US20070003069A1 - Perceptual synthesis of auditory scenes - Google Patents

Perceptual synthesis of auditory scenes Download PDF

Info

Publication number
US20070003069A1
US20070003069A1 US11/470,314 US47031406A US2007003069A1 US 20070003069 A1 US20070003069 A1 US 20070003069A1 US 47031406 A US47031406 A US 47031406A US 2007003069 A1 US2007003069 A1 US 2007003069A1
Authority
US
United States
Prior art keywords
signal
audio
auditory scene
different
signals
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.)
Abandoned
Application number
US11/470,314
Inventor
Christof Faller
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Agere Systems LLC
Original Assignee
Agere Systems LLC
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Agere Systems LLC filed Critical Agere Systems LLC
Priority to US11/470,314 priority Critical patent/US20070003069A1/en
Assigned to AGERE SYSTEMS INC. reassignment AGERE SYSTEMS INC. ASSIGNMENT OF ASSIGNORS INTEREST (SEE DOCUMENT FOR DETAILS). Assignors: FALLER, CHRISTOF
Publication of US20070003069A1 publication Critical patent/US20070003069A1/en
Abandoned legal-status Critical Current

Links

Images

Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/56Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M3/00Automatic or semi-automatic exchanges
    • H04M3/42Systems providing special services or facilities to subscribers
    • H04M3/56Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
    • H04M3/568Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities audio processing specific to telephonic conferencing, e.g. spatial distribution, mixing of participants
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S3/00Systems employing more than two channels, e.g. quadraphonic
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04SSTEREOPHONIC SYSTEMS 
    • H04S2420/00Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/03Application of parametric coding in stereophonic audio systems

Definitions

  • the present invention relates to the synthesis of auditory scenes, that is, the generation of audio signals to produce the perception that the audio signals are generated by one or more different audio sources located at different positions relative to the listener.
  • an audio signal i.e., sounds
  • the audio signal will typically arrive at the person's left and right ears at two different times and with two different audio (e.g., decibel) levels, where those different times and levels are functions of the differences in the paths through which the audio signal travels to reach the left and right ears, respectively.
  • the person's brain interprets these differences in time and level to give the person the perception that the received audio signal is being generated by an audio source located at a particular position (e.g., direction and distance) relative to the person.
  • An auditory scene is the net effect of a person simultaneously hearing audio signals generated by one or more different audio sources located at one or more different positions relative to the person.
  • This processing by the brain can be used to synthesize auditory scenes, where audio signals from one or more different audio sources are purposefully modified to generate left and right audio signals that give the perception that the different audio sources are located at different positions relative to the listener.
  • FIG. 1 shows a high-level block diagram of conventional binaural signal synthesizer 100 , which converts a single audio source signal (e.g., a mono signal) into the left and right audio signals of a binaural signal, where a binaural signal is defined to be the two signals received at the eardrums of a listener.
  • synthesizer 100 receives a set of spatial parameters corresponding to the desired position of the audio source relative to the listener.
  • the set of spatial parameters comprises an interaural level difference (ILD) value (which identifies the difference in audio level between the left and right audio signals as received at the left and right ears, respectively) and an interaural time delay (ITD) value (which identifies the difference in time of arrival between the left and right audio signals as received at the left and right ears, respectively).
  • ILD interaural level difference
  • ITD interaural time delay
  • some synthesis techniques involve the modeling of a direction-dependent transfer function for sound from the signal source to the eardrums, also referred to as the head-related transfer function (HRTF). See, e.g., J. Blauert, The Psychophysics of Human Sound Localization, MIT Press, 1983, the teachings of which are incorporated herein by reference.
  • the mono audio signal generated by a single sound source can be processed such that, when listened to over headphones, the sound source is spatially placed by applying an appropriate set of spatial parameters (e.g., ILD, ITD, and/or HRTF) to generate the audio signal for each ear.
  • an appropriate set of spatial parameters e.g., ILD, ITD, and/or HRTF
  • Binaural signal synthesizer 100 of FIG. 1 generates the simplest type of auditory scenes: those having a single audio source positioned relative to the listener. More complex auditory scenes comprising two or more audio sources located at different positions relative to the listener can be generated using an auditory scene synthesizer that is essentially implemented using multiple instances of binaural signal synthesizer, where each binaural signal synthesizer instance generates the binaural signal corresponding to a different audio source. Since each different audio source has a different location relative to the listener, a different set of spatial parameters used to generate the binaural audio signal for each different audio source.
  • FIG. 2 shows a high-level block diagram of conventional auditory scene synthesizer 200 , which converts a plurality of audio source signals (e.g., a plurality of mono signals) into the left and right audio signals of a single combined binaural signal, using a different set of spatial parameters for each different audio source.
  • the left audio signals are then combined (e.g., by simple addition) to generate the left audio signal for the resulting auditory scene, and similarly for the right.
  • conferencing Assume, for example, a desktop conference with multiple participants, each of whom is sitting in front of his or her own personal computer (PC) in a different city.
  • PC personal computer
  • each participant's PC is equipped with (1) a microphone that generates a mono audio source signal corresponding to that participant's contribution to the audio portion of the conference and (2) a set of headphones for playing that audio portion.
  • Displayed on each participant's PC monitor is the image of a conference table as viewed from the perspective of a person sitting at one end of the table. Displayed at different locations around the table are real-time video images of the other conference participants.
  • a server In a conventional mono conferencing system, a server combines the mono signals from all of the participants into a single combined mono signal that is transmitted back to each participant.
  • the server can implement an auditory scene synthesizer, such as synthesizer 200 of FIG. 2 , that applies an appropriate set of spatial parameters to the mono audio signal from each different participant and then combines the different left and right audio signals to generate left and right audio signals of a single combined binaural signal for the auditory scene. The left and right audio signals for this combined binaural signal are then transmitted to each participant.
  • an auditory scene synthesizer such as synthesizer 200 of FIG. 2
  • the present invention is directed to a technique for synthesizing auditory scenes that addresses the transmission bandwidth problem of the prior art.
  • an auditory scene corresponding to multiple audio sources located at different positions relative to the listener is synthesized from a single combined (e.g., mono) audio signal.
