EP2094031A2 - Device and method for creating an encoding stereo signal of an audio section or audio data stream - Google Patents

Abstract

The device has a multi-channel decoder (11) to make more than two multi-channels available from a multi-channel representation. A headphone signal processor (12) processes a headphone signal, in order to produce an uncoded stereo signal with an uncoded first stereo channel (10a) and an uncoded second stereo channel (10b). A stereo coder (13) codes the first uncoded stereo channels, in order to receive a coded stereo signal (14). The stereo coder has a data rate for transferring the coded stereo signal being smaller than a data rate for transferring the uncoded stereo signal. An independent claim is included for a method for producing a coded stereo signal of an audio piece or an audio data stream with a first stereo channel and a second stereo channel from a multi-channel representation of the audio piece or audio data stream, and a computer program.

Description

The present invention relates to multi-channel audio technology, and more particularly to multi-channel audio applications in conjunction with headphone techniques.

The two international patent applications WO 99/49574 and WO 99/14983 disclose audio signal processing techniques for driving a pair of oppositely located headphone speakers to provide a user with spatial perception of the audio scene through the two headphones, which is not just a stereo representation but a multi-channel presentation. Thus, the listener receives via his or her headphones a spatial perception of an audio piece that is equal to its spatial perception at best, if the user would sit in a playback room, which is equipped for example with a 5.1 audio system. For this purpose, for each headphone speaker, each channel of the multi-channel audio track or multi-channel audio stream, as shown in Fig. 2 is shown, fed to its own filter, after which then the respective filtered belonging together channels are added, as shown below.

On a left side in Fig. 2 are the multi-channel inputs 20, which together represent a multi-channel representation of the audio track or audio data stream. Such a scenario is for example in Fig. 10 shown schematically. Fig. 10 shows a playback room 200 in which a so-called 5.1 audio system is arranged. The 5.1 audio system includes a center speaker 201, a front-left speaker 202, a front-right speaker 203, a rear-left speaker 204, and a rear-right speaker 205. A 5.1 audio system has an additional subwoofer 206, which is also referred to as a low-frequency enhancement channel. In the so-called "sweet spot" of the playback room 200 is a listener 207 wearing a headset 208 having a left headphone speaker 209 and a right headphone speaker 210.

In the Fig. 2 Processing means shown is now adapted to each channel 1, 2, 3 of the multi-channel inputs 20 with a filter H _iL , the sound channel from the speaker to the left speaker 209 in Fig. 10 describes to filter and to further filter the same channel with a filter H _{iR representing} the sound from one of the five speakers to the right ear 210 to the right speaker 210 of the headphone 208.

For example, if channel 1 were in Fig. 2 the front left channel, which passes through the speaker 202 in Fig. 10 is emitted, the filter H _iL would represent the channel indicated by a dashed line 212, while the filter H _1R would represent the channel represented by a dashed line 213. As it is in Fig. 10 For example, indicated by a dashed line 214, the left earphone speaker 209 receives not only the direct sound, but also early reflections at an edge of the playback room and, of course, late reflections that are expressed in a diffuse reverberation.

Such a filter representation is in Fig. 11 shown. In particular shows Fig. 11 a schematic example of an impulse response of a filter, for example, the filter H _1L of Fig. 2 The direct sound coming through line 212 in FIG Fig. 11 is represented by a peak at the beginning of the filter, whereas early reflections, such as those shown by 214 in FIG Fig. 10 are represented by a central region with a plurality of (discrete) smaller peaks in Fig. 11 be reproduced. The diffuse reverberation is then typically not more resolved to individual peaks, since the sound of the speaker 202 is reflected in principle as often as desired, the energy of course, with each reflection and additional propagation distance continues to decrease, as by the decreasing energy in the rear section, with "diffuse reverberation" in Fig. 11 is indicated, is shown.

Each of the in Fig. 2 Therefore, the filter shown comprises a filter impulse response, which has approximately a course, as shown by the schematic impulse response representation in Fig. 11 is reproduced. Of course, the individual filter impulse response from the playback room, the positioning of the speakers, any attenuation properties in the playback room z. B. due to several persons present or furniture located in the playback room, etc., and ideally also depend on the characteristics of the individual speakers 201 to 206.

The fact that the signals from all speakers are superimposed on the ear of the listener 207 is determined by the adders 22 and 23 in FIG Fig. 2 shown. Thus, each channel is filtered with a corresponding filter for the left ear to then simply add up the signals output from the filters, which are all for the left ear, to obtain the headphone output signal for the left ear L. Similarly, addition is made by the right ear adder 23 and the right headphone speaker 210 in FIG Fig. 10 in order to obtain the headphone output signal for the right ear by superimposing all the loudspeaker signals filtered with a corresponding filter for the right ear.

