Perceptual Coding of Narrowband Audio Signals
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[1] K. H. Barratt. Digital Coding of Waveforms , 1985 .
[2] Allen Gersho,et al. Vector quantization and signal compression , 1991, The Kluwer international series in engineering and computer science.
[3] Andrzej Drygajlo,et al. Perceptual speech coding and enhancement using frame-synchronized fast wavelet packet transform algorithms , 1999, IEEE Trans. Signal Process..
[4] P. Kabal,et al. Perceptual coding of narrowband audio signals at 8 kbit/s , 1997, 1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding.
[5] Peter Monta,et al. Low rate audio coder with hierarchical filterbanks and lattice vector quantization , 1994, Proceedings of ICASSP '94. IEEE International Conference on Acoustics, Speech and Signal Processing.
[6] Takehiro Moriya,et al. Scalable audio coder based on quantizer units of MDCT coefficients , 1999, 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258).
[7] Takehiro Moriya,et al. A design of transform coder for both speech and audio signals at 1 bit/sample , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.
[8] J. Makhoul,et al. Vector quantization in speech coding , 1985, Proceedings of the IEEE.
[9] Raymond N. J. Veldhuis,et al. Bit Rates in Audio Source Coding , 1992, IEEE J. Sel. Areas Commun..
[10] Jürgen Herre,et al. Bridging the Gap: Extending MPEG Audio Down to 8 kbit/s , 1997 .
[11] Thilo Thiede,et al. A New Perceptual Quality Measure for Bit-Rate Reduced Audio , 1996 .
[12] Louis Dunn Fielder,et al. ISO/IEC MPEG-2 Advanced Audio Coding , 1997 .
[13] David L. Neuhoff,et al. Quantization , 2022, IEEE Trans. Inf. Theory.
[14] Yuan-Hao Huang,et al. A new forward masking model and its application to perceptual audio coding , 1999, 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258).
[15] Allen Gersho,et al. Auditory distortion measure for speech coding , 1991, [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing.
[16] Jean-Pierre Petit,et al. High-quality audio transform coding at 64 kbps , 1994, IEEE Trans. Commun..
[17] P. Mermelstein. G.722: a new CCITT coding standard for digital transmission of wideband audio signals , 1988, IEEE Communications Magazine.
[18] Michael G. Perkins,et al. Application of the Princen-Bradley filter bank to speech and image compression , 1990, IEEE Trans. Acoust. Speech Signal Process..
[19] Karlheinz Brandenburg,et al. The iso/mpeg-audio codec: A generic standard for coding of high quality digital audio , 1992 .
[20] Thomas P. Barnwell,et al. The design of perfect reconstruction nonuniform band filter banks , 1991, [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing.
[21] Tor A. Ramstad,et al. Fully vector-quantized subband coding with adaptive codebook allocation , 1984, ICASSP.
[22] E. Terhardt,et al. Algorithm for extraction of pitch and pitch salience from complex tonal signals , 1982 .
[23] Julius O. Smith,et al. Audio representations for data compression and compressed domain processing , 1998 .
[24] Louis Dunn Fielder,et al. AC-2 and AC-3: The Technology and Its Application , 1995 .
[25] Jont B. Allen,et al. Micromechanical Models of the Cochlea , 1992 .
[26] Daniel Schulz. Improving audio codecs by noise substitution , 1996 .
[27] Henrique S. Malvar. Lapped transforms for efficient transform/subband coding , 1990, IEEE Trans. Acoust. Speech Signal Process..
[28] Deepen Sinha,et al. Low bit rate transparent audio compression using adapted wavelets , 1993, IEEE Trans. Signal Process..
[29] Akihiko Sugiyama,et al. A 128 kb/s Hi-Fi Audio CODEC Based on Adaptive Transform Coding with Adaptive Block Size MDCT , 1992, IEEE J. Sel. Areas Commun..
