87 results about "Modified discrete cosine transform" patented technology

Calculate the log frame energy value of each of a pre-determined number n of frames of an input speech signal and apply a matrix transform to the n log frame energy values to form a temporal matrix representing the input speech signal. The matrix transform may be a discrete cosine transform.

Estimates of spectral magnitude and phase are obtained by an estimation process using spectral information from analysis filter banks such as the Modified Discrete Cosine Transform. The estimation process may be implemented by convolution-like operations with impulse responses. Portions of the impulse responses may be selected for use in the convolution-like operations to trade off between computational complexity and estimation accuracy. Mathematical derivations of analytical expressions for filter structures and impulse responses are disclosed.

A memory-efficient system converting a signal from a first transform domain to a second transform domain. The system includes a first mechanism that obtains an input signal expressed via a first transform-domain signal representation. A second mechanism expresses the input signal via a second transform-domain signal representation without intermediate time-domain conversion. In the specific embodiment, the input signal is a Modified Discrete Cosine Transform (MDCT) signal. The second transform-domain signal representation is a Discrete Fourier Transform (DFT) signal. The second mechanism further includes a third mechanism that combines effects of an inverse MDCT, a synthesis window function, and an analysis window function, and provides a first signal in response thereto. A fourth mechanism converts the MDCT signal to the DFT signal based on the first signal. In a more specific embodiment, the synthesis window function is an MDCT synthesis window function, while the analysis window function is a DFT analysis window function. The fourth mechanism includes a mechanism for performing a fast transform on the MDCT signal and providing a first transformed signal in response thereto. The fourth mechanism further includes a mechanism for selectively delaying and updating the first transformed signal to yield second and third transformed signals, respectively, in response thereto. The fourth mechanism further includes a mechanism for operating on the first, second, and third transformed signals via third, second, and first combined window functions, respectively, and providing third, second, and first windowed signals, respectively, in response thereto. An adder adds the first, second, and third windowed functions to provide an added digital signal. An inverse DFT circuit performs an inverse DFT on the added digital signal to provide the DFT signal as output.

Provided are a scalable wide-band speech coding/decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.

The invention provides a compensation method for audio frame loss in a MDCT domain, the method comprising: when a frame currently lost is a P^th frame, obtaining a set of frequencies to be predicted, and for each frequency in the set, using phases and amplitudes of a plurality of frames before a (P−1)^th frame in a MDCT-MDST domain to predict a phase and an amplitude of the P^th frame, and using the predicted phase and amplitude to obtain a MDCT coefficient of the P^th frame at each corresponding frequency; for a frequency outside the set, using MDCT coefficients of a plurality of frames before the P^th frame to calculate a MDCT coefficient value of the P^th frame at the frequency; performing an IMDCT for the MDCT coefficients of the P^th frame to obtain a time domain signal of the P^th frame.

A method of and an apparatus to restore high frequency of a moving picture experts group audio layer 3 (MP3) audio signal within a decoder. The method includes: setting modified discrete cosine transform (MDCT) coefficients of low bands and high bands of an audio signal, based on scale factor information of each band; extracting MDCT coefficients of low bands per band based on scale factors of each band after dequantizing inputted compressed audio bitstream; selecting the MDCT coefficients of the set low bands that corresponds to patterns of MDCT coefficients of low bands of the inputted compressed audio bitstream, and selecting the MDCT coefficients of the high bands that matches with the MDCT coefficients of the selected low bands; and performing an inverse MDCT by adding the MDCT coefficients of the selected high bands with the MDCT coefficients of the low bands.

An audio encoding apparatus that allows a decoded signal exhibiting an excellent sound quality to be obtained on a decoding side. In the audio encoding apparatus (1000A), a time-frequency transform unit (1001) uses a time-frequency transform, such as a discrete Fourier transform (DFT) or a modified discrete cosine transform (MDCT), to transform a time domain signal (S(n)) to a frequency domain signal (spectrum factor) (S(f)). A psychoacoustic model analyzing unit (1002) performs a psychoacoustic model analysis of the frequency domain signal (S(f)), thereby obtaining a masking curve. An acoustic sense weighting unit (1003) estimates, based on the masking curve, an importance degree of acoustic sense, and determines and applies the weighting factors of respective spectrum factors to the respective spectrum factors. An encoding unit (1004) encodes the frequency domain signal (S(f)) as weighted in terms of the acoustic sense. A multiplexing unit (1005) multiplexes and transmits the encoded parameters.

Provided is a speech coding apparatus and method. A band divider divides an input signal into a high-band signal and a low-band signal, a narrowband encoder encodes the low-band signal using a Code Excited Linear Prediction (CELP)-based narrowband speech codec, a frequency characteristic collector converts the high-band signal to a signal in a frequency domain and obtains Modified Discrete Cosine Transform (MDCT) coefficients, a subband determiner determines subbands in a final stage based on the MDCT coefficients and determines subbands for quantization based on the subbands in a final stage, a gain quantizer performs gain quantization of the subbands, a bit assignment unit assigns bits to the subbands according to the magnitude of the gain quantization, and a shape quantizer performs shape quantization of the subbands in an algebraic method. Accordingly, algorithm consistency can be maintained and a complexity can be reduced by extending a bandwidth with a small number of bits in a speech codec.

