EP1697930B1 - Vorrichtung und verfahren zum verarbeiten eines multikanalsignals - Google Patents

Vorrichtung und verfahren zum verarbeiten eines multikanalsignals Download PDF

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EP1697930B1
EP1697930B1 EP05715611A EP05715611A EP1697930B1 EP 1697930 B1 EP1697930 B1 EP 1697930B1 EP 05715611 A EP05715611 A EP 05715611A EP 05715611 A EP05715611 A EP 05715611A EP 1697930 B1 EP1697930 B1 EP 1697930B1
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channel
prediction
block
similarity
spectral
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EP1697930A1 (de
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Jürgen HERRE
Michael Schug
Alexander Groeschl
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/03—Spectral prediction for preventing pre-echo; Temporary noise shaping [TNS], e.g. in MPEG2 or MPEG4
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders

Definitions

  • the present invention relates to audio encoders, and more particularly to audio encoders that are transformation-based, that is, where a temporal representation is converted to a spectral representation at the beginning of the encoder pipeline.
  • FIG. 3 A known transform-based audio encoder is shown in FIG.
  • the encoder shown in Fig. 3 is in International Standard ISO / IEC 14496-3: 2001 (E), Subpart 4, page 4 , and also known in the art as an AAC encoder.
  • an audio signal to be coded is fed. This is first supplied to a scaling stage 1002 in which a so-called AAC gain control is performed to set the level of the audio signal. Scaling page information is provided to a bitstream formatter 1004, as indicated by the arrow between block 1002 and block 1004. The scaled audio signal is then applied to an MDCT filter bank 1006.
  • the filter bank implements a modified discrete cosine transform with 50% overlapping windows, the window length being determined by a block 1008.
  • block 1008 is for windowing transient signals with shorter windows, and for windowing stationary signals with longer windows. This serves to achieve a higher time resolution (at the expense of frequency resolution) due to the shorter transient signal windows while For more stationary signals, a higher frequency resolution (at the expense of time resolution) is achieved by longer windows, with longer windows tending to be preferred because they promise greater coding gain.
  • temporally successive blocks of spectral values are present, which, depending on the embodiment of the filter bank, may be MDCT coefficients, Fourier coefficients or even subband signals, each subband signal having a certain limited bandwidth passing through the corresponding subband channel in the filter bank 1006, and wherein each subband signal has a certain number of subband samples.
  • the filter bank outputs temporally successive blocks of MDCT spectral coefficients, which generally represent successive short-term spectra of the audio signal to be encoded at input 1000.
  • a block of MDCT spectral values is then fed into a TNS processing block 1010 in which temporal noise shaping (TNS) takes place.
  • TNS temporal noise shaping
  • the TNS technique is used to shape the temporal shape of the quantization noise within each window of the transform. This is achieved by applying a filtering process to parts of the spectral data of each channel.
  • the coding is performed on a window basis.
  • the following steps are performed to apply the TNS tool to a window of spectral data, that is, to a block of spectral values.
  • a frequency range is selected for the TNS tool.
  • a suitable choice is to cover a frequency range of 1.5 kHz up to the highest possible scale factor band with a filter. It should be noted that this frequency range depends on the sampling rate as specified in the AAC standard (ISO / IEC 14496-3: 2001 (E)).
  • LPC linear predictive coding
  • the expected prediction gain PG is obtained. Further, the reflection coefficients or Parcor coefficients are obtained.
  • the TNS tool is not applied. In this case, control information is written in the bit stream for a decoder to know that no TNS processing has been performed.
  • TNS processing is applied.
  • the reflection coefficients are quantized.
  • the order of the noise shaping filter used is determined by removing all the reflection coefficients having an absolute value less than a threshold from the "tail" of the reflection coefficient array. The number of remaining reflection coefficients is on the order of the noise shaping filter.
  • a suitable threshold is 0.1.
  • the remaining reflection coefficients are typically converted to linear prediction coefficients, which technique is also known as a "step-up" procedure.
