This paper proposes an unequal error protection（UEP）coding method to improve the transmission performance of three-dimensional（3D）audio based on expanding window fountain（EWF）.Different from other transmissions ...This paper proposes an unequal error protection（UEP）coding method to improve the transmission performance of three-dimensional（3D）audio based on expanding window fountain（EWF）.Different from other transmissions with equal error protection（EEP）when transmitting the 3D audio objects.An approach of extracting the important audio object is presented,and more protection is given to more important audio object and comparatively less protection is given to the normal audio objects.Objective and subjective experiments have shown that the proposed UEP method achieves better performance than equal error protection method,while the bits error rates（BER）of the important audio object can decrease from 10^（–3） to 10^（–4）,and the subjective quality of UEP is better than that of EEP by 14%.展开更多
Object-based audio coding is the main technique of audio scene coding. It can effectively reconstruct each object trajectory, besides provide sufficient flexibility for personalized audio scene reconstruction. So more...Object-based audio coding is the main technique of audio scene coding. It can effectively reconstruct each object trajectory, besides provide sufficient flexibility for personalized audio scene reconstruction. So more and more attentions have been paid to the object-based audio coding. However, existing object-based techniques have poor sound quality because of low parameter frequency domain resolution. In order to achieve high quality audio object coding, we propose a new coding framework with introducing the non-negative matrix factorization(NMF) method. We extract object parameters with high resolution to improve sound quality, and apply NMF method to parameter coding to reduce the high bitrate caused by high resolution. And the experimental results have shown that the proposed framework can improve the coding quality by 25%, so it can provide a better solution to encode audio scene in a more flexible and higher quality way.展开更多
Lattice vector quantization （LVQ） has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages： It has a simple and fast encoding process,...Lattice vector quantization （LVQ） has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages： It has a simple and fast encoding process, and it significantly reduces the amount of memory required. Therefore, LVQ is suitable for use in low-complexity speech and audio coding. In this paper, we describe the basic concepts of LVQ and its advantages over conventional vector quantization. We also describe some LVQ techniques that have been used in speech and audio coding standards of international standards developing organizations （SDOs）.展开更多
A Hi Fi audio coding technology for ISDN and Internet is introduced. It is the ISO/MPEG Audio Layer III digital audio compression scheme coding at 64 kbit/s. First, the paper implements C language simulation accordin...A Hi Fi audio coding technology for ISDN and Internet is introduced. It is the ISO/MPEG Audio Layer III digital audio compression scheme coding at 64 kbit/s. First, the paper implements C language simulation according to the algorithm and gets satisfactory quality of the reconstructed music signal. The estimation of operation steps and simulation of decoder finished by a TMS 320C548 simulator are presented. The result is the same as that of the C language simulation.展开更多
Audio Video Coding Standard （AVS） is a second-generation source coding standard and the first standard for audio and video coding in China with independent intellectual property rights. Its performance has reached t...Audio Video Coding Standard （AVS） is a second-generation source coding standard and the first standard for audio and video coding in China with independent intellectual property rights. Its performance has reached the international standard. Its coding efficiency is 2 to 3 times greater than that of MPEG -2. This technical solution is more simple, and it can greatly save channel resource. After more than ten years＇ development, AVS has achieved great success. The latest version of the AVS audio coding standard is ongoing and mainly aims at the increasing demand for low bitrate and high quality audio services. The paper reviews the history and recent development of AVS audio coding standard in terms of basic features, key techniques and performance. Finally, the future development of AVS audio coding standard is discussed.展开更多
