76 results about "Waveform coding" patented technology

An audio encoding apparatus capable of improving the frame cancellation error tolerance, without increasing the number of bits of a fixed code book, in a CELP type audio encoding. In this apparatus, a low frequency component waveform encoding part (210) calculates, based on a quantized LPC received from an LPC encoding part (202), a linear prediction residual signal of a digital audio signal received from an A/D converter (112), then performs a down sampling of the calculation result to extract the low frequency components comprising bands, which are lower than a predetermined frequency, in the audio signal, and then waveform encodes the extracted low frequency components to produce encoded low-frequency component information. Then, the low frequency component waveform encoding part (210) inputs this encoded low-frequency component information to a packetizing part (231), while inputting the quantized low-frequency component waveform encoded signal (sound source waveform), which has been produced by the waveform encoding, to a high frequency component encoding part (220).

Linear predictive speech coding system with classification of frames and a hybrid coder using both waveform coding and parametric coding for different classes of frames. Phase alignment for a parametric coder aligns synthesized speech frames with adjacent waveform coder synthesized frames. Zero phase alignment of speech prior to waveform coding aligns synthesized speech frames of a waveform coder with frames synthesized with a parametric coder. Inter-frame interpolation of LP coefficients suppresses artifacts in resultant synthesized speech frames.

A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in terms of the encoding units, whereby explosive and fricative consonants can be impeccably reproduced, while there is an attenuation of the occurrence of foreign sounds being generated at a transient portion between voiced (V) and unvoiced (UV) portions, so that the speech with high clarity devoid of “stuffed” feeling may be produced. The encoding apparatus includes a first encoding unit for finding residuals of linear predictive coding (LPC) of an input speech signal for performing harmonic coding and a second encoding unit for encoding the input speech signal by waveform coding. The first encoding unit and the second encoding unit are used for encoding a voiced (V) portion and an unvoiced (UV) portion of the input signal, respectively. Code excited linear prediction (CELP) encoding employing vector quantization by a closed loop search of an optimum vector using an analysis-by-synthesis method is used for the second encoding unit. A corresponding decoding method and apparatus is also provided.

Hybrid linear predictive speech coding system with phase alignment predictive quantization zero phase alignment of speech prior to waveform coding aligns synthesized speech frames of a waveform coder with frames synthesized with a parametric coder. Inter-frame interpolation of LP coefficients suppresses artifacts in resultant synthesized speech frames.

The invention discloses a single acoustic path sound coding device using an analysis sub-band filter set module to decompound a single acoustic path sound signal into a low-frequency sub-band domain signal comprising at least two sub-bands and a high-frequency sub-band domain signal comprising at least two sub-bands; the low-frequency sub-band domain signal is performed with prediction analysis by a low-frequency sub-band domain time-varying prediction analysis module, is performed with time domain transformation by a low-frequency sub-band domain time-frequency transformation module, and is performed with quantification coding by a low-frequency sub-band domain waveform coding module so as to obtain low-frequency sub-band domain waveform coding data; high-frequency sub-band domain parameter coding data is extracted from the high-frequency sub-band domain signal by parameter coding of a high-frequency sub-band domain parameter coding module; the low-frequency sub-band domain waveform coding data and the high-frequency sub-band domain parameter coding data are multiplexed by a bit-stream multiplexing module to output a sound code stream. The invention also discloses a single acoustic path sound decoding device, a single acoustic path sound coding/decoding method, a dimensional sound coding/decoding device and method, all of which can realize high-quality coding of wide band sound in a low code rate.

The invention provides an absolute time delay detection device of an intelligent substation mutual inductor data collection system. The absolute time delay detection device comprises a three-phase programmable power supply, an electronic transformer, a merging unit, a Hall sensor and a control workstation. The invention further provides an absolute time delay detection method of the intelligent substation mutual inductor data collection system. The three-phase power supply with the output voltage waveform modulated is adopted, and voltage output with coded information is achieved, and the Hall sensor is used for recording a corresponding waveform coding time sequence. Meanwhile, the control workstation detects and records corresponding output signals of the detected electronic current transformer and the merging unit, and the corresponding output signals are compared with the waveform coding time sequence recorded by the Hall sensor so as to obtain the absolute time delay of the mutual inductor data collection system. According to the absolute time delay detection device and method of the intelligent substation mutual inductor data collection system, accuracy of measurement can be ensured, meanwhile, accurate judgment and detection can be performed on the full period time delay with the detection method, and in addition, the phase difference of the current and the voltage can be detected under the condition of the same source.

