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Card-sized portable audio player using high quality audio coding technology twin VQ

IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 2000年第3期46卷 907-913页

作者： Takahashi, T Morita, T Shimoda, T Kaneko, T Moriya, T Kobe Steel Ltd Kobe Hyogo Japan NTT Cyber Space Labs Tokyo Japan

This paper presents a new implementation of the solid state audio player equipped with TwinVQ decoder. TwinVQ is an advanced high quality coding technology, and its basic algorithm is utilized in MPEG-4/audio standardized by ISO (International Standardization Organization). Using TwinVQ coding, 44.1 kHz sampled audio data can be compressed at a very low bitrate of 40 kbit/s/ch with better sound quality than conventional coding methods as MPEG-1/audio layer-3. Another feature of the player is that it is extremely compact, business-card-sized, 10 milt thick, including barren, as a result of the efficient design. Decoding algorithms are optimized and implemented into 16 bit fired-point DSP, and a vent thin rechargeable lithium polymer battery is contained integral with the player. As a result, the new player has very practical performance with low power consumption. and 6 hours of 44 kHz stereo audio can be continuously played when the battery is fully charged.

关键词： .solid state HIGH QUALITY MPEG audio PLAYER USING audio coding TECHNOLOGY TWIN VQ audio Player

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Robust signal/noise discrimination for wideband speech and audio coding

Robust signal/noise discrimination for wideband speech and a...

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7th IEEE Workshop on Speech coding

作者： Jelinek, M Labonté, F Univ Sherbrooke Dept Elect & Comp Engn Sherbrooke PQ J1K 2R1 Canada

ISBN: (纸本)0780364163

We present a robust discrimination method to separate information carrying signals from ambient noise for wideband coding applications. The method consists of a two-stage procedure. First. a local decision is made based on a set of extracted parameters to update the estimated noise level. The parameters has been chosen to detect reliably speech as well as music signals. Then, the final decision is made based only on a frequency dependent Signal to Noise Ratio (SNR). The noise level update does not depend on the final decision to prevent the discriminator from locking when noise level changes suddenly. The performance is compared with the performance of the Voice Activity Detector (VAD) of G.729, Annex G.

关键词： audio coding Multiple signal classification Noise level Noise robustness Signal detection Signal processing Signal to noise ratio Speech coding Speech enhancement Wideband

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Trellis-based optimization of MPEG-4 advanced audio coding

Trellis-based optimization of MPEG-4 advanced audio coding

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7th IEEE Workshop on Speech coding

作者： Aggarwal, A Regunathan, SL Rose, K Univ Calif Santa Barbara Dept Elect & Comp Engn Santa Barbara CA 93106 USA

ISBN: (纸本)0780364163

We outline a method to perform efficient low rate quantization for MPEG-4 Advanced audio coding (AAC). The AAC bit stream consists of indices for quantized spectral coefficients as well as side information about quantizer step sizes and Huffman codebooks. The MPEG-4 Verification Model does not explicitly account far side information bits in its optimization and suffers from poor compression efficiency at low bit rates. We reformulate the encoding problem as one of optimal parameter selection, where the side information bits are taken into account, so as to minimize the noise to mask ratio for the given target bit rate. The optimal solution is determined by a dynamic programming procedure that efficiently searches through a trellis. This trellis-based optimization greatly improves the low bit rate performance of AAC and, consequently, the perfomance of a multi-laver AAC system. The resulting bit stream is standard-compatible, and additional complexity due to the proposed optimization is only incurred at the encoder.

关键词： audio coding audio compression Bit rate Dynamic programming Encoding MPEG 4 Standard Quantization Signal to noise ratio Streaming media Virtual manufacturing

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A design of lossy and lossless scalable audio coding 25

A design of lossy and lossless scalable audio coding

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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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Combined speech and audio coding by discrimination

Combined speech and audio coding by discrimination

引用

7th IEEE Workshop on Speech coding

作者： Tancerel, L Ragot, S Ruoppila, VT Lefebvre, R Univ Sherbrooke Dept Elect Engn Sherbrooke PQ J1K 2R1 Canada

ISBN: (纸本)0780364163

We propose in this paper a general solution for combined speech and audio coding. Particularly, we describe a speech/music discrimination procedure for multi-mode wideband coding. The speech/music decision is updated only when a low-energy frame is detected, and kept unchanged otherwise. The signal is classified using second-order statistics of discriminant parameters. An experimental CELP/transform coder operating at 16 kbit/s is demonstrated. Results show improved performance when compared to single-mode encoding.

关键词： audio coding Bit rate Delay estimation Forward contracts Linear predictive coding Multiple signal classification Speech coding Statistics Switches Wideband

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Binaural cue coding: a novel and efficient representation of spatial audio

Binaural cue coding: a novel and efficient representation of...

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

作者： Christof Faller Frank Baumgarte Media Signal Processing Research Agere Systems Murray Hill NJ USA

We present a novel concept for representing multi-channel audio signals: Binaural Cue coding (BCC). BCC aims at separating the basic audio content and the information relevant for spatial perception. A multi-channel audio signal is represented as a mono signal and BCC parameters. We present two types of applications of BCC. Firstly, a number of separate sound source signals are reduced to a mono signal and BCC parameters. In this case, the decoder has control over the location of each source in auditory space. In other words, the decoder can render spatial images as if the separate source signals were given. Secondly, a multi-channel audio signal is reduced to a mono signal and BCC parameters. In this case the decoder generates a multi-channel signal with a spatial image similar to the spatial image of the input signal of the encoder. Results from a subjective test suggest that BCC, combined with existing mono audio coders, offers better quality than conventional stereo and multi-channel perceptual transform audio coders for a wide range of bitrates.

