JP2007529021A - Fidelity optimized variable frame length coding - Google Patents

Abstract

A method of encoding multi-channel audio signals comprises generating of a first output signal (x' mono ), being encoding (38) parameters representing a main signal (x mono ). The main signal (x mono ) is a first linear combination (34) of signals (16A,16B) of at least a first and a second channel. The method further comprises generating (30) of a second output signal (p side ), being encoding parameters representing a side signal (x side ). The side signal (x side ) is a second linear combination (36) of signals (16A,16B) of at least the first and the second channel within an encoding frame. The method is characterised in that the generating of the second output signal further comprises scaling of the side signal (x side ) to an energy contour of the main signal (x mono ). A method of decoding is also presented as well as an encoder, a decoder and audio system, all according to the same basic idea.

Description

  The present invention relates to encoding audio signals, and more particularly to encoding multi-channel audio signals.

  There is a high market demand for transmitting or storing audio signals at a low bit rate while maintaining high audio quality. Specifically, when there are restrictions on transmission resources and storage devices, low bit rate operation is an essential cost factor. This is a common recognition in streaming and messaging applications in mobile communication systems such as GSM, UMTS, CDMA, for example.

  At present, there are no standardized codecs that can be used in mobile communication systems that provide high stereo audio quality at economically interesting bit rates. What is possible with the current codec is mono transmission of audio signals. Stereo transmission is also available to some extent. However, due to bit rate restrictions, the current situation is that very limited stereo representations are unavoidable.

  The simplest method of stereo encoding or multi-channel encoding of an audio signal is to separately encode signals of different channels as individual independent signals. Another basic method used in stereo FM radio transmission and coexisting with a conventional monaural radio receiver is to transmit two-channel sum and difference signals.

  Current audio codecs such as MPEG-1 / 2 Layer III and MPEG-2 / 4 AAC use so-called joint stereo coding. According to this method, the signals of different channels are processed jointly rather than separately and individually. The two most commonly used joint stereo coding methods are “Mid / Side” (M / S) stereo coding and intensity stereo coding. Known as intensity stereo coding, these are generally applied to the subbands of the stereo signal or multichannel signal to be encoded.

  M / S stereo coding is described in stereo FM radio in the sense that it encodes and transmits the sum and difference signals of the channel sub-bands, thereby exploiting the redundancy between the channel and sub-bands. The procedure is the same. The structure and operation of an encoder based on M / S stereo coding is described, for example, in US Pat. No. 5,285,498 by J. D. Johnston.

  On the other hand, intensity stereo can take advantage of stereo independence. This transmits the combined strength of the channels (in different subbands) along with some position information indicating how the strength is distributed in the channel. Intensity stereo outputs only the spectral amplitude information of the channel. Phase information is not transmitted. For this reason and because time-channel information (more specifically, channel-to-channel time difference) exhibits a major psychoacoustic relevance especially at low frequencies, intensity stereo is only at frequencies above 2 kHz, for example. Can be used. Intensity stereo coding is described, for example, in European Patent No. 0497413 by R. Veldhuis et al.

  A recently developed stereo coding method is, for example, the name “Binaural cue coding applied to stereo and multi channel audio compression” by C. Faller et al. (Conference Literature). This method is a parametric multi-channel audio coding method. The basic principle is that on the encoding side, the input signals from the N channels c1, c2,..., CN are combined into a single mono signal m. The mono signal is audio encoded using any conventional mono audio codec. In parallel, parameters are derived from the channel signal that describes the multi-channel image. The parameters are encoded and sent to the decoder along with the audio bit stream. The decoder first decodes the monaural signal m 'and then regenerates the channel signals c1', c2 ', ..., cN' based on the parameter description of the multichannel image.

  The principle of the binaural cue coding (BCC) method transmits an encoded mono signal and so-called BCC parameters. The BCC parameters have coded inter-channel level difference and inter-channel time difference information for the subbands of the original multi-channel input signal. The decoder regenerates the difference channel signal by applying the subband level and phase adjustment of the mono signal based on the BCC parameters. An advantage over M / S or intensity stereo etc. is that stereo information with time channel information is transmitted at a much lower bit rate. However, this technique requires a time-frequency transform that is cumbersome to compute for each of the channels in both the encoder and the decoder.

  Furthermore, BCC does not deal with the fact that a lot of stereo information is spread, i.e. not coming from any particular direction, especially at low frequencies. A diffuse sound field exists in both channels of stereo recording, but the phases are greatly shifted with respect to each other. If an algorithm such as BCC is under the influence of a recording with a large amount of diffuse sound field, the regenerated stereo image will interfere and the BCC algorithm will only pan a signal in a particular frequency band to the left or right Jump from left to right.

  One way to encode a stereo signal and to achieve a good reproduction of the diffuse sound field is to use a coding scheme very similar to the technique used in FM stereo broadcasting, ie mono (left + right) Some encode the signal and the difference (left-right) signal separately.

  The technique described in US Pat. No. 5,434,948 by C. E. Holt et al. Uses a technique similar to BCC to encode mono signals and side information. In this case, the sub information consists of a prediction filter and optionally a residual signal. In the prediction filter, evaluation is performed by a Least-Mean-Square algorithm, and when applied to a monaural signal, a multi-channel audio signal can be predicted. This technique can achieve very low bit rate coding for multi-channel sound sources, but with a loss of quality, as discussed further below.

  Finally, for completeness, the techniques used for 3D audio are described. This technique combines a right channel signal and a left channel signal by filtering the sound source signal with so-called head-related filters. However, this technique requires that different sound source signals be separated and is therefore generally not applicable to stereo or multi-channel coding.

