摘要:
Apparatus for decoding an encoded audio signal comprising an encoded core signal (1), comprising: a core decoder (1400) for decoding the encoded core signal (1401) to obtain a decoded core signal; a tile generator (1404) for generating one or more spectral tiles having frequencies not included in the decoded core signal using a spectral portion of the decoded core signal; and a cross-over filter (1406) for spectrally cross-over filtering the decoded core signal and a first frequency tile having frequencies extending from a gap filling frequency (309) to an upper border frequency or for spectrally cross-over filtering a first frequency tile and a second frequency tile.
摘要:
Apparatus for decoding an encoded audio signal comprising an encoded core signal (1), comprising: a core decoder (1400) for decoding the encoded core signal (1401) to obtain a decoded core signal; a tile generator (1404) for generating one or more spectral tiles having frequencies not included in the decoded core signal using a spectral portion of the decoded core signal; and a cross-over filter (1406) for spectrally cross-over filtering the decoded core signal and a first frequency tile having frequencies extending from a gap filling frequency (309) to an upper border frequency or for spectrally cross-over filtering a first frequency tile and a second frequency tile.
摘要:
An apparatus for generating an encoded signal, comprises: a window sequence controller (808) for generating a window sequence information (809) for windowing an audio or image signal, the window sequence information indicating a first window (1500) for generating a first frame of spectral values, a second window function (1502) and at least one third window function (1503) for generating a second frame of spectral values, wherein the first window function (1500), the second window function (1502) and the one or more third window functions overlap within a multi-overlap region (1300); a preprocessor (802) for windowing (902) a second block of samples corresponding to the second window function and the at least one third window functions using an auxiliary window function (1 100) to obtain a second block of windowed samples, and for preprocessing (904) the second block of windowed samples using a folding-in operation of a portion of the second block overlapping with a first block into the multi-overlap portion (1300) to obtain a preprocessed second block of windowed samples having a modified multi-overlap portion; a spectrum converter (804) for applying an aliasing-introducing transform (906) to the first block of samples using the first window function to obtain the first frame of spectral values, for applying the aliasing introducing transform to a first portion of the preprocessed second block of windowed samples using the second window function to obtain a first portion of spectral samples of a second frame and for applying the aliasing introducing transform to a second portion of the preprocessed second block of windowed samples using the one or more third window functions (1503) to obtain a second portion of spectral samples of the second frame; and a processor (806) for processing the first frame and the second frame to obtain encoded frames of the audio or image signal.
摘要:
An apparatus for generating a frequency enhancement signal (130), comprises: a signal generator (200) for generating an enhancement signal from a core signal (120), the enhancement signal comprising an enhancement frequency range not included in the core signal, wherein a time portion of the enhancement signal comprises subband signals for a plurality of subbands; a synthesis filterbank (300) for generating the frequency enhanced signal (140) using the enhancement signal (130), wherein the signal generator (200) is configured for performing an energy limitation in order to make sure that the frequency enhanced signal (140) obtained by the synthesis filterbank (300) is so that an energy of a higher band is, at the most, equal to an energy in a lower band or is greater than an energy of a higher band, at the most, by a predefined threshold.
摘要:
The invention provides an audio encoder for encoding an audio signal (AS) so as to produce therefrom an encoded signal (ES), the audio encoder (1) comprising: a framing device (2) configured to extract frames (F) from the audio signal (AS); a quantizer (3) configured to map spectral lines (SL 1-32 ) of a spectrum signal (SPS) derived from the frame (F) of the audio signal (AS) to quantization indices (I 0 , I 1 ), wherein the quantizer (3) has a dead-zone (DZ), in which the input spectral lines (SL) are mapped to quantization index zero (I 0 ); and a control device (4) configured to modify the dead-zone (DZ); wherein the control device (4) comprises a tonality calculating device (5) configured to calculate at least one tonality indicating value (TI 5.32 ) for at least one spectrum line (SL 1-32 ) or for at least one group of spectral lines (SL 1-32 ), wherein the control device (4) is configured to modify the dead-zone (DZ) for the at least one spectrum line (SL 1-32 ) or the at least one group of spectrum lines (SL 1-32 ) depending on the respective tonality indicating value (TI 5-32 ).
摘要:
An apparatus for encoding a speech signal by determining a codebook vector of a speech coding algorithm is provided. The apparatus includes a matrix determiner for determining an autocorrelation matrix R, and a codebook vector determiner for determining the codebook vector depending on the autocorrelation matrix R. The matrix determiner is configured to determine the autocorrelation matrix R by determining vector coefficients of a vector r, wherein the autocorrelation matrix R includes a plurality of rows and a plurality of columns, wherein the vector r indicates one of the columns or one of the rows of the autocorrelation matrix R, wherein R(i, j)=r(|i−j|), wherein R(i, j) indicates the coefficients of the autocorrelation matrix R, wherein i is a first index indicating one of a plurality of rows of the autocorrelation matrix R, and wherein j is a second index indicating one of the plurality of columns of the autocorrelation matrix R.
摘要:
A signal processor for providing a processed version of an input signal in dependence on the input signal comprises a windower configured to window a portion of the input signal, or of a pre-processed version thereof, in dependence on a signal processing window described by signal processing window values for a plurality of window value index values, in order to obtain the processed version of the input signal. The signal processor also comprises a window provider for providing the signal processing window values for a plurality of window value index values in dependence on one or more window shape parameters.
摘要:
An audio encoder and an audio decoder are based on a combination of two audio channels (201, 202) to obtain a first combination signal (204) as a mid signal and a residual signal (205) which can be derived using a predicted side signal derived from the mid signal. The first combination signal and the prediction residual signal are encoded (209) and written (212) into a data stream (213) together with the prediction information (206) derived by an optimizer (207) based on an optimization target (208). A decoder uses the prediction residual signal, the first combination signal and the prediction information to derive a decoded first channel signal and a decoded second channel signal. In an encoder example or in a decoder example, a real-to-imaginary transform can be applied for estimating the imaginary part of the spectrum of the first combination signal. For calculating the prediction signal used in the derivation of the prediction residual signal, the real-valued first combination signal is multiplied by a real portion of the complex prediction information and the estimated imaginary part of the first combination signal is multiplied by an imaginary portion of the complex prediction information.
摘要:
A frequency-domain audio codec is provided with the ability to additionally support a certain transform length in a backward-compatible manner, by the following: the frequency-domain coefficients of a respective frame are transmitted in an interleaved manner irrespective of the signalization signaling for the frames as to which transform length actually applies, and additionally the frequency-domain coefficient extraction and the scale factor extraction operate independent from the signalization. By this measure, old-fashioned frequency-domain audio coders/decoders, insensitive for the signalization, would be able to nevertheless operate without faults and with reproducing a reasonable quality. Concurrently, frequency-domain audio coders/decoders able to support the additional transform length would offer even better quality despite the backward compatibility. As far as coding efficiency penalties due to the coding of the frequency domain coefficients in a manner transparent for older decoders are concerned, same are of comparatively minor nature due to the interleaving.