Sampling rate is a most important parameter that determine audio quality. The sampling rate is the number of samples of a sound that are taken per second to represent the audio. The more samples taken per second, the more accurate the digital representation of the sound can be.
There are two types of resampling:
- Upsampling - when the sampling rate is increased, interpolation is used to add intermediate samples to the audio signal.
- Downsampling - in this case, the new sampling rate is lower than the old rate and a decimation is performed. Excess segments are removed from the audio signal.
Why do I need to convert sample rate?
All audio formats and audio hardware support only limited range of sampling rates. Here is a few cases when you need to convert sampling rate:
- HD audio formats support up to 192 Khz sampling rates, but common audio formats support only sampling rates from 8000Hz to 48000Hz
- MP3 compression quality depends from sampling rate, you can not use 256Kbps or 320Kbps bitrates for low quality audio files.
- AMR codec support only 8000Hz and 16000Hz sampling rates
- Audio CD support only 44100Hz sampling rate
- Opus format does not support the 44100 sampling rate, which is the most common. Conversion between 44100 and 48000Hz is mathematically inconvenient
- Despite the essential differences, the conversion between DSD and PCM is nothing more than a sampling rate conversion
Audio Converter Plus will take care about sampling rate conversion automatically, but you can force it to use any frequency in codec settings.