US Pat. No. 5,285,498 European Patent No. 0497413 C. Faller et al., “Binaural cue coding applied to stereo and multi-channel audio compression”, 112th AES Conference, May 2002, Munich, Germany US Pat. No. 5,434,948

(Overview)
Specifically, problems associated with existing coding schemes based on the coding of a main signal and one or more sub-signal signals are the division of audio information into frames, resulting in unpleasant noise ( perceptual artefacts) can occur. By dividing the information into frames of relatively long duration, the average bit rate is generally reduced. This may be useful, for example, for music that contains a large amount of diffuse sound. However, in temporary music and voice with a high volume, fine temporal changes become unclear over the frame duration, resulting in ghostly sound and pre-echo problems. If encoding is performed with a shorter frame length, on the contrary, sound can be expressed more precisely and energy is minimized, but the transmission bit rate is increased and the amount of calculation is increased. Therefore, even if the frame length is very short, the encoding efficiency may be reduced. In addition, the discontinuity of the encoding parameter may occur due to an increase in the frame boundary, and this may be perceived as an abnormal sound.

  Another problem associated with schemes based on the encoding of the main signal and one or more sub-signals is an increase in computational complexity. Specifically, when short frames are used, dealing with parameter discontinuities from frame to frame is a complex task. If a long frame is used, a very large sub-signal may be generated due to a temporary sound evaluation error, which increases the demand for transmission rate.

  Accordingly, an object of the present invention is to provide an encoding method and apparatus capable of improving the audible quality of a multi-channel audio signal. It is to avoid. Another object of the present invention is to provide an encoding method and apparatus that requires less processing power and has a more constant transmission bit rate.

  The above objective is accomplished by a method and device according to the appended claims. In general, polyphonic signals are used to generate a main signal and a sub-signal, which are typically monaural signals. The main signal is encoded according to the encoding principle of the prior art. Several schemes are provided for encoding the side signals. Each coding scheme is characterized by a set of subframes of different lengths. The total length of the subframe corresponds to the length of the encoding frame of the encoding scheme. The set of subframes has at least one subframe. The coding scheme used for the sub-signal is selected depending at least in part on the current signal content of the polyphonic signal.

  In one embodiment, the selection is made prior to encoding based on signal feature analysis. In other embodiments, the sub-signal is encoded by each of the encoding schemes, and an optimal encoding scheme is selected based on a measurement of the encoding quality.

  In the preferred embodiment, the sub-residual signal is generated as the difference between the sub-signal scaled by the balance factor and the main signal. The balance factor is selected to minimize the secondary residual signal. The optimized sub-residual signal and balance factor are encoded and provided as parameters representing the sub-signal. On the decoder side, the sub-signal is recovered using the balance coefficient, the sub-residual signal, and the main signal.

  In other preferred embodiments, the sub-signal encoding has iso-energy curve scaling to avoid pre-echo effects. Furthermore, different coding schemes can have different coding procedures for different subframes.

  The main advantage of the present invention is that the audible quality of the audio signal is improved. Furthermore, the present invention enables multi-channel signal transmission at very low bit rates.

(Detailed explanation)
FIG. 1 is a diagram illustrating an exemplary system 1 according to a preferred embodiment of the present invention. The transmitter 10 includes an antenna 12 that includes associated hardware and software so that the radio signal 5 can be transmitted to the receiver 20. The transmitter 10 comprises, among other things, a multi-channel encoder 14, which converts several input channel 16 signals into output signals suitable for wireless transmission. An example of a suitable multi-channel encoder 14 will be described in detail later. The input channel 16 signal may be provided by an audio signal storage device 18, such as a digital representation data file for recording, a magnetic tape, or an audio vinyl disk recording. The signal on the input channel 16 may be provided “live” from a set of microphones 19, for example. If the audio signal is not already in digital form, it is digitized before entering multichannel encoder 14.

  On the receiver 20 side, an antenna 22 with associated hardware and software handles the actual reception of the radio signal 5 representing the polyphonic audio signal. Here, normal functions such as error correction are implemented. A decoder 24 decodes the received radio signal 5, thereby converting the conveyed audio data into a number of output channel 26 signals. The output signal can be provided to the speaker 29 for immediate output, or can be stored in any type of audio signal storage device 28, for example.

  The system 1 can be, for example, a system for providing a conference call system, voice service or other audio application. In some systems, such as teleconference systems, communication must be a dual transmission system, while the distribution of musical sounds from service providers to subscribers is basically a unidirectional transmission system. It's okay. Transmission of the signal from the transmitter 10 to the receiver 20 may be performed by any other means, such as by different types of electromagnetic waves, cables, or fibers, or combinations thereof.

FIG. 2a shows an embodiment of an encoder according to the invention. In this embodiment, the polyphonic signal is a stereo signal with two channels a and b received at inputs 16A and 16B, respectively. The signals of channels a and b are output to the preprocessing unit 32, where different signal conditioning processes can be applied. The (possibly modified) signal from the output of the preprocessor 32 is added in an adder 34. The adder 34 also divides the addition result by 2. The signal X _mono generated in this way is the main signal of the stereo signal because it basically has all the data from both channels. Thus, in this embodiment, the main signal represents a pure “mono” signal. The main signal X _mono is provided to a main signal encoding unit 38 that encodes the main signal according to any suitable encoding principle. Since the prior art can be applied to the principle, the description is omitted in this specification. The main signal encoding unit 38 outputs an output signal p _mono which is an encoding parameter representing the main signal.