A novel frame error concealment scheme is proposed to improve the decoded audio quality of the receiver for transform coded excitation（TCX）audio codec.This scheme,which is a gain control approach based on the stabil...A novel frame error concealment scheme is proposed to improve the decoded audio quality of the receiver for transform coded excitation（TCX）audio codec.This scheme,which is a gain control approach based on the stability of linear predictive coding（LPC）filter,predicts the lost frames by utilizing the linear spectrum frequency and different continuous attenuation factor of different kinds of lost frames.Signal noise ratio（SNR）test and multiple stimuli with hidden reference and anchor（MUSHRA）test are conducted to evaluate the performance of this approach in adaptive multi-rate wideband plus（AMR-WB＋）audio codec.Compared with the original frame error concealment scheme,our scheme achieves better audio recovery quality in AMR-WB＋audio codec.展开更多
A new three-dimensional(3D) audio coding approach is presented to improve the spatial perceptual quality of 3D audio. Different from other audio coding approaches, the distance side information is also quantified, and...A new three-dimensional(3D) audio coding approach is presented to improve the spatial perceptual quality of 3D audio. Different from other audio coding approaches, the distance side information is also quantified, and the non-uniform perceptual quantization is proposed based on the spatial perception features of the human auditory system, which is named as concentric spheres spatial quantization(CSSQ) method. Comparison results were presented, which showed that a better distance perceptual quality of 3D audio can be enhanced by 5.7%~8.8% through extracting and coding the distance side information comparing with the directional audio coding, and the bit rate of our coding method is decreased of 8.07% comparing with the spatial squeeze surround audio coding.展开更多
A Bark-band residual noise model integrated with the human hearing mechanism is proposed to efficiently complement sinusoidal model in parametric audio coding. The time-varying spectrum of the residual noise is retrie...A Bark-band residual noise model integrated with the human hearing mechanism is proposed to efficiently complement sinusoidal model in parametric audio coding. The time-varying spectrum of the residual noise is retrieved by Bark-scale piecewise constant magnitude estimates along with random phases. In the proposed noise model, Bark bands information is obtained by short-time FFT method and window overlap-add technique is exploited to remove boundary discontinuities. SVQ is also incorporated into parameter quantization process for the low bit-rate coding demand. Simulation results and informal listening tests show that when the sinusoidal model is combined with the Bark-band noise model, better synthesis audio quality can be achieved compared with the original sinusoidal modeling audio codec.展开更多
This paper proposed improvements to the low bit rate parametric audio coder with sinusoid model as its kernel. Firstly, we propose a new method to effectively order and select the perceptually most important sinusoids...This paper proposed improvements to the low bit rate parametric audio coder with sinusoid model as its kernel. Firstly, we propose a new method to effectively order and select the perceptually most important sinusoids. The sinusoid which contributes most to the reduction of overall NMR is chosen. Combined with our improved parametric psychoacoustic model and advanced peak riddling techniques, the number of sinusoids required can be greatly reduced and the coding efficiency can be greatly enhanced. A lightweight version is also given to reduce the amount of computation with only little sacrifice of performance. Secondly, we propose two enhancement techniques for sinusoid synthesis: bandwidth enhancement and line enhancement. With little overhead, the effective bandwidth can be extended one more octave; the timbre tends to sound much brighter, thicker and more beautiful.展开更多
Abstract The method of quantization noise control of audio coding in the wavelet domain is proposed. Using the inverse Discrete Fourier Transform （DFT）, it converts the masking threshold coming from MPEG psycho-acou...Abstract The method of quantization noise control of audio coding in the wavelet domain is proposed. Using the inverse Discrete Fourier Transform （DFT）, it converts the masking threshold coming from MPEG psycho-acoustic model in the frequency domain to the signal in the time domain; the Discrete Wavelet Packet Transform （DWPF） is performed; the energy in each subband is regarded as the maximum allowed quantization noise energy. The experimental result shows that the proposed method can attain the nearly transparent audio quality below 64kbps for the most testing audio signals.展开更多
