Title of Invention	METHOD AND APPARATUS FOR SYNTHESIZING A PLURALITY OF AUDIO CHANNELS
Abstract	The invention relates to Method for synthesizing a plurality of audio channels, comprising retrieving from an audio stream at least one sum signal representing a sum of source signals, retrieving from the audio stream statistical information about one or more source signals, receiving from the audio stream, or determining locally, parameters describing an output audio format and source mixing parameters, computing output mixer parameters from the received statistical information, the parameters describing an output audio format, and the source mixing parameters, synthesizing the plurality of audio channels from the at least one sum signal based on the computed output mixer parameters.

Title of Invention

METHOD AND APPARATUS FOR SYNTHESIZING A PLURALITY OF AUDIO CHANNELS

Abstract

The invention relates to Method for synthesizing a plurality of audio channels, comprising retrieving from an audio stream at least one sum signal representing a sum of source signals, retrieving from the audio stream statistical information about one or more source signals, receiving from the audio stream, or determining locally, parameters describing an output audio format and source mixing parameters, computing output mixer parameters from the received statistical information, the parameters describing an output audio format, and the source mixing parameters, synthesizing the plurality of audio channels from the at least one sum signal based on the computed output mixer parameters.

Full Text	PARAMETRIC JOINT-CODING OF AUDIO SOURCES 1. INTRODUCTION In a general coding problem, we have a number of (mono) source signals s,(n) (1 ' >M) and a scene description vector S(n), where n is the time index. The scene description vector contains parameters such as (virtual) source positions, source widths, and acoustic parameters such as (virtual) room parameters. The scene description may be time-invariant or may be changing over time. The source signals and scene description are coded and transmitted to a decoder. The coded source signals, s\(n) are successively mixed as a function of the scene description, S{n), to generate wavefield synthesis, multi-channel, or stereo signals as a function of the scene description vector. The decoder output signals are denoted x\(n) (0 >i Note that the scene description vector S(n) may not be transmitted but may be determined at the decoder. In this document, the term "stereo audio signal" always refers to two-channel stereo audio signals. ISO/IEC MPEG-4 addresses the described coding scenario. It defines the scene description and uses for each ("natural") source signal a separate mono audio coder, e.g. an AAC audio coder. However, when a complex scene with many sources is to be mixed, the bitrate becomes high, i.e. the bitrate scales up with the number of sources. Coding one source signal with high quality requires about 60 - 90 kb/s. Previously, we addressed a special case of the described coding problem [1][2] with a scheme denoted Binaural Cue Coding (BCC) for Flexible Rendering. By transmitting only the sum of the given source signals plus low bitrate side information, low bitrate is achieved. However, the source signals can not be recovered at the decoder and the scheme was limited to stereo and multi-channel surround signal generation. Also, only simplistic mixing was used, based on amplitude and delay panning. Thus, the direction of sources could be controlled but no other auditory spatial image attributes. Another limitation of this scheme was its limited audio quality. Especially, a decrease in audio quality as the number of source signals is increased. The document [1], (Binaural Cue Coding, Parametric Stereo, MP3 Surround, MPEG Surround) covers the case where N audio channels are encoded and N audio channels with similar cues then the original audio channels are decoded. The transmitted side information includes inter-channel cue parameters relating to differences between the input channels. The channels of stereo and multi-channel audio signals contain mixes of audio sources signals and are thus different in nature than pure audio source signals. Stereo and multi-channel audio signals are mixed such that when played back over an appropriate playback system, the listener will perceive an auditory spatial image ("sound stage") as captured by the recording setup or designed by the recording engineer during mixing. A number of schemes for joint-coding for the channels of a stereo or multi-channel audio signal have been proposed previously. SUMMARY OF THE INVENTION The aim of the invention is to provide a method to transmit a plurality of source signals while using a minimum bandwidth. In most of known methods, the playback format (e.g. stereo, 5.1) is predefined and has a direct influence on the coding scenario. The audio stream on the decoder side should use only this predefined playback format, therefore binding the user to a predefined playback scenario (e.g. stereo). The proposed invention encodes N audio source signals, typically not channels of a stereo or multi-channel signals, but independent signals, such as different speech or instrument signals. The transmitted side information includes statistical parameters relating to the input audio source signals. The proposed invention decodes M audio channels with different cues than the original audio source signals. These different cues are either implicitly synthesized by applying a mixer to the received sum signal. The mixer is controlled as a function of the received statistical source information and the received (or locally determined) audio format parameters and mixing parameters. Alternatively, these different cues are explicitly computed as a function of the received statistical source information and the received (or locally determined) audio format parameters and mixing parameters. These computed cues are used to control a prior art decoder (Binaural Cue Coding, Parametric Stereo, MPEG Surround) for synthesizing the output channels given the received sum signal. The proposed scheme for joint-coding of audio source signals is the first of its kind. It is designed for joint-coding of audio source signals. Audio source signals are usually mono audio signals which are not suitable for playback over a stereo or multi-channel audio system. For brevity, in the following, audio source signals are often denoted source signals. Audio source signals first need to be mixed to stereo, multi-channel, or wavefield synthesis audio signals prior to playback. An audio source signal can be a single instrument or talker, or the sum of a number of instruments and talkers. Another type of audio source signal is a mono audio signal captured with a spot microphone during a concert. Often audio source signals are stored on multi-track recorders or in harddisk recording systems. The claimed scheme for joint-coding of audio source signals, is based on only transmitting the sum of the audio source signals, or a weighted sum of the source signals. Optionally, weighted summation can be carried out with different weights in different subbands and the weights may be adapted in time. Summation with equalization, as described in Chapter 3.3.2 in [1], may also be applied. In the following, when we refer to the sum or sum signal, we always mean a signal generate by (1) or generated as described. In addition to the sum signal, side information is transmitted. The sum and the side information represent the outputted audio stream. Optionally, the sum signal is coded using a conventional mono audio coder. This stream can be stored in a file (CD, DVD, Harddisk) or broadcasted to the receiver. The side information represents the statistical properties of the source signals which are the most important factors determining the perceptual spatial cues of the mixer output signals. It will be shown that these properties are temporally evolving spectral envelopes and auto-correlation functions. About 3 kb/s of side information is transmitted per source signal. At the receiver, source signals si(n) (1 statistical properties approximating the corresponding properties of the original source signals and the sum signal. BRIEF DESCRIPTION OF THE DRAWINGS The invention will be better understood thanks to the attached Figures in which: - figure 1 shows a scheme in which the transmission of each source signal is made independently for further processing, - figure 2 shows a number of sources transmitted as sum signal plus side information, - figure 3 shows a block diagram of a Binaural Cue Coding (BCC) scheme, - figure 4 shows a mixer for generating stereo signals based on several source signals, - figure 5 shows the dependence between ICTD, ICLD and ICC and the source signal subband power, - figure 6 shows the process of side information generation, - figure 7 shows the process of estimating the LPC parameters of each source signal, - figure 8 shows the process of re-creating the source signals from a sum signal, - figure 9 shows an alternative scheme for the generation of each signal from the sum signal, - figure 10 shows a mixer for generating stereo signals based on the sum signal, - figure 11 shows an amplitude panning algorithm preventing that the source levels depends on the mixing parameters, - figure 12 shows a loudspeaker array of a wavefield synthesis playback system, - figure 13 shows how to recover an estimate of the source signals at the receiver by processing the downmix of the transmitted channels, - figure 14 shows how to recover an estimate of the source signals at the receiver by processing the transmitted channels. II. DEFINITIONS, NOTATION, AND VARIABLES X The following notation and variables are used in this paper: n time index; i audio channel or source index; d delay index; M number of encoder input source signals; N number of decoder output channels; X,(n) mixed original source signals; xf (n) mixed decoder output signals; s,(«) encoder input source signals; si,-(«) transmitted source signals also called pseudo-source signals; s(ri) transmitted sum signal; yt(ri) L-channel audio signal; (audio signal to be re-mixed); s(k) one subband signal of st(n) (similarly defined for other signals); E{5. («)} short- time estimate of st (n) (similarly defined for other signals); ICLD inter-channel level difference; ICTD inter-channel time difference; ICC inter-channel coherence; AL(n) estimated subband ICLD; x(n) estimated subband ICTD; c(n) estimated subband ICC; pt(n) relative source subband power; a,, bi mixer scale factors; Cj, d\ mixer delays; Mj, x{ri) mixer level and time difference; G, mixer source gain; III. JOINT-CODING OF AUDIO SOURCE SIGnalS First, Binaural Cue Coding (BCC), a parametric multi-channel audio coding tech- nique, is described. Then it is shown that with the same insight as BCC is based on one can devise an algorithm for jointly coding the source signals for a coding scenario. A. Binaural Cue Coding (BCC) A BCC scheme [1][2] for multi-channel audio coding is shown in the figure bellow. The input multi-channel audio signal is downmixed to a single channel. As opposed to coding and transmitting information about all channel waveforms, only the downmixed signal is coded (with a conventional mono audio coder) and transmitted. Additionally, perceptually motivated "audio channel differences" are estimated between the original audio channels and also transmitted to the decoder. The decoder generates its output channels such that the audio channel differences approximate the corresponding audio channel differences of the original audio signal. Summing localization implies that perceptually relevant audio channel differences for a loudspeaker signal channel pair are the inter-channel time difference (ICTD) and inter-channel level difference (ICLD). ICTD and ICLD can be related to the perceived direction of auditory events. Other auditory spatial image attributes, such as apparent source width and listener envelopment, can be related to interaural coherence (IC). For loudspeaker pairs in the front or back of a listener, the interaural coherence is often directly related to the inter-channel coherence (ICC) which is thus considered as third audio channel difference measure by BCC. ICTD, ICLD, and ICC are estimated in subbands as a function of time. Both, the spectral and temporal resolution that is used, are motivated by perception. B. Parametric joint-coding of audio sources A BCC decoder is able to generate a multi-channel audio signal with any auditory spatial image by taking a mono signal and synthesizing at regular time intervals a single specific ICTD, ICLD, and ICC cue per subband and channel pair. The good performance of BCC schemes for a wide range of audio material [see 1] implies that the perceived auditory spatial image is largely determined by the ICTD, ICLD, and ICC. Therefore, as opposed to requiring "clean" source signals st(ri) as mixer input in Figure 1, we just require pseudo-source signals si,.(n) with the property that they result in similar ICTD, ICLD, and ICC at the mixer output as for the case of supplying the real source signals to the mixer. There are three goals for the generation of si,.(n) • If si,(n) are supplied to a mixer, the mixer output channels will have approximately the same spatial cues (ICLD, ICTD, ICC) as if s,-(n) were supplied to the mixer. • si,(n) are to be generated with as little as possible information about the original source signals s(n) (because the goal is to have low bitrate side information). • si.(n)are generated from the transmitted sum signal s(ri) such that a minimum amount of signal distortion is introduced. For deriving the proposed scheme we are considering a stereo mixer (M = 2). A further simplification over the general case is that only amplitude and delay panning are applied for mixing. If the discrete source signals were available at the decoder, a stereo signal would be mixed as shown in Figure 4, i.e. In this case, the scene description vector S(n) contains just source directions which determine the mixing parameters, where T is the transpose of a vector. Note that for the mixing parameters we ignored the time index for convenience of notation. More convenient parameters for controlling the mixer are time and level difference, Ti and ^L which are related to a,, bi c,, and di by where Gi is a source gain factor in dB. In the following, we are computing ICTD, ICLD, and ICC of the stereo mixer output as a function of the input source signals si(n). The obtained expressions will give indication which source signal properties determine ICTD, ICLD, and ICC (together with the mixing parameters). si,-(n) are then generated such that the identified source signal properties approximate the corresponding properties of the original source signals. B.1 ICTD, ICLD, and ICC of the mixer output The cues are estimated in subbands and as a function of time. In the following it is assumed that the source signals si{n) are zero mean and mutually independent. A pair of subband signals of the mixer output (2) is denoted x,(n) and x2{ri). Note that for simplicity of notation we are using the same time index n for time-domain and subband-domain signals. Also, no subband index is used and the described analysis/processing is applied to each subband independently. The subband power of the two mixer output signals is where si(n) 's one subband signal of source s,(n) and E{.