In the subtractor 36, the difference of the channel signals (divided by 2) is provided as the _side signal x _side . In this embodiment, the sub signal represents the difference between the two channels of the stereo signal. The sub signal x _side is provided to the sub signal encoding unit 30. A preferred embodiment of the sub-signal encoding unit 30 will be further described below. According to the sub-signal encoding procedure described in detail below, the sub-signal x _side is converted into an encoding parameter p _side representing the sub-signal x _side . In an embodiment, this encoding is also performed using information of the main signal x _mono . Arrow 42 indicates such a provision and the original _uncoded main signal x _mono is used. In another embodiment, the main signal information used in the sub-signal encoding unit 30 can be inferred from the encoding parameter p _mono representing the main signal, as indicated by the broken line 44.

Encoding parameters p _mono representing the main signal x _mono is a first output signal, encoded parameters p _side representing the side signal x _side is a second output signal. In a normal case, these two output signals p _mono and p _side both represent perfect stereo sound and are multiplexed into one transmission signal 52 by the multiplexer 40. However, in other embodiments, the transmission of the first output signal p _mono and the second output signal p _side can be performed separately.

In FIG. 2b, an embodiment of a decoder 24 according to the invention is shown in a block diagram. The received signal 54 includes encoding parameters representing main signal information and sub-signal information, and is output to a demultiplexer 56 that separates the first input signal and the second input signal. The first input signal corresponds to the main signal encoding parameter p _mono and is provided to the main signal decoding unit 64. In the conventional method, the encoding parameter p _mono representing the main signal is obtained by decoding the main signal x ″ _mono decoded so as to be as similar as possible to the main signal x _mono (FIG. 2 a) of the encoder 14 (FIG. 2 a). Used to generate.

Similarly, the second input signal corresponding to the sub signal is supplied to the sub signal decoding unit 60. Here, the encoding parameter p _side representing the sub-signal is used to recover the decoded sub-signal x ″ _side . In some embodiments, the encoding procedure is as indicated by arrow 65. , Use information about the main signal x " _mono .

The decoded main signal x ″ _mono and the decoded sub signal x ″ _side are supplied to the adder 70. Adder 70 outputs an output signal that is an indication of the original signal on channel a. Similarly, the difference supplied by subtractor 68 outputs an output signal that is an indication of the original signal on channel b. These channel signals can be post-processed in the post-processing unit 74 in accordance with a prior art signal processing procedure. Finally, channel signals a and b are output from decoder signal outputs 26A and 26B.

  As described in the overview, encoding is basically performed frame by frame. A frame has audio samples for a predetermined time. In the lower part of FIG. 3a, a frame SF2 of duration L is shown. Audio samples within the unhatched part are encoded together. The preceding and subsequent samples are encoded in other frames. By dividing the sample into frames, in some cases, some discontinuities will occur at the frame boundaries. If the sound changes, the encoding parameter also changes. The change of the encoding parameter is basically at each frame boundary. This causes an auditory error. One way to compensate for this to some extent is to encode based not only on the samples to be encoded, but also on samples near the boundaries of the frame, as indicated by the hatched portion. By doing so, the transition between different frames becomes more flexible. Alternatively, or as a complement to it, interpolation techniques may be used to reduce abnormal noise caused by frame boundaries. However, each of these processes requires a large amount of additional calculation, and depending on the encoding method, it may be difficult to realize with any resource.

  From this point of view, it is desirable to use a frame that is as long as possible. The reason is that the number of frame boundaries is reduced. In general, the coding efficiency is increased and the required transmission bit rate is minimized. However, when the frame length is increased, there arises a problem that abnormal sound or ghost sound due to pre-echo is perceived.

  Instead, using shorter frames such as SF1 or even SF0 with durations of L / 2 and L / 4, respectively, can reduce coding efficiency and increase the transmission bit rate, and Those skilled in the art will appreciate that the problem of boundary noise increases. However, if the frame length is shortened, for example, an adverse effect due to abnormal sound due to ghost sound or pre-echo is reduced. As short a frame length as possible should be used so that coding errors are minimized as much as possible.

  According to the present invention, the audio audibility quality is improved by using the frame length for encoding the sub-signal depending on the current signal content. Since the influence of the frame length on the audio audibility quality differs depending on the nature of the sound to be encoded, it can be improved by making the nature of the signal itself affect the frame length used. Since the encoding of the main signal is not the object of the present invention, a detailed description is omitted. However, the frame length used for the main signal may or may not be equal to the frame length used for the sub-signal.

  In some cases, it may be beneficial to encode the sub-signal using a relatively long frame because of the small temporal variation. This is the case for recordings with a large amount of diffuse sound field, such as recordings of concerts. For frames such as stereo speech, a short frame may be preferred. Which frame length is preferred can be determined in two basic ways.

  One embodiment of the sub-signal encoder 30 according to the present invention is shown in FIG. 3b, where a closed loop decision is used. Here, a basic encoded frame of length L is used. Several encoding schemes 81 featuring a set 80 of subframes 90 are provided. Each set 80 of subframes 90 has one or more subframes 90 of equal or different lengths. However, the total length of the set 80 of subframes 90 is always equal to the basic encoded frame length L. Referring to FIG. 3b, the top encoding scheme features a set of subframes comprising only one subframe of length L. The next set of subframes comprises two frames of length L / 2. The third set comprises two frames of length L / 4, followed by L / 2 frames.

The signal x _side supplied to the sub signal encoding unit 30 is encoded by all the encoding methods 81. In the uppermost encoding scheme, the entire basic encoding frame is encoded. However, in other encoding schemes, the signal x _side is encoded separately in each subframe. The result of each encoding method is supplied to the selector 85. The fidelity measuring means 83 calculates a fidelity measure for each encoded signal. The fidelity measure is an objective quality value and is preferably a signal to noise measure or a weighted signal to noise ratio. The fidelity measures associated with the respective coding schemes are compared, and the switching means 87 is controlled in accordance with the result, so that the coding parameters representing the sub-signals from the coding schemes that obtain the optimum fidelity measures are subtracted. The output signal p _side is selected from the signal encoding unit 30.

