Embodiments relate to an MDCT-based multi-signal encoding and decoding system with signal-adaptive joint channel processing, wherein the signal can be a channel, and the multisignal is a multichannel signal or, alternatively an audio signal being a component of a sound field description such as an Ambisonics component, i.e., W, X, Y, Z in first order Ambisonics or any other component in a higher order Ambisonics description. The signal can also be a signal of an A-format or B-format or any other format description of a sound field.
In the 3D Audio context, loudspeaker channels are distributed in several height layers, resulting in horizontal and vertical channel pairs. Joint coding of only two channels as defined in USAC is not sufficient to consider the spatial and perceptual relations between channels. MPEG Surround is applied in an additional pre-/postprocessing step, residual signals are transmitted individually without the possibility of joint stereo coding, e.g. to exploit dependencies between left and right vertical residual signals. In AC-4 dedicated N-channel elements are introduced that allow for efficient encoding of joint coding parameters, but fail for generic speaker setups with more channels as proposed for new immersive playback scenarios (7.1+4, 22.2). MPEG-H Quad Channel element is also restricted to only 4 channels and cannot be dynamically applied to arbitrary channels but only a pre-configured and fixed number of channels. MCT introduces the flexibility of signal-adaptive joint channel coding of arbitrary channels, but stereo processing is conducted on windowed and transformed non-normalized (non whitened) signals. Furthermore, coding of the prediction coefficients or angles in each band for each stereo box needs a significant number of bits.
According to an embodiment, a multisignal encoder for encoding at least three audio signals may have: a signal preprocessor for individually preprocessing each audio signal to acquire at least three preprocessed audio signals, wherein the preprocessing is performed so that a pre-processed audio signal is whitened with respect to the signal before preprocessing; an adaptive joint signal processor for performing a processing of the at least three preprocessed audio signals to acquire at least three jointly processed signals or at least two jointly processed signals and an unprocessed signal; a signal encoder for encoding each signal to acquire one or more encoded signals; and an output interface for transmitting or storing an encoded multi-signal audio signal including the one or more encoded signals, side information relating to the preprocessing and side information relating to the processing.
According to another embodiment, a multisignal decoder for decoding an encoded signal may have: a signal decoder for decoding at least three encoded signals; a joint signal processor for performing a joint signal processing in accordance with side information included in the encoded signal to acquire at least three processed decoded signals; and a post processor for post processing the at least three processed decoded signals in accordance with side information included in the encoded signal, wherein the post processing is performed so that the post processed signals are less white than the signals before post processing, and wherein the post processed signals represent a decoded audio signal.
According to another embodiment, a method for performing multisignal encoding at least three audio signals may have the steps of: individually preprocessing each audio signal to acquire at least three preprocessed audio signals, wherein the preprocessing is performed so that a preprocessed audio signal is whitened with respect to the signal before preprocessing; performing a processing of the at least three preprocessed audio signals to acquire at least three jointly processed signals or at least two jointly processed signals and a signal to be encoded individually; encoding each signal to acquire one or more encoded signals; and transmitting or storing an encoded multisignal audio signal including the one or more encoded signals, side information relating to the preprocessing and side information relating to the processing.
According to another embodiment, a method for multisignal decoding an encoded signal may have the steps of: individually decoding at least three encoded signals; performing a joint signal processing in accordance with side information included in the encoded signal to acquire at least three processed decoded signals; and post processing the at least three processed decoded signals in accordance with side information included in the encoded signal, wherein the post processing is performed so that the post processed signals are less white than the signals before post processing, and wherein the post processed signals represent a decoded audio signal.
Another embodiment may have a non-transitory digital storage medium having a computer program stored thereon to perform the method for performing multisignal encoding at least three audio signals, the method having the steps of: individually preprocessing each audio signal to acquire at least three preprocessed audio signals, wherein the preprocessing is performed so that a preprocessed audio signal is whitened with respect to the signal before pre-processing; performing a processing of the at least three preprocessed audio signals to acquire at least three jointly processed signals or at least two jointly processed signals and a signal to be encoded individually; encoding each signal to acquire one or more encoded signals; and transmitting or storing an encoded multisignal audio signal including the one or more encoded signals, side information relating to the preprocessing and side information relating to the processing, when said computer program is run by a computer.
Another embodiment may have a non-transitory digital storage medium having a computer program stored thereon to perform the method for multisignal decoding an encoded signal, the method having the steps of: individually decoding at least three encoded signals; performing a joint signal processing in accordance with side information included in the encoded signal to acquire at least three processed decoded signals; and post processing the at least three processed decoded signals in accordance with side information included in the encoded signal, wherein the post processing is performed so that the post processed signals are less white than the signals before post processing, and wherein the post processed signals represent a decoded audio signal, when said computer program is run by a computer.
Another embodiment may have an encoded signal, having: at least three individually encoded signals; side information related to a preprocessing performed in order to acquire the three individually encoded signals; and side information related to a pairwise processing performed for acquiring the at least three individually encoded signals, and wherein the encoded signal includes, for each of the at least three encoded signals acquired by multisignal encoding, an energy scaling value or, for each one of the individually encoded signals, a bit distribution value.