} denotes short-time expectation, e.g. where K determines the length of the moving average. Note that the subband power values E {s22 (n)} represent for each source signal the spectral envelope as a function of time. The ICLD, AL(n), is For estimating ICTD and ICC the normalized cross-correlation function, is estimated. The ICC, c(n), is computed according to For the computation of the ICTD, J{n), the location of the highest peak on the delay axis is computed, Now the question is, how can the normalized cross-correlation function be computed as a function of the mixing parameters. Together with (2), (8) can be written as which is equivalent to where the normalized auto-correlation function φ(n,e) is and Ti= di- ci. Note that for computing (12) given (11) it has been assumed that the signals are wide sense stationary within the considered range of delays, i.e. A numerical example for two source signals, illustrating the dependence between ICTD, ICLD, and ICC and the source subband power, is shown in Figure 5. The top, middle, and bottom panel of Figure 5 show ΔL(n), T(n), and c(n), respectively, as a function of the ratio of the subband power of the two source signals, for different mixing parameters (4) ΔL1 , ΔL2 , 7i and T2. Note that when only one source has power in the subband (a = 0 or a = 1), then the computed ΔL (n) and T(n) are equΔL to the mixing parameters (ΔL,, ΔL2, 71 , T2). B.2 Necessary side information The ICLD (7) depends on the mixing parameters (a,, si>,, ch di) and on the short-time subband power of the sources, The normΔlized subband cross- correlation function ®(n,d) (12), that is needed for ICTD (10) and ICC (9) computation, depends on and additionally on the normΔlized subband auto-correlation function, Oj(n, e) (13), for each source signal. The maximum of φ(n,d) lies within the range mirii{7i} = di - c, the corresponding range for which the source signal subband property Oj(n, e) (13) is needed is Since 0the ICTD, ICLD, and ICC cues depend on the source signal subband properties and φi(n, e) in the range (14), in principle these source signal subband properties need to be transmitted as side information. We assume that any other kind of mixer (e.g. mixer with effects, wavefield synthesis mixer/convoluter, etc.) has similar properties and thus this side information is useful ΔLso when other mixers than the described one are used. For reducing the amount of side information, one could store a set of predefined auto-correlation functions in the decoder and only transmit indices for choosing the ones most closely matching the source signal properties. A first version of our ΔLgorithm assumes that within the range (14) ®,{n, e) = 1 and thus (12) is computed using only the subband power vΔLues (6) as side information. The data shown in Figure 5 has been computed assuming In order to reduce the amount of side information, the relative dynamic range of the source signals is limited. At each time, for each subband the power of the strongest source is selected. We found it sufficient to lower bound the corresponding subband power of ΔLl the other sources at a vΔLue 24 dB lower than the strongest subband power. Thus the dynamic range of the quantizer can be limited to 24 dB. Assuming that the source signals are independent, the decoder can compute the sum of the subband power of ΔLl sources as . Thus, in principle it is enough to transmit to the decoder only the subband power vΔLues of M - 1 sources, while the subband power of the remaining source can be computed locally. Given this idea, the side information rate can be slightly reduced by transmitting the subband power of sources with indices 2 s / Note that dynamic range limiting as described previously is carried out prior to (15). As an ΔLternative, the subband power vΔLues could be normΔlized relative to the sum signal subband power, as opposed to normΔlization relative to one source's subband power (15). For a sampling frequency of 44.1 kHz we use 20 subbands and transmit for each subband ApX") (2 s / s M) about every 12 ms. 20 subbands corresponds to hΔLf the spectraL resolution of the auditory system (one subband is two "critical bandwidths" wide). InformΔL experiments indicate that only slight improvement is achieved by using more subbands than 20, e.g. 40 subbands. The number of subbands and subband bandwidths are chosen according to the time and frequency resolution of the auditory system. A low quΔlity implementation of the scheme requires at least three subbands (low, medium, high frequencies). According to a particular embodiment, the subbands have different bandwidths, subbands at lower frequencies have smΔLler bandwidth than subbands at higher frequencies. The relative power vΔLues are quantized with a scheme similar to the ICLD quantizer described in [2], resulting in a bitrate of approximately 3(M -1) kb/s. Figure 6 illustrates the process of side information generation (corresponds to the "Side infor- mation generation" block in Figure 2). Side information rate can be additionally reduced by analyzing the activity for each source signal and only transmitting the side information associated with the source if it is active. As opposed to transmitting the subband power vΔLues E {s] (n)} as statistical information, other information representing the spectraL envelopes of the source signals could be transmitted. For example, linear predictive coding (LPC) parameters could be transmitted, or corresponding other parameters such as lattice filter parameters or line spectraL pair (LSP) parameters. The process of estimating the LPC parameters of each source signal is illustrated in Figure 7. B.3 Computing st(n) Figure 8 illustrates the process that is used to re-create the source signals, given the sum signal (1). This process is part of the "Synthesis" block in Figure 2. The individuΔL source signals are recovered by scaling each subband of the sum signal with g,{/i) and by applying a de-correlation filter with impulse response hi (n), where * is the linear convolution operator and is computed with the side information by As de-correlation filters h{n), complementary comb filters, ΔLl-pass filters, delays, or filters with random impulse responses may be used. The goΔL for the de-correlation process is to reduce correlation between the signals while not modifying how the individuΔL waveforms are perceived. Different de-correlation techniques cause different artifacts. Complementary comb filters cause coloration. ΔLl the described techniques are spreading the energy of transients in time causing artifacts such as "pre-echoes". Given their potentiΔL for artifacts, de-correlation techniques should be applied as little as possible. The next section describes techniques and strategies which require less de-correlation processing than simple generation of independent signals si,(n). An ΔLternative scheme for generation of the signals i,.(n) is shown in Figure 9. First the spectrum of s(n) is flattened by means of computing the linear prediction error e(n). Then, given the LPC filters estimated at the encoder, fh the corresponding ΔLl- pole filters are computed as the inverse z-transform of The resulting ΔLl-pole filters, fn represent the spectraL envelope of the source signals. If other side information than LPC parameters is transmitted, the LPC parameters first need to be computed as a function of the side information. As in the other scheme, de-correlation filters h, are used for making the source signals independent. IV. IMPLEMENTATIONS CONSIDERING PRACTIcal CONSTRAINTS In the first part of this section, an implementation example is given, using a BCC synthesis scheme as a stereo or multi-channel mixer. This is particularly interesting since such a BCC type synthesis scheme is part of an upcoming ISO/IEC MPEG standard, denoted "spatiΔL audio coding". The source signals si,(n) are not explicitly computed in this case, resulting in reduced computational complexity. ΔLso, this scheme offers the potentiΔL for better audio quΔlity since effectively less de- correlation is needed than for the case when the source signals s.(ri) are explicitly computed. The second part of this section discusses issues when the proposed scheme is applied with any mixer and no de-correlation processing is applied at ΔLl. Such a scheme has a lower complexity than a scheme with de-correlation processing, but may have other drawbacks as will be discussed. IdeΔLly, one would like to apply de-correlation processing such that the generated si,(n)can be considered independent. However, since de-correlation processing is problematic in terms of introducing artifacts, one would like to apply de-correlation processing as little as possible. The third part of this section discusses how the amount of problematic de-correlation processing can be reduced while getting benefits as if the generated i, (n) were independent. A. Implementation without explicit computation of s,.(n) Mixing is directly applied to the transmitted sum signal (1) without explicit computation of si.(n). A BCC synthesis scheme is used for this purpose. In the following, we are considering the stereo case, but ΔLl the described principles can be applied for generation of multi-channel audio signals as well. A stereo BCC synthesis scheme (or a "parametric stereo" scheme), applied for processing the sum signal (1), is shown in Figure 10. Desired would be that the BCC synthesis scheme generates a signal that is perceived similarly as the output signal of a mixer as shown in Figure 4. This is so, when ICTD, ICLD, and ICC between the BCC synthesis scheme output channels are similar as the corresponding cues appearing between the mixer output (4) signal channels. The same side information as for the previously described more generaL scheme is used, ΔLlowing the decoder to compute the short-time subband power vΔLues E of the sources. Given the gain factors gi and 0/2 in Figure 10 are computed as such that the output subband power and ICLD (7) are the same as for the mixer in Figure 4. The ICTD T{n) is computed according to (10), determining the delays D1 and D2 in Figure 10, The ICC c(n) is computed according to (9) determining the de-correlation processing in Figure 10. De-correlation processing (ICC synthesis) is described in [1]. The advantages of applying de-correlation processing to the mixer output channels compared to applying it for generating independent si,(n) are: • UsuΔLly the number of source signals M is larger than the number of audio output channels N. Thus, the number of independent audio channels that need to be generated is smΔLler when de-correlating the N output channels as opposed to de-correlating the M source signals. • Often the N audio output channels are correlated (ICC > 0) and less de- correlation processing can be applied than would be needed for generating independent M or N channels. Due to less de-correlation processing better audio quΔlity is expected. Best audio quΔlity is expected when the mixer parameters are constrained such that In this case, the power of each source in the transmitted sum signal (1) is the same as the power of the same source in the mixed decoder output signal. The decoder output signal (Figure 10) is the same as if the mixer output signal (Figure 4) were encoded and decoded by a BCC encoder/decoder in this case. Thus, ΔLso similar quΔlity can be expected. The decoder can not only determine the direction at which each source is to appear but ΔLso the gain of each source can be varied. The gain is increased by choosing and decreased by choosing B. Using no de-correlation processing The restriction of the previously described technique is that mixing is carried out with a BCC synthesis scheme. One could imagine implementing not only ICTD, ICLD, and ICC synthesis but additionally effects processing within the BCC synthesis. However, it may be desired that existing mixers and effects processors can be used. This ΔLso includes wavefield synthesis mixers (often denoted "convoluters"). For using existing mixers and effects processors, the i,(n)are computed explicitly and used as if they were the original source signals. When applying no de-correlation processing (hj(n) = 8(n) in (16)) good audio quΔlity can ΔLso be achieved. It is a compromise between artifacts introduced due to de- correlation processing and artifacts due to the fact that the source signals st(n)are correlated. When no de-correlation processing is used the resulting auditory spatiΔL image may suffer from instability [1]. But the mixer may introduce itself some de- correlation when reverberators or other effects are used and thus there is less need for de-correlation processing. If si,(n)are generated without de-correlation processing, the level of the sources depends on the direction to which they are mixed relative to the other sources. By replacing amplitude panning ΔLgorithms in existing mixers with an ΔLgorithm compensating for this level dependence, the negative effect of loudness dependence on mixing parameters can be circumvented. A level compensating amplitude ΔLgorithm is shown in Figure 11 which aims to compensate the source level dependence on mixing parameters. Given the gain factors of a conventional amplitude panning ΔLgorithm (e.g. Figure 4), a, and bh the weights in Figure 11, a, and bt, are computed by Note that ai and b{ are computed such that the output subband power is the same as if st(n) were independent in each subband. c. Reducing the amount of de-correlation processing As mentioned previously, the generation of independent si,(n) is problematic. Here strategies are described for applying less de-correlation processing, while effectively getting a similar effect as if the i,(n)were independent. Consider for example a wavefield synthesis system as shown in Figure 12. The desired virtuΔL source positions for Si, s2, ..., Se {M = 6) are indicated. A strategy for computing si,-(n) (16) without generating M fully independent signals is: 1. Generate groups of source indices corresponding to sources close to each other. For example in Figure 8 these could be: {1}, {2, 5}, {3}, and {4, 6}. 