Variable Rate Characteristic Waveform Interpolation Speech Coder Based on Phonetic Classification

A variable-bit-rate characteristic waveform interpolation （VBR-CWI） speech codec with about 1.8 kbit/s average bit rate which integrates phonetic classification into characteristic waveform （CW） decomposition is proposed. Each input frame is classified into one of 4 phonetic classes. Non-speech frames are represented with Bark-band noise model. The extracted CWs become rapidly evolving waveforms （REWs） or slowly evolving waveforms （SEWs） in the cases of unvoiced or stationary voiced frames respectively, while mixed voiced frames use the same CW decomposition as that in the conventional CWI. Experimental results show that the proposed codec can eliminate most buzzy and noisy artifacts existing in the fixed-bit-rate characteristic waveform interpolation （FBR-CWI） speech codec, the average bit rate can be much lower, and its reconstructed speech quality is much better than FS 1 016 CELP at 4.8 kbit/s and similar to G. 723.1 ACELP at 5.3 kbit/s.

Very Low Bit Rate Video Coding Using Motion Compensated 3 D Wavelet Transform

A new motion compensated 3 D wavelet transform (MC 3DWT) video coding scheme is presented in this paper. The new coding scheme has a good performance in average PSNR, compression ratio and visual quality of reconstructions compared with the existing 3 D wavelet transform (3DWT) coding methods and motion compensated 2 D wavelet transform (MC WT) coding method. The new MC 3DWT coding scheme is suitable for very low bit rate video coding.

Variable Bit Rate Fuzzy Control for Low Delay Video Coding

Rate control plays a critical role in achieving perceivable video quality under a variable bit rate,limited buffer sizes and low delay applications.Since a rate control system exhibits non-linear and unpredictable characteristics,it is difficult to establish a very accurate rate-distortion(R-D)model and acquire effective rate control performance.Considering the excellent control ability and low computing complexity of the fuzzy logic in non-linear systems,this paper proposes a bitrate control algorithm based on a fuzzy controller,named the Fuzzy Rate Control Algorithm(FRCA),for All-Intra(AI)and low-delay(LD)video source coding.Contributions of the proposed FRCA mainly consist of four aspects.First,fuzzy logic is adopted to minimize the deviation between the actual and the target buffer size in the hypothetical reference decoder(HRD).Second,a fast lookup table is employed in fuzzy rate control,which reduces computing cost of the control process.Third,an input domain determination scheme is proposed to improve the precision of the fuzzy controller.Fourth,a novel scene change detection is introduced and integrated in the FRCA to adaptively adjust the Group-of-Pictures(GOP)length when the source content fluctuates.The FRCA can be transplanted and implemented in various industry coders.Extensive experiments show that the FRCA has accurate variable bit-rate control ability and maintains a steady buffer size during the encoding processes.Compared with the default configuration encoding under AI and LD,the proposed FRCA can achieve the target bit rates more accurately in various classical encoders.

AVariablebitrateVideoCodingSchemeBasedonMorphologicalOperationsforATMNetworks

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Very Low Bit-Rate Video Coding Based on Wavelet Transform

Two video coding schemes based on wavelet transform achieving very low bit rate are presented in this paper. The first is a hybrid motion compensated wavelet transform(MC WT)system which behaves better at very low bit rates than the block DCT residual coder. The second is a new efficient coding system based on a simple frame differencing wavelet transform(FD WT)which performs well in both PSNR and visual quality with substantially reduced complexity.

3-D Wavelet Transform for Very Low Bit-Rate Video Coding

A new improved Goh's 3 D wavelet transform(WT) coding scheme is presented in this paper. The new scheme has great advantages including a simple code structure, low computation cost and good performance in PSNR, compression ratios and visual quality of reconstructions, when compared to the other existing 3 D WT coding methods and the 2 D WT based coding methods. The new 3 D WT coding scheme is suitable for very low bit rate video coding.