  It is preferable to try all combinations of possible frame lengths and select a set of subframes that provide the best objective quality, such as signal-to-noise ratio.

  In the present embodiment, the length of the subframe to be used is selected according to the following equation.

l _sf ＝ l _f / 2 ⁿ

Here, l _sf is the length of the subframe, l _f is the length of the encoded frame, and n is an integer. In the present embodiment, n is selected between 0 and 3. However, any frame length can be used as long as the total length of the set is kept constant.

In FIG. 3c, another embodiment of the sub-signal encoding unit 30 according to the present invention is shown. Here, the determination of the frame length is an open loop determination based on signal statistics. That is, the spectral characteristics of the sub-signal are used as a basis for determining which encoding scheme is used. As before, different coding schemes characterized by different sets of subframes are available. However, in this embodiment, the selector 85 is arranged before the actual encoding. The input sub signal x _side is input to the selector 85 and the signal analysis unit 84. The result of the analysis becomes the input of the switch 86 and only one of the encoding methods 81 is used. The output from the encoding method is also the output signal p _side from the sub-signal encoding unit 30.

  The advantage of an open loop decision is that only one actual encoding is performed. However, the disadvantage is that the analysis of signal characteristics can be very complex and it is difficult to predict possible behavior before the switch 86 can be given the appropriate choices. There is a possibility. The signal analyzer 84 needs to perform many speech statistical analyses. Any minor change in the coding scheme can disrupt the statistical behavior.

  By using closed loop selection (FIG. 3b), it is possible to change the coding scheme without changing the subsequent units. On the other hand, when many encoding methods are investigated, the amount of calculation increases.

  The advantage of the variable frame length encoding of the sub-signal as described above is that a fine time resolution and a coarse frequency resolution can be selected, while a coarse time resolution and a fine frequency resolution can be selected. The above embodiment can maintain a stereo image optimally.

  There are also several requirements for the actual coding used in different coding schemes. Specifically, when closed loop selection is used, the computational resources to perform some degree of simultaneous encoding must be large. As the encoding process becomes more complex, higher computational power is required. Furthermore, a low bit rate in transmission is also preferred.

  The method disclosed in US Pat. No. 5,434,948 uses a filtered monaural signal (main signal) to approximate a sub-signal or difference signal. Filter parameters can be optimized over time. A filter parameter representing the encoding of the side signal is then transmitted. In one embodiment, a secondary residual signal is also transmitted. In many cases, such an approach can be used as sub-signal coding within the scope of the present invention. However, this approach has several drawbacks. Filter coefficient quantization and any sub-residual signals often require relatively high bit rates for transmission. The reason is that the order of the filter needs to be high to provide an accurate side signal estimate. The evaluation of the filter itself can be a problem, especially for a large amount of transient music. The evaluation error provides a modified sub-signal that can be larger than the uncorrected signal. This increases the demand for bit rate. Furthermore, if a new set of filter coefficients is calculated every N samples, the filter coefficients need to be interpolated so that the transition from one set of filter coefficients to another is smooth as described above. Interpolating filter coefficients is a complex task, and interpolation errors appear in large error sub-signals, which requires higher bit rates for different error signal encoders.

  A means of avoiding the need for interpolation is to rely on backward adaptive analysis, updating the filter coefficients for each sample. In order for this to work well, the bit rate of the remaining encoders needs to be quite high. This is therefore not a good alternative to low bit rate stereo coding.

  For example, there are cases where the monaural signal and the difference signal are almost uncorrelated, which is very common in music. Thus, the evaluation of the filter, the additional risk of simply exacerbating the situation for different error signal encoders, is very cumbersome.

  The solution according to US Pat. No. 5,434,948 can work very well when the filter coefficients change very slowly over time, such as in conference phone systems. In the case of music signals, this technique does not work very well. The reason is that it is necessary to change the filter very quickly in order to track the stereo image. This means that subframe lengths of very different sizes need to be used, which means that the number of combinations tested increases rapidly. This means that the requirement to calculate all possible coding schemes is impractically high.

  Thus, in the preferred embodiment, the sub-signal encoding reduces the redundancy between the monaural signal and the sub-signal by using a simple balance factor instead of a predictive filter that consumes a complex bit rate. Based on the concept of reducing. Therefore, the residual of this operation is encoded. The magnitude of such a residual is relatively small and a high bit rate is not necessary for its transmission. This concept is suitable for practical combination with the variable frame set approach described above. The reason is that the computational complexity is mild.

  The use of a balance factor combined with a variable frame length approach eliminates the need for complex interpolation and related problems that may arise from interpolation. Furthermore, by using a simple balance factor instead of a complex filter, there are fewer problems associated with the evaluation. This is because the influence of possible evaluation errors on the balance coefficient is smaller. The preferred solution can regenerate the panned signal and diffuse sound field with good quality and with limited bit rate requirements and computational resources.