2. At each time in each subband select the source index of the strongest source, Apply no de-correlation processing for the source indices part of the group containing /'max, i.e. h{n) = 8(n). 3. For each other group choose the same h{n) within the group. The described ΔLgorithm modifies the strongest signal components least. Additionally, the number of different hj(n) that are used are reduced. This is an advantage because de-correlation is easier the less independent channels need to be generated. The described technique is ΔLso applicable when stereo or multi-channel audio signals are mixed. V. ScalABIliTY IN TERMS OF QUΔliTY AND BITRATE The proposed scheme transmits only the sum of ΔLl source signals, which can be coded with a conventional mono audio coder. When no mono backwards compatibility is needed and capacity is available for transmission/storage of more than one audio waveform, the proposed scheme can be scaled for use with more than one transmission channel. This is implemented by generating severaL sum signals with different subsets of the given source signals, i.e. to each subset of source signals the proposed coding scheme is applied individuΔLly. Audio quΔlity is expected to improve as the number of transmitted audio channels is increased because less independent channels have to be generated by de-correlation from each transmitted channel (compared to the case of one transmitted channel). VI. BACKWARDS COMPATIBIliTY TO EXISTING STEREO AND SURROUND AUDIO FORMATS Consider the following audio delivery scenario. A consumer obtains a maximum quΔlity stereo or multi-channel surround signal (e.g. by means of an audio CD, DVD, or on-line music store, etc.). The goΔL is to optionally deliver to the consumer the flexibility to generate a custom mix of the obtained audio content, without compromising standard stereo/surround playback quΔlity. This is implemented by delivering to the consumer (e.g. as optional buying option in an on-line music store) a bit stream of side information which ΔLlows computation of i.(n)as a function of the given stereo or multi-channel audio signal. The consumer's mixing ΔLgorithm is then applied to the si,(n) In the following, two possibilities for computing si,-(n), given stereo or multi-channel audio signals, are described. A. Estimating the sum of the source signals at the receiver The most straight forward way of using the proposed coding scheme with a stereo or multi-channel audio transmission is illustrated in Figure 13, where y{n) (1 s i the L channels of the given stereo or multi-channel audio signal. The sum signal of the sources is estimated by downmixing the transmitted channels to a single audio channel. Downmixing is carried out by means of computing the sum of the channels y{n) (1 si i si L) or more sophisticated techniques may be applied. For best performance, it is recommended that the level of the source signals is adapted prior to estimation (6) such that the power ratio between the source signals approximates the power ratio with which the sources are contained in the given stereo or multi-channel signal. In this case, the downmix of the transmitted channels is a relatively good estimate of the sum of the sources (1) (or a scaled version thereof). An automated process may be used to adjust the level of the encoder source signal inputs Sj(n) prior to computation of the side information. This process adaptively in time estimates the level at which each source signal is contained in the given stereo or multi-channel signal. Prior to side information computation, the level of each source signal is then adaptively in time adjusted such that it is equΔL to the level at which the source is contained in the stereo or multi-channel audio signal. B. Using the transmitted channels individuΔLly Figure 14 shows a different implementation of the proposed scheme with stereo or multi-channel surround signal transmission. Here, the transmitted channels are not downmixed, but used individuΔLly for generation of the st(n). Most generaLly, the subband signals of s((n) are computed by where w,(ri) are weights determining specific linear combinations of the transmitted channels' subbands. The linear combinations are chosen such that the si,•(n) are ΔLready as much decorrelated as possible. Thus, no or only a smΔLl amount of de- correlation processing needs to be applied, which is favorable as discussed earlier. VII. APPliCATIONS ΔLready previously we mentioned a number of applications for the proposed coding schemes. Here, we summarize these and mention a few more applications. A. Audio coding for mixing Whenever audio source signals need to be stored or transmitted prior to mixing them to stereo, multi-channel, or wavefield synthesis audio signals, the proposed scheme can be applied. With prior art, a mono audio coder would be applied to each source signal independently, resulting in a bitrate which scales with the number of sources. The proposed coding scheme can encode a high number of audio source signals with a single mono audio coder plus relatively low bitrate side information. As described in Section V, the audio quΔlity can be improved by using more than one transmitted channel, if the memory/capacity to do so is available. B. Re-mixing with meta-data As described in Section VI, existing stereo and multi-channel audio signals can be re- mixed with the help of additional side information (i.e. "meta-data"). As opposed to only selling optimized stereo and multi-channel mixed audio content, meta data can be sold ΔLlowing a user to re-mix his stereo and multi-channel music. This can for example ΔLso be used for attenuating the vocals in a song for karaoke, or for attenuating specific instruments for playing an instrument ΔLong the music. Even if storage would not be an issue, the described scheme would be very attractive for enabling custom mixing of music. That is, because it is likely that the music industry would never be willing to give away the multi-track recordings. There is too much a danger for abuse. The proposed scheme enables re-mixing capability without giving away the multi-track recordings. Furthermore, as soon as stereo or multi-channel signals are re-mixed a certain degree of quΔlity reduction occurs, making illegΔL distribution of re-mixes less attractive. c. Stereo/multi-channel to wavefield synthesis conversion Another application for the scheme described in Section VI is described in the following. The stereo and multi-channel (e.g. 5.1 surround) audio accompanying moving pictures can be extended for wavefield synthesis rendering by adding side information. For example, Dolby AC-3 (audio on DVD) can be extended for 5.1 backwards compatibly coding audio for wavefield synthesis systems, i.e. DVDs play back 5.1 surround sound on conventional legacy players and wavefield synthesis sound on a new generation of players supporting processing of the side information. VIII. SUBJECTIVE EVΔLUATIONS We implemented a reΔL-time decoder of the ΔLgorithms proposed in Section IV-A and IV-B. An FFT-based STFT filterbank is used. A 1024-point FFT and a STFT window size of 768 (with zero padding) are used. The spectraL coefficients are grouped together such that each group represents signal with a bandwidth of two times the equivΔLent rectangular bandwidth (ERB). InformΔL listening reveΔLed that the audio quΔlity did not notably improve when choosing higher frequency resolution. A lower frequency resolution is favorable since it results in less parameters to be transmitted. For each source, the amplitude/delay panning and gain can be adjusted individuΔLly. The ΔLgorithm was used for coding of severaL multi-track audio recordings with 12 - 14 tracks. The decoder ΔLlows 5.1 surround mixing using a vector base amplitude panning (VBAP) mixer. Direction and gain of each source signal can be adjusted. The software ΔLlows on the-fly switching between mixing the coded source signal and mixing the original discrete source signals. CasuΔL listening usuΔLly reveΔLs no or little difference between mixing the coded or original source signals if for each source a gain G, of zero dB is used. The more the source gains are varied the more artifacts occur. Slight amplification and attenuation of the sources (e.g. up to ± 6 dB) still sounds good. A critical scenario is when ΔLl the sources are mixed to one side and only a single source to the other opposite side. In this case the audio quΔlity may be reduced, depending on the specific mixing and source signals. IX. CONCLUSIONS A coding scheme for joint-coding of audio source signals, e.g. the channels of a multi-track recording, was proposed. The goΔL is not to code the source signal waveforms with high quΔlity, in which case joint-coding would give minimΔL coding gain since the audio sources are usuΔLly independent. The goΔL is that when the coded source signals are mixed a high quΔlity audio signal is obtained. By considering statistical properties of the source signals, the properties of mixing schemes, and spatiΔL hearing it was shown that significant coding gain improvement is achieved by jointly coding the source signals. The coding gain improvement is due to the fact that only one audio waveform is transmitted. Additionally side information, representing the statistical properties of the source signals which are the relevant factors determining the spatiΔL perception of the final mixed signal, are transmitted. The side information rate is about 3 kbs per source signal. Any mixer can be applied with the coded source signals, e.g. stereo, multi-channel, or wavefield synthesis mixers. It is straight forward to scale the proposed scheme for higher bitrate and quΔlity by means of transmitting more than one audio channel. Furthermore, a variation of the scheme was proposed which ΔLlows re-mixing of the given stereo or multi-channel audio signal (and even changing of the audio format, e.g. stereo to multi-channel or wavefield synthesis). The applications of the proposed scheme are manifold. For example MPEG-4 could be extended with the proposed scheme to reduce bitrate when more than one "naturaL audio object" (source signal) needs to be transmitted. ΔLso, the proposed scheme offers compact representation of content for wavefield synthesis systems. As mentioned, existing stereo or multi-channel signals could be complemented with side information to ΔLlow that the user re-mixes the signals to his liking. REFERENCES [1] C. FΔLler, Parametric Coding of SpatiΔL Audio, Ph.D. thesis, Swiss FederaL Institute of Technology Lausanne (EPFL), 2004, Ph.D. Thesis No. 3062. [2] C. FΔLler and F. Baumgarte, "BinauraL Cue Coding - Part II: Schemes and applications," IEEE Trans, on Speech and Audio Proa, vol. 11, no. 6, Nov. 2003. We Claim: 1. Method for synthesizing a plurality of audio channels, comprising: retrieving from an audio stream at least one sum signal representing a sum of source signals, retrieving from the audio stream statistical information about one or more source signals, receiving from the audio stream, or determining locally, parameters describing an output audio format and source mixing parameters, computing output mixer parameters from the received statistical information, the parameters describing an output audio format, and the source mixing parameters, synthesizing the plurality of audio channels from the at least one sum signal based on the computed output mixer parameters. 2. Method as claimed in claim 1, wherein the statistical information represent spectraL envelopes of the source signals, or the spectraL envelopes of the one or more audio source signals comprise lattice filter parameters or line spectraL parameters or in which the statistical information represent a relative power as a function of frequency and time of the plurality of source signals. 3. Method as claimed in claim 1, wherein the step of computing the output mixer parameters comprises computing the cues of the plurality of audio channels and computing the output mixer parameters using the calculated cues of the plurality of audio channels. 4. Method as claimed in claim 1, wherein the audio channels are synthesized in a subband domain of a filterbank. 5. Method as claimed in claim 4, wherein a number and bandwidths of the subband domain are determined according to a spectraL and temporaL resolution of an human auditory system. 6. Method as claimed in claim 4, wherein a number of subbands is between 3 and 40. 7. Method as claimed in claim 4, wherein subbands in the subband domain have different bandwidths, wherein subbands at lower frequencies have smΔLler bandwidths than subbands at higher frequencies. 8. Method as claimed in claim 4, wherein a short time Fourier transform (STFT) based filterbank is used and spectraL coefficients are combined to form groups of spectraL coefficients such that each group of spectraL coefficients forms a subband. 9. Method as claimed in claim 1, wherein the statistical information ΔLso comprises auto-correlation functions. 10. Method as claimed in claim 2, wherein spectraL envelopes are represented as linear predictive coding (LPC) parameters. 11. Method as claimed in claim 3, wherein the computed cues are level difference, time difference, or coherence fcr different frequencies and time instants. 12.Apparatus for synthesizing a plurality of audio channels, wherein the apparatus is operative for: retrieving from an audio stream at least one; sum signal representing a sum of source signals, retrieving from the audio stream statistical information about one or more source signals, receiving from the audio stream, or determining locally, parameters describing an output audio format and source mixing parameters, computing output mixer parameters from the received statistical information, the parameters describing an output audio format, and the source mixing parameters, synthesizing the plurality of audio channels from the at least one sum signal based on the computed output mixer parameters. ABSTRACT TITLE: Method and Apparatus for synthesizing a plurality of audio channels The invention relates to Method for synthesizing a plurality of audio channels, comprising retrieving from an audio stream at least one sum signal representing a sum of source signals, retrieving from the audio stream statistical information about one or more source signals, receiving from the audio stream, or determining locally, parameters describing an output audio format and source mixing parameters, computing output mixer parameters from the received statistical information, the parameters describing an output audio format, and the source mixing parameters, synthesizing the plurality of audio channels from the at least one sum signal based on the computed output mixer parameters.