[30] P. Noll,et al. Wideband speech and audio coding , 1993, IEEE Communications Magazine.
[31] G.G. Langdon,et al. Data compression , 1988, IEEE Potentials.
[32] P. Noll,et al. Adaptive transform coding of speech signals , 1977 .
[33] T. Q. Nguyen,et al. A simple design method for nonuniform multirate filter banks , 1994, Proceedings of 1994 28th Asilomar Conference on Signals, Systems and Computers.
[34] Schuyler R. Quackenbush. Coding of natural audio in MPEG-4 , 1998, Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP '98 (Cat. No.98CH36181).
[35] Deepen Sinha,et al. Audio compression at low bit rates using a signal adaptive switched filterbank , 1996, 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings.
[36] G.C.P. Lokhoff. DCC-digital compact cassette , 1991 .
[37] Kenneth C. Pohlmann,et al. Principles of Digital Audio , 1986 .
[38] Allen Gersho,et al. Constrained-storage quantization of multiple vector sources by codebook sharing , 1991, IEEE Trans. Commun..
[39] J. D. Johnston,et al. Estimation of perceptual entropy using noise masking criteria , 1988, ICASSP-88., International Conference on Acoustics, Speech, and Signal Processing.
[40] Kenzo Akagiri,et al. ATRAC: Adaptive Transform Acoustic Coding for MiniDisc , 1992 .
[41] Rainer Dipl.-Ing. Buchta,et al. The WorldStar- Sound Format , 1996 .
[42] Eric D. Scheirer. The MPEG-4 Structured Audio standard , 1998, Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, ICASSP '98 (Cat. No.98CH36181).
[43] James D. Johnston,et al. Transform coding of audio signals using perceptual noise criteria , 1988, IEEE J. Sel. Areas Commun..
[44] B. Atal,et al. Optimizing digital speech coders by exploiting masking properties of the human ear , 1978 .
[45] Mark J. T. Smith,et al. Time-varying analysis-synthesis systems based on filter banks and post filtering , 1995, IEEE Trans. Signal Process..
[46] S. Merrill Weiss. MPEG Audio Coding , 1996 .
[47] Peter Kabal,et al. Improving perceptual coding of narrowband audio signals at low rates , 1999, 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258).
[48] Louis Dunn Fielder,et al. AC-2 and AC-3: Low-Complexity Transform-Based Audio Coding , 1996 .
[49] Vladimir Cuperman. On adaptive vector transform quantization for speech coding , 1989, IEEE Trans. Commun..
[50] Gerhard Eckel,et al. The Perception of Audio Signals Reduced by Overmasking to the Most Prominent Spectral Amplitudes (Peaks) , 1992 .
[51] Gilbert A. Soulodre,et al. Adaptive Methods for Removing Camera Noise from Film Soundtracks , 1998 .
[52] Yair Shoham. Vector predictive quantization of the spectral parameters for low rate speech coding , 1987, ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing.
[53] Marina Bosi,et al. Use of Low Bit-Rate Coding for High Quality Audio Over Telephone Lines , 1992 .
[54] Schuyler Quackenbush,et al. Objective measures of speech quality , 1995 .
[55] Paul M. McCourt. Critical band quantisation analysis for masked distortion speech coding , 1996, 1996 8th European Signal Processing Conference (EUSIPCO 1996).
[56] Henrique S. Malvar,et al. The LOT: transform coding without blocking effects , 1989, IEEE Trans. Acoust. Speech Signal Process..
[57] T. Ramstad,et al. Cosine-modulated analysis-synthesis filterbank with critical sampling and perfect reconstruction , 1991, [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing.
[58] William M. Hartmann,et al. Psychoacoustics: Facts and Models , 2001 .
[59] Marina Bosi,et al. High-Quality, Low-Rate Audio Transform Coding for Transmission and Multimedia Applications , 1992 .
[60] Sadaoki Furui,et al. Advances in Speech Signal Processing , 1991 .
