EP3058564A1 - Spatialisation sonore avec effet de salle, optimisee en complexite - Google Patents
Spatialisation sonore avec effet de salle, optimisee en complexiteInfo
- Publication number
- EP3058564A1 EP3058564A1 EP14796814.3A EP14796814A EP3058564A1 EP 3058564 A1 EP3058564 A1 EP 3058564A1 EP 14796814 A EP14796814 A EP 14796814A EP 3058564 A1 EP3058564 A1 EP 3058564A1
- Authority
- EP
- European Patent Office
- Prior art keywords
- frequency
- signal
- transfer function
- room
- sound
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
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- 230000003595 spectral effect Effects 0.000 claims abstract description 58
- 230000005236 sound signal Effects 0.000 claims abstract description 22
- 230000006870 function Effects 0.000 claims description 88
- 230000000694 effects Effects 0.000 claims description 41
- 238000000034 method Methods 0.000 claims description 20
- 238000004364 calculation method Methods 0.000 claims description 11
- 238000005070 sampling Methods 0.000 claims description 9
- 230000006835 compression Effects 0.000 claims description 6
- 238000007906 compression Methods 0.000 claims description 6
- 230000003447 ipsilateral effect Effects 0.000 claims description 5
- 238000012216 screening Methods 0.000 claims description 5
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- 238000010276 construction Methods 0.000 description 2
- 230000003111 delayed effect Effects 0.000 description 2
- 210000005069 ears Anatomy 0.000 description 2
- 210000002837 heart atrium Anatomy 0.000 description 2
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- 210000005245 right atrium Anatomy 0.000 description 2
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Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
- H04S7/307—Frequency adjustment, e.g. tone control
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
- H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
- H04S7/306—For headphones
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
- G10K15/00—Acoustics not otherwise provided for
- G10K15/08—Arrangements for producing a reverberation or echo sound
- G10K15/12—Arrangements for producing a reverberation or echo sound using electronic time-delay networks
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
- G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
- H04S2400/01—Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
- H04S2420/01—Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
- H04S2420/07—Synergistic effects of band splitting and sub-band processing
Definitions
- the present invention relates to a sound spatialization with room effect.
- the invention finds an advantageous but non-limiting application to a processing of sound signals respectively coming from L channels associated with virtual loudspeakers (for example in a multichannel representation, or in a surround representation, of the sound to be reproduced), for a spatialized reproduction on real loudspeakers (for example two headsets of a headset in binaural restitution, or two separate speakers in transaural restitution).
- L channels associated with virtual loudspeakers for example in a multichannel representation, or in a surround representation, of the sound to be reproduced
- real loudspeakers for example two headsets of a headset in binaural restitution, or two separate speakers in transaural restitution.
- the signal of one of these channels can be processed to have a first contribution on the left atrium and a second contribution on the right atrium, binaural restitution, applying in particular a transfer function with effect of room for each of these contributions.
- the application of these room effect transfer functions thus contributes to offering the listener a feeling of immersion, practically allowing him to "locate in space” the virtual speaker associated with this channel.
- a room effect transfer function is applied to each sound signal of a corresponding channel, in the time domain in the form of a BRIR type impulse response (for "Binaural Room Impulse Response "or” Binaural Impulse Room Response ").
- this transfer function BRIR is constructed as the combination:
- a second transfer function global, common to all signals and characterized in particular a diffuse field, whose presence usually occurs in a room after a certain time, typically after the first reflections of a sound wave.
- Such an embodiment advantageously makes it possible to apply a treatment common to all the signals, which corresponds, in a physical reality, to a "mixture" of the waves acoustic as and reverberations, so beyond a given duration (characterizing a beginning of presence of the diffuse field).
- Such an embodiment then makes it possible to reduce the complexity of spatialization processing with room effect on several initial channels.
- the channel signals are received in encoded form by a decoder in compression.
- This decoder sends the signals of the channels, once decoded, to a spatialization module for a sound reproduction with room effect, on two speakers. It is then appropriate for this spatialization step (which follows the decoding of the received signals) to be of reduced processing complexity so as not to delay the overall set of decoding and spatialization steps on reception of the signals before restitution.