Full Text

PARAMETRIC JOINT-CODING OF AUDIO SOURCES
1. INTRODUCTION
In a general coding problem, we have a number of (mono) source signals s,(n) (1 '
>M) and a scene description vector S(n), where n is the time index. The scene
description vector contains parameters such as (virtual) source positions, source
widths, and acoustic parameters such as (virtual) room parameters. The scene
description may be time-invariant or may be changing over time. The source signals
and scene description are coded and transmitted to a decoder. The coded source
signals, s\(n) are successively mixed as a function of the scene description, S{n), to
generate wavefield synthesis, multi-channel, or stereo signals as a function of the
scene description vector. The decoder output signals are denoted x\(n) (0 >i Note that the scene description vector S(n) may not be transmitted but may be
determined at the decoder. In this document, the term "stereo audio signal" always
refers to two-channel stereo audio signals.
ISO/IEC MPEG-4 addresses the described coding scenario. It defines the scene
description and uses for each ("natural") source signal a separate mono audio coder,
e.g. an AAC audio coder. However, when a complex scene with many sources is to
be mixed, the bitrate becomes high, i.e. the bitrate scales up with the number of
sources. Coding one source signal with high quality requires about 60 - 90 kb/s.
Previously, we addressed a special case of the described coding problem [1][2] with
a scheme denoted Binaural Cue Coding (BCC) for Flexible Rendering. By
transmitting only the sum of the given source signals plus low bitrate side information,
low bitrate is achieved. However, the source signals can not be recovered at the
decoder and the scheme was limited to stereo and multi-channel surround signal
generation. Also, only simplistic mixing was used, based on amplitude and delay
panning. Thus, the direction of sources could be controlled but no other auditory
spatial image attributes. Another limitation of this scheme was its limited audio
quality. Especially, a decrease in audio quality as the number of source signals is
increased.