  • the calculated LPC coefficients are then used as coder noise shaping filter coefficients, ie as prediction filter coefficients.
  • This FIR filter is routed over the specified target frequency range.
  • the decoding uses an autoregressive filter, while the coding uses a so-called moving average filter.
  • the page information for the TNS tool is also supplied to the bit stream formatter as shown by the arrow shown between the block TNS processing 1010 and the bitstream formatter 1004 in FIG.
  • the center / side encoder 1012 is active when the audio signal to be encoded is a multi-channel signal, that is, a stereo signal having a left channel and a right channel. So far, that is, in the processing direction before the block 1012 in Fig. 3, the left and right stereo channels have been separately processed, that is, scaled, transformed by the filter bank, subjected to TNS processing or not, etc.
  • middle / side encoder In the middle / side encoder is then first checked whether a middle / side encoding makes sense, that brings a coding gain at all. A middle / side encoding will then bring a coding gain if the left and the right channel are more similar, because then the center channel, that is the sum of the left and the right channel is almost equal to the left or the right channel, apart from the scaling by the factor 1/2, while the page channel has only very small values, since it is equal to the difference between the left and the right channel.
  • the quantizer 1014 is given a allowed perturbation per scale factor band by a psycho-acoustic model 1020.
  • the quantizer operates iteratively, d. H.
  • An external iteration loop is called first, which then calls an inner iteration loop.
  • a quantization of a block of values is made at the input of the quantizer 1014.
  • the inner loop quantizes the MDCT coefficients, consuming a certain number of bits.
  • the outer loop calculates the distortion and modified energy of the coefficients using the scale factor to again invoke an inner loop. This process is iterated until a certain conditional set is met.
  • the signal is reconstructed to compute the perturbation introduced by the quantization and to compare it with the allowable perturbation provided by the psycho-acoustic model 1020. Furthermore, the scale factors are increased from iteration to iteration by one step, for each iteration of the outer iteration loop.
  • the iteration ie the analysis-by-synthesis procedure is terminated, and the resulting scale factors are encoded as set forth in block 1014 and supplied in encoded form to bitstream formatter 1004, as indicated by the arrow drawn between the block 1014 and the block 1004.
  • the quantized values are then fed to entropy coder 1016, which typically performs entropy coding using several Huffman code tables for different scale factor bands to transmit the quantized values into a binary format.
  • entropy coding in the form of Huffman coding relies on code tables that are created on the basis of expected signal statistics and in which frequently occurring values get shorter code words than more rarely occurring values.
  • the entropy-coded values are then also supplied as actual main information to the bit stream formatter 1004, which then outputs the coded audio signal on the output side in accordance with a specific bit stream syntax.
  • predictive filtering is used in TNS processing block 1010 to time-shaping the quantization noise within an encoding frame.
  • the temporal shaping of the quantization noise is performed by filtering the spectral coefficients over the frequency in the encoder before the quantization and subsequent inverse filtering in the decoder.
  • TNS processing causes the quantization noise envelope to be timed below the envelope of the signal to avoid pre-echo artifacts.
  • the application of the TNS results from an estimation of the prediction gain of the filtering as stated above.
  • the filter coefficients for each encoding frame are determined via a correlation measure. The calculation of the filter coefficients is done separately for each channel. They are also transmitted separately in the coded bit stream.
  • a disadvantage of the activation / deactivation of the TNS concept is the fact that for each stereo channel, if Once TNS processing has been activated due to the good expected coding gain, the TNS filtering for each channel takes place separately. So this is still unproblematic with relatively different channels.
  • the left and right channels are relatively similar, then the left and right channels in an extreme example have exactly the same payload, such as a speaker, and differ only in terms of the noise inevitably contained in the channels, so when standing Nevertheless, the technology calculates and uses a separate TNS filter for each channel.
