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In early 2012 the ISO/IEC JTC1/SC29/WG11 (MPEG) finalized the new MPEG-D Unified Speech and Audio Coding standard. The new codec brings together the previously separated worlds of general audio coding and speech coding. It does so by integrating elements from audio coding and speech coding into a unified system. The present publication outlines all aspects(More)
We propose a low-complexity Bandwidth Extension (BWE) method operating in the Modified Discrete Cosine Transform (MDCT) domain to reduce the bitrate of wideband and super-wideband speech codecs. The proposed method generates a high-frequency signal by copying the MDCT spectrum from the low frequency part, and then adjusts tonality to improve the subjective(More)
The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music,(More)
EVS, the newly standardized 3GPP Codec for Enhanced Voice Services (EVS) was developed for mobile services such as VoLTE, where error resilience is highly essential. The presented paper outlines all aspects of the advances brought during the EVS development on packet loss concealment, by presenting a high level description of all technical features present(More)
1 At recent AES conventions, authors have had the option of submitting complete 4-to 10-page manuscripts for peer-review by subject-matter experts. The following two papers have been recognized as co-winners of the AES 132nd Convention Peer-Reviewed Paper Award. The AES has launched a new opportunity to recognize student members who author technical papers.(More)
This paper describes new time domain techniques for concealing packet loss in the new 3GPP Enhanced Voice Services codec. Enhancements to the existing ACELP concealment methods include guided, improved pitch prediction, increased flexibility and accuracy of pulse resynchronization. Furthermore, the new method of separate linear predictive (LP) filter(More)
This paper proposes a vector quantization (VQ) method based on composite permutation coding for transform audio coding. VQ is widely used for audio data compression. It requires mean square error computation or a similar metric for finding the nearest neighbor in the codebook, which generally incurs a lot of operations. To reduce such operations, we focus(More)