  • a solution can be implemented in which each participant's PC receives only a single mono audio signal corresponding to a combination of the mono audio source signals from all of the participants.
  • the present invention is based on an assumption that, for those frequency bands in which the energy of the source signal from a particular audio source dominates the energies of all other source signals in the combined audio signal, from the perspective of the perception by the listener, the combined audio signal can be treated as if it corresponded solely to that particular audio source.
  • different sets of spatial parameters are applied to different frequency bands in the combined audio signal where different audio sources dominate, to synthesize an auditory scene.
  • the present invention is a method for synthesizing an auditory scene, comprising the steps of (a) dividing an input audio signal into a plurality of different frequency bands; and (b) applying two or more different sets of one or more spatial parameters to two or more of the different frequency bands in the input audio signal to generate two or more synthesized audio signals of the auditory scene, wherein for each of the two or more different frequency bands, the corresponding set of one or more spatial parameters is applied to the input audio signal as if the input audio signal corresponded to a single audio source in the auditory scene.
  • the present invention is an apparatus for synthesizing an auditory scene, comprising (1) an auditory scene synthesizer configured to (a) divide an input audio signal into a plurality of different frequency bands; and (b) apply two or more different sets of one or more spatial parameters to two or more of the different frequency bands in the input audio signal to generate two or more synthesized audio signals of the auditory scene, wherein for each of the two or more different frequency bands, the corresponding set of one or more spatial parameters is applied to the input audio signal as if the input audio signal corresponded to a single audio source in the auditory scene; and (2) one or more inverse time-frequency transformers configured to convert the two or more synthesized audio signals from a frequency domain into a time domain.
  • an auditory scene synthesizer configured to (a) divide an input audio signal into a plurality of different frequency bands; and (b) apply two or more different sets of one or more spatial parameters to two or more of the different frequency bands in the input audio signal to generate two or more synthesized audio signals of the auditory
  • the present invention is a method for processing two or more input audio signals, comprising the steps of (a) converting the two or more input audio signals from a time domain into a frequency domain; (b) generating a set of one or more auditory scene parameters for each of two or more different frequency bands in the two or more converted input audio signals, where each set of one or more auditory scene parameters is generated as if the corresponding frequency band corresponded to a single audio source in an auditory scene; and (c) combining the two or more input audio signals to generate a combined audio signal.
  • the present invention is an apparatus for processing two or more input audio signals, comprising (a) a time-frequency transformer configured to convert the two or more input audio signals from a time domain into a frequency domain; (b) an auditory scene parameter generator configure to generate a set of one or more auditory scene parameters for each of two or more different frequency bands in the two or more converted input audio signals, where each set of one or more auditory scene parameters is generated as if the corresponding frequency band corresponded to a single audio source in an auditory scene; and (c) a combiner configured to combine the two or more input audio signals to generate a combined audio signal.
  • FIG. 1 shows a high-level block diagram of conventional binaural signal synthesizer that converts a single audio source signal (e.g., a mono signal) into the left and right audio signals of a binaural signal;
  • a single audio source signal e.g., a mono signal
  • FIG. 2 shows a high-level block diagram of conventional auditory scene synthesizer that converts a plurality of audio source signals (e.g., a plurality of mono signals) into the left and right audio signals of a single combined binaural signal;
  • a plurality of audio source signals e.g., a plurality of mono signals
  • FIG. 3 shows a block diagram of a conferencing system, according to one embodiment of the present invention
  • FIG. 4 shows a block diagram of the audio processing implemented by the conference server of FIG. 3 , according to one embodiment of the present invention
  • FIG. 5 shows a flow diagram of the processing implemented by the auditory scene parameter generator of FIG. 4 , according to one embodiment of the present invention
  • FIG. 6 shows a graphical representation of the power spectra of the audio signals from three different exemplary sources
  • FIG. 7 shows a block diagram of the audio processing performed by each conference node in FIG. 3 ;
  • FIG. 8 shows a graphical representation of the power spectrum in the frequency domain for the combined signal generated from the three mono source signals in FIG. 6 ;
  • FIG. 9 shows a representation of the analysis window for the time-frequency domain, according to one embodiment of the present invention.
  • FIG. 10 shows a block diagram of the transmitter for an alternative application of the present invention, according to one embodiment of the present invention.
  • FIG. 3 shows a block diagram of a conferencing system 300 , according to one embodiment of the present invention.
  • Conferencing system 300 comprises conference server 302 , which supports conferencing between a plurality of conference participants, where each participant uses a different conference node 304 .
  • each node 304 is a personal computer (PC) equipped with a microphone 306 and headphones 308 , although other hardware configurations are also possible. Since the present invention is directed to processing of the audio portion of conferences, the following description omits reference to the processing of the video portion of such conferences, which involves the generation, manipulation, and display of video signals by video cameras, video signal processors, and digital monitors that would be included in conferencing system 300 , but are not explicitly represented in FIG. 3 .
  • the present invention can also be implemented for audio-only conferencing.
  • each node 304 transmits a (e.g., mono) audio source signal generated by its microphone 306 to server 302 , where that source signal corresponds to the corresponding participant's contribution to the conference.
  • Server 302 combines the source signals from the different participants into a single (e.g., mono) combined audio signal and transmits that combined signal back to each node 304 .
  • the combined signal transmitted to each node 304 may be either unique to that node or the same as the combined signal transmitted to every other node.
  • server 302 transmits an appropriate set of auditory scene parameters to each node 304 .
  • Each node 304 applies the set of auditory scene parameters to the combined signal in a manner according to the present invention to generate a binaural signal for rendering by headphones 308 and corresponding to the auditory scene for the conference.
  • conference server 302 may be implemented within a distinct node of conferencing system 300 .
  • server processing may be implemented in one of the conference nodes 304 , or even distributed among two or more different conference nodes 304 .
  • FIG. 4 shows a block diagram of the audio processing implemented by conference server 302 of FIG. 3 , according to one embodiment of the present invention.
  • auditory scene parameter generator 402 generates one or more sets of auditory scene parameters from the plurality of source signals generated by and received from the various conference nodes 304 of FIG. 3 .
  • signal combiner 404 combines the plurality of source signals (e.g., using straightforward audio signal addition) to generate the combined signal that is transmitted back to each conference node 304 .
