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检索条件"主题词=Audio coding"

共 1197 条记录，以下是121-130 订阅

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Jointly optimal quantization of parameters in sinusoidal audio coding

Jointly optimal quantization of parameters in sinusoidal aud...

引用

Workshop on Applications of Signal Processing to audio and Acoustics

作者： Vafin, R Kleijn, WB Royal Inst Technol Dept Signals Sensors & Syst Stockholm Sweden

ISBN: (纸本)0780391543

[it this paper, we present new methods for jointly optimal scalar quantization of parameters in sinusoidal audio coding. The sinusoidal amplitudes, frequencies, and phases used to represent (the tonal component of) an audio signal are quantized Such as to minimize a single-letter perceptual distortion measure under a given bit-rate constraint. The asymptotically optimal scalar quantizers are derived analytically using high-rate theory. We present different practical methods, where quantization accuracy of sonic parameters is made dependent on quantized values of other parameters'. The choice of method depends on the availability of the frequency-masking threshold, defining the perceptual importance of the individual sinusoids, at the decoder. The high performance of our quantizers is demonstrated by means of a listening test.

关键词： audio coding quantisation (signal) bit-rate constraint frequency-masking threshold jointly optimal scalar quantization optimal scalar quantizers single-letter perceptual distortion sinusoidal audio coding audio coding quantizer Sinusoidal Quantization audio signals

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A fully scalable audio coding structure with embedded psychoacoustic model

A fully scalable audio coding structure with embedded psycho...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Li, Te Rahardja, Susanto Koh, Soo Ngee Inst Infocomm Res 21 Heng Mui Keng Terrace Singapore 119613 Singapore Nanyang Technol Univ Singapore 639798 Singapore

ISBN: (纸本)9781424414833

A fully scalable audio coding structure based on a novel combination of the non-core MPEG-4 scalable lossless audio coding (SLS), the state-of-the-art psychoacoustic model, joint stereo coding and the perceptually prioritized bit-plane coding is presented in this paper. The psychoacoustic information is implicitly embedded in the scalable bitstream with negligible amount of side information and trivial modification to the standardized SLS decoder. Results of extensive evaluation show that the subjective quality of scalable audio is improved significantly.

关键词： audio coding joint stereo coding bitplane coding

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Model-based deadzone optimization for stack-run audio coding with uniform scalar quantization

Model-based deadzone optimization for stack-run audio coding...

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33rd IEEE International Conference on Acoustics, Speech and Signal Processing

作者： Oger, Marie Ragot, Stephane Antonini, Marc France Telecom R&D TECH SSTP Av Pierre Marzin F-22307 Lannion France CNRS Lab I3S UMR 6070 F-06903 Sophia Antipolis France Univ Nice Sophia Antipolis F-06903 Sophia Antipolis France

ISBN: (纸本)9781424414833

In this paper we present a model-based method to optimize the dead-zone of uniform scalar quantization in transform audio coding. The input signal is coded in modified discrete cosine transform (MDCT) domain by uniform scalar quantization followed by context-based arithmetic coding. The optimal deadzone is derived using a non-asymptotic method, assuming that the distribution of MDCT coefficients is approximated by a generalized Gaussian model. We show that deadzone optimization improves slightly quality, especially at low bitrates.

关键词： transform coding audio coding

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AVS2 SPEECH AND audio coding SCHEME FOR HIGH QUALITY AT LOW BITRATES

AVS2 SPEECH AND AUDIO CODING SCHEME FOR HIGH QUALITY AT LOW ...

引用

IEEE International Conference on Multimedia and Expo Workshops (ICMEW)

作者： Jiang, Lin Hu, Ruimin Wang, Xiaochen Zhang, Maoshen Wang, Zhongyuan Wuhan Univ Coll Comp Natl Engn Res Ctr Multimedia Software Wuhan Hubei Peoples R China E China Inst Technol Software Coll Taiyuan Peoples R China

ISBN: (纸本)9781479947171

With the increasing number of portable and wireless devices and the expanding of wireless network bandwidth, there is a growing demand for high quality and low bitrate speech and audio codec. This paper presents a speech and audio codec, which is based upon AVS P10. In our presented codec, ACELP and TCX technology is adopted, and also presents a hybrid bandwidth extension to coding high band signal. The paper gives an overview of the codec architecture and presents results of formal listening tests, objective test, and complicity of algorithm comparing this new codec with ITU-T G.718 and AVS P10. This result shows a comparable performance and forms the basis of the reference model in the ongoing AVS standardization activity for AVS2 speech and audio coding.

