Method and apparatus for sampling-rate conversion of audio signals

摘要

Generally, performing sampling-rate conversion from a higher sampling frequency fs1 to a lower sampling frequency fs2 results in aliasing. It is known to use a low-pass filter, known as anti-alias filter, for avoiding this alias distortion. Its effect is to remove spectral contents above fs2/2 from the digital signal. According to the invention these signal parts are suppressed at spectral decoding resulting in a bandwidth of the signal to be re-sampled which is less than half of the second sampling frequency fs2. This can be done for MPEG encoded audio signals by limiting the decoding to a certain number of subbands, for DOLBY AC-3 encoded audio signals by setting certain spectral lines to zero at decoding. The inventive method not only totally removes the processing power needed for calculating an anti-alias filter, but also limits the decoding work needed. <IMAGE>