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.

Performance evaluation of EVRC-encoded voice traffic over CDMA EVDO Rev. A

Abstract: Voice traffic is not efficiently supported by CDMA EVDO Rev. A from its original design, but an efficient approach is needed to support voice traffic over CDMA EVDO Rev. A for graceful migration from CDMA2000 1x to CDMA EVDO Rev. A as voice service revenue is still the dominant income source for cellular service providers. To understand the system design tradeoffs, we first investigate the traffic characteristics of EVRC-encoded real voice traces and we find that the correlation structure of voice traffic exhibits strong long-range dependence. We further observe that the burst length distribution of voice traffic is heavy-tailed. Traditional Markov-model could not accurately characterize the true nature of the voice traffic. In the major part of our study, we examine the tradeoff between the initial play out delay and voice quality for real voice traces over CDMA EVDO Rev. A. We also examine the tradeoff between voice quality and RF conditions, e.g., the location of mobiles and the number of mobiles in a sector, etc. Finally, we address the performance aspects of voice traffic over CDMA EVDO Rev. A with adaptive frame bundling (AFB) techniques.

Wavelet scalable speech coding using algebraic quantization

Abstract: This paper proposes a new structure for a scalable codec. Our proposed codec works with 10 ms input frame for wideband speech and audio signals at bit rates ranging from 8 to 32 kbit/s. The core layer is the ITU-T G.729 at 8 kbit/s producing a narrowband output. The first enhancement layer is a band-width extension providing a wideband output with 2 kbit/s. The second enhancement layer is based on algebraic quantization of wavelet packet coefficients and improves gradually the synthesized signal as the bitrate increases. For speech signals, at bitrates of 24 and 32 kbit/s, the codec is shown to be equivalent to the ITU-T G.722 codec at 56 and 64 kbit/s, respectively. Moreover, the codec at 32 kbit/s is assessed to be equivalent to the recently standardized embedded codec ITU-T G.729.1 at the same bitrate with a lower algorithmic delay.

New piggybacking algorithm in VoIP using enhanced G.722.2 codec with larger frames

Abstract: In this paper, we present the design of a new piggybacking algorithm for VoIP implemented using the G.722.2 codec. In piggybacking, multiple speech frames that include those transmitted in the past are encapsulated in a single packet. Because redundant copies of each frame are transmitted to the receiver, the receiver can recover those lost frames when one or more packets are lost or arrive late in their transmission. In this paper, we have enhanced the G.722.2 codec by removing further redundancies in the multiple frames encapsulated in piggybacking and by having only one set of LP coefficients for all the subframes encapsulated. We create multiple versions of the codec, each using a different frame size. Our new codec can encode the multiple frames with little degradation in PESQ, while having substantial bit savings. Its performance is evaluated against the original method of piggybacking over random losses, as well as that using packet traces collected in the PlanetLab.

A low bit rate speech codec using multi band excitation and LPC modeling of harmonic magnitudes for private mobile radio

Abstract: This paper discusses a low-bit-rate speech codec that can be operated in the 3.125-kHz band, which is half the minimum frequency band occupied at present, for use in next-generation private mobile radio. In the speech codec under consideration, the frame length is 20 ms (10 ms × 2 subframes), and the pitch, the harmonic magnitudes, and the voiced/unvoiced (V/UV) information are analyzed and extracted from each subframe. Then, the logarithmic pitch is uniform/prediction quantized. A linear model is constructed by correcting the DC component of the harmonic magnitude, and the harmonic magnitude is quantized separately for the prediction coefficient and the gain. The V/UV information is quantized by means of a representative pattern number. Thus, the speech is encoded with a coding rate of 1.6 kbit/s. Error correction is applied at 1.6 kbit/s. The coding scheme is divided into three classes according to the error sensitivity of the speech coding bits, and is encoded at different coding rate by combining CRC (cyclic redundancy check) and RCPC (rate compatible punctured convolutional code). Finally, clean speech is used and speech quality tests employing the MOS (mean opinion score) and the articulation test are performed in an error generation environment. It is shown that although the MOS under error-free conditions is a little worse than FS1016 (CELP, 4.8 kbit/s), the quality degradation is slow when errors are generated. Single sound articulation above 80% is obtained for an error rate of 7%, which is considered sufficient for speech communications. © 2004 Wiley Periodicals, Inc. Electron Comm Jpn Pt 2, 88(1): 40–50, 2005; Published online in Wiley InterScience (www.interscience.wiley.com). DOI 10.1002/ecjb.20140

Adaptive multimedia codec parameter adaptation

Abstract: There is provided determining of multimedia codec parameters for a multimedia communications session. A channel quality indicator value representing current throughput characteristics of a communications channel used for the multimedia communications session is acquired. The channel quality indicator value is compared to a previous channel quality indicator value of the communications channel. It is determined, based on the comparing, whether to delay adaptation of values of a set of multimedia codec parameters to be used by a multimedia codec during the multimedia communications session or not, and which values of the set of multimedia codec parameters to adapt.