Due to the fact that in addition to the direct sound, there are also early reflections and, in particular, a diffuse reverberation, which are of great importance, in particular for the perception of space, so that the sound does not sound synthetic or "wooden" but gives the listener the feeling that he is really sitting in a concert hall with its acoustic properties, the impulse responses of each filter 21 will all take a considerable length. The folding of each individual multi-channel of the multi-channel representation with two filters therefore already leads to a considerable computational task. Since two filters are required for each individual multi-channel, namely one for the left ear and another for the right ear, if the subwoofer channel is also treated separately, for a headphone playback of a 5.1 multi-channel display in total 12 different filters needed. All filters have how it looks Fig. 11 It can be seen, a very long impulse response to not only the direct sound but also to be able to consider early reflections and the diffuse reverberation, which gives an audio piece actually only the correct sound reproduction and a good spatial impression.

Therefore, to put the well-known concept into reality, it becomes, as in Fig. 10 in addition to a multi-channel player 220 requires the very complex virtual sound processing 222, the signals for the two speakers 209 and 210, through lines 224 and 226 in Fig. 10 are shown supplies.

Headphone systems for generating a multi-channel headphone sounds are therefore complicated, bulky and expensive, due to the high computing power, the high power requirements for the necessary high computing power and high memory requirements for the ratings to be performed with the impulse response and the associated large volume or expensive blocks for the player is. Such applications are therefore tied to home PC sound cards or laptop sound cards or home stereo systems.

In particular, the ever-expanding market for mobile players, such as mobile CD players or in particular the hardware players remains the multi-channel headphone sound closed because the computational requirements for filtering the multi-channels with z. B. 12 different filters both in terms of processor resources as well as in terms of power consumption of the typically battery-powered devices are not feasible in the price segment. This is about a price segment at the lower (low) end of the scale. Especially this price segment is economically very interesting because of the large quantities.

The object of the present invention is to provide an efficient signal processing concept that enables multi-channel headphone reproduction even on simple playback devices.

This object is achieved by a device for generating a coded stereo signal according to claim 1 or by a method for generating a coded stereo signal according to claim 11 or a computer program according to claim 12.

The present invention is based on the finding that the high-quality and attractive multi-channel headphone sound can be made available to all available playback devices, such as CD players or hardware players, by providing a multi-channel display of an audio track or audio data stream So, for example, a 5.1 representation of an audio piece outside of a hardware player, so z. B. in a high-power computer of a provider of a headphone signal processing. However, according to the invention, the result of earphone signal processing is not simply played back but fed to a typical audio stereo coder, which then generates a coded stereo signal from the left earphone channel and the right earphone channel.

This coded stereo signal can then, like any other coded stereo signal, which has no multi-channel representation, the hardware player or z. B. a mobile CD player in the form of a CD. The player will then provide the user with a headphone multi-channel sound without having to add any additional resources or equipment to existing devices. According to the invention, the result of the headphone signal processing, ie the left and the right headphone signal is not reproduced in a headphone, as in the prior art, but coded and output as coded stereo data.

Such an output may be a store, a transfer, or anything like that. Such a file with coded stereo data can then be readily supplied to any player designed for stereo playback without the user having to make any changes to his device.

The concept according to the invention of generating a coded stereo signal from the result of the headphone signal processing thus makes it possible for the multichannel presentation, which provides a much better and more lifelike quality for a user, to also be simple and very widely used in the future even more widespread hardware players can be used.

In a preferred embodiment of the present invention is based on a coded multi-channel representation, that is, a parameter representation having one or typically two base channels, and further comprising parameter data to on the basis of the base channels and the parameter data to the multi-channels of the multi-channel representation produce. After a frequency-domain-based method for multi-channel decoding is preferred, the headphone signal processing according to the invention is performed not in the time domain by convolution of the time signal with the impulse response, but in the frequency domain by multiplication with the filter transfer function.

This makes it possible to save at least one inverse transformation before the headphone signal processing and is particularly advantageous when the subsequent stereo coder operates in the frequency domain, so that then, without ever having to go into the time domain, the stereo coding of the headphone Stereo signal can also be done without gear in the time domain. The processing from the multi-channel representation to the coded stereo signal without the intervention of a time domain or at least a reduced number of transformations is not only interesting in terms of computational time efficiency, but limits quality losses, since fewer processing stages also introduce fewer artifacts into the audio signal.