Hybrid linear predictive speech coding system with phase alignment predictive quantization zero phase alignment of speech prior to waveform coding aligns synthesized speech frames of a waveform coder with frames synthesized with a parametric coder. Inter-frame interpolation of LP coefficients suppresses artifacts in resultant synthesized speech frames.

The invention relates to a sound coding and decoding apparatus and a sound coding and decoding method, and particularly relates to a monophony sound coding and decoding apparatus, a monophony sound coding and decoding method, a stereo sound coding and decoding apparatus, and a stereo sound coding and decoding method. The sound coding and decoding method is characterized in that low frequency waveform coding is combined with high frequency parameter coding, and a specific frequency range of a low frequency spectrum is mapped to a specific frequency range of a high frequency spectrum; at a coding end, the high frequency spectrum before and after spectrum mapping operation is subjected to mapping domain boundary preprocessing, the high frequency spectrum after parameter decoding operation is subjected to mapping domain boundary post-processing, problems brought by frequency band dividing operation and low frequency spectrum to high frequency spectrum mapping operation in high frequency parameter coding operation are overcome, frequency spectrum continuity and frequency band signal naturalness are improved, harmonic wave interference noise and aliasing noise caused by sidelobe leakage are eliminated, and high frequency parameter coding quality is further improved under the condition of a low code rate.

The invention discloses a method integrating a radar detection function with a microwave communication function, which is characterized by comprising the following steps of: measuring the speed of high-speed movement equipment by transmitting a measuring signal to obtain corresponding Doppler frequency shift; providing a frequency source by using a signal processing device, and selecting a waveform code corresponding to a frequency fdata; and transmitting carrier frequency signals with frequencies f0 to fd at a transmitting end during communication, wherein the frequency of a receiver is the same as that of the carrier frequency signals transmitted at the transmitting end, and a signal receiving device of the high-speed movement equipment and a signal receiving device of the common low-speed movement or static equipment have the same structure. The technology overcomes the defect that the signal receiving device in the high-speed movement equipment and the signal receiving device in the low-speed movement or static equipment have different structures because compensation is performed at a signal receiving end of the high-speed movement equipment in the conventional radio communication system.

Hybrid linear predictive speech coding system with phase alignment predictive quantization zero phase alignment of speech prior to waveform coding aligns synthesized speech frames of a waveform coder with frames synthesized with a parametric coder. Inter-frame interpolation of LP coefficients suppresses artifacts in resultant synthesized speech frames.

The invention discloses a decoding method and a decoding system of a fluorescent encoded microsphere. The decoding method of the fluorescent encoded microsphere comprises the step of exciting, namely, irradiating a fluorescent microsphere with laser; receiving a spectrum waveform emitted by a certain fluorescent microsphere which is excited by the laser; and determining, namely, selecting a certain reference spectrum waveform which is similar to the spectrum waveform from a plurality of reference spectrum waveforms in a reference spectrum waveform library and determining that the fluorescent microsphere is a reference fluorescent microsphere which corresponds to the certain reference spectrum waveform. According to the decoding method and the decoding system, more fluorescent materials can be utilized for encoding the microsphere; the fluorescent microsphere distinguishing accuracy is improved; the fluorescent microspheres can be simply prepared, transported and preserved; more waveform codes can be adopted, i.e., the number of microsphere codes is increased.

The invention discloses a multi-waveform phase coding method based on a mode search algorithm, belongs to the technical field of radars, and especially relates to a radar waveform coding method. A multi-waveform autocorrelation and cross-correlation function obtained through the method has a low sidelobe level in a specific time delay region, thereby solving a problem that the calculation work is large in the prior art. The implementation process of the method comprises the steps: 1, constructing a target function of a multi-dimensional optimization problem; 2, simplifying the target function, and employing the mode search algorithm to solve the simplified target function; 3, carrying out iteration for many times till a system cannot accept any phase change, i.e., obtaining a waveform design meeting the demands of related characteristics. Compared with a Weighted Cyclic Algorithm New, WeCAN method, the method is high in rate of convergence, is small in calculation work, and can effectively design a phase-coding multiple waveforms with lower autocorrelation and cross-correlation sidelobe in a specific time delay region.

