# Sampling frequency

A friend of mine who lives in Thailand helps me to translate phrases in Thai. Today he sent me some audiofiles that I opened in Audacity for analyzing. His voice seems to be in some way distorted, or at least not clear. He has been using X-touch tab to record.

Here in Sweden my girlfriend helps me to record a woman's voice and we are using a Samsung Galaxy S3. Here the voice is almost perfect.

What I could see in Audacity is that the files where the sound is not clear have a sample rate of 8000 Hz, compared to the clear soundfiles from samsung which is 44100 Hz.

Does the lower sampling rate produce low quality sound files? I read that if one records speech, a lower sampling rate is ok.

In both cases - the amplitude resolution is the same - that is a 32 bit float.

Lowering the sampling rate affects directly the highest frequency that is captured or played back, in line with Nyquist's sampling theorem, which defines the phenomenon.

The sampling theorem states that:
sampling frequency / 2 = highest audible frequency in the captured/played digital signal

A voice needs around 4-5kHz of bandwidth (around 1kHz-6kHz in audio frequencies) to be tangible. Therefore an 8kHz sample rate (divided by 2 gives 4kHz of bandwidth) suffices and minimizes the need of bandwidth, therefore it needs less storage space and is faster and more efficient (by electrical means) to transfer and thus a low sampling rate such as 8kHz is suitable for mobile phones and other transmission by radio waves.

If the sampling rate was reduced to reduce filesize for transfer, I would suggest looking at bit-rate instead. A lot of podcasts I listen to can get away with 64 kbps mp3s, but I would suggest higher for translation purposes. Still, a 256 kbps mp3 would be a big space saver compared to uncompressed WAV.

@Internet Human, true those are the most important frequencies, but I think 8 kHz is cutting it a bit close. There's still a lot of information being lost. If you're listening to a foreign language or a thick accent, your brain will probably be less likely to fill in the gaps. I would opt for a 22 kHz sampling rate, or record uncompressed and use MP3 encoding.

There is also the question of the quality of hardware (built in MIC and preamp). On my laptop, the built in electret MIC produces horrible, distorted, garbled sound - even when recording to 16bit/44.1kHz. There is all sorts of interference too, like whining from the HD,etc. Bad preamp, dirty power supply. This might be the case with tour friends recording too.

To help with intelligibility, sampling at such low rates would require some emphasis (pre- or de-), though i doubt any of these devices apply it. The other thing to watch out for is aliasing. Here's an interesting paper for more details: http://www.wescottdesign.com/articles/Sampling/sampling.pdf

8 khz sampling rate are not enough for "clear" voice. With 8k you will lose the high frequencies which give clarity to the sound.

44.1khz are needed to capture high-frequencies up to 22khz (human ear typically hear to 16khz) so that will be enough regarding sampling rate.

If you getting the voice files in mp3 format, i suggest 192kbps. Most people will say 128kbps or even 64kbps, but i can hear the difference between 192 and 128 kbps, even though my hearing is suffering a bit. 192kbps should be enough to have clear voice.