In particular, in block-based methods that quantize considering a psychoacoustic masking threshold, as is preferred for the stereo encoder, it is important to prevent as much tandem coding artifacts as possible.

In a particularly preferred embodiment of the present invention is used as a multi-channel representation of a BCC representation with one or preferably two base channels. After the BCC method works in the frequency domain, the multi-channels are not transformed into the time domain after their synthesis, as is usual with the BCC decoder. Instead, the spectral representation of the multichannels is used in blocks and subjected to the headphone signal processing. For this purpose, the transfer functions of the filters are used, that is, the Fourier transforms the impulse responses to perform a multiplication between the spectral representation of the multi-channels and the filter transfer functions. If the impulse responses of the filters are longer in time than one block of spectral components at the output of the BCC decoder, a block-wise filter processing is preferred in which the impulse responses of the filters in the time domain be separated and be transformed block by block to then perform necessary for such measures weights of the spectra, as described for example in the WO 94/01933 are disclosed.

Fig. 1

ein Blockschaltbild der erfindungsgemäßen Vorrichtung zum Erzeugen eines codierten Stereo-Signals;

Fig. 2

eine Detaildarstellung einer Implementierung der Kopfhörer-Signalverarbeitung von Fig. 1;

Fig. 3

einen bekannten Joint-Stereo-Codierer zum Erzeugen von Kanaldaten und parametrischen Multikanal-Informationen;

Fig. 4

eine Darstellung eines Schemas zum Bestimmen von ICLD-, ICTD- und ICC-Parametern für eine BCC-Codierung/Decodierung;

Fig. 5

eine Blockdiagrammdarstellung einer BCC-Codierer/Decodierer-Kette;

Fig. 6

ein Blockdiagramm einer Implementierung des BCC-Synthese-Blocks von Fig. 5;

Fig. 7

eine Kaskadierung zwischen einem Multikanal-Decodierer und der Kopfhörer-Signalverarbeitung ohne Transformation in den Zeitbereich;

Fig. 8

eine Kaskadierung zwischen der Kopfhörer-Signalverarbeitung und einem Stereo-Codierer ohne Transformation in den Zeitbereich;

Fig. 9

ein Prinzip-Blockdiagramm eines bevorzugten Stereo-Codierers;

Fig. 10

eine Prinzipdarstellung eines Wiedergabe-Szenarios zum Bestimmen der Filterfunktionen von Fig. 2;

Fig. 11

eine prinzipielle Darstellung einer zu erwartenden Impulsantwort eines Filters, das gemäß Fig. 10 bestimmt ist.

Preferred embodiments of the present invention will be explained below in detail with reference to the accompanying drawings. Show it:

Fig. 1

a block diagram of the inventive apparatus for generating a coded stereo signal;

Fig. 2

a detailed representation of an implementation of the headphone signal processing of Fig. 1 ;

Fig. 3

a known joint stereo encoder for generating channel data and multi-channel parametric information;

Fig. 4

a representation of a scheme for determining ICLD, ICTD and ICC parameters for BCC encoding / decoding;

Fig. 5

a block diagram representation of a BCC encoder / decoder chain;

Fig. 6

a block diagram of an implementation of the BCC synthesis block of Fig. 5 ;

Fig. 7

a cascading between a multi-channel decoder and the headphone signal processing without transformation into the time domain;

Fig. 8

a cascading between the headphone signal processing and a stereo encoder without transformation into the time domain;

Fig. 9

a schematic block diagram of a preferred stereo encoder;

Fig. 10

a schematic representation of a playback scenario for determining the filter functions of Fig. 2 ;

Fig. 11

a schematic representation of an expected impulse response of a filter according to Fig. 10 is determined.

Fig. 1 shows a schematic block diagram of an inventive device for generating a coded stereo signal of an audio track or audio data stream. The stereo signal comprises in uncoded form an uncoded first stereo channel 10a and an uncoded second stereo channel 10b, and is generated from a multi-channel representation of the audio piece or audio data stream, the multi-channel representation comprising information on more than two multi-channels. As will be seen later, the multi-channel representation may be in uncoded or encoded form. If the multi-channel representation is present in uncoded form, it includes three or more multi-channels. In a preferred application scenario, the multichannel presentation includes five channels and a subwoofer channel.

On the other hand, if the multi-channel representation is in encoded form, this encoded form typically includes one or more base channels and parameters for synthesizing the three or more multi-channels from the one or both base channels. A multi-channel decoder 11 is therefore one example of a means for providing the more than two multi-channels from the multi-channel representation. On the other hand, is the multichannel presentation already in uncoded form, ie z. B. in the form of 5 + 1 PCM channels, the means for providing an input terminal for a device 12 for Performing headphone signal processing to produce the uncoded stereo signal with the uncoded first stereo channel 10a and the uncoded second stereo channel 10b.