[61] Bernd Edler. Current Status of the MPEG-4 Audio Verification Model Development , 1996 .
[62] Mark J. T. Smith,et al. Time-domain filter bank analysis: a new design theory , 1992, IEEE Trans. Signal Process..
[63] Ag Armin Kohlrausch,et al. Waveform coding and auditory masking , 1995 .
[64] Brian C. J. Moore. Masking in the Human Auditory System , 1996 .
[65] P. Jacobs,et al. Qcelp: The North American Cdma Digital Cellular Variable Rate Speech Coding Standard , 1993, Proceedings., IEEE Workshop on Speech Coding for Telecommunications,.
[66] James David Johnston,et al. Enhancing the Performance of Perceptual Audio Coders by Using Temporal Noise Shaping (TNS) , 1996 .
[67] Davis Pan,et al. A Tutorial on MPEG/Audio Compression , 1995, IEEE Multim..
[68] Deepen Sinha,et al. AT&T Perceptual Audio Coding (PAC) , 1996 .
[69] Todor Cooklev,et al. Compression of High-Quality Audio Signals, Including Recent Methods Using Wavelet Packets , 1996, Digit. Signal Process..
[70] Jun Matsumoto,et al. Harmonic and noise coding of LPC residuals with classified vector quantization , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.
[71] E. Owens,et al. An Introduction to the Psychology of Hearing , 1997 .
[72] Francis Rumsey. Putting Low-Bit-Rate Audio to Work , 1996 .
[73] Yair Shoham,et al. Hierarchical vector quantization of speech with dynamic codebook allocation , 1984, ICASSP.
[74] Pierrick Philippe,et al. Wavelet packet filterbanks for low time delay audio coding , 1999, IEEE Trans. Speech Audio Process..
[75] E. Zwicker,et al. Audio engineering and psychoacoustics: matching signals to the final receiver, the human auditory system , 1991 .
[76] Marcus Purat,et al. Audio coding with a dynamic wavelet packet decomposition based on frequency-varying modulated lapped transforms , 1996, 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings.
[77] Allen Gersho,et al. Advances in speech and audio compression , 1994, Proc. IEEE.
[78] Hugo Fastl,et al. Psychoacoustics: Facts and Models , 1990 .
[79] Robert M. Gray,et al. An Algorithm for Vector Quantizer Design , 1980, IEEE Trans. Commun..
[80] John Princen. The design of nonuniform modulated filterbanks , 1995, IEEE Trans. Signal Process..
[81] Eliathamby Ambikairajah,et al. Comparison of auditory masking models for speech coding , 1997, EUROSPEECH.
[82] H. Bastian. Sensation and Perception.—I , 1869, Nature.
[83] Thomas Sporer,et al. Evaluating a Measurement System , 1995 .
[84] Peter No,et al. Digital Coding of Waveforms , 1986 .
[85] Martin Vetterli,et al. Perfect reconstruction FIR filter banks: some properties and factorizations , 1989, IEEE Trans. Acoust. Speech Signal Process..
[86] Henrique S. Malvar,et al. Signal processing with lapped transforms , 1992 .
[87] Shing-Chow Chan. The generalized lapped transform (GLT) for subband coding applications , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.
[88] P. Noll,et al. Approaches to adaptive transform speech coding at low bit rates , 1979 .
[89] Takehiro Moriya,et al. Extension and complexity reduction of TwinVQ audio coder , 1996, 1996 IEEE International Conference on Acoustics, Speech, and Signal Processing Conference Proceedings.
[90] John Princen,et al. Analysis/Synthesis filter bank design based on time domain aliasing cancellation , 1986, IEEE Trans. Acoust. Speech Signal Process..
[91] K. Brandenburg. Audio coding for TV and multimedia , 1995 .
[92] James A. Storer,et al. Data Compression , 1992, Inf. Process. Manag..