The present invention is based on the finding that a multi-signal encoding efficiency is substantially enhanced by performing the adaptive joint signal processing not on the original signals but on preprocessed audio signals where this pre-processing is performed so that a pre-processed audio signal is whitened with respect to the signal before pre-processing. With respect to the decoder side, this means that a post processing is performed subsequent to the joint signal processing to obtain at least three processed decoded signals. These at least three processed decoded signals are post processed in accordance with side information included in the encoded signal, wherein the post processing is performed in such a way that the post processed signals are less white than the signals before post processing. The post processed signals finally represent, either directly or subsequent to further signal processing operations, the decoded audio signal, i.e., the decoded multi-signal.
Especially for immersive 3D audio formats, efficient multichannel coding exploiting the properties of a plurality of signals are obtained to reduce the amount of transmission data while pre-serving the overall perceptual audio quality. In an implementation, a signal adaptive joint coding within a multichannel system is performed using perceptually whitened and, additionally, inter-channel level difference (ILD) compensated spectra. A joint coding is performed advantageously using a simple per band M/S transform decision that is driven based on an estimated number of bits for an entropy coder.
A multi-signal encoder for encoding at least three audio signals comprises a signal preprocessor for individually preprocessing each audio signal to obtain at least three preprocessed audio signals, where the preprocessing is performed so that the preprocessed audio signal is whitened with respect to the signal before preprocessing. An adaptive joint signal processing of the at least three preprocessed audio signals is performed to obtain at least three jointly processed signals. This processing operates on whitened signals. The preprocessing results in the extraction of certain signal characteristics such as a spectral envelope or so that, if not extracted, would reduce the efficiency of the joint signal processing such as a joint stereo or a joint multichannel processing. Additionally, in order to enhance the joint signal processing efficiency, a broadband energy normalization of the at least three preprocessed audio signals is performed so that each preprocessed audio signal has a normalized energy. This broadband energy normalization is signaled into the encoded audio signal as side information so that this broadband energy normalization can be reversed on the decoder side subsequent to inverse joint stereo or joint multichannel signal processing. By means of this advantageous additional broadband energy normalization procedure, the adaptive joint signal processing efficiency is enhanced so that the number of bands or even the number of full frames that can be subjected to mid/side processing in contrast to left/right processing (dual mono processing) is substantially enhanced. The efficiency of the whole stereo encoding process is enhanced more and more the higher the number of bands or even full frames that are subjected to common stereo or multichannel processing such as mid/side processing becomes.
The lowest efficiency is obtained, from the stereo processing view, when the adaptive joint signal processor has to adaptively decide, for a band or for a frame that this band or frame is to be processed by “dual mono” or left/right processing. Here, the left channel and the right channel are processed as they are, but naturally in the whitened and energy normalized domain. When, however, the adaptive joint signal processor adaptively determines, for a certain band or frame that a mid/side processing is performed, the mid signal is calculated by adding the first and the second channel and the side signal is calculated by calculating the difference from the first and the second channel of the channel pair. Typically, the mid signal is, with respect to its value range, comparable to one of the first and the second channels, but the side signal will typically be a signal with a small energy that can be encoded with high efficiency or, even in the most advantageous situation, the side signal is zero or close to zero so that spectral regions of the side signal can even be quantized to zero and, therefore, be entropy encoded in a highly efficient way. This entropy encoding is performed by the signal encoder for encoding each signal to obtain one or more encoded signals and the output interface of the multi-signal encoder transmits or stores an encoded multi-signal audio signal comprising the one or more encoded signals, side information relating to the preprocessing and side information relating to the adaptive joint signal processing.
On the decoder-side, the signal decoder that typically comprises an entropy decoder decodes the at least three encoded signals typically relying on an advantageous included bit distribution information. This bit distribution information is included as side information in the encoded multi-signal audio signal and can, for example, be derived in the encoder-side by looking at the energy of the signals at the input into the signal (entropy) encoder. The output of the signal decoder within the multi-signal decoder is input into a joint signal processor for performing a joint signal processing in accordance with side information included in the encoded signal to obtain at least three processed decoded signals. This joint signal processor advantageously undoes the joint signal processing performed on the encoder-side and, typically, performs an inverse stereo or inverse multichannel processing. In the advantageous implementation, the joint signal processor applies a processing operation to calculate left/right signals from mid/side signals. When, however, the joint signal processor determines from the side information that, for a certain channel pair, a dual mono processing is already there, this situation is noted and used in the decoder for further processing.
The joint signal processor on the decoder-side can be, as the adaptive joint signal processor on the encoder-side, a processor operating in the mode of a cascaded channel-pair tree or a simplified tree. A simplified tree also represents some kind of cascaded processing, but the simplified tree is different from the cascaded channel pair tree in that the output of a processed pair cannot be an input into another to be processed pair.
It can be the case that, with respect to a first channel pair that is used by the joint signal processor on the multi-signal decoder side in order to start the joint signal processing, this first channel pair that was the last channel pair processed on the encoder side has, for a certain band, a side information indicating dual mono but, these dual mono signals can be used, later on in a channel pair processing as a mid signal or a side signal. This is signaled by the corresponding side information related to a pair-wise processing performed for obtaining the at least three individually encoded channels to be decoded on the decoder-side.
Embodiments relate to an MDCT-based multi-signal encoding and decoding system with signal-adaptive joint channel processing, wherein the signal can be a channel, and the multisignal is a multichannel signal or, alternatively an audio signal being a component of a sound field description such as an Ambisonics component, i.e., W, X, Y, Z in first order Ambisonics or any other component in a higher order Ambisonics description. The signal can also be a signal of an A-format or B-format or any other format description of a sound field.