Preferably, the headphone signal processing means 12 is arranged to evaluate the multi-channels of the multi-channel display respectively with a first filter function for the first stereo channel and a second filter function for the second stereo channel and weighted multi-channels in each case to obtain the uncoded first stereo channel and the uncoded second stereo channel, as determined by Fig. 2 has been shown. The device 12 for performing the headphone signal processing is followed by a stereo encoder 13, which is configured to encode the first uncoded stereo channel 10a and the second uncoded stereo channel 10b to the encoded stereo signal at an output 14th of the stereo encoder 13. The stereo encoder performs data rate reduction so that a data rate necessary to transmit the encoded stereo signal is less than a data rate necessary to transmit the uncoded stereo signal.

According to the invention, a concept is thus achieved which makes it possible to supply multi-channel sound, which is also referred to as "surround", via simple playback devices, such as hardware players, stereo headphones.

As a simple headphone signal processing z. B. the sum of certain channels are formed to obtain the output channels for the stereo data. Improved methods work with more complex algorithms, which in turn achieve a better quality of reproduction.

It should be noted that the inventive concept allows the computationally expensive steps to multi-channel decode and perform headphone signal processing not be performed in the player itself, but are performed externally. The result of the inventive concept is a coded stereo file, which is for example an MP3 file, an AAC file, a HE AAC file or any other stereo file.

In other embodiments, the multi-channel decoding, the headphone signal processing and the stereo encoding can be performed on different devices, since the output data or input data of the individual blocks are easily portable and standardized generated and stored.

Subsequently, reference will be made to Fig. 7 a preferred embodiment of the present invention is shown, in which the multi-channel decoder 11 has a filter bank or an FFT function, such that the multi-channel representation is provided in the frequency domain. Specifically, the individual multi-channels are generated separately as blocks of spectral values for each channel. According to the headphone signal processing is then not performed in the time domain by folding the temporal channels with the filter impulse responses, but it is performed a multiplication of the frequency domain representation of the multi-channels with a spectral representation of the filter impulse response. At the output of the headphone signal processing then an uncoded stereo signal is reached, which is not present in the time domain, but comprising a left and a right stereo channel, wherein such a stereo channel is given as a result of blocks of spectral values, each block of spectral values represents a short-term spectrum of the stereo channel.

At the in Fig. 8 In the embodiment shown, the headphone signal processing block 12 is supplied on the input side either with time domain or frequency domain data. On the output side are the uncoded stereo channels in the frequency domain, ie again generated as a sequence of blocks of spectral values. In this case, a stereo coder which is transformation-based, that processes spectral values, without a frequency / time conversion and a subsequent time between the headphone signal processor 12 and the stereo coder 13 is preferred as the stereo coder 13 Frequency conversion is required. On the output side, the stereo encoder 13 then outputs a file with the coded stereo signal, which in addition to page information comprises a coded form of spectral values.

In a particularly preferred embodiment of the present invention is en route from the multi-channel representation at the entrance of the block 11 of Fig. 1 to the coded stereo file at the output 14 of the device of Fig. 1 a continuous frequency domain processing performed without a conversion into the time domain and, if appropriate, again has to be implemented in the frequency domain. If an MP3 coder or an AAC coder is used as the stereo coder, it is preferable to convert the Fourier spectrum at the output of the headphone signal processing block into an MDCT spectrum. This ensures, according to the invention, that the phase information which is required exactly for the convolution / evaluation of the channels in the headphone signal processing block is converted into the MDCT representation which does not operate in phase-correct manner, so that for the stereo encoder, in contrast to a normal MP3 encoder or a normal AAC encoder no means for converting time domain in the frequency domain, ie in the MDCT spectrum is needed.

Fig. 9 shows a general block diagram for a preferred stereo encoder. The stereo encoder comprises on the input side a joint stereo module 15, which preferably adaptively determines whether a common stereo coding, for example in the form of a middle / side encoding, provides a higher coding gain than a separate processing from left and right channel. The joint stereo module 15 may further be configured to perform intensity stereo coding, wherein intensity stereo coding provides a significant coding gain, especially at higher frequencies, without audible artifacts occurring. The output of the joint stereo module 15 is then further processed using various other redundancy-reducing measures, such as TNS filtering, noise substitution, etc., and then applying the results to a quantizer 16 which quantizes using a psychoacoustic masking threshold of the spectral values. The quantizer step size is chosen such that the noise introduced by the quantization remains below the psychoacoustic marker threshold, so that a data rate reduction is achieved without the distortions introduced by the lossy quantization becoming audible. The quantizer 16 is finally followed by an entropy coder 17, which performs a lossless entropy coding of the quantized spectral values. At the output of the entropy coder, the encoded stereo signal is then present, which comprises side information necessary for decoding in addition to the entropy-coded spectral values.