  • FIG. 5 shows a flow diagram of the processing implemented by auditory scene parameter generator 402 of FIG. 4 , according to one embodiment of the present invention.
  • Generator 402 applies a time-frequency (TF) transform, such as a discrete Fourier transform (DFT), to convert each node's source signal to the frequency domain (step 502 of FIG. 5 ).
  • TF time-frequency
  • DFT discrete Fourier transform
  • Generator 402 compares the power spectra of the different converted source signals to identify one or more frequency bands in which the energy one of the source signals dominates all of the other signals (step 504 ).
  • a particular source signal may be said to dominate all of the other source signals when the energy of that source signal exceeds the sum of the energies in the other source signals by either a specified factor or a specified amount of power (e.g., in dBs).
  • a particular source signal may be said to dominate when the energy of that source signal exceeds the second most powerful source signal by a specified factor or a specified amount of power.
  • Other criteria are, of course, also possible, including those that combine two or more different comparisons. For example, in addition to relative domination, a source signal might have to have an absolute energy level that exceeds a specified energy level before qualifying as a dominating source signal.
  • FIG. 6 shows a graphical representation of the power spectra of the audio signals from three different exemplary sources (labeled A, B, and C).
  • FIG. 6 identifies eight different frequency bands in which one of the three source signals dominates the other two. Note that, in FIG. 6 , there are particular frequency ranges in which none of the three source signals dominate. Note also that the lengths of the dominated frequency ranges (i.e., frequency ranges in which one of the source signals dominates) are not uniform, but rather are dictated by the characteristics of the power spectra themselves.
  • a set of auditory scene parameters is generated for each frequency band, where those parameters correspond to the node whose source signal dominates that frequency band (step 506 ).
  • the processing of step 506 implemented by generator 402 generates the actual spatial parameters (e.g., ILD, ITD, and/or HRTF) for each dominated frequency band.
  • generator 402 receives (e.g., a priori) information about the relative spatial placement of each participant in the auditory scene to be synthesized (as indicated in FIG. 4 ).
  • at least the following auditory scene parameters are transmitted to each conference node 304 of FIG. 3 for each dominated frequency band:
  • the generation of the spatial parameters for each dominated frequency band is implemented independently at each conference node 304 .
  • generator 402 does not need any information about the relative spatial placements of the various participants in the synthesized auditory scene. Rather, in addition to the combined signal, only the following auditory scene parameters need to be transmitted to each conference node 304 for each dominated frequency band:
  • the processing of FIG. 5 is preferably repeated at a specified interval (e.g., once for every 20-msec frame of audio data).
  • a specified interval e.g., once for every 20-msec frame of audio data.
  • the number and definition of the dominated frequency ranges as well as the particular source signals that dominate those ranges will typically vary over time (e.g., from frame to frame), reflecting the fact that the set of conference participants who are speaking at any given time will vary over time as will the characteristics of their own individual voices (e.g., intonations and/or volumes).
  • the spatial parameters corresponding to each conference participant may be either static (e.g., for synthesis of stationary participants whose relative positions do not change over time) or dynamic (e.g., for synthesis of mobile participants who relative positions are allowed to change over time).
  • a set of spatial parameters can be generated that reflects the contributions of two or more—or even all—of the participants. For example, weighted averaging can be used to generate an ILD value that represents the relative contributions for the two or more most dominant participants. In such cases, each set of spatial parameters is a function of the relative dominance of the most dominant participants for a particular frequency band.
  • FIG. 7 shows a block diagram of the audio processing performed by each conference node 304 in FIG. 3 to convert a single combined mono audio signal and corresponding auditory scene parameters received from conference server 302 into the binaural signal for a synthesized auditory scene.
  • time-frequency (TF) transform 702 converts each frame of the combined signal into the frequency domain.
  • auditory scene synthesizer 704 For each dominated frequency band, auditory scene synthesizer 704 applies the corresponding auditory scene parameters to the converted combined signal to generate left and right audio signals for that frequency band in the frequency domain. In particular, for each audio frame and for each dominated frequency band, synthesizer 704 applies the set of spatial parameters corresponding to the participant whose source signal dominates the combined signal for that dominated frequency range. If the auditory scene parameters received from the conference server do not include the spatial parameters for each conference participant, then synthesizer 704 receives information about the relative spatial placement of the different participants in the synthesized auditory scene as indicated in FIG. 7 , so that the set of spatial parameters for each dominated frequency band in the combined signal can be generated locally at the conference node.
  • An inverse TF transform 706 is then applied to each of the left and right audio signals to generate the left and right audio signals of the binaural signal in the time domain corresponding to the synthesized auditory scene.
  • the resulting auditory scene is perceived as being approximately the same as for an ideally synthesized binaural signal with the same corresponding spatial parameters but applied over the whole spectrum of each individual source signal.
  • FIG. 8 shows a graphical representation of the power spectrum in the frequency domain for the combined signal generated from the three mono source signals from sources A, B, and C in FIG. 6 .
  • FIG. 8 also shows the same frequency bands identified in FIG. 6 in which the power of one of the three source signals dominates the other two. It is to these dominated frequency bands to which auditory scene synthesizer 704 applies appropriate sets of spatial parameters.
  • not all of the conference participants will dominate at least one frequency band, since not all of the participants will typically be talking at the same time. If only one participant is talking, then only that participant will typically dominate any of the frequency bands. By the same token, during an audio frame corresponding to relative silence, it may be that none of the participants will dominate any frequency bands. For those frequency bands for which no dominating participant is identified, no spatial parameters are applied and the left and right audio signals of the resulting binaural signal for those frequency bands are identical.
  • TF transform 702 in FIG. 7 converts the combined mono audio signal to the spectral (i.e., frequency) domain frame-wise in order for the system to operate for real-time applications.
  • a level difference ⁇ L n [k] a time difference ⁇ n [k] and/or an HRTF is to be introduced into the underlying audio signal.
  • TF transform 702 is a DFT-based transform, such as those described in A. V. Oppenheim and R. W. Schaefer, Discrete - Time Signal Processing, Signal Processing Series, Prentice Hall, 1989, the teachings of which are incorporated herein by reference.
  • the transform is derived based on the desire for the ability to synthesize frequency-dependent and time-adaptive time differences ⁇ n [k].