Subsequently, further advantages of embodiments are indicated. The codec uses new concepts to merge the flexibility of signal adaptive joint coding of arbitrary channels as described in [6] by introducing the concepts described in [7] for joint stereo coding. These are:
a) Use of perceptually whitened signals for further coding (similar to the way they are used in a speech coder). This has several advantages:
b) Use of a ILD parameters of arbitrary channels to efficiently code panned sources
c) Flexible bit distribution among the processed channels based on the energy.
The codec furthermore uses Frequency Domain Noise Shaping (FDNS) to perceptually whiten the signal with the rate-loop as described in [8] combined with the spectral envelope warping as described in [9]. The codec further normalized the FDNS-whitened spectrum towards the mean energy level using ILD parameters. Channel pairs for joint coding are selected in an adaptive manner as described in [6], where the stereo coding consist of a band-wise M/S vs UR decision. The band-wise M/S decision is based on the estimated bitrate in each band when coded in the L/R and in the M/S mode as described in [7]. Bitrate distribution among the band-wise M/S processed channels is based on the energy.
Embodiments of the present invention will be detailed subsequently referring to the ap-pended drawings, in which:
In an implementation, the signal encoder 300 comprises a rate loop processor that is controlled by bit distribution information 536 that is generated by the adaptive joint signal processor 200 and that is not only forwarded from block 200 to block 300 but that is also forwarded, within the side information 530, to the output interface 400 and, therefore, into the encoded multi-signal audio signal. The encoded multi-signal audio signal 500 is typically generated in a frame-by-frame way where the framing and, typically, a corresponding windowing and time-frequency conversion is performed within the signal preprocessor 100.
An exemplary illustration of a frame of the encoded multi-signal audio signal 500 is illustrated in
As will be illustrated later on, the preprocessing comprises a temporal noise shaping processing and/or a frequency domain noise shaping processing or LTP (long term prediction) processing or windowing processing operations. The corresponding preprocessing side information 550 may comprise at least one of the temporal noise shaping (TNS) information, frequency domain noise shaping (FDNS) information, long term prediction (LTP) information or windowing or window information.
Temporal noise shaping comprises a prediction of a spectral frame over frequency. A spectral value with a higher frequency is predicted using a weighted combination of spectral values having lower frequencies. The TNS side information comprises the weights of the weighted combination that are also known as LPC coefficients derived by the prediction over frequency. The whitened spectral values are the prediction residual values, i.e., the differences, per spectral value, between the original spectral value and the predicted spectral value. On the decoder side, an inverse prediction of an LPC synthesis filtering is performed in order to undo the TNS processing on the encoder side.
FDNS processing comprises weighting spectral values of a frame using weighting factors for the corresponding spectral values, where the weighting values are derived from the LPC coefficients calculated from a block/frame of the windowed time domain signal. The FDNS side information comprises a representation of the LPC coefficients derived from the time domain signal.
Another whitening procedure also useful for the present invention is a spectral equalization using scale factors so that the equalized spectrum represents a version being whiter than a non-equalized version. The side information would be the scale factors used for weighting and the inverse procedure comprises undoing the equalization on the decoder side using the transmitted scale factors.
Another whitening procedure comprises performing an inverse filtering of the spectrum using an inverse filter controlled by the LPC coefficients derived from the time domain frame as known in the art of speech coding. The side information is the inverse filter information and this inverse filtering is undone in the decoder using the transmitted side information.
Another whitening procedure comprises performing an LPC analysis in the time domain and calculating time domain residual values that are then converted into the spectral range. Typically, the thus obtained spectral values are similar to the spectral values obtained by FDNS. On the decoder side, the postprocessing comprises performing the LPC synthesis using the transmitted LPC coefficients representation.
The joint processing side information 530 comprises, in an implementation, a pair-wise processing side information 532, an energy scaling information 534 and a bit distribution information 536. The pairwise processing side information may comprise at least one of the channel pair side information bits, a full mid/side or dual mono or band-wise mid/side information and, in case of a band-wise mid/side indication, a mid/side mask indicating, for each bandwidth in a frame, whether the band is processed by mid/side or L/R processing. The pairwise processing side information may additionally comprise intelligent gap filling (IGF) or other bandwidth extension information such as SBR (spectral band replication) information or so.
The energy scaling information 534 may comprise, for each whitened, i.e., preprocessed signal 180, an energy scaling value and a flag, indicating, whether the energy scaling is an upscaling or a downscaling. In case of eight channels, for example, block 534 would comprise eight scaling values such as eight quantized ILD values and eight flags indicating, for each of the eight channels, whether an upscaling or downscaling has been done within the encoder or has to be done within the decoder. An upscaling in the encoder is necessary, when the actually energy of a certain preprocessed channel within a frame is below the mean energy for the frame among all channels, and a downscaling is necessary, when the actual energy of a certain channel within the frame is above the mean energy over all channels within the frame. The joint processing side information may comprise a bit distribution information for each of the jointly processed signals or for each jointly processed signals and, if available, an unprocessed signal, and this bit distribution information is used by the signal encoder 300 as illustrated in
In step 213, each channel is normalized. To this end, a scaling factor or value and an up- or downscaling information is determined. Block 213, therefore, is configured to output the scaling flag for each channel indicated at 534a. In block 214, the actual quantization of the scaling ratio determined in block 212 is performed, and this quantized scaling ratio is output at 534b for each channel. This quantized scaling ratio is also indicated as inter-channel level difference (k), i.e., for a certain channel k with respect to a reference channel having the mean energy. In block 215, the spectrum of each channel is scaled using the quantized scaling ratio. The scaling operation in block 215 is controlled by the output of block 213, i.e., by the information whether an upscaling or downscaling is to be performed. The output of block 215 represents a scaled spectrum for each channel.
