If you do not use any form of compression (or use lossless compression) I would keep the files in the format I got them, because any manipulation in sampling frequency will cause a small degradation in quality (most noticeable when downsampling = from a higher frequency to a lower and upsampling will give you no benefits - you cannot improve the quality of the sound above the quality of the original). ), if so, then most of the time (except when you use a lossless compression format) this compression will influence the quality more than the chosen sampling frequency. )Īnother thing that will influence quality is: Do you use compression on this audio files (MP3, WMA, Flac. Most of the quality loss experienced with digital audio starts after the conversion from digital sound to analog, and depends on the used filters (a good digital -> analog conversion needs a filter to remove the high frequency digitalization effects), the amplifier, and most important the speakers (or headphones or. And as user55325 states, anything above 44.1 kHz is not really needed for the (normal) human ear. Most modern sound card can natively support the most common sampling frequencies (the ones you mention) without quality loss.
#WAV 8 BIT VS 16 BIT SOFTWARE#
I believe most integrated audio hardware's native sample rate is 48 kHz, so it would be best to resample everything to that rate, since software algorithms are likely to be better than cheap integrated hardware. If you choose a different rate, it'll get upsampled (or downsampled, if you choose 96 kHz) in hardware, which may or may not cause audible quality loss depending on how it's done. However, we are dealing with imperfect hardware, and so the answer depends on your sound card's native sample rate. Since the range of human hearing does not extend past 20 kHz - in fact, for an adult human it's probably less than 19 kHz - 44.1 kHz is more than adequate. In theory, the Nyquist theorem tells us that a given sampling rate can accurately reproduce any frequency less than half that rate. If this question can be reworded to fit the rules in the help center, please edit the question. Many good questions generate some degree of opinion based on expert experience, but answers to this question will tend to be almost entirely based on opinions, rather than facts, references, or specific expertise. Just to be clear, I am not talking about converting files, I'm talking about what happens when a given sound is played on the fly using the default format set in Windows:ġ2.5k 29 29 gold badges 108 108 silver badges 179 179 bronze badges closed as primarily opinion-based by Dave, DavidPostill ♦, Gaff, mdpc, G-Man Nov 5 '16 at 6:12 Would it be terrible to just choose that as my default format? Is it better to 'upsample' 44.1 KHz audio to 48 KHz, or 'downsample' 48 KHz audio to 44.1 KHz?Īlso, I do have a handful of 24-bit, 96 KHz albums that I rarely listen to. I can't tell the difference in testing between using DVD quality vs CD quality as the default format, but if I were able to which would sound better? Which makes more sense for me to use from a technical point of view?
#WAV 8 BIT VS 16 BIT WINDOWS#
The Windows system sounds are 16-bit at 22050 Hz, which seem to fit nice and evenly into 44.1 KHz.
Most of my music is in this format, but most of my movies use a 16-bit at 48 KHz format (named DVD quality in the Sound panel).
#WAV 8 BIT VS 16 BIT 32 BIT#
A decent audio program would probably import lossy audio as 32 bit (probably even import 16 bit wave files as 32 bit too, or at least apply 32 bit processing to them), as then you can edit and apply effects etc with minimum quality loss due to rounding errors, but most of the time you'd probably export the finished audio as a 16 bit wave file. You can write 24-bit WAV files to an optical disc, of course, but it will not be a standards complying audio CD. Higher bitrates are generally used for editing, not for playback. Audio CDs are encoded with 16-bit values.
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