The document [1], (Binaural Cue Coding, Parametric Stereo, MP3 Surround, MPEG
Surround) covers the case where N audio channels are encoded and N audio
channels with similar cues then the original audio channels are decoded. The
transmitted side information includes inter-channel cue parameters relating to
differences between the input channels.
The channels of stereo and multi-channel audio signals contain mixes of audio
sources signals and are thus different in nature than pure audio source signals.
Stereo and multi-channel audio signals are mixed such that when played back over
an appropriate playback system, the listener will perceive an auditory spatial image
("sound stage") as captured by the recording setup or designed by the recording
engineer during mixing. A number of schemes for joint-coding for the channels of a
stereo or multi-channel audio signal have been proposed previously.
SUMMARY OF THE INVENTION
The aim of the invention is to provide a method to transmit a plurality of source
signals while using a minimum bandwidth. In most of known methods, the playback
format (e.g. stereo, 5.1) is predefined and has a direct influence on the coding
scenario. The audio stream on the decoder side should use only this predefined
playback format, therefore binding the user to a predefined playback scenario (e.g.
stereo).
The proposed invention encodes N audio source signals, typically not channels of a
stereo or multi-channel signals, but independent signals, such as different speech or
instrument signals. The transmitted side information includes statistical parameters
relating to the input audio source signals.
The proposed invention decodes M audio channels with different cues than
the original audio source signals. These different cues are either implicitly
synthesized by applying a mixer to the received sum signal. The mixer is
controlled as a function of the received statistical source information and the
received (or locally determined) audio format parameters and mixing
parameters. Alternatively, these different cues are explicitly computed as a
function of the received statistical source information and the received (or

locally determined) audio format parameters and mixing parameters. These
computed cues are used to control a prior art decoder (Binaural Cue Coding,
Parametric Stereo, MPEG Surround) for synthesizing the output channels
given the received sum signal.
The proposed scheme for joint-coding of audio source signals is the first of its kind. It
is designed for joint-coding of audio source signals. Audio source signals are usually
mono audio signals which are not suitable for playback over a stereo or multi-channel
audio system. For brevity, in the following, audio source signals are often denoted
source signals.
Audio source signals first need to be mixed to stereo, multi-channel, or wavefield
synthesis audio signals prior to playback. An audio source signal can be a single
instrument or talker, or the sum of a number of instruments and talkers. Another type
of audio source signal is a mono audio signal captured with a spot microphone during
a concert. Often audio source signals are stored on multi-track recorders or in
harddisk recording systems.
The claimed scheme for joint-coding of audio source signals, is based on only
transmitting the sum of the audio source signals,

or a weighted sum of the source signals. Optionally, weighted summation can be
carried out with different weights in different subbands and the weights may be
adapted in time. Summation with equalization, as described in Chapter 3.3.2 in [1],
may also be applied. In the following, when we refer to the sum or sum signal, we
always mean a signal generate by (1) or generated as described. In addition to the
sum signal, side information is transmitted. The sum and the side information
represent the outputted audio stream. Optionally, the sum signal is coded using a
conventional mono audio coder. This stream can be stored in a file (CD, DVD,
Harddisk) or broadcasted to the receiver. The side information represents the
statistical properties of the source signals which are the most important factors
determining the perceptual spatial cues of the mixer output signals. It will be shown

that these properties are temporally evolving spectral envelopes and auto-correlation
functions. About 3 kb/s of side information is transmitted per source signal. At the
receiver, source signals si(n) (1 statistical properties approximating the corresponding properties of the original
source signals and the sum signal.
BRIEF DESCRIPTION OF THE DRAWINGS
The invention will be better understood thanks to the attached Figures in which:
- figure 1 shows a scheme in which the transmission of each source signal is made
independently for further processing,
- figure 2 shows a number of sources transmitted as sum signal plus side
information,
- figure 3 shows a block diagram of a Binaural Cue Coding (BCC) scheme,
- figure 4 shows a mixer for generating stereo signals based on several source
signals,
- figure 5 shows the dependence between ICTD, ICLD and ICC and the source
signal subband power,
- figure 6 shows the process of side information generation,
- figure 7 shows the process of estimating the LPC parameters of each source signal,
- figure 8 shows the process of re-creating the source signals from a sum signal,
- figure 9 shows an alternative scheme for the generation of each signal from the sum
signal,
- figure 10 shows a mixer for generating stereo signals based on the sum signal,
- figure 11 shows an amplitude panning algorithm preventing that the source levels
depends on the mixing parameters,
- figure 12 shows a loudspeaker array of a wavefield synthesis playback system,
- figure 13 shows how to recover an estimate of the source signals at the receiver by
processing the downmix of the transmitted channels,

- figure 14 shows how to recover an estimate of the source signals at the receiver by
processing the transmitted channels.
II. DEFINITIONS, NOTATION, AND VARIABLES
X
The following notation and variables are used in this paper:
n time index;
i audio channel or source index;
d delay index;
M number of encoder input source signals;
N number of decoder output channels;
X,(n) mixed original source signals;
xf (n) mixed decoder output signals;
s,(«) encoder input source signals;
si,-(«) transmitted source signals also called pseudo-source signals;
s(ri) transmitted sum signal;
yt(ri) L-channel audio signal; (audio signal to be re-mixed);
s(k) one subband signal of st(n) (similarly defined for other signals);
E{5. («)} short- time estimate of st (n) (similarly defined for other
signals);
ICLD inter-channel level difference;
ICTD inter-channel time difference;
ICC inter-channel coherence;
AL(n) estimated subband ICLD;
x(n) estimated subband ICTD;
c(n) estimated subband ICC;
pt(n) relative source subband power;
a,, bi mixer scale factors;

Cj, d\ mixer delays;
Mj, x{ri) mixer level and time difference;
G, mixer source gain;
III. JOINT-CODING OF AUDIO SOURCE SIGnalS
First, Binaural Cue Coding (BCC), a parametric multi-channel audio coding tech-
nique, is described. Then it is shown that with the same insight as BCC is based on
one can devise an algorithm for jointly coding the source signals for a coding
scenario.
A. Binaural Cue Coding (BCC)
A BCC scheme [1][2] for multi-channel audio coding is shown in the figure bellow.
The input multi-channel audio signal is downmixed to a single channel. As opposed
to coding and transmitting information about all channel waveforms, only the
downmixed signal is coded (with a conventional mono audio coder) and transmitted.
Additionally, perceptually motivated "audio channel differences" are estimated
between the original audio channels and also transmitted to the decoder. The
decoder generates its output channels such that the audio channel differences
approximate the corresponding audio channel differences of the original audio signal.
Summing localization implies that perceptually relevant audio channel differences for
a loudspeaker signal channel pair are the inter-channel time difference (ICTD) and
inter-channel level difference (ICLD). ICTD and ICLD can be related to the perceived
direction of auditory events. Other auditory spatial image attributes, such as apparent
source width and listener envelopment, can be related to interaural coherence (IC).
For loudspeaker pairs in the front or back of a listener, the interaural coherence is
often directly related to the inter-channel coherence (ICC) which is thus considered
as third audio channel difference measure by BCC. ICTD, ICLD, and ICC are
estimated in subbands as a function of time. Both, the spectral and temporal
resolution that is used, are motivated by perception.

B. Parametric joint-coding of audio sources
A BCC decoder is able to generate a multi-channel audio signal with any auditory
spatial image by taking a mono signal and synthesizing at regular time intervals a
single specific ICTD, ICLD, and ICC cue per subband and channel pair. The good
performance of BCC schemes for a wide range of audio material [see 1] implies that
the perceived auditory spatial image is largely determined by the ICTD, ICLD, and
ICC. Therefore, as opposed to requiring "clean" source signals st(ri) as mixer input
in Figure 1, we just require pseudo-source signals si,.(n) with the property that they
result in similar ICTD, ICLD, and ICC at the mixer output as for the case of supplying
the real source signals to the mixer. There are three goals for the generation of si,.(n)
• If si,(n) are supplied to a mixer, the mixer output channels will have
approximately the same spatial cues (ICLD, ICTD, ICC) as if s,-(n) were
supplied to the mixer.
• si,(n) are to be generated with as little as possible information about the
original source signals s(n) (because the goal is to have low bitrate side
information).
• si.(n)are generated from the transmitted sum signal s(ri) such that a
minimum amount of signal distortion is introduced.
For deriving the proposed scheme we are considering a stereo mixer (M = 2). A
further simplification over the general case is that only amplitude and delay panning
are applied for mixing. If the discrete source signals were available at the decoder, a
stereo signal would be mixed as shown in Figure 4, i.e.

In this case, the scene description vector S(n) contains just source directions which
determine the mixing parameters,

where T is the transpose of a vector. Note that for the mixing parameters we ignored
the time index for convenience of notation.
More convenient parameters for controlling the mixer are time and level difference, Ti
and ^L which are related to a,, bi c,, and di by

where Gi is a source gain factor in dB.
In the following, we are computing ICTD, ICLD, and ICC of the stereo mixer output as
a function of the input source signals si(n). The obtained expressions will give
indication which source signal properties determine ICTD, ICLD, and ICC (together
with the mixing parameters). si,-(n) are then generated such that the identified source
signal properties approximate the corresponding properties of the original source
signals.
B.1 ICTD, ICLD, and ICC of the mixer output
The cues are estimated in subbands and as a function of time. In the following it is
assumed that the source signals si{n) are zero mean and mutually independent. A
pair of subband signals of the mixer output (2) is denoted x,(n) and x2{ri). Note that
for simplicity of notation we are using the same time index n for time-domain and
subband-domain signals. Also, no subband index is used and the described
analysis/processing is applied to each subband independently. The subband power
of the two mixer output signals is

where si(n) 's one subband signal of source s,(n) and E{.} denotes short-time
expectation, e.g.

where K determines the length of the moving average. Note that the subband power
values E {s22 (n)} represent for each source signal the spectral envelope as a
function of time. The ICLD, AL(n), is

For estimating ICTD and ICC the normalized cross-correlation function,

is estimated. The ICC, c(n), is computed according to

For the computation of the ICTD, J{n), the location of the highest peak on the delay
axis is computed,

Now the question is, how can the normalized cross-correlation function be computed
as a function of the mixing parameters. Together with (2), (8) can be written as

which is equivalent to

where the normalized auto-correlation function φ(n,e) is

and Ti= di- ci. Note that for computing (12) given (11) it has been assumed that the
signals are wide sense stationary within the considered range of delays, i.e.