In block 235, 238, the cascaded processing with the full tree or a simplified tree processing or a non-cascaded processing are continued until a certain termination criterion. At the certain termination criterion, a pair indication output by, for example, block 229 and a stereo mode processing information output by block 232a are generated and input into the bit stream in the pairwise processing side information 532 explained with respect to
In block 284, a total energy among all signals output by the adaptive joint signal processor 200 is calculated. A bit distribution information is calculated in block 286 for each signal based on the signal energy for each stereo processed signal or, if available, an energy reverted or energy weighted signal and based on the total energy output by block 284. This side information 536 generated by block 286 is, on the one hand, forwarded to the signal encoder 300 of
The actual bit allocation is performed in an embodiment based on the procedures illustrated in
In step 292, a refinement is performed. When the quantization was so that the remaining bits are assigned and the result is higher than the available number of bits, a subtraction of bits assigned in block 291 has to be performed. When, however, the quantization of the energy ratio was so that the assignment procedure in block 291 is so that there are still bits to be further assigned, these bits can be additionally given or distributed in the refinement step 292. If, subsequent to the refinement step, there still exist any bits to use by the signal encoder, a final donation step 293 is performed, and the final donation is done to the channel with the maximum energy. At the output of step 293, the assigned bit budget for each signal is available.
In step 300, the quantization and entropy encoding of each signal using the assigned bit budget generated by the process of steps 290, 291, 292, 293 is performed. Basically, the bit allocation is performed in such a way that a higher energy channel/signal is quantized more precise than a lower energy channel/signal. Importantly the bit allocation is not done using the original signals or the whitened signals but is done using the signals at the output of the adaptive joint signal processor 200 that have different energies than the signals input into the adaptive joint signal processing due to the joint channel processing. In this context, it is also to be noted that, although a channel pair processing is the advantageous implementation, other groups of channels can be selected and processed by means of the cross correlation. For example, groups of three or even four channels can be formed by means of the adaptive joint signal processor and correspondingly processed in a cascaded full procedure or a cascaded procedure with a simplified tree or within a non-cascaded procedure.
The bit allocation illustrated in blocks 290, 291, 292, 293 is performed in the same way on the decoder-side by means of the signal decoder 700 of
In this implementation, the codec uses new concepts to merge the flexibility of signal adaptive joint coding of arbitrary channels as described in [6] by introducing the concepts described in [7] for joint stereo coding. These are:
The codec uses Frequency Domain Noise Shaping (FDNS) to perceptually whiten the signal with the rate-loop as described in [8] combined with the spectral envelope warping as described in [9]. The codec further normalized the FDNS-whitened spectrum towards the mean energy level using ILD parameters. Channel pairs for joint coding are selected in an adaptive manner as described in [6], where the stereo coding consist of a band-wise M/S vs L/R decision. The band-wise M/S decision is based on the estimated bitrate in each band when coded in the L/R and in the M/S mode as described in [7]. Bitrate distribution among the band-wise M/S processed channels is based on the energy.
Embodiments relate to an MDCT-based multi-signal encoding and decoding system with signal-adaptive joint channel processing, wherein the signal can be a channel, and the multisignal is a multichannel signal or, alternatively an audio signal being a component of a sound field description such as an Ambisonics component, i.e., W, X, Y, Z in first order Ambisonics or any other component in a higher order Ambisonics description. The signal can also be a signal of an A-format or B-format or any other format description of a sound field. Hence, the same disclosure given for “channels” is also valid for “components” or other “signals” of the multi-signal audio signal.
Each single channel k is analyzed and transformed to a whitened MDCT-domain spectrum following the processing steps as shown in the block diagram of Error! Reference source not found.
The processing blocks of the time-domain Transient Detector, Windowing, MDCT, MDST and OLA are described in [8]. MDCT and MDST form Modulated Complex Lapped Transform (MCLT); performing separately MDCT and MDST is equivalent to performing MCLT; “MCLT to MDCT” represents taking just the MDCT part of the MCLT and discarding MDST.
Temporal Noise Shaping (TNS) is done similar as described in [8] with the addition that the order of the TNS and the Frequency domain noise shaping (FDNS) is adaptive. The existence of the 2 TNS boxes in the figures is to be understood as the possibility to change the order of the FDNS and the TNS. The decision of the order of the TNS and the FDNS can be for example the one described in [9].
Frequency domain noise shaping (FDNS) and the calculation of FDNS parameters are similar to the procedure described in [9]. One difference is that the FDNS parameters for frames where TNS is inactive are calculated from the MCLT spectrum. In frames where the TNS is active, the MDST spectrum is estimated from the MDCT spectrum.
