Lossless Switcher is available free of charge from Neo’s Github repository. Joined Mar 4, Posts Likes I recommend Bitperfect mainly for its value. Vincent Neo warns that “there may be short interruptions to your audio playback, during the time where the app attempts to switch the sample rates.” He also says that his app has yet to be tested on an M1 Mac. But bitperfect vs audirvana free some reason the site isnt working now. Lossless Switcher reads the sample rate of the incoming Apple Music stream and then automatically sets the corresponding sample rate in the Audio MIDI Setup app. Once installed and running, the app runs in the menu bar at the top of the screen, next to the clock. Vincent Neo’s LosslessSwitcher app “switches your current audio device’s sample rate to match the currently playing a lossless song on your Apple Music app, automatically.” And this is only achievable on MacOS by manually specifying the Audio MIDI Setup app’s sample rate on a track-by-track (or album-by-album) basis. Only when no resampling takes place do we have what is known as ‘bit-perfect playback’. Resampling can subtly alter the sound of a stream heard on a more resolving hi-fi or head-fi system. If we hit play on a 44.1kHz file but the Audio MIDI Setup app is set to 96kHz, we gain dummy data (and arguably lose again). If we hit play on, say, a 192kHz hi-res lossless stream but the Audio MIDI Setup app is set to 44.1kHz, we lose data. Upsampling and downsampling are collectively referred to as resampling. audio samples are thrown away by the operating system when the incoming sample rate sits above the Audio MIDI app’s output setting.audio samples are interpolated by the operating system when the incoming sample rate sits below the Audio MIDI app’s output setting.Why? MacOS will adjust the data’s output according to the settings specified in the Audio MIDI Setup app. Unless we are extremely vigilant, the audio data that we pulled from Apple’s servers isn’t necessarily that which reaches our DAC. Of course, it may be another twenty years until it happens, and I may be drooling into my cream of wheat by then.Running Apple Music on MacOS? For hi-res enthusiasts looking to get the very best from Apple’s lossless content, the gotcha is an absence of automatic sample-rate switching. I am hoping that UA will be on the cutting edge of this as it emerges, so that we can have the benefit of UA's portfolio of excellent plugins to use when mixing DXD at 384 kHz. This is, as I see it, a potential game changer for digital audio which could herald the advent of DSD as a mainstream digital format for audio capture. It is proprietary and only available under license, as it is to PS Audio, who are using it to record for their new record label Octave Records, which specializes in DSD only audiophile recordings. Unfortunately this filter is not generally available. What this means is that the audio may now be mixed as DXD with all of the benefits of PCM based processing, without losing any of the inherent DSD sound quality. The interesting development is that there is a new zero phase filter (the Zephir filter) that has been designed by the BitPerfect Ltd company (as reported in his blog by Paul McGowan of PS Audio in this thread: ), which will allow for a bit perfect (yeah, I know) translation of the orginal DSD signal over to DXD, without the loss of quality it has always suffered. The drawback being that the process degrades the sound quality somewhat ('somewhat' being completely subjective). This is a method of converting the DSD to a form of PCM at 384kHz, which can be processed like all other PCM. It can't be mixed, processed with plugins, or altered in any way - not even level change in short all the ways that PCM can be, and which has made PCM the defacto standard for virtually everyone.įor most people who use DSD the method then has been to treat it like an analog tape recorder - to record to it, then play back from it, converting back to analog, mixing it in the analog domain, with no digital processing.įor those who want to process it in the digital domain, there is a workaround - the use of DXD. It represents the audio digitally by the density of the digital 1's - ie where the level of signal is louder, the number of 1's is greater.Īlthough DSD is believed by many (myself included) to be the most natural and accurate sounding format - superior sounding to PCM - it has a fatal drawback, one that has relagated it to the weeds in terms of mainstream digital audio - It can't be processed in any way in it's native form. It does this by using a very high sample rate - the lowest DSD sample rate (DSD64) is 64 times the sampling rate of 44.1kHz at roughly 2.8 million samples per second. Not sure how much traction this will get, but here goes.Īs many of you will know, DSD is a one bit digital audio format which uses PDM (pulse density modulation) to encode analog audio to digital.
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