  • the same transform can be used advantageously for the synthesis of frequency-dependent and time-adaptive level differences ⁇ L n [k] and for HRTFs.
  • the described scheme works as long as the time-shift d does not vary in time. Since the desired d usually varies over time, the transitions are smoothed by using overlapping windows for the analysis transform. A frame of N samples is multiplied with the analysis window before an N-point DFT is applied.
  • FIG. 9 shows a representation of the analysis window, which was chosen such that it is additive to one when windows of adjacent frames are overlapped by W/2 samples.
  • the time-span of the window shown in FIG. 9 is shorter than the DFT length such that non-circular time-shifts within the range [ ⁇ Z,Z] are possible.
  • a higher factor of oversampling can be used by choosing the time-span of the window to be smaller and/or by overlapping the windows more.
  • the zero padding of the analysis window shown in FIG. 9 allows the implementation of convolutions with HRTFs as simple multiplications in the frequency domain. Therefore, the transform is also suitable for the synthesis of HRTFs in addition to time and level differences.
  • a more general and slightly different point of view of a similar transform is given by J. B. Allen, “Short-term spectral analysis, synthesis and modification by discrete fourier transform, ” IEEE Trans. on Speech and Signal Processing, vol. ASSP-25, pp. 235-238, June 1977, the teachings of which are incorporated herein by reference.
  • auditory scene synthesizer 704 of FIG. 7 applies different sets of specified level and time differences to the different dominated frequency bands in the combined signal to generate the left and right audio signals of the binaural signal for the synthesized auditory scene.
  • each dominated frequency band n is associated with a level difference ⁇ L n [k] and a time difference ⁇ n [k].
  • these level and time differences are applied symmetrically to the spectrum of the combined signal to generate the spectra of the left and right audio signals according to Equations (4) and (5), respectively, as follows:
  • S n L 10 ⁇ ⁇ ⁇ L n 10 1 + 10 ⁇ ⁇ ⁇ L n 10 ⁇ S n ⁇ e - 2 ⁇ ⁇ ⁇ ⁇ n ⁇ ⁇ ⁇ n 2 ⁇ N ⁇ ⁇ and ( 4 )
  • S n R 1 1 + 10 2 ⁇ ⁇ ⁇ ⁇ L n 10 ⁇ S n ⁇ e - 2 ⁇ ⁇ ⁇ n ⁇ ⁇ n 2 ⁇ N ( 5 )
  • ⁇ S n ⁇ are the spectral coefficients of the combined signal
  • ⁇ S n L ⁇ and ⁇ S n R ⁇ are the spectral coefficients of the resulting binaural signal.
  • the level differences ⁇ L n ⁇ are expressed in dB and the time differences
  • a weighted sum of the frequency responses of the HRTFs of all sources is applied with weights w m,n .
  • the level differences ⁇ L n , time differences ⁇ n , and HRTF weights w m,n are preferably smoothed in frequency and time to prevent artifacts.
  • a typical sentence of the corpus is “READY LAKER, GO TO BLUE FIVE NOW,” where LAKER is the call sign and BLUE FIVE is a color-number combination. Combinations of the eight different call signs, four different colors, and eight different numbers were chosen randomly with the restriction that the call sign assigned to the participant occurred in 50% of the cases.
  • each participant was instructed to respond when his or her call sign was called by indicating the color-number combination by the talker who called the call sign.
  • One out of four female talkers was randomly chosen for each of the two talkers in each test item.
  • ILD ⁇ 16 dB
  • TTD ⁇ 500 ⁇ sec
  • Table I shows the results for the case when the listeners were called by their call signs.
  • the upper row shows the percentage of correct identification of the call sign, and the lower row shows the conditional percentage of the correct color-number combination given that the listener's call sign was correctly identified.
  • Test 1 Test 2
  • Test 3 Test 4
  • Test 5 call sign 70% 78% 85% 77% 78% color-number 64% 98% 88% 96% 91% Alternative Embodiments
  • the present invention was described in the context of a desktop conferencing application.
  • the present invention can also be employed for other applications.
  • the present invention can be applied where the input is a binaural signal corresponding to an (actual or synthesized) auditory scene, rather than the input being individual mono source signals as in the previous application.
  • the binaural signal is converted into a single mono signal and auditory scene parameters (e.g., sets of spatial parameters).
  • this application of the present invention can be used to reduce the transmission bandwidth requirements for the auditory scene since, instead of having to transmit the individual left and right audio signals for the binaural signal, only a single mono signal plus the relatively small amount of spatial parameter information need to be transmitted to a receiver, where the receiver performs processing similar to that shown in FIG. 7 .
  • FIG. 10 shows a block diagram of transmitter 1000 for such an application, according to one embodiment of the present invention.
  • a TF transform 1002 is applied to corresponding frames of each of the left and right audio signals of the input binaural signal to convert the signals to the frequency domain.
  • Auditory scene analyzer 1004 processes the converted left and right audio signals in the frequency domain to generate a set of auditory scene parameters for each of a plurality of different frequency bands in those converted signals.
  • analyzer 1004 divides the converted left and right audio signals into a plurality of frequency bands.
  • each of the left and right audio signals can be divided into the same number of equally sized frequency bands.
  • the size of the frequency bands may vary with frequency, e.g., larger frequency bands for higher frequencies or smaller frequency bands for higher frequencies.
  • analyzer 1004 compares the converted left and right audio signals to generate one or more spatial parameters (e.g., an ILD value, an ITD value, and/or an HRTF).
  • the cross-correlation between the converted left and right audio signals is estimated.
  • the maximum value of the cross-correlation which indicates how much the two signals are correlated, can be used as a measure for the dominance of one source in the band. If there is 100% correlation between the left and right audio signals, then only one source's energy is dominant in that frequency band. The less the cross-correlation maximum is, the less is just one source dominant.
  • the location in time of the maximum of the cross-correlation can be used to correspond to the ITD.
  • the ILD can be obtained by computing the level difference of the power spectral values of the left and right audio signals.
  • each set of spatial parameters is generated by treating the corresponding frequency range as if it were dominated by a single source signal.
  • the generated set of spatial parameters will be fairly accurate.
  • the generated set of spatial parameters will have less physical significance to the actual auditory scene.
