Showing papers on "Enhanced Variable Rate Codec published in 2010"

The VISNET II DVC codec: Architecture, tools and performance

Abstract: This paper introduces the VISNET II DVC codec. This codec achieves very high RD performance thanks to the efficient combination of many state-of-the-art coding tools into a fully practical video codec. Experimental results show that the proposed DVC codec consistently outperforms H.264/AVC Intra. For sequences with coherent motion, it even surpasses H.264/AVC zero-motion. Finally, it is also always better than the DISCOVER DVC codec. Therefore, it is expected that the proposed high performing DVC codec will be used by other researchers in the field as a reference to benchmark their results.

Dual-mode switching used for unified speech and audio codec

Abstract: This paper presents a dual-mode switching method between time-domain codec and transform-domain codec of audio coding. It is a key technique of unified speech and audio (music) coding, since the replaying audio quality corresponds to the suitable codec selection and smooth switching between them. The proposed method consists of two steps, codec mode selection and switching. The binary decision trees (BDTs) algorithm is used to take a decision of mode selection, because of its advantages of high accuracy, low delay and low complexity. For smoothing transition between two codec, a pre-coding strategy is suggested in this paper. The classical speech codec, Algebraic Code Excited Linear Prediction (ACELP) and the Advanced Audio Coding (AAC) of MPEG are used for validating the proposed method. The scores of PESQ with 11 testing sequences show that the proposed switching method will not bring additional noise and can get higher objective evaluation of audio quality than single codec.

Complexity Scalability Design in the Internet Low Bit Rate Codec (iLBC) for Speech Coding

Abstract: Differing from the long-term prediction used in the modern speech codec, the standard of the internet low bit rate codec (iLBC) independently encodes the residual of the linear predictive coding (LPC) frame by frame. In this paper, a complexity scalability design is proposed for the coding of the dynamic codebook search in the iLBC speech codec. In addition, a trade-off between the computational complexity and the speech quality can be achieved by dynamically setting the parameter of the proposed approach. Simulation results show that the computational complexity can be effectively reduced with imperceptible degradation of the speech quality.

Third generation mobile network mobile phone sound end-to-end encrypting device and method

Abstract: The invention discloses a third generation mobile network mobile phone sound end-to-end encrypting device which is a sound end-to-end encrypting device designed aiming to a CDMA2000 (Code Division Multiple Access2000) mobile phone. The device provides a selectable independent sound encryption hardware module and a corresponding sound input/output device for the mobile phone, can provide end-to-end encryption/decryption capability for the CDMA2000 mobile phone, and realizes the function of encrypting/decrypting sound signals compressed by a 8K velocity ratio EVRC (Enhanced Variable Rate Codec) code and a 13K velocity ratio QCELP (Qualcomm Code Excited Linear Predictive Coding) code, wherein an FPGA (Field Programmable Gate Array) module (1) of an encrypting device is a hardware platform for carrying out an encryption/decryption process and is the core of the whole system; an encryption/decryption algorithm module (2) is a related algorithm program for realizing the encryption/decryption process; a signal preprocessing module (3) connects the FPGA module with the CDMA2000 mobile phone, preprocesses input/output signals of a chip, and also provides clock signals for the chip; and a power source management module (4) is connected with the FPGA module and the signal preprocessing module and is used for providing a power source with required specific voltage for the whole system.

The study of video acquisition and compression codec system based on DaVinci

Abstract: This paper achieves the design of a video acquisition and compression codec system, which takes dual-core chips TMS320DM6446 as its core and Linux as its operating system. This is because Linux can be reduced and transplanted. The video capture device driver V4L2 and Codec Engine are introduced in detail, and through H.264 algorithm functions of video compression codec are successfully realized. Relevant experiments show that the codec algorithm has strong performance of anti-errors and the videos are clear and reliable after compression codec. Moreover the mount of the video data is largely reduced.

Hybrid test compression architecture using multiple codecs for low pin count and high compression devices

Abstract: This invention uses multiple codecs to efficiently achieve the right balance between compression and coverage for a given design. This application illustrates a simple example using two codecs including a high compression codec and a low compression codec. The test engineer generates a first set of test patterns using the high compression codec. If this high compression results in unacceptable fault coverage loss, the top-up patterns for additional coverage are generated using the low compression codec. The invention may use multiple codecs serially one after the other. The codecs can be of different types or parameters (such as compression ratio, debug tolerance and combinational codec versus sequential codec).

Design for New Kind of Low Rate Speech Codec in VoIP

Abstract: An efficient low rate Speech Codec can get well utilized to solve the problem of network congestion under the circumstance of current network. To the speech characteristic analysed, the voice activity detection (VAD)algorithm that is much fewer operational quantity and very effective is designed based on conjugate structure algebraic code excited linear prediction (CS-ACELP),the Speech Codec in this paper is implemented on the Digital Signal Processor(TMS320VC5410)with low bit rate and good timbre. That will lower down the average bit rate to about 4kb/s. As a result, the demand of real-time full-duplex communication in VoIP is fulfilled.

Robust Detection of Underwater Transient Signals Using EVRC Noise Suppression Module

Abstract: Detection of transient signals is generally done by examining power and spectral variation of the received signal, but it becomes a difficult task when the background noise gets large. In this paper, we propose a robust transient detection algorithm using the EVRC noise suppression module. We define new parameters from the outputs of the EVRC noise suppression module for transient detection. Experimental results with various types of underwater transients have shown that the proposed method outperforms the conventional energy-based method and achieved performance improvement of detection rate by 7% to 15% for various types of background noise.

Using Noise Substitution for Backwards-Compatible Audio Codec Improvement

Abstract: A method for representing error in perceptual audio coding as filtered noise is presented. Various techniques are compared for analyzing and re-synthesizing the noise representation. A focus is placed on improving the perceived audio quality with minimal data overhead. In particular, it is demonstrated that per-critical-band energy levels are sufficient to provide an increase in quality. Methods for including the coded error data in an audio file in a backwards-compatible manner are also discussed. The MP3 codec is treated as a case study, and an implementation of this method is presented.

Perbandingan Kualitas Layanan Wireless VOIP pada Codec G.711, G.723 dan G.729

Abstract: Voice over Internet Protocol (VoIP) is a technology that enable voice message transmission over data network (internet protocol). Codec is algorithm or special computer program to reduce number of bytes. The usage of appropriate codec at implementation VoIP is one thing determining in attainment of quality VoIP communications. This research implement and analys the usage of codec G711, G723 and G729 at protocol H323 for VoIP service. Examination by through simulation using ns- 2. Codec G723 has smaller delay value and jitter value than codec G711 and G729. The result from simulation is codec G723 has smaller bit rate value, so more efficient for implementation at network that having not big bandwidth or network capacity. In communications process usage of

An Efficient Transcoding Scheme in Parameters for ITU-T G.729 and G.723.1 Codec

Abstract: We proposed an efficient method of converting speech codec formats between the G.729 and G.723.1 speech codec. The G.729 three frames is converted to one frame of the G.723.1, transcoding is completed through four processes： LSP and pitch conversions are used to linear interpolation processing, respectively, fast adaptive-codebook search processing adopt to predict the range of adaptive-codebook gain in the G.723.1, and a fast stochastic excitation pulses estimates method. Simulation results show that the proposed method is able to reduce the computation complexity about 84.87% with shorter delay than the tandem. And the perceptual speech quality (PESQ) evaluated better than the latter.

Low-complexity JPEG-based progressive video codec for wireless video transmission

Abstract: This paper discusses the question of video codec enhancement for wireless video transmission of high definition video data taking into account constraints on memory and complexity. Starting from parameter adjustment for JPEG2000 compression algorithm used for wireless transmission and achieving the best possible results by tuning settings, this work proceeds to develop a low-complexity progressive codec based on JPEG, which is compared to the tuned JPEG2000. Comparison to H.264/SVC for this codec is also given. As the results show, our simple solution on low rates can compete with JPEG2000 and H.264/SVC for specific video content.

Delay-Reduced MDCT for Scalable Speech Codec with Cascaded Transforms

Abstract: A scalable speech codec consisting of a harmonic codec as the core layer and MDCT-based transform codec as the enhancement layer is often required to provide both very low-rate core communication and fine granular scalability. This structure, however, has a serious drawback for practical use because a time delay caused by transform in each layer is accumulated, resulting in a long overall codec delay. In this letter, a new MDCT structure is proposed to reduce the overall codec delay by eliminating the accumulation of time delay by each transform. In the proposed structure, the time delay is first reduced by forcing two transforms to share a common look-ahead. The error components of MDCT caused by the look-ahead sharing are then analyzed and compensated in the decoder, resulting in perfect reconstruction. The proposed structure reduces the codec delay by the frame size, with an equivalent coding efficiency.