BACKGROUND OF THE INVENTION

The present application is concerned with a downscaled decoding concept.

The MPEG-4 Enhanced Low Delay AAC (AAC-ELD) usually operates at sample rates up to 48 kHz, which results in an algorithmic delay of 15 ms. For some applications, e.g. lip-sync transmission of audio, an even lower delay is desirable. AAC-ELD already provides such an option by operating at higher sample rates, e.g. 96 kHz, and therefore provides operation modes with even lower delay, e.g. 7.5 ms. However, this operation mode comes along with an unnecessary high complexity due to the high sample rate.

The solution to this problem is to apply a downscaled version of the filter bank and therefore, to render the audio signal at a lower sample rate, e.g. 48 kHz instead of 96 kHz. The downscaling operation is already part of AAC-ELD as it is inherited from the MPEG-4 AAC-LD codec, which serves as a basis for AAC-ELD.

The question which remains, however, is how to find the downscaled version of a specific filter bank. That is, the only uncertainty is the way the window coefficients are derived whilst enabling clear conformance testing of the downscaled operation modes of the AAC-ELD decoder.

In the following the principles of the down-scaled operation mode of the AAC-(E)LD codecs are described.

The downscaled operation mode or AAC-LD is described for AAC-LD in ISO/IEC 14496-3:2009 in section 4.6.17.2.7 “Adaptation to systems using lower sampling rates” as follows:

“In certain applications it may be necessary to integrate the low delay decoder into an audio system running at lower sampling rates (e.g. 16 kHz) while the nominal sampling rate of the bitstream payload is much higher (e.g. 48 kHz, corresponding to an algorithmic codec delay of approx. 20 ms). In such cases, it is favorable to decode the output of the low delay codec directly at the target sampling rate rather than using an additional sampling rate conversion operation after decoding.

This can be approximated by appropriate downscaling of both, the frame size and the sampling rate, by some integer factor (e.g. 2, 3), resulting in the same time/frequency resolution of the codec. For example, the codec output can be generated at 16 kHz sampling rate instead of the nominal 48 kHz by retaining only the lowest third (i.e. 480/3=160) of the spectral coefficients prior to the synthesis filterbank and reducing the inverse transform size to one third (i.e. window size 960/3=320).

As a consequence, decoding for lower sampling rates reduces both memory and computational requirements, but may not produce exactly the same output as a full-bandwidth decoding, followed by band limiting and sample rate conversion.

Please note that decoding at a lower sampling rate, as described above, does not affect the interpretation of levels, which refers to the nominal sampling rate of the AAC low delay bitstream payload.”

Please note that AAC-LD works with a standard MDCT framework and two window shapes, i.e. sine-window and low-overlap-window. Both windows are fully described by formulas and therefore, window coefficients for any transformation lengths can be determined.

Compared to AAC-LD, the AAC-ELD codec shows two major differences:The Low Delay MDCT window (LD-MDCT)The possibility of utilizing the Low Delay SBR tool

The IMDCT algorithm using the low delay MDCT window is described in 4.6.20.2 in [1], which is very similar to the standard IMDCT version using e.g. the sine window. The coefficients of the low delay MDCT windows (480 and 512 samples frame size) are given in Table 4.A.15 and 4.A.16 in [1]. Please note that the coefficients cannot be determined by a formula, as the coefficients are the result of an optimization algorithm.FIG. 9shows a plot of the window shape for frame size 512.

In case the low delay SBR (LD-SBR) tool is used in conjunction with the AAC-ELD coder, the filter banks of the LD-SBR module are downscaled as well. This ensures that the SBR module operates with the same frequency resolution and, therefore, no more adaptions are implemented.