It is not necessary in any case to have a complex transform within block 108. Additionally, a time-to-spectral converter only performing an MDCT is also sufficient for certain applications and, if an imaginary part of the transform is required, this imaginary part can also be estimated from the real part, as the case may be. A feature of the TNS/FDNS processing is that, in case of TNS being inactive, the FDNS parameters are calculated from the complex spectrum, i.e., from the MCLT spectrum while, in frames, where TNS is active, the MDST spectrum is estimated from the MDCT spectrum so that one has, for the frequency domain noise shaping operation, the full complex spectrum available.
In the described system, after each channel is transformed to the whitened MDCT domain, signal-adaptive exploitation of varying similarities between arbitrary channels for joint coding is applied, based on the algorithm described in [6]. From this procedure, the respective channel-pairs are detected and chosen to be jointly coded using a band-wise M/S transform.
An overview of the encoding system is given in Error! Reference source not found. For simplicity block arrows represent single channel processing (i.e. the processing block is applied to each channel) and block “MDCT-domain analysis” is represented in detail in Error! Reference source not found.
In the following paragraphs, the individual steps of the algorithm applied per frame are described in detail. A data flow graph of the algorithm described is given in Error! Reference source not found.
It should be noted, in the initial configuration of the system, there is a channel mask indicating for which channels the multi-channel joint coding tool is active. Therefore, for input where LFE (Low-Frequency Effects/Enhancement) channels are present, they are not taken into account in the processing steps of the tool.
An M/S transform is not efficient if ILD exists, that is if channels are panned. We avoid this problem by normalizing the amplitude of the perceptually whitened spectra of all channels to a mean energy level Ē.
where N is the total number of spectral coefficients.
where α is the scaling ratio. The scaling ratio is uniformly quantized and sent to the decoder as side information bits.
(k)=max(1,min(ILDRANGE−1, └(ILDRANGE+αk+0.5┘)
where ILDRANGE=1<<ILDbits
Then the quantized scaling ratio with which the spectrum is finally scaled is given by
if Ek<Ē (upscaling)
where (k) is calculated as in previous case.
To distinguish whether we have downscaling/upscaling at decoder and in order to revert the normalization, besides the values for each channel, a 1-bit flag (0=downscaling/1=upscaling) is sent. ILDRANGE indicates the number of bits used for the transmitted quantized scaling value , and this value is known to the encoder and the decoder and does not have to be transmitted in the encoded audio signal.
In this step, in order to decide and select which channel pair has the highest degree of similarities and therefore is suitable to be selected as a pair for stereo joint coding, the inter-channel normalized cross-correlation value for each possible channel pair is calculated. The normalized cross-correlation value for each channel pair is given by the cross-spectrum as follows:
N being the total number of spectral coefficients per frame XMDCT and YMDCT being the respective spectra of the channel-pair under consideration.
The normalized cross-correlation values for each channel paired are stored in the cross-correlation vector
CC=[{tilde over (r)}0, {tilde over (r)}1, . . . , {tilde over (r)}P]
where P=(Ctotal*(Ctotal−1))/2 is the maximum number of possible pairs.
As seen in Error! Reference source not found, depending on the transient detector we can have different block sizes (e.g. 10 or 20 ms window block sizes). Therefore, the inter-channel cross-correlation is calculated given that the spectral resolution for both channels is the same. If otherwise, then the value is set to 0, thus ensuring that no such channel pair is selected for joint coding.
An indexing scheme to uniquely represent each channel pair is used. An example of such a scheme for indexing six input channels is shown in Error! Reference source not found.
The same indexing scheme is held throughout the algorithm at is used also to signal channel pairs to the decoder. The number of bits needed for signaling one channel-pair amount to
bitsidx=└log2(P−1)┘+1
After calculating the cross-correlation vector, the first channel-pair to be considered for joint-coding is the respective with the highest cross-correlation value and higher than a minimum value threshold advantageously of 0.3.
The selected pair of channels serve as input to a stereo encoding procedure, namely a band-wise M/S transform. For each spectral band, the decision whether the channels will be coded using M/S or discrete L/R coding depends on the estimated bitrate for each case. The coding method that is less demanding in terms of bits is selected. This procedure is described in detail in [7].
The output of this process results to an updated spectrum for each of the channels of the selected channel-pair. Also, information that need to be shared with the decoder (side information) regarding this channel-pair are created, i.e. which stereo mode is selected (Full M/S, dual-mono, or band-wise M/S) and if band-wise M/S is the mode selected the respective mask of indicating whether M/S coding is chosen (1) or L/R (0).
For the next steps there are two variations of the algorithm:
resulting in additional computational complexity (coming from the M/S decision process of the stereo operation) and also additional metadata that needs to be transmitted to the decoder for each channel pair.
It should be noted, that there may be cases where the stereo operation of a selected channel-pair does not alter the spectra of the channels. That happens when the M/S decision algorithm decides the coding mode should be “dual-mono”. In this case, the arbitrary channels involved are not considered a channel-pair anymore as they will be coded separately. Also, updating the cross-correlation vector will have no effect. To continue with the process, the channel-pair with the next highest value is considered. The steps in this case continue as described above.
In many cases the normalized cross-correlation values of arbitrary channel-pairs from frame to frame can be close and therefore the selection can switch often between this close values. That may cause frequent channel-pair tree switching, which may result to audible instabilities to the output system. Therefore, it is opted to use a stabilization mechanism, where a new set of channel pairs is selected only when there is a significant change to the signal and the similarities between arbitrary channels change. To detect this, the cross-correlation vector of the current frame with the vector of the previous frame is compared and when the difference is larger than a certain threshold then the selection of new channel pairs is allowed.
