Abstract: The present disclosure relates to an adaptive jitter buffer for buffering audio data received via a network. The adaptive jitter buffer comprises an adaptive audio sample buffer, which comprises an adaptive resampler that receives a number of audio samples of the audio data and that outputs a first number of audio samples, which are resampled from the received number of audio samples according to a resampling factor, an audio sample buffer that buffers audio samples, wherein the outputted first number of audio samples are written to the audio sample buffer during an input access event and a second number of audio samples are read from the audio sample buffer during an output access event, and an audio sample buffer fill quantity controller that controls a fill quantity of the audio sample buffer based on controlling the resampling factor of the adaptive resampler.

Abstract: The present invention relates to a speech signal encoding method for encoding an inputted first speech signal into a second speech signal having a narrower available bandwidth than the first speech signal. The method comprises generating a pitch-scaled version of higher frequencies of the first speech signal and including in the second speech signal lower frequencies of the first speech signal and the pitch-scaled version of the higher frequencies. At least a part of the higher frequencies are frequencies that are outside the available bandwidth of the second speech signal. The pitch-scaled version of the higher frequencies is preferably included in the second speech signal with a gain factor having a value of 1 or a value higher than 1. The present invention further relates to a corresponding speech signal decoding method for decoding an inputted first speech signal into a second speech signal having a wider available bandwidth than the first speech signal.