Hereinafter, preferred implementations of the multi-channel decoder or on preferred multi-channel representations using the Fig. 3 to 6 received.

Thus, many techniques exist in the art for reducing the amount of data needed to transmit a multi-channel audio signal. Such techniques are called joint stereo techniques. For this purpose is on Fig. 3 which shows a joint stereo device 60. This device may be a device implementing, for example, the intensity stereo (IS) technique or the binaural cue coding technique (BCC). Such a device usually receives as input at least two Channels CH1, CH2, .... CHn, and outputs a single carrier channel and multi-channel parametric information. The parametric data is defined so that an approximation of an original channel (CH1, CH2, ..., CHn) can be calculated in a decoder.

Normally, the carrier channel will include subband samples, spectral coefficients, time domain samples, etc., which provide a relatively fine representation of the underlying signal, while the parametric data does not include such samples or spectral coefficients, but control parameters for controlling a particular reconstruction algorithm, such as multiplying by weight. by time shifting, by frequency shifting, etc. The parametric multi-channel information therefore comprises a relatively rough representation of the signal or the associated channel. Expressed in numbers, the amount of data needed by a carrier channel is about 60 to 70 kbps, while the amount of data required by one channel parametric page information is in the range of 1.5 to 2.5 kbps. It should be noted that the above figures apply to compressed data. Of course, a non-compressed CD channel requires data rates on the order of about ten times. An example of parametric data is the known scale factors, intensity stereo information, or BCC parameters, as set forth below.

The technique of intensity stereo coding is described in the AES Preprint 3799, " Intensity Stereo Coding ", J. Herre, KH Brandenburg, D. Lederer, February 1994 , Amsterdam. In general, the concept of Intensity Stereo is based on a major axis transformation that is to be performed on data from both stereophonic audio channels. When most data points are concentrated around the first major axis, a gain in coding can be achieved by rotating both signals through a certain angle before the coding takes place. However, this is not always true for real stereophonic reproduction techniques. Therefore, this technique is modified to exclude the second orthogonal component from transmission in the bitstream. Thus, the reconstructed signals for the left and right channels consist of differently weighted or scaled versions of the same transmitted signal. Nevertheless, the reconstructed signals differ in their amplitude, but they are identical in terms of their phase information. However, the energy-time envelopes of both original audio channels are maintained by the selective scaling operation, which typically operates in a frequency-selective manner. This corresponds to the human perception of sound at high frequencies, where the dominant spatial information is determined by the energy envelopes.

In addition, in practical implementations, the transmitted signal, i. H. the carrier channel is generated from the sum signal of the left channel and the right channel instead of the rotation of both components. Furthermore, this processing, i. H. generating intensity-stereo parameters to perform the scaling operations in a frequency-selective manner, i. H. independent for each scale factor band, d. H. for each encoder frequency partition. Preferably, both channels are combined to form a combined or "carrier" channel and, in addition to the combined channel, the intensity stereo information. The intensity stereo information depends on the energy of the first channel, the energy of the second channel or the energy of the combined channel.

The BCC technique is described in the AES convention paper 5574 " Binaural Cue Coding Applied to Stereo and Multichannel Audio Compression ", T. Faller, F. Baumgarte, May 2002 , Munich. In BCC coding, a number of audio input channels are converted into a spectral representation, using a DFT-based transformation with overlapping windows. The resulting spectrum is divided into non-overlapping sections, each of which has an index. Each partition has a bandwidth proportional to the equivalent rectangular bandwidth (ERB). The Inter Channel Level Differences (ICLD) and the Inter Channel Time Differences (ICTD) are determined for each partition and for each frame k. The ICLD and ICTD are quantized and encoded to eventually arrive as page information in a BCC bitstream. The inter-channel level differences and the inter-channel time differences are given for each channel relative to a reference channel. Then, the parameters are calculated according to predetermined formulas that depend on the particular partitions of the signal to be processed.

On the decoder side, the decoder typically receives a mono signal and the BCC bit stream. The mono signal is transformed into the frequency domain and input to a spatial synthesis block which also receives decoded ICLD and ICTD values. In the spatial synthesis block, the BCC parameters (ICLD and ICTD) are used to perform a weighting operation of the mono signal to synthesize the multi-channel signals representing, after a frequency / time conversion, a reconstruction of the original multi-channel audio signal.

