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.

High Quality Coding of Wideband Speech at 24 Kbit/s

Abstract: This paper proposes a Wideband-CELP-Coding scheme (bandwidth 7kHz) at 24 kbit/s. The codec introduces a delay of just 10 ms. This fulfills the requirements of a possible codec candidate for wideband speech coding within DECT or video applications [1]. The analysis-by-synthesis structure of the proposed Wideband-CELP-Codec includes an alternative LPC analysis concept, where the autocorrelation function is calculated recursively [2]. This special LPC scheme provides an improved speech quality and a reduction of computational complexity in comparison to conventional algorithms for the LPC analysis. In addition a stochastic sparse codebook with extremely low computational effort is presented, comparable to the one presented in [12] resulting in a neglectable amount of storage. The CCITT G.722 standard was applied as reference codec, in order to compare the new coding scheme in terms of subjective quality. With the proposed Wideband-CELP a speech quality is achieved, which is equivalent to the reference codec operating at 56 kbit/s.

Variable bit rate speech encoding after gain suppression

Abstract: A speech coding apparatus having a speech input unit for receiving input speech, a speech coding rate selector for selecting an appropriate speech coding rate according to the power of the input speech, a speech analyzer for processing the input speech to estimate a transfer function of the speaker's oral cavity, and a speech coding unit forming a synthesis filter based on the transfer function of the oral cavity. The speech coding unit also codes an excitation signal of the synthesis filter on the basis of an estimation result supplied by the speech analyzer. A gain suppressor interposed between the speech input unit and the speech coding unit suppresses the gain of a signal supplied from the speech input unit to the speech coding unit during an unvoiced period according to information from the speech coding rate selector.

Gain smoothing for speech coding

Abstract: PROBLEM TO BE SOLVED: To provide an optimal speech codec for solving the problem that, although a speech signal contains a considerable amount of noise contents, it is often difficult to suitably make a model of noise in a conventional method for coding noise, resulting in undesirable interrupt and discontinuity including in speech, and that background noise is not suitably coded in a speech coder of analysis by synthesis, especially in a reduced bit rate. SOLUTION: A decoder processing circuit corrects a gain, and vector contribution from an adaptive code book 515 is emphasized in Block 527. Adaptive tilt compensation is applied on the vector connected to a target for smoothing an excitation spectrum in Block 529. The decoder processing circuit performs synthesis filtering by using a smoothed excitation signal. COPYRIGHT: (C)2010,JPO&INPIT

Systems and Methods for Controlling Audio Volume in the Processor of a High Definition Audio Codec

Abstract: Systems and methods for controlling the audio volume of an audio signal in an HDA codec having a programmable processor such as a DSP, wherein the codec receives digital audio signals and audio volume control verbs over an HDA bus, and the audio volume levels associated with the audio volume control verbs are used by the processor in the generation pulse width modulated (PWM) output signals, thereby controlling the audio volume levels of the output signals. The processor may be configured to adjust non-volume parameters such as PWM deadtime, in addition to adjusting audio volume, based on the audio volume levels. The codec may be implemented in a PC or other system that implements an HDA system that includes the HDA bus and HDA codec.

A high quality ADM LSI codec at 32 kbit/s for digital speech communications

Abstract: This paper describes a 32kbit/s adaptive delta modulation (ADM) CODEC which can be applied to digital portable telephones. This ADM CODEC employs a new adaptation algorithm and parameters adjusted for speech signals. The encoding algorithm makes it possible to achieve higher quality with the 32 kbit/s ADM CODEC than with 6 bit µ-law PCM coding. The developed ADM CODEC including a coder, a decoder and switched-capacitor filters on a chip can operate on a single power supply(3.6∼ 5V). The power dissipation is only 30 mW during normal operation and less than 10 mW in the power-down mode. The CMOS chip is an a 16-pin package, 7 × 4.3mm.