- the present invention improves the situation.
- the invention proposes to reduce the complexity of the application of the room effect transfer function, in particular by reducing this complexity in the spectral domain.
- the convolution by a transfer function becomes a multiplication of spectral components of the signal on the one hand, and a filter representing the transfer function on the other hand ( Figure 1 commented in detail later). ).
- the invention then starts from the advantageous observation that, after a direct propagation, a sound wave tends to attenuate in the high frequencies due to progressive reflections on surfaces (walls typically, the listener's face, etc.) absorbing. the wave especially in the high frequencies.
- the air itself absorbs the spectral components of the highest frequencies of sound during its propagation. This phenomenon is all the more increased for example for the diffuse sound field, for which it is not necessary to have a frequency representation for very high frequencies (for example, higher than a frequency in a range of 5 to 15 kHz ).
- the invention thus aims at a sound spatialization method, comprising the application of at least one room effect transfer function to at least one sound signal, said application being used to multiply, in the spectral domain, the spectral components of the signal sound by the spectral components of a filter corresponding to the aforementioned transfer function.
- Each spectral component of the filter comprises a temporal evolution in a time-frequency representation (as detailed below with reference to FIG. 3).
- these spectral components of the filter are ignored, for the aforementioned component multiplications, beyond a threshold frequency and after at least one given instant in said time-frequency representation.
- the spectral components of the filter are taken into account up to a cutoff frequency which can be chosen for example between 5 and 15 kHz (depending on the room effect to be applied and / or the signal to spatialize, as described below). Beyond the cutoff frequency, the multiplication is not even performed, which mathematically amounts to the same as multiplying the signal by zero.
- This given instant typically represents the moment when a sound wave begins to undergo reverberations (by successive reflections, or, even later, from a presence of a diffuse sound field).
- the transfer function takes into account reverberations in the room effect (taking into account, for example, a diffuse sound field)
- the instant given above can be chosen as a function of such reverberations.
- the instant given above may be posterior, in the room effect, to a direct sound propagation with first reflections, and then correspond to a beginning of presence of diffuse sound field.
- provision may be made in which the aforementioned threshold frequency decreases as a function of time in said time-frequency representation.
- the signal may be provided, for example, to preserve the spectral components present in the signal, in the multiplication of the components, for a first block, and then to ignore them at the same time. beyond a first threshold frequency for a second block following the first block, then ignoring them beyond a second threshold frequency for a third block following the second block, etc., the second threshold frequency being more bass than the first.
- the spectral components of the filter can be ignored, for the multiplication of the components:
- the second threshold frequency being lower than the first threshold frequency
- the given block mentioned above may include, for example, samples located temporally at times that correspond to times when a sound wave has undergone one or more reflections, with even a beginning of presence of diffuse sound field.
- the block that follows this given block may include, for example, samples located temporally after or from a beginning of presence of diffuse sound field.
- Such an embodiment makes it possible, for example, to limit possibly audible signal limiting artifacts in the high frequencies for reverberations, this realization being accomplished progressively over several blocks. It also makes it possible to consider several forms of transfer functions (denoted hereinafter B i (m), m being a block index) characterizing a diffuse sound field. Indeed, it is possible, for example, to apply a transfer function B k mean to a given given block, and to apply a temporally progressive (fade out) breaking window to this transfer function B for the block that follows, to "finish" the presence of the diffuse sound field.
- B i m
- m being a block index
- W K (I) being a weighting weight chosen, and 6 ⁇ J (i)), a predetermined energy compensation gain,
- DD ⁇ being a delay application, counted in number of sample blocks, corresponding to a time difference between a sound emission in a room corresponding to the room effect, and a beginning of diffuse field presence in this room. room, the index m corresponding to a number of blocks of samples of duration corresponding to this delay, M being the total number of blocks that a transfer function lasts in a time-frequency representation,
- a multiplication calculation limitation beyond a first threshold frequency as soon as the first sample block or blocks, as a function of the characteristics of the signal (for example its sampling frequency, or the most frequent frequency). high represented in the spectral components of the signal) or as a function of the applied spatialization characteristics (with for example a limitation of the high frequency components for a contra-lateral acoustic path as detailed below).
