Converting Flac to Wav & Upconversion

When FLAC and wav sound different, its probably more due to the different software in play for each and how that is written, designed, and performs more so than the format. Decompressing FLAC files will probably require more CPU processing, but should not be a problem if done right. Of course, things are not always done right, and many factors can come into play when playing digital music files, so differences in performance between the two in any particular case would not surprise me and reasons why may not be apparent.

FWIW, I think it's odd that different software players (that can be tested to show bitperfect output) can sound different playing back the same WAV or FLAC file. Heck, even different versions of the software can sound different playing back the same file.

It is thus not inconceivable that WAV and FLAC despite having bitperfect data can sound different. I remember that the Pure Music designer mentioned the need to minimize sudden/minute spikes in CPU load to improve performance and that was one of the goals of their software update. So it is not just a % of the CPU load that is averaged over time that we need to see but those sudden spikes. There was a talk at RMAF about 1-2 years back by an ESS engineer which talked about the need to look beyond steady states but also how the system reaches steady states (ie does it oscillate through large swings in values before reaching steady states). He found that large swings seemed to have a negative impact on the sound quality. I think there's a lot to the computer playback chain that we are only just beginning to understand.

For example, 96k files require twice as many mixing resources as 48k, which means that only half the number of channels on the console would be available.

Mr.Jazz1959,I do not know a lot about recording ,could you please explain to me how recording at 96k will use twice as many of available channels on the recording console as recording at 48k.Thank you

IT boils down to the software must be able able to run correctly and fast enough to maintain a in-memory cache of data that is available at the exact time needed for playback. Playback happens in real time, so data must be continuously streamed, made available and applied at precisely the correct time in order for things to sound best. Real time applications like computer audio up the ante in regards to what is needed for optimum performance.

Memory is shared and virtual on most general purpose computers. MAny programs may compete for available memory. WHen there is not sufficient physical (fast) memory available, virtual disk based (slow) memory is applied as a supplement to allow things to run though not as fast.

If data is not availble at the exact moment needed for playback, software has three choices:

1) pause or wait for memory to become available again. THis may result in an interruption or delay in playback until data is once again available

2) reduce the bit rate of the data stream. This would result in not all bits being used and would affect sound quality accordingly, although teh music may continue to play. A program/system designed for audiophiles would not chose this approach, but it might be applied otherwise for more casual listeners without concern.

3) some combination of 1) and 2)

With software/computer programs anything is possible and may well occur unless care is taken in design to avoid it. Specialized audio streaming devices like Squeezebox are essentially specialized and dedicated (not general purpose) computers designed to optimize performance. They take a lot of the mystery and variables that can affect the sound quality out of the equation.

SO in my opinion, the system used to stream and play FLAC or WAV files is a much bigger factor in regards to sound quality than the format itself, both of which are lossless and essentially equivalent in terms of information content. Its what content gets delivered and how well that matters most.

Unfortunately, digital playback mechanisms are not transparent to the user. There is nothing other than the hearing the resulting sound apparent to determine if if all this is occurring well or not. That's one advantage of vinyl. There is more there to see, feel and touch in addition to hear. It gives you something more physically tangible to sink your teeth into and perhaps adjust or tweak for better performance, if that is your thing. With digital audio, your listening fate is more largely determined by the equipment designers. Luckily, there are many good ones out there. There is an advantage to placing your fate in the hands of a trusted expert as well. Most will likely prefer that approach.

Mapman - with many people using aysnc USB the real time nature of the PC is minimal. All it really has to do is keep the buffer full and not get in the way of the aysnc USB requests. Filling the buffer is just not that hard. It is hard to understand how a computer with little else running other than the music player and with the CPU running at only a few percent of usage can have a significant effect on the timing of the aysnc USB over several minutes of music unless there is a serious flaw in the design of the player. There may be computer effects that influence wav and flac playback, but it seems that aysnc USB takes the real time aspect of the PC out of the equation. This is especially true for DACs that re-clock the data. Before async USB, the real time nature of the PC could definitely be an issue. But that seems minimal with async USB.