scispace - formally typeset

Topic

Adaptive Multi-Rate audio codec

About: Adaptive Multi-Rate audio codec is a(n) research topic. Over the lifetime, 1467 publication(s) have been published within this topic receiving 19736 citation(s). The topic is also known as: AMR & Adaptive Multi-Rate.


Papers
More filters
Patent
09 Jun 1998
Abstract: The present invention proposes a new method and apparatus for the enhancement of source coding systems. The invention employs bandwidth reduction (101) prior to or in the encoder (103), followed by spectral-band replication (105) at the decoder (107). This is accomplished by the use of new transposition methods, in combination with spectral envelope adjustments. Reduced bitrate at a given perceptual quality or an improved perceptual quality at a given bitrate is offered. The invention is preferably integrated in a hardware or software codec, but can also be implemented as a separate processor in combination with a codec. The invention offers substantial improvements practically independent of codec type and technological progress.

488 citations

Book
01 Jan 2002
TL;DR: This paper presents a meta-modelling framework for building a Perceptual Audio Decoder that automates the very labor-intensive and therefore time-heavy, and therefore expensive, and expensive, process of Audio Coding.
Abstract: Foreword. Preface. I: Audio Coding Methods. 2. Quantization. 3. Representation of Audio Signals. 4. Time to Frequency Mapping Part I: The PQMF. 5. Time to Frequency Mapping Part II: The MDCT. 6. Introduction to Psychoacoustics. 7. Psychoacoustic Models for Audio Coding. 8. Bit Allocation Strategies. 9. Building a Perceptual Audio Decoder. 10. Quality Measurement of Perceptual Audio Codecs. II: Audio Coding Standards. 11. MPEG-1 Audio. 12. MPEG-2 Audio. 13. MPEG-2 AAC. 14.Dolby AC-3. 15. MPEG-4 Audio. Index.

347 citations

Journal ArticleDOI
Bishnu S. Atal1
TL;DR: A new class of speech coders are described which allow one to realize the precise optimum noise spectrum which is crucial to achieving very low bit rates, but also represent the important first step in bridging the gap between waveform coders and vocoders without suffering from their limitations.
Abstract: Predictive coding is a promising approach for speech coding. In this paper, we review the recent work on adaptive predictive coding of speech signals, with particular emphasis on achieving high speech quality at low bit rates (less than 10 kbits/s). Efficient prediction of the redundant structure in speech signals is obviously important for proper functioning of a predictive coder. It is equally important to ensure that the distortion in the coded speech signal be perceptually small. The subjective loudness of quantization noise depends both on the short-time spectrum of the noise and its relation to the short-time spectrum of the Speech signal. The noise in the formant regions is partially masked by the speech signal itself. This masking of quantization noise by speech signal allows one to use low bit rates while maintaining high speech quality. This paper will present generalizations of predictive coding for minimizing subjective distortion in the reconstructed speech signal at the receiver. The quantizer in predictive coders quantizes its input on a sample-by-sample basis. Such sample-by-sample (instantaneous) quantization creates difficulty in realizing an arbitrary noise spectrum, particularly at low bit rates. We will describe a new class of speech coders in this paper which could be considered to be a generalization of the predictive coder. These new coders not only allow one to realize the precise optimum noise spectrum which is crucial to achieving very low bit rates, but also represent the important first step in bridging the gap between waveform coders and vocoders without suffering from their limitations.

312 citations

Journal ArticleDOI
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.

293 citations

01 Jan 2002
TL;DR: 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 is described.
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.

265 citations

Network Information
Related Topics (5)
Signal processing

73.4K papers, 983.5K citations

79% related
Data compression

43.6K papers, 756.5K citations

78% related
Decoding methods

65.7K papers, 900K citations

78% related
Computational complexity theory

30.8K papers, 711.2K citations

76% related
Hidden Markov model

28.3K papers, 725.3K citations

75% related
Performance
Metrics
No. of papers in the topic in previous years
YearPapers
20201
20193
20183
201721
201645
201558