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
Proceedings ArticleDOI
04 Jun 2016
TL;DR: A steganography scheme using a 3D-sudoku matrix to enlarge capacity and improve quality of speech and theoretical analysis is provided to demonstrate that the concealment and the hidden capacity are greatly improved with the proposed scheme.
Abstract: Redundant information of low-bit rate speech is extremely small, thus it's very difficult to implement large capacity steganography on the low-bit rate speech. Based on multiple vector quantization characteristics of the Line Spectrum Pair (LSP) of the speech codec, this paper proposes a steganography scheme using a 3D-sudoku matrix to enlarge capacity and improve quality of speech. A cyclically moving algorithm to construct 3D-Sudoku matrix for steganography is proposed in this paper, as well as an embedding and an extracting algorithm of steganography based on 3D-Sudoku matrix in low-bit rate speech codec. Theoretical analysis is provided to demonstrate that the concealment and the hidden capacity are greatly improved with the proposed scheme. Experimental results show the hidden capacity is raised to 200bps in ITU-T G.723.1 codec. Moreover, the quality of steganography speech in Perceptual Evaluation of Speech Quality (PESQ) reduces no more than 4%, indicating little impact on the quality of speech.

8 citations

Proceedings ArticleDOI
01 May 1977
TL;DR: Variable data rate LPC speech compression schemes are employed to transmit LPC parameters only when speech characteristics have changed sufficiently since the last transmission, yielding improved speech quality relative to fixed-rate schemes for a given average transmission rate.
Abstract: Variable data rate LPC speech compression schemes are employed to transmit LPC parameters only when speech characteristics have changed sufficiently since the last transmission, yielding improved speech quality relative to fixed-rate schemes for a given average transmission rate. Transmission of variable-rate LPC speech over fixed-rate channels is accomplished using transmit and receive buffers, with resulting transmission delays. Development of proper buffer control strategy is essential to minimize losses caused by exhausting either buffer, or by corrective actions, namely, forced or suppressed transmission. Certain aspects of such strategy and their impact on speech quality and data rate are discussed for a narrowband (2400 bps) speech transmission system.

8 citations

Proceedings ArticleDOI
09 Jul 2006
TL;DR: Initial results in determining the recognition accuracy that can be achieved with five widely used speech coding standards are presented and show that performance does not strictly depend on coding rate or codec speech quality.
Abstract: Compressed-domain automatic speaker recognition is based on the analysis of the compressed parameters of speech coders. The objective is to perform low-complexity on-line speaker recognition for VoIP in the compressed domain, without the need to decode or resynthesize the speech bitstream. In this paper, we present initial results in determining the recognition accuracy that can be achieved with five widely used speech coding standards. Experiments with a database of 14 speakers obtain a recognition ratio close to 100% after the analysis of 30 seconds of active speech for most of the considered speech coders and rates. In particular, the results show that performance does not strictly depend on coding rate or codec speech quality.

8 citations

Proceedings ArticleDOI
Wen Xu1
08 Nov 1998
TL;DR: The basic idea is to convert the residual redundancy of the source encoded parameters into the bit redundancy such that it can be more efficiently utilized in the channel decoding and the resulting parameters are less vulnerable to digital errors.
Abstract: The optimal binary mappings for converting the signal redundancy of the zero-th order (nonuniformity) and the first order (correlation) into individual bits are described. By employing a mapping matched to the residual redundancy inherent in the source-encoded parameters further gains can be obtained in the joint source-channel coding. The basic idea is to convert the residual redundancy of the source encoded parameters into the bit redundancy such that it can be more efficiently utilized in the channel decoding and the resulting parameters are less vulnerable to digital errors. The approach is successfully applied to the GSM full rate (FR) codec to achieve a more reliable transmission of speech signals.

8 citations

Proceedings ArticleDOI
01 Nov 2011
TL;DR: The simulation results show that when all the improvement schemes are combined, the performance is improved at all the bit rates compared to the previous results despite the fact that the Huffman table structure is significantly simplified.
Abstract: The internet Low Bit-rate Codec (iLBC) inherently possesses high robustness to packet loss which is one of the essential properties of Voice over Internet Protocol (IP) applications. Another important feature is the rate flexibility, which allows the speech codec to adapt its bit rate to constantly changing network condition. Previously, the multi-rate operation of the iLBC was enabled by utilizing the Discrete Cosine Transform (DCT) and entropy coding. In this paper, various approaches to improve performance are presented. The simulation results show that when all the improvement schemes are combined, the performance is improved at all the bit rates compared to the previous results despite the fact that the Huffman table structure is significantly simplified.

8 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