FIG. 4 shows a preferred embodiment of a stereo encoder according to the invention. This embodiment is very similar to the embodiment shown in FIG. 2a, but details of the sub-signal encoding unit 30 are revealed. The encoder 14 of this embodiment does not have a preprocessing unit, and the input signal is directly supplied to the adder 34 and the subtracter 36. The monaural signal x _mono is multiplied by the balance coefficient g _sm in the multiplexer 33. The multiplied monaural signal is subtracted in the subtractor 35 from the sub-signal x _side , that is, essentially the difference between the two channels to generate a sub-residual signal. The balance coefficient g _sm is determined by the optimization unit 37 based on the contents of the monaural signal and the sub signal so that the sub residual signal is minimized according to the quality standard. The quality standard is preferably based on the Least-Mean-Square (LMS) method. The sub residual signal is encoded in the sub residual encoder 39 according to an arbitrary encoding procedure. The sub-residual encoder 39 is preferably a low bit rate transform encoder or a CELP (Codebook Driven Linear Prediction) encoder. Therefore, the coding parameter p _side representing the _side signal includes the coding parameter p _{side residual} representing the _{side residual} signal and the optimization balance coefficient 49.

In the embodiment of FIG. 4, the monaural signal 42 used for the synthesis of the sub-signal is the target signal x _mono of the monaural encoder 38. As mentioned above (in conjunction with FIG. 2a), the local composite signal of the mono encoder 38 can also be used. In the latter case, the overall coding delay may increase and the complexity of the sub-signal calculation may also increase. On the other hand, the quality may be good. The reason is that it is possible to repair the coding error that occurs in the monaural encoder.

  Mathematically, the basic coding scheme can be described as follows: The stereo left channel and right channel are denoted as channel signals a and b, respectively. The channel signal is converted into a monaural signal by addition, and is sub-signaled by subtraction. The operation is described as follows.

x _mono (n) = 0.5 (a (n) + b (n))
x _side (n) = 0.5 (a (n)-b (n))

It is beneficial to scale the x _mono and x _side signals by 2. Here we show that there are other _ways to generate x _mono and x _side . For example, the following can be used:

x _mono (n) = γa (n) + (1-γ) b (n)
x _side (n) = γa (n)-(1-γ) b (n)
0 ≦ γ ≦ 1.0

  In the block of input signals, the modified or sub-residual signal is calculated according to the following equation:

x _{side residual} (n) = x _side (n)-f (x _mono , x _side ) x _mono (n)

In the above equation, f (x _mono , x _side ) is a balance coefficient function to be removed from the sub signal as much as possible based on a block of N samples from the sub signal and the monaural signal, that is, a subframe. That is, the balance factor is used to minimize the secondary residual signal. In the special case where the secondary residual signal is minimized by the LMS, this is equivalent to minimizing the energy of the secondary residual signal x _{side residual} .

In the special case described above, f (x _mono , x _side ) is expressed as follows.

However, x _side is a sub signal and x _mono is a monaural signal. Note that the function is based on a block that starts at "frame start" and ends at "frame end".

Frequency domain weighting can be added to the calculation of the balance factor. This is done by _convolving the x _side signal and the x _mono signal with the impulse response of the weighting filter. It is then possible to move to a frequency range where it is difficult to hear the evaluation error. This is called auditory weighting.

A quantized version of the balance coefficient value given by the function f (x _mono , x _side ) is sent to the decoder. If a modified sub-signal is generated, it is preferable to consider quantization. Thus, the following equation is achieved:

Q _g (..) is a quantization function applied to the balance coefficient given by the function f (x _mono , x _side ). The balance factor is transmitted on the transmission channel. In a normal left / right pan signal, the balance coefficient is limited to [−1.0, 1.0]. On the other hand, if the channels are out of phase with respect to each other, the balance factor may extend beyond these limits.

  As an optional means of stabilizing the stereo image, the balance factor can be limited if the normalized cross-correlation between the monaural signal and the sub-signal is insufficient as given by:

ただし、

However,

  These situations occur very often, for example in classical music or studio music with a large amount of diffused sound, and in some cases the a and b channels can sometimes be almost erased from each other when a mono signal is generated There is sex. The effect on the balance factor is that it can jump quickly, which causes stereo images to interfere. The above correction alleviates this problem.

  The filter-based approach of US Pat. No. 5,434,948 has similar problems, but in this case the solution is not so simple.

E _s is the encoding function of the sub residual signal (e.g., transcoder), and when E _m is the encoding function of the mono signal is decoded in the decoder side a "signal and b" signals Can be written as follows (assuming γ = 0.5):

  One important advantage derived from calculating the balance factor for each frame is that the use of interpolation is avoided. Instead, typically, as described above, frame processing is performed on overlapping frames.

  Coding principles that use balance factors work particularly well in the case of musical signals that usually require rapid changes to track a stereo image.

  Recently, multi-channel coding has become common. An example is 5.1 channel surround sound of a DVD movie. In this case, the channel is configured as front left, front center, front right, back left, back right, and subwoofer. In FIG. 5, an embodiment of an encoder for encoding three front channels configured to take advantage of inter-channel redundancy according to the present invention is shown.

Three channel signals L, C, R are supplied to the three inputs 16A-C, and the monaural signal x _mono is generated by the sum of all three signals. A center signal encoding unit 130 that receives the center signal x _center is added. The monaural signal 42 in this embodiment is an encoded and decoded monaural signal x ″ _mono and is multiplied by a balance factor g _Q in the multiplexer 133. In the subtractor 135, the multiplied monaural signal is To generate the center residual signal, it is subtracted from the center signal x _centre, and the balance factor g _Q is optimized by the optimization unit 137 to minimize the center residual signal according to the quality criterion. The central residual signal is encoded in a central residual encoder 139 according to an arbitrary encoder procedure, which is a low bit rate transform encoding. preferably a vessel or CELP encoder. Accordingly, coding parameters p _center representing the center signal, the center residual signal and optimization balance coefficient . Center residual signal and the scaled mono signal including the encoded parameters p _{center residual_prediction_flag} representing the 49 are summed in adder 235 to produce a modified center signal 142 being compensated for encoding errors.