- the signal resulting from the reverberations does not normally comprise spectral components of higher frequency than the initial signal.
- the aforementioned threshold frequency can not be greater than this highest frequency.
- the highest frequency spectral component information in the sound signal is obtained, and the aforementioned threshold frequency is selected as the minimum of a predetermined threshold frequency (for example between 5 and 15 kHz) and said highest frequency.
- the highest frequency spectral component information may be provided by the decoder.
- the spatialization is carried out with a module capable of supporting different signal formats, especially in terms of the sampling frequency of such signals, the highest frequency, mentioned above, can not be greater than half of the sampling frequency, and thus the threshold frequency for the implementation of the invention may be further selected according to this sampling frequency.
- the sound signal is spatialized on at least first and second virtual speakers, respectively associated with a first and a second channel, respectively first and second room effect transfer functions are applied to these first and second channels respectively.
- channels as explained above in introduction (for example by adapting signals on surround channels to move to a binaural or transaural restitution).
- one of the first and second transfer functions applies an ipsi-lateral acoustic path effect
- the other of the first and second transfer functions applies a contra-lateral acoustic path effect
- This "screening" frequency is explained by the fact that for a contra-lateral path between a virtual loudspeaker and an ear of the listener, the listener's head masks the acoustic path and absorbs the acoustic tones. More acute acoustic wave (thus eliminates the spectral components associated with the highest frequencies of the acoustic wave).
- the aforementioned threshold frequency for the transfer function applying a contra-lateral path effect, can be chosen as a minimum of a predetermined threshold frequency (for example chosen between 5 and 15 kHz) and this screening frequency.
- a predetermined threshold frequency for example chosen between 5 and 15 kHz
- This embodiment is advantageous for being applied already for the first block of samples.
- it does not exclude the possibility of increasing again the threshold frequency for the next block to simulate a first reflection on a wall situated opposite the considered ear, this first reflection being received at this ear by a path ipsilateral.
- the cutoff frequency can be chosen to be common to all the signals, in one possible embodiment, after a given instant which corresponds, for example, to the presence of the diffuse field.
- each transfer function applied to a signal comprises:
- the instant given above may be common to all the signals and correspond for example to a beginning of presence of diffuse sound field.
- the signals comprise successive blocks of samples, of the same size between signals
- at least one given instant is provided for limiting the consideration of the frequency components up to a cut-off frequency, this given instant being located temporally at the beginning of a different block of a first block in a succession of blocks. This given moment therefore intervenes after a direct propagation, and at the time of sound reflections or of presence of diffuse field.
- FIG. 5 also illustrating, in an exemplary embodiment, a possible algorithm of a computer program that would execute a processor of a spatialization module operating the method within the meaning of the invention.
- the present invention also aims, in general, a computer program comprising instructions for implementing the method above, when they are executed by a processor.
- the present invention also aims at a sound spatialization module, comprising calculation means for applying at least one room effect transfer function to at least one input sound signal, said application being used to multiply, in the spectral domain, spectral components of the sound signal by the spectral components of a filter corresponding to said transfer function, each spectral component of the filter comprising a temporal evolution in a time-frequency representation.
- these calculation means are configured to ignore said spectral components of the filter for said multiplications of components, beyond a threshold frequency and after at least a given time in said time-frequency representation.
- the sound spatialization module receiving a plurality of input signals, delivers at least two output signals, the calculation means being configured to apply a room effect transfer function, at each input signal, each of said output signals being given by applying a formula of the type:
- O k being an output signal
- k being the index relating to an output signal
- a k ⁇ 1) being a room effect transfer function specific to an input signal
- J3 ⁇ 4 SG5S (m) being a global transfer function, with room effect, common to the input signals
- z ⁇ ⁇ lDD TM being a delay application, counted in number of sample blocks, corresponding to a time difference between a sound emission in a room corresponding to the room effect, and a beginning of diffuse field presence in this room. room, the index m corresponding to a number of blocks of samples of duration corresponding to this delay, M being the total number of blocks that a transfer function lasts in a time-frequency representation,
- This module can be integrated into a decoding device in compression, or more generally in a rendering system.
