow binary is optimized with oneAPI Deep Neural Network Library (oneDNN) to use the following CPU instructions in performance-critical operations: AVX2 FMA To enable them in other operations, rebuild TensorFlow with the appropriate compiler flags. Running TensorFlow Graph Passes: 100% Converting TF Frontend ==> MIL Ops: 56% Traceback (most recent call last): File "~/yamnet-tensorflow2-yamnet-v1/convertItScript.py", line 19, in <module> mlmodel = ct.convert( File "~/.pyenv/versions/3.10.12/lib/python3.10/site-packages/coremltools/converters/_converters_entry.py", line 444, in convert mlmodel = mil_convert( File "~/.pyenv/versions/3.10.12/lib/python3.10/site-packages/coremltools/converters/mil/converter.py", line 190, in mil_convert return _mil_convert(model, convert_from, convert_to, ConverterRegistry, MLModel, compute_units, **kwargs) File "~/.pyenv/versions/3.10.12/lib/python3.10/site-packages/coremltools/converters/mil/converter.py", line 217, in _mil_convert ... File "~/.pyenv/versions/3.10.12/lib/python3.10/site-packages/coremltools/converters/mil/mil/ops/defs/iOS15/elementwise_unary.py", line 870, in value_inference return self.get_cast_value(self.x, self.dtype.val) File "~/.pyenv/versions/3.10.12/lib/python3.10/site-packages/coremltools/converters/mil/mil/ops/defs/iOS15/elementwise_unary.py", line 896, in get_cast_value return input_var.val.astype(dtype=type_map[dtype_val]) AttributeError: 'float' object has no attribute 'astype'

Thank you for the insights on fine-tuning SNSoundClassifier with AudioFeaturePrint and logistic regression. However, I’m still unclear on how to effectively integrate embeddings from SNSoundClassifier into this pipeline, given that they aren’t directly accessible. Are there specific steps or methodologies to consider for augmenting the base model with user-supplied audio data, and how can I ensure the classifier accurately reflects custom sound classes? What specific pipeline do you recommend? Base model seems to be necessary while fine-tuning on CreateML. If SNSoundClassifier can be used then how? If it cannot be used as base model then its going to be either TF or PyTorch model (which one) Any additional guidance would be greatly appreciated!

Hi Greg, Thank you for taking the time to look into this. Unfortunately, I wasn't able to upload the file directly due to the "Add File" option not allowing me to select the 19.9 MB compressed zip file. To make it accessible, I’ve uploaded the file to an S3 bucket, and you can download it here: https://curvsort-public.s3.ap-south-1.amazonaws.com/AVCamBuildingACameraApp_1.zip. Please let me know once you've downloaded it, and I will delete it from the bucket. The key files with the focused issue are ESoundClassifier and CaptureService (specifically resultsObserver1 and the sections marked Works 1/2/3). To Reproduce the Issue: I am running the app on an iPhone 11 with Swift 5 and Xcode 16 Problem Case: In this case, there are no logs for the analyze and analyzeAudio methods. Working Case: To observe the working flow, comment out the code at // Does not work 1 and uncomment the three lines at // Works 3. You should then see logs for both analyze and analyzeAudio. Swift 6 Concurrency Issues: After switching the build setting to Swift 6, the following concurrency-related errors are reported: ThumbnailButton Issue: The PhotosPicker throws an error when displaying the thumbnail: "Main actor-isolated property 'thumbnail' cannot be referenced from a Sendable closure." AVAudioBuffer: The AVAudioBuffer is also flagged as a non-sendable type, which causes concurrency errors. Apple Source Code References: AVCam: Building a Camera App Classifying Sounds in an Audio Stream I hope this provides the necessary context. Let me know if you need any further details. Best regards