The sub signal x _side , that is, the difference between the left L channel and the right R channel is supplied to the sub signal encoding unit 30 as in the above-described embodiment. However, here, the optimization unit 37 also depends on the modified center signal 142 supplied by the center signal encoding unit 130. Accordingly, the sub residual signal is generated in the subtractor 35 as an optimal linear combination of the monaural signal 42, the modified center signal 142, and the sub signal.

  The variable frame length concept described above can be applied to either or both of the side signal and the center signal.

FIG. 6 shows a decoder suitable for receiving an encoded audio signal from the encoder of FIG. The received signal 54 is divided into a coding parameter p _mono representing the main signal, a coding parameter p _center representing the center signal, and a coding parameter p _side representing the sub signal. In the decoder 64, the encoding parameter p _mono representing the main signal is used to generate the main signal x ″ _mono . In the decoder 160, the encoding parameter p _center representing the center signal is used as the main signal. Used to generate x " _center based on x" _mono . In decoder 60, the coding parameter _pside representing the _side signal is decoded and based on the main signal x " _mono and the center signal x" _center . Thus, the sub signal x " _side is generated.

  The procedure can be expressed mathematically as follows:

Input signals x _left , x _right , x _center are combined into a mono channel according to the following.

  α, β, and χ are set to 1.0 in the remaining sections for simplicity, but can be set to arbitrary values. The values of α, β, χ can be constant or dependent on the signal content to emphasize one or two channels in order to achieve optimal quality.

  The normalized cross-correlation between the monaural signal and the center signal is calculated as follows:

ただし、

However,

x _center is the center signal and x _mono is the monaural signal. The monaural signal comes from the monaural target signal, but it is equally possible to use local synthesis of a monaural encoder.

  The center residual signal is encoded as follows.

Q _g (..) is a quantization function applied to the balance coefficient. The balance factor is transmitted on the transmission channel.

If E _c is the encoding function of the center residual signal (eg, transform encoder) and E _m is the encoding function of the monaural signal, the decoded x " _center signal on the decoder side is Can be described as:

  The sub residual signal to be encoded is as follows.

Here, g _Qsm and g _Qsc are quantized values of the parameters g _sm and g _sc that minimize the following equation.

η can be equal to, for example, 2 for LMS minimization of errors. The g _sm and g _sc parameters can be quantized jointly or separately.

If E _s is the coding function of the sub-residual signal, the decoded x " _left and x" _right channel signals are given as follows:

  One of the most troublesome noises is the pre-echo effect. Such abnormal noise is shown in FIGS. Assume a signal component having a time progression as shown by curve 100. At the start from t0, there is no signal component in the audio sample. At time t between t1 and t2, the signal component suddenly appears. If the signal component is encoded using a frame length of t 2 -t 1, the appearance of the signal component becomes “blurred” over the entire frame, as shown in curve 101. When decoding is performed on curve 101, the signal component appears at a time Δt before the intended appearance of the signal component, and “pre-echo” is perceived.

  Pre-echo noise is accentuated when long encoded frames are used. By using a short frame, the noise is somewhat reduced. Another scheme that addresses the above-mentioned pre-echo problem takes advantage of the fact that monaural signals are available on both the encoder side and the decoder side. This makes it possible to scale the sub-signal according to the isoenergy curve of the monaural signal. On the decoder side, inverse scaling is performed, so the pre-echo problem can be somewhat mitigated.

  The isoenergy curve of the monaural signal is calculated over the frame as follows:

  However, w (n) is a window function. The simplest window function is a rectangular window, but other window types such as a Hamming window may be more preferred.

  The sub-residual signal is then scaled as follows:

  In a more general form, the above equation can be written as:

  In the above equation, f (...) is a monotone continuous function. In the decoder, an isoenergy curve is calculated for the decoded monaural signal and applied to the decoded sub-signal as follows:

  In a sense, this isoenergetic curve scaling is an alternative to the use of shorter frame lengths, so this concept is particularly well suited to be combined with the variable frame length concept, as further described above. Yes. Encoding by having some coding schemes that apply iso-energy curve scaling, some coding schemes that do not apply, and some coding schemes that apply iso-energy curve scaling only during certain subframes It is possible to supply a more flexible set of schemes. In FIG. 8, an embodiment of a signal encoding unit 30 according to the present invention is shown. Here, the different encoding schemes 81 have hatched subframes 91 that represent encodings that apply equal energy curve scaling and unhatched subframes 92 that represent encoding procedures that do not apply equal energy curve scaling. . In this way, not only subframes of different lengths but also combinations of subframes of different coding principles can be used. In this illustrative example, the application of isoenergy curve scaling is different between different coding schemes. In the more general case, any coding principle can be combined with the variable length concept in a similar manner.

  The set of encoding schemes in FIG. 8 includes schemes that deal with pre-echo abnormal noise and the like using different schemes. In some schemes, longer subframes with pre-echo minimization according to the isoenergy curve principle are used. In other schemes, shorter subframes without isoenergy curve scaling are used. Depending on the signal content, one of the alternatives may be more advantageous. For very severe pre-echo, an encoding scheme that uses short subframes with iso-energy curve scaling may be necessary.

  The proposed solution can be used in the whole frequency band or in one or more individual subbands. The use of subbands can be applied for both main and subsignals or separately for one of them. In the preferred embodiment, the sub-signal is divided in several frequency bands. This is simply because it is easier to remove the possible redundancy in the isolated frequency band than in the entire frequency band. This is particularly important when encoding musical signals having a large amount of spectral content.