- the latter comprises in the example shown an input interface IN for receiving the decoded signals, as well as calculation means such as a processor PROC and a working memory MEM cooperating with the IN / OUT interfaces to spatialize the signals. (/) and output from the output interface OUT only two signals O d and O 8 for feeding the respective ear cups of a CAS helmet.
- Figure 1 illustrates a general embodiment of the method according to the invention
- FIG. 2 illustrates an example of application of the method according to an embodiment where the transfer functions are in the form of a combination of two transfer functions, one of which is applied with a delay on the signal to be processed;
- FIG. 3 represents an example of a time-frequency representation of a transfer function with cutoff frequencies (or "threshold frequencies” mentioned above) that vary in particular as a function of time;
- FIG. 4 illustrates a flowchart corresponding to a possible general algorithm of the computer program within the meaning of the invention
- FIG. 5 represents a particular embodiment resulting from the mode represented in FIG.
- FIG. 6 illustrates an exemplary spatialization module within the meaning of the invention
- Figure 7 schematically illustrates the virtual loudspeakers and the room effect to apply an appropriate transfer function, with imitation of the components frequency of this transfer function to an appropriate cutoff frequency.
- a plurality of virtual speakers surround, in the example shown, the head TE of a listener.
- Each of the virtual speakers HPV is initially powered by a signal ⁇ 1) with 'G [1. ⁇ i] for example previously decoded as indicated above with reference to Figure 6.
- the arrangement of the virtual speakers can concern a multichannel or also a surround representation of the signals 1 (1) to be processed in order to render them spatially together with a room effect on a CAS headset (FIG. 6)
- the maximum frequency F c d (0) of a filter representing the transfer function specific to the right ear can be lower than the frequency maximum F c 8 (0) of a filter representing the transfer function specific to the left ear.
- a designer of such a filter can thus limit the components of his filter for the right ear up to the cut-off frequency F c d (0) (corresponding to a head-catching frequency) even though the signal to be processed I (/) may have higher spectral components and up to at the frequency F c 8 (0) at least.
- a filter designer representing these transfer functions can provide for limiting the filter components for the right ear to the cutoff frequency F c d (1) and for the left ear to the cutoff frequency F c 8 (l).
- F c d (1) the cutoff frequency
- F c 8 (l) the cutoff frequency
- a filter designer representing these transfer functions can provide for limiting the filter components for the right ear to the cutoff frequency F c d (2) and for the left ear to the cutoff frequency F C 8 (2).
- L input signals 1 (1), 1 (2), I (L) are transformed in the frequency domain, respectively at the steps TF11, TF12, TF1L.
- input signals may already be available in frequency form (for example at the decoder).
- step BA11 a complete impulse response of spatialization (typically type B I for "Binaural Room Impulse Response") in time form corresponding to the signal 1 (1) of the channel 1 is stored.
- this impulse response is converted into frequency form to obtain a corresponding filter in the spectral domain.
- the filter is stored in its spectral form to avoid repeating the calculation of the transform.
- This filter is then multiplied by the frequency-domain input signal of channel 1 (which is tantamount to convol ution in the time domain). We therefore have the signal spatialized for the signal 1 (1) of the channel 1.
- the same operations are performed for the other L-1 channels.
- We thus have a total of L spatialized channels. These channels are then summed to obtain a single output signal representing the L channels, and one goes back to the time domain at the step ITF11, to deliver one of the signals O k (with k d, g) feeding an atrium. .
- a similar treatment is performed for the other atrium.
- the L spatialized channels are not accessible independently before summation: the single output signal is constructed by summing up each channel spatially with the previous output signal. These operations are performed for each output signal O k to be constructed.
- the L input signals may typically correspond to the L channels of multichannel audio content intended to power speakers ("virtual").
- the L input signals can, for example, correspond to the L ambiophonic signals of an audio content in ambiophonic representation.
