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Inversion of short-time fourier transform magnitude spectrograms with adaptive window lengths

Inversion of short-time fourier transform magnitude spectrog...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Volker Gnann Martin Spiertz Institut für Nachrichtentechnik RWTH Aachen University Aachen Germany

ISBN: (纸本)9781424423538

In this paper, we extend the real-time iterative spectrogram inversion method (RTISI) for generating a time-domain audio signal from a magnitude spectrogram such that it can handle changing spectrogram window lengths. For each desired window length, we use a separate buffer structure and synchronize the buffers each time the window length changes. This way, the proposed method helps to improve the time/frequency-resolution trade-off for algorithms that operate on magnitude-only spectra.

关键词： Fourier transforms Spectrogram Phase estimation Signal resolution Bridges Frequency audio coding Iterative methods Signal generators Time domain analysis

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Watermarking of AAC Bit-streams for Forensic Tracking

Watermarking of AAC Bit-streams for Forensic Tracking

引用

IEEE Signal Processing and Communications Applications (SIU)

作者： Serap Kirbiz Mehmet Celik Aweke Lemma Stefan Katzenbeisser ITÜ Elektrik Elektronik Fakültesi Telekomünikasyon Mühendisliği Bölümü Maslak Istanbul Turkey Philips Research Europe Eindhoven Netherlands

We propose a novel method for embedding robust forensic tracking watermarks to complement encryption in electronic music distribution applications. The watermark is embedded at the player during the AAC decoding process through modification of scale-factors in pre-defined frequency bands. It thereby modulates the short-time envelope of the decoded audio in the corresponding band. The resulting watermark is robust to various attacks such as re-compression and acoustic transmission. The method, due to its negligible computational overhead, can be used in resource constrained devices such as portable players.

关键词： Watermarking Forensics Internet audio coding Robustness Decoding Remuneration Information security Europe Cryptography

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Quantifying perceptual distortion in scalably compressed MPEG audio

Quantifying perceptual distortion in scalably compressed MPE...

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Asilomar Conference on Signals, Systems & Computers

作者： C.D. Creusere Klipsch School of Electrical & Computer Engineering New Mexico State University USA

A scalably compressed bitstream is one which can be streamed and decoded at a wide variety of bitrates, and it is therefore compatible with communications channels of varying capacity. The audio coding portions of the MPEG 2 and 4 standards support fine-grained scalability through the use of bit slice arithmetic coding (BSAC). Human subjective analysis of BSAC, however, has shown that it performs poorly at low bitrates; seemingly random tonal patterns are superimposed on the actual audio. Here, we develop a new approach for objectively characterizing such distortion and validate it with human subjective trials. Unlike most other objective performance metrics, the proposed approach does not require sample-accurate sequence synchronization. As a comparison, we also apply the ITU-R BS.1387-1 objective testing recommendation to the same audio sequences and quantify how well it predicts the observed subjective quality.

关键词： Transform coding Bit rate Humans Streaming media Decoding Communication channels Channel capacity audio coding Scalability Arithmetic

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Mobile audio visual terminal (MAVT)

Mobile audio visual terminal (MAVT)

引用

International Conference on Universal Personal Communications

作者： D. Lappe Robert Bosch GmbH Hildesheim Germany

The project MAVT within the European Research program on advanced communication (RACE) has the objective to develop a demonstrator for a Mobile audio Visual Terminal. Further researches are conducted for the development of a terminal simulation in the Universal Mobile Telecommunication System (UMTS). The main objectives of the project are: To develop powerful video and audio coding algorithms for transmission of moving video, still video and speech in UMTS. To implement these algorithms on a demonstrator called MAVT in a DECT environment and in an UMTS simulation. The project duration was from February 1992 to December 1994. The MAVT main achievements of the first year have been: Clear definition of services and user requirements. Description of transition between different networks. Introduction of the mobile test sequences MAVT carphone and MAVT foreman. First reference model for px8kbit/s video codec. Due to the noisy channels and high error rates in a mobile environment the source data stream is protected by RCPC Codes (Rate Compatible Punctured Convolutional Codes).

关键词： 3G mobile communication Mobile communication audio coding Speech coding Testing Video codecs Working environment noise Error analysis Streaming media Protection

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Perfect reconstruction time-varying IIR filter banks

Perfect reconstruction time-varying IIR filter banks

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Asilomar Conference on Signals, Systems & Computers

作者： C.D. Creusere Naval Air Warfare Center China Lake CA USA

We consider here the general problem of implementing perfect reconstruction IIR filter banks, allowing the use of both causal and noncausal filtering stages. Past work including that of Smith & Chung (1995) and Chen & Vaidyanathan (1993) has focused exclusively on filter banks having only noncausal filtering stages, probably because the computationally efficient modulated polyphase allpass filter bank falls into this category. We use a canonical direct form II framework here which is closely related to that of Chen and Vaidyanathan. In addition to considering extensions of this framework to include causal filtering stages, we point out some constraints analogous to those found by Smith and Chung for their direct form I implementation. Finally, we consider in detail the problem of selecting initial filter states to guarantee perfect cancellation of startup transients for finite length sequences in the time-varying case. We derive here a result for circular convolution with time-varying IIR filters analogous to that found by Smith & Eddins (1990) for time-invariant filters.

关键词： IIR filters Filter bank Finite impulse response filter Image reconstruction Filtering Convolution Time varying systems Delay Lakes audio coding

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R-TTT Module with Modified Residual Signal for Improving Multichannel audio Signal Accuracy

R-TTT Module with Modified Residual Signal for Improving Mul...