The variation in time of the cross-correlation vector is calculated as follows:
If Cdiff>t, then the selection of new channel-pairs to be jointly coded, as described in the previous step, is allowed. A threshold chosen is given by
t=0.15Ctot(Ctot−1)/2
If, on the other hand, the differences are small, then the same channel-pair tree as in the previous frame is used. For each given channel-pair, the band-wise M/S operation is applied as previously described. If, however, the normalized cross-correlation value of the given channel-pair does not exceed the threshold of 0.3 then the selection of new channel pairs creating a new tree is initiated.
After the termination of the iteration process for the channel pair selection there may be channels that are not part of any channel/pair and therefore are coded separately. For those channels the initial normalization of the energy level towards the mean energy level is reverted back to their original energy level. Depending on the flag signaling upscaling or downscaling the energy of these channels are reverted using the inverse of the quantized scaling ratio
Regarding IGF analysis, in case of stereo channel pairs an additional joint stereo processing is applied, as is thoroughly described in [10]. This is necessary, because for a certain destination range in the IGF spectrum the signal can be a highly correlated panned sound source. In case the source regions chosen for this particular region are not well correlated, although the energies are matched for the destination regions, the spatial image can suffer due to the uncorrelated source regions.
Therefore, for each channel pair stereo IGF is applied if the stereo mode of the core region is different to the stereo mode of the IGF region or if the stereo mode of the core is flagged as band-wise M/S. If these conditions do not apply, then single channel IGF analysis is performed. If there are single channels, not coded jointly in a channel-pair, then they also undergo a single channel IGF analysis.
After the process of joint channel-pair stereo processing, each channel is quantized and coded separately by an entropy coder. Therefore, for each channel the available number of bits should be given. In this step, the total available bits are distributed to each channel using the energies of the processed channels.
The energy of each channel, the calculation of which is described above in the normalization step, is recalculated as the spectrum for each channel may have changed due to the joint processing. The new energies are denoted {tilde over (E)}k, k=0, 1, . . . , Ctot. As a first step the energy-based ratio with which the bits will be distributed is calculated:
Here it should be noted, that in the case where the input consists also from an LFE channel, it is not taken into account for the ratio calculations. For the LFE channel, a minimal amount of bits bitsLFE is assigned only if the channel has non-zero content. The ratio is uniformly quantized:
k=max(1,min(rtRANGE−1, └rRANGE·rtk+0.5┘))
rt
RANGE=1<<rtbits
The quantized ratios {circumflex over (r)}k are stored in the bitstream to be used from the decoder to assign the same amount of bits to each channel to read the transmitted channel spectra coefficients.
The bit distribution scheme is described below:
The exact same procedure is followed from the decoder in order to determine the amount of bits to be read to decode the spectrum coefficients of each channel. rtRANGE indicates the number of bits used for the bit distribution information bitsk and this value is known to the encoder and the decoder and does not have to be transmitted in the encoded audio signal.
Quantization, noise filling and the entropy encoding, including the rate-loop, are as described in [8]. The rate-loop can be optimized using the estimated Gest. The power spectrum P (magnitude of the MCLT) is used for the tonality/noise measures in the quantization and Intelligent Gap Filling (IGF) as described in [8]. Since whitened and band-wise M/S processed MDCT spectrum is used for the power spectrum, the same FDNS and M/S processing has to be done on the MDST spectrum. The same normalization scaling based on the ILD has to be done for the MDST spectrum as it was done for the MDCT. For the frames where TNS is active, MDST spectrum used for the power spectrum calculation is estimated from the whitened and M/S processed MDCT spectrum.
The normalized channels are input into a block 220 for performing a cross correlation vector calculation and channel pair selection. Based on the procedure in block 220 which is advantageously an iterative procedure using a cascaded full tree or a cascaded simplified tree processing or which is, alternatively, a non-iterative non-cascaded processing, the corresponding stereo operations are performed in block 240 that may perform a full band or a band-wise mid/side processing or any other corresponding stereo processing operation such as rotations, scalings, any weighted or non-weighted linear or non-linear combinations, etc.
At the output of the blocks 240, a stereo intelligent gap filling (IGF) processing or any other bandwidth extension processing such as spectral band replication processing or harmonic bandwidth processing can be performed. The processing of the individual channel pairs is signaled via channel pair side information bits and, although not illustrated in
The final stage of
In step 222, the calculation is performed in order to determine whether the tree as determined for the preceding frame is to be maintained or not. To this end, the variation in time of the cross-correlation vector is calculated and, Advantageously, the sum of the individual differences of the cross-correlation vectors and, particularly, the magnitudes of the differences is calculated. In step 223, it is determined whether the sum of the differences is greater than the threshold. If this is the case, then, in step 224, the flag keepTree is set to 0, which means that the tree is not kept, but a new tree is calculated. When, however, it is determined that the sum is smaller than the threshold, block 225 sets the flag keepTree=1 so that the tree is determined from the previous frame is also applied for the current frame.
