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Showing papers on "Enhanced Variable Rate Codec published in 2011"


Patent
24 Mar 2011
TL;DR: In this article, an integrated echo canceller and speech codec for voice-over-internet (VoI) protocol is presented. But the speech codec is not integrated with the echo canceler.
Abstract: A method includes operating an integrated echo canceller and speech codec for voice-over internet protocol. An apparatus includes an echo canceller and a speech codec, wherein the speech codec includes a decoder and an encoder, and wherein the echo canceller and the speech codec are integrated for voice-over-internet protocol.

8 citations


Patent
21 Dec 2011
TL;DR: In this article, a method for measuring voice quality in a wireless communication network includes measuring an MOS of a signal using a narrowband voice codec and a MOS using a wideband voice decoder in a cable loopback environment.
Abstract: A method for measuring voice quality in a wireless communication network includes measuring an MOS of a signal using a narrowband voice codec and an MOS of a signal using a wideband voice codec in a cable loopback environment, calculating a wideband voice codec correction coefficient using the measured MOS, measuring an MOS of a signal using the narrowband voice codec and an MOS of a signal using the wideband voice codec in a terminal connection environment; and outputting a value obtained by adding the wideband voice codec correction coefficient to the measured MOS in the terminal connection environment.

7 citations


Proceedings ArticleDOI
26 Jul 2011
TL;DR: Two FEC schemes are proposed which take the advantage of the codec's structure characteristics and do not introduce extra delay to enhance the robustness of packet loss recovery for AVS Mobile speech and audio (AVS-M) codec.
Abstract: In this paper, we utilize sender-based Forward Error Correction (FEC) techniques to enhance the robustness of packet loss recovery for AVS Mobile speech and audio (AVS-M) codec. Two FEC schemes are proposed which take the advantage of the codec's structure characteristics and do not introduce extra delay. The objective and subjective listening tests results show that the two methods achieve higher reconstructed quality than the codec's original frame erasure scheme in the case of packet loss.

4 citations


Journal ArticleDOI
TL;DR: In the experimental results with SNR, ASDM, transmission bit rate measurement, it is demonstrated that voice quality was improved by using the proposed algorithm.
Abstract: To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC`s basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.

1 citations


Proceedings ArticleDOI
01 Dec 2011
TL;DR: A modification to the EVRC algorithm is proposed, which further improves the performance of it tremendously and increase in performance of the proposed modified algorithm is ascertained by implementing the same on the DSK and obtaining relevant results.
Abstract: This paper intends to analyze the performance of the widely used Enhanced Variable Rate Codec (EVRC) algorithm for suppressing background noise in speech Signals and propose a method for improving the current noise suppression capability of the algorithm. Firstly, the current EVRC algorithm is implemented on the TMS320C6713 based DSK in order to verify the algorithm and also to validate the real time performance of it. Then, based on the results and further analysis of the algorithm, a modification to the EVRC algorithm is proposed, which further improves the performance of it tremendously. This increase in performance of the proposed modified algorithm is ascertained by implementing the same on the DSK and obtaining relevant results.

1 citations


Proceedings ArticleDOI
12 May 2011
TL;DR: This work focuses on optimizing C code by assembly in Basic Operators (BASOP) that shows the significant improvement of complexity and execution time of G.711.1 codec on ARM processor.
Abstract: ITU-T wideband speech codec G.711.1[1] extends and interoperates with widely-used narrowband G.711 codec. This results in an efficient deployment of G.711.1 codec over existing G.711-based VoIP to provide higher voice quality. However, processing capacity of commercial processors is very limited to implement wideband codec; therefore the codec needs to be optimized for real-time processing. This work focuses on optimizing C code by assembly in Basic Operators (BASOP) [2], that shows the significant improvement of complexity and execution time of G.711.1 codec on ARM processor.