Topic
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.
Papers published on a yearly basis
Papers
More filters
14 May 2006
TL;DR: A technique to extend narrowband (NB) speech communication systems, using e.g. the GSM enhanced full rate (EFR) codec, with wideband (WB, 50-7000 Hz) capability, by applying split vector quantization in a transformed domain.
Abstract: We present a technique to extend narrowband (NB) speech communication systems, using e.g. the GSM enhanced full rate (EFR) codec [1], with wideband (WB, 50–7000 Hz) capability. The limited acoustic bandwidth of narrowband speech coding is extended using a fairly coarse description of the missing high frequency band (3.4–7 kHz) in terms of temporal and spectral envelopes. The high-band parameters are quantized, transmitted and then used at the receiver side to regenerate the high frequency components. The parameter encoding is done by applying split vector quantization in a transformed domain. This quantization scheme can be scaled to match any given target bit rate. Several example configurations have been implemented and tested in MUSHRA-style listening tests.
13 citations
01 Dec 2007
TL;DR: The artifacts introduced by STSA methods will be described, and how the spectral subtraction method is modified to counter these artifacts.
Abstract: The spectral subtraction method is a well-known noise reduction technique Most implementations and variations of the basic technique advocate subtraction of the noise spectrum estimate over the entire speech spectrum However, real world noise is mostly colored and does not affect the speech signal uniformly over the entire spectrum In this paper, the artifacts introduced by STSA methods will describe, and how the spectral subtraction method is modified to counter these artifacts The EVRC noise suppression algorithm will discuss in some detail Finally, some of the methods will compare based on the attenuation given an estimate of the SNR and the results will show with diagrams
13 citations
TL;DR: There are no significant benefits to lowering the mode-sets or deploying dynamic codec rate adaptation, and controlled laboratory experiments indicated that there was an improvement in voice quality when mode-set eight was employed.
Abstract: In this paper, we examine the impact of four voice over long term evolution adaptive multi-rate wideband codec mode-sets on coverage at pedestrian and vehicular speeds Industry-standardized mean opinion scores were used as a metric for voice quality Controlled laboratory experiments simulating pedestrian speeds indicated that there was an improvement in voice quality when mode-set eight was employed At vehicular speeds, mode-set eight outperformed the other mode-sets for path losses less than 130 dB; however, all four mode-sets experienced a significant decline in voice quality when the path loss was greater than 130 dB Based on the current implementations, there are no significant benefits to lowering the mode-sets or deploying dynamic codec rate adaptation
13 citations
27 May 2007
TL;DR: An audio codec for portable application, which supports MPEG-2/MPEG-4 AAC LC profile, implemented using UMC/HJTC 0.18mum CMOS technology, which is capable of real-time encoding/decoding AAC streams at only 5MHz working frequency.
Abstract: In this paper, we present an audio codec for portable application, which supports MPEG-2/MPEG-4 AAC LC profile. The presented codec has been implemented using UMC/HJTC 0.18mum CMOS technology. In this implementation, optimized algorithms are developed and most memory and calculation units are shared between encoder and decoder so that the features of low power and low cost are achieved. The codec uses a total of 160k gates, 21KB RAM and 1.7KB ROM. Experiment shows that it is capable of real-time encoding/decoding AAC streams at only 5MHz working frequency. The codec's interface is simple so that it can be reused as IP macro and easily embedded into various multimedia systems
12 citations
TL;DR: The subjective evaluation results show that the speech quality of the proposed codec is equivalent to that of state-of-the-art codec, G.718, under both a clean channel condition and lossy channel conditions, which is significant considering that development of the proposal is still in early stage.
Abstract: High quality speech at low bit rates makes code excited linear prediction (CELP) the dominant choice for a narrowband coding technique despite the susceptibility to packet loss. One of the few techniques which received attention after the introduction of CELP coding technique is the internet low bitrate codec (iLBC) because of inherent high robustness to packet loss. Addition of rate flexibility and scalability makes the iLBC an attractive choice for voice communication over IP networks. In this paper, performance improvement schemes of multi-rate iLBC and its scalable structure are proposed, and the proposed codec enhanced from the previous work is re-designed based on the subjective listening quality instead of the objective quality. In particular, perceptual weighting and the modified discrete cosine transform (MDCT) with short overlap in weighted signal domain are employed along with the improved packet loss concealment (PLC) algorithm. The subjective evaluation results show that the speech quality of the proposed codec is equivalent to that of state-of-the-art codec, G.718, under both a clean channel condition and lossy channel conditions. This result is significant considering that development of the proposed codec is still in early stage.
12 citations