  • the TNS filter depends directly on the left or right channel, and in particular reacts relatively sensitively to the spectral data of the left and the right channel, the signal is also very similar in the case of a signal in which the left and the right channel are very similar
  • a TNS processing with a separate prediction filter is carried out for each channel. This leads to a different temporal noise shaping taking place in the two stereo channels due to the different filter coefficients.
  • the known procedure has a far possibly even more serious disadvantage.
  • the TNS output values that is, the spectral residuals
  • the spectral residuals are subjected to center / side coding in the center / side encoder 1002 of FIG. While the two channels were still relatively the same before TNS processing, this can not be said after TNS processing.
  • the described stereo effect introduced by the separate TNS processing, makes the spectral residuals of the two channels more dissimilar than they would actually be. This leads to a immediate drop in coding gain due to the mid / side coding, which is particularly disadvantageous for applications where a low bit rate is required.
  • the known TNS activation is thus problematic for stereo signals that use similar but not exactly identical signal information in both channels, such as mono-like speech signals. If different filter coefficients are determined for both channels in the case of TNS detection, this leads to a temporally different shaping of the quantization noise in the channels. This can lead to audible artifacts because z. B. the original mono-like sound image gets an unwanted stereo character through these temporal differences. Furthermore, as has been stated, the TNS-modified spectrum is subjected to center / side encoding in a subsequent step. Different filters in both channels additionally reduce the similarity of the spectral coefficients and thus the center / side gain.
  • the DE 19829284C2 discloses a method and apparatus for processing a temporal stereo signal and a method and apparatus for decoding an audio bitstream encoded using prediction over frequency.
  • the left, the right and the mono channel can be subjected to their own prediction over the frequency, ie a TNS processing.
  • a separate complete prediction can be performed for each channel.
  • calculation of the prediction coefficients for the left channel may be performed, which are then used to filter the right channel and the mono channel.
  • the object of the present invention is to provide a concept for processing a multi-channel signal, the lower artifacts and still allows a good compression of the information.
  • the present invention is based on the finding that if the left and the right channel are similar, that is to say exceed a similarity measure, the same TNS filtering is to be used for both channels. This ensures that the TNS processing no pseudo-stereo artifacts are introduced into the multi-channel signal, as is achieved by using the same prediction filter for both channels that the temporal shaping of the quantization noise for both channels takes place identically, so that no pseudo Stereo artifacts are heard.
  • the similarity of the signals after the TNS filtering ie the similarity of the residual spectral values corresponds to the similarity of the input signals in the filter and not, as in the prior art, the similarity of the input signals, which is still reduced by different filters.
  • Fig. 1 shows an apparatus for processing a multi-channel signal, wherein the multi-channel signal is represented by one block of spectral values for at least two channels, as shown by L and R.
  • the blocks of spectral values are represented by z.
  • the blocks of spectral values for each channel are then fed in a preferred embodiment of the present invention to a means 12 for determining a similarity between the two channels.
  • the means for determining the similarity between the two channels may also be as shown in FIG Using time domain samples 1 (t) or r (t) for each channel.
  • the means 12 for determining the similarity between the first and the second channel is operative to generate, based on a similarity measure or alternatively a measure of dissimilarity, a control signal on a control line 14 having at least two states, one of which expresses that Blocks of spectral values of the two channels are similar, or that in its other state states that the blocks of spectral values are dissimilar for each channel.
  • the decision as to whether similarity or dissimilarity prevails can be made using a preferably numerical similarity measure.
  • Both the block of spectral values for the left channel and the block of spectral values for the right channel are fed to a means 16 for performing a prediction filtering.
  • predictive filtering is performed over the frequency, the means being adapted to perform, to perform the prediction versus frequency, a common prediction filter 16a for the block of spectral values of the first channel and for the block of spectral values of the first channel second channel if the similarity is greater than a threshold similarity.
  • the means 16 for performing the prediction filtering is notified by the similarity determining means 12 that the two blocks of spectral values are dissimilar for each channel, that is, similar to less than a threshold similarity, then the means 16 is to perform the prediction filtering apply different filters 16b to the left and right channels.