  One possible use is to encode the frequency band below a predetermined threshold in the manner described above. The predetermined threshold is preferably 2 kHz, or even more preferably 1 kHz. For the rest of the frequency range of interest, other additional frequency bands can be encoded in the manner described above or in a completely different manner.

  One motivation for using the above method, preferably for low frequencies, is because the diffuse sound field generally has a low energy content at high frequencies. The reason for this is that sound absorption usually increases with frequency. Also, diffuse sound field components seem to play a less important role for human hearing systems at higher frequencies. It is therefore beneficial to use this solution at low frequencies (below 1 kHz or 2 kHz) and rely on other more bit efficient coding schemes at higher frequencies. By applying this scheme only at low frequencies, the bit rate is greatly saved because the required bit rate for the proposed method is proportional to the required bandwidth. In most cases, the mono encoder can encode the entire frequency band, while the proposed sub-signal encoding is in the lower part of the frequency band, as shown schematically in FIG. It is suggested that only be implemented. Reference numeral 301 refers to the coding scheme of the sub-signal according to the invention, reference numeral 302 refers to any other coding scheme of the sub-signal, and reference numeral 303 refers to the coding scheme of the sub-signal.

  There is also the possibility to use the proposed method for several individual frequency bands.

  In FIG. 10, the main steps of an embodiment of the encoding method according to the invention are shown as a flowchart. The procedure begins at step 200. In step 210, the main signal inferred from the polyphonic signal is encoded. In step 212, an encoding scheme with different length and / or order subframes is provided. The sub-signal inferred from the polyphonic signal in step 214 is encoded with a selected encoding scheme that depends at least in part on the actual signal content of the polyphonic signal present. The procedure ends at step 299.

  In FIG. 11, the main steps of an embodiment of the encoding method according to the invention are shown as a flowchart. The procedure begins at step 200. In step 220, the received encoded main signal is decoded. In step 222, a decoding scheme with different lengths and / or orders of subframes is provided. The received sub-signal is decoded at step 224 according to the selected coding scheme. In step 226, the decoded main and sub signals are combined into a polyphonic signal. The procedure ends at step 299.

  The above-described embodiments should be understood as some examples of the invention. Those skilled in the art will be able to make various modifications, combinations and changes to the above-described embodiments without departing from the scope of the present invention. In particular, different partial solutions of different embodiments can be combined in other configurations where technically possible. In any case, the scope of the present invention is defined by the appended claims.

1 is a block diagram illustrating a system for transmitting a polyphonic signal. It is a block diagram of the encoder in a transmitter. It is a block diagram of the decoder in a receiver. It is a figure which shows the encoding of the frame of a different length. It is a block diagram of an embodiment of a sub-signal encoding unit according to the present invention. It is a block diagram of an embodiment of a sub-signal encoding unit according to the present invention. FIG. 3 is a block diagram of an embodiment of an encoder that uses balance factor encoding of sub-signals. 2 is a block diagram of an embodiment of an encoder for a multi-signal system. FIG. FIG. 6 is a block diagram of one embodiment of a decoder suitable for decoding signals from the apparatus of FIG. It is a figure which shows pre-echo noise. It is a figure which shows pre-echo noise. FIG. 6 is a block diagram of an embodiment of a sub-signal encoding unit according to the present invention that uses different encoding principles for different subframes. FIG. 4 illustrates using different encoding principles for different frequency subbands. 4 is a flowchart of basic steps of an embodiment of an encoding method according to the present invention; 4 is a flowchart of basic steps of an embodiment of a decoding method according to the present invention;

Claims (26)

Generating a first output signal (p _mono ), which is an encoding parameter representing a main signal, based on at least the signals of the first channel and the second channel (a, b; L, R);
A second output signal (p _side ) that is an encoding parameter representing a sub-signal based on at least the signals of the first channel and the second channel (a, b; L, R) in the encoded frame (80). A method for encoding a polyphonic signal, comprising:
The encoded frame (80) is constituted by a set of subframes (90) including at least one subframe (90), and the sum of the lengths of the subframes (90) is the length of the encoded frame (80). Further comprising the step (212) of providing at least two encoding schemes (81) equal to the length;
The step (214) of generating the second output signal (p _side ) includes the step of selecting an encoding scheme (81) that depends at least in part on the signal content of the current sub-signal (x _side ),
The second output signal (p _side ) is encoded separately for each subframe (90) of a set of selected subframes (90). The step (214) of generating the second output signal (p _side ) includes:
A first linear combination of signals of at least the first and second channels (a, b; L, R) within every subframe (90) of each of at least two sets of subframes (90) Separately generating encoding parameters representing the _side signal (x _side );
Calculating a fidelity measure for each of the at least two encoding schemes (81);
Selecting a signal encoded as the encoding parameter (p _side ) representing the sub-signal from the encoding scheme (81) having the optimal fidelity measure. The method according to 1.   The method of claim 2, wherein the fidelity measure is based on a signal-to-noise measure. If l _f is the length of the coded frame (80) and n is an integer, the length l _sf of the subframe (90) is
l _sf ＝ l _f / 2 ⁿ
The method according to claim 1, wherein the method is represented by:   The method of claim 4, wherein n is less than a predetermined value.   The method according to claim 5, characterized in that the at least two encoding schemes (81) have a pre-permutation of the length of the subframe (90). The step (210) of generating a coding parameter (p _mono ) representing the main signal is:
Generating a main signal (x _mono ) as a second linear combination of signals of at least the first and second channels (a, b; L, R);
Encoding the main signal into an encoding parameter (p _mono ) representing the main signal;
The step of encoding the sub-signal includes:
Generating a _{side residual} signal (x _{side residual} ) as a difference between the _side signal and the main signal (x _mono ) scaled by a balance factor (g _sm );
Encoding the sub-residual signal and the balance factor (g _sm ) into the encoding parameter (p _side ) representing the sub-signal,
The method according to any of the _preceding claims, wherein the balance factor (g _sm ) is determined as a factor that minimizes the sub-residual signal according to a quality criterion.   The method according to claim 7, wherein the quality standard is based on Least-Mean-Square (LMS). 9. The method according to claim 1, wherein the step of encoding the sub-signal further includes a step of scaling the sub-signal (x _side ) according to an isoenergy curve of the main signal (x _mono ). The method of crab. The method according to claim 9, characterized in that the scaling of the sub-signal (x _side ) is divided by a factor that is a monotone continuous function of the iso-energy curve of the main signal (x _mono ).   The method of claim 10, wherein the monotonic continuous function is a square root function. Wherein such energy curve E _c of said main signal (x _mono) The method of claim 10 or 11, characterized in that it is computed over a sub-frame according to the following equation.

