摘要:
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.
摘要:
Exemplary embodiments may provide a method of encoding an audio signal. The method includes: segmenting the audio signal into a plurality of frames, wherein each of the frames includes M samples and M is a natural number greater than one; applying a first window, a second window, and at least one third window to the frames, wherein a length of the second window is longer than a length of the first window, and a length of the third window is longer than the length of the first window and shorter than the length of the second window; time-frequency transforming the frames to which the first window, the second window, and the at least one third window have been applied; and generating a bitstream including the time-frequency transformed frames.
摘要:
An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.
摘要:
Methods and systems for echo modulation are described. In one embodiment, intensities of a plurality of values in multiple windows of an audio signal may be obtained. The windows may be subject to a periodic boundary condition. A plurality of echo values may be calculated for each of the respective windows. The audio signal may be altered in one or more of the windows using a windowing function and echo values. Additional methods and systems are disclosed.
摘要:
A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.
摘要:
An apparatus for decoding an encoded audio signal including bandwidth extension control data indicating either a first harmonic bandwidth extension mode or a second non-harmonic bandwidth extension mode, includes: an input interface for receiving the encoded audio signal including the bandwidth extension control data indicating either the first harmonic bandwidth extension mode or the second non-harmonic bandwidth extension mode; a processor for decoding the audio signal using the second non-harmonic bandwidth extension mode; and a controller for controlling the processor to decode the audio signal using the second non-harmonic bandwidth extension mode, even when the bandwidth extension control data indicates the first harmonic bandwidth extension mode for the encoded signal.
摘要:
A system and method of adjusting digital audio sampling used with wideband audio includes: performing audio sampling on an analog audio signal at an initial sampling rate and an initial bit rate over a wideband audio frequency range; generating a digital audio signal based on the audio sampling; detecting a qualitative error rate between the analog audio signal and the digital audio signal; and decreasing the initial sampling rate, the initial bit rate, or both for sampling subsequent analog audio when the qualitative error is below a threshold.
摘要:
An apparatus for generating a bandwidth extended audio signal from an input signal, includes a patch generator for generating one or more patch signals from the input signal, wherein the patch generator is configured for performing a time stretching of subband signals from an analysis filterbank, and wherein the patch generator further includes a phase adjuster for adjusting phases of the subband signals using a filterbank-channel dependent phase correction.
摘要:
Provided are systems, methods and techniques for processing frame-based data. A frame of data, an indication that a transient occurs within the frame, and a location of the transient within the frame are obtained. Based on the indication of the transient, a block size is set for the frame, thereby effectively defining a plurality of equal-sized blocks within the frame. In addition, different window functions are selected for different ones of the plurality of equal-sized blocks based on the location of the transient, and the frame of data is processed by applying the selected window functions.
摘要:
The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank (501) comprising an analysis transformation unit (601) having a frequency resolution of Δf; and an analysis window (611) having a duration of DA; the analysis filter bank (501) being configured to provide a set of analysis subband signals from the low frequency component of the signal; a nonlinear processing unit (502, 650) configured to determine a set of synthesis subband signals based on a portion of the set of analysis subband signals, wherein the portion of the set of analysis subband signals is phase shifted by a transposition order T; and a synthesis filter bank (504) comprising a synthesis transformation unit (602) having a frequency resolution of QΔf; and a synthesis window (612) having a duration of DS; the synthesis filter bank (504) being configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein Q is a frequency resolution factor with Q≧1 and smaller than the transposition order T; and wherein the value of the product of the frequency resolution Δf and the duration DA of the analysis filter bank is selected based on the frequency resolution factor Q.