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.
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08 Dec 2008TL;DR: This paper describes an embedded speech and audio codec which is based on ITU-T Recommendation G.722.1; it can process 7 kHz bandwidthspeech and audio signal at scalable bit rates and adds two modules: the energy ordering of sub-band and the processing of bit-stream truncation.
Abstract: This paper describes an embedded speech and audio codec which is based on ITU-T Recommendation G.722.1; it can process 7 kHz bandwidth speech and audio signal at scalable bit rates. Based on the G.722.1 of ITU-T, this algorithm adds two modules: the energy ordering of sub-band and the processing of bit-stream truncation. Furthermore, it does some modification on the categorization and noise-fill modules. It makes sure that the codec could produce embedded bit-stream, so this codec had more robustness in the transmission. The test results by ITU-T PESQ show that this codec has good performance as G.722.1 at the same bit-rates.
1 citations
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01 Jan 2009TL;DR: This paper describes an efficient way to exploit arithmetic coding to entropy compress quantized spectral magnitudes of the sub-band FDLP residuals and provides 11% bit-rate reduction compared to the Huffman coding algorithm.
Abstract: A speech/audio codec based on Frequency Domain Linear Prediction (FDLP) exploits auto-regressive modeling to approximate instantaneous energy in critical frequency sub-bands of relatively long input segments. The current version of the FDLP codec operating at 66 kbps has been shown to provide comparable subjective listening quality results to state-of-the-art codecs on similar bit-rates even without employing standard blocks such as entropy coding or simultaneous masking. This paper describes an experimental work to increase compression efficiency of the FDLP codec by employing entropy coding. Unlike conventional Huffman coding employed in current speech/audio coding systems, we describe an efficient way to exploit arithmetic coding to entropy compress quantized spectral magnitudes of the sub-band FDLP residuals. Such an approach provides 11% (∼ 3 kbps) bit-rate reduction compared to the Huffman coding algorithm (∼ 1 kbps).
1 citations
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28 Dec 2011TL;DR: In this paper, an audio coding method and apparatus are disclosed, where the method includes: dividing an input audio signal into a lowband signal and a high-band signal; identifying types of the low-band signals and the highband signals; adaptively allocating a total input rate of the audio signal to the lowband signals.
Abstract: An audio coding method and apparatus are disclosed, where the method includes: dividing an input audio signal into a low-band signal and a high-band signal; identifying types of the low-band signal and the high-band signal; adaptively allocating a total input rate of the audio signal to the low-band signal and the high-band signal according to different coding modes corresponding to the low-band signal and the high-band signal; and coding the low-band signal through a coding mode corresponding to the low-band signal according to the low-band rate, and coding the high-band signal through a coding mode corresponding to the high-band signal according to the high-band rate. In embodiments of the present application, when the low-band signal and the high-band signal are coded, the coding is not performed according to a rate given in the standard or set before the coding, and coding rates are adaptively adjusted according to different types of the signals, thereby improving overall audio coding performance.
1 citations
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22 Aug 2001TL;DR: In this paper, a single coding algorithm based on backward adaptive linear predictive coding (BA LPC) is proposed for compressing wideband speech and audio signals (0-8kHz) operating with scalable bit rates and low delay.
Abstract: The integration of fixed and wireless networks have expanded the range in which speech and audio coders were designed to operate. As a result, current research is focusing on a number of developments including algorithms which are able to adapt to different transmission environments and to operate under multiple constraints of bit rate, complexity, delay, robustness to bit errors and diversity of input signals. In this paper, we propose a single coding algorithm for compressing wideband speech and audio signals (0-8kHz) operating with scalable bit rates and with low delay. The algorithm is based on the backward-adaptive linear predictive coding (BA LPC) technique in conjunction with an efficient closedloop optimised excitation structure consisting of sparse pulses of ternary values. The output bit rates range from 17 to 68kb/s. The scalability feature is achieved by means of discrete quantisation layers representing various levels of enhancements of the base-line coder and also flexibility in terms of complexity and bit allocation requirements depending on the particular application and on the network resources. An evaluation of the performance of the coder operating at 17kb/s is carried out using the G.722 standard as a reference.
1 citations
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TL;DR: This paper describes and implements wideband speech codec algorithms on the DSP56156 digital signal processor and describes the method proposed by the British Telecom Research Laboratory in the GSM Pan-European study on digital cellular land mobile radio.
Abstract: This paper describes and implements wideband speech codec algorithms on the DSP56156 digital signal processor. Two wideband speech codec algorithms are described. The first algorithm is based on the G722 recommendation provided by the International Telegraph and Telephone Consultative Committee Study Group XVIII for 7 kHz audiocoding at 64 kbits/s. To reduce the bit rate and improve the speech quality, a second algorithm is proposed for 7 kHz audiocoding at 40 kbits/s. The second algorithm is based on the method proposed by the British Telecom Research Laboratory in the GSM Pan-European study on digital cellular land mobile radio. The results obtained for the two wideband codec algorithms are included.
1 citations