Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.

GSM enhanced full rate speech codec

Abstract: This paper describes the GSM enhanced full rate (EFR) speech codec that has been standardised for the GSM mobile communication system. The GSM EFR codec has been jointly developed by Nokia and University of Sherbrooke. It provides speech quality at least equivalent to that of a wireline telephony reference (32 kbit/s ADPCM). The EFR codec uses 12.2 kbit/s for speech coding and 10.6 kbit/s for error protection. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec provides substantial quality improvement compared to the existing GSM full rate and half rate codecs. The old GSM codecs lack wireline quality even in error-free channel conditions, while the EFR codec provides wireline quality not only for error-free conditions but also for the most typical error conditions. With the EFR codec, wireline quality is also sustained in the presence of background noise and in tandem connections (mobile to mobile calls).

Video coding for low bit rate communications (H.263)

Abstract: The H.263 Recommendation specifies a coded representation that can be used for compressing the moving picture components of audio-visual services at low bit rates. Detailed specifications of the first generation of this codec under the test model (TM) to verify the performance and compliance of this codec were finalised in 1995. The basic configuration of the video source algorithm in this codec is based on ITU-T Recommendation H.261, which is a hybrid of interpicture prediction to utilise temporal redundancy and transform coding of the residual signal to reduce spatial redundancy. However, during the course of the development of H.261 and the subsequent advances on video coding in MPEG-1 and MPEG-2 video codecs, substantial experience was gained, which has been exploited to make H.263 an efficient encoder. In this chapter those parts of the H.263 standard that make this codec more efficient than its predecessors will be explained.

Bandwidth Extension of Telephone Speech Using a Neural Network and a Filter Bank Implementation for Highband Mel Spectrum

Abstract: The limited audio bandwidth used in narrowband telephone systems degrades both the quality and the intelligibility of speech. This paper presents a new method for the bandwidth extension of telephone speech. Frequency components are added to the frequency band 4-8 kHz using only the information in the narrowband speech. A neural network is used to estimate the mel spectrum in the extension band in short time frames based on features calculated from the narrowband speech. A wideband excitation signal is generated by spectral folding from the narrowband linear prediction residual and a filter bank is utilized to divide the excitation into four sub-bands that cover the extension band. These sub-bands are weighted such that the estimated mel spectrum is realized. Bandwidth-extended speech is obtained by summing the weighted sub-bands and the original narrowband signal. Listening tests show that this new method improves speech quality compared with narrowband telephone speech and with a previously published bandwidth extension method.

Codebook tables for encoding and decoding

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Spatial Audio Coding: Next-generation efficient and compatible coding of multi-channel audio

Abstract: Reference LCAV-CONF-2005-033 URL: www.aes.org Record created on 2005-10-07, modified on 2017-05-12