A numerical example for two source signals, illustrating the dependence between
ICTD, ICLD, and ICC and the source subband power, is shown in Figure 5. The top,
middle, and bottom panel of Figure 5 show ΔL(n), T(n), and c(n), respectively, as a
function of the ratio of the subband power of the two source signals,
for different mixing parameters (4) ΔL1 , ΔL2 , 7i and T2.
Note that when only one source has power in the subband (a = 0 or a = 1), then the
computed ΔL (n) and T(n) are equΔL to the mixing parameters (ΔL,, ΔL2, 71 , T2).
B.2 Necessary side information
The ICLD (7) depends on the mixing parameters (a,, si>,, ch di) and on the short-time
subband power of the sources, The normΔlized subband cross-
correlation function ®(n,d) (12), that is needed for ICTD (10) and ICC (9)
computation, depends on and additionally on the normΔlized subband
auto-correlation function, Oj(n, e) (13), for each source signal. The maximum of
φ(n,d) lies within the range mirii{7i} = di - c, the corresponding range for which the source signal subband property Oj(n,
e) (13) is needed is

Since 0the ICTD, ICLD, and ICC cues depend on the source signal subband
properties and φi(n, e) in the range (14), in principle these source signal
subband properties need to be transmitted as side information. We assume that any
other kind of mixer (e.g. mixer with effects, wavefield synthesis mixer/convoluter, etc.)
has similar properties and thus this side information is useful ΔLso when other mixers
than the described one are used. For reducing the amount of side information, one
could store a set of predefined auto-correlation functions in the decoder and only
transmit indices for choosing the ones most closely matching the source signal
properties. A first version of our ΔLgorithm assumes that within the range (14) ®,{n, e)
= 1 and thus (12) is computed using only the subband power vΔLues (6) as side
information. The data shown in Figure 5 has been computed assuming In order to reduce the amount of side information, the relative dynamic range of the
source signals is limited. At each time, for each subband the power of the strongest
source is selected. We found it sufficient to lower bound the corresponding subband
power of ΔLl the other sources at a vΔLue 24 dB lower than the strongest subband
power. Thus the dynamic range of the quantizer can be limited to 24 dB.
Assuming that the source signals are independent, the decoder can compute the
sum of the subband power of ΔLl sources as . Thus, in principle it is
enough to transmit to the decoder only the subband power vΔLues of M - 1 sources,
while the subband power of the remaining source can be computed locally. Given this
idea, the side information rate can be slightly reduced by transmitting the subband
power of sources with indices 2 s /
Note that dynamic range limiting as described previously is carried out prior to (15).
As an ΔLternative, the subband power vΔLues could be normΔlized relative to the sum
signal subband power, as opposed to normΔlization relative to one source's subband
power (15). For a sampling frequency of 44.1 kHz we use 20 subbands and transmit
for each subband ApX") (2 s / s M) about every 12 ms. 20 subbands corresponds to
hΔLf the spectraL resolution of the auditory system (one subband is two "critical

bandwidths" wide). InformΔL experiments indicate that only slight improvement is
achieved by using more subbands than 20, e.g. 40 subbands. The number of
subbands and subband bandwidths are chosen according to the time and frequency
resolution of the auditory system. A low quΔlity implementation of the scheme
requires at least three subbands (low, medium, high frequencies).
According to a particular embodiment, the subbands have different bandwidths,
subbands at lower frequencies have smΔLler bandwidth than subbands at higher
frequencies.
The relative power vΔLues are quantized with a scheme similar to the ICLD quantizer
described in [2], resulting in a bitrate of approximately 3(M -1) kb/s. Figure 6
illustrates the process of side information generation (corresponds to the "Side infor-
mation generation" block in Figure 2).
Side information rate can be additionally reduced by analyzing the activity for each
source signal and only transmitting the side information associated with the source if
it is active.
As opposed to transmitting the subband power vΔLues E {s] (n)} as statistical
information, other information representing the spectraL envelopes of the source
signals could be transmitted. For example, linear predictive coding (LPC) parameters
could be transmitted, or corresponding other parameters such as lattice filter
parameters or line spectraL pair (LSP) parameters. The process of estimating the
LPC parameters of each source signal is illustrated in Figure 7.
B.3 Computing st(n)
Figure 8 illustrates the process that is used to re-create the source signals, given the
sum signal (1). This process is part of the "Synthesis" block in Figure 2. The
individuΔL source signals are recovered by scaling each subband of the sum signal
with g,{/i) and by applying a de-correlation filter with impulse response hi (n),

where * is the linear convolution operator and is computed with the side
information by

As de-correlation filters h{n), complementary comb filters, ΔLl-pass filters, delays, or
filters with random impulse responses may be used. The goΔL for the de-correlation
process is to reduce correlation between the signals while not modifying how the
individuΔL waveforms are perceived. Different de-correlation techniques cause
different artifacts. Complementary comb filters cause coloration. ΔLl the described
techniques are spreading the energy of transients in time causing artifacts such as
"pre-echoes". Given their potentiΔL for artifacts, de-correlation techniques should be
applied as little as possible. The next section describes techniques and strategies
which require less de-correlation processing than simple generation of independent
signals si,(n).
An ΔLternative scheme for generation of the signals i,.(n) is shown in Figure 9. First
the spectrum of s(n) is flattened by means of computing the linear prediction error
e(n). Then, given the LPC filters estimated at the encoder, fh the corresponding ΔLl-
pole filters are computed as the inverse z-transform of

The resulting ΔLl-pole filters, fn represent the spectraL envelope of the source
signals. If other side information than LPC parameters is transmitted, the LPC
parameters first need to be computed as a function of the side information. As in the
other scheme, de-correlation filters h, are used for making the source signals
independent.
IV. IMPLEMENTATIONS CONSIDERING PRACTIcal CONSTRAINTS

In the first part of this section, an implementation example is given, using a BCC
synthesis scheme as a stereo or multi-channel mixer. This is particularly interesting
since such a BCC type synthesis scheme is part of an upcoming ISO/IEC MPEG
standard, denoted "spatiΔL audio coding". The source signals si,(n) are not explicitly
computed in this case, resulting in reduced computational complexity. ΔLso, this
scheme offers the potentiΔL for better audio quΔlity since effectively less de-
correlation is needed than for the case when the source signals s.(ri) are explicitly
computed.
The second part of this section discusses issues when the proposed scheme is
applied with any mixer and no de-correlation processing is applied at ΔLl. Such a
scheme has a lower complexity than a scheme with de-correlation processing, but
may have other drawbacks as will be discussed.
IdeΔLly, one would like to apply de-correlation processing such that the generated
si,(n)can be considered independent. However, since de-correlation processing is
problematic in terms of introducing artifacts, one would like to apply de-correlation
processing as little as possible. The third part of this section discusses how the
amount of problematic de-correlation processing can be reduced while getting
benefits as if the generated i, (n) were independent.
A. Implementation without explicit computation of s,.(n)
Mixing is directly applied to the transmitted sum signal (1) without explicit
computation of si.(n). A BCC synthesis scheme is used for this purpose. In the
following, we are considering the stereo case, but ΔLl the described principles can be
applied for generation of multi-channel audio signals as well.
A stereo BCC synthesis scheme (or a "parametric stereo" scheme), applied for
processing the sum signal (1), is shown in Figure 10. Desired would be that the BCC
synthesis scheme generates a signal that is perceived similarly as the output signal
of a mixer as shown in Figure 4. This is so, when ICTD, ICLD, and ICC between the
BCC synthesis scheme output channels are similar as the corresponding cues
appearing between the mixer output (4) signal channels.

