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.

Lossless Compression at Zero Delay of the Electrical Stimulation Patterns of Cochlear Implants for Wireless Streaming of Audio Using Artificial Neural Networks

Abstract: Cochlear implants (CIs) are battery-powered, surgically implanted hearing-aids capable of restoring a sense of hearing in people suffering from moderate to profound hearing loss. To achieve this, audio signals captured by the microphone of the CI are processed by its signal processor and converted into electrical pulses, the stimulation patterns, which then excite certain areas of the cochlear. Nowadays wireless transmission of audio from external devices, like remote microphones and smartphones, is used to improve speech understanding and localization or for the convenience of the CI user. To conserve energy or channel capacity in this wireless transmission, data compression is commonly applied. In this work, zero delay lossless compression of the so called clinical units of the CIs is proposed and a zero delay lossless codec (ZDLLC) based on artificial neural networks is investigated for this purpose. The ZDLLC is compared to the lossless compression algorithms PAQ and PPM as well as the lossy Opus audio codec. On the TIMIT speech corpus and various acoustic scenarios the ZDLLC achieved a mean bitrate of 28.6 kbit/s at zero algorithmic latency compared to 33.6 kbit/s to 35.2 kbit/s for the Opus audio codec at 5 ms to 7.5 ms algorithmic latency. In contrast, at very high latency, PPM achieved a mean bitrate of 37.3 kbit/s and PAQ achieved a mean bitrate of 25.1 kbit/s. It was found that lossless compression of the stimulation patterns could be useful for wireless streaming of audio.

Performance characterization of VMR-WB, the cdma2000 wideband speech codec

Abstract: The voice services sub-working group of 3GPP2 conducted a standardization effort to select and characterize the first wideband speech codec for cdma2000reg. This article describes the selection process, the results of the selection process, and the subsequent tests to characterize the performance of the selected wideband speech codec, VMR-WB

Progressive multichannel audio codec (PMAC) with rich features

Abstract: MPEG Advanced Audio Coding (AAC) is one of the most distinguished multichannel digital audio compression systems. Based on AAC, we develop a progressive syntax-rich multichannel audio codec in this work. It not only supports fine grain bit rate scalability for multichannel audio bitstreams but also provides several desirable functionalities. MPEG-4 version-2 audio coding supports fine grain bit rate scalablility in its Generic Audio Coder with the Bit-Sliced Arithmetic Coding (BSAC) tool that provides scalability for mono or stereo audio, but not for multichannel audio. A formal subjective listening test shows that, the proposed algorithm achieves a better performance at several different bit rates when compared with MPEG-4 BSAC for mono audio sources.

Data transfer in audio codec controllers

Abstract: An audio codec control technique is provided with improved multichannel data ordering capabilities. An audio codec controller comprises a first interface unit for performing data transfer to and from an audio codec, a second interface unit for performing data transfer from an external memory, and a data buffer for buffering data received from the external memory via the second interface unit. The controller further comprises a capture register for receiving from the data buffer data requested by the audio codec, and temporarily storing the received data. The first interface unit is connected to receive temporarily stored data from the capture register. The operation of the audio codec controller may be done in several operational modes including 2, 4, and 6-channel full-rate and half-rate modes.