Topic
Adaptive Multi-Rate audio codec
About: Adaptive Multi-Rate audio codec is a research topic. Over the lifetime, 1467 publications have been published within this topic receiving 19736 citations. The topic is also known as: AMR & Adaptive Multi-Rate.
Papers published on a yearly basis
Papers
More filters
07 Oct 2001
TL;DR: The speech quality of the UMTS adaptive multi-rate speech codec is investigated and relevant results are presented, showing that the speech quality is not affected appreciably by transmission errors when the FER is kept below 1%.
Abstract: We present a methodology and a tool for objective characterization of voice service quality in W-CDMA networks. Specifically, the speech quality of the UMTS adaptive multi-rate speech codec is investigated and relevant results are presented. The results show that the speech quality is not affected appreciably by transmission errors when the FER is kept below 1%. However, the codec can be operated at up to FER=7% in the highest codec mode while keeping a voice quality as good as or better than lower codec rates at lower FER.
7 citations
25 Jun 1992
TL;DR: The paper reviews the recent activities of SGXV/WP2 on 8-kbit/s speech coding algorithm and introduces one of the probable algorithms and presents this working group's plans for future work.
Abstract: The CCITT (International Telegraph and Telephone Consultative Committee) Study Group XV is in charge of studies on speech processing. SGXV has already drafted standard G.728 for a 16-kbit/s low-delay speech coding algorithm and is now studying 8-kbit/s speech coding algorithms. The paper reviews the recent activities of SGXV/WP2 on 8-kbit/s speech coding algorithm. These activities include discussion of the algorithm's application, of performance requirements and objectives, and of the procedure for evaluating the candidate algorithms. It also introduces one of the probable algorithms and presents this working group's plans for future work. >
7 citations
05 Dec 2005
TL;DR: The informal subject test result is given and it is shown that AVS audio codec is enough for Hi-Fi audio applications.
Abstract: AVS Audio Coding Standard is the first standard for Hi-Fi audio in China. The framework of AVS Audio was introduced. Many key technologies are described in details, including long/short window switch decision based on energy and unpredictability, integer MDCT for lossless time-frequency transform, square polar stereo coding, and context-dependent bitplane coding for scalable entropy coding. The informal subject test result is given between AVS audio codec and several dominating audio codecs. It is shown that AVS audio codec is enough for Hi-Fi audio applications.
7 citations
TL;DR: A novel signal modification method for wide-band code-excited linear prediction (CELP) speech codecs to improve pitch prediction at low bit rates and preserves the original time scale at the end of each frame is introduced.
Abstract: This paper introduces a novel signal modification method for wide-band code-excited linear prediction (CELP) speech codecs to improve pitch prediction at low bit rates. The method is enabled only in stable voiced speech frames, and preserves the original time scale at the end of each frame. This feature helps to avoid artifacts and simplifies an encoder implementation. The signal modification includes a classification algorithm as an integral part. The classification algorithm detects the frames most suitable for signal modification and low bit rate coding, and can be employed in a rate selection module of variable bit rate (VBR) codecs. In this paper, the signal modification method is applied in an experimental VBR wide-band speech codec derived from the 3GPP adaptive multirate wideband (AMR-WB) standard (ITU-T Recommendation G.722.2). The codec fulfills the system requirements of IS-95/CDMA2000 Rate Set II, operating at source coding bit rates 12.65, 6.2, and 1.0 kb/s. The signal modification is used in the 6.2 kb/s mode dedicated for voiced speech frames. Listening test results demonstrate the good performance of the proposed method. The signal modification method is used in the Nokia/VoiceAge codec that was declared in April 2003 as the winner of the selection phase in the 3GPP2 CDMA2000 wide-band speech codec standardization.
7 citations
TL;DR: High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme, which achieves high speech quality, close to that of a 7-bit mu -law PCM.
Abstract: High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme. An optimized APC-AB algorithm is studied that reduces processing complexity while maintaining speech quality. The coding algorithm is implemented in two digital signal processors (DSPs). The DSP chips, a framing LSI circuit, a PCM codec, and some peripheral ICs are integrated in each of two compact packages, i.e. codec modules, operating at 16 or 8 kb/s. The codec module size is as small as 80 mm*50 mm*12 mm, and its typical power consumption is 500 mW using 2- mu m CMOS LSI technology. At 16 kb/s this APC-AB codec achieves high speech quality, close to that of a 7-bit mu -law PCM. The codec modules are expected to be used for various applications such as customer premises multiplexers for digital leased lines, digital mobile radio, and stored-and-forward-message systems (voice-mail systems). >
6 citations