The same side information as for the previously described more generaL scheme is
used, ΔLlowing the decoder to compute the short-time subband power vΔLues E
of the sources. Given the gain factors gi and 0/2 in Figure 10 are
computed as

such that the output subband power and ICLD (7) are the same as for the mixer in
Figure 4. The ICTD T{n) is computed according to (10), determining the delays D1
and D2 in Figure 10,

The ICC c(n) is computed according to (9) determining the de-correlation processing
in Figure 10. De-correlation processing (ICC synthesis) is described in [1]. The
advantages of applying de-correlation processing to the mixer output channels
compared to applying it for generating independent si,(n) are:
• UsuΔLly the number of source signals M is larger than the number of audio output
channels N. Thus, the number of independent audio channels that need to be
generated is smΔLler when de-correlating the N output channels as opposed to
de-correlating the M source signals.
• Often the N audio output channels are correlated (ICC > 0) and less de-
correlation processing can be applied than would be needed for generating
independent M or N channels.
Due to less de-correlation processing better audio quΔlity is expected.
Best audio quΔlity is expected when the mixer parameters are constrained such that
In this case, the power of each source in the transmitted
sum signal (1) is the same as the power of the same source in the mixed decoder
output signal. The decoder output signal (Figure 10) is the same as if the mixer

output signal (Figure 4) were encoded and decoded by a BCC encoder/decoder in
this case. Thus, ΔLso similar quΔlity can be expected.
The decoder can not only determine the direction at which each source is to appear
but ΔLso the gain of each source can be varied. The gain is increased by choosing
and decreased by choosing
B. Using no de-correlation processing
The restriction of the previously described technique is that mixing is carried out with
a BCC synthesis scheme. One could imagine implementing not only ICTD, ICLD, and
ICC synthesis but additionally effects processing within the BCC synthesis.
However, it may be desired that existing mixers and effects processors can be used.
This ΔLso includes wavefield synthesis mixers (often denoted "convoluters"). For
using existing mixers and effects processors, the i,(n)are computed explicitly and
used as if they were the original source signals.
When applying no de-correlation processing (hj(n) = 8(n) in (16)) good audio quΔlity
can ΔLso be achieved. It is a compromise between artifacts introduced due to de-
correlation processing and artifacts due to the fact that the source signals st(n)are
correlated. When no de-correlation processing is used the resulting auditory spatiΔL
image may suffer from instability [1]. But the mixer may introduce itself some de-
correlation when reverberators or other effects are used and thus there is less need
for de-correlation processing.
If si,(n)are generated without de-correlation processing, the level of the sources
depends on the direction to which they are mixed relative to the other sources. By
replacing amplitude panning ΔLgorithms in existing mixers with an ΔLgorithm
compensating for this level dependence, the negative effect of loudness dependence
on mixing parameters can be circumvented. A level compensating amplitude
ΔLgorithm is shown in Figure 11 which aims to compensate the source level
dependence on mixing parameters. Given the gain factors of a conventional

amplitude panning ΔLgorithm (e.g. Figure 4), a, and bh the weights in Figure 11, a,
and bt, are computed by

Note that ai and b{ are computed such that the output subband power is the same
as if st(n) were independent in each subband.
c. Reducing the amount of de-correlation processing
As mentioned previously, the generation of independent si,(n) is problematic. Here
strategies are described for applying less de-correlation processing, while effectively
getting a similar effect as if the i,(n)were independent.
Consider for example a wavefield synthesis system as shown in Figure 12. The
desired virtuΔL source positions for Si, s2, ..., Se {M = 6) are indicated. A strategy for
computing si,-(n) (16) without generating M fully independent signals is:
1. Generate groups of source indices corresponding to sources close to each
other. For example in Figure 8 these could be: {1}, {2, 5}, {3}, and {4, 6}.
2. At each time in each subband select the source index of the strongest source,

Apply no de-correlation processing for the source indices part of the group containing
/'max, i.e. h{n) = 8(n).
3. For each other group choose the same h{n) within the group.
The described ΔLgorithm modifies the strongest signal components least. Additionally,
the number of different hj(n) that are used are reduced. This is an advantage
because de-correlation is easier the less independent channels need to be

generated. The described technique is ΔLso applicable when stereo or multi-channel
audio signals are mixed.
V. ScalABIliTY IN TERMS OF QUΔliTY AND BITRATE
The proposed scheme transmits only the sum of ΔLl source signals, which can be
coded with a conventional mono audio coder. When no mono backwards
compatibility is needed and capacity is available for transmission/storage of more
than one audio waveform, the proposed scheme can be scaled for use with more
than one transmission channel. This is implemented by generating severaL sum
signals with different subsets of the given source signals, i.e. to each subset of
source signals the proposed coding scheme is applied individuΔLly. Audio quΔlity is
expected to improve as the number of transmitted audio channels is increased
because less independent channels have to be generated by de-correlation from
each transmitted channel (compared to the case of one transmitted channel).
VI. BACKWARDS COMPATIBIliTY TO EXISTING STEREO AND SURROUND
AUDIO FORMATS
Consider the following audio delivery scenario. A consumer obtains a maximum
quΔlity stereo or multi-channel surround signal (e.g. by means of an audio CD, DVD,
or on-line music store, etc.). The goΔL is to optionally deliver to the consumer the
flexibility to generate a custom mix of the obtained audio content, without
compromising standard stereo/surround playback quΔlity.
This is implemented by delivering to the consumer (e.g. as optional buying option in
an on-line music store) a bit stream of side information which ΔLlows computation of
i.(n)as a function of the given stereo or multi-channel audio signal. The consumer's
mixing ΔLgorithm is then applied to the si,(n) In the following, two possibilities for
computing si,-(n), given stereo or multi-channel audio signals, are described.
A. Estimating the sum of the source signals at the receiver
The most straight forward way of using the proposed coding scheme with a stereo or
multi-channel audio transmission is illustrated in Figure 13, where y{n) (1 s i the L channels of the given stereo or multi-channel audio signal. The sum signal of

the sources is estimated by downmixing the transmitted channels to a single audio
channel. Downmixing is carried out by means of computing the sum of the channels
y{n) (1 si i si L) or more sophisticated techniques may be applied.
For best performance, it is recommended that the level of the source signals is
adapted prior to estimation (6) such that the power ratio between the
source signals approximates the power ratio with which the sources are contained in
the given stereo or multi-channel signal. In this case, the downmix of the transmitted
channels is a relatively good estimate of the sum of the sources (1) (or a scaled
version thereof).
An automated process may be used to adjust the level of the encoder source signal
inputs Sj(n) prior to computation of the side information. This process adaptively in
time estimates the level at which each source signal is contained in the given stereo
or multi-channel signal. Prior to side information computation, the level of each
source signal is then adaptively in time adjusted such that it is equΔL to the level at
which the source is contained in the stereo or multi-channel audio signal.
B. Using the transmitted channels individuΔLly
Figure 14 shows a different implementation of the proposed scheme with stereo or
multi-channel surround signal transmission. Here, the transmitted channels are not
downmixed, but used individuΔLly for generation of the st(n). Most generaLly, the
subband signals of s((n) are computed by

where w,(ri) are weights determining specific linear combinations of the transmitted
channels' subbands. The linear combinations are chosen such that the si,•(n) are
ΔLready as much decorrelated as possible. Thus, no or only a smΔLl amount of de-
correlation processing needs to be applied, which is favorable as discussed earlier.
VII. APPliCATIONS

ΔLready previously we mentioned a number of applications for the proposed coding
schemes. Here, we summarize these and mention a few more applications.
A. Audio coding for mixing
Whenever audio source signals need to be stored or transmitted prior to mixing them
to stereo, multi-channel, or wavefield synthesis audio signals, the proposed scheme
can be applied. With prior art, a mono audio coder would be applied to each source
signal independently, resulting in a bitrate which scales with the number of sources.
The proposed coding scheme can encode a high number of audio source signals
with a single mono audio coder plus relatively low bitrate side information. As
described in Section V, the audio quΔlity can be improved by using more than one
transmitted channel, if the memory/capacity to do so is available.
B. Re-mixing with meta-data
As described in Section VI, existing stereo and multi-channel audio signals can be re-
mixed with the help of additional side information (i.e. "meta-data"). As opposed to
only selling optimized stereo and multi-channel mixed audio content, meta data can
be sold ΔLlowing a user to re-mix his stereo and multi-channel music. This can for
example ΔLso be used for attenuating the vocals in a song for karaoke, or for
attenuating specific instruments for playing an instrument ΔLong the music.
Even if storage would not be an issue, the described scheme would be very attractive
for enabling custom mixing of music. That is, because it is likely that the music
industry would never be willing to give away the multi-track recordings. There is too
much a danger for abuse. The proposed scheme enables re-mixing capability without
giving away the multi-track recordings.
Furthermore, as soon as stereo or multi-channel signals are re-mixed a certain
degree of quΔlity reduction occurs, making illegΔL distribution of re-mixes less
attractive.
c. Stereo/multi-channel to wavefield synthesis conversion
Another application for the scheme described in Section VI is described in the
following. The stereo and multi-channel (e.g. 5.1 surround) audio accompanying

moving pictures can be extended for wavefield synthesis rendering by adding side
information. For example, Dolby AC-3 (audio on DVD) can be extended for 5.1
backwards compatibly coding audio for wavefield synthesis systems, i.e. DVDs play
back 5.1 surround sound on conventional legacy players and wavefield synthesis
sound on a new generation of players supporting processing of the side information.
VIII. SUBJECTIVE EVΔLUATIONS
We implemented a reΔL-time decoder of the ΔLgorithms proposed in Section IV-A and
IV-B. An FFT-based STFT filterbank is used. A 1024-point FFT and a STFT window
size of 768 (with zero padding) are used. The spectraL coefficients are grouped
together such that each group represents signal with a bandwidth of two times the
equivΔLent rectangular bandwidth (ERB). InformΔL listening reveΔLed that the audio
quΔlity did not notably improve when choosing higher frequency resolution. A lower
frequency resolution is favorable since it results in less parameters to be transmitted.
For each source, the amplitude/delay panning and gain can be adjusted individuΔLly.
The ΔLgorithm was used for coding of severaL multi-track audio recordings with 12 -
14 tracks.
The decoder ΔLlows 5.1 surround mixing using a vector base amplitude panning
(VBAP) mixer. Direction and gain of each source signal can be adjusted. The
software ΔLlows on the-fly switching between mixing the coded source signal and
mixing the original discrete source signals.
CasuΔL listening usuΔLly reveΔLs no or little difference between mixing the coded or
original source signals if for each source a gain G, of zero dB is used. The more the
source gains are varied the more artifacts occur. Slight amplification and attenuation
of the sources (e.g. up to ± 6 dB) still sounds good. A critical scenario is when ΔLl the
sources are mixed to one side and only a single source to the other opposite side. In
this case the audio quΔlity may be reduced, depending on the specific mixing and
source signals.
IX. CONCLUSIONS
A coding scheme for joint-coding of audio source signals, e.g. the channels of a
multi-track recording, was proposed. The goΔL is not to code the source signal

