Speech coding

About: Speech coding is a research topic. Over the lifetime, 14245 publications have been published within this topic receiving 271964 citations.

Improving sound quality of established low bit-rate audio coding systems without loss of decoder compatibility

Abstract: A multi-channel audio compression technology is presented that extends the range of sampling frequencies compared to existing technologies and/or lowers the noise floor while remaining compatible with those earlier generation technologies. The high-sampling frequency multi-channel audio (12) is decomposed into core audio up to the existing sampling frequencies and a difference signal up to the sampling frequencies of the next generation technologies. The core audio is encoded (18) using the first generation technology such as DTS, DOLBY AC-3 or MPEG I or MPEG II such that the encoded core bit stream (20) is fully compatible with a comparable decoder in the market. The difference signal (34) is encoded (36) using technologies that extend the sampling frequency and/or improve the quality of the core audio. The compressed difference signal (38) is attached as an extension to the core bit stream (20). The extension data will be ignored by the first generation decoders but can be decoded by the second generation decoders. By summing the decoded core and extension audio signals together (28), a second generation decoder can effectively extend the audio signal bandwidth and/or improve the signal to noise ratio beyond that available through the core decoder alone.

Reducing noise in audio systems

Abstract: Two or more microphones receive acoustic signals and generate audio signals that are processed to determine what portion of the audio signals result from (i) incoherence between the audio signals and/or (ii) audio-signal sources having propagation speeds different from the acoustic signals. The audio signals are filtered to reduce that portion of one or more of the audio signals. The present invention can be used to reduce turbulent wind-noise resulting from wind or other airjets blowing across the microphones. Time-dependent phase and amplitude differences between the microphones can be compensated for based on measurements made in parallel with routine audio system processing.

Noise-dependent postfiltering

Abstract: The paper introduces a modification of the commonly used postfilter that improves performance when acoustic background noise is present. The modification consists of replacing the nonadaptive postfilter parameters that govern the degree of spectral emphasis (commonly denoted as /spl gamma//sub 1/ and /spl gamma//sub 2/) with parameters that adapt to the noise statistics. We describe an effective mapping from the noise statistics to the emphasis parameters and provide a low complexity noise estimation algorithm that is sufficient for this application. The resulting noise-adaptive postfilter successfully attenuates the background noise and naturally converges to the conventional postfilter at high SNR conditions. Thus, the speech enhancement problem is solved with minimal modification of legacy codecs, since the existing structure of the speech codec is used. Test results indicate that the presented algorithm significantly outperforms the standard postfilter with non-adaptive parameters.

Image coding using vector quantization

Abstract: An image is partitioned into cells of pxp pixels. Each cell is regarded as a vector of dimension p2and is encoded by searching through a codebook for a nearest matching representative vector. A binary word identifying the selected representative vector is assigned as the codeword to describe the original cell. The decoder uses this codeword to address a codebook. Each entry of the codebook contains a full precision digital representation of one of the N representative vectors. The codebook design is based on a clustering technique for vector quantizer design preceded by a classification of training cells into edge or shade cells. Results for coding rates from 0.5 to 1.5 bits/pixel are discussed. Vector quantization appears to be a powerful and promising technique for image coding.

Multi-mode audio recognition and auxiliary data encoding and decoding

Abstract: Audio signal processing enhances audio watermark embedding and detecting processes. Audio signal processes include audio classification and adapting watermark embedding and detecting based on classification. Advances in audio watermark design include adaptive watermark signal structure data protocols, perceptual models, and insertion methods. Perceptual and robustness evaluation is integrated into audio watermark embedding to optimize audio quality relative the original signal, and to optimize robustness or data capacity. These methods are applied to audio segments in audio embedder and detector configurations to support real time operation. Feature extraction and matching are also used to adapt audio watermark embedding and detecting.