The invention relates to an acoustic decoding method. The method comprises the following steps: demultiplexing an acoustic code stream to obtain low-frequency waveform code data and high-frequency parameter code data; decoding the low-frequency waveform code data to obtain low-frequency spectrum data in an MDCT domain; converting the low-frequency spectrum data in the MDCT domain into an MDFT domain; mapping part of the spectrum data in the low-frequency spectrum data in the MDFT domain to the high-frequency part; conducting parameter decoding on the mapped high-frequency spectrum data according to the high-frequency parameter code data; combining the low-frequency spectrum data in the MDCT domain and the high-frequency spectrum data subjected to parameter decoding in the MDFT domain to form full-frequency spectrum data in the MDFT domain, and finally obtaining acoustic decoding data in a time domain through IMDFT. By means of the acoustic decoding method, an MDFT low frequency spectrum is directly calculated through an MDCT low frequency spectrum, conversion between IMDCT and MDFT is avoided, errors caused by introduction of low-frequency MDFT estimation during MDFT frequency-time domain transformation are avoided by splicing the low-frequency DMCT spectrum and a high-frequency MDFT spectrum into a full-frequency-band MDFT spectrum, computation complexity and computation errors of a decoder are reduced on the premise of maintaining coding quality.

An encoder (109) comprises a receiver (201) which receives a time domain audio signal. A filter bank (203) generates a first subband signal from the time domain audio signal where the first subband signal corresponds to a non-critically sampled complex subband domain representation of the time domain signal. A conversion processor (205) generates a second subband signal from the first subband signal by subband processing. The second subband signal corresponds to a critically sampled complex subband domain representation of the time domain audio signals. An encode processor (207) then generates a waveform encoded data stream by encoding data values of the second subband signal. The conversion processor (205) generates the second subband signal by direct subband conversion without converting back to the time domain. The invention allows an oversampled subband signal typically generated in parametric encoding to be waveform encoded with reduced complexity. A decoder performs the inverse operation.

There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

An audio signal is encoded by a first waveform encoder (103) to generate a first waveform based bit-stream component. A second encoder (105) encodes the audio signal to generate a second bit-stream component comprising first enhancement data and a third encoder (107) encodes the audio signal to generate a third bit-stream component comprising second enhancement data for the first waveform based bit-stream component. The first and second bit-stream components correspond to a first representation of the audio signal and the first and third bit-stream components correspond to a second representation of the audio signal. A scalable audio bit-stream is generated by a bit-stream generator (109). The different representations may be selected between by a decoder thereby allowing a flexible and scalable bit-stream to be communicated. The second encoder (105) may specifically be a waveform encoder and the third encoder (107) may specifically be a parametric encoder.

Waveform-coding is applied to map successive on-off-keying (OOK) data bits onto successive multicarrier modulated symbols in time domain, wherein each multicarrier modulated symbol includes a set of sub-carriers in which alternating sub-carriers are set to ones and zeros in frequency domain. The waveform coded multicarrier modulated symbols are up-converted to a carrier frequency to provide a datasignal that is transmitted over a wireless channel.

The invention discloses an automatic proving program product including the circuit function of an automatic proving tool modular and an emulator tool modular, which comprises the following steps: inputting storage unit circuit description and storage unit circuit function description to the automatic proving tool modular; generating the proving input code and the proving result code; calling the emulator tool modular by the automatic proving tool, inputting the proving input code and getting the emulator result code; comparing the emulator result code and the proving result code to the automatic proving tool modular if selecting the internal detection, generating the right report if the same, generating the wrong report if the different; converting the emulator result code and the proving result code to the other computer language by the automatic proving tool modular and analyzing and judging by the circuit function proving product of the other converted language circuit function if selecting the external detection. The storage unit circuit function includes the logical relation and the sequential relationship file of the input and output signal; the code includes various waveform codes.