  • the assumption is that those frequency bands contribute less significantly to the overall perception of the auditory scene. As such, the application of such “less significant” spatial parameters will have little if any adverse affect on the resulting auditory scene.
  • transmitter 1000 transmits these auditory scene parameters to the receiver for use in reconstructing the auditory scene from the mono audio signal.
  • Auditory scene remover 1006 combines the converted left and right audio signals in the frequency domain to generate the mono audio signal.
  • remover 1006 simply averages the left and right audio signals.
  • more sophisticated processing is performed to generate the mono signal.
  • the spatial parameters generated by auditory scene analyzer 1004 can be used to modify both the left and right audio signals in the frequency domain as part of the process of generating the mono signal, where each different set of spatial parameters is used to modify a corresponding frequency band in each of the left and right audio signals.
  • the left and right audio signals in each frequency band can be appropriately time shifted using the corresponding ITD value to make the ITD between the left and right audio signals become zero.
  • the power spectra for the time-shifted left and right audio signals can then be added such that the perceived loudness of each frequency band is the same in the resulting mono signal as in the original binaural signal.
  • An inverse TF transform 1008 is then applied to the resulting mono audio signal in the frequency domain to generate the mono audio signal in the time domain.
  • the mono audio signal can then be compressed and/or otherwise processed for transmission to the receiver. Since a receiver having a configuration similar to that in FIG. 7 converts the mono audio signal back into the frequency domain, the possibility exists for omitting inverse TF transform 1008 of FIG. 10 and TF transform 702 of FIG. 7 , where the transmitter transmits the mono audio signal to the receiver in the frequency domain.
  • the receiver applies the received auditory scene parameters to the received mono audio signal to synthesize (or, in this latter case, reconstruct an approximation of) the auditory scene.
  • the frequency bands are selected in an open-loop manner, but processed with the same underlying assumption as the previous application: that is, that each frequency band can be treated as if it corresponded to a single source using a corresponding set of spatial parameters.
  • the present invention may be implemented as circuit-based processes, including possible implementation on a single integrated circuit.
  • various functions of circuit elements may also be implemented as processing steps in a software program.
  • Such software may be employed in, for example, a digital signal processor, micro-controller, or general-purpose computer.
  • the present invention can be embodied in the form of methods and apparatuses for practicing those methods.
  • the present invention can also be embodied in the form of program code embodied in tangible media, such as floppy diskettes, CD-ROMs, hard drives, or any other machine-readable storage medium, wherein, when the program code is loaded into and executed by a machine, such as a computer, the machine becomes an apparatus for practicing the invention.
  • the present invention can also be embodied in the form of program code, for example, whether stored in a storage medium, loaded into and/or executed by a machine, or transmitted over some transmission medium or carrier, such as over electrical wiring or cabling, through fiber optics, or via electromagnetic radiation, wherein, when the program code is loaded into and executed by a machine, such as a computer, the machine becomes an apparatus for practicing the invention.
  • program code When implemented on a general-purpose processor, the program code segments combine with the processor to provide a unique device that operates analogously to specific logic circuits.

Abstract

An auditory scene is synthesized by applying two or more different sets of one or more spatial parameters (e.g., an inter-ear level difference (ILD), inter-ear time difference (ITD), and/or head-related transfer function (HRTF)) to two or more different frequency bands of a combined audio signal, where each different frequency band is treated as if it corresponded to a single audio source in the auditory scene. In one embodiment, the combined audio signal corresponds to the combination of two or more different source signals, where each different frequency band corresponds to a region of the combined audio signal in which one of the source signals dominates the others. In this embodiment, the different sets of spatial parameters are applied to synthesize an auditory scene comprising the different source signals.

Description

    CROSS-REFERENCE TO RELATED APPLICATIONS
  • This is a continuation of co-pending application Ser. No. 09/848,877, filed on May 4, 2001 as attorney docket no. Faller 5, the teachings of which are incorporated herein by reference.
  • BACKGROUND OF THE INVENTION
  • 1. Field of the Invention
  • The present invention relates to the synthesis of auditory scenes, that is, the generation of audio signals to produce the perception that the audio signals are generated by one or more different audio sources located at different positions relative to the listener.
  • 2. Description of the Related Art
  • When a person hears an audio signal (i.e., sounds) generated by a particular audio source, the audio signal will typically arrive at the person's left and right ears at two different times and with two different audio (e.g., decibel) levels, where those different times and levels are functions of the differences in the paths through which the audio signal travels to reach the left and right ears, respectively. The person's brain interprets these differences in time and level to give the person the perception that the received audio signal is being generated by an audio source located at a particular position (e.g., direction and distance) relative to the person. An auditory scene is the net effect of a person simultaneously hearing audio signals generated by one or more different audio sources located at one or more different positions relative to the person.
  • The existence of this processing by the brain can be used to synthesize auditory scenes, where audio signals from one or more different audio sources are purposefully modified to generate left and right audio signals that give the perception that the different audio sources are located at different positions relative to the listener.
  • FIG. 1 shows a high-level block diagram of conventional binaural signal synthesizer 100, which converts a single audio source signal (e.g., a mono signal) into the left and right audio signals of a binaural signal, where a binaural signal is defined to be the two signals received at the eardrums of a listener. In addition to the audio source signal, synthesizer 100 receives a set of spatial parameters corresponding to the desired position of the audio source relative to the listener. In typical implementations, the set of spatial parameters comprises an interaural level difference (ILD) value (which identifies the difference in audio level between the left and right audio signals as received at the left and right ears, respectively) and an interaural time delay (ITD) value (which identifies the difference in time of arrival between the left and right audio signals as received at the left and right ears, respectively). In addition or as an alternative, some synthesis techniques involve the modeling of a direction-dependent transfer function for sound from the signal source to the eardrums, also referred to as the head-related transfer function (HRTF). See, e.g., J. Blauert, The Psychophysics of Human Sound Localization, MIT Press, 1983, the teachings of which are incorporated herein by reference.
  • Using binaural signal synthesizer 100 of FIG. 1, the mono audio signal generated by a single sound source can be processed such that, when listened to over headphones, the sound source is spatially placed by applying an appropriate set of spatial parameters (e.g., ILD, ITD, and/or HRTF) to generate the audio signal for each ear. See, e.g., D. R. Begault, 3-D Sound for Virtual Reality and Multimedia, Academic Press, Cambridge, Mass., 1994.