关键词： MONOS devices Variable speed drives Transforms Bit rate Media audio coding

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audio coding using a psychoacoustic pre- and post-filter

Audio coding using a psychoacoustic pre- and post-filter

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

作者： B. Edler G. Schuller Multimedia Communications Research Laboratory Bell Laboratories Lucent Technologies Inc. Murray Hill NJ USA

A novel concept for perceptual audio coding is presented which is based on the combination of a pre- and post-filter, controlled by a psychoacoustic model, with a transform coding scheme. This paradigm allows modeling of the temporal and spectral shape of the masked threshold with a resolution independent of the used transform. By using frequency warping techniques the maximum possible detail for a given filter order can be made frequency-dependent and thus better adapted to the human auditory system. The filter coefficients are represented efficiently by LSF parameters which can be adaptively interpolated over time. First experiments with a system obtained by extending an existing transform codec showed that this approach can significantly improve the performance for speech signals, while the performance for other signals remained the same.

关键词： audio coding Psychology Psychoacoustic models Frequency Transform coding Spectral shape Humans Auditory system Adaptive filters Codecs

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Estimation of auditory spatial cues for binaural cue coding

Estimation of auditory spatial cues for binaural cue coding

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Baumgarte, F Faller, C Agere Syst Media Signal Proc Res Murray Hill NJ USA

ISBN: (纸本)0780374029

Binaural Cue coding (BCC) offers a compact parametric representation of auditory spatial information. This representation can be applied to stereophonic or multi-channel audio compression. It allows to reconstruct the spatial image given a mono audio signal and spatial cues that require only a low data volume. This paper focuses on the extraction of the spatial cues from a stereophonic signal with a BCC analyzer. Results of a first subjective quality evaluation confirm that this technique is able to approximate the spatial image of critical reference signals. Thus BCC has the potential to offer highly efficient stereophonic or multi-channel coding, even at very low bit rates that currently permit coding of only one mono channel.

关键词： audio coding signal reconstruction signal representation data compression channel coding auditory spatial cues estimation binaural cue coding parametric representation auditory spatial information stereophonic audio compression multichannel audio compression spatial image reconstruction mono audio signal low data volume spatial cues extraction stereophonic signal subjective quality evaluation reference signals multichannel coding stereophonic coding audio coding

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Robust speech coding under packet-loss conditions using recovery sub-codec for broad-band IP network

Robust speech coding under packet-loss conditions using reco...

引用

IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Morinaga, T Sasaki, S Mano, K Kaneko, T NTT Corp NTT Cyber Space Labs Tokyo Japan

ISBN: (纸本)0780374029

A speech-coding method called "recovery sub-codec (RSC)" is described that improves speech quality under packet-loss conditions, which can occur even in high-speed wide-band networks. The RSC uses a sub bit-stream as additional information in the next speech frame in addition to the main bit-stream in the current frame. The main bit-stream is coded using a low-compression high-quality main coder and the sub bit-stream is coded using a high-compression sub-coder. The internal states of the recovery sub-coder can be updated using the output from the high-quality main coder, so they are independent of previous frames and are not affected by previous packet loss. Subjective quality tests of several speech codecs based on the proposed method show that it significantly improves the concealment of packet loss, both for narrow- and wide-band speech and audio signals.

关键词： audio coding speech coding broadband networks speech codecs telecommunication network reliability voice communication speech coding packet-loss conditions recovery sub-codec broadband IP network bitstream audio signals

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Software optimization of the MPEG-audio decoder using a 32-bit MCU RISC processor

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 2002年第3期48卷 671-676页

作者： Lee, KS Park, YC Youn, DH Yonsei Univ Dept Elect & Elect Engn MCSP Lab Seoul 120749 South Korea Yonsei Univ Dept Div Informat Technol Wonju South Korea

This paper proposes optimization techniques to implement MPEG audio decoding algorithms in real-time using general-purpose 32-bit MCU RISC processor. Both MP3 and AAC LC profile decoding algorithms can be partitioned into two parts: control-intensive part and computation-intensive part. Optimization techniques in this study are focused on developing methodologies that are suitable for each part. We implemented MP3/AAC decoder using ARM-based RISC MPU and its test board. The both designed decoders with proposed optimizations could achieve decoding processes in real-time within an operating frequency of 35 MHz. And ISO 13818-4 compliance test results confirmed that the proposed decoders ensured full compliance with ISO 13818-3 audio decoder. These implementation results illustrate the efficiency of the proposed design methodology in both performance and cost.

关键词： code standards telecommunication standards audio coding decoding digital signal processing chips reduced instruction set computing optimisation software optimization MPEG-audio decoder MPEG audio decoding algorithms general-purpose MCU RISC code standards audio coding RISC processors decoders digital signal processing chips micro-controller unit MCU gene DEcoding

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