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 full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality
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 in the final standardized codec. Coupled with jitter buffer management, the EVS codec provides robustness against late or lost packets. The advantages of the new EVS codec over reference codecs are further discussed based on listening test results
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 on the permutation representation and easy indexing of vectors in the codebook. The proposal consists of constrained composite permutation codes, which are distinguished by the number of components quantized into each quantization level. This scheme makes the output bit stream take the same form as a parallel array of scalar quantization (SQ). Simulation results show that the proposal almost matches the performance of VQ at 2 bit/scalar bitrates with lower computational complexity. Its structure yields the efficient representation of tones that are important for auditory perception.
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 synthesis aims for sound quality improvement in case of multiple packet loss, especially for noisy signals. Another enhancement consists of a guided LP concealment approach to limit the risk of creating artifacts during recovery. These enhancements are also used in the presented advanced TCX concealment method. Subjective listening tests show that quality is significantly increased with these methods
scite is a Brooklyn-based organization that helps researchers better discover and understand research articles through Smart Citations–citations that display the context of the citation and describe whether the article provides supporting or contrasting evidence. scite is used by students and researchers from around the world and is funded in part by the National Science Foundation and the National Institute on Drug Abuse of the National Institutes of Health.
hi@scite.ai
10624 S. Eastern Ave., Ste. A-614
Henderson, NV 89052, USA
Copyright © 2024 scite LLC. All rights reserved.
Made with 💙 for researchers
Part of the Research Solutions Family.