The adaptive multi-rate wideband codec: history and performance (original) (raw)
Related papers
The adaptive multirate wideband speech codec (AMR-WB
IEEE Transactions on Speech and Audio Processing, 2002
This paper describes the Adaptive Multirate 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. 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
A candidate proposal for a 3GPP adaptive multi-rate wideband speech codec
2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221), 2001
This paper describes an adaptive multi-rate wideband (AMR-WB) speech codcc proposcd 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 24kbit/s down to S.lkbit/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.
AMR wideband codec - leap in mobile communication voice quality
2001
The Third Generation Partnership Project (3GPP) and European Telecommunications Standards Institute (ETSI) have carried out development and standardisation of a wideband speech codec for GSM and the third generation mobile communication WCDMA system since 1999. The Adaptive Multi-Rate Wideband (AMR-WB) codec algorithm was selected in December 2000, and the corresponding specifications were approved in March 2001. The AMR-WB codec was jointly developed by Nokia and VoiceAge. AMR-WB extends the audio bandwidth from 3.4 kHz to 7 kHz and gives superior speech quality and voice naturalness compared to existing 2nd and 3rd generation mobile communication systems. The wideband speech service provided by the AMR-WB codec will give mobile communication speech quality that even exceeds (narrowband) wireline quality.
An embedded adaptive multi-rate wideband speech coder
2001
This paper presents a multi-rate wideband speech coder with bit rates from 8 to 32 kb/s. The coder uses a splitband approach, where the input signal, sampled at 16 kHz, is split into two equal frequency bands from 0-4 kHz and 4-8 kHz, each of which is decimated to an 8 kHz sampling rate. The lower band is coded using the adaptive multi-rate (AMR) family of high-quality narrowband speech coders, while the higher band is represented by a simple but effective parametric model. A complete solution including this wideband speech coder, channel coding for various GSM channels, and dynamic rate adaptation, easily passed all Selection Rules and ranked second overall in the 3GPP AMR Wideband Selection Testing. Besides the high performance, additional advantages of the embedded split-band approach include ease of implementation, reduced complexity, and simplified interoperation with narrowband speech coders
International Conference on Acoustics, Speech, and Signal Processing, 2004
Description and design of the source-controlled variable-rate multimode wideband (VMR-WB) codec recently selected by the 3 rd Generation Partnership Project 2 (3GPP2) for the cdma2000® system in Rate-Set II are presented. This paper gives an overview of the codec and the methodologies that enabled high quality wideband coding at average data rates ranging from TIA/EIA/IS-733 ADR to that of TIA/EIA/IS-127. The codec has three modes of operation at different average data rates and a fourth mode that is interoperable with 3GPP/AMR-WB (ITU-T/G.722.2). Despite the interoperability constraint, the codec was capable of meeting the aggressive performance requirements through the use of novel techniques such as noise suppression, efficient signal classification, new coding types optimized for stable voiced and unvoiced frames, novel post-processing technique for periodicity enhancement in the lower frequency band, and improved frame erasure concealment mechanisms.
2004 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2004
Description and design of the source-controlled variable-rate multimode wideband (VMR-WB) codec recently selected by the 3 rd Generation Partnership Project 2 (3GPP2) for the cdma2000® system in Rate-Set II are presented. This paper gives an overview of the codec and the methodologies that enabled high quality wideband coding at average data rates ranging from TIA/EIA/IS-733 ADR to that of TIA/EIA/IS-127. The codec has three modes of operation at different average data rates and a fourth mode that is interoperable with 3GPP/AMR-WB (ITU-T/G.722.2). Despite the interoperability constraint, the codec was capable of meeting the aggressive performance requirements through the use of novel techniques such as noise suppression, efficient signal classification, new coding types optimized for stable voiced and unvoiced frames, novel post-processing technique for periodicity enhancement in the lower frequency band, and improved frame erasure concealment mechanisms.
IEEE Communications Magazine, 2006
This article is an overview of the architecture and operation of the VMR-WB5 a source- and network-controlled variable-rate multimode codec designed for robust processing of wideband speech. To enable a smooth transition from legacy narrowband voice services, VMR-WB is also capable of processing conventional telephone-bandwidth speech. The VMR-WB codec is interoperable with AMR-WB at certain bit rates, thus eliminating quality degradation and additional delay due to transcoding
ADVANCES IN SOURCE-CONTROLLED VARIABLE BIT RATE WIDEBAND SPEECH CODING
2000
This paper presents novel techniques for source controlled variable rate wideband speech coding. These techniques have been used in the variable-rate multimode wideband (VMR-WB) speech codec recently selected by 3GPP2 for wideband (WB) speech telephony, streaming, and multimedia messaging services in the cdma2000 third generation wireless system. The coding algorithm contains several innovations that enable very good performance at average bit rates as low as 4.0 kbit/s in typical conversational operating conditions. These innovations include: Efficient noise suppression algorithm, signal classification and rate selection algorithm that enables high quality operation at low average bit rates, efficient post-processing techniques tailored for wideband signals, and novel frame erasure concealment techniques including supplementary information for reconstruction of lost onsets and improving decoder convergence. Further, the coder utilizes efficient coding types optimized for different classes of speech signal including a generic coding type based on AMR-WB for transients and onsets, voiced coding type optimized for stable voiced signals and utilizing novel signal modification procedure resulting in good wideband quality at 6.2 kbit/s, unvoiced coding types optimized for unvoiced segments, and efficient comfort noise generation coding. The article describes in detail some of the codec novel features.
Study and Performance of AMR codecs for GSM
IJARCCE
In wireless communication system, limited bandwidth and power is the primary restriction. The existing wireless systems involved in transmission of speech visualized that efficient and effective methods be developed to transmit and receive the same while maintaining quality of speech, especially at the receiving end. Speech coding technique is a material of research for the scientific and academic community since the era of digitization (digital). Amongst all elements of the communication systems (transmitter, channel and receiver), transmission channel is the most critical and plays a key role in the transmission and reception of information. The quality of speech at receiver end decides by channel conditions. Modelling a channel is a multifarious task. A number of techniques are adopted to alleviate the effect of the channel. Adaptive Multi Rate is one of the techniques that neutralize the deleterious effect of the channel on speech. This technique utilizes variable bit rate that dynamically switches to specific modes of operation depending upon the channel conditions. For example, Low bit rate mode of operation is selected in adverse channel conditions, this helps to provide more error protection bits for channel coding and vice versa. Therefore, in this paper, application of Code Excited Linear Prediction (CELP) source codec on speech followed by AMR codec is studied. Further, higher the bit rate used, the better is the quality of speech. In this paper apart from speech codec about AMR is also studied that why the AMR is proposed for the GSM, how the bits rates are reduced in AMR, operation of AMR and other applications of AMR.