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检索条件"主题词=linear predictive coding"

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Diphone synthesis for phonetic vocoding

Diphone synthesis for phonetic vocoding

引用

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

作者： R. Schwartz J. Klovstad J. Makhoul D. Klatt V. Zue Bolt Beranek and Newman 슠Inc. Cambridge MA USA

We report on the synthesis of speech in the context of a phonetic vocoder operating at 100 b/s. With each phoneme, the vocoder transmits the duration and a single pitch value. The synthesizer uses a large inventory of diphone "models" to synthesize a desired phoneme string. The diphone inventory has been selected to differentiate between prevocalic and postvocalic allophones of sonorants, to account for changes in vowel color conditioned by postvocalic liquids, to allow exact specification of voice onset time, and to permit synthesis of glottal stops alveolar flaps and syllabic consonants. The diphones are extracted from carefully constructed short utterances and are stored as a sequence of LPC parameters. During synthesis, the requisite diphone models are time-warped, abutted and smoothed to produce a complete sequence of LPC parameters that are used in the synthesis. The algorithms used are described and compared with more conventional methods. Examples of the synthesized speech will be played.

关键词： Speech synthesis linear predictive coding Synthesizers Vocoders Fasteners Liquids Assembly systems Interpolation Joining processes Steady-state

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A preprocessing filter for enhancing LPC analysis/Synthesis of noisy speech

A preprocessing filter for enhancing LPC analysis/Synthesis ...

引用

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

作者： M. Sambur Defense Communications Division ITT Nutley NJ USA

The goal of this study was to develop an effective and computationally inexpensive method of enhancing the linear prediction analysis/synthesis of noisy speech. To this end, a preprocessing filter has been proposed that is capable of perfectly removing the "expected" noise signal when the input speech spectrum is closely approximated by the noisy speech spectrum. The proposed filter has been evaluated by the linear prediction distance measure, perceptual listening, and spectrograms. This evaluation has demonstrated the effectiveness of the filter for broadband noise removal. The filter has also been implemented as a preprocessing filter in a real time LPC system. The total processing time for the filtering is only 2.6 msec per 22.5 msec frame. In this system, the LPC analysis and synthesis takes a combined time of 13 msec.

关键词： Speech analysis Speech synthesis linear predictive coding Speech enhancement Signal synthesis Data preprocessing Nonlinear filters Spectrogram Real time systems Filtering

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Narrowband LPC speech transmission over noisy channels

Narrowband LPC speech transmission over noisy channels

引用

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

作者： E. Blackman R. Viswanathan W. Russell J. Makhoul Bolt Beranek and Newman 슠Inc. Cambridge MA USA

Recently we described a variable-frame-rate LPC vocoder designed to transmit good quality speech over 2400 bps fixed-rate noisy channels with bit-error probabilities ranging up to 5% [3]. The basic idea was to lower the data rate by transmitting LPC parameters only when speech characteristics have changed sufficiently since the last transmission, and to employ the resulting bit-rate savings for protecting important transmission data against channel noise. This paper describes our continuing efforts which have concentrated on minimizing loss of synchronization between the receiver and the transmitter. In one approach, we emphasize heavy protection of header, and rapid resynchronization. Alternatively, we apply constraints which guarantee synchronization at a cost of some freedom in the selection of data for transmission. Results from the first approach are presented; results from both methods will be compared at the conference.

关键词： Narrowband linear predictive coding Protection Speech analysis Vocoders Costs Channel capacity Speech synthesis Fasteners Speech enhancement

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Microprocessor Implementation Of A linear predictive Coder

Microprocessor Implementation Of A Linear Predictive Coder

引用

IEEE Compcon

作者： A.J. Goldberg M. Leonardi L. Bergeron W. Lynch J. deLellis GTE Sylvania Inc. Needham Heights MA USA

This paper describes a linear predictive coder (LPQ and its microprocessor fabrication. The LPC has an audio bandwidth of 3200 Hz, uses the autocorrelation formulation of LPC to determine the short term spectrum, and an Average Magni- tude Difference Function (AMDF) to extract pitch. A two multiplier/stage lattice filter at the receiver recreates the speech. The 2400 b/s full duplex LPC is implemented in the firmware of a horizontally coded microprocessor having a 48-bit instruction word and 16-bit data word. The processor architecture uses a 4-bit TTL ALU slice and a hardwared 16 x 16 bit parallel multiplier to rapidly process the data with relatively slow multiplication circuitry. With the chosen architecture, the LPC requires only 60 percent of the proces sor's capacity, while the processor itself has fewer than 150 integrated components. The low cost of the voice digitizer has resulted in commercial sales in a market that appears to be growing.

关键词： Microprocessors linear predictive coding Costs Delay estimation Bandwidth Speech analysis Fabrication Autocorrelation Data mining Lattices

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A frequency domain noise cancelling preprocessor for narrowband speech communications systems

A frequency domain noise cancelling preprocessor for narrowb...

