The Experts below are selected from a list of 58956 Experts worldwide ranked by ideXlab platform
Ryota Miyauchi - One of the best experts on this subject based on the ideXlab platform.
-
method of Digital Audio watermarking based on cochlear delay characteristics
2013Co-Authors: Masashi Unoki, Ryota MiyauchiAbstract:This paper proposes a state-of-the-art method of Digital-Audio watermarking, based on the properties of the human cochlear. It is based on the concept of embedding inaudible watermarks (e.g., copyright data) into an original sound by controlling the phase characteristics of the sound in relation to the characteristics of cochlear delay. The method involves two processes, i.e., embedding and detecting data. The data-embedding process was carried out by applying two types of IIR all-pass filters with cochlear delays, and then selecting the filtered signal from these according to the watermarks. The data-detection process involved estimating the group delay of the filter from the phase difference between the original and watermarked sounds to detect the embedded data. We designed optimal group delays for the filters. We experimentally evaluated the proposed method by carrying out subjective detection tests, two objective detection tests (PEAQ and LSD), bit-detection tests, and three objective tests for robustness (against down sampling, amplitude compression, and mp3-compression). We also comparatively evaluated the proposed method with four other methods (LSB, DSS, ECHO, and PPM). The results revealed that subjects could not detect the embedded data in any of the watermarked sounds we used, and that the proposed approach could precisely and robustly detect the embedded data from the watermarked sounds.
-
reversible watermarking for Digital Audio based on cochlear delay characteristics
2011Co-Authors: Masashi Unoki, Ryota MiyauchiAbstract:There have recently been serious social issues involved in multimedia signal processing such as Digital rights management, secure authentication, malicious attacks, and tampering with Digital Audio/speech signals. Reversible watermarking is a technique that enables these signals to be authenticated and then restored to their original signals by removing watermarks from them. We previously proposed an inaudible Digital-Audio watermarking approach based on cochlear delay (CD). We investigated how the proposed approach could be developed as reversible watermarking by considering blind detection for inaudible watermarks and the reversibility of Audio watermarking. We evaluated inaudible and reversible watermarking with the proposed approach by carrying out three objective tests (PEAQ, LSD, and bit-detection or SNR). The results revealed that reversible watermarking based on CD could be accomplished.
B A Wooley - One of the best experts on this subject based on the ideXlab platform.
-
a 1 8 v Digital Audio sigma delta modulator in 0 8 spl mu m cmos
1997Co-Authors: S Rabii, B A WooleyAbstract:Oversampling techniques based on sigma-delta (/spl Sigma//spl Delta/) modulation offer numerous advantages for the realization of high-resolution analog-to-Digital (A/D) converters in low-voltage environment. This paper examines the design and implementation of a CMOS /spl Sigma//spl Delta/ modulator for Digital-Audio A/D conversion that operates from a single 1.8-V power supply. A cascaded modulator that maintains a large full-scale input range while avoiding signal clipping at internal nodes is introduced. The experimental modulator has been designed with fully differential switched-capacitor integrators employing different input and output common-mode levels and boosted clock drivers in order to facilitate low voltage operation. Precise control of common-mode levels, high power supply noise rejection, and low power dissipation are obtained through the use of two-stage, class A/AB operational amplifiers. At a sampling rate of 4 MHz and an oversampling ratio of 80, an implementation of the modulator in a 0.8-/spl mu/m CMOS technology with metal-to-polycide capacitors and NMOS and PMOS threshold voltages of +0.65 V and -0.75 V, respectively, achieves a dynamic range of 99 dB at a Nyquist conversion rate of 50 kHz. The modulator can operate from supply voltages ranging from 1.5-2.5 V, occupies an active area of 1.5 mm/sup 2/, and dissipates 2.5 mW from a 1.8-V supply.
-
second order sigma delta modulation for Digital Audio signal acquisition
1991Co-Authors: B P Brandt, D E Wingard, B A WooleyAbstract:The authors compare the second-order sigma-delta ( Sigma Delta ) modulator to several alternative modulator architectures in the context of Digital-Audio signal acquisition. Design details and experimental results are presented for a 1 mu m CMOS implementation that does not require error correction or component trimming to achieve virtually ideal 16 b performance at a conversion rate of 50 kHz. The experimental modulator is a fully differential circuit that operates from a single 5 V power supply and does not require the use of precision sample-and-hold circuitry. With an oversampling ratio of 256 and a clock rate of 12.8 MHz, the modulator achieves a 98 dB dynamic range and a peak signal-to-(noise+distortion) ratio (SNDR) of 94 dB. Measurements and simulations of discrete noise peaks in the output spectrum that result from limit-cycle oscillations are also presented and discussed. >
Hongying Yang - One of the best experts on this subject based on the ideXlab platform.
