Patent Description:
The present invention relates to the field of audio signal processing, and discloses methods and systems for efficient estimation of dialogue components, in particular for audio signals having spatialization components, sometimes referred to as immersive audio content.

Content creation, coding, distribution and reproduction of audio are traditionally performed in a channel based format, that is, one specific target playback system is envisioned for content throughout the content ecosystem. Examples of such target playback systems audio formats are mono, stereo, <NUM>, <NUM>, and the like, and we refer to these formats as different presentations of the original content. The above mentioned presentations are typically played back over loudspeakers but a notable exception is the stereo presentation which also commonly is played back directly over headphones.

One specific presentation is the binaural presentation, typically targeting playback on headphones. Distinctive to a binaural presentation is that it is a two-channel signal with each signal representing the content as perceived at, or close to, the left and right eardrum respectively. A binaural presentation can be played back directly over loudspeakers, but preferably the binaural presentation is transformed into a presentation suitable for playback over loudspeakers using cross-talk cancellation techniques.

Different audio reproduction systems have been introduced above, like loudspeakers in different configurations, for example stereo, <NUM>, and <NUM>, and headphones. It is understood from the examples above that a presentation of the original content has a natural, intended, associated audio reproduction system, but can of course be played back on a different audio reproduction system.

If content is to be reproduced on a different playback system than the intended one, a downmixing or upmixing process can be applied. For example, <NUM> content can be reproduced over a stereo playback system by employing specific downmix equations. Another example is playback of stereo encoded content over a <NUM> speaker setup, which may comprise a so-called upmixing process, that could or could not be guided by information present in the stereo signal. A system capable of upmixing is Dolby Pro Logic from Dolby Laboratories Inc (Roger Dressler, "Dolby Pro Logic Surround Decoder, Principles of Operation", www.

An alternative audio format system is an audio object format such as that provided by the Dolby Atmos system. In this type of format, objects or components are defined to have a particular location around a listener, which may be time varying. Audio content in this format is sometimes referred to as immersive audio content. It is noted that within the context of this application an audio object format is not considered a presentation as described above, but rather a format of the original content that is rendered to one or more presentations in an encoder, after which the presentation(s) is encoded and transmitted to a decoder.

When multi-channel and object based content is to be transformed into a binaural presentation as mentioned above, the acoustic scene consisting of loudspeakers and objects at particular locations is simulated by means of head-related impulse responses (HRIRs), or binaural room impulse responses (BRIRs), which simulate the acoustical pathway from each loudspeaker/object to the ear drums, in an anechoic or echoic (simulated) environment, respectively. In particular, audio signals can be convolved with HRIRs or BRIRs to re-instate inter-aural level differences (ILDs), inter-aural time differences (ITDs) and spectral cues that allow the listener to determine the location of each individual loudspeaker/object. The simulation of an acoustic environment (reverberation) also helps to achieve a certain perceived distance. <FIG> illustrates a schematic overview of the processing flow for rendering two object or channel signals xi <NUM>, <NUM>, being read out of a content store <NUM> for processing by <NUM> HRIRs e.g. <NUM>. The HRIR outputs are then summed <NUM>, <NUM>, for each channel signal, so as to produce headphone speaker outputs for playback to a listener via headphones <NUM>. The basic principle of HRIRs is, for example, explained in <NPL>.

The HRIR/BRIR convolution approach comes with several drawbacks, one of them being the substantial amount of convolution processing that is required for headphone playback. The HRIR or BRIR convolution needs to be applied for every input object or channel separately, and hence complexity typically grows linearly with the number of channels or objects. As headphones are often used in conjunction with battery-powered portable devices, a high computational complexity is not desirable as it may substantially shorten battery life. Moreover, with the introduction of object-based audio content, which may comprise say more than <NUM> objects active simultaneously, the complexity of HRIR convolution can be substantially higher than for traditional channel-based content.

For this purpose, co-pending and non-published <CIT>, describes a dual-ended approach for presentation transformations that can be used to efficiently transmit and decode immersive audio for headphones. The coding efficiency and decoding complexity reduction are achieved by splitting the rendering process across encoder and decoder, rather than relying on the decoder alone to render all objects.