- FIG. 2 illustrating an implementation in the sense of the invention, the principle of a spatialization of L channels as shown in FIG. 1 is taken again. Nevertheless, the presentation of FIG. 2 is simplified in this respect.
- L input signals 1 (1), 1 (2), I (L) are transformed in the frequency domain in step S21.
- such input signals may alternatively already be available in frequency form.
- an impulse response A k (/) of spatialization typically of type BI
- This impulse response A k (/) is incomplete in the representation of FIG.
- this impulse response may already be available in frequency form.
- the components of this filter are then multiplied by the spectral signal of the corresponding channel 1 (/).
- This multiplication is parameterized (as indicated below with reference to FIG. 4) so that certain frequency components are ignored, within the meaning of the invention. Typically, the highest frequency components will be ignored to limit computational complexity.
- the multiplication of the components limited to a cut-off frequency is indicated by the sign:
- a cut-off frequency f cA ( i) from which the frequency components are ignored is defined (for example the maximum frequency represented in the channel signal l (/), or half of its sampling frequency).
- the summation is performed in a particular way, because it takes into account a delay on the channels to characterize the reverberations (reflections and diffuse field), as detailed below.
- the L spatialized channels are not accessible independently before summation: the single output signal is constructed by summing each channel spatially with the previous output signal.
- step S24 an incomplete impulse response B k m (/) of spatialization (typically of type BI) corresponding to the signal I (/) of the channel / is transformed in the spectral domain to obtain a frequency filter.
- this impulse response may alternatively already be available in frequency form.
- this filter B k m (/) is then multiplied by the signal I (/) of the channel /.
- the cutoff frequencies are different for this second time block. As shown with reference to FIG. 3, measurements show that the high frequencies are more attenuated in the distant time blocks (corresponding to diffuse sounds and to multiple reverberations). The cutoff frequencies for these remote blocks may therefore be lower than for the first blocks. However, the lower the cutoff frequency, the more the number of operations is limited. Thus, the complexity of the calculations is advantageously reduced.
- the same operations are performed for the L channels and the multiplication operations of the filter are repeated on the progressively delayed spectral signals by summing the contributions at the repetitive step S25 for each delay m until obtaining a single signal representing the L channels on the set M of the time blocks m considered.
- the single output signal is constructed by summing each spatial channel with the previous output signal as will now be seen with reference to FIG. 4.
- step S26 we go back to the time domain in step S26 to obtain an output signal for feeding one of the headsets of the helmet.
- step S40 the output signal S is initialized to 0.
- This output signal is expressed in the frequency domain. It has a limited size longer than the cutoff frequency fc (/). For example, this signal is set to [0; fs (/) / 2], fs (/) being the sampling frequency of this signal I (/).
- a first count variable / is also initialized to 1. This first count variable identifies one of the channel 1 (1), 1 (2), ..., 1 (/), ..., 1 (L) signals. on the time block [0; Nl] for the right ear.
- a second count variable j is initialized to 0. This second count variable identifies a frequency component of a signal I (/) on the time block [0; Nl] for the right ear.
- step S42 the coefficient c BR! R (j; /) is stored in memory. This coefficient corresponds to the frequency component j of the filter BI (/) on the time block [0; Nl] for the right ear. Similarly, the coefficient c, (j; /) is stored. This coefficient corresponds to the frequency component j of the signal I (/) on the time block [0; Nl] for the right ear. Thus, the coefficients c BR
- the frequency corresponding to the variable j is lower (for example strictly) at the cut-off frequency fc (/).
- This cutoff frequency corresponds to the cutoff frequency of the signal I (/) for the time block [0; N-1] for the right ear. If the frequency j is less than the cutoff frequency fc (/), go to step S44.
- step S44 a value MULT (j) corresponding to the multiplication of the coefficients is calculated; /) and q (j; /). These coefficients are well multiplied term by term because they correspond to the same frequency component j (for the same channel, on the same block and for the same ear).
- this value MULT (j) is incremented to the signal S at the position of the frequency j.