In step 226, the iteration termination criterion is checked. In case it is determined that the maximum number of channel pairs (CP) is not reached, which is, of course, the case when block 226 is accessed for the first time, and when the flag keepTree is set to 0 as determined by block 228, the procedure goes on with block 229 for the selection of the channel pair with the maximum cross-correlation from the cross-correlation vector. When, however, the tree of the earlier frame is maintained, i.e., when keepTree is equal to 1 as has been checked in block 225, block 230 determines whether the cross-correlation of the “forced” channel pair is greater than the threshold. If this is not the case, the procedure is continued with step 227, which means, nevertheless, that a new tree is to be determined although the procedure in block 223 determined the opposite. The evaluation in block 230 and the corresponding consequence in block 227 can overturn the determination in block 223 and 225.
In block 231, it is determined whether the channel pair with the maximum cross-correlation is above 0.3. If this is the case, the stereo operation in block 232 is performed, which is also indicated as 240 in
When, however, the check in block 226 or the check in block 231 results in a “no” answer, the control goes to block 236 in order to check whether a single channel exists. If this is the case, i.e., if a single channel has been found that has not been processed together with another channel in a channel-pair processing, the ILD normalization is reversed in block 237. Alternatively, the reversal in block 237 can only be a part reversal or can be some kind of weighting.
In case the iteration is completed and in case blocks 236 and 237 are completed as well, the procedure ends and all channel pairs have been processed and, at the output of the adaptive joint signal processor, there are at least three jointly processed signals in case of block 236 resulting in a “no” answer, or there are at least two jointly processed signals and an unprocessed signal corresponding to a “single channel”, when block 236 has resulted in a “yes” answer.
The decoding process starts with decoding and inverse quantization of the spectrum of the jointly coded channels, followed by the noise filling as described in 6.2.2 “MDCT based TCX” in [11] or [12]. The number of bits allocated to each channel is determined based on the window length, the stereo mode and the bitrate ratio k that are coded in the bitstream. The number of bits allocated to each channel has to be known before fully decoding the bitstream.
In the intelligent gap filling (IGF) block, lines quantized to zero in a certain range of the spectrum, called the target tile are filled with processed content from a different range of the spectrum, called the source tile. Due to the band-wise stereo processing, the stereo representation (i.e. either L/R or M/S) might differ for the source and the target tile. To ensure good quality, if the representation of the source tile is different from the representation of the target tile, the source tile is processed to transform it to the representation of the target file prior to the gap filling in the decoder. This procedure is already described in [10]. The IGF itself is, contrary to [11] and [12], applied in the whitened spectral domain instead of the original spectral domain. In contrast to the known stereo codecs (e.g. [10]), the IGF is applied in the whitened, ILD compensated spectral domain.
From the bitstream signaling it is also known if there are channel-pairs that were jointly coded. The inverse processing should start with the last channel-pair formed in the encoder, especially for the cascaded channel pair-tree, in order to convert back to the original whitened spectra of each channel. For each channel pair the inverse stereo processing is applied based on the stereo mode and the band-wise M/S decision.
For all channels that were involved in channel pairs and were jointly coded, the spectrum is de-normalized to the original energy level based on the (k) values that were sent from the encoder.
The side information extracted by the input interface 600 and forwarded to the joint signal processor 800 is the side information 530 illustrated in
The joint signal processor 800 is configured to extract or to receive from the input interface 600 an energy normalization value for each joint stereo decoded signal. This energy normalization value for each joint stereo decoded signal corresponds to the energy scaling information 530 of
In order to make sure that a channel that has received a reverse ILD normalization as explained with respect to
In an embodiment, the signal decoder 700 is configured to receive, from the input interface 600, a bit distribution value for each encoded signal as indicated in block 620. This bit distribution value illustrated at 536 in
The joint signal processor 800 has a band replication, bandwidth extension or intelligent gap filling processing functionality using certain side information included in the side information block 532. This side information is forwarded to block 810 and block 820 performs the joint stereo (decoder) processing using the result of the bandwidth extension procedure as applied by block 810. In block 810, the intelligent gap filling procedure is configured to transform a source range from one stereo representation to another stereo representation, when a destination range of the bandwidth extension or IGF processing is indicated as having the other stereo representation. When the destination range is indicated to have a mid/side stereo mode, and when the source range is indicated to have an L/R stereo mode, the L/R source range stereo mode is transformed into a mid/side source range stereo mode and, then, the IGF processing is performed with the mid/side stereo mode representation of the source range.
In block 820, the joint signal processor performs an advantageously cascaded inverse processing starting with a last signal pair, where the term “last” refers to the processing order determined and performed by the encoder. In the decoder, the “last” signal pair is the one that is processed first. Block 820 receives side information 532 which indicate, for each signal pair indicated by the signal pairs information illustrated in block 630 and, for example, implemented in the way as explained with respect to
Subsequently to the inverse processing in block 820, a de-normalization of the signals involved in the channel pairs is performed in the block 830 once again relying on side information 534 indicating a normalization information per channel. The de-normalization illustrated with respect to block 830 in
The signal decoder 700 comprises a decoder and dequantizer stage 710 for the spectra included in the encoded signal 500. The signal decoder 700 comprises a bit allocator 720 that receives, as a side information, advantageously the window length, the certain stereo mode and the bit allocation information per encoded signal. The bit allocator 720 performs the bit allocation particularly using, in an implementation, steps 290, 291, 292, 293, where the bit allocation information per encoded signal is used in step 291, and where information on the window length and the stereo mode are used in block 290 or 291.