waveforms with high quΔlity, in which case joint-coding would give minimΔL coding
gain since the audio sources are usuΔLly independent. The goΔL is that when the
coded source signals are mixed a high quΔlity audio signal is obtained. By
considering statistical properties of the source signals, the properties of mixing
schemes, and spatiΔL hearing it was shown that significant coding gain improvement
is achieved by jointly coding the source signals.
The coding gain improvement is due to the fact that only one audio waveform is
transmitted.
Additionally side information, representing the statistical properties of the source
signals which are the relevant factors determining the spatiΔL perception of the final
mixed signal, are transmitted.
The side information rate is about 3 kbs per source signal. Any mixer can be applied
with the coded source signals, e.g. stereo, multi-channel, or wavefield synthesis
mixers.
It is straight forward to scale the proposed scheme for higher bitrate and quΔlity by
means of transmitting more than one audio channel. Furthermore, a variation of the
scheme was proposed which ΔLlows re-mixing of the given stereo or multi-channel
audio signal (and even changing of the audio format, e.g. stereo to multi-channel or
wavefield synthesis).
The applications of the proposed scheme are manifold. For example MPEG-4 could
be extended with the proposed scheme to reduce bitrate when more than one
"naturaL audio object" (source signal) needs to be transmitted. ΔLso, the proposed
scheme offers compact representation of content for wavefield synthesis systems. As
mentioned, existing stereo or multi-channel signals could be complemented with side
information to ΔLlow that the user re-mixes the signals to his liking.
REFERENCES
[1] C. FΔLler, Parametric Coding of SpatiΔL Audio, Ph.D. thesis, Swiss FederaL
Institute of Technology Lausanne (EPFL), 2004, Ph.D. Thesis No. 3062.

[2] C. FΔLler and F. Baumgarte, "BinauraL Cue Coding - Part II: Schemes and
applications," IEEE Trans, on Speech and Audio Proa, vol. 11, no. 6, Nov. 2003.

We Claim:
1. Method for synthesizing a plurality of audio channels, comprising:
retrieving from an audio stream at least one sum signal representing a sum
of source signals,
retrieving from the audio stream statistical information about one or more
source signals,
receiving from the audio stream, or determining locally, parameters
describing an output audio format and source mixing parameters,
computing output mixer parameters from the received statistical information,
the parameters describing an output audio format, and the source mixing
parameters,
synthesizing the plurality of audio channels from the at least one sum signal
based on the computed output mixer parameters.
2. Method as claimed in claim 1, wherein the statistical information represent
spectraL envelopes of the source signals, or the spectraL envelopes of the
one or more audio source signals comprise lattice filter parameters or line
spectraL parameters or in which the statistical information represent a
relative power as a function of frequency and time of the plurality of
source signals.

3. Method as claimed in claim 1, wherein the step of computing the output
mixer parameters comprises computing the cues of the plurality of audio
channels and computing the output mixer parameters using the calculated
cues of the plurality of audio channels.
4. Method as claimed in claim 1, wherein the audio channels are synthesized
in a subband domain of a filterbank.
5. Method as claimed in claim 4, wherein a number and bandwidths of the
subband domain are determined according to a spectraL and temporaL
resolution of an human auditory system.
6. Method as claimed in claim 4, wherein a number of subbands is between
3 and 40.
7. Method as claimed in claim 4, wherein subbands in the subband domain
have different bandwidths, wherein subbands at lower frequencies have
smΔLler bandwidths than subbands at higher frequencies.

8. Method as claimed in claim 4, wherein a short time Fourier transform
(STFT) based filterbank is used and spectraL coefficients are combined to
form groups of spectraL coefficients such that each group of spectraL
coefficients forms a subband.
9. Method as claimed in claim 1, wherein the statistical information ΔLso
comprises auto-correlation functions.
10. Method as claimed in claim 2, wherein spectraL envelopes are represented
as linear predictive coding (LPC) parameters.
11. Method as claimed in claim 3, wherein the computed cues are level
difference, time difference, or coherence fcr different frequencies and
time instants.
12.Apparatus for synthesizing a plurality of audio channels, wherein the
apparatus is operative for:

retrieving from an audio stream at least one; sum signal representing a
sum of source signals,
retrieving from the audio stream statistical information about one or more
source signals,
receiving from the audio stream, or determining locally, parameters
describing an output audio format and source mixing parameters,
computing output mixer parameters from the received statistical information,
the parameters describing an output audio format, and the source mixing
parameters,
synthesizing the plurality of audio channels from the at least one sum signal
based on the computed output mixer parameters.

ABSTRACT

TITLE: Method and Apparatus for synthesizing a plurality of audio channels
The invention relates to Method for synthesizing a plurality of audio channels,
comprising retrieving from an audio stream at least one sum signal
representing a sum of source signals, retrieving from the audio stream
statistical information about one or more source signals, receiving from the
audio stream, or determining locally, parameters describing an output audio
format and source mixing parameters, computing output mixer parameters
from the received statistical information, the parameters describing an output
audio format, and the source mixing parameters, synthesizing the plurality of
audio channels from the at least one sum signal based on the computed
output mixer parameters.

Documents:

02778-kolnp-2007-abstract.pdf

02778-kolnp-2007-claims.pdf

02778-kolnp-2007-correspondence others 1.1.pdf

02778-kolnp-2007-correspondence others.pdf

02778-kolnp-2007-description complete.pdf

02778-kolnp-2007-drawings.pdf

02778-kolnp-2007-form 1.pdf

02778-kolnp-2007-form 18.pdf

02778-kolnp-2007-form 2.pdf

02778-kolnp-2007-form 3.pdf

02778-kolnp-2007-form 5.pdf

02778-kolnp-2007-international publication.pdf

02778-kolnp-2007-international search report.pdf

02778-kolnp-2007-pct request form.pdf

02778-kolnp-2007-priority document.pdf

2778-KOLNP-2007-(19-09-2012)-ABSTRACT.pdf

2778-KOLNP-2007-(19-09-2012)-AMANDED CLAIMS.pdf

2778-KOLNP-2007-(19-09-2012)-ANNEXURE TO FORM 3.pdf

2778-KOLNP-2007-(19-09-2012)-DESCRIPTION (COMPLETE).pdf

2778-KOLNP-2007-(19-09-2012)-DRAWING.pdf

2778-KOLNP-2007-(19-09-2012)-EXAMINATION REPORT REPLY RECEIVED.pdf

2778-KOLNP-2007-(19-09-2012)-FORM-1.pdf

2778-KOLNP-2007-(19-09-2012)-FORM-2.pdf

2778-KOLNP-2007-(19-09-2012)-OTHERS.pdf

2778-KOLNP-2007-ASSIGNMENT.pdf

2778-KOLNP-2007-CANCELLED PAGES.pdf

2778-KOLNP-2007-CORRESPONDENCE 1.4.pdf

2778-KOLNP-2007-CORRESPONDENCE OTHERS 1.2.pdf

2778-KOLNP-2007-CORRESPONDENCE OTHERS 1.3.pdf

2778-KOLNP-2007-CORRESPONDENCE.pdf

2778-KOLNP-2007-EXAMINATION REPORT.pdf

2778-KOLNP-2007-FORM 13.pdf

2778-KOLNP-2007-FORM 18.pdf

2778-KOLNP-2007-GRANTED-ABSTRACT.pdf

2778-KOLNP-2007-GRANTED-CLAIMS.pdf

2778-KOLNP-2007-GRANTED-DESCRIPTION (COMPLETE).pdf

2778-KOLNP-2007-GRANTED-DRAWINGS.pdf

2778-KOLNP-2007-GRANTED-FORM 1.pdf

2778-KOLNP-2007-GRANTED-FORM 2.pdf

2778-KOLNP-2007-GRANTED-FORM 3.pdf

2778-KOLNP-2007-GRANTED-FORM 5.pdf

2778-KOLNP-2007-GRANTED-SPECIFICATION-COMPLETE.pdf

2778-KOLNP-2007-INTERNATIONAL SEARCH REPORT & OTHERS.pdf

2778-KOLNP-2007-OTHERS 1.1.pdf

2778-KOLNP-2007-OTHERS.pdf

2778-KOLNP-2007-PA.pdf

2778-KOLNP-2007-REPLY TO EXAMINATION REPORT.pdf

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Patent Number

256863

Indian Patent Application Number

2778/KOLNP/2007

PG Journal Number

32/2013

Publication Date

09-Aug-2013

Grant Date

05-Aug-2013

Date of Filing

30-Jul-2007

Name of Patentee

FRAUNHOFER-GESELLSCHAFT ZUR FORDERUNG DER ANGEWANDTEN FORSCHUNG E.V.

Applicant Address

HANSASTRASSE 27 C 80686 MUNCHEN

Inventors:

#	Inventor's Name	Inventor's Address
1	FALLER CHRISTOF	GUETRAIN 1 8274 TAGERWILEN

PCT International Classification Number

G10L 19/00,H04S 3/00

PCT International Application Number

PCT/EP2006/050904

PCT International Filing date

2006-02-13

PCT Conventions:

#	PCT Application Number	Date of Convention	Priority Country
1	05101055.1	2005-02-14	EPO