In the case of BCC, the joint stereo module 60 operates to output the channel-side information such that the parametric channel data is quantized and coded ICLD or ICTD parameters using one of the original channels as the reference channel for encoding the channel side information.

Normally, the carrier signal is formed from the sum of the participating source channels.

Of course, the above techniques provide only a monodic representation for a decoder that can only process the carrier channel, but is unable to process the parametric data to produce one or more approximations of more than one input channel.

The BCC technique is also disclosed in the US patent publications US 2003/0219130 A1 . US 2003/0026441 A1 and US 2003/0035553 A1 described. In addition, the specialist publication " Binaural cue coding. Part II: Schemes and Applications ", T. Faller and F. Baumgarte, IEEE Trans. On Audio and Speech Proc. Vol. 11, No. 6, November 2003 directed.

In the following, a typical BCC scheme for multi-channel audio coding will be described in more detail, referring to FIGS 4 to 6 ,

Fig. 5 shows such a BCC scheme for encoding / transmission of multi-channel audio signals. The multi-channel audio input signal at an input 110 of a BCC encoder 112 is down-mixed in a so-called downmix block 114. In this example, the original multi-channel signal at the input 110 is a 5-channel surround signal having a front left channel, a front right channel, a left surround channel, a right surround channel and a center channel. In the preferred embodiment of the present invention, the downmix block 114 generates a sum signal by simply adding these five channels into a mono signal.

Other downmixing schemes are known in the art such that a single channel downmix channel is obtained using a multi-channel input signal.

This single channel is output on a sum signal line 115. Page information provided by the BCC analysis block 116 is output on a page information line 117.

In the BCC analysis block, inter-carial level differences (ICLD) and inter-channel time differences (ICTD) are calculated, as shown above. Recently, the BCC analysis block 116 is also capable of calculating inter-channel correlation (ICC) values. The sum signal and the page information are transmitted in a quantized and encoded format to a BCC decoder 120. The BCC decoder decomposes the transmitted sum signal into a number of subbands and performs scaling, delays and other processing to provide the subbands of the multichannel audio channels to be output. This processing is performed so that the ICLD, ICTD and ICC parameters (cues) of a reconstructed multichannel signal at output 121 match the corresponding cues for the original multichannel signal at input 110 in BCC encoder 112. For this purpose, the BCC decoder 120 includes a BCC synthesis block 122 and a page information revision block 123.

Hereinafter, the internal structure of the BCC synthesis block 122 will be referred to Fig. 6 shown. The sum signal on line 115 is fed to a time / frequency conversion unit or filter bank FB 125. At the output of the block 125 there exists a number N of subband signals or, in an extreme case, a block of spectral coefficients, when the audio filter bank 125 performs a 1: 1 transform, ie a transform producing N spectral coefficients from N time domain samples.

The BCC synthesis block 122 further includes a delay stage 126, a level modification stage 127, a correlation processing stage 128, and an inverse filter bank stage IFB 129. At the output of stage 129, the reconstructed Multi-channel audio signal with, for example, five channels in the case of a 5-channel surround system are output to a set of speakers 124 as shown in FIG Fig. 5 or Fig. 4 are shown.

The input signal sn is converted into the frequency domain or the filter bank region by means of the element 125. The signal output by element 125 is copied so as to obtain multiple versions of the same signal, as represented by copy node 130. The number of versions of the original signal is equal to the number of output channels in the output signal. Then each version of the original signal at node 130 undergoes a certain delay d ₁ , d ₂ , ..., d _i , ... d _N. The delay parameters are determined by the page information processing block 123 in FIG Fig. 5 and from the inter-channel time differences as determined by the BCC analysis block 116 of FIG Fig. 5 have been calculated derived.

The same applies to the multiplication parameters a ₁ , a ₂ ,..., A _i ,..., A _N , which are also calculated by the page information processing block 123 based on the interchannel level differences as calculated by the BCC analysis block 116. be calculated.

The ICC parameters calculated by BCC analysis block 116 are used to control the functionality of block 128 so that certain correlations between the delayed and level manipulated signals are obtained at the outputs of block 128. It should be noted here that the order of stages 126, 127, 128 is different from the one in FIG Fig. 6 may differ.