  • the output signals of device 16 are thus left-channel spectral residuals at output 18a as well as right-channel spectral residuals at output 18b, where, depending on the similarity of the left and right channels, the spectral residuals of the two channels using the same prediction filter (Case 16a) or using different prediction filters (Case 16b).
  • the spectral residuals of the left and right channels may be either directly or after multiple processing such as described in US Pat. B.
  • AAC standard are supplied to a center / side stereo encoder, which outputs at an output 21 a, the center signal as half of the sum of left and right channel, while the side signal as half of the difference of left and right right channel is output.
  • the page signal is now smaller due to the synchronization of the TNS processing of the two channels than in the case where different TNS filters are used for similar channels.
  • this promises a higher coding gain.
  • Fig. 2 there is shown a preferred embodiment of the present invention in which the first stage of the TNS calculation is already performed in the similarity determining means 12, namely the calculation of the parc reflection coefficients and the prediction gain for both the left channel and the right channel, as represented by blocks 12a, 12b.
  • This TNS processing thus provides both the filter coefficients for the final prediction filter to be used and the prediction gain, and this prediction gain is also needed to decide whether or not TNS processing should be performed at all.
  • the prediction gain for the first, left channel, denoted by PG1 in FIG. 2, as well as the prediction gain for the right channel, denoted PG2 in FIG. 2, are fed to a similarity measure determiner, shown in FIG 12c is designated.
  • This similarity determining means is operable to calculate the absolute amount of the difference or the relative difference of the two prediction gains and to see if it is below a predetermined deviation threshold S. If the absolute amount of the difference of the prediction gains is below the threshold S, then it is assumed that the two signals are similar, and the question in the block 12c is answered with Yes. If, on the other hand, it is determined that the difference is greater than the similarity threshold S, the question is answered with no.
  • a common filter is used for both channels L and R, while in the case of answering the question in block 12c with No separate filters are used, ie a TNS processing, as in the state the technique can be performed.
  • the device 16 is supplied with a set of filter coefficients FKL for the left channel and a set of filter coefficients FKR for the right channel from the devices 12a and 12b, respectively.
  • a particular selection is made in a block 16c.
  • block 16c it is decided which channel has the greater energy. If it is determined that the left channel has the greater energy, the filter coefficients FKL calculated by the left channel device 12a are used for the common filtering. On the other hand, if it is determined in block 16c that the right channel has the greater energy, then for common filtering, the set of filter coefficients FKR calculated for the right channel in the device 12b is used.
  • both the time signal and the spectral signal can be used for energy determination. Due to the fact that transformation artifacts that may have already taken place in the spectral signal are preferred, it is preferable to use the spectral signals of the left and right channels for the "energy decision" in block 16c.
  • a TNS synchronization that is, the use of the same filter coefficients for both channels is used when the prediction gains for the left and right channels differ by less than three percent. If both channels differ by more than three percent, the question is answered in block 12c of FIG. 2 with "no".
  • the similarity determination may also be achieved using other details of the signal, so that when a similarity has been determined, only the TNS filter coefficient set needs to be calculated for the channel that will be used for the prediction filtering of both stereo channels. This has the advantage that, if Fig. 2 is considered, and if the signals are similar, only either block 12a or block 12b will be active.
  • the concept according to the invention can also be used to further reduce the bit rate of the coded signal. While different TNS page information is transmitted for both channels when using two different reflection coefficients, when filtering the two channels with the same prediction filter, TNS information must be transmitted only once for both channels. Therefore, the concept of the invention can also achieve a reduction of the bit rate such that a set of TNS page information is "saved" if the left and right channels are similar.
  • the inventive concept is not basically limited to stereo signals, but could be applied in a multi-channel environment between different channel pairs or even groups of more than 2 channels.
  • a determination of the left-right channel cross-correlation measure k or a determination of the TNS prediction gain and the TNS filter coefficients may be made separately for each channel.
