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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Journal Article•
TL;DR: A method for representing error in perceptual audio coding as filtered noise as well as methods for including the coded error data in an audio file in a backwards-compatible manner are discussed.
Abstract: A method for representing error in perceptual audio coding as filtered noise is presented. Various techniques are compared for analyzing and re-synthesizing the noise representation. A focus is placed on improving the perceived audio quality with minimal data overhead. In particular, it is demonstrated that per-critical-band energy levels are sufficient to provide an increase in quality. Methods for including the coded error data in an audio file in a backwards-compatible manner are also discussed. The MP3 codec is treated as a case study, and an implementation of this method is presented.
Patent•
01 Jul 2011
TL;DR: In this paper, a scalable audio codec for a processing device determines first and second bit allocations for each frame of input audio, and the allocations are made on a frame-by-frame basis based on the energy ratio between the two bands.
Abstract: A scalable audio codec for a processing device determines first and second bit allocations for each frame of input audio. First bits are allocated for a first frequency band, and second bits are allocated for a second frequency band. The allocations are made on a frame-by-frame basis based on the energy ratio between the two bands. For each frame, the codec transform codes both frequency bands into two sets of transform coefficients, which are then packetized based on the bit allocations. The packets are then transmitted with the processing device. Additionally, the frequency regions of the transform coefficients can be arranged in order of importance determined by power levels and perceptual modeling. Should bit stripping occur, the decoder at a receiving device can produce audio of suitable quality given that bits have been allocated between the bands and the regions of transform coefficients have been ordered by importance.
Patent•
20 May 2015TL;DR: In this article, an audio coding device includes a memory; and a processor configured to execute a plurality of instructions stored in the memory, the instructions comprising: selecting a main lobe among a small number of lobes detected from a frequency signal configuring an audio signal on a basis of bandwidth and power of the lobes; and coding the audio signal in such a manner that a first amount of bits per a unit frequency domain allocated to coding of the main lobe is larger than a second amount allocated to the coding of a side lobe as a lobe other than the main one.
Abstract: An audio coding device includes a memory; and a processor configured to execute a plurality of instructions stored in the memory, the instructions comprising: selecting a main lobe among a plurality of lobes detected from a frequency signal configuring an audio signal on a basis of bandwidth and power of the lobes; and coding the audio signal in such a manner that a first amount of bits per a unit frequency domain allocated to coding of the frequency signal of the main lobe is larger than a second amount of bits per the unit frequency domain allocated to the coding of the frequency signal of a side lobe as a lobe other than the main lobe.
24 Jun 1988
TL;DR: An overview is given of low bit rate (m*64 kb/s, m=1.6) video coding using a generalized video codec that uses a hybrid coding algorithm combining differential pulse-code modulation and transform coding or vector quantization in the spatial domain to code sequences of spatially and temporally subsampled images.
Abstract: An overview is given of low bit rate (m*64 kb/s, m=1.6) video coding using a generalized video codec. This codec uses a hybrid coding algorithm combining differential pulse-code modulation (DPCM) in the temporal domain and transform coding or vector quantization (VQ) in the spatial domain to code sequences of spatially and temporally subsampled images. The coder uses motion detection and estimation for more efficient prediction; the resulting motion vectors are also used in the decoder for motion-compensated interpolation of the images left out in the subsampling process. All the elements of the codec are discussed and some of the options available for each element are noted. >
TL;DR: Implementing the turbo codec on FPGA Virtex 5 by optimizing the logic design and minimizing the area requirements for the codec design has been thereby addressed in this paper.
Abstract: Satellite communication systems require forward error correction techniques for their correct functioning. Satellite communications necessitate an efficient error control technique which can provide high reliability, low coding redundancy, and high coding gain. Coding technique is needed to send or receive information from ground station with minimal or no error. In this paper, turbo coding, a very dominant and robust error correcting coding scheme is presented. The reformed Viterbi algorithm called SOVA (Soft Output Viterbi Algorithm) which is used at the decoder is also presented. Efficiently designing the turbo codec based on CCSDS defined standards and parameters has been the focus of this paper. The modern satellite communication systems also demand low power and lesser area design techniques. Optimization of area reduces the delay and henceforth increases the device speed which is a critical parameter while designing satellites and employing the error control techniques in spacecraft. Implementing the turbo codec on FPGA Virtex 5 by optimizing the logic design and minimizing the area requirements for the codec design has been thereby addressed in