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This paper presents a novel frequency domain bandwidth extension (BWE) scheme with relaxed synchronization, optimized for coding inactive and music/mixed content signals. The algorithm achieves high subjective quality at low and medium bitrates and it has a low algorithmic delay. The algorithm is part of the 3GPP Enhanced Voice Services (EVS) codec. In addition to the presented algorithm, the EVS...
This paper presents the bandwidth extension (BWE) method developed for the AMR-WB interoperable (AMR-WB IO) modes of the 3GPP EVS codec. The low-band signal (0–6.4 kHz) is coded using an enhanced version of ACELP as in AMR-WB and post-processed; the high-band (above 6.4 kHz) in contrast to AMR-WB is represented with a new BWE method. The decoded low-band excitation is adaptively extended to high frequencies...
This paper describes the time-domain bandwidth extension (TBE) framework employed to code wideband and super-wideband speech in the newly standardized 3GPP EVS codec. The TBE algorithm uses a nonlinear harmonic modeling technique that incorporates principles of time-domain envelope-modulated noise mixing. At 13.2 kbps, the super-wideband coding of speech uses as low as 1.55 kbps for encoding the spectral...
A Discontinuous transmission (DTX) system, which is widely adopted in speech codecs, is an important function for speech communication systems that can reduce the transmission bandwidth by at least a half. Within a DTX system, the comfort noise generation (CNG) plays a key role in the overall quality. Critical performance parameters with respect to the CNG including the transition quality from active...
In this paper a novel technique is presented to efficiently mix traditional ACELP time domain coding with a frequency domain coding model to improve the quality of generic audio signals coded at low bitrates without additional delay. The paper discusses how to integrate parts of a traditional Algebraic Code Excited Linear Prediction (ACELP) speech codec to create a time-domain contribution which coexists...
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, including support for wideband, super-wideband and...
A highly error resilient mode of the newly standardized 3GPP EVS speech codec is described. Compared to the AMR-WB codec and other conversational codecs, the EVS channel aware mode offers significantly improved error resilience in voice communication over packet-switched networks such as Voice-over-IP (VoIP) and Voice-over-LTE (VoLTE). The error resilience is achieved using a form of in-band forward...
This paper presents a low bit-rate MDCT coder, which is adopted as a part of the recently standardized codec for Enhanced Voice Services. To maximize codec performance for NB to SWB input signals for low bit-rates (7.2 to 16.4 kbps), new adaptive bit-allocation and spectrum quantization schemes, which emphasize perceptually important spectrum while efficiently coding full spectrum, was introduced...
This paper presents high quality monaural superwideband extensions to G.711.1 and G.722, recently standardized as Recommendations ITU-T G.711.1 Annex D and G.722 Annex B. The superwideband (50-14000 Hz) functionality is achieved using embedded scalable structure that adds extension layers on top of the wideband core codecs. The bit rates are extended to 96/112/128 and 64/80/96 kbit/s for G.711.1 and...
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