  • the synchronization decision is made if k exceeds a threshold (e.g., 0.6) and MS stereo coding is enabled.
  • a threshold e.g., 0.6
  • MS stereo coding is enabled.
  • the MS criterion can also be omitted.
  • TNS prediction gain and TNS filter coefficients are made separately for each channel. Then a decision is made. If the prediction gain of both channels differs by no more than a degree, e.g. B. 3%, the synchronization takes place.
  • the reference channel can also be chosen arbitrarily, if one can assume a similarity of the channels. Again, there is a copying of the TNS filter coefficients from the reference channel to the other channel, whereupon an application of the synchronized or unsynchronized TNS filters to the spectrum takes place.
  • TNS in a channel is always activated depends on the prediction gain in this channel. If this exceeds a certain threshold, TNS is activated for this channel. Alternatively, a TNS synchronization is made for 2 channels if TNS was activated in only one of the two channels. Condition is then that e.g. the prediction gain is similar, ie one channel just above the activation limit, and one channel just below the activation limit. From this comparison, the activation of TNS for both channels with equal coefficients is derived, or possibly the deactivation for both channels.
  • the inventive method for processing a multi-channel signal can be implemented in hardware or in software.
  • the implementation may be on a digital storage medium, in particular a floppy disk or CD with electronically readable control signals, which may interact with a programmable computer system such that the method is performed.
  • the invention thus also consists in a computer program product with a program code stored on a machine-readable carrier for carrying out the method according to the invention, when the computer program product runs on a computer.
  • the invention can thus be realized as a computer program with a program code for carrying out the method when the computer program runs on a computer.

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EP05715611A 2004-03-01 2005-02-28 Vorrichtung und verfahren zum verarbeiten eines multikanalsignals Active EP1697930B1 (de)

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DE102004009954A DE102004009954B4 (de) 2004-03-01 2004-03-01 Vorrichtung und Verfahren zum Verarbeiten eines Multikanalsignals
PCT/EP2005/002110 WO2005083678A1 (de) 2004-03-01 2005-02-28 Vorrichtung und verfahren zum verarbeiten eines multikanalsignals

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EP (1) EP1697930B1 (ko)
JP (1) JP4413257B2 (ko)
KR (1) KR100823097B1 (ko)
CN (1) CN1926608B (ko)
AT (1) ATE364882T1 (ko)
AU (1) AU2005217517B2 (ko)
BR (1) BRPI0507207B1 (ko)
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AU2005217517B2 (en) 2008-06-26
BRPI0507207A8 (pt) 2018-06-12
US20070033056A1 (en) 2007-02-08
PT1697930E (pt) 2007-09-25
NO20064431L (no) 2006-09-29
DK1697930T3 (da) 2007-10-08
KR100823097B1 (ko) 2008-04-18
BRPI0507207A (pt) 2007-06-12
ES2286798T3 (es) 2007-12-01
CN1926608A (zh) 2007-03-07
DE102004009954B4 (de) 2005-12-15
ATE364882T1 (de) 2007-07-15
BRPI0507207B1 (pt) 2018-12-26
NO339114B1 (no) 2016-11-14
DE102004009954A1 (de) 2005-09-29
CN1926608B (zh) 2010-05-05
RU2332727C2 (ru) 2008-08-27
AU2005217517A1 (en) 2005-09-09
JP2007525718A (ja) 2007-09-06
RU2006134641A (ru) 2008-04-10
IL177213A (en) 2011-10-31
KR20060121982A (ko) 2006-11-29
US7340391B2 (en) 2008-03-04
EP1697930A1 (de) 2006-09-06
HK1095194A1 (en) 2007-04-27
DE502005000864D1 (de) 2007-07-26
CA2558161C (en) 2010-05-11
WO2005083678A1 (de) 2005-09-09
IL177213A0 (en) 2006-12-10
CA2558161A1 (en) 2005-09-09
JP4413257B2 (ja) 2010-02-10

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