  • Binaural signal synthesizer 100 of FIG. 1 generates the simplest type of auditory scenes: those having a single audio source positioned relative to the listener. More complex auditory scenes comprising two or more audio sources located at different positions relative to the listener can be generated using an auditory scene synthesizer that is essentially implemented using multiple instances of binaural signal synthesizer, where each binaural signal synthesizer instance generates the binaural signal corresponding to a different audio source. Since each different audio source has a different location relative to the listener, a different set of spatial parameters used to generate the binaural audio signal for each different audio source.
  • FIG. 2 shows a high-level block diagram of conventional auditory scene synthesizer 200, which converts a plurality of audio source signals (e.g., a plurality of mono signals) into the left and right audio signals of a single combined binaural signal, using a different set of spatial parameters for each different audio source. The left audio signals are then combined (e.g., by simple addition) to generate the left audio signal for the resulting auditory scene, and similarly for the right.
  • One of the applications for auditory scene synthesis is in conferencing, Assume, for example, a desktop conference with multiple participants, each of whom is sitting in front of his or her own personal computer (PC) in a different city. In addition to a PC monitor, each participant's PC is equipped with (1) a microphone that generates a mono audio source signal corresponding to that participant's contribution to the audio portion of the conference and (2) a set of headphones for playing that audio portion. Displayed on each participant's PC monitor is the image of a conference table as viewed from the perspective of a person sitting at one end of the table. Displayed at different locations around the table are real-time video images of the other conference participants.
  • In a conventional mono conferencing system, a server combines the mono signals from all of the participants into a single combined mono signal that is transmitted back to each participant. In order to make more realistic the perception for each participant that he or she is sitting around an actual conference table in a room with the other participants, the server can implement an auditory scene synthesizer, such as synthesizer 200 of FIG. 2, that applies an appropriate set of spatial parameters to the mono audio signal from each different participant and then combines the different left and right audio signals to generate left and right audio signals of a single combined binaural signal for the auditory scene. The left and right audio signals for this combined binaural signal are then transmitted to each participant. One of the problems with such conventional stereo conferencing systems relates to transmission bandwidth, since the server has to transmit a left audio signal and a right audio signal to each conference participant.
  • SUMMARY OF THE INVENTION
  • The present invention is directed to a technique for synthesizing auditory scenes that addresses the transmission bandwidth problem of the prior art. According to the present invention, an auditory scene corresponding to multiple audio sources located at different positions relative to the listener is synthesized from a single combined (e.g., mono) audio signal. As such, in the case of the conference described previously, a solution can be implemented in which each participant's PC receives only a single mono audio signal corresponding to a combination of the mono audio source signals from all of the participants.
  • The present invention is based on an assumption that, for those frequency bands in which the energy of the source signal from a particular audio source dominates the energies of all other source signals in the combined audio signal, from the perspective of the perception by the listener, the combined audio signal can be treated as if it corresponded solely to that particular audio source. According to implementations of the present invention, different sets of spatial parameters (corresponding to different audio sources) are applied to different frequency bands in the combined audio signal where different audio sources dominate, to synthesize an auditory scene.
  • In one embodiment, the present invention is a method for synthesizing an auditory scene, comprising the steps of (a) dividing an input audio signal into a plurality of different frequency bands; and (b) applying two or more different sets of one or more spatial parameters to two or more of the different frequency bands in the input audio signal to generate two or more synthesized audio signals of the auditory scene, wherein for each of the two or more different frequency bands, the corresponding set of one or more spatial parameters is applied to the input audio signal as if the input audio signal corresponded to a single audio source in the auditory scene.
  • In another embodiment, the present invention is an apparatus for synthesizing an auditory scene, comprising (1) an auditory scene synthesizer configured to (a) divide an input audio signal into a plurality of different frequency bands; and (b) apply two or more different sets of one or more spatial parameters to two or more of the different frequency bands in the input audio signal to generate two or more synthesized audio signals of the auditory scene, wherein for each of the two or more different frequency bands, the corresponding set of one or more spatial parameters is applied to the input audio signal as if the input audio signal corresponded to a single audio source in the auditory scene; and (2) one or more inverse time-frequency transformers configured to convert the two or more synthesized audio signals from a frequency domain into a time domain.
  • In yet another embodiment, the present invention is a method for processing two or more input audio signals, comprising the steps of (a) converting the two or more input audio signals from a time domain into a frequency domain; (b) generating a set of one or more auditory scene parameters for each of two or more different frequency bands in the two or more converted input audio signals, where each set of one or more auditory scene parameters is generated as if the corresponding frequency band corresponded to a single audio source in an auditory scene; and (c) combining the two or more input audio signals to generate a combined audio signal.
  • In yet another embodiment, the present invention is an apparatus for processing two or more input audio signals, comprising (a) a time-frequency transformer configured to convert the two or more input audio signals from a time domain into a frequency domain; (b) an auditory scene parameter generator configure to generate a set of one or more auditory scene parameters for each of two or more different frequency bands in the two or more converted input audio signals, where each set of one or more auditory scene parameters is generated as if the corresponding frequency band corresponded to a single audio source in an auditory scene; and (c) a combiner configured to combine the two or more input audio signals to generate a combined audio signal.
  • BRIEF DESCRIPTION OF THE DRAWINGS
  • Other aspects, features, and advantages of the present invention will become more fully apparent from the following detailed description, the appended claims, and the accompanying drawings in which:
  • FIG. 1 shows a high-level block diagram of conventional binaural signal synthesizer that converts a single audio source signal (e.g., a mono signal) into the left and right audio signals of a binaural signal;
  • FIG. 2 shows a high-level block diagram of conventional auditory scene synthesizer that converts a plurality of audio source signals (e.g., a plurality of mono signals) into the left and right audio signals of a single combined binaural signal;
  • FIG. 3 shows a block diagram of a conferencing system, according to one embodiment of the present invention;
  • FIG. 4 shows a block diagram of the audio processing implemented by the conference server of FIG. 3, according to one embodiment of the present invention;
  • FIG. 5 shows a flow diagram of the processing implemented by the auditory scene parameter generator of FIG. 4, according to one embodiment of the present invention;
  • FIG. 6 shows a graphical representation of the power spectra of the audio signals from three different exemplary sources;
  • FIG. 7 shows a block diagram of the audio processing performed by each conference node in FIG. 3;
  • FIG. 8 shows a graphical representation of the power spectrum in the frequency domain for the combined signal generated from the three mono source signals in FIG. 6;
  • FIG. 9 shows a representation of the analysis window for the time-frequency domain, according to one embodiment of the present invention; and
  • FIG. 10 shows a block diagram of the transmitter for an alternative application of the present invention, according to one embodiment of the present invention.
