07-27-2023, 03:23 PM
(07-27-2023, 06:26 AM)Convert Wrote: Hi All and apologies for the delayed response. I agree with Tim and many others who say that as long as the information is in digital format no sound quality can be lost as it is always 1 and 0 and when a 1 is lost it is re-read and corrected. Only when converted to analog sound quality can be lost.
However what bugs me for years: why is so much energy spend on digital devices to keep SQ high when the cheapest hardware and the bulkiest operation system does not matter? Why Apple products for example, famous for digital quality in video and sound for years?
You may want to revisit your assumptions.
Digital audio signals can be distorted by timing errors, electromagnetic interference (noise) and software/hardware bugs all of which would be evident by examining signal traces or the software code containing the bug.
Digital audio data is transmitted between the source (renderer) and the destination (audio device) using Isochronous transfer mode which has no error detection/correction mechanism because its sole purpose is to deliver data at a desired rate.
This is in contrast to Bulk data transfer mode which is used for transferring file data and has error detection/correction, retransmit and guaranteed delivery or fail. Another way to look at it is that for file transfer whats most important is that the data is unaltered and that each transfer of a chunk of data completes successfully not how long the transfer takes. For audio data whats most important is that the desired data rate is maintained not whether a bit here or there might be dropped.
For example bulk data transfer is used for getting an audio file from your NAS or attached USB drive to the player software's input buffer. Isochronous transfer is use to get the decoded PCM data from the player to the audio device across a USB or I2S interface.
What about getting audio data from a remote source to the player software's input buffer across a network? This is done using so called "streaming" protocols. Some operate more like Bulk data transfer and others more like Isochronous transfer. In practice which operational mode is used doesn't matter.
Claims that the OS or playback software introduces distortion in the audio data needs to be accompanied by proof that shows the distortion in a signal trace and identifies exactly what in the code is causing the distortion otherwise consider the the claims to be bogus.