The performance of the speaker recognition system declines when training and testing audio codecs are mismatched. In this paper, based on analyzing the effect of mismatched audio codecs in the linear prediction cepstr...The performance of the speaker recognition system declines when training and testing audio codecs are mismatched. In this paper, based on analyzing the effect of mismatched audio codecs in the linear prediction cepstrum coefficients, a method of MAP-based audio coding compensation for speaker recognition is proposed. The proposed method firstly sets a standard codec as a reference and trains the speaker models in this codec format, then learns the deviation distributions between the standard codec format and the other ones, next gets the current bias via using a small number adaptive data and the MAP-based adaptive technique, and then adjusts the model parameters by the type of coming audio codec format and its related bias. During the test, the features of the coming speaker are used to match with the adjusted model. The experimental result shows that the accuracy reached 82.4% with just one second adaptive data, which is higher 5.5% than that in the baseline system.展开更多
With the rapid expansion of multimedia data,protecting digital information has become increasingly critical.Reversible data hiding offers an effective solution by allowing sensitive information to be embedded in multi...With the rapid expansion of multimedia data,protecting digital information has become increasingly critical.Reversible data hiding offers an effective solution by allowing sensitive information to be embedded in multimedia files while enabling full recovery of the original data after extraction.Audio,as a vital medium in communication,entertainment,and information sharing,demands the same level of security as images.However,embedding data in encrypted audio poses unique challenges due to the trade-offs between security,data integrity,and embedding capacity.This paper presents a novel interpolation-based reversible data hiding algorithm for encrypted audio that achieves scalable embedding capacity.By increasing sample density through interpolation,embedding opportunities are significantly enhanced while maintaining encryption throughout the process.The method further integrates multiple most significant bit(multi-MSB)prediction and Huffman coding to optimize compression and embedding efficiency.Experimental results on standard audio datasets demonstrate the proposed algorithm’s ability to embed up to 12.47 bits per sample with over 9.26 bits per sample available for pure embedding capacity,while preserving full reversibility.These results confirm the method’s suitability for secure applications that demand high embedding capacity and perfect reconstruction of original audio.This work advances reversible data hiding in encrypted audio by offering a secure,efficient,and fully reversible data hiding framework.展开更多
Non-blind audio bandwidth extension is a standard technique within contemporary audio codecs to efficiently code audio signals at low bitrates. In existing methods, in most cases high frequencies signal is usually gen...Non-blind audio bandwidth extension is a standard technique within contemporary audio codecs to efficiently code audio signals at low bitrates. In existing methods, in most cases high frequencies signal is usually generated by a duplication of the corresponding low frequencies and some parameters of high frequencies. However, the perception quality of coding will significantly degrade if the correlation between high frequencies and low frequencies becomes weak. In this paper, we quantitatively analyse the correlation via computing mutual information value. The analysis results show the correlation also exists in low frequency signal of the context dependent frames besides the current frame. In order to improve the perception quality of coding, we propose a novel method of high frequency coarse spectrum generation to improve the conventional replication method. In the proposed method, the coarse high frequency spectrums are generated by a nonlinear mapping model using deep recurrent neural network. The experiments confirm that the proposed method shows better performance than the reference methods.展开更多
In this paper we present a motion compensation (MC) design for the newest Audio Video coding Standard (AVS) of China. Because of compression-efficient techniques of variable block size (VBS) and sub-pixel interpolatio...In this paper we present a motion compensation (MC) design for the newest Audio Video coding Standard (AVS) of China. Because of compression-efficient techniques of variable block size (VBS) and sub-pixel interpolation, intensive pixel calculation and huge memory access are required. We propose a parallel serial filtering mixed luma interpolation data flow and a three-stage multiplication free chroma interpolation scheme. Compared to the conventional designs, the integrated architecture supports about 2.7 times filtering throughput. The proposed MC design utilizes Vertical Z processing order for reference data re-use and saves up to 30% memory bandwidth. The whole design requires 44.3k gates when synthesized at 108 MHz clock frequency using 0.18-μm CMOS technology and can support up to 1920×1088@30 fps AVS HDTV video decoding.展开更多