关键词： speech coding audio coding AVS

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Recent Advances in Broadcast audio coding

Recent Advances in Broadcast Audio Coding

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8th IEEE International Symposium on Broadband Multimedia Systems and Broadcasting (BMSB)

作者： Popp, Julian Neuendorf, Max Fuchs, Harald Forster, Christian Heuberger, Albert Univ Erlangen Nurnberg Informat Technol Commun Elect Wolfsmantel 33 D-91058 Erlangen Germany Fraunhofer Inst Integrated Circuits IIS D-91058 Erlangen Germany

ISBN: (纸本)9781467360470

In the last decades there have been major advances in broadcasting technologies as well as a change in consumer behaviour and demands. The on-demand concepts and streaming technology, together with widely available fixed and mobile broadband Internet connections, had an eminent impact on the broadcasters to fit the consumer's needs with respect to content variety, quality and availability. Due to the limited capacity available for broadcasting content, especially over the air, more and more efficient codecs were necessary. This paper will give a short overview of the history of broadcast audio coding and point out which codecs are currently used in the major broadcasting systems, i.e. DVB, DAB and DRM. Recent developments in broadcast audio coding will be presented, focusing on spatial audio object coding and unified speech and audio coding technologies.

关键词： Digital audio broadcasting audio compression audio coding

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Novel Ideas for Lossless audio coding

Novel Ideas for Lossless Audio Coding

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2012 International Conference on Signals and Electronic Systems (ICSES)

作者： Ulacha, Grzegorz Stasinski, Ryszard West Pomeranian Univ Technol Dept Comp Sci PL-71210 Szczecin Poland

ISBN: (纸本)9781467317092

The novel ideas for lossless audio coding analyzed in the paper are linked with forward predictor adaptation, and concern optimization of predictors on the basis of zero-order entropy and MMAE criterions, and context sound coding. Direct use of the former criterion is linked with exponential growth of optimization procedure, hence, a suboptimal algorithm having polynomial complexity is proposed. It is shown that on average the new types of predictors are better than those obtained by MMSE technique. Then, a two-context only system is on average better than a single predictor one. It also appears that 7-bit precision of PARCOR coefficients in the MPEG-4 ALS standard is insufficient for some recordings and predictor length, and that for very long frames coding results improve with the predictor rank practically in unlimited way.

关键词： audio coding Lossless Optimization Procedure Exponential Growth Recordings

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A design of lossy and lossless scalable audio coding 25

A design of lossy and lossless scalable audio coding

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Moriya, T Iwakami, N Jin, A Mori, T NTT Labs Musashino Tokyo 1808585 Japan

ISBN: (纸本)0780362934

This paper proposes a lossless audio coding scheme making use of high-compression lossy coding such as the ISO/IEC MPEG-4 audio standard. The encoding process consists of lossy compression, bit slice lossless data conversion, and lossless coding. The lossy compression significantly reduces the amplitude of the error signal between the input signal and the reconstructed signal. In addition, bit slice data conversion makes the lossless coding more efficient. As a result of preliminary experiment, a total file size can be reduced to 70 - 50% of the original size, which is significantly more efficient than conventional universal lossless coding schemes such as "gzip". The resultant compressed data is useful for various applications, since it has bitrate scalability and can be used as a highly compressed signal as well as a lossless signal.

关键词： Lossless audio coding lossy compression lossless coding lossless compression input signal ERROR SIGNAL File Size data conversion Scalability

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PRE-ECHO REDUCTION IN TRANSFORM audio coding VIA TEMPORAL ENVELOPE CONTROL WITH MACHINE LEARNING BASED ESTIMATION 49

PRE-ECHO REDUCTION IN TRANSFORM AUDIO CODING VIA TEMPORAL EN...

