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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01 Jan 2000
TL;DR: This paper describes a CODEC implementation on an ultra low-power miniature application specific signal processor (ASSP) designed for mobile signal processing applications that consumes only 1 mW in a package small enough for use in a range of demanding portable applications.
Abstract: This paper describes a CODEC implementation on an ultra low-power miniature application specific signal processor (ASSP) designed for mobile a udio signal processing applications. The CODEC records s peech to and plays back from a serial flash memory at data rates of 16 and 8 kb/s, with a bandwidth of 4 kHz. This CODEC consumes only 1 mW in a package small enough for use in a range of demanding portable applications. Results, improvements and applications are a lso d iscussed.
1 citations
01 Aug 2011
TL;DR: This document specifies the payload format for packetization of SPIRIT IP-MR encoded speech signals into the Real-time Transport Protocol (RTP).
Abstract: This document specifies the payload format for packetization of SPIRIT
IP-MR encoded speech signals into the Real-time Transport Protocol
(RTP). The payload format supports transmission of multiple frames per
payload and introduced redundancy for robustness against packet loss.
1 citations
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06 Sep 2015TL;DR: An extensive investigation of state-of-the-art multi-channel signal processing algorithms used for dereverberation, noise reduction and speaker localization based on a microphone array with 16 microphones which are part of a fixed ceiling-mounted device.
Abstract: Automatic speech recognition in a home environment is currently of large interest for many practical applications. It is also challenging for signal processing, since several problems have to be solved satisfactorily. First, a large degradation of speech recognition performance is caused by room reverberations. Second, environmental noises like a radio, television or kitchen devices may further degrade speech recognition. And third, the problem of concurrent and possibly moving speakers has to be addressed. In this paper we present our results of an extensive investigation of state-of-the-art multi-channel signal processing algorithms used for dereverberation, noise reduction and speaker localization. The investigation is based on a microphone array with 16 microphones which are part of a fixed ceiling-mounted device. All signal processing algorithms are implemented on a PC platform to run in real-time and produce an enhanced signal, which is finally used for speech recognition.
1 citations
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TL;DR: The characteristic, performance and application of TLV320AIC23, which is a high performance stereo audio CODEC with highly integrated analog functionality, are introduced.
Abstract: The characteristic, performance and application of TLV320AIC23, which is a high performance stereo audio CODEC with highly integrated analog functionality,are introduced.How to initiate TLV320AIC23 through I2C and how to set the registers of the MCBSP in DSP according to the characteristic of TLV320AIC23 are described in detail.
1 citations
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01 Oct 2014TL;DR: The quality of the sound and the degree of its damage is evaluated by subjective experiments on the audio that has different coding format and different bit rate compression, in order to sort the quality ofThe sound for different coding scheme of the audio.
Abstract: This article evaluates the quality of the sound and the degree of its damage by subjective experiments on the audio that has different coding format and different bit rate compression, in order to sort the quality of the sound for different coding scheme of the audio. The experimental signals include language, music and environmental sound, so as to show the difference of the quality of the sound between the common audio from different angles.
1 citations