Audio Video coding Standard (AVS) is the latest audio and video coding standard of China. AVS Part 7 (also known as AVS-M) targets mobility applications where error concealment is of great importance. This paper first...Audio Video coding Standard (AVS) is the latest audio and video coding standard of China. AVS Part 7 (also known as AVS-M) targets mobility applications where error concealment is of great importance. This paper first briefly introduces the general concept of error concealment. Then two error concealment schemes are proposed and implemented on AVS-M decoder under different test conditions. Simulation results of the schemes and suggestions on how to use these tools are also provided.展开更多
基金Supported by the National High Technology Research and Development Program of China(863 Program,2015AA016306)the National Natural Science Foundation of China(61662010,61231015,61471271)+1 种基金Science and Technology Plan Projects of Shenzhen(ZDSYS2014050916575763)Science and Technology Foundation of Guizhou Province(LKS[2011]1)
文摘This paper proposes an unequal error protection（UEP）coding method to improve the transmission performance of three-dimensional（3D）audio based on expanding window fountain（EWF）.Different from other transmissions with equal error protection（EEP）when transmitting the 3D audio objects.An approach of extracting the important audio object is presented,and more protection is given to more important audio object and comparatively less protection is given to the normal audio objects.Objective and subjective experiments have shown that the proposed UEP method achieves better performance than equal error protection method,while the bits error rates（BER）of the important audio object can decrease from 10^（–3） to 10^（–4）,and the subjective quality of UEP is better than that of EEP by 14%.
基金supported by National High Technology Research and Development Program of China (863 Program) (No.2015AA016306)National Nature Science Foundation of China (No.61231015)National Nature Science Foundation of China (No.61671335)
文摘Object-based audio coding is the main technique of audio scene coding. It can effectively reconstruct each object trajectory, besides provide sufficient flexibility for personalized audio scene reconstruction. So more and more attentions have been paid to the object-based audio coding. However, existing object-based techniques have poor sound quality because of low parameter frequency domain resolution. In order to achieve high quality audio object coding, we propose a new coding framework with introducing the non-negative matrix factorization(NMF) method. We extract object parameters with high resolution to improve sound quality, and apply NMF method to parameter coding to reduce the high bitrate caused by high resolution. And the experimental results have shown that the proposed framework can improve the coding quality by 25%, so it can provide a better solution to encode audio scene in a more flexible and higher quality way.
文摘Lattice vector quantization （LVQ） has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages： It has a simple and fast encoding process, and it significantly reduces the amount of memory required. Therefore, LVQ is suitable for use in low-complexity speech and audio coding. In this paper, we describe the basic concepts of LVQ and its advantages over conventional vector quantization. We also describe some LVQ techniques that have been used in speech and audio coding standards of international standards developing organizations （SDOs）.
文摘A Hi Fi audio coding technology for ISDN and Internet is introduced. It is the ISO/MPEG Audio Layer III digital audio compression scheme coding at 64 kbit/s. First, the paper implements C language simulation according to the algorithm and gets satisfactory quality of the reconstructed music signal. The estimation of operation steps and simulation of decoder finished by a TMS 320C548 simulator are presented. The result is the same as that of the C language simulation.
文摘Audio Video Coding Standard （AVS） is a second-generation source coding standard and the first standard for audio and video coding in China with independent intellectual property rights. Its performance has reached the international standard. Its coding efficiency is 2 to 3 times greater than that of MPEG -2. This technical solution is more simple, and it can greatly save channel resource. After more than ten years＇ development, AVS has achieved great success. The latest version of the AVS audio coding standard is ongoing and mainly aims at the increasing demand for low bitrate and high quality audio services. The paper reviews the history and recent development of AVS audio coding standard in terms of basic features, key techniques and performance. Finally, the future development of AVS audio coding standard is discussed.
