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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.


Papers
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Proceedings ArticleDOI
15 Mar 1999
TL;DR: The HE-LPC coder has the potential of producing high quality speech at 4.8 kb/s and below and employs a new pitch estimation and voicing technique, and new DCT based LPC and residual amplitude quantization techniques have been developed.
Abstract: The harmonic excitation linear predictive speech coder (HE-LPC) is a technique derived from MBE and MB-LPC type of speech coding algorithms. The HE-LPC coder has the potential of producing high quality speech at 4.8 kb/s and below. This coder employs a new pitch estimation and voicing technique. In addition, new DCT based LPC and residual amplitude quantization techniques have been developed. The 4 kb/s HE-LPC coder with a 14th order LPC filter was found to produce much better speech quality than the various low rate speech coding standards, including 3.6 kb/s INMARSAT Mini-M AMBE vocoder. During formal ITU ACR test, the 4 kb/s HE-LPC vocoder was found to produced equivalent performance to 32 kb/s ADPCM and G.729 for both flat and modified IRS filtered clean input speech conditions. The HE-LPC algorithm can also be extended to cover bit rates between 1.2 and 8 kb/s range depending on the application.

12 citations

Proceedings Article
01 Jan 2001
TL;DR: A perceptually-based upper bound for phase errors of spectral components of spectral Components is introduced, which delivers good quality for narrowband speech and non-speech inputs.
Abstract: This paper presents a bi-modal coding paradigm to compress narrowband audio signals at 8 kbit/s. In the general mode, the Enhanced Narrowband Audio Coder (ENPAC) exploits the characteristics of the human hearing system to adaptively code the perceptually important spectral components of the input audio. The other mode is employed to handle audio inputs with a strong harmonic structure. In that mode, the input block is represented by its audible harmonics. The spectral magnitude is modeled by the linear prediction analysis in the time domain. The phase of each harmonic is predicted and the phase residues are quantized using an adaptive bit allocation algorithm. This paper introduces a perceptually-based upper bound for phase errors of spectral components. The ENPAC encoder delivers good quality for narrowband speech and non-speech inputs.

12 citations

Proceedings ArticleDOI
25 Mar 2012
TL;DR: The method uses a number of speech features which are then used to train a CART classifier and can identify a codec and its bit rate to an accuracy of 92% and detect the presence of a codec with a accuracy of 97% at -5 dB SNR.
Abstract: We present a non-intrusive data driven method for codec detection and identification in the presence of background noise. The method uses a number of speech features which are then used to train a CART classifier. We demonstrate the performance of the method using several different noise types over a wide range of SNRs. Our results show that we can identify a codec and its bit rate to an accuracy of 92% and we are able to detect the presence of a codec with an accuracy of 97% at −5 dB SNR.

12 citations

Proceedings ArticleDOI
28 Dec 2015
TL;DR: Objective measures show that a more reliable switching decision is achievable and a reliable speech and music discriminator (SMD) for such an application is designed.
Abstract: Switching between speech coding and generic audio coding schemes was recently proven to be very efficient for coding a large range of audio materials at low bit-rates. However, it strongly relies on a robust classification of the input signal. The aim of the paper is to design a reliable speech and music discriminator (SMD) for such an application. Main attention was laid on getting a good tradeoff between accuracy, reactivity and stability of the decision while keeping the delay and complexity reasonably low. To this end, short-term and long-term features are dissociated before being conveyed to two different classifiers. The two classifier outputs are combined in a final decision using a hysteresis. Objective measures show that a more reliable switching decision is achievable. The SMD was successfully implemented in MPEG Unified Speech and Audio Coding (USAC). It allows the codec to show unprecedented audio quality.

12 citations

Proceedings ArticleDOI
27 Aug 2007
TL;DR: Efficient methods to provide the full wideband frequency bandwidth already for the lower bit rates of 8 and 12 kbit/s while maintaining interoperability with the standard implementation of G.729.1 are investigated.
Abstract: This paper discusses a potential extension of the ITU-T G.729.1 speech and audio codec. The G.729.1 coder is hierarchically organized, i.e., the obtained quality increases with the amount of bits that is received for each frame. In particular, the bit rates of 8 and 12 kbit/s offer narrowband (50 Hz – 4 kHz) speech transmission. With a received bit rate of at least 14 kbit/s, the output bandwidth is increased to the wideband frequency range (50 Hz – 7 kHz). Here, we investigate efficient methods to provide the full wideband frequency bandwidth already for the lower bit rates of 8 and 12 kbit/s while maintaining interoperability with the standard implementation of G.729.1. These techniques are not necessarily limited to G.729.1 and thus may serve in other applications as well.

12 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202310
202214
20201
20193
20183
201721