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.

Audio codec post-filter

Abstract: Techniques and tools are described for processing reconstructed audio signals. For example, a reconstructed audio signal is filtered in the time domain using filter coefficients that are calculated, at least in part, in the frequency domain. As another example, producing a set of filter coefficients for filtering a reconstructed audio signal includes clipping one or more peaks of a set of coefficient values. As yet another example, for a sub-band codec, in a frequency region near an intersection between two sub-bands, a reconstructed composite signal is enhanced.

Soft-clipping postprocessor scaling decoded audio signal frame saturation regions to approximate original waveform shape and maintain continuity

Abstract: An audio coder/decoder ("codec") that is suitable for real-time applications due to reduced computational complexity, and a novel adaptive sparse vector quantization (ASVQ) scheme and algorithms for general purpose data quantization. The codec provides low bit-rate compression for music and speech, while being applicable to higher bit-rate audio compression. The codec includes an in-path implementation of psychoacoustic spectral masking, and frequency domain quantization using the novel ASVQ scheme and algorithms specific to audio compression. More particularly, the inventive audio codec employs frequency domain quantization with critically sampled subband filter banks to maintain time domain continuity across frame boundaries. The input audio signal is transformed into the frequency domain in which in-path spectral masking can be directly applied. This in-path spectral masking usually results in sparse vectors. The ASVQ scheme is a vector quantization algorithm that is particularly effective for quantizing sparse signal vectors. In the preferred embodiment, ASVQ adaptively classifies signal vectors into six different types of sparse vector quantization, and performs quantization accordingly. The ASVQ technique applies to general purpose data quantization as well as to quantization in the context of audio compression. The invention also includes a "soft clipping" algorithm in the decoder as a post-processing stage. The soft clipping algorithm preserves the waveform shapes of the reconstructed time domain audio signal in a frame- or block-oriented stateless manner while maintaining continuity across frame or block boundaries. The invention includes related methods, apparatus, and computer programs.

Multimode Embedded Compression Codec Engine for Power-Aware Video Coding System

Abstract: In a typical portable multimedia system, external access, which is usually dominated by block-based video content, induces more than half of total system power. Embedded compression (EC) effectively reduces external access caused by video content by reducing the data size. In this paper, an algorithm and a hardware architecture of a new type EC codec engine with multiple modes are presented. Lossless mode, and lossy modes with rate control modes and quality control modes are all supported by single algorithm. The proposed four-tree pipelining scheme can reduce 83% latency and 67% buffer size between transform and entropy coding. The proposed EC codec engine can save 62%, 66%, and 77% external access at lossless mode, half-size mode, and quarter-size mode and can be used in various system power conditions. With TSMC 0.18 mum 1P6M CMOS logic process, the proposed EC codec engine can encode or decode CIF 30 frame per second video data and achieve power saving of more than 109 mW. The EC codec engine itself consumes only 2 mW power.

Speech codec for the European mobile radio system

Abstract: The speech coding scheme which will be used as the standard for the European mobile radio system has been selected by the CEPT Groupe Special-Mobile (GSM) as a result of formal subjective listening tests. It is based on the regular-pulse excitation linear predictive coding technique (RPE-LPC) combined with long-term prediction (LTP). The solution is called the RPE-LTP codec. The codec algorithm and the error protection scheme are presented. The net bit rate is 13.0 kb/s, and the gross bit rate, including error protection, is 22.8 kb/s. The experimental implementation based on VLSI signal processors is described. The speech quality obtained with the technique considered is far superior to that obtainable with present-day analog mobile radio systems. A duplex speech codec including error protection can be implemented with two VLSI sign processors with external data memories of about 1 K*16 b. >

A 128 Kb/s Hi-Fi audio codec based on adaptive transform coding with adaptive block size MDCT

Abstract: A Hi-Fi audio codec with an improved adaptive transform coding (ATC) algorithm is presented using digital signal processors (DSPs). An audio signal with a 20 kHz bandwidth sampled at 48 kHz is coded at a rate of 128 kb/s. The algorithm utilizes adaptive block size selection, which is effective for preecho suppression. A modified discrete cosine transform (MDCT) with a simple window set is employed to reduce block boundary noise without decreasing the performance of transform coding. In addition, a fast MDCT calculation algorithm, based on a fast Fourier transform, is adopted. Weighted bit allocation is employed to quantize the transformed coefficients. The codec was realized by a multiprocessor system composed of newly developed DSP boards. Subjective tests with the codec show that the coding quality is comparable to that of compact disc signals. >