  • DETAILED DESCRIPTION
  • FIG. 3 shows a block diagram of a conferencing system 300, according to one embodiment of the present invention. Conferencing system 300 comprises conference server 302, which supports conferencing between a plurality of conference participants, where each participant uses a different conference node 304. In preferred embodiments of the present invention, each node 304 is a personal computer (PC) equipped with a microphone 306 and headphones 308, although other hardware configurations are also possible. Since the present invention is directed to processing of the audio portion of conferences, the following description omits reference to the processing of the video portion of such conferences, which involves the generation, manipulation, and display of video signals by video cameras, video signal processors, and digital monitors that would be included in conferencing system 300, but are not explicitly represented in FIG. 3. The present invention can also be implemented for audio-only conferencing.
  • As indicated in FIG. 3, each node 304 transmits a (e.g., mono) audio source signal generated by its microphone 306 to server 302, where that source signal corresponds to the corresponding participant's contribution to the conference. Server 302 combines the source signals from the different participants into a single (e.g., mono) combined audio signal and transmits that combined signal back to each node 304. (Depending on the type of echo-cancellation performed, if any, the combined signal transmitted to each node 304 may be either unique to that node or the same as the combined signal transmitted to every other node.) In addition to the combined signal, server 302 transmits an appropriate set of auditory scene parameters to each node 304. Each node 304 applies the set of auditory scene parameters to the combined signal in a manner according to the present invention to generate a binaural signal for rendering by headphones 308 and corresponding to the auditory scene for the conference.
  • The processing of conference server 302 may be implemented within a distinct node of conferencing system 300. Alternatively, the server processing may be implemented in one of the conference nodes 304, or even distributed among two or more different conference nodes 304.
  • FIG. 4 shows a block diagram of the audio processing implemented by conference server 302 of FIG. 3, according to one embodiment of the present invention. As shown in FIG. 4, auditory scene parameter generator 402 generates one or more sets of auditory scene parameters from the plurality of source signals generated by and received from the various conference nodes 304 of FIG. 3. In addition, signal combiner 404 combines the plurality of source signals (e.g., using straightforward audio signal addition) to generate the combined signal that is transmitted back to each conference node 304.
  • FIG. 5 shows a flow diagram of the processing implemented by auditory scene parameter generator 402 of FIG. 4, according to one embodiment of the present invention. Generator 402 applies a time-frequency (TF) transform, such as a discrete Fourier transform (DFT), to convert each node's source signal to the frequency domain (step 502 of FIG. 5). Generator 402 then compares the power spectra of the different converted source signals to identify one or more frequency bands in which the energy one of the source signals dominates all of the other signals (step 504).
  • Depending on the implementation, different criteria may be applied to determine whether a particular source signal dominates the other source signals. For example, a particular source signal may be said to dominate all of the other source signals when the energy of that source signal exceeds the sum of the energies in the other source signals by either a specified factor or a specified amount of power (e.g., in dBs). Alternatively, a particular source signal may be said to dominate when the energy of that source signal exceeds the second most powerful source signal by a specified factor or a specified amount of power. Other criteria are, of course, also possible, including those that combine two or more different comparisons. For example, in addition to relative domination, a source signal might have to have an absolute energy level that exceeds a specified energy level before qualifying as a dominating source signal.
  • FIG. 6 shows a graphical representation of the power spectra of the audio signals from three different exemplary sources (labeled A, B, and C). FIG. 6 identifies eight different frequency bands in which one of the three source signals dominates the other two. Note that, in FIG. 6, there are particular frequency ranges in which none of the three source signals dominate. Note also that the lengths of the dominated frequency ranges (i.e., frequency ranges in which one of the source signals dominates) are not uniform, but rather are dictated by the characteristics of the power spectra themselves.
  • Returning to FIG. 5, after generator 402 identifies one or more frequency bands in which one of the source signals dominates, a set of auditory scene parameters is generated for each frequency band, where those parameters correspond to the node whose source signal dominates that frequency band (step 506). In some implementations, the processing of step 506 implemented by generator 402 generates the actual spatial parameters (e.g., ILD, ITD, and/or HRTF) for each dominated frequency band. In those cases, generator 402 receives (e.g., a priori) information about the relative spatial placement of each participant in the auditory scene to be synthesized (as indicated in FIG. 4). In addition to the combined signal, at least the following auditory scene parameters are transmitted to each conference node 304 of FIG. 3 for each dominated frequency band:
      • (1) Frequency of the start of the frequency band;
      • (2) Frequency of the end of the frequency band; and
      • (3) One or more spatial parameters (e.g., ILD, ITD, and/or HRTF) for the frequency band.
        Although the identity of the particular node/participant whose source signal dominates the frequency band can be transmitted, such information is not required for the subsequent synthesis of the auditors scene. Note that, for those frequency bands, for which no source signal is determined to dominate, no auditory scene parameters or other special information needs to be transmitted to the different conference nodes 304.
  • In other implementations, the generation of the spatial parameters for each dominated frequency band is implemented independently at each conference node 304. In those cases, generator 402 does not need any information about the relative spatial placements of the various participants in the synthesized auditory scene. Rather, in addition to the combined signal, only the following auditory scene parameters need to be transmitted to each conference node 304 for each dominated frequency band:
      • (1) Frequency of the start of the frequency band;
      • (2) Frequency of the end of the frequency band; and
      • (3) Identity of the node/participant whose source signal dominates the frequency band.
        In such implementations, each conference node 304 is responsible for generating the appropriate spatial parameters for each dominated frequency range. Such implementation enables each different conference node to generate a unique auditory scene (e.g., corresponding to different relative placements of the various conference participants within the synthesized auditory scene).
