Voice-over Internet Protocol (VoIP) telephony which is the transmission of real-time voice over an
Internet Protocol (IP) data network, is increasingly becoming a variant telecommunication technology that
one day may surpass the old analog and digital telephone systems. The Quality-of-Service (QoS) factor is
an important parameter to be considered when measuring the performance of a VoIP system. Algorithmic
delay (latency) may influence the QoS of a VoIP system. This paper presents the hardware implementation
of the Linear Prediction (LP) Analysis and Quantization component of the Conjugate-Structure
Algebraic-Code-Excited Linear-Prediction (CS-ACELP) speech compression algorithms. The LP Analysis and
Quantization units were implemented in hardware using Xilinx 11.1 ISE, after which ITU-T test vectors
were used to determine whether they were equivalent implementations of the LP Analysis and Quantization.
The algorithmic delays of hardware-implementations were obtained via simulation using Modelsim XE 6.4b
while the algorithmic delays of the software-implemented speech compression algorithms were obtained by
software profiling using the GPROF profiler.
After comparison the algorithmic delays of the hardware-implementations of the LP Analysis and
Quantization was found to be shorter than the algorithmic delays of their software-implemented and
hardware-implemented counterparts.

Linear Prediction Analysis and Quantization for the Conjugate-Structure Algebraic-Code-Excited Linear-Prediction Speech Compression Algorithm

Data security is one of the major concerns for digital data as well as voice transmission through non-privatized networks. Hence, a significant amount of research work is needed to provide reliable and secured data/voice transmission. The CS-ACELP algorithm based G.729 codec is standardized as voice codec by ITU-T for multimedia and Voice over Internet Protocol (VoIP) applications. The encoding scheme described in this paper, for CS-ACELP speech codec provides on-fly (while processing the speech frame) security for the encoded data, which avoids the use of various complex encryption, and decryption algorithms in order to provide secured speech transmission. This method is based on the periodically changing the track sequence of fixed codebook index for every 100 ms speech samples sampled at a rate of 8000 samples per second and also, adapting a new technique to encode the fixed codebook index. This paper, presents an easy and efficient way of providing data security for G.729 codec. The experimental results show that similar speech quality can be obtained without compromising with additional cost and transmission bandwidth.

Analysis of the encoding scheme for CS-ACELP codec for secured VoIP communication

Conjugate-structure algebraic CELP (CS-ACELP) is a type of voice coder algorithm that compresses speech signal based on model parameters of human voice. Fully Software or Hardware implementation of CS-ACELP is not satisfactory in some applications such as multi-channel implementation. So in this paper a hardware-software Co-design implementation of the voice encoder is described. This design is based on idea of parallelism and fast access to data memory using hardware. The proposed system is implemented on Virtex-II ProTM FPGA using EDK8.1. The results of experiment show that co-design implantation can reduce the execution time of algorithm and make it possible to multi-channel implementation of it.

Hardware-software co-design of G729 voice encoder using Virtex-II Pro TM FPGA

In this work, ITU-T G.729 speech codec is precisely modeled using basic building blocks of SIMULINK such as simple adders and multipliers. Therefore, a golden model of the codec which best suits as a reference for its hardware implementation is developed. In a custom hardware design, the precision of the computations, i.e. the optimum number of the bits of the computational blocks should be calculated. The optimum number of the bits of the computational blocks is the minimum word-width that maintains the quality of the output with minimum chip area. Three blocks are selected to extract their bit-true models. The optimum bit-width of the coefficients and the internal computations are measured. These blocks are the pre-processing block, the open loop pitch estimator, and the fixed codebook search block. The former is important in terms of error propagation to the output and the two others are the blocks that demand the most computational power respectively. As such, these blocks are suitable choices to be hardware implemented in a hardware-software co-design project.

Modeling of ITU-T G.729 codec with bit-width optimization for intensive computation blocks

This paper proposes a method to extract speech features for content integrity verification of G 729 coded speech, which could minimize the computation cost and reduce the sizes of the features Experiments results show that the method is robust to common operations in speech transmission such as recompression, volume changing, re-sampling and transcoding, and also does well in tamper detection for its ability to distinguish content preserving operations from malicious modifications

Content Integrity Verification for G.729 Coded Speech

Modern mobile communications require optimum bandwidth utilization with minimum loss, delay and good quality of speech transmission This triggered the usage of low bit rate voice codecs, among which CELP codecs inherit the merits of both waveform and source codecs, like toll quality, low bit rate etc In this paper several hardware and software optimization techniques are presented for efficient implementation of ITU G729 standard (CS-ACELP, conjugate structure algebraic code excited linear prediction) of 8 Kbit/s bit rate on a real time digital signal processor (DSP), with the aim of overcoming the limitation of computational burden and also scaling this application for enhanced speed to process more channels These techniques are in general applicable to any speech codec and DSP processor platform

Optimizations of ITU G.729 speech codec

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Optimizations of ITU G.729 speech codec