基于可逆元胞自动机加密的扩展码索引调制方案

针对直接序列扩频系统伪随机(Pseudo Noise,PN)码资源有限且码索引调制系统误码率性能下降的问题,提出了一种基于可逆的初等元胞自动机加密的扩展码索引调制方案。首先,针对PN码资源有限的问题,提出使用混沌规则的初等元胞自动机对PN码进行迭代的方案,从而实现扩展PN码的目的;其次,针对码索引调制误码率性能下降的问题,提出了一种可逆元胞自动机加密的码索引调制方案,在发射端,把信息比特切割成调制比特和映射比特,并分别映射为调制符号和扩频码索引,使用对应索引的扩频码对同相分量进行扩频,同时使用可逆元胞自动机对映射比特进行加密,加密后的映射比特为调制符号的正交分量选取对应索引的扩频码进行扩频。仿真与分析结果表明,在相同频谱效率条件下,在加性高斯白噪声信道中,当误码率为10-5时,扩展码索引调制方案的误码率性能优于码索引调制和广义码索引调制方案约2~4 dB,优于非正交码移键控码索引调制方案约0.5 dB。

Neural-Polar码:一种基于深度学习的新型信道编码方案

为应对新型移动通信系统智能性的需求以及在难以进行人工建模的复杂信道环境下进行可靠通信的问题,基于Polar码的编译码递归结构提出一种新型神经网络信道编码方案,即Neural-Polar码。该方案利用神经网络将Polar码编译码递归结构中父、子节点间的线性映射变成非线性映射,引入快速连续抵消(successive cancellation, SC)译码的思想,解决在完全二叉树上构建Neural-Polar码造成网络结构过大的问题。仿真实验表明,Neural-Polar码可以获得优于经典SC译码算法的误码率(bit error rate, BER)和误块率(block error rate, BLER)性能,对网络的联合训练使得Neural-Polar码能够自动学习信道特性,具有更好的信道适应性和鲁棒性。Neural-Polar码将传统的对复杂信道进行人工建模分析的难题交给机器,充分体现出其编译码的智能性。

一种基于视觉显著性的码率控制算法

针对当前高效视频编码的码率控制算法未能充分确保视频显著区域的图像质量问题,提出了一种基于视觉显著性的码率控制算法。首先,采用基于图的流形排序显著检测算法获取视频的显著区域;其次,在图像编码过程中,为每个编码单元(Coding Unit,CU)建立基于视觉显著性的加权失真公式,将显著性值作为权重;最后,通过求解得到拉格朗日乘子λ,并根据原始码率控制算法中的λ与量化参数(Quantization Parameter,QP)的关系为每个CU确定QP。试验结果表明,该算法有效地根据视频内容的显著性权重实现了码率的合理分配,并在保持码率控制准确性的同时,提升了视频显著区域的图像质量。

一种基于混合MELP/CELP的4 kbit/s声码器

利用混合激励线性预测(mixed excitation linear prediction,MELP)算法和码激励线性预测(code excitation linear prediction,CELP)算法的优点,提出了一种混合MELP/CELP语音编码模型。编码端对强浊音帧采用MELP编码,对弱浊音帧和清音帧进行CELP编码。MELP编码器采用相位对齐技术提取强浊音帧的相位参数,解决了合成语音与原始语音在时间上不同步的问题。对实现的4 kbit/s混合MELP/CELP声码器进行客观MOS(mean opinion score)值和主观DRT(diagnostic rhythm test)清晰度测试,结果表明,该声码器的合成语音具有较高的可懂度和清晰度。

基于Gamma-Gamma湍流信道下的卷积码性能研究

为了研究不同码型的卷积码在水下湍流信道中的误码率(BER)性能,采用接受-拒绝采样模拟湍流信道乘性干扰,并选择二进制相移键控(BPSK)调制方式,建立Gamma-Gamma湍流信道通信系统仿真模型。仿真结果表明:在不同强度的湍流信道中,采用卷积码编码均能提升系统的BER性能;卷积码的码率越小,系统BER性能提升越显著;随着信噪比(SNR)增大,记忆深度越长,系统BER下降速度越快;采用软译码比采用硬译码时增益至少提升2.82 d B;卷积码的解码不仅受当前信息的影响,还与之前的码元信息有关。

无人机航拍视频码率控制算法优化

高效率视频编码(HEVC)在实际应用中很难满足无人机航拍领域的需求。针对此问题,提出一种基于缓冲区状态及视频显著性的码率控制算法。对预处理后的帧进行颜色、纹理特征提取,并采用log-Gabor滤波器对其进行特征滤波,得到当前帧的显著图;计算每帧和每个最小单元(CTU)的显著性权重;以帧的显著性权重和缓冲区状态指导帧层比特分配,以CTU的显著性权重指导CTU层比特分配。实验结果表明,提出的算法可以在缓冲区维持在一个平稳状态的前提下,提高0.61 dB视频显著区域的峰值信噪比,更适用于无人机航拍的场景。

A rate control scheme for H.264 video under low bandwidth channel

The dilemma of the quantization parameter (QP) being involved in both rate control and rate-distortion optimization (RDO) prevents using the traditional rate control scheme. Although some rate control schemes are proposed to circumvent the dilemma, the inaccurate prediction model and improper bit allocation deter H.264 application on low bandwidth channel. To resolve this issue, this paper proposes a novel rate control scheme by considering the macroblock (MB) encoding complexity variation and buffer variation and by exploiting the spatio-temporal correlation sufficiently well. Simulations showed that this scheme improves the perceptual quality of the pictures with similar or smaller PSNR deviations when compared to that of rate control in JVT-O016.