A part of the content which during creation is associated with a specific spatial location is referred to as an audio component. The spatial location can be a point in space or a distributed location. Audio components can be thought of as all the individual audio sources that a sound artist mixes, i.e., positions spatially, into a soundtrack. Typically a semantic meaning (e.g. dialogue) is assigned to the components of interest so that the goal of the processing (e.g. dialogue enhancement) becomes defined. It is noted that audio components that are produced during content creation are typically present throughout the processing chain, from the original content to different presentations. For example, in an object format there can be dialogue objects with associated spatial locations. And in a stereo presentation there can be dialogue components that are spatially located in the horizontal plane.

In some applications, it is desirable to extract dialogue components in the audio signal, in order to e.g. enhance or amplify such components. The goal of dialogue enhancement (DE) may be to modify the speech part of a piece of content that contains a mix of speech and background audio so that the speech becomes more intelligible and/or less fatiguing for an end-user. Another use of DE is to attenuate dialogue that for example is perceived as disturbing by an end-user. There are two fundamental classes of DE methods: encoder side and decoder side DE. Decoder side DE (called single ended) operates solely on the decoded parameters and signals that reconstruct the non-enhanced audio, i.e., no dedicated side-information for DE is present in the bitstream. In encoder side DE (called dual ended), dedicated side-information that can be used to do DE in the decoder is computed in the encoder and inserted in the bitstream.

<FIG> shows an example of dual ended dialogue enhancement in a conventional stereo example. Here, dedicated parameters <NUM> are computed in the encoder <NUM> that enable extraction of the dialogue <NUM> from the decoded non-enhanced stereo signal <NUM> in the decoder <NUM>. The extracted dialogue is level modified, e.g. boosted <NUM> (by an amount partially controlled by the end-user) and added to the non-enhanced output <NUM> to form the final output <NUM>. The dedicated parameters <NUM> can be extracted blindly from the non-enhanced audio <NUM> or exploit a separately provided dialogue signal <NUM> in the parameter computations.

Another approach is disclosed in <CIT>. Here, the bitstream to the decoder includes an object downmix signal (e.g. a stereo presentation), object parameters to enable reconstruction of the audio objects, and object based metadata allowing manipulation of the reconstructed audio objects. As indicated in <FIG> of <CIT>, the manipulation may include amplification of speech related objects. This approach thus requires the reconstruction of the original audio objects on the decoder side, which typically is computationally demanding.

The international search report cites <NPL> (hereinafter "D1") and <CIT> (hereinafter "D2").

D1 describes the architecture and features of the Spatial Audio Object Coding for Dialogue Enhancement (SAOC-DE), which was closely derived from the well-known MPEG-D Spatial Audio Object Coding (SAOC).

D2 describes a decoder for generating an audio output signal having one or more audio output channels, having a receiving interface for receiving an audio input signal having a plurality of audio object signals, for receiving loudness information on the audio object signals, and for receiving rendering information indicating whether one or more of the audio object signals shall be amplified or attenuated, further having a signal processor for generating the one or more audio output channels of the audio output signal, configured to determine a loudness compensation value depending on the loudness information and depending on the rendering information, and configured to generate the one or more audio output channels of the audio output signal from the audio input signal depending on the rendering information and depending on the loudness compensation value. One or more by-pass audio object signals are employed for generating the audio output signal.

There is a general desire to provide dialogue estimation efficiently also in a binaural context.

It is an object of the invention to provide efficient dialogue enhancement in a binaural context, i.e. when at least one of the audio presentations that the dialogue component(s) is extracted from, or the audio presentation to which the extracted dialogue is added to, is a (echoic or anechoic) binaural representation.

In accordance with a first aspect of the present invention, there is provided a method for dialogue enhancing audio content having one or more audio components, wherein each component is associated with a spatial location, according to claim <NUM>.

In accordance with a second aspect of the present invention, there is provided a method for dialogue enhancing audio content having one or more audio components, wherein each component is associated with a spatial location, according to claim <NUM>.

In accordance with a third aspect of the present invention, there is provided a decoder for dialogue enhancing audio content having one or more audio components, wherein each component is associated with a spatial location, according to claim <NUM>.

In accordance with a fourth aspect of the present invention, there is provided a decoder for dialogue enhancing audio content having one or more audio components, wherein each component is associated with a spatial location, according to claim <NUM>.

The invention is based on the insight that a dedicated parameter set provides an efficient way to extract a dialogue presentation from one audio signal presentation which is then combined with another audio signal presentation, where at least one of the presentations is a binaural presentation. It is noted that according to the invention, it is not necessary to reconstruct the original audio objects in order to enhance dialogue. Instead, the dedicated parameters are applied directly on a presentation of the audio objects, e.g. a binaural presentation, a stereo presentation, etc. The inventive concept enables a variety of specific embodiments, each with specific advantages.