基金Supported by the National High Technology Research and Development Program of China(863 Program)(2015AA016306)the Foundation of Outstanding Middle-aged and Young Scientific and Technological Innovation Team Program of Department of Education of Hubei Province(T201516)the Foundation of Department of Education of Hubei Province(Q20132207)
文摘A novel frame error concealment scheme is proposed to improve the decoded audio quality of the receiver for transform coded excitation（TCX）audio codec.This scheme,which is a gain control approach based on the stability of linear predictive coding（LPC）filter,predicts the lost frames by utilizing the linear spectrum frequency and different continuous attenuation factor of different kinds of lost frames.Signal noise ratio（SNR）test and multiple stimuli with hidden reference and anchor（MUSHRA）test are conducted to evaluate the performance of this approach in adaptive multi-rate wideband plus（AMR-WB＋）audio codec.Compared with the original frame error concealment scheme,our scheme achieves better audio recovery quality in AMR-WB＋audio codec.
基金supported by National High Technology Research and Development Program of China (863 Program, No. 2015AA016306)National Nature Science Foundation of China (No. 61662010, 61231015, 61471271, 61761044, 61762005)
文摘A new three-dimensional(3D) audio coding approach is presented to improve the spatial perceptual quality of 3D audio. Different from other audio coding approaches, the distance side information is also quantified, and the non-uniform perceptual quantization is proposed based on the spatial perception features of the human auditory system, which is named as concentric spheres spatial quantization(CSSQ) method. Comparison results were presented, which showed that a better distance perceptual quality of 3D audio can be enhanced by 5.7%~8.8% through extracting and coding the distance side information comparing with the directional audio coding, and the bit rate of our coding method is decreased of 8.07% comparing with the spatial squeeze surround audio coding.
文摘A Bark-band residual noise model integrated with the human hearing mechanism is proposed to efficiently complement sinusoidal model in parametric audio coding. The time-varying spectrum of the residual noise is retrieved by Bark-scale piecewise constant magnitude estimates along with random phases. In the proposed noise model, Bark bands information is obtained by short-time FFT method and window overlap-add technique is exploited to remove boundary discontinuities. SVQ is also incorporated into parameter quantization process for the low bit-rate coding demand. Simulation results and informal listening tests show that when the sinusoidal model is combined with the Bark-band noise model, better synthesis audio quality can be achieved compared with the original sinusoidal modeling audio codec.
文摘This paper proposed improvements to the low bit rate parametric audio coder with sinusoid model as its kernel. Firstly, we propose a new method to effectively order and select the perceptually most important sinusoids. The sinusoid which contributes most to the reduction of overall NMR is chosen. Combined with our improved parametric psychoacoustic model and advanced peak riddling techniques, the number of sinusoids required can be greatly reduced and the coding efficiency can be greatly enhanced. A lightweight version is also given to reduce the amount of computation with only little sacrifice of performance. Secondly, we propose two enhancement techniques for sinusoid synthesis: bandwidth enhancement and line enhancement. With little overhead, the effective bandwidth can be extended one more octave; the timbre tends to sound much brighter, thicker and more beautiful.
文摘Abstract The method of quantization noise control of audio coding in the wavelet domain is proposed. Using the inverse Discrete Fourier Transform （DFT）, it converts the masking threshold coming from MPEG psycho-acoustic model in the frequency domain to the signal in the time domain; the Discrete Wavelet Packet Transform （DWPF） is performed; the energy in each subband is regarded as the maximum allowed quantization noise energy. The experimental result shows that the proposed method can attain the nearly transparent audio quality below 64kbps for the most testing audio signals.
文摘The performance of the speaker recognition system declines when training and testing audio codecs are mismatched. In this paper, based on analyzing the effect of mismatched audio codecs in the linear prediction cepstrum coefficients, a method of MAP-based audio coding compensation for speaker recognition is proposed. The proposed method firstly sets a standard codec as a reference and trains the speaker models in this codec format, then learns the deviation distributions between the standard codec format and the other ones, next gets the current bias via using a small number adaptive data and the MAP-based adaptive technique, and then adjusts the model parameters by the type of coming audio codec format and its related bias. During the test, the features of the coming speaker are used to match with the adjusted model. The experimental result shows that the accuracy reached 82.4% with just one second adaptive data, which is higher 5.5% than that in the baseline system.
