Drive space is an issue for SSD's and RAMdisks, and for easy quick access to a large music collection.
Drive space is an issue for SSD's and RAMdisks, and for easy quick access to a large music collection.
I suppose it depends on how much drive space you need. In my case, I have enough on each of my 2 TB SSDs that it would take me in excess of a year to listen to each and every track...I would need to just sit in a room day/night after day/night. I installed the 2TB in my McIntosh MB100. I have not violated the internals of my Aurender but have access to one of the 2TB drives. The other two are just backups. I have a 16TB Promise Raid but do not use it for steaming but just as a means for collecting data. Because RAID is not a back up solution. I have two additional separate drives [2 sets/different manufactures] off site as backups too. The RAID collects both audio and video. Cable directly connected to one of my OPPO BDP 105Ds. I do not bother with managing it over wifi/ethernet.
That was a confusing post. First, MQA is a lossy compression as opposed to lossless FLAC. Second, FLAC provides an internal checksum, verified by software than hardware, it is true, but the idea of anything being dependent on computer "hardware" is counter-productive.
Because of FLACs built-in checksum, it's easier and faster for a computer to decode FLAC from a hard drive to WAV.
Interesting discussion! For the super slow, I still seem to be missing something: if the wav and flac files are "mathematically equivalent" and dac chips are very powerful and can easily handle this calculation, what are some reasons for the files not sounding the same? Why do so many folks in this thread prefer to convert their flac to wav?
All I know is that on both my computers it is faster (so I presume easier) to decode FLAC to WAV in RAM than to copy (or load?) WAV to WAV in RAM. Also I know that the process of decoding FLAC includes using the checksum. Do you have a different explanation for this phenomenon (FLAC is faster)?
The answer is that digital audio is digital as long as we talk file format (CD, file on a HD) or data transmission to a DAC. This is the domain of the bits. This is fully digital.
The moment we start to listen all kind of analog components in to play.
The sample rate is generated by a clock, indeed a crystal oscillating. This is about as analog as you can get hence there will be analog imperfection (intrinsic jitter).
Likewise any component does what is supposed to do, e.g. the head of a HD moving to read the data, but at the same time might generate some noise be if RFI, EMI, a ripple on the power rail, etc.
If this creeps into a DAC, it might affect sound quality.
Hence WAV requires very little processor power (almost raw PCM) but a lot of I/O and FLAC does exactly the reverse.
The result is still bit identical but the noise pattern produced in generating this bits might differ.
If your system in insufficiently isolated against this noise it might become audible.
That is the fun of memory playback, read a track into memory, hence do all the decoding and I/O before playback starts. This will eliminates all the FLAC/WAV differences.
It can, of course, not eliminate expectation bias![]()