It is noted that the expression "dialogue enhancement" here is not restricted to amplifying or boosting dialogue components, but may also relate to attenuation of selected dialogue components. Thus, in general the expression "dialogue enhancement" refers to a level-modification of one or more dialogue related components of the audio content. The gain factor G of the level modification may be less than zero in order to attenuate dialogue, or greater than zero in order to enhance dialogue.

In some embodiments, the first and second presentations are both (echoic or anechoic) binaural presentations. In case only one of them binaural, the other presentation may be a stereo or surround audio signal presentation.

In the case of different presentations, the dialogue estimation parameters may be configured to also perform a presentation transform, so that the dialogue presentation corresponds to the second audio signal presentation.

The invention may advantageously be implemented in a particular type of a so called simulcast system, where the encoded bit stream also includes a set of transform parameters suitable for transforming the first audio signal presentation to a second audio signal presentation.

Systems and methods disclosed in the following may be implemented as software, firmware, hardware or a combination thereof. In a hardware implementation, the division of tasks referred to as "stages" in the below description does not necessarily correspond to the division into physical units; to the contrary, one physical component may have multiple functionalities, and one task may be carried out by several physical components in cooperation. Certain components or all components may be implemented as software executed by a digital signal processor or microprocessor, or be implemented as hardware or as an application-specific integrated circuit. Such software may be distributed on computer readable media, which may comprise computer storage media (or non-transitory media) and communication media (or transitory media). As is well known to a person skilled in the art, the term computer storage media includes both volatile and non-volatile, removable and non-removable media implemented in any method or technology for storage of information such as computer readable instructions, data structures, program modules or other data. Computer storage media includes, but is not limited to, RAM, ROM, EEPROM, flash memory or other memory technology, CD-ROM, digital versatile disks (DVD) or other optical disk storage, magnetic cassettes, magnetic tape, magnetic disk storage or other magnetic storage devices, or any other medium which can be used to store the desired information and which can be accessed by a computer. Further, it is well known to the skilled person that communication media typically embodies computer readable instructions, data structures, program modules or other data in a modulated data signal such as a carrier wave or other transport mechanism and includes any information delivery media.

Various ways to implement embodiments of the invention will be discussed with reference to <FIG>. All these embodiments generally relate to a system and method for applying dialogue enhancement to an input audio signal having one or more audio components, wherein each component is associated with a spatial location. The illustrated blocks are typically implemented in a decoder.

In the presented embodiments the input signals are preferably analyzed in time/frequency tiles, for example by means of a filter bank such as a quadrature mirror filter (QMF) bank, a discrete Fourier transform (DFT), a discrete cosine transform (DCT), or any other means to split input signals into a variety of frequency bands. The result of such a transform is that an input signal xi[n] for input with index i and discrete-time index n is represented by sub-band signals xi[b, k] for time slot (or frame) k and sub-band b. Consider for example the estimation of the binaural dialogue presentation from a stereo presentation. Let xj[b, k], j = <NUM>,<NUM> denote the sub-band signals of the left and right stereo channels, and d̂i[b,k], i = <NUM>,<NUM> denote the sub-band signals of the estimated left and right binaural dialogue signals. The dialogue estimate may be computed like <MAT> with Bp, K sets of frequency (b) and time (k) indices corresponding to a desired time/frequency tile, p the parameter band index, and m a convolution tap index, and <MAT> a matrix coefficient belonging to input index j, parameter band Bp, sample range or time slot K, output index i, and convolution tap index m. Using the above formulation, the dialogue is parameterized by the parameters w (relative to the stereo signal; J=<NUM> in this case of a stereo signal). The number of time slots in the set K can be independent of, and constant with respect to frequency and is typically chosen to correspond to a time interval of <NUM>-<NUM>. The number P of sets of frequency indices is typically between <NUM>-<NUM> with the number of frequency indices in each set typically increasing with increasing frequency to reflect properties of hearing (higher frequency resolution in the parameterization toward low frequencies).

The dialogue parameters w may be computed in the encoder, and encoded using techniques disclosed in <CIT>. The parameters w are then transmitted in the bitstream and decoded by a decoder prior to application using the above equation. Due to the linear nature of the estimate the encoder computation can be implemented using minimum mean squared error (MMSE) methods in cases where the target signal (the clean dialogue or an estimate of the clean dialogue) is available.