引用

49th IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Kim, Jae-Won Jo, Byeongho Beack, Seungkwon Park, Hochong Kwangwoon Univ Seoul South Korea ETRI Daejeon South Korea

ISBN: (纸本)9798350344868;9798350344851

This paper proposes a new method for pre-echo reduction in transform-based audio coding by controlling the temporal envelope of the waveform. The proposed method comprises two operating modes: temporal envelope flattening and temporal envelope correction of a target signal. The proposed method estimates signal levels with a low temporal resolution from side information using machine learning and converts them into a signal to be applied to the target signal to flatten and correct the temporal envelope. It also adjusts the signals to maintain signal continuity between the non-transient and transient frames. The proposed method differs from conventional methods in that it directly modifies the waveform before encoding and after decoding, which makes it useful as a new coding tool for legacy codecs. A subjective performance evaluation confirms that the proposed method uses fewer bits to provide sound quality equivalent to that of the short-window transform.

关键词： audio coding machine learning pre-echo temporal envelope transform coding

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Low Delay Robust audio coding by Noise Shaping, Fractional Sampling, and Source Prediction

Low Delay Robust Audio Coding by Noise Shaping, Fractional S...

引用

Data Compression Conference (DCC)

作者： Ostergaard, Jan Aalborg Univ Dept Elect Syst Fredrik Bajers Vej 7 DK-9000 Aalborg Denmark

ISBN: (纸本)9781665403337

It was recently shown that the combination of source prediction, two-times oversampling, and noise shaping, can be used to obtain a robust (multiple-description) audio coding framework for networks with packet loss probabilities less than 10%. Specifically, it was shown that audio signals could be encoded into two descriptions (packets), which were separately sent over a communication channel. Each description yields a desired performance by itself, and when they are combined, the performance is improved. This paper extends the previous work to an arbitrary number of descriptions (packets) by using fractional oversampling and a new decoding principle. We demonstrate that, due to source aliasing, existing MSE optimized reconstruction rules from noisy sampled data, performs poorly from a perceptual point of view. A simple reconstruction rule is proposed, that improves the PEAQ objective difference grades (ODG) by more than 2 points. The proposed audio coder enables low-delay high-quality audio streaming on networks with late packet arrivals or packet losses. With a coding delay of 2.5 ms, and a total bitrate of 300 kbps, it is demonstrated that mean PEAQ ODGs around -0.65 can be obtained for 48 kHz (mono) music (pop & rock), and packet loss probabilities of 20%.

关键词： audio coding fractional sampling low delay Multiple descriptions noise shaping source predictions

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Efficient audio coding with optimized subband configurations

Efficient audio coding with optimized subband configurations

引用

Workshop on Applications of Signal Processing to audio and Acoustics

作者： Ali, MT Mian, MS Univ Engn & Technol Dept Elect Engn Lahore Pakistan

ISBN: (纸本)0780391543

A dynamic subbands design selection scheme for perceptual audio coding is presented which optimizes the coding efficiency in association with all improved utilization of the masking potential of the signal. The choices for subbands design Selection include critical hands, uniform subbands, uniform subbands with a non-uniform last hand and a non-uniform-non-critical (NUNC) subbands distribution. A particular consideration is given to corresponding critical band boundaries while formulating NUNC subbands, to minimize the perception of coding noise. The design selection decision is made by First identifying the best tradeoff between empty quantizer slots and amount of side information, followed by a minimization or the perceptual entropy (PE) estimates. A further analysis is carried Out to identify the prominent spectral peaks (PSPs), with a view to efficiently utilize their masking potential by the bit allocation algorithm which makes a bit cut first to subbands with PSPs. Test results based oil ITU-R recommendation BS.1 116 show that Our coding scheme performs slightly better as compared to MPEG-4 AAC verification model (V-M) for a majority of signal types from the SQAM database. The paper Concludes with a discussion Of future research implications of the work.

关键词： audio coding entropy ITU-R recommendation BS.1116 MPEG-4 AAC verification model SQAM database bit allocation algorithm coding efficiency coding noise critical bands dynamic subbands design selection scheme nonuniform last band nonuniform-noncritical subb audio coding Subbands Allocation Algorithm critical band entropy code efficiency side information

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