In block 730, a noise filling also advantageously using noise filling side information is performed for ranges in the spectrum that are quantized to zero and that are not within the IGF range. Noise filling is advantageously limited to a low band portion of the signal output by block 710. In block 810, and using certain side information, an intelligent gap filling or generally bandwidth extension processing is performed that, importantly, operates on whitened spectra.
In block 820, and using side information, the inverse stereo processor performs the procedures to undo the processing performed in
Subsequently, further advantages and specific features of embodiments are indicated.
The scope of this invention is to provide a solution for principles from [6] when processing perceptually whitened and ILD compensated signals.
As described in previous paragraphs, in this implementation, the codec uses new means to merge the flexibility of signal adaptive joint coding of arbitrary channels as described in [6] by introducing the concepts described in [7] for joint stereo coding. The novelty of the proposed invention is summarized in the following differences:
Embodiments of the invention relate to a signal adaptive joint coding of a multichannel system with perceptually whitened and ILD compensated spectra, where joint coding consists of a simple per band M/S transform decision based on the estimated number of bits for an entropy coder.
Although some aspects have been described in the context of an apparatus, it is clear that these aspects also represent a description of the corresponding method, where a block or device corresponds to a method step or a feature of a method step. Analogously, aspects described in the context of a method step also represent a description of a corresponding block or item or feature of a corresponding apparatus. Some or all of the method steps may be executed by (or using) a hardware apparatus, like for example, a microprocessor, a programmable computer or an electronic circuit. In some embodiments, some one or more of the most important method steps may be executed by such an apparatus.
The inventive encoded audio signal can be stored on a digital storage medium or can be transmitted on a transmission medium such as a wireless transmission medium or a wired transmission medium such as the Internet.
Depending on certain implementation requirements, embodiments of the invention can be implemented in hardware or in software. The implementation can be performed using a digital storage medium, for example a floppy disk, a DVD, a Blu-Ray, a CD, a ROM, a PROM, an EPROM, an EEPROM or a FLASH memory, having electronically readable control signals stored thereon, which cooperate (or are capable of cooperating) with a programmable computer system such that the respective method is performed. Therefore, the digital storage medium may be computer readable.
Some embodiments according to the invention comprise a data carrier having electronically readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods described herein is performed.
Generally, embodiments of the present invention can be implemented as a computer program product with a program code, the program code being operative for performing one of the methods when the computer program product runs on a computer. The program code may for example be stored on a machine readable carrier.
Other embodiments comprise the computer program for performing one of the methods described herein, stored on a machine readable carrier.
In other words, an embodiment of the inventive method is, therefore, a computer program having a program code for performing one of the methods described herein, when the computer program runs on a computer.
A further embodiment of the inventive methods is, therefore, a data carrier (or a digital storage medium, or a computer-readable medium) comprising, recorded thereon, the computer program for performing one of the methods described herein. The data carrier, the digital storage medium or the recorded medium are typically tangible and/or non-transitionary.
A further embodiment of the inventive method is, therefore, a data stream or a sequence of signals representing the computer program for performing one of the methods described herein. The data stream or the sequence of signals may for example be configured to be transferred via a data communication connection, for example via the Internet.
A further embodiment comprises a processing means, for example a computer, or a programmable logic device, configured to or adapted to perform one of the methods described herein.
A further embodiment comprises a computer having installed thereon the computer program for performing one of the methods described herein.
A further embodiment according to the invention comprises an apparatus or a system configured to transfer (for example, electronically or optically) a computer program for performing one of the methods described herein to a receiver. The receiver may, for example, be a computer, a mobile device, a memory device or the like. The apparatus or system may, for example, comprise a file server for transferring the computer program to the receiver.
In some embodiments, a programmable logic device (for example a field programmable gate array) may be used to perform some or all of the functionalities of the methods described herein. In some embodiments, a field programmable gate array may cooperate with a microprocessor in order to perform one of the methods described herein. Generally, the methods are advantageously performed by any hardware apparatus.
The apparatus described herein may be implemented using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer.
The methods described herein may be performed using a hardware apparatus, or using a computer, or using a combination of a hardware apparatus and a computer.
While this invention has been described in terms of several embodiments, there are alterations, permutations, and equivalents which fall within the scope of this invention. It should also be noted that there are many alternative ways of implementing the methods and compositions of the present invention. It is therefore intended that the following appended claims be interpreted as including all such alterations, permutations and equivalents as fall within the true spirit and scope of the present invention.
These references are all incorporated herein by reference in their entirety.
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[6] F. Schuh, S. Dick, R. Füg, C. R. Helmrich, N. Rettelbach and T. Schwegler, “Efficient Multichannel Audio Transform Coding with Low Delay and Complexity,” in AES Convention, Los Angeles, Sep. 20, 2016.
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[8] 3GPP TS 26.445, Codec for Enhanced Voice Services (EVS); Detailed algorithmic description.
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Number | Date | Country | Kind |
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18181767.7 | Jul 2018 | EU | national |
This application is a continuation of copending International Application No. PCT/EP2019/067256, filed Jun. 27, 2019, which is incorporated herein by reference in its entirety, and additionally claims priority from European Application No. EP 18 181 767.7, filed Jul. 4, 2018, which is incorporated herein by reference in its entirety.
Number | Date | Country | |
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Parent | PCT/EP2019/067256 | Jun 2019 | US |
Child | 17124628 | US |