Thus, the above description reveals that there is a need for downscaling decoding operations such as, for example, downscaling a decoding at an AAC-ELD. It would be feasible to find out the coefficients for the downscaled synthesis window function anew, but this is a cumbersome task, necessitates additional storage for storing the downscaled version and renders a conformity check between the non-downscaled decoding and the downscaled decoding more complicated or, from another perspective, does not comply with the manner of downscaling requested in the AAC-ELD, for example. Depending on the downscale ratio, i.e. the ratio between the original sampling rate and the downscaled sampling rate, one could derive the downscaled synthesis window function simply by downsampling, i.e. picking out every second, third, . . . window coefficient of the original synthesis window function, but this procedure does not result in a sufficient conformity of the non-downscaled decoding and downscaled decoding, respectively. Using more sophisticated decimating procedures applied to the synthesis window function, lead to unacceptable deviations from the original synthesis window function shape. Therefore, there is a need in the art for an improved downscaled decoding concept.

SUMMARY

According to an embodiment, an audio decoder configured to decode an audio signal at a first sampling rate from a data stream into which the audio signal is transform coded at a second sampling rate, the first sampling rate being 1/Fthof the second sampling rate, may have: a receiver configured to receive, per frame of length N of the audio signal, N spectral coefficients; a grabber configured to grab-out for each frame, a low-frequency fraction of length N/F out of the N spectral coefficients; a spectral-to-time modulator configured to subject, for each frame, the low-frequency fraction to an inverse transform having modulation functions of length (E+2)·N/F temporally extending over the respective frame and E+1 previous frames so as to obtain a temporal portion of length (E+2)·N/F; a windower configured to window, for each frame, the temporal portion using a synthesis window of length (E+2)·N/F having a zero-portion of length 1/4·N/F at a leading end thereof and having a peak within a temporal interval of the synthesis window, the temporal interval succeeding the zero-portion and having length 7/4·N/F so that the windower obtains a windowed temporal portion of length (E+2)·N/F; and a time domain aliasing canceler configured to subject the windowed temporal portion of the frames to an overlap-add process so that a trailing-end fraction of length (E+1)/(E+2) of the windowed temporal portion of a current frame overlaps a leading end of length (E+1)/(E+2) of the windowed temporal portion of a preceding frame, wherein the inverse transform is an inverse MDCT or inverse MDST, and wherein the synthesis window is a downsampled version of a reference synthesis window of length (E+2)·N, downsampled by a factor of F by a segmental interpolation in segments of length 1/4·N.

Another embodiment may have an audio decoder for generating a downscaled version of a synthesis window of the above inventive audio decoder, wherein E=2 so that the synthesis window function has a kernel related half of length 2·N/F preceded by a reminder half of length 2·N/F and wherein the spectral-to-time modulator, the windower and the time domain aliasing canceler are implemented so as to cooperate in a lifting implementation according to which the spectral-to-time modulator confines the subjecting, for each frame, the low-frequency fraction to the inverse transform having modulation functions of length (E+2)·N/F temporally extending over the respective frame and E+1 previous frames, to a transform kernel coinciding with the respective frame and one previous frame so as to obtain the temporal portion xk,nwith n=0 . . . 2M−1 with M=N/F being a sample index and k being a frame index; the windower windowing, for each frame, the temporal portion xk,naccording to zk,n=ωn·xk,nfor n=0, . . . , 2M−1 so as to obtain the windowed temporal portion zk,nwith n=0 . . . 2M−1; the time domain aliasing canceler generates intermediate temporal portions mk(0), . . . mk(M−1) according to mk,n=zk,n+zk-1,n+Mfor n=0, . . . , M−1, and the audio decoder has a lifter configured to obtain the frames uk,nwith n=0 . . . M−1 according to uk,n=mk,n+ln−M/2·mk-1,M-1-nfor n=M/2, . . . , M−1, and uk,n=mk,n+lM-1-n·outk-1,M-1-nfor n=0, . . . , M/2−1, wherein lnwith n=0 . . . M−1 are lifting coefficients, and wherein lnwith n=0 . . . M−1 and ωnwith n=0, . . . , 2M−1 depend on coefficients wnwith n=0 . . . (E+2)M−1 of the synthesis window.