-
a robust Digital Audio watermarking using higher order statistics
2011Co-Authors: Hongying Yang, Xiangyang WangAbstract:Abstract Based on higher-order statistics and synchronization code, we propose a new Digital Audio watermarking algorithm with good auditory quality and reasonable resistance toward desynchronization attacks in this paper. Firstly, the wavelet de-noising is performed on the original host Audio, the de-noised Digital Audio is segmented, and then each segment is cut into two parts. Secondly, with the spatial watermarking technique, synchronization code is embedded into the statistics average value of Audio samples in the first part. And then, the higher-order statistics are obtained by using the Hausdorff distance. Finally, the Digital watermark is embedded into the original Audio signal in wavelet domain by using the higher-order statistics. Simulation results show that the proposed watermarking scheme is not only inaudible and robust against common signal processing such as MP3 compression, noise addition, resampling, re-quantization, etc., but also robust against the desynchronization attacks such as random cropping, amplitude variation, pitch shifting, jittering, etc.
-
a robust Digital Audio watermarking based on statistics characteristics
2009Co-Authors: Xiangyang Wang, Hongying YangAbstract:In Digital Audio watermarking, the watermark's vulnerability to desynchronization attacks has long been a difficult problem. According to the Audio statistics characteristics and synchronization code technique, a new robust Audio watermarking scheme against desynchronization attacks is proposed in this paper. Firstly, the original Digital Audio is segmented and then each Audio segment is cut into two parts. Secondly, with the spatial watermarking technique, the synchronization code is embedded into the statistics average value of Audio samples in the first part. Finally, the second part of Audio segment is cut into Audio sections, the DWT is performed on the Audio sections, and the watermark bit is embedded into the statistics average value of low frequency components. Experimental results show that the proposed scheme is inaudible and robust against common signals processing, including MP3 compression, low-pass filtering, noise addition, and equalization, etc. Moreover, it also survives several desynchronization attacks, such as random cropping, amplitude variation, pitch shifting, time-scale modification, and jittering, etc.
-
a robust Digital Audio watermarking method
2009Co-Authors: Xiangyang Wang, Hongying YangAbstract:Most of the previous Audio-watermarking schemes are robust to common signal-processing attacks, but show severe problems when faced with desynchronization attacks. To solve the problems associated with these approaches, we propose an Audio-watermarking scheme based on support-vector-machine (SVM) theory to protect against desynchronization attacks by using Audio statistics characteristics and a synchronization code technique. Experimental results with SVM show that our proposed scheme is inaudible, robust against common signal processing, and robust against desynchronization attacks.
Masashi Unoki - One of the best experts on this subject based on the ideXlab platform.
-
method of Digital Audio watermarking based on cochlear delay characteristics
2013Co-Authors: Masashi Unoki, Ryota MiyauchiAbstract:This paper proposes a state-of-the-art method of Digital-Audio watermarking, based on the properties of the human cochlear. It is based on the concept of embedding inaudible watermarks (e.g., copyright data) into an original sound by controlling the phase characteristics of the sound in relation to the characteristics of cochlear delay. The method involves two processes, i.e., embedding and detecting data. The data-embedding process was carried out by applying two types of IIR all-pass filters with cochlear delays, and then selecting the filtered signal from these according to the watermarks. The data-detection process involved estimating the group delay of the filter from the phase difference between the original and watermarked sounds to detect the embedded data. We designed optimal group delays for the filters. We experimentally evaluated the proposed method by carrying out subjective detection tests, two objective detection tests (PEAQ and LSD), bit-detection tests, and three objective tests for robustness (against down sampling, amplitude compression, and mp3-compression). We also comparatively evaluated the proposed method with four other methods (LSB, DSS, ECHO, and PPM). The results revealed that subjects could not detect the embedded data in any of the watermarked sounds we used, and that the proposed approach could precisely and robustly detect the embedded data from the watermarked sounds.
-
reversible watermarking for Digital Audio based on cochlear delay characteristics
2011Co-Authors: Masashi Unoki, Ryota MiyauchiAbstract:There have recently been serious social issues involved in multimedia signal processing such as Digital rights management, secure authentication, malicious attacks, and tampering with Digital Audio/speech signals. Reversible watermarking is a technique that enables these signals to be authenticated and then restored to their original signals by removing watermarks from them. We previously proposed an inaudible Digital-Audio watermarking approach based on cochlear delay (CD). We investigated how the proposed approach could be developed as reversible watermarking by considering blind detection for inaudible watermarks and the reversibility of Audio watermarking. We evaluated inaudible and reversible watermarking with the proposed approach by carrying out three objective tests (PEAQ, LSD, and bit-detection or SNR). The results revealed that reversible watermarking based on CD could be accomplished.
C E W Sundberg - One of the best experts on this subject based on the ideXlab platform.
