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
TL;DR: A novel steganography algorithm is proposed that performs well in imperceptibility with a hiding capacity of 550 bits/s and the real-time and anti-detection performances are also satisfactory.
Abstract: The rapid development of speech communication technology has made it possible for low bit-rate speech to become appropriate steganographic cover media. To incorporate data hiding into the low bit-rate speech codec, a novel steganography algorithm is proposed in this paper. By analyzing the encoding rule of fixed codebook vector, the way of transposing encoding locations of adjacent pulses is found to be suitable for data embedding with good imperceptibility. Based on encoding location transposition of adjacent pulses, the relationship between adjacent pulse locations is used to embed secret data while the fixed codebook search is being conducted during the encoding process of G.729 codec, which can maintain synchronization between data embedding and speech encoding. The experimental results demonstrate that the proposed steganography algorithm performs well in imperceptibility with a hiding capacity of 550 bits/s. Furthermore, the real-time and anti-detection performances are also satisfactory.
17 citations
12 May 1996
TL;DR: A novel approach to a two-stage wavelet packet based scalable audio coding system is presented and two different structures have been designed and implemented, one in the time-domain, and its dual in the wavelet-domain; these are compared with an MPEG based scalable codec.
Abstract: Scalability, a well known concept in video coding, has only recently been introduced to audio coding. In this paper, a novel approach to a two-stage wavelet packet based scalable audio coding system is presented. Two different structures have been designed and implemented, one in the time-domain, and its dual in the wavelet-domain; these are compared with an MPEG based scalable codec. Results at different bit-rates are shown, while trade-offs and limitations together with future developments for further reduced bit-rates are discussed.
17 citations
TL;DR: Experimental results indicate that the coding scheme ensures transparent coding of one channel CD-quality audio signals at bit rates below 64 kbps for most audio signals, and the results confirm that the best way to achieve maximum compression rate and transparent coding is the usage of perceptual-entropy-based decompositions.
17 citations
Patent•
26 Feb 2007
TL;DR: In this article, a method for transcoding a CELP-based compressed voice bitstream from source codec to destination codec is proposed, which includes processing a source codec input cELP bitstream to unpack at least one or more CELPs from the input bitstream and interpolating a plurality of unpacked cELPs.
Abstract: A method for transcoding a CELP based compressed voice bitstream from source codec to destination codec. The method includes processing a source codec input CELP bitstream to unpack at least one or more CELP parameters from the input CELP bitstream and interpolating one or more of the plurality of unpacked CELP parameters from a source codec format to a destination codec format if a difference of one or more of a plurality of destination codec parameters including a frame size, a subframe size, and/or sampling rate of the destination codec format and one or more of a plurality of source codec parameters including a frame size, a subframe size, or sampling rate of the source codec format exist. The method includes encoding the one or more CELP parameters for the destination codec and processing a destination CELP bitstream by at least packing the one or more CELP parameters for the destination codec.
17 citations
19 Apr 2015
TL;DR: The time-domain bandwidth extension (TBE) framework employed to code wideband and super-wideband speech in the newly standardized 3GPP EVS codec shows significantly improved quality compared to the other standardized SWB codecs under both clean speech and speech with background noise.
Abstract: This paper describes the time-domain bandwidth extension (TBE) framework employed to code wideband and super-wideband speech in the newly standardized 3GPP EVS codec. The TBE algorithm uses a nonlinear harmonic modeling technique that incorporates principles of time-domain envelope-modulated noise mixing. At 13.2 kbps, the super-wideband coding of speech uses as low as 1.55 kbps for encoding the spectral content from 6.4–14.4 kHz. Subjective evaluation results from ITU-T P.800 Mean Opinion Score (MOS) tests are provided, showing significantly improved quality compared to the other standardized SWB codecs under both clean speech and speech with background noise.
17 citations