///////////////////////// ///////////////////////// // ensures all operations happen off of the `@MainActor`. actor CaptureService { ... nonisolated private let resultsObserver1 = ResultsObserver1() ... private func setUpSession() throws { .. } ... setVideoCaptureMode() throws { captureSession.beginConfiguration() defer { captureSession.commitConfiguration() } /* -- Works fine (analyzeAudio is printed) Task { self.resultsObserver1.startAudioEngine() } */ self.resultsObserver1.startAudioEngine() // Does not work - analyzeAudio not printed captureSession.sessionPreset = .high try addOutput(movieCapture.output) if isHDRVideoEnabled { setHDRVideoEnabled(true) } updateCaptureCapabilities() } ///////////////////////// ///////////////////////// class ResultsObserver1 { let resultsObserver2 = ResultsObserver2() var classifiedText: String = "" var confidence: Double = 0.0 private var audioEngine: AVAudioEngine? private var soundAnalyzer: SNAudioStreamAnalyzer? private var inputFormat: AVAudioFormat? let analysisQueue = DispatchQueue(label: "com.example.AnalysisQueue") func analyzeAudio(buffer: AVAudioBuffer, at time: AVAudioTime) { print("analyzeAudio") analysisQueue.async { print("analyze") self.soundAnalyzer?.analyze(buffer, atAudioFramePosition: time.sampleTime) } } func stopAudioEngine() { print("stopAudioEngine") soundAnalyzer?.removeAllRequests() audioEngine?.inputNode.removeTap(onBus: 0) audioEngine?.stop() soundAnalyzer = nil audioEngine = nil inputFormat = nil } // Setup audio analysis using SNAudioStreamAnalyzer func startAudioEngine() { print("startAudioEngine") // Create a new audio engine. audioEngine = AVAudioEngine() print("audioEngine: \(String(describing: audioEngine))") // Get the native audio format of the engine's input bus. let inputBus = AVAudioNodeBus(0) inputFormat = audioEngine?.inputNode.inputFormat(forBus: inputBus) print("inputFormat: \(String(describing: inputFormat))") guard let inputFormat = inputFormat else { print("Failed to get input format") return } do { // Start the stream of audio data. try audioEngine?.start() print("audio engine started") } catch { print("Unable to start AVAudioEngine: \(error.localizedDescription)") } // Create a new stream analyzer. soundAnalyzer = SNAudioStreamAnalyzer(format: inputFormat) print("soundAnalyzer: \(String(describing: soundAnalyzer))") // Use Apple's built-in classifier version 1 let version1 = SNClassifierIdentifier.version1 do { // Create a classification request for version 1 let request = try SNClassifySoundRequest(classifierIdentifier: version1) // Add a sound classification request that reports to an observer. try soundAnalyzer?.add(request, withObserver: resultsObserver2) print("Added request to soundAnalyzer with the specified observer") } catch { print("Error setting up sound analysis: \(error)") return } audioEngine?.inputNode.installTap(onBus: 0, bufferSize: 8192, format: inputFormat, block: { buffer, when in self.analyzeAudio(buffer: buffer, at: when) }) /* audioEngine?.inputNode.installTap(onBus: 0, bufferSize: 8192, format: inputFormat, block: { buffer, when in print("buffer recieved") self.soundAnalyzer?.analyze(buffer, atAudioFramePosition: AVAudioFramePosition(buffer.frameLength)) }) */ do { // Prepare and start the audio engine audioEngine?.prepare() try audioEngine?.start() } catch { print("Error starting audio engine: \(error)") } } } class ResultsObserver2: NSObject, SNResultsObserving { /// Notifies the observer when a request generates a prediction. func request(_ request: SNRequest, didProduce result: SNResult) { // Downcast the result to a classification result. guard let result = result as? SNClassificationResult else { return } // Get the prediction with the highest confidence. guard let classification = result.classifications.first else { return } // Get the starting time. let timeInSeconds = result.timeRange.start.seconds // Convert the time to a human-readable string. let formattedTime = String(format: "%.2f", timeInSeconds) print("Analysis result for audio at time: \(formattedTime)") // Convert the confidence to a percentage string. let percent = classification.confidence * 100.0 let percentString = String(format: "%.2f%%", percent) // Print the classification's name (label) with its confidence. print("\(classification.identifier): \(percentString) confidence.\n") } /// Notifies the observer when a request generates an error. func request(_ request: SNRequest, didFailWithError error: Error) { print("The analysis failed: \(error.localizedDescription)") } /// Notifies the observer when a request is complete. func requestDidComplete(_ request: SNRequest) { print("The request completed successfully!") } }