[Show abstract][Hide abstract] ABSTRACT: 3GPP and ITU-T have standardized a multi-rate codec for wideband speech conversational applications. Following a competitive selection process, the adaptive multirate wideband (AMR-WB) specifications were approved in March 2001 as part of 3GPP release 5. The ITU-T Study group 16 approved the same wideband coding algorithm as Recommendation G.722.2 and its Annexes
[Show abstract][Hide abstract] ABSTRACT: The standardization of the AMR-WB speech codec by 3GPP opens a new area on audio pre-and post-processing integrated in telecommunication terminals. It appears that the transition from narrowband to wideband signals is not a trivial task, but can rather be seen as a challenge, as acoustic quality requirements, scalability and computational limitations have to be faced at the same time. This article presents key examples to address the transition issue as well as proposals for low-complexity implementations of acoustic echo cancellation and noise reduction
[Show abstract][Hide abstract] ABSTRACT: This paper examines the application of UDP-lite for unequal error detection in packet-switched speech transmission via Internet protocols (voice-over-IP) over UMTS radio channels. Traditionally, UDP is used as transport layer protocol, which contains a checksum that covers the complete packet. Thus, any packet with residual bit errors is discarded. Speech codecs like AMR, however, can tolerate bit errors in less sensitive parts of the bitstream. A more recent development, UDP-lite, provides unequal error detection with a partial checksum that covers only the sensitive parts of a packet. Thus, only packets with errors in important bits are discarded. We compare the use of UDP-lite for UMTS channels with convolutional and channels with turbo coding. The results show that the achievable quality improvement by applying UDP-lite depends on the residual bit error distribution of the chosen UMTS channel coding method. While we determined a quality improvement for channels with convolutional coding, we did not get an improvement for turbo coding. Furthermore, when combined with header compression, the convolutional coder with use of UDP-lite can reach the performance of the turbo coder with use of UDP
Personal, Indoor and Mobile Radio Communications, 2005. PIMRC 2005. IEEE 16th International Symposium on; 10/2005
[Show abstract][Hide abstract] ABSTRACT: The introduction of packet switched (PS) networks (e.g., IMS) creates a need to evaluate speech transmission quality when using the 3GPP default speech codecs. 3GPP SA4 has created a work item in 3GPP Release 6 on "Performance characterization of default codecs for PS conversational application". France Telecom R&D and Siemens proposed a test framework consisting of a UMTS simulator for the air interface and an IP network simulator. ITU-T recommendation P.800 (1996) is used for the quality estimation of the transmission. The real-time conversational test results show that the AMR-NB (adaptive multirate narrow band) and AMR-WB (AMR wide band) speech codecs are well suited for PS conversational applications. Furthermore, the results clearly show a higher understanding when using AMR-WB rather than AMR-NB.
Multimedia and Expo, 2004. ICME '04. 2004 IEEE International Conference on; 07/2004
[Show abstract][Hide abstract] ABSTRACT: In order to make hidden Markov model (HMM) speech recognition suitable for mobile phone applications, Siemens developed a recognizer, Very Smart Recognizer (VSR), for deployment in future mobile phone generations. Typical applications will be name dialling, command and control operations suited for different environments, for example in cars. The paper describes research and development issues of a speech recognizer in mobile devices focusing on noise robustness, memory efficiency and integer implementation. The VSR is shown to reach a word error rate as low as 4.1% on continuous digits recorded in a car environment. Furthermore by means of discriminative training and HMM-parameter coding, the memory requirements of the VSR HMMs are smaller than 64 kBytes.
IEEE Transactions on Speech and Audio Processing 12/2002; 10(8-10):562 - 569. DOI:10.1109/TSA.2002.804548 · 2.29 Impact Factor
[Show abstract][Hide abstract] ABSTRACT: Speech coding at low and medium bit rates benefits from the assumption that the speech signal is stationary inside a subframe or even inside consecutive subframes. Speech signals however comprise a considerable amount of transient segments, such as onsets, which are usually difficult to encode with good quality. This paper proposes a method to improve the encoding of transient (sub-)frames by disregarding the adaptive codebook contribution and strengthening the fixed codebook contribution. This can principally be applied to any CELP-like standard speech coder while keeping the same bit rate. The proposed technique has been successfully employed in the ITU-T 4 kbps speech coder candidate commonly proposed by AT&T, Conexant currently Mindspeed, Deutsche Telekom, France Telecom, Matsushita, NTT, and Siemens in November 2001. Simulation results show the improvements achieved by this approach.
[Show abstract][Hide abstract] ABSTRACT: This paper describes an adaptive multi-rate wideband (AMR-WB) speech codec proposed for the GSM system and also for the evolving third generation (3G) mobile speech services. The speech codec is based on SB-CELP (splitband-code-excited linear prediction) with five modes operating bit rates from 24 kbit/s down to 9.1 kbit/s. The respective channel coding schemes are based on RSC (recursive systematic code) and UEP (unequal error protection). Both, source and channel codec are designed as homogenous as possible to guarantee robust transmission on current and future mobile radio channels.
Proceedings - ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing 10/2001; 2:757-760. DOI:10.1109/ICASSP.2001.941025
[Show abstract][Hide abstract] ABSTRACT: In this paper we describe a noise reduction preprocessing algorithm for the adaptive multirate (AMR) speech codec of the GSM system. The algorithm is based on spectral weighting and explicitly takes into account the properties of the human auditory system. The weighting rule results in the smallest possible speech distortion under the constraint that the background noise should exhibit no audible distortions. The algorithm was implemented in 16 Bit fixed-point arithmetic and submitted to the ETSI AMR noise reduction standardization contest. Compared to other algorithms, our noise reduction method gave very good results in CCR tests and good results in ACR tests.
KONVENS 2000 / Sprachkommunikation, Vorträge der gemeinsamen Veranstaltung 5. Konferenz zur Verarbeitung natürlicher Sprache (KONVENS), 6. ITG-Fachtagung "Sprachkommunikation", 9. bis 12. Oktober 2000, Technische Universität Ilmenau; 01/2000