  • In either type of implementation, the processing of FIG. 5 is preferably repeated at a specified interval (e.g., once for every 20-msec frame of audio data). As a result, the number and definition of the dominated frequency ranges as well as the particular source signals that dominate those ranges will typically vary over time (e.g., from frame to frame), reflecting the fact that the set of conference participants who are speaking at any given time will vary over time as will the characteristics of their own individual voices (e.g., intonations and/or volumes). Depending on the implementation, the spatial parameters corresponding to each conference participant may be either static (e.g., for synthesis of stationary participants whose relative positions do not change over time) or dynamic (e.g., for synthesis of mobile participants who relative positions are allowed to change over time).
  • In alternative embodiments, rather than selecting a set of spatial parameters that corresponds to a single source, a set of spatial parameters can be generated that reflects the contributions of two or more—or even all—of the participants. For example, weighted averaging can be used to generate an ILD value that represents the relative contributions for the two or more most dominant participants. In such cases, each set of spatial parameters is a function of the relative dominance of the most dominant participants for a particular frequency band.
  • FIG. 7 shows a block diagram of the audio processing performed by each conference node 304 in FIG. 3 to convert a single combined mono audio signal and corresponding auditory scene parameters received from conference server 302 into the binaural signal for a synthesized auditory scene. In particular, time-frequency (TF) transform 702 converts each frame of the combined signal into the frequency domain.
  • For each dominated frequency band, auditory scene synthesizer 704 applies the corresponding auditory scene parameters to the converted combined signal to generate left and right audio signals for that frequency band in the frequency domain. In particular, for each audio frame and for each dominated frequency band, synthesizer 704 applies the set of spatial parameters corresponding to the participant whose source signal dominates the combined signal for that dominated frequency range. If the auditory scene parameters received from the conference server do not include the spatial parameters for each conference participant, then synthesizer 704 receives information about the relative spatial placement of the different participants in the synthesized auditory scene as indicated in FIG. 7, so that the set of spatial parameters for each dominated frequency band in the combined signal can be generated locally at the conference node.
  • An inverse TF transform 706 is then applied to each of the left and right audio signals to generate the left and right audio signals of the binaural signal in the time domain corresponding to the synthesized auditory scene. The resulting auditory scene is perceived as being approximately the same as for an ideally synthesized binaural signal with the same corresponding spatial parameters but applied over the whole spectrum of each individual source signal.
  • FIG. 8 shows a graphical representation of the power spectrum in the frequency domain for the combined signal generated from the three mono source signals from sources A, B, and C in FIG. 6. In addition to showing the three different source signals (dotted lines), FIG. 8 also shows the same frequency bands identified in FIG. 6 in which the power of one of the three source signals dominates the other two. It is to these dominated frequency bands to which auditory scene synthesizer 704 applies appropriate sets of spatial parameters.
  • In a typical audio frame, not all of the conference participants will dominate at least one frequency band, since not all of the participants will typically be talking at the same time. If only one participant is talking, then only that participant will typically dominate any of the frequency bands. By the same token, during an audio frame corresponding to relative silence, it may be that none of the participants will dominate any frequency bands. For those frequency bands for which no dominating participant is identified, no spatial parameters are applied and the left and right audio signals of the resulting binaural signal for those frequency bands are identical.
  • Time-Frequency Transform
  • As indicated above, TF transform 702 in FIG. 7 converts the combined mono audio signal to the spectral (i.e., frequency) domain frame-wise in order for the system to operate for real-time applications. For each frequency band n at each time k (e.g., frame number k), a level difference ΔLn[k], a time difference τn[k], and/or an HRTF is to be introduced into the underlying audio signal. In a preferred embodiment, TF transform 702 is a DFT-based transform, such as those described in A. V. Oppenheim and R. W. Schaefer, Discrete-Time Signal Processing, Signal Processing Series, Prentice Hall, 1989, the teachings of which are incorporated herein by reference. The transform is derived based on the desire for the ability to synthesize frequency-dependent and time-adaptive time differences τn[k]. The same transform can be used advantageously for the synthesis of frequency-dependent and time-adaptive level differences ΔLn[k] and for HRTFs.
  • When W samples s0 . . . , sW−1 in the time domain are converted to W samples S0. . . SW−1 in a complex spectral domain with a DFT transform, then a circular time-shift of d time-domain samples can be obtained by modifying the W spectral values according to Equation (1) as follows: S ^ n = S n 2 π nd W . ( 1 )
    In order to introduce a non-circular time-shift within each frame (as opposed to a circular time-shift), the time-domain samples s0 . . . , sW−1 are padded with zeros at the beginning and at the end of the frame and a DFT of size N=2Z+W is then used. By modifying the resulting spectral coefficients, a non-circular time-shift within the range dε[−Z,Z] can be implemented by modifying the resulting N spectral coefficients according to Equation (2) as follows: S ^ n = S n 2 π nd N . ( 2 )
  • The described scheme works as long as the time-shift d does not vary in time. Since the desired d usually varies over time, the transitions are smoothed by using overlapping windows for the analysis transform. A frame of N samples is multiplied with the analysis window before an N-point DFT is applied. The following Equation (3) shows the analysis window, which includes the zero padding at the beginning and at the end of the frame: w a [ k ] = 0 for k < Z w a [ k ] = sin 2 ( ( k - Z ) π W ) for Z k < Z + W w a [ k ] = 0 for Z + W k ( 3 )
    where Z is the width of the zero region before and after the window. The non-zero window span is W, and the size of the transform s N=2Z+W.
  • FIG. 9 shows a representation of the analysis window, which was chosen such that it is additive to one when windows of adjacent frames are overlapped by W/2 samples. The time-span of the window shown in FIG. 9 is shorter than the DFT length such that non-circular time-shifts within the range [−Z,Z] are possible. To gain more flexibility in changing time differences, level differences, and HRTFs in time and frequency, a higher factor of oversampling can be used by choosing the time-span of the window to be smaller and/or by overlapping the windows more.
  • The zero padding of the analysis window shown in FIG. 9 allows the implementation of convolutions with HRTFs as simple multiplications in the frequency domain. Therefore, the transform is also suitable for the synthesis of HRTFs in addition to time and level differences. