Hi all,  Apple dropping on-going development for FireWire devices that were supported with the Core Audio driver standard is a catastrophe for a lot of struggling musicians who need to both keep up to date on security updates that come with new OS releases, and continue to utilise their hard earned investments in very expensive and still pristine audio devices that have been reduced to e-waste by Apple's seemingly tone-deaf ignorance in the cries for on-going support.  I have one of said audio devices, and I'd like to keep using it while keeping my 2019 Intel Mac Book Pro up to date with the latest security updates and OS features.  Probably not the first time you gurus have had someone make the logical leap leading to a request for something like this, but I was wondering if it might be somehow possible of shoe-horning the code used in previous versions of Mac OS that allowed the Mac to speak with the audio features of such devices to run inside the Ventura version of the OS.  Would it possible? Would it involve a lot of work? I don't think I'd be the only person willing to pay for a third party application or utility that restored this functionality. There has to be 100's of thousands of people who would be happy to spare some cash to stop their multi-thousand dollar investment in gear to be so thoughtlessly resigned to the scrap heap.  Any comments or layman-friendly explanations as to why this couldn’t happen would be gratefully received!  Thanks,  em

Hi all, I’ve implemented the new Core Audio Tap API (AudioHardwareCreateProcessTap with CATapDescription) and I’m seeing consistent level attenuation that scales with the number of stereo output pairs exposed by the target device. What I observe Device with 4 stereo pairs (8 outs) → tap shows −12.04 dB relative to source. True 2-ch devices (built-in speakers, AirPods) → ~0 dB attenuation. The attenuation appears regardless of whether I: Create a global (default-output) tap via initStereoGlobalTapButExcludeProcesses: Or create a per-process/per-device tap via initWithProcesses:andDeviceUID:withStream: Additionally, the routing choice inside the sending app matters: App output to “System/Default Output” → I often see no attenuation. App output directly to a multi-out interface (e.g., RME Fireface) → I see the pair-count-scaled attenuation. I can query Core Audio for the number of output channels/pairs and gain-compensate (+20·log10(N_pairs) dB) and that matches my measurements for many cases. However, this compensation is not universally correct because it seems to depend on where each process routes its audio (Default Output vs. direct device), even when those processes are included in the same tap aggregate. Question Is there a supported way to obtain the raw, unattenuated streams for all processes through the Tap API—i.e., to bypass this automatic headroom/attenuation behavior entirely? If this attenuation is expected by design: Is there a documented rule for when it applies (global vs. device taps, per-process taps, stream selection, etc.)? Is there a property/flag to disable it, or a reliable, official method to compute the exact compensation (beyond counting stereo pairs)? Any guidance on ensuring consistent levels when multiple processes route differently (Default Output vs. direct device) but are captured by the same tap? Environment API: AudioHardwareCreateProcessTap + CATapDescription Devices: built-in output (2-ch), RME Fireface (8+ outs / 4+ stereo pairs) Behavior reproducible with both global and per-process/per-device tap descriptions. Attenuation example: 4 stereo pairs → −12.04 dB observed. Happy to provide a minimal sample, measurements, and device logs. Thanks! — David

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!