Improving the BER performance of turbo codes with short frame size based on union bound

In order to improve the bit error rate(BER) performance of turbo codes with short frame size at a wide range of signal to noise ratio(SNR),a new method by optimizing the bit energy is proposed.At first,a formula derived from the Union Bound is introduced.It describes the relations between the bit error rate distribution and the minimum weight distribution.And then,by mathematically optimizing the formula,the energy for every bit of the codeword is optimized to achieve the minimum BER at high SNR region.At last,an adjustable parameter is employed to compensate the degradations of BER at low and moderate SNR regions.Case studies indicate that the improvements of BER for turbo codes with short frame size are significant at a wide range of SNR.

Joint Source-Channel Bit Allocation Based on Expected End-to-End Distortion Model

An adaptive joint source channel bit allocation method for video communications over error-prone channel is proposed.To protect the bit-streams from the channel bit errors,the rate compatible punctured convolution(RCPC)code is used to produce coding rates varying from 4/5 to 1/2 using the same encoder and the Viterbi decoder.An expected end-to-end distortion model was presented to estimate the distortion introduced in compressed source coding due to quantization and channel bit errors jointly.Based on the proposed end-to-end distortion model,an adaptive joint source-channel bit allocation method was proposed under time-varying error-prone channel conditions.Simulated results show that the proposed methods could utilize the available channel capacity more efficiently and achieve better video quality than the other fixed coding-based bit allocation methods when transmitting over error-prone channels.

Frame-layer bit allocation for multi-view video coding based on frame complexity estimation

Current multi-view video coding （MVC） reference model in joint video team （JVT） does not provide efficient rate control schemes. This paper presents a rate control algorithm for MVC by improving the quadratic rate-distortion （R-D） model. We reasonably allocate bit-rate among views based on the correlation analysisl The proposed algorithm consists of three levels to control the rate bits more accurately, of which the frame layer allocates bits according to the frame complexity and the temporal activity. Extensive experiments show that the proposed algorithm can control the bit rate efficiently.

OPTIMAL PUNCTURING OF TURBO CODES BASED ON ADS CRITERION

Puncturing is the predominant strategy to construct high code rate turbo codes. Puncturing tables are crucial to the performance of punctured turbo codes(PTC). This paper developed a new searching algorithm of optimal puncturing tables based on average distance spectrum(ADS) criterion. Consequently, some optimal puncturing tables were presented as the searching results. Finally, it presented the performance comparison among some optimal and bad puncturing tables by simulation.

A Self-Terminated Interleaver Design for Turbo Codes

Turbo codes can achieve excellent performance at low signal-to-noise ratio (SNR), but the performance can be severely degraded if no trellis termination is employed. This paper proved that if trellis termination bits were appended to RSC1, trellis of RSC2 could be terminated by designing the interleaver properly, consequently, derived the designing condition of such self-terminated interleaver (STI). Then we presented an algorithm of implementing a kind of STI, which could terminate RSC2 as well on condition that the RSC1 was terminated. We verified the performance of STI for turbo codes by simulation, and the simulation results showed that turbo codes with STI outperformed interleavers that could not terminate RSC2 as well.

Performance analysis and design of MIMO-OFDM system using concatenated forward error correction codes

This work investigates the performance of various forward error correction codes, by which the MIMO-OFDM system is deployed. To ensure fair investigation, the performance of four modulations, namely, binary phase shift keying(BPSK), quadrature phase shift keying(QPSK), quadrature amplitude modulation(QAM)-16 and QAM-64 with four error correction codes(convolutional code(CC), Reed-Solomon code(RSC)+CC, low density parity check(LDPC)+CC, Turbo+CC) is studied under three channel models(additive white Guassian noise(AWGN), Rayleigh, Rician) and three different antenna configurations(2×2, 2×4, 4×4). The bit error rate(BER) and the peak signal to noise ratio(PSNR) are taken as the measures of performance. The binary data and the color image data are transmitted and the graphs are plotted for various modulations with different channels and error correction codes. Analysis on the performance measures confirm that the Turbo + CC code in 4×4 configurations exhibits better performance.