[93] Mark B. Sandler,et al. On the performance of wavelets for low bit rate coding of audio signals , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.
[94] N. Spencer. An overview of digital telephony standards , 1998 .
[95] Bernhard Feiten,et al. Dynamically Scalable Internet Audio Transmission , 1998 .
[96] Mark J. T. Smith,et al. Analysis-synthesis systems with time-varying filter bank structures , 1992, [Proceedings] ICASSP-92: 1992 IEEE International Conference on Acoustics, Speech, and Signal Processing.
[97] Akihiko Sugiyama,et al. Adaptive transform coding with an adaptive block size (ATC-ABS) , 1990, International Conference on Acoustics, Speech, and Signal Processing.
[98] P. P. Vaidyanathan,et al. Multirate digital filters, filter banks, polyphase networks, and applications: a tutorial , 1990, Proc. IEEE.
[99] Ronald E. Crochiere,et al. Frequency domain coding of speech , 1979 .
[100] P. Urcun,et al. A MUSICAM source codec for digital audio broadcasting and storage , 1991, [Proceedings] ICASSP 91: 1991 International Conference on Acoustics, Speech, and Signal Processing.
[101] Jean-Bernard Rault,et al. A New Noise Injection Model for Audio Compression Algorithms , 1996 .
[102] Robert Friedrich,et al. Audio Compression for Network Transmission , 1996 .
[103] Bernd Edler. Speech coding in MPEG-4 , 1999, Int. J. Speech Technol..
[104] S. Wada. Design of nonuniform division multirate FIR filter banks , 1995 .
[105] Bernd Edler. Very Low Bit Rate Audio Coding Development , 1997 .
[106] John Princen,et al. Audio coding with signal adaptive filterbanks , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.
[107] Henrique S. Malvar. Lapped biorthogonal transforms for transform coding with reduced blocking and ringing artifacts , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.
[108] Andreas Spanias,et al. Speech coding: a tutorial review , 1994, Proc. IEEE.
[109] Andreas Spanias,et al. A review of algorithms for perceptual coding of digital audio signals , 1997, Proceedings of 13th International Conference on Digital Signal Processing.
[110] Michel C. Lavoie,et al. Subjective evaluation of state-of-the-art two-channel audio codecs , 1998 .
[111] Bernd Edler,et al. Object-Based Analysis/Synthesis Audio Coder for Very Low Bit Rates , 1998 .
[112] John Princen,et al. Subband/Transform coding using filter bank designs based on time domain aliasing cancellation , 1987, ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing.
[113] Karlheinz Brandenburg. OCF--A new coding algorithm for high quality sound signals , 1987, ICASSP '87. IEEE International Conference on Acoustics, Speech, and Signal Processing.
[114] S. P. Lloyd,et al. Least squares quantization in PCM , 1982, IEEE Trans. Inf. Theory.
[115] Takehiro Moriya,et al. High-quality audio-coding at less than 64 kbit/s by using transform-domain weighted interleave vector quantization (TwinVQ) , 1995, 1995 International Conference on Acoustics, Speech, and Signal Processing.
[116] William C. Treurniet,et al. Objective Perceptual Measurement of Audio Quality , 1996 .
[117] Ricardo L. de Queiroz,et al. Time-varying lapped transforms and wavelet packets , 1993, IEEE Trans. Signal Process..
[118] Redwan Salami,et al. GSM enhanced full rate speech codec , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.
[119] Bob Novorita,et al. Incorporation of temporal masking effects into bark spectral distortion measure , 1999, 1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258).
[120] John G. Beerends,et al. A Perceptual Audio Quality Measure Based on a Psychoacoustic Sound Representation , 1992 .
[121] Jean-Pierre Adoul,et al. Enhanced full rate speech codec for IS-136 digital cellular system , 1997, 1997 IEEE International Conference on Acoustics, Speech, and Signal Processing.