The choice of P, and the choice of the number of time slots in K is a trade-off between quality and bit rate. Furthermore, the parameters w can be constrained in order to lower the bit rate (at the cost of lower quality), e.g., by assuming <MAT> when i ≠ j and simply not transmitting those parameters. The choice of M is also a quality/bitrate trade-off, see <CIT>. The parameters w are in general complex valued since the binauralization of the signals introduces ITDs (phase differences). However, the parameters can be constrained to be real-valued in order to lower the bit rate. Furthermore, it is well-known that humans are insensitive to phase and time differences between the signals in the left and right ear above a certain frequency, the phase/magnitude cut-off frequency, around <NUM>-<NUM>, thus above that frequency, binaural processing is typically done so that no phase difference is introduced between the left and right binaural signals, and hence parameters can be real-valued with no loss in quality (cf. The above quality/bit rate trade-offs can be done independently in each time/frequency tile.

In general it is proposed to use estimators of the form <MAT> where at least one of ŷ and x is a binaural signal, i.e., I=<NUM> or J=<NUM> or I=J=<NUM>. For notational convenience we will in the following often omit the time/frequency tile indexing Bp, K as well as the i,j,m indexing when referring to different parameter sets used to estimate dialogue.

The above estimator can conveniently be expressed in matrix notation as (omitting the time/frequency tile indexing for ease of notation) <MAT> where Xm = [x<NUM> (m). xJ(m)] and Ŷ = [ŷ<NUM>. ŷI] contain vectorized versions of xj[b, k - m] and ŷi[b, k] respectively in the columns, and Wm is a parameter matrix with J rows and I columns. The above form of the estimator may be used when performing only dialogue extraction, or when performing only a presentation transform, as well as in the case where both extraction and presentation transform is done using a single set of parameters as is detailed in embodiments below.

With reference to <FIG>, a first audio signal presentation <NUM> has been rendered from an immersive audio signal including a plurality of spatialized audio components. This first audio signal presentation is provided to a dialogue estimator <NUM>, in order to provide a presentation <NUM> of one or several extracted dialogue components. The dialogue estimator <NUM> is provided with a dedicated set of dialogue estimation parameters <NUM>. The dialogue presentation is level modified (e.g. boosted) by gain block <NUM>, and then combined with a second presentation <NUM> of the audio signal to form a dialogue enhanced output <NUM>. As will be discussed below, the combination may be a simple summation, but may also involve a summation of the dialogue presentation with the first presentation, before applying a transform to the sum, thereby forming the dialogue enhanced second presentation.

According to the present invention, at least one of the presentations is a binaural presentation (echoic or anechoic). As will be further discussed in the following, the first and second presentations may be different, and the dialogue presentation may or may not correspond to the second presentation. For example, the first audio signal presentation may be intended for playback on a first audio reproduction system, e.g. a set of loudspeakers, while the second audio signal presentation may be intended for playback on a second audio reproduction system, e.g. headphones.

In the decoder embodiment in <FIG>, the first and second presentations <NUM>, <NUM>, as well as the dialogue presentation <NUM>, are all (echoic or anechoic) binaural presentations. The (binaural) dialogue estimator <NUM> - and the dedicated parameters <NUM> - is thus configured to estimate binaural dialogue components which are level modified in block <NUM> and added to the second audio presentation <NUM> to form output <NUM>.

In the embodiment in <FIG>, the parameters <NUM> are not configured to perform any presentation transform. Still, for best quality, the binaural dialogue estimator <NUM> should be complex valued in frequency bands up to the phase/magnitude cut-off frequency. To explain why complex valued estimators can be needed even when no presentation transform is done consider estimation of binaural dialogue from a binaural signal that is a mix of binaural dialogue and other binaural background content. Optimal extraction of dialogue often includes subtracting portions of say the right binaural signal from the left binaural signal to cancel background content. Since the binaural processing, by nature, introduces time (phase) differences between left and right signals, those phase differences must be compensated for prior to any subtraction can be done, and such compensation requires complex valued parameters. Indeed, when studying the result of MMSE computation of parameters the parameters in general come out as complex valued if not constrained to be real valued. In practice the choice of complex vs real valued parameters is a trade-off between quality and bit rate. As mentioned above, parameters can be real-valued above the frequency phase/magnitude cut-off frequency without any loss in quality by exploiting the insensitivity to fine-structure waveform phase differences at high frequencies.