It should be noted that in a frame-by-frame processing of the audio signal, the BCC analysis is performed frame by frame, so temporally variable, and that further a frequency-wise BCC analysis is obtained as indicated by the filter bank partitioning Fig. 6 is apparent. This means that the BCC parameters are obtained for each spectral band. This also means that in the case where the audio filter bank 125 decomposes the input signal into, for example, 32 bandpass signals, the BCC analysis block obtains a set of BCC parameters for each of the 32 bands. Of course, the BCC synthesis block 122 performs Fig. 5 who is detailed in Fig. 6 is a reconstruction, which is based on the exemplified 32 bands.

Subsequently, reference will be made to Fig. 4 presented a scenario that is used to determine individual BCC parameters. Normally the ICLD, ICTD and ICC parameters can be defined between channel pairs. However, it is preferred to determine the ICLD and ICTD parameters between a reference channel and each other channel. This is in Fig. 4A shown.

ICC parameters can be defined in several ways. Generally speaking, one can determine ICC parameters in the encoder between all possible channel pairs, as shown in FIG Fig. 4B is shown. However, it has been proposed to calculate only ICC parameters between the strongest two channels at a time, as in Fig. 4C where an example is shown in which one ICC parameter between channels 1 and 2 is calculated one at a time, and at another time an ICC parameter between channels 1 and 5 is calculated. The decoder then synthesizes the inter-channel correlation between the strongest channels in the decoder and uses certain heuristic rules to compute and synthesize the inter-channel coherence for the remaining channel pairs.

With regard to the calculation, for example, of the multiplication parameters a ₁ , a _N based on the transmitted ICLD parameters Reference is made to AES Convention Paper No. 5574. The ICLD parameters represent an energy distribution of an original multichannel signal. Without loss of generality, it is preferred as shown in FIG Fig. 4A shown to take four ICLD parameters representing the energy difference between the respective channels and the front left channel. In the page information processing block 122, the multiplication parameters a ₁ , ..., a _{N are derived} from the ICLD parameters such that the total energy of all reconstructed output channels is the same (or proportional to the energy of the transmitted sum signal).

At the in Fig. 7 The embodiment shown is based on the frequency-time conversion, which by the inverse filter banks IFB 129 of Fig. 6 be achieved, waived. Instead, the spectral representations of the individual channels are used at the input of these inverse filter banks and the headphone signal processing device of Fig. 7 supplied to perform the evaluation of the individual multi-channels with the two filters per multi-channel without an additional frequency / time transformation.

With regard to a complete occurring in the frequency domain processing, it should be noted that then the multi-channel decoder, so z. B. the filter bank 125 of Fig. 6 and the stereo encoder should have the same time / frequency resolution. Furthermore, it is preferred to use one and the same filter bank, which is particularly advantageous in that for the entire processing, as shown in FIG Fig. 1 represents, only a single filter bank is needed. In this case, a particularly efficient processing, since the transformation in the multi-channel decoder and in the stereo encoder do not need to be calculated.

The input data or output data in the inventive concept are therefore preferably coded in the frequency domain by means of transformation / filter bank and are coded according to psychoacoustic specifications taking advantage of masking effects, in which case a spectral representation of the signals should be present in particular in the decoder. Examples include MP3 files, AAC files or AC3 files. However, the input data or output data can also be encoded by summing and subtraction, as is the case with so-called matrixed methods. Examples are Dolby ProLogic, Logic7 or Circle Surround. The data, in particular the multichannel representation, can additionally be coded with parametric methods, as is the case with MP3 surround, this method being based on the BCC technique.

Depending on the circumstances, the inventive method for generating can be implemented in hardware or in software. The implementation may be on a digital storage medium, in particular a floppy disk or CD with electronically readable control signals, which may interact with a programmable computer system such that the method is performed. In general, the invention thus also consists in a computer program product with a program code stored on a machine-readable carrier for carrying out a method according to the invention, when the computer program product runs on a computer. In other words, the invention can thus be realized as a computer program with a program code for carrying out the method when the computer program runs on a computer.

Claims (11)

