Showing papers on "Enhanced Variable Rate Codec published in 2002"

The adaptive multirate wideband speech codec (AMR-WB)

Abstract: This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.

The adaptive multirate wideband speech codec (AMR-WB)

Abstract: This paper describes the adaptive multirate wideband (AMR-WB) speech codec selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000 and the corresponding specifications were approved in March 2001. The AMR-WB codec was also selected by the International Telecommunication Union-Telecommunication Sector (ITU-T) in July 2001 in the standardization activity for wideband speech coding around 16 kb/s and was approved in January 2002 as Recommendation G.722.2. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 50 Hz to 7 kHz and gives superior speech quality and voice naturalness compared to existing second- and third-generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that also substantially exceeds (narrowband) wireline quality. The paper details AMR-WB standardization history, algorithmic description including novel techniques for efficient ACELP wideband speech coding and subjective quality performance of the codec.

Noise Suppression with High Speech Quality Based on Weighted Noise Estimation and MMSE STSA

Abstract: A noise suppression algorithm with high speech quality based on weighted noise estimation and MMSE STSA is proposed. The proposed algorithm continuously updates the noise estimate by noisy speech weighted in accordance with an estimated SNR. The spectral gain is modified with the estimated SNR so that it can better utilize the improvement in noise estimation. Subjective evaluation results show that five-grade mean opinion scores of the new algorithm are improved by as much as 0.93 and 0.35, compared with the original MMSE STSA and the EVRC noise suppression algorithm, respectively.

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.

Changing a codec or MAC size without affecting the encryption key in packetcable communication

Abstract: A technique is described to change the codec or MAC (message authentication code) size in a packet security unit for PacketCable communications during realtime voice transmissions is described. An algorithm that provides fast RC4 key advancing to prevent MIPS (millions of instruction cycles per second) overflow is used to perform codec or MAC size changes. The invention is performed without changing the keying material, where the sender and receiver must continue the RC4 encryption process from its state prior to the codec or MAC size change. A sender needs to preserve continuity of the timestamp across a codec change, since the timestamp reflects realtime. Changing the codec or MAC size is likely to change the frame parameters. To preserve continuity of the RC4 state and the timestamp across the codec/MAC size change, the sender TX and receiver RX generates a new frame number. The new frame number is applied to the first frame generated by the new codec or MAC size.

Internet Low Bit Rate Codec

Abstract: {PG,1 A roughing machine having a support for supporting bottom-up a shoe assembly formed of a last having an insole located on its bottom and an upper mounted thereon with the margin of the upper secured to the bottom of the insole and a roughing tool located above the shoe assembly and yieldably urged downwardly against the shoe assembly so as to rough the upper margin during movement of the support to move the upper margin past the roughing tool. The shoe assembly bottom has its heel seat portion, its shank portion and its forepart portion lying in different planes. The support is so constructed as to be rockable in such a manner that the heel seat portion, the shank portion, and the forepart portion of the upper margin all lie in horizontal planes during their movement past the roughing tool.

The adaptive multi-rate wideband codec: history and performance

Abstract: This paper gives the history and performance of the adaptive multi-rate wideband (AMR-WB) speech codec recently selected by the Third Generation Partnership Project (3GPP) for GSM and the third generation mobile communication WCDMA system for providing wideband speech services. The AMR-WB speech codec algorithm was selected in December 2000, and the corresponding specifications were approved in March 2001. In July 2001, the AMR-WB codec was also selected by ITU-T in the standardization activity for wideband speech coding around 16 kbit/s. The adoption of AMR-WB by ITU-T is of significant importance since for the first time the same codec is adopted for wireless as well as wireline services. AMR-WB uses an extended audio bandwidth from 3.4 kHz to 7 kHz and gives superior speech quality and voice naturalness compared to 2/sup nd/ and 3/sup rd/ generation mobile communication systems.

All assembly implementation of G.729 Annex B speech codec on a fixed point DSP

Abstract: A lot of effort has been spent over the last few years in the development of digital speech coding methods and their subsequent standardization. Algorithms have evolved which provide good quality speech at sub 8 kbps bit rates although at a much higher computational expense. DSP processors have also improved with time providing specific signal processing functionalities aiding in easier codec implementations along with lower power consumption at higher clock speeds. Software development tools and compilers have also improved although they still do not work well in high volume, low cost systems. The cost of development tools may also be prohibitive for nonvendors and at times high level code conversion tools may not be present at all. This paper describes techniques and approaches commonly used to realize such systems where the codec implementation in all assembly is necessary. The specific codec implemented was International Telecommunication Union (ITU-T) G.729 Annex B. The techniques described in this paper are applicable to any other speech codec.

Joint capacity and quality evaluation for AMR telephony speech in WCDMA systems

Abstract: The adaptive multi-rate (AMR) speech codec is the mandatory speech codec for WCDMA systems. The codec supports eight different source rates ranging from 12.2 kbit/s down to 4.75 kbit/s. This paper evaluates different ways of allocating AMR rates to users on the downlink in a WCDMA system. A novel system performance concept is introduced based on a user satisfaction metric that tries to combine the user experience from different speech quality levels as well as events like blocking and dropping. System simulations show that the AMR codec introduces a significant trade-off between capacity and quality for the speech service. By allocating AMR modes based on the system load, the quality and capacity trade-off can be efficiently balanced and high system performance can be achieved for a wide range of offered loads.

Context-based error recovery technique for GSM AMR speech codec

Abstract: GSM AMR speech codec being used for both Internet and mobile networks, robustness to both frame erasures and random bit errors assumes significance. This paper proposes a new context-based error recovery technique for the CELP-based speech codec accomplishing recovery of erased frames, updating decoder state during erasure spells and reliable estimation of codec parameters in case of bit errors. Previous error concealment techniques do not adequately make use of the context in which concealment is being done. The proposed error concealment technique is intended to retrieve and use contextual information for better performance. The method is solely receiver based, uses no look ahead, makes use of implicitly available codec parameters and buffers for parameter estimation and is hence computationally efficient. Segmental Itakura-Saito measure and MOS scores are used to compare the output speech quality of the proposed technique with those of the basic techniques as recommended by the standard.

Improved performance speech codec for mobile communications.

Abstract: This paper presents the application of a Voice Gender Normalization algorithm to the GSM Speech Codec and describes the refinements that can be made to the Codec as a result. By reducing the dynamic range of the speech signals entering the Codec gender specific adaptations can be made to the Codec to improve its performance in terms of subjective sound quality or its transmitted bit rate.

Method of Providing Chorus Effect Using EVRC Vocoder in Karaoke Service through Mobile Phone and System therefor

Abstract: PURPOSE: A method for supporting an effective sound using an EVRC vocoder in a Noraebang(karaoke) service through a cellular phone and a system therefor are provided to support the effective sound using the EVRC vocoder and have a chorus effect using only a software without adding a certain device to the cellular phone in the Noraebang service through the cellular phone. CONSTITUTION: A user connects a wireless Internet service to a server of a mobile communication company connected to a CP(Contents Provider) server in which Noraebang contents are stored using a cellular phone. The user selects a Noraebang service. The user selects the Noraebang contents for being downloaded in the cellular phone. The user receives the Noraebang contents from the CP server. The user executes the downloaded Noraebang contents by a program through the cellular phone.