引用

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

作者： R. Preuss MITRE Corporation Bedford MA USA

Performance of narrowband speech communications systems, such as linear predictive coding (LPC), is often severely degraded by the presence of ambient acoustic noise in the input speech signal. Spectral subtraction techniques show promise in improving the overall performance of LPC in acoustic noise environments, but typically present annoying musical tones at the output. A spectral subtraction technique is described, which includes a biased estimate of the noise, that does not present musical tones at the output. In addition, an automatic speech activity detector is described and used to adapt the noise estimate to changing noise environments.

关键词： Frequency domain analysis Noise cancellation Narrowband Oral communication Acoustic noise linear predictive coding Working environment noise Speech enhancement Subtraction techniques Degradation

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Description of a hybrid 7.2 kbps vocoder

Description of a hybrid 7.2 kbps vocoder

引用

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

作者： H. Watkins KETRON Inc. Wayne Pennsylvania

A speech bandwidth compression system is described which employs a hybrid processing approach combining a baseband system and a linear predictive coding system to produce high quality speech at a transmission rate of 7.2 Kbps. The system requires the extraction and transmission of excitation parameters, but is not very sensitive to errors in those parameters nor is it particularly sensitive to errors in the baseband portion of the processing. Since the system is composed of two independent processes, errors in one process have no effect on the other process and the system is remarkably robust. The system is essentially a modification of a similar system which was operated at 16 Kbps (1) which utilizes unique coding techniques to reduce the bit rate to 7.2 Kbps.

关键词： Vocoders Baseband Signal processing linear predictive coding Speech coding Speech processing Robustness Frequency Transmitters Bandwidth

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Real-time text processing for Italian speech synthesis

Real-time text processing for Italian speech synthesis

引用

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

作者： E. Vivalda S. Sandri C. Miotti CSELT Torino Italy

The paper describes the software architecture of an Italian text-to-speech synthesis system based on the joining of LPC coded diphones. The automatic voice response system is designed according to multichannel and real time criteria. For each output channel, the following operations are performed: pre-processing of the input string of characters, translation into the proper sequence of diphones, generation of prosodic contours and real-time control of a hardware speech synthesizer.

关键词： Text processing Speech synthesis Synthesizers Computer interfaces Signal synthesis Software architecture linear predictive coding Real time systems Character generation Automatic generation control

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CCD adaptive filtering for robust LPC speech processing

CCD adaptive filtering for robust LPC speech processing

引用

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

作者： G. Borsuk M. White Westinghouse Advanced Technology Laboratories Baltimore MD USA

A monolithic CCD adaptive filter chip is described which implements the Widrow-Hoff "clipped-data" LMS adaptive algorithm. The chip can be used as a pre-filter noise canceller, analysis filter, or pre-whiten... 详细信息

关键词： Charge coupled devices Adaptive filters linear predictive coding Speech processing Noise robustness Least squares approximation Adaptive algorithm Noise cancellation Speech analysis Nonlinear filters

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Development of a 4.8-9.6 kbps RELP vocoder

Development of a 4.8-9.6 kbps RELP vocoder

引用

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

作者： M. Dankberg D. Wong Space and Secure Telecommnications Systems Division Rockwell International Newport Beach CA USA

A new version of the Residual Excited linear predictive (RELP) vocoder has been simulated. The objective has been to reduce the data rate required for good quality speech to 4.8 kbps. Results have indicated that it is possible to remove the hoarseness currently associated with low data rate RELP speech. Development of a pitch predictive ADPCM residual encoder and preliminary results on new harmonic generation techniques are discused. Taped demonstrations will be played at the conference.

关键词： Vocoders Baseband linear predictive coding Space technology Speech coding Speech enhancement Noise robustness Acoustic waves Humans Telecommunications

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A two-step speech compression system with vector quantizing

A two-step speech compression system with vector quantizing

引用

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

作者： A. Buzo A. Gray R. Gray J. Markel Signal Technology Inc. SantaBarbara CA Information Systems Laboratory University of Stanford Stanford CA USA

A training sequence of speech data is used to design a two-step speech compression system, based upon either single speakers or multiple speakers. The system is designed to minimize an average spectral distortion over the training sequence, leading to an identification step using linear prediction techniques followed by a vector quantizer. The system is then used to compress test sequences of speech data, leading to much lower bit rates than obtained using scalar quantization for equivalent distortions. For the same numerical distortion, 20-bits/frame were required using "optimal" scalar bit allocation and quantization, whereas 8-bits/frame were required using vector quantization. Results are presented in the form of numerical distortion measures and analog tapes of synthesized speech.

关键词： Distortion measurement Switches Speech processing Quantization linear predictive coding Random processes Noise measurement Vectors Bit rate Speech synthesis

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