Apparatus for generating a coded stereo signal of an audio piece or audio data stream having a first stereo channel and a second stereo channel from a multi-channel representation of the audio piece or audio data stream having information over more than two multi-channels, comprising: means (11) for providing the more than two multi-channels from the multi-channel representation; wherein the multi-channel representation has one or more base channels and parameter information for calculating the multi-channels from one or more base channels,
wherein the means (11) for providing is adapted to calculate from the one or more base channels and the parameter information the at least three multi-channels.
means (12) for performing headphone signal processing to produce an uncoded stereo signal having an uncoded first stereo channel (10a) and an uncoded second stereo channel (10b), the means (12) being adapted to pass through,
for each multi-channel having a first filter function (H _iL ) derived from a virtual position of a loudspeaker for reproducing the multi-channel and a virtual first ear position of a listener for the first stereo channel and with a second filter function (H _iR ) derived from a virtual position of the loudspeaker and a virtual second ear position of the listener for the second Evaluate a stereo channel to produce for each multi-channel a first weighted channel and a second weighted channel, the two virtual ear positions of the listener being different,
to add up the weighted first channels (22) to obtain the uncoded first stereo channel (10a), and
to add up the weighted second channels (23) to obtain the uncoded second stereo channel (10b); and
a stereo encoder (13) for encoding the first uncoded stereo channel (10a) and the uncoded second stereo channel (10b) to obtain the encoded stereo signal (14), the stereo encoder being arranged a data rate necessary for transmitting the encoded stereo signal is less than a data rate necessary for transmitting the uncoded stereo signal. The apparatus of claim 1, wherein the means (12) is _adapted to perform to take into account the first filter function (H _iL ), the direct sound, reflections and diffuse reverberation, and the second filter function (H _iR ), the direct sound, reflections and diffuse Reverberation considered, to use. The apparatus of claim 2, wherein the first and second filter functions correspond to a filter impulse response having a peak at a low time value representing the direct sound, a plurality of smaller peaks at mean time values representing the reflections, and a continuous range that is no longer is resolved after individual peaks and represents the diffuse reverberation. Device according to claim 1,
wherein the means (11) for providing is adapted to provide a block-wise frequency domain representation for each multichannel on the output side, and
in which the means (12) for performing is designed to evaluate the block-wise frequency domain representation with a frequency domain representation of the first and the second filter function. Device according to one of the preceding claims,
wherein the means (12) is arranged to perform to provide a block-wise frequency domain representation of the uncoded first stereo channel and the uncoded second stereo channel, and
wherein the stereo encoder (13) is a transform based encoder and is further configured to process the block-wise frequency domain representation of the uncoded first stereo channel and the uncoded second stereo channel without conversion from the frequency domain representation to a temporal representation. Device according to one of the preceding claims,
wherein the stereo encoder (13) is adapted to perform a common stereo coding (15) of the first and the second stereo channel. Device according to one of the preceding claims,
wherein the stereo encoder (13) is adapted to quantize a block of spectral values using a psychoacoustic masking threshold (16) and an entropy coding (17) to obtain the coded stereo signal. Device according to one of the preceding claims,
in which the device (11) is designed to be provided as a BCC decoder. Device according to one of the preceding claims,
in which the means (11) for providing is designed as a multi-channel decoder which has a filter bank with a plurality of outputs,
wherein the means (12) is adapted to perform to evaluate signals at the filter bank outputs with the first and the second filter function, and
in which the stereo coder (13) is designed to quantize (16) and to apply an entropy coding (17) to the uncoded first stereo channel present in the frequency domain and the uncoded second stereo channel present in the frequency domain to get encoded stereo signal. A method of generating a coded stereo signal of an audio piece or audio data stream having a first stereo channel and a second stereo channel from a multi-channel representation of the audio piece or audio data stream having information over more than two multi-channels, comprising the steps of: Providing (11) the more than two multi-channels from the multi-channel representation; wherein the multi-channel representation includes one or more base channels and parameter information for calculation having multi-channels of one or more base channels,
wherein when providing from the one or more base channels and the parameter information, the at least three multi-channels are calculated;
Performing (12) headphone signal processing to produce an uncoded stereo signal having an uncoded first stereo channel (10a) and an uncoded second stereo channel (10b), the step of performing (12) comprising the steps of: Evaluating each multi-channel with a first filter function (H _iL ) derived from a virtual position of a loudspeaker to reproduce the multi-channel and a virtual first ear position of a listener, for the first stereo channel and with a second filter function (H _iR ) derived from a virtual position of the loudspeaker and a virtual second ear position of the listener for the second stereo channel to generate a first weighted channel and a second weighted channel for each multi-channel, the two virtual ear positions of the listener are different, Adding (22) the weighted first channels to obtain the uncoded first stereo channel (10a), and Adding (23) the weighted second channels to obtain the uncoded second stereo channel (10b); and Stereo coding (13) the first uncoded stereo channel (10a) and the uncoded second stereo channel (10b) to obtain the encoded stereo signal (14), wherein the step of stereo encoding is performed such that a data rate for transmission of the coded stereo signal is less than a data rate necessary to transmit the uncoded stereo signal. A computer program comprising program code for performing the method of generating a coded stereo signal according to claim 10 when the computer program is run on a computer.

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