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.

Voice activity detection for GSM adaptive multi-rate codec

Abstract: This paper describes the VAD (voice activity detection) for controlling DTX (discontinuous transmission) of the GSM AMR (adaptive multi-rate) speech codec. The algorithm is based on spectral estimation and periodicity detection. The VAD contains a 9-band IIR filter bank, which divides input signals into frequency bands. The signal level at each band is calculated. Background noise is estimated in each sub-band. The VAD decision is computed by comparing input signal level and background noise estimate. The algorithm incorporates novel methods to estimate background noise and to detect periodic components based on open-loop pitch gain. A new method is also derived to detect correlated complex signals like music.

A 90 mW MPEG4 video codec LSI with the capability for core profile

Abstract: A single-chip MPEG4 video codec LSI with 20 Mb embedded DRAM performs a QCIF 15 Hz H.263 codec, a Simple at L1 codec, and Core at L1 decoding. It consumes 90 mW at 54 MHz. This chip integrates a programmable DSP, 8 dedicated hardware engines, and interface units on a 75.68 mm/sup 2/ die using 0.18 /spl mu/m 1.8 V quad-metal CMOS technology.

Systems and methods for interfacing with codecs across an architecture optimized for audio

Abstract: Systems and methods for interfacing with codec(s) on an architecture optimized for audio are described. In one aspect, a device driver accesses an application programming interface (API). The API facilitates communications between the device driver and one or more codec(s) via a controller coupled to the codec(s). The codec(s) and the controller are implemented in an environment that is substantially optimized for audio. Such communication includes, for example, registering for event(s), transferring data to or from the codec(s), obtaining information about the capability of the codec(s), and/or managing bus or codec resources.

A full-bandwidth audio codec with low complexity and very low delay

Abstract: We propose an audio codec that addresses the low-delay requirements of some applications such as network music performance. The codec is based on the modified discrete cosine transform (MDCT) with very short frames and uses gain-shape quantization to preserve the spectral envelope. The short frame sizes required for low delay typically hinder the performance of transform codecs. However, at 96 kbit/s and with only 4 ms algorithmic delay, the proposed codec out-performs the ULD codec operating at the same rate. The total complexity of the codec is small, at only 17 WMOPS for real-time operation at 48 kHz.

The effect of source based rate adaptation extension in AMR-WB speech codec

Abstract: This paper presents a source based rate adaptation concept for AMR wideband speech codec. The source based rate adaptation algorithm selects the multi rate codec mode based on the input speech characteristics and coding parameters to minimise the average bit rate. The presented concept introduces up to 50% reduction in average bit rate without any degradation in speech quality. The benefit of source based adaptation is in increasing the system capacity in conversational services as well as storage size in messaging type of applications.