5,228 research outputs found
Wavenet based low rate speech coding
Traditional parametric coding of speech facilitates low rate but provides
poor reconstruction quality because of the inadequacy of the model used. We
describe how a WaveNet generative speech model can be used to generate high
quality speech from the bit stream of a standard parametric coder operating at
2.4 kb/s. We compare this parametric coder with a waveform coder based on the
same generative model and show that approximating the signal waveform incurs a
large rate penalty. Our experiments confirm the high performance of the WaveNet
based coder and show that the speech produced by the system is able to
additionally perform implicit bandwidth extension and does not significantly
impair recognition of the original speaker for the human listener, even when
that speaker has not been used during the training of the generative model.Comment: 5 pages, 2 figure
Speech coding at 4800 bps for mobile satellite communications
A speech compression project has recently been completed to develop a speech coding algorithm suitable for operation in a mobile satellite environment aimed at providing telephone quality natural speech at 4.8 kbps. The work has resulted in two alternative techniques which achieve reasonably good communications quality at 4.8 kbps while tolerating vehicle noise and rather severe channel impairments. The algorithms are embodied in a compact self-contained prototype consisting of two AT and T 32-bit floating-point DSP32 digital signal processors (DSP). A Motorola 68HC11 microcomputer chip serves as the board controller and interface handler. On a wirewrapped card, the prototype's circuit footprint amounts to only 200 sq cm, and consumes about 9 watts of power
Neural network based speech synthesizer: A preliminary report
A neural net based speech synthesis project is discussed. The novelty is that the reproduced speech was extracted from actual voice recordings. In essence, the neural network learns the timing, pitch fluctuations, connectivity between individual sounds, and speaking habits unique to that individual person. The parallel distributed processing network used for this project is the generalized backward propagation network which has been modified to also learn sequences of actions or states given in a particular plan
Low bit rate speech coding methods and a new interframe differential coding scheme for line spectrum pairs
Ankara : Department of Electrical and Electronics Engineering and the Institute of Engineering and Sciences of Bilkent University, 1992.Thesis (Master's) -- Bilkent University, 1992.Includes bibliographical references leaves 30-32.Low bit rate speech coding techniques and a new coding scheme for vocal
tract parameters are presented. Linear prediction based voice coding techniques
(linear predictive coding and code excited linear predictive coding) are
examined and implemented. A new interframe differential coding scheme for
line spectrum pairs is developed. The new scheme reduces the spectral distortion
of the linear predictive filter while maintaining a high compression ratio.Erzin, EnginM.S
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