About: Enhanced full rate is a research topic. Over the lifetime, 32 publications have been published within this topic receiving 581 citations. The topic is also known as: EFR & GSM‐EFR.
TL;DR: The GSM enhanced full rate (EFR) speech codec that has been standardised for the GSM mobile communication system provides wireline quality not only for error-free conditions but also for the most typical error conditions.
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).
TL;DR: A two stage hybrid embedded speech/audio coding structure uses a speech coder as a core to provide the minimal bitrate and an acceptable performance on speech inputs and a transform coder using a modified discrete cosine transform and perceptual coding principles is proposed.
Abstract: A two stage hybrid embedded speech/audio coding structure is proposed. The structure uses a speech coder as a core to provide the minimal bitrate and an acceptable performance on speech inputs. The second stage is a transform coder using a modified discrete cosine transform (MDCT) and perceptual coding principles. This stage is itself embedded both in complexity and bitrate, and provides various levels of enhancement of the core output, particularly for general audio signals like music. Informal A-B comparison tests show that the performance of the structure at 16 kb/s is between that of the GSM enhanced full rate coder at 12.2 kb/s, and the G.728 LD-CELP coder at 16 kb/s.
TL;DR: This work proposes an improved low bit rate bandwidth extension algorithm along with a robust watermarking scheme for CELP-type speech codecs which is especially tailored to state-of-the-art narrowband speech communication networks such as GSM or UMTS.
Abstract: We consider the problem of transmitting a wideband speech signal with a cut-off frequency of fc = 7 kHz over a standardized narrowband (fc = 3.4 kHz) communication link in a backwards compatible manner. In a previous contribution we have shown that backwards compatibility can be achieved by using digital watermarking: we embedded compact side information about the missing high frequency band (3.4 - 7 kHz) into the narrowband speech signal. Here, we present a related system which is especially tailored to state-of-the-art narrowband speech communication networks such as GSM or UMTS. Therefore, we propose an improved low bit rate bandwidth extension algorithm along with a robust watermarking scheme for CELP-type speech codecs. The practical relevance of our system is shown by speech quality evaluations and by link-level simulations for the "enhanced full rate traffic channel" (TCH/EFS) of the GSM cellular communication system.
TL;DR: The enhanced full rate (EFR) speech codec that has recently been standardised for the North American TDMA digital cellular system (IS-136) offers speech quality close to that of wireline telephony and provides a substantial improvement over the quality of the current speech channel.
Abstract: In this paper, we describe the enhanced full rate (EFR) speech codec that has recently been standardised for the North American TDMA digital cellular system (IS-136). The EFR codec, specified in the IS-641 standard, has been jointly developed by Nokia and University of Sherbrooke. The codec consists of 7.4 kbit/s speech (source) coding and 5.6 kbit/s channel coding (error protection) resulting in a 13.0 kbit/s gross bit-rate in the channel. Speech coding is based on the ACELP algorithm (algebraic code excited linear prediction). The codec offers speech quality close to that of wireline telephony (G.726 32 kbit/s ADPCM used as a wireline reference) and provides a substantial improvement over the quality of the current speech channel. The improved speech quality is not only achieved in error-free conditions, but also in typical cellular operating conditions including transmission errors, environmental noise, and tandeming of speech codecs.