In the decoder embodiment in <FIG>, the first and second presentations are different. In the illustrated example, the first presentation <NUM> is a non-binaural presentation (e.g. stereo <NUM>, or surround <NUM>), while the second presentation <NUM> is a binaural presentation. In this case, the set of dialogue estimation parameters <NUM> are configured to allow the binaural dialogue estimator <NUM> to estimate a binaural dialogue presentation <NUM> from a non-binaural presentation <NUM>. It is noted that the presentations could be reversed, in which case the binaural dialogue estimator would e.g. estimate a stereo dialogue presentation from a binaural audio presentation. In either case, the dialogue estimator needs to extract dialogue components and perform a presentation transform. The binaural dialogue presentation <NUM> is level modified by block <NUM> and added to the second presentation <NUM>.

As indicated in <FIG>, the binaural dialogue estimator <NUM> receives one single set of parameters <NUM>, configured to perform the two operations of dialogue extraction and presentation transform. However, as indicated in <FIG>, it is also possible that an (echoic or anechoic) binaural dialogue estimator <NUM> receives two sets of parameters D1, D2; one set (D1) configured to extract dialogue (dialogue extraction parameters) and one set (D2) configured to perform the dialogue presentation transform (dialogue transform parameters). This may be advantageous in an implementation where one or both of these subsets D1, D2 are already available in the decoder. For example, the dialogue extraction parameters D1 may be available for conventional dialogue extraction as illustrated in <FIG>. Further, the parameter transform parameters D2 may be available in a simulcast implementation, as discussed below.

In <FIG>, the dialogue extraction (block 62a) is indicated as occurring before the presentation transform (block 62b), but this order may of course equally well be reversed. It is also noted that for reasons of computational efficiency, even if the parameters are provided as two separate sets D1, D2, it may be advantageous to first combine the two sets of parameters into one combined matrix transform, before applying this combined transform to the input signal <NUM>.

Further, it is noted that the dialogue extraction can be one dimensional, such that the extracted dialogue is a mono representation. The transform parameters D2 are then positional metadata, and the presentation transform comprises rendering the mono dialogue using HRTFs, HRIRs or BRIRs corresponding to the position. Alternatively, if the desired rendered dialogue presentation is intended for loudspeaker playback, the mono dialogue could be rendered using loudspeaker rendering techniques such as amplitude panning or vector-based amplitude panning (VBAP).

<FIG> show embodiments of the present invention in the context of a simulcast system, i.e. a system where one audio presentation is encoded and transmitted to a decoder together with a set of transform parameters which enable the decoder to transform the audio presentation into a different presentation adapted to the intended playback system (e.g. as indicated a binaural presentation for headphones). Various aspects of such a system is described in detail in co-pending and non-published <CIT>. For simplicity, <FIG> only illustrate the decoder side.

As illustrated in <FIG>, a core decoder <NUM> receives an encoded bitstream <NUM> including an initial audio signal presentation of the audio components. In the illustrated case this initial presentation is a stereo presentation z, but it may also be any other presentation. The bitstream <NUM> also includes a set of presentation transform parameters w(y) which are used as matrix coefficients to perform a matrix transform <NUM> of the stereo signal z to generate a reconstructed anechoic binaural signal ŷ. The transform parameters w(y) have been determined in the encoder as discussed in <CIT>. In the illustrated case, the bitstream <NUM> also includes a set of parameters w(f) which are used as matrix coefficients to perform a matrix transform <NUM> of the stereo signal z to generate a reconstructed input signal f̂ for an acoustic environment simulation, here a feedback delay network (FDN) <NUM>. These parameters w(f) have been determined in a similar way as the presentation transform parameters w(y). The FDN <NUM> receives the input signal f̂ and provides an acoustic environment simulation output FDNout which may be combined with the anechoic binaural signal ŷ to provide an echoic binaural signal.

In the embodiment in <FIG>, the bitstream further includes a set of dialogue estimation parameters w(D) which are used as matrix coefficients in a dialogue estimator <NUM> to perform a matrix transform of the stereo signal z to generate an anechoic binaural dialogue presentation D. The dialogue presentation D is level modified (e.g. boosted) in block <NUM>, and combined with the reconstructed anechoic signal ŷ and the acoustic environment simulation output FDNout in summation block <NUM>.

<FIG> is essentially an implementation of the embodiment in <FIG> in a simulcast context.

In <FIG>, a stereo signal z, a set of transform parameters w(y) and a further set of parameters w(f) are received and decoded just as in <FIG>, and elements <NUM>, <NUM>, <NUM>, <NUM>, and <NUM> are equivalent to those discussed with respect to <FIG>. Further, the bitstream <NUM> here also includes a set of dialogue estimation parameters w(D1) which are applied by a dialogue estimator <NUM> on the signal z. However, in this embodiment, the dialogue estimation parameters w(D1) are not configured to provide any presentation transform. The dialogue presentation output Dstereo from the dialogue estimator <NUM> therefore corresponds to the initial audio signal presentation, here a stereo presentation. This dialogue presentation Dstereo is level modified in block <NUM>, and then added to the signal z in the summation <NUM>. The dialogue enhanced signal (z + Dstereo) is then transformed by the set of transform parameters w(y).

<FIG> can be seen as an implementation of the embodiment in <FIG> in a simulcast context, where w(D1) is used as D1 and w(y) is used as D2. However, while in <FIG> both sets of parameters are applied in the dialogue estimator <NUM>, in <FIG> the extracted dialogue Dstereo is added to the signal z and the transform w(y) is applied to the combined signal (z + D).

It is noted that the set of parameters w(D1) may be identical to the dialogue enhancement parameters used to provide dialogue enhancement of the stereo signal in a simulcast implementation. This alternative is illustrated in <FIG>, where the dialogue extraction 96a is indicated as forming part of the core decoder <NUM>. Further, in <FIG>, a presentation transform 96b using the parameter set w(y) is performed before the gain, separately from the transformation of the signal z. This implementation is thus even more similar to the case shown in <FIG>, with the dialogue estimator <NUM> comprising both transforms 96a, 96b.

<FIG> shows a modified version of <FIG>. In this case the presentation transform is not performed using the parameter set w(y), but with an additional set of parameters w(D2) which is provided in a part of the bitstream dedicated to binaural dialogue estimation.

In one aspect related to the invention but not comprising all features of the independent claims, the aforementioned dedicated presentation transform w(D2) in <FIG> is a real-valued, single-tap (M=<NUM>), full-band (P=<NUM>) matrix.

<FIG> shows a modified version of <FIG>. In this case, the dialogue extractor 96a again provides a stereo dialogue presentation Dstereo, and is again indicated as forming part of the core decoder <NUM>. Here, however, the stereo dialogue presentation Dstereo, after level modification in block <NUM>, is added directly to the anechoic binaural signal ŷ (together with the acoustic environment simulation from the FDN).

It is noted that combining signals with different presentations, e.g., summing a stereo dialogue signal to a binaural signal (which contains non-enhanced binaural dialogue components) naturally leads to spatial imaging artifacts since the non-enhanced binaural dialogue components are perceived to be spatially different compared to a stereo presentation of the same components.

It is further noted that combining signals with different presentations can lead to constructive summing of dialogue components in certain frequency bands, and destructive summing in other frequency bands. The reason for this is that binaural processing introduces ITDs (phase differences) and we are summing signals that are in-phase in certain frequency bands and out-of-phase in other bands, leading to coloring artifacts in the dialogue components (moreover the coloring can be different in the left and right ear). In one embodiment, phase differences above the phase/magnitude cut-off frequency are avoided in the binaural processing so as to reduce this type of artifact.

As a final note to the case of combining signals with different presentations it is acknowledged that in general, binaural processing can reduce the intelligibility of dialogue. In cases where the goal of dialogue enhancement is to maximize intelligibility, it may be advantageous to extract and level modify (e.g. boost) a dialogue signal that is non-binaural. To elaborate further, even if the final presentation intended for playback is binaural, it may be advantageous in such a case to extract and level modify (e.g. boost) a stereo dialogue signal and combine that with the binaural presentation (trading off coloring artifacts and spatial imaging artifacts as described above, for increased intelligibility).

In the embodiment in <FIG>, a stereo signal z, a set of transform parameters w(y) and a further set of parameters w(f) are received and decoded just as in <FIG>. Further, similar to <FIG>, the bitstream also includes a set of dialogue estimation parameters w(D1) which are not configured to provide any presentation transform. However, in this embodiment, the dialogue estimation parameters w(D1) are applied by the dialogue estimator <NUM> on the reconstructed anechoic binaural signal ŷ to provide an anechoic binaural dialogue presentation D. This dialogue presentation D is level modified by a block <NUM> and added in summation <NUM> to the signal ŷ together with FDNout.

<FIG> is essentially an implementation of the single presentation embodiment in <FIG> in a simulcast context. However, it can also be seen as an implementation of <FIG> with a reversed order of D1 and D2, where again w(D1) is used as D1 and w(y) is used as D2. However, while in <FIG> both sets of parameters are applied in the dialogue estimator, in <FIG> the transform parameters D2 have already been applied in order to obtain ŷ, and the dialogue estimator <NUM> only needs to apply the parameters w(D1) to the signal ŷ in order to obtain the echoic binaural dialogue presentation D.

In some applications, it may be desirable to apply different processing depending on the desired value of the dialogue level modification factor G. In one embodiment, example, appropriate processing is selected based on a determination of whether the factor G is greater than or smaller than a given threshold. Of course, there may also be more than one threshold, and more than one alternative processing. For example, a first processing when G<th1, a second processing when th1<=G<th2, and a third processing when G>=th2, where th1 and th2 are two given threshold values.

In a specific example, illustrated in <FIG>, the threshold is zero, and first processing is applied when G<<NUM> (attenuation of dialogue), while a second processing is applied when G><NUM> (enhancement of dialogue). For this purpose, the circuit in <FIG> includes selection logic in the form of a switch <NUM> with two positions A and B. The switch is provided with the value of the gain factor G from block <NUM>, and is configured to assume position A when G<<NUM>, and position B when G><NUM>.

When the switch is in position A, the circuit is here configured to combine the estimated stereo dialogue from matrix transform <NUM> with the stereo signal z, and then perform the matrix transform <NUM> on the combined signal to generate a reconstructed anechoic binaural signal. The output from the feedback delay network <NUM> is then combined with this signal in <NUM>. It is noted that this processing essentially corresponds to <FIG> discussed above.

When the switch is in position B, the circuit is here configured to apply transform parameters w(D2) to the stereo dialogue from matrix transform <NUM> in order to provide a binaural dialogue estimation. This estimation is then added to the anechoic binaural signal from transform <NUM>, and output from the feedback delay network <NUM>. It is noted that this processing essentially corresponds to <FIG> discussed above.

The skilled person will realize many other alternatives for the processing in position A and B, respectively. For example, the processing when the switch is in position B could instead correspond to that in <FIG>. However, the main contribution of the embodiment in <FIG> is the introduction of the switch <NUM>, which enables alternative processing depending on the value of the gain factor G.

As used herein, unless otherwise specified the use of the ordinal adjectives "first", "second", "third", etc., to describe a common object, merely indicate that different instances of like objects are being referred to, and are not intended to imply that the objects so described must be in a given sequence, either temporally, spatially, in ranking, or in any other manner.

As used herein, the term "exemplary" is used in the sense of providing examples, as opposed to indicating quality. That is, an "exemplary embodiment" is an embodiment provided as an example, as opposed to necessarily being an embodiment of exemplary quality.

It should be appreciated that in the above description of exemplary embodiments of the invention, various features of the invention are sometimes grouped together in a single embodiment, FIG. , or description thereof for the purpose of streamlining the disclosure and aiding in the understanding of one or more of the various inventive aspects. Thus, the claims following the Detailed Description are hereby expressly incorporated into this Detailed Description, with each claim standing on its own as a separate embodiment of this invention.

Claim 1:
A method for dialogue enhancing audio content having one or more audio components, wherein each component is associated with a spatial location, comprising:
providing a first audio signal presentation (<NUM>, <NUM>, <NUM>) of the audio components intended for reproduction on a first audio reproduction system;
providing a second audio signal presentation (<NUM>, <NUM>, <NUM>) of the audio components intended for reproduction on a second audio reproduction system;
receiving a set of dialogue estimation parameters (<NUM>, <NUM>, <NUM>, D1) configured to enable estimation of dialogue components from the first audio signal presentation;
applying said set of dialogue estimation parameters to said first audio signal presentation, to form a dialogue presentation (<NUM>, <NUM><NUM>) of the dialogue components; and
combining the dialogue presentation with said second audio signal presentation to form a dialogue enhanced audio signal presentation (<NUM>, <NUM>) for reproduction on the second audio reproduction system;
wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.