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How should I decide on a default audio format? [closed]
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Closed 7 years ago .
I see that Windows sets the default output format at 16-bit, 44100 Hz (CD Quality). 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). The Windows system sounds are 16-bit at 22050 Hz, which seem to fit nice and evenly into 44.1 KHz. 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? Is it better to «upsample» 44.1 KHz audio to 48 KHz, or «downsample» 48 KHz audio to 44.1 KHz? Also, I do have a handful of 24-bit, 96 KHz albums that I rarely listen to. Would it be terrible to just choose that as my default format? 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:
Louis Waweru
asked Jan 8, 2014 at 6:04
Louis Waweru Louis Waweru
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Incidentally, using anything above DVD quality makes iTunes 11 stutter when playing a large number of tracks in my collection.
Jun 30, 2014 at 8:45
3 Answers 3
In theory, the Nyquist theorem tells us that a given sampling rate can accurately reproduce any frequency less than half that 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.
However, we are dealing with imperfect hardware, and so the answer depends on your sound card’s native sample rate. 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.
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.
answered Jan 8, 2014 at 6:45
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user55325, thanks for sharing that. I’m still a little confused by your last sentence though, «since software algorithms are likely to be better than cheap integrated hardware.» @Johan & user55325, It kind of gave me the feeling that you thought I was talking about converting my files for storage. I’ve updated the question to make it clear that I’m not talking about converting files, but how they are dealt with during live playback.
Jan 8, 2014 at 16:28
Also from the Wiki article, can I take «a bandlimited function can be perfectly reconstructed from a countable sequence of samples if the bandlimit, B, is no greater than half the sampling rate (samples per second)» to mean that I should choose the highest output format available. Since this would have the best chance of having at least twice the sample rate of a given format?
Jan 8, 2014 at 16:32
The idea is that the DAC chip in your sound hardware supports one sample rate, so if your sound driver is sending it a different one, the hardware has to resample it before it gets sent to the DAC. Whatever 0.01-cent chip the manufacturer decided to use for resampling may not be as good as the algorithms provided by your OS, and may introduce audible artifacts. @Johan may be right that modern hardware is good enough that no audible loss would take place — I don’t know because I haven’t seen any specifications myself.
Jan 8, 2014 at 16:42
If you’re recording audio that you are going to postprocess in some way, then yes, you should choose the highest sample rate available. For playback, though, this is not necessary (and may adversely affect quality, or it may not if your hardware can handle it, but it won’t improve quality any).
Jan 8, 2014 at 16:45
Sorry for the late reply. I wasn’t looking to improve quality. I was looking to avoid quality loss. I expected quality loss for the rare 24bit/96KHz files playing back at CD or DVD quality. I was looking to see if there would no quality loss when playing back lower bitrate music, movies and system sounds at a higher bitrate.
Jun 30, 2014 at 9:01
Most modern sound card can natively support the most common sampling frequencies (the ones you mention) without quality loss. And as user55325 states, anything above 44.1 kHz is not really needed for the (normal) human ear. 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 . )
Another thing that will influence quality is: Do you use compression on this audio files (MP3, WMA, Flac . ), 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.
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). Upsampling gives also the disadvantage of larger files, so you will need more disk space to hold the same files.
About the 96kHz files: those can perfectly be used for further processing, this is (one of) the format(s) studios use when they do a recording (most multi-track) and then combine all this to put it on a CD (at 44.1 kHz) or in a movie (48 kHz). When you have them in this format, I would keep them like this, except when you need the disk space, but I would never use it as format for all my files.