- a stepwise construction of a signal S which comprises (at the end of the loop of length fc (/)) all the frequency components up to the cut-off frequency fc (/) (for this signal l (/) on the block [0; N-1] and for a right ear).
- a buffer initially zero up to the cutoff frequency to successively build the signal S.
- each multiplication MULT (j) of coefficients is added step by step to the signal S under construction.
- step S46 the variable j is incremented and it is resumed at step S42. If the variable j is greater (for example or equal to) the cutoff frequency fc (/), the test T48 is passed. Thus, the signal S has been filled over the interval [0; fc (/)].
- this signal can be set to an interval greater than [0; fc (/)] (for example [0; fs (/) / 2]).
- this signal was initialized to 0 over its entire definition range. Therefore, it is null on the rest of the interval that has not been filled (for example [fc (/); fs (/) / 2]). The complexity is therefore improved here because steps of filling the signal S have not been performed, which reduces the number of calculations required.
- test T48 it is verified that the count variable / corresponding to the signal l (/) of the channel / is lower (for example strictly) to the number L of channels. If the variable / is less than or equal to L, the variable / in step S49 is incremented and the method is repeated in step S41. If the variable / is greater than L, the signal S corresponding to the spatialized signal for the time block [0; N-1] for the right ear is available in step S50.
- This signal S corresponding to the time block [0; N-1] is then summed to the other signals generated in a similar way for other time blocks [N; 2N-1], [2N; 3N-1], etc. (and for which an appropriate delay has been applied in accordance with the above DBD step of Fig. 2, for example).
- a filter is applied in the frequency domain corresponding to a transfer function common to all the input signals I (/), representing the diffuse field, with a cut-off frequency fc in the multiplication of the spectral components which is the minimum between:
- the maximum frequency f max represented in each input signal (for example its sampling frequency or the maximum frequency whose spectral component is not zero, this value being usually given by a decoder in compression).
- this low-pass filter is not applied to the audio signal, but to the BRIR filter (which is itself convoluted to the audio signal) which is already composed of multiple reflections; the artifacts produced will therefore, at worst, be perceived as additional reflections of the original BRIR filter, and in practice rarely perceptible.
- FIG. 5 illustrates a complete algorithmic form of the processing, in accordance with the formula giving an output node O k as shown above:
- the weights W k (/) and the gains G (l (/)) can be set to 1.
- the gains G (l (/)) are not shown in FIG. to read this figure as an integration of the gains at the weights 1 / W k (/).
- these two parameters are determined, fixed and multiplied one by one once and for all.
Abstract
Description
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EP22211949.7A EP4184505B1 (fr) | 2013-10-18 | 2014-10-14 | Spatialisation sonore avec effet de salle, optimisee en complexite |
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FR1360185A FR3012247A1 (fr) | 2013-10-18 | 2013-10-18 | Spatialisation sonore avec effet de salle, optimisee en complexite |
PCT/FR2014/052617 WO2015055946A1 (fr) | 2013-10-18 | 2014-10-14 | Spatialisation sonore avec effet de salle, optimisee en complexite |
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EP22211949.7A Division EP4184505B1 (fr) | 2013-10-18 | 2014-10-14 | Spatialisation sonore avec effet de salle, optimisee en complexite |
EP22211949.7A Division-Into EP4184505B1 (fr) | 2013-10-18 | 2014-10-14 | Spatialisation sonore avec effet de salle, optimisee en complexite |
