Abstract:
A method for processing an audio signal, including: sound is converted to an analog audio input signal and converted into a digital audio signal; a windowed time domain signal is obtained and then a twiddled signal is obtained; the twiddled signal is pre-rotated and then an FFT is performed; an in-place fixed rotate compensation is performed on the FFT signal and then an post-rotated is performed; a quantized signal is obtained and then wrote into a bitstream for transmitting or storing.
Abstract:
A data processing method is disclosed, including: twiddling input data, so as to obtain twiddled data; pre-rotating the twiddled data by using a symmetric rotate factor, where the rotate factor is a·W4L2p+1, p=0, . . . , L/2−1, and a is a constant; performing a Fast Fourier (Fast Fourier Transform, FFT) transform of L/2 point on the pre-rotated data, where L is the length of the input data; post-rotating the data that has undergone the FFT transform by using a symmetric rotate factor, where the rotate factor is b·W4L2q+1, q=0, . . . , L/2−1, and b is a constant; and obtaining output data.
Abstract:
A method and a device for encoding a high frequency signal, and a method and a device for decoding a high frequency signal are provided, which relate to encoding and decoding technology. The method for encoding a high frequency signal includes: determining a signal type of a high frequency signal of a current frame; smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded, if the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal; and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.
Abstract:
Embodiments of the present invention provide a signal classification method and device, and encoding and decoding methods and devices. The encoding method includes dividing a current frame into a low-frequency band signal and a high-frequency band signal, attenuating the high-frequency band signal or a to-be-encoded characteristic parameter of the high-frequency band signal according to an energy attenuation value of the low-frequency band signal, where the energy attenuation value indicates energy attenuation of the low-frequency band signal caused by encoding of the low-frequency band signal, and encoding the attenuated high-frequency band signal or the attenuated to-be-encoded characteristic parameter of the high-frequency band signal. The technical solutions according to the embodiments of the present invention can improve the effect of combining the low-frequency band signal and the high-frequency band signal at the decoder.
Abstract:
An embodiment of the present invention discloses a data processing method, including: twiddling input data, so as to obtain twiddled data; pre-rotating the twiddled data by using a symmetric rotate factor, where the rotate factor is a·W4L2p+1, p=0, . . . , L/2−1, and a is a constant; performing a Fast Fourier (Fast Fourier Transform, FFT) transform of L/2 point on the pre-rotated data, where L is the length of the input data; post-rotating the data that has undergone the FFT transform by using a symmetric rotate factor, where the rotate factor is b·W4L2q+1, q=0, . . . , L/2−1, and b is a constant; and obtaining output data.
Abstract:
A voice or audio signal processor for processing received network packets received over a communication network to provide an output signal, the voice or audio signal processor comprising a jitter buffer being configured to buffer the received network packets, a voice or audio decoder being configured to decode the received network packets as buffered by the jitter buffer to obtain a decoded voice or audio signal, a controllable time scaler being configured to amend a length of the decoded voice or audio signal to obtain a time scaled voice or audio signal as the output voice or audio signal, and an adaptation control means being configured to control an operation of the time scaler in dependency on a processing complexity measure.
Abstract:
The present disclosure relates to a signal analyzer for processing an overlapped input signal frame comprising 2N subsequent input signal values. The signal analyzer comprises: a windower adapted to window the overlapped input signal frame to obtain a windowed signal, wherein the windower is adapted to zero M+N/2 subsequent input signal values of the overlapped input signal frame, wherein M is equal or greater than 1 and smaller than N/2; and a transformer adapted to transform the remaining 3N/2−M subsequent windowed signal values of the windowed signal using N−M sets of transform parameters to obtain a transformed-domain signal comprising N−M transformed-domain signal values.
Abstract:
A method for encoding a high frequency signal includes determining a signal type of a high frequency signal of a current frame, smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded when the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal, and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.
Abstract:
A method and apparatus for providing signal processing coefficients for processing an input signal at a predetermined signal processing sampling rate, wherein the input signal is received at an input signal sampling rate, the method comprising the steps of computing a correlation or covariance function based on the received input signal at the input signal sampling rate to provide correlation or covariance coefficients at the input signal sampling rate, re-sampling the computed correlation or covariance coefficients having the input signal sampling rate to provide correlation or covariance coefficients at the predetermined signal processing sampling rate, and calculating the signal processing coefficients based on the correlation or covariance coefficients at the predetermined signal processing sampling rate.
Abstract:
A voice quality enhancement (VQE) detector for a network element receiving an audio signal from a previous network element of a network, wherein said voice quality enhancement detector is adapted: to perform a voice quality enhancement detection based on the received audio signal, wherein said voice quality enhancement detection comprises detecting that at least one voice quality enhancement function, VQEE comprising of a noise cancellation or an echo cancellation function using a Gaussian Mixture Model was applied to the received audio signal by at least one previous network element of the network; and to control a voice quality enhancement processing of the received audio signal depending on the detection result.