According to another embodiment, an audio decoder configured to decode an audio signal at a first sampling rate from a data stream into which the audio signal is transform coded at a second sampling rate, the first sampling rate being 1/Fthof the second sampling rate, may have: a receiver configured to receive, per frame of length N of the audio signal, N spectral coefficients; a grabber configured to grab-out for each frame, a low-frequency fraction of length N/F out of the N spectral coefficients; a spectral-to-time modulator configured to subject, for each frame, the low-frequency fraction to an inverse transform having modulation functions of length 2·N/F temporally extending over the respective frame and a previous frame so as to obtain a temporal portion of length 2·N/F; a windower configured to window, for each frame, the temporal portion xk,naccording to zk,n=ωn·xk,nfor n=0, . . . , 2M−1 so as to obtain a windowed temporal portion zk,nwith n=0 . . . 2M−1; a time domain aliasing canceler configured to generate intermediate temporal portions mk(0), . . . mk(M−1) according to mk,n=zk,n+zk-1,n+Mfor n=0, . . . , M−1, and the lifter configured to obtain frames uk,nof the audio signal with n=0 . . . M−1 according to uk,n=mk,n+ln−M/2·mk-1,M-1-nfor n=M/2, . . . , M−1, and uk,n=mk,n+lM-1-n·outk-1,M-1-nfor n=0, . . . , M/2−1, wherein lnwith n=0 . . . M−1 are lifting coefficients, wherein the inverse transform is an inverse MDCT or inverse MDST, and wherein lnwith n=0 . . . M−1 and ωnwith n=0, . . . , 2M−1 depend on coefficients wnwith n=0 . . . (E+2)M−1 of a synthesis window, and the synthesis window is a downsampled version of a reference synthesis window of length 4·N, downsampled by a factor of F by a segmental interpolation in segments of length 1/4·N.

Another embodiment may have an apparatus for generating a downscaled version of a synthesis window of one of the above inventive audio decoders, wherein the apparatus is configured to downsample a reference synthesis window of length (E+2)·N by a factor of F by a segmental interpolation in 4·(E+2) segments of equal length.

Still another embodiment may have a method for generating a downscaled version of a synthesis window of one of the above inventive audio decoders, wherein the method has downsampling a reference synthesis window of length (E+2)·N by a factor of F by a segmental interpolation in 4·(E+2) segments of equal length.

According to another embodiment, a method for decoding an audio signal at a first sampling rate from a data stream into which the audio signal is transform coded at a second sampling rate, the first sampling rate being 1/Fthof the second sampling rate, may have the steps of: receiving, per frame of length N of the audio signal, N spectral coefficients; grabbing-out for each frame, a low-frequency fraction of length N/F out of the N spectral coefficients; performing a spectral-to-time modulation by subjecting, for each frame, the low-frequency fraction to an inverse transform having modulation functions of length (E+2)·N/F temporally extending over the respective frame and E+1 previous frames so as to obtain a temporal portion of length (E+2)·N/F; windowing, for each frame, the temporal portion using a synthesis window of length (E+2)·N/F having a zero-portion of length 1/4·N/F at a leading end thereof and having a peak within a temporal interval of the synthesis window, the temporal interval succeeding the zero-portion and having length 7/4·N/F so that the windower obtains a windowed temporal portion of length (E+2)·N/F; and performing a time domain aliasing cancellation by subjecting the windowed temporal portion of the frames to an overlap-add process so that a trailing-end fraction of length (E+1)/(E+2) of the windowed temporal portion of a current frame overlaps a leading end of length (E+1)/(E+2) of the windowed temporal portion of a preceding frame, wherein the inverse transform is an inverse MDCT or inverse MDST, and wherein the synthesis window is a downsampled version of a reference synthesis window of length (E+2)·N, downsampled by a factor of F by a segmental interpolation in segments of length 1/4·N.

Another embodiment may have a non-transitory digital storage medium having stored thereon a computer program for performing the above inventive methods, when said computer program is run by a computer.

The present invention is based on the finding that a downscaled version of an audio decoding procedure may more effectively and/or at improved compliance maintenance be achieved if the synthesis window used for downscaled audio decoding is a downsampled version of a reference synthesis window involved in the non-downscaled audio decoding procedure by downsampling by the downsampling factor by which the downsampled sampling rate and the original sampling rate deviate, and downsampled using a segmental interpolation in segments of 1/4 of the frame length.

