scispace - formally typeset
Search or ask a question
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.


Papers
More filters
Patent
01 Sep 2008
TL;DR: In this article, a shadow codec is proposed to snoop the audio data and commands on the High Definition Audio (HDA) bus that are targeted to the conventional codec to generate a second audio output.
Abstract: Systems and methods for “shadowing” a target codec to provide additional features that are not available in the target codec. In one embodiment, an audio amplification system includes a High Definition Audio (HDA) bus, and an HDA controller, a conventional HDA codec and a shadow HDA codec coupled to the HDA bus. The conventional codec receives audio data and commands from the HDA controller via the bus and processes them to generate an output audio signal. The shadow codec snoops the audio data and commands on the HDA bus that are targeted to the conventional codec. The shadow codec processes the snooped audio data and commands to generate a second audio output. The shadow codec does not communicate with the HDA controller and is transparent to the controller. The shadow codec does not request enumeration from the HDA controller and does not receive an address from the HDA controller.

10 citations

Proceedings ArticleDOI
22 Aug 1999
TL;DR: A high quality audio coding scheme based on a novel hybrid algorithm combining warped linear prediction (WLP) and subband coding is proposed, able to provide a performance comparable to that of the MPEG audio layer II while operating at a lower bit rate.
Abstract: In this paper, a high quality audio coding scheme based on a novel hybrid algorithm combining warped linear prediction (WLP) and subband coding is proposed. In the proposed scheme, subband coding is used to quantize the WLP residual. A modified bit allocation algorithm is used for subband bit allocation so that the overall quantization noise to masking threshold ratio (NMR) in the coded signal is minimized. The proposed codec is capable of providing high quality audio output at low bit rate. Subjective tests have shown that the proposed codec is able to provide a performance comparable to that of the MPEG audio layer II while operating at a lower bit rate.

10 citations

Proceedings ArticleDOI
21 Aug 2000
TL;DR: A high-speed speech codec which can process concurrently 18 voice channels with a single TMS320C6201 chip in the IP telephony gateway is implemented and the performance of the resulting ITU-T G.723.1 speech codec is summarized.
Abstract: This paper describes how to implement the G.723.1 recommendation in the IP telephony gateway and studies in detail the programming of the TMS320C6201 DSP and optimization methods for reducing the speech processing delay of the G.723.1 codec. As a result of adopting these optimization and programming methods, we have implemented a high-speed speech codec which can process concurrently 18 voice channels with a single TMS320C6201 chip in the IP telephony gateway. Finally, the paper summarizes the performance of the resulting ITU-T G.723.1 speech codec.

10 citations

Proceedings ArticleDOI
28 Dec 2015
TL;DR: Stereo coding aspect of this block is demonstrated that, by using specially chosen spectral configurations when deriving the parametric side-information in the encoder, perceptual artifacts can be reduced and the spatial processing in the decoder can remain real-valued.
Abstract: Traditional audio codecs based on real-valued transforms utilize separate and largely independent algorithmic schemes for parametric coding of noise-like or high-frequency spectral components as well as channel pairs. It is shown that in the frequency-domain part of coders such as Extended HE-AAC, these schemes can be unified into a single algorithmic block located at the core of the modified discrete cosine transform path, enabling greater flexibility like semi-parametric coding and large savings in codec delay and complexity. This paper focuses on the stereo coding aspect of this block and demonstrates that, by using specially chosen spectral configurations when deriving the parametric side-information in the encoder, perceptual artifacts can be reduced and the spatial processing in the decoder can remain real-valued. Listening tests confirm the benefit of our proposal at intermediate bit-rates.

10 citations

Book ChapterDOI
27 Aug 2012
TL;DR: This paper proposes an adaptive end-to-end based codec switching scheme that fully conforms to the SIP standard, and evaluation with a real-world prototype based on Linphone shows that the scheme adapts well to changing network conditions, improving overall speech quality.
Abstract: Contemporary Voice-Over-IP (VoIP) systems typically negotiate only one codec for the entire VoIP session life time. However, as different codecs perform differently well under certain network conditions like delay, jitter or packet loss, this can lead to a reduction of quality if those conditions change during the call. This paper makes two core contributions: First, we compare the speech quality of a set of standard VoIP codecs given different network conditions. Second, we propose an adaptive end-to-end based codec switching scheme that fully conforms to the SIP standard. Our evaluation with a real-world prototype based on Linphone shows that our codec switching scheme adapts well to changing network conditions, improving overall speech quality.

10 citations


Network Information
Related Topics (5)
Signal processing
73.4K papers, 983.5K citations
79% related
Data compression
43.6K papers, 756.5K citations
78% related
Decoding methods
65.7K papers, 900K citations
78% related
Computational complexity theory
30.8K papers, 711.2K citations
76% related
Hidden Markov model
28.3K papers, 725.3K citations
75% related
Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202310
202214
20201
20193
20183
201721