-
soft selection combining for terrestrial Digital Audio broadcasting in the fm band
2001Co-Authors: J N Laneman, C E W SundbergAbstract:Methods of adaptive soft combining and channel decoding are developed to combat the effects of multipath fading and nonuniform interference channels, with particular application to Digital reception in hybrid in-band on-channel (HIBOC) Digital Audio broadcast (DAB) systems in the FM band. These systems transmit near CD quality Digital Audio and analog FM simultaneously within the same license band, requiring the Digital Audio to be protected with powerful channel codes and sophisticated decoding algorithms to provide broad coverage under a variety of fading and potentially severe interference conditions created by first adjacent FM stations. In an example HIBOC DAB system, Digital transmissions are DQPSK/OFDM modulated in two sidebands of the analog FM host signal, and a complementary punctured pair convolutional (CPPC) inner coding scheme allows for higher diversity benefit than code combining when both sidebands are interference-free as well as full recovery of the Audio information when one of the sidebands is severely corrupted by first adjacent interference. For the intermediate cases in which one of the sidebands is partially useful, we demonstrate via simulations that an unmodified receiver designed for a Gaussian channel and corresponding to equal-gain combining performs ineffectively for moderate to high interference levels. Motivated by a clear need for more sophisticated receivers, we examine soft combiners derived from the maximum-likelihood principle and provide simulated performance bounds for the case in which perfect channel parameter estimates are available. We then discuss more practical methods for performing adaptive soft combining and channel decoding, focusing in particular on an appealing soft selection combining technique, based on successive erasures and Viterbi decoding, that requires only coarse estimates of the channel parameters. An outer code used for error concealment may be further utilized to perform the selection function. The performance of this soft selection combining scheme under a variety of interference scenarios is also evaluated via simulation. Further improvements may be obtained with a list Viterbi decoder.
-
reed solomon decoding algorithms for Digital Audio broadcasting in the am band
2001Co-Authors: J N Laneman, C E W SundbergAbstract:Digital Audio broadcasting (DAB) systems for the AM band are being developed to provide higher-quality radio broadcasts, broader coverage, and data services. Transmission is typically done by means of multistreaming with QAM/OFDM in the high bits/sec/Hz regime in which Reed-Solomon codes, either alone or in concatenation with inner trellis coded modulation (TCM), are natural choices for error control. For the AM DAB application, Reed-Solomon codes with suboptimal decoders of the bounded-distance type offer error correction important for bringing down the error probability, but also offers error detection important for generating block detected-error flags for use in the error concealment, or error mitigation, algorithm in the Audio decoder. For multistream systems, the block detected-error flags can also be used to select the component streams retained by the Audio decoder. In this paper, we evaluate the performance of Reed-Solomon codes in terms of both detected (flag) and undetected error rates for DAB applications. We provide simulation results for a variety of decoding algorithms, including hard-decision decoding (HDD), successive-erasure decoding (SED), and fixed-erasure decoding (FED), and discuss the channel environments in which each of these algorithms may be appropriate. Among other results, for example, we address the issue of matching the blocklength of the Reed-Solomon code to the Audio decoder with its variable frame structure. From our simulation results for HDD, we conclude that the increased error correction capability of longer Reed-Solomon codes more than compensates for the mismatch in terms of error mitigation in the Audio decoder. We also observe that SED provides only small improvements for uniform interference channels, while FED offers considerable improvements for some partial-band interference channels. We describe an appealing two-mode adaptive configuration using HDD and FED for mitigating partial-band interference caused by some second-adjacent AM broadcasts.
-
complementary punctured pair convolutional codes for Digital Audio broadcasting
2000Co-Authors: Brian Chen, C E W SundbergAbstract:Hybrid in-band on-channel (IBOC) broadcasting systems for Digital Audio radio have the capability of simultaneously transmitting analog FM and Digital Audio of CD-like quality. Due to fading and interference in the already-crowded FM band, the signal design for the hybrid IBOC system is very challenging. It has been proposed to use a method of double sideband transmission where the Digital information is transmitted by means of orthogonal frequency-division multiplexing (OFDM) on both sides of the analog host FM and where the Digital information can be recovered when one sideband is partially or even totally lost. This leads to an interesting channel coding problem, where we search for optimal pairs of high-rate codes that form good combined low-rate codes, which are better than classic code combining techniques. Furthermore, we also search for rate-compatible punctured convolutional codes which can be used for two-level unequal error protection (UEP) of Digital Audio. Since some of the tones in the multitone modem (OFDM) are more exposed to interference than others, optimal assignments of convolutional code bits to tones depending on their spectral position are also found. A large number of new codes with memory 6 and 8 are presented both for equal error protection and UEP.
-
Digital Audio broadcasting in the fm band
1997Co-Authors: C E W SundbergAbstract:It is envisioned that all broadcasting will be Digital in the future. There are standards work and system development going on in various parts of the world to make this vision reality. In Europe, the emphasis has been on broadband systems (EUREKA) with multiple program channels in a new frequency band. In the United States, the broadcasting industry is favoring a solution where broadcasting is compatible with the existing analog FM and AM systems. A number of new Digital Audio radio systems have been proposed and evaluated. In this paper we present a conceptual overview of some of the properties of these systems. We also present the transmission channel properties of the FM systems. The most challenging part of the transmission is broadcasting to vehicles. We discuss details of key subsystems such as Audio coding, channel coding, and Digital modulation in the proposed in-band on channel (IBOC) and in-band adjacent channel (IBAC) as well as in-band reserved channel (IBRC) systems for the FM band.