Here, L is an arbitrary coefficient, n is a sum index, m is a sample in the subframe, and w (n) is a window function.   The method of claim 12, wherein the window function is a rectangular window function.   The method of claim 12, wherein the window function is a Hamming window function. 15. The method according to claim 1, wherein the at least two encoding schemes (81) comprise different encoding principles of the sub-signal (x _side ). At least a first encoding scheme of the at least two encoding schemes (81) includes a first encoding principle of the sub-signal (x _side ) of all subframes (90), and the at least two encoding schemes The at least second encoding scheme of the encoding scheme (81) includes a second encoding principle of the sub-signal (x _side ) of all subframes (90). Method. At least one of the at least two encoding schemes (81) includes the first encoding principle for the sub-signal (x _side ) of one subframe and the sub-signal ( _17. Method according to claim 15 or 16, characterized in that it comprises the second encoding principle for _xside ). The step (214) of generating the second output signal (p _side ) includes:
Analyzing a spectral characteristic of a _side signal (x _side ) that is a first linear combination of signals of at least the first and second channels (a, b; L, R);
Selecting a set of subframes (90) based on the analyzed spectral characteristics;
The method of claim 1, comprising: encoding each of said sub-signals (x _side ) of all sub-frames (90) in said selected set of sub-frames (90). The second output signal (p _side) the step of generating (214) A method according to any one of claims 1 to 18, characterized in that it is applied in a limited frequency band. The second output signal step (214) for generating a (p _side) The method of claim 19, characterized in that it is applied only for frequencies below 2 kHz. 21. The method according to claim 20, wherein the step (214) of generating the second output signal (p _side ) is applied only for frequencies below 1 kHz.   22. A method as claimed in any preceding claim, wherein the polyphonic signal represents a musical sound signal. Decoding a coding parameter (p _mono ) representing the main signal (220);
A step (224) of decoding a coding parameter (p _side ) representing a sub-signal in the coded frame (80);
Combining at least the decoded main signal (x " _mono ) and the decoded sub signal (x" _side ) into signals of at least first and second channels (a, b; L, R) (226) A method for decoding a polyphonic signal comprising:
The encoded frame (80) is composed of a set of subframes (90) including at least one subframe (90), and the sum of the lengths of the subframes (90) is the length of the encoded frame (80). Providing (222) providing at least two encoding schemes (81) equal to the length;
The step (224) of decoding the coding parameter (p _side ) representing the sub-signal includes the step of representing the sub-signal in the sub-frame (90) of any of the at least two coding schemes (81). A method comprising the step of decoding an encoding parameter (p _side ). Input means (16; 16A-C) for inputting a polyphonic signal (a, b; L, R, C) having at least first and second channels (a, b; L, R);
Means (38) for generating a first output signal (p _mono ), which is a coding parameter representing a main signal, based on at least the signals of the first and second channels (a, b; L, R);
In the encoded frame (80), a second output signal (p _side ), which is an encoding parameter representing a sub-signal, is generated based on at least the signals of the first and second channels (a, b; L, R). Means for generating (30);
An encoding device (14) having output means (52),
The encoded frame (80) is constituted by a set of subframes (90) including at least one subframe (90), and the sum of the lengths of the subframes (90) is the length of the encoded frame (80). Means for providing at least two encoding schemes (81) equal to length,
The means (30) for generating the second output signal (p _side ) includes means (86; 87) for selecting an encoding method depending at least in part on the signal content of the current sub signal (x _side ). ,
The encoding apparatus further comprising means for separately encoding the sub-signal (x _side ) in each of the selected sub-frames (90) of the encoding method. An input means (54) for inputting a coding parameter (p _mono ) representing a main signal and a coding parameter (p _side ) representing a sub signal;
Means (64) for decoding the encoding parameter (p _mono ) representing the main signal;
Means (60) for decoding the encoding parameter (p _side ) representing the sub-signal in an encoded frame (80);
Means for combining at least the decoded main signal (x " _mono ) and the decoded sub signal (x" _side ) into signals of at least first and second channels (a, b; L, R) (68, 70) and
A decoding device (24) having output means (26; 26A-C),
The means (60) for decoding the encoding parameter (p _side ) representing the sub-signal is:
The encoded frame (80) is constituted by a set of subframes (90) including at least one subframe (90), and the sum of the lengths of the subframes (90) is the length of the encoded frame (80). Means for providing at least two encoding schemes (81) equal to length;
Means for separately decoding the coding parameter (p _side ) representing the sub-signal in any sub-frame (90) of the at least two coding schemes (81). apparatus. Encoding device (14) according to claim 24;
An audio system (1) comprising at least one of the decoding device (24) according to claim 25.

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