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Large-scale mobile communication systems tend to contain legacy transmissionchannels with narrowband bottlenecks, resulting in characteristic"telephone-quality" audio. While higher quality codecs exist, due to the scaleand heterogeneity of the networks, transmitting higher sample rate audio withmodern high-quality audio codecs can be difficult in practice. This paperproposes an approach where a communication node can instead extend thebandwidth of a band-limited incoming speech signal that may have been passedthrough a low-rate codec. To this end, we propose a WaveNet-based modelconditioned on a log-mel spectrogram representation of a bandwidth-constrainedspeech audio signal of 8 kHz and audio with artifacts from GSM full-rate (FR)compression to reconstruct the higher-resolution signal. In our experimentalMUSHRA evaluation, we show that a model trained to upsample to 24kHz speechsignals from audio passed through the 8kHz GSM-FR codec is able to reconstructaudio only slightly lower in quality to that of the Adaptive Multi-RateWideband audio codec (AMR-WB) codec at 16kHz, and closes around half the gap inperceptual quality between the original encoded signal and the original speechsampled at 24kHz. We further show that when the same model is passed 8kHz audiothat has not been compressed, is able to again reconstruct audio of slightlybetter quality than 16kHz AMR-WB, in the same MUSHRA evaluation.

Speech bandwidth extension with WaveNet