WAV vs. FLAC vs. AIFF

Although in principle any half-way decent modern computer should have no trouble de-compressing and processing any audio format, my suspicion is that a major reason for reported sonic differences between lossless formats relates to the fact that in modern computers the clock rate, voltage, and power draw of the cpu chip are usually dynamically varied as a function of the processing requirements at any instant of time. That is done for purposes of minimizing power consumption and heat generation, and in the case of laptops to prolong battery run-time.

That switching involves changes in current that are both large and abrupt, which can be expected to cause significant noise transients to propagate through a lot of the circuitry on the computer's motherboard. That in turn can be expected to contribute to jitter, or even outright mis-clocking and breakups, on the signals that are used to output the audio data.

See my post here for a description of how to disable that switching in Windows 7. That change did in fact resolve the audio breakup problem the OP in that thread was having. Some computer BIOS's also allow "EIST" (aka "Speed Step") to be disabled, which may accomplish the same thing. I'm not particularly familiar with Mac's, but I believe that third-party software might be needed to do this.

The sonic significance of all of this will obviously be dependent on the particular computer that is being used, on what kind of output is being used (USB, S/PDIF, Ethernet, etc.), and on the jitter sensitivity of the component to which the signal is being sent. Ethernet and wireless presumably have little if any sensitivity to these issues, because of the packetized and buffered nature of the data transmission.

None of this necessarily correlates with the resolution or quality of the audio system.

It would be interesting to know if those who report sonic differences between these formats perceive the same differences when these power conservation features are disabled, and the cpu is running at the same speed and voltage all the time.

Regards,
-- Al

Hi tom,
Maybe there are many reasons why one file sounds different on system than another.

I like to think I have a reasonable set up. The only differences I have heard between the same file is in the conversion. I.e coverting from 44.1 to 48k etc. Whether a WAV or Aiff they sound the same if taken from the same master file. While flac has some live unpacking it too sounds the same unless there is some terrible software being used.

If someone believes any different send me a PM with the original music file, and I will gladly convert it on professional equipment to whatever you fancy. WAV to AIFF or whatever & send it back to you. Then see.

Al - I agree that minimizing system changes is a good practice, as is removing unnecessary software and processes. I take a different route from many - I use a Netbook with Windows 7 Starter. There are very few processes running, not even virus software (which is OK because it is a dediated system and I almost never have it on the Internet directly). You can even turn off networking if you have the files local. I still think these tricks are less important with a good aysns USB converter/DAC. I use a async USB converter and a DAC that does its own reclocking, so hopefully I am immune to PC induced jitter. If you have galvanic isolation you can elimiate other types of noise also. Unfornately, when people hear differences it is very difficutlt to find out why.

Al:

Thanks for the clarification. I've read the same elsewhere and it seems to all boil down to the competition for power and the sound card that goes on inside of a computer. It is not optimized for music playback as currently set up. That's probably the reason why a third party add on like Bitperfect (for those of us with shallow pockets) dramatically cleans up the sound.

Once you have an optimized platform, like a well designed music server, then differences between files should be minimal, at best, when it comes to the better ones. It will be nice when some sort of standardization takes effect so makers can concentrate on perfecting software/hardware and we can just rip and download to our hearts content and sit back and enjoy.

I have yet to hear all that's out there so this is conjecture, on my part, but it seems to make sense.

Al the best,
Nonoise

Okay, here's my hypothesis:

When using networked servers, such as Sonos and Logitech, the decoding processes of FLAC and ALAC have plenty of time to execute because the other processes are not real-time. This is because the networked stream is packetized and transmitted very quickly, and does not get involved with the S/W audio stack in the computer. The data processing in the computer is minimal and happens very quickly making the latency very low for these transfers. This allows the CODEC to run as slow or fast as it needs to run to achieve accurate results. As a result, the sound quality differences with these lossless formats is usually minimal if even detectable when using network protocol.

On the other hand, when you use Firewire or USB for data streaming, some of the audio stack is involved and the CODEC must run in real-time and keep-up with the stream rate. Because the audio stack creates a lot of latency, even when playing uncompressed files, there is evidently not much time left for FLAC decoding to keep up with the bit stream. As a result, the timings are very tight and sound quality suffers as a result. This is why I believe on a resolving system using USB or Firewire, FLAC files sound like listening through a tunnel and the same FLAC uncompressed to .wav sounds normal. I dont know if this is a result of poor programming for the FLAC and ALAC CODECs or maybe just the way that they execute when they are competing for resources and repeatedly queue and stall in the execution sequence. With multi-threading in computer OS now, these applications dont run continuously ever.

There is a lot of anecdotal evidence to support the above hypothesis. I have no technical proof however.

Steve N.
Empirical Audio