Compressed data are decompressed using an inverse discrete cosine transform (IDCT). A first one directional (1D) IDCT is performed resulting in a plurality of first 1D IDCT coefficients followed by a second 1D IDCT resulting in a plurality of second 1D IDCT coefficients. In performing the first 1D IDCT and the second 1D IDCT a first plurality of intermediate butterfly computations are performed which include performing a plurality of intermediate multiplications resulting in a plurality of initial products and performing a plurality of intermediate additions resulting in intermediate product which are maintained at no more than 16-bits utilizing a round near positive (RNP) rounding scheme. Following the second 1D IDCT a rounding and shifting of the plurality of second 1D IDCT coefficients is performed utilizing a round away from zero (RAZ) rounding scheme resulting in a plurality of output coefficients which comply with the IEEE 1180 standard.

The present invention provides a method and a system with which information embedded in compressed digital audio data can be directly operated. An embodiment of the system for embedding additional information in compressed audio data includes: means for extracting MDCT (Modified Discrete Cosine Transform) coefficients from the compressed audio data; means for employing the MDCT coefficients to calculate a frequency component for the compressed audio data; means for embedding additional information in the frequency component obtained in a frequency domain; means for transforming into MDCT coefficients the frequency component in which the additional information is embedded; and means for using the MDCT coefficients, in which the additional information is embedded, to generate compressed audio data.

The present document relates to methods and systems for encoding an audio signal. The method comprises determining a spectral representation of the audio signal. The determining a spectral representation step may comprise determining modified discrete cosine transform, MDCT, coefficients, or a Quadrature Mirror Filter, QMF, filter bank representation of the audio signal. The method further comprises encoding the audio signal using the determined spectral representation; and classifying parts of the audio signal to be speech or non-speech based on the determined spectral representation. Finally, a loudness measure for the audio signal based on the speech parts is determined.

Provided is an apparatus and method for encoding a voice and audio signal by expanding a modified discrete cosine transform (MDCT) based CODEC to a wideband and a super-wideband in a communication system. The apparatus for encoding a signal in a communication system, includes a converter configured to convert a time domain signal corresponding to a service to be provided to users to a frequency domain signal, a quantization and normalization unit configured to calculate and quantize gain of each subband in the converted frequency domain signal and normalize a frequency coefficient of the each subband, a search unit configured to search patch information of each subband in the converted frequency domain signal using the normalized frequency coefficient, and a packetizer configured to packetize the quantized gain and the searched patch information and encode gain information of each subband in the frequency domain signal.

The invention discloses an audio watermarking method based on the MP3 encoding principle, which belongs to the technical field of multimedia digital watermarks. The method includes two processes, i.e. watermark embedment and watermark extraction; a watermark is embedded into a low-frequency MDCT (Modified Discrete Cosine Transform) coefficient as an encoding process is carried out synchronously; in order to enhance the robustness of the watermark, an appropriate audio segment is chosen in combination with the frequency domain masking effect in a human auditory system to be embedded; and in order to resist desynchronization attacks, a synchronization mechanism is introduced. By processing the encoding process, the method fulfills the injection of the watermark, and the method can effectively resist common audio attacks, and also has good robustness on desynchronization attacks (such as shearing attacks, time scaling and the like). While taking audio watermark robustness into consideration, the method guarantees the auditory invisibility of audio contents, the computational complexity of the algorithm is low, and the method is easy to implement.

The invention discloses a three-dimensional discrete cosine transform (DCT)-based geometric attack resistant volume data watermark realization method, which belongs to the field of multimedia signal processing. The method comprises the steps of watermark embedding and watermark extraction. The watermark embedding step further comprises the following steps of: (1) performing global three-dimensional DCT on original volume data, and extracting a geometric attack resistant characteristic vector from a transform coefficient; and (2) obtaining a binary logic sequence through a Hash function by utilizing the characteristic vector and watermarks to be embedded, and storing the binary sequence to a third party. The watermark extraction step further comprises the following steps of: (3) performing the global three-dimensional DCT on tested volume data, and extracting the geometric attack resistant characteristic vector from the object; and (4) extracting the watermarks by utilizing the characteristics of the Hash function and the binary logic sequence stored in the third party. The method relates to a three-dimensional DCT-based volume data digital watermarking technology, is proved by experiments to have relatively higher geometric and conventional attach resistance, also relates to a zero-watermark technology, and avoids the watermark embedding changing the contents of the volume data.

The invention provides an encoding method and a decoding method for lossless audio compression. The encoding method includes following steps: S1, adopting a self-adaption framing technology to frame original PCM (pulse-code modulation) audio signals to obtain framing information and framing signals, and bringing the framing information into side information; S2, performing modified discrete cosine transforming to each frame signal so as to obtain residual frequency spectrums and transformation signals; S3, performing audio track decorrelation to each frame transformation signal; S4, utilizing linear prediction parameters to conduct linear prediction to each frame after the audio track decorrelation so as to obtain residual error signals; S5, performing entropy encoding to the residual error signals and the residual frequency spectrums together so as to obtain a codebook of entropy encoding, and converting the codebook into a compressed bit stream for outputting; and S6, bringing the linear prediction parameters used in S4 linear prediction and the codebook of the entropy encoding obtained in S5 into the side information in S1.