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US (1) | US9641953B2 (fr) |
EP (2) | EP3058564B1 (fr) |
JP (1) | JP6518661B2 (fr) |
KR (1) | KR102156650B1 (fr) |
CN (1) | CN105706162B (fr) |
ES (1) | ES2959534T3 (fr) |
FR (1) | FR3012247A1 (fr) |
WO (1) | WO2015055946A1 (fr) |
Families Citing this family (2)
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GB201609089D0 (en) * | 2016-05-24 | 2016-07-06 | Smyth Stephen M F | Improving the sound quality of virtualisation |
CN110428802B (zh) * | 2019-08-09 | 2023-08-08 | 广州酷狗计算机科技有限公司 | 声音混响方法、装置、计算机设备及计算机存储介质 |
Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US7835535B1 (en) * | 2005-02-28 | 2010-11-16 | Texas Instruments Incorporated | Virtualizer with cross-talk cancellation and reverb |
EP2840811A1 (fr) * | 2013-07-22 | 2015-02-25 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Procédé de traitement d'un signal audio, unité de traitement de signal, rendu binaural, codeur et décodeur audio |
Family Cites Families (8)
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FR1357299A (fr) | 1962-05-16 | 1964-04-03 | Ampoule pour phares de véhicules automobiles | |
US5917917A (en) * | 1996-09-13 | 1999-06-29 | Crystal Semiconductor Corporation | Reduced-memory reverberation simulator in a sound synthesizer |
JP4627880B2 (ja) * | 1997-09-16 | 2011-02-09 | ドルビー ラボラトリーズ ライセンシング コーポレイション | リスナーの周囲にある音源の空間的ひろがり感を増強するためのステレオヘッドホンデバイス内でのフィルタ効果の利用 |
EP1072089B1 (fr) * | 1998-03-25 | 2011-03-09 | Dolby Laboratories Licensing Corp. | Procede et appareil de traitement de signaux audio |
US20080085008A1 (en) * | 2006-10-04 | 2008-04-10 | Earl Corban Vickers | Frequency Domain Reverberation Method and Device |
TWI475896B (zh) * | 2008-09-25 | 2015-03-01 | Dolby Lab Licensing Corp | 單音相容性及揚聲器相容性之立體聲濾波器 |
US8976972B2 (en) * | 2009-10-12 | 2015-03-10 | Orange | Processing of sound data encoded in a sub-band domain |
EP2503800B1 (fr) * | 2011-03-24 | 2018-09-19 | Harman Becker Automotive Systems GmbH | Système d'ambiophonie constant spatialement |
-
2013
- 2013-10-18 FR FR1360185A patent/FR3012247A1/fr not_active Withdrawn
-
2014
- 2014-10-14 CN CN201480060448.0A patent/CN105706162B/zh active Active
- 2014-10-14 KR KR1020167012795A patent/KR102156650B1/ko active IP Right Grant
- 2014-10-14 EP EP14796814.3A patent/EP3058564B1/fr active Active
- 2014-10-14 ES ES14796814T patent/ES2959534T3/es active Active
- 2014-10-14 US US15/029,458 patent/US9641953B2/en active Active
- 2014-10-14 JP JP2016523910A patent/JP6518661B2/ja active Active
- 2014-10-14 EP EP22211949.7A patent/EP4184505B1/fr active Active
- 2014-10-14 WO PCT/FR2014/052617 patent/WO2015055946A1/fr active Application Filing
Patent Citations (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US7835535B1 (en) * | 2005-02-28 | 2010-11-16 | Texas Instruments Incorporated | Virtualizer with cross-talk cancellation and reverb |
EP2840811A1 (fr) * | 2013-07-22 | 2015-02-25 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Procédé de traitement d'un signal audio, unité de traitement de signal, rendu binaural, codeur et décodeur audio |
Non-Patent Citations (1)
Title |
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See also references of WO2015055946A1 * |
Also Published As
Publication number | Publication date |
---|---|
CN105706162B (zh) | 2019-06-11 |
WO2015055946A1 (fr) | 2015-04-23 |
EP4184505A1 (fr) | 2023-05-24 |
JP6518661B2 (ja) | 2019-05-22 |
KR20160073394A (ko) | 2016-06-24 |
US9641953B2 (en) | 2017-05-02 |
EP4184505B1 (fr) | 2024-02-28 |
US20160269850A1 (en) | 2016-09-15 |
FR3012247A1 (fr) | 2015-04-24 |
EP3058564B1 (fr) | 2023-07-26 |
ES2959534T3 (es) | 2024-02-26 |
CN105706162A (zh) | 2016-06-22 |
JP2016537866A (ja) | 2016-12-01 |
KR102156650B1 (ko) | 2020-09-16 |
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