引用

International Conference on Automation, Cognitive Science, Optics, Micro Electro-Mechanical System, and Information Technology

作者： Ikhwana Elfitri Amirul Luthfi Fitrilina Department of Electrical Engineering Faculty of Engineering Andalas University

ISBN: (纸本)9781467374095

Spatial audio coding is a technique that capable of representing multichannel audio signals as a lower number of audio channels accompanied by spatial parameters and residual signal which will be useful for recreating the original multi-channel audio signals. Moving Picture Expert Group (MPEG) Surround, an international standard developed based on spatial audio coding, specifies Reverse Two-To-Three (R-TTT) module to extend stereo audio, consisted of left and right channels, into three audio channels: left, centre, and right channels based on Channel Prediction Coefficient (CPC) as spatial parameter and residual signal. In this paper, a modified residual signal is proposed to provide a better audio waveform reconstruction in the decoder side by minimising distortion caused by quantisation of CPC. Our experiments show that the waveform accuracy in terms of Signal-to-Noise Ratio (SNR) gets improved as high as 11 dB while the subjective test shows that the proposed method does not reduce perceptual quality, in terms of Subjective Difference Grade (SDG) score, of the reconstructed audio signals.

关键词： MPEG Surround Spatial audio coding Multichannel audio Signals residual signal audio coding audio signals audio channels Moving Pictures Experts Group Subjective testing Quantization

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Frequency-domain Comfort Noise Generation for Discontinuous Transmission in EVS

Frequency-domain Comfort Noise Generation for Discontinuous ...

引用

IEEE International Conference on Acoustics, Speech and Signal Processing

作者： A. Lombard S. Wilde E. Ravelli S. Dohla G. Fuchs M. Dietz Fraunhofer IIS Erlangen Germany

ISBN: (纸本)9781467369985

Discontinuous Transmission (DTX) is an efficient way to drastically reduce the transmission rate of a communication codec in the absence of voice input. In this mode, most frames that are determined to consist of background noise only are dropped from transmission and replaced by some Comfort Noise Generation (CNG) in the decoder. In this paper, we propose a novel CNG approach combining information gained about the actual background noise at both encoder and decoder side. It is able to better reproduce background noise types showing a pronounced spectral tilt, which is difficult for traditional schemes based on a linear prediction model. The proposed technique operates in the frequency domain. It is part of the Enhanced Voice Services (EVS) codec, where it is known as FD-CNG. Listening tests show the superior quality of FD-CNG over existing approaches for certain background noise such as car noise.

关键词： CNG DTX EVS audio coding speech coding

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A Novel audio Codec for Mobile Multimedia Applications

A Novel Audio Codec for Mobile Multimedia Applications

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第三届IEEE无线通讯、网络技术暨移动计算国际会议

作者： Cong Zhang RuiMin Hu School of Computer Wuhan University Wuhan China National Engineering Research Center for Multimedia Software Wuhan University Wuhan China

During the last decade, new mobile multimedia applications have emerged for mobile and network multimedia, wireless multimedia communication, audio/video teleconferencing, remote assistance, digital storage systems, secure audio transmission and so on. In order to meet these requirements, tremendous research efforts have been put in the development of efficient digital audio coding technologies. In China, AVS-M audio standard is such an audio technology targeting for mobile multimedia applications which is developed and owned by China audio and Video coding Standard Workgroup. In this paper, AVS-M audio standard is discussed by revealing the technical principles of the en- and decoding, the standardization situation and the suitability of the codec in relation to technology available, economical feasibility and the market needs. Finally it concludes with a brief discussion of future research directions.

关键词： AVS-M audio coding 3GPP

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Perceptual phase redundancy in speech

Perceptual phase redundancy in speech

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Doh-Suk Kim Human & Computer Interaction Laboratory Samsung Advanced Institute of Technology Yongin si Kyunggi South Korea

For efficient quantization of speech representation, it is common to incorporate perceptual characteristics of human hearing. However, the focus has been confined only to the magnitude information of speech, and little attention has been paid to phase information. This paper presents a novel approach, termed perceptually irrelevant phase elimination (PIPE), to find out irrelevant phase information of acoustic signals in terms of perception. The proposed method, which is based on the observation that the relative phase relationship within a critical band is perceptually important, is derived not only for stationary Fourier signal but also for harmonic signal. The proposed method is then incorporated into the analysis-by-synthesis system based on harmonic representation of speech, and subjective test results demonstrate its effectiveness.

关键词： Humans Auditory system Speech coding Quantization Speech analysis System testing Timbre Laboratories Harmonic analysis audio coding

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Scalable and progressive audio codec

Scalable and progressive audio codec

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： M.S. Vinton L.E. Atlas Department of Electrical Engineering University of Washington Seattle WA USA

A source coding technique for variable, bandwidth-constrained channels such as the Internet must do two things: offer high quality at low data rates, and adapt gracefully to changes in available bandwidth. Here we propose an audio coding algorithm that is superior on both counts. It is inherently scalable, meaning that channel conditions can be matched without the need for additional computation. Moreover, it is compact: in subjective tests our algorithm, coded at 32 kb/s/channel, outperformed MPEG-1 Layer 3 (MP3) coded at 56 kb/s/channel (both at 44.1 kHz). We achieve this simultaneous increase in compression and scalability through use of a two-dimensional transform that concentrates relevant information into a small number of coefficients.

关键词： Codecs audio coding Scalability Frequency modulation Bandwidth Bit rate Source coding Testing Transform coding Digital audio players

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