Enhanced Variable Rate Codec

About: Enhanced Variable Rate Codec is a research topic. Over the lifetime, 357 publications have been published within this topic receiving 4842 citations. The topic is also known as: EVRC.

The adaptive multirate wideband speech codec: system characteristics, quality advances, and deployment strategies

Abstract: Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications

On the architecture, operation, and applications of VMR-WB: the new cdma2000 wideband speech coding standard

Abstract: This article is an overview of the architecture and operation of the VMR-WB5 a source- and network-controlled variable-rate multimode codec designed for robust processing of wideband speech. To enable a smooth transition from legacy narrowband voice services, VMR-WB is also capable of processing conventional telephone-bandwidth speech. The VMR-WB codec is interoperable with AMR-WB at certain bit rates, thus eliminating quality degradation and additional delay due to transcoding

A full-bandwidth audio codec with low complexity and very low delay

Abstract: We propose an audio codec that addresses the low-delay requirements of some applications such as network music performance. The codec is based on the modified discrete cosine transform (MDCT) with very short frames and uses gain-shape quantization to preserve the spectral envelope. The short frame sizes required for low delay typically hinder the performance of transform codecs. However, at 96 kbit/s and with only 4 ms algorithmic delay, the proposed codec out-performs the ULD codec operating at the same rate. The total complexity of the codec is small, at only 17 WMOPS for real-time operation at 48 kHz.

The effect of source based rate adaptation extension in AMR-WB speech codec

Abstract: This paper presents a source based rate adaptation concept for AMR wideband speech codec. The source based rate adaptation algorithm selects the multi rate codec mode based on the input speech characteristics and coding parameters to minimise the average bit rate. The presented concept introduces up to 50% reduction in average bit rate without any degradation in speech quality. The benefit of source based adaptation is in increasing the system capacity in conversational services as well as storage size in messaging type of applications.

System for improved use of pitch enhancement with subcodebooks

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 codec 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. The overall quality of the system is strongly related to the excitation. In order to enhance the excitation, the system contains a fixed codebook comprising several subcodebooks. The invention reveals a way to apply a pitch enhancement efficiently and differently for different subcodebooks without using additional bits. The technique is particularly applicable to selectable mode vocoder (SMV) systems.