基金funded by theNational Science and Technology Council of Taiwan under the grant number NSTC 113-2221-E-035-058.
文摘With the rapid expansion of multimedia data,protecting digital information has become increasingly critical.Reversible data hiding offers an effective solution by allowing sensitive information to be embedded in multimedia files while enabling full recovery of the original data after extraction.Audio,as a vital medium in communication,entertainment,and information sharing,demands the same level of security as images.However,embedding data in encrypted audio poses unique challenges due to the trade-offs between security,data integrity,and embedding capacity.This paper presents a novel interpolation-based reversible data hiding algorithm for encrypted audio that achieves scalable embedding capacity.By increasing sample density through interpolation,embedding opportunities are significantly enhanced while maintaining encryption throughout the process.The method further integrates multiple most significant bit(multi-MSB)prediction and Huffman coding to optimize compression and embedding efficiency.Experimental results on standard audio datasets demonstrate the proposed algorithm’s ability to embed up to 12.47 bits per sample with over 9.26 bits per sample available for pure embedding capacity,while preserving full reversibility.These results confirm the method’s suitability for secure applications that demand high embedding capacity and perfect reconstruction of original audio.This work advances reversible data hiding in encrypted audio by offering a secure,efficient,and fully reversible data hiding framework.
基金supported by the National Natural Science Foundation of China under Grant No. 61762005, 61231015, 61671335, 61702472, 61701194, 61761044, 61471271National High Technology Research and Development Program of China (863 Program) under Grant No. 2015AA016306+2 种基金 Hubei Province Technological Innovation Major Project under Grant No. 2016AAA015the Science Project of Education Department of Jiangxi Province under No. GJJ150585The Opening Project of Collaborative Innovation Center for Economics Crime Investigation and Prevention Technology, Jiangxi Province, under Grant No. JXJZXTCX-025
文摘Non-blind audio bandwidth extension is a standard technique within contemporary audio codecs to efficiently code audio signals at low bitrates. In existing methods, in most cases high frequencies signal is usually generated by a duplication of the corresponding low frequencies and some parameters of high frequencies. However, the perception quality of coding will significantly degrade if the correlation between high frequencies and low frequencies becomes weak. In this paper, we quantitatively analyse the correlation via computing mutual information value. The analysis results show the correlation also exists in low frequency signal of the context dependent frames besides the current frame. In order to improve the perception quality of coding, we propose a novel method of high frequency coarse spectrum generation to improve the conventional replication method. In the proposed method, the coarse high frequency spectrums are generated by a nonlinear mapping model using deep recurrent neural network. The experiments confirm that the proposed method shows better performance than the reference methods.
基金(No. Y106574) supported by the Natural Science Foundationof Zhejiang Province, China
文摘In this paper we present a motion compensation (MC) design for the newest Audio Video coding Standard (AVS) of China. Because of compression-efficient techniques of variable block size (VBS) and sub-pixel interpolation, intensive pixel calculation and huge memory access are required. We propose a parallel serial filtering mixed luma interpolation data flow and a three-stage multiplication free chroma interpolation scheme. Compared to the conventional designs, the integrated architecture supports about 2.7 times filtering throughput. The proposed MC design utilizes Vertical Z processing order for reference data re-use and saves up to 30% memory bandwidth. The whole design requires 44.3k gates when synthesized at 108 MHz clock frequency using 0.18-μm CMOS technology and can support up to 1920×1088@30 fps AVS HDTV video decoding.
基金Project (No. 60333020) supported by the National Natural Science Foundation of China
文摘Audio Video coding Standard (AVS) is the latest audio and video coding standard of China. AVS Part 7 (also known as AVS-M) targets mobility applications where error concealment is of great importance. This paper first briefly introduces the general concept of error concealment. Then two error concealment schemes are proposed and implemented on AVS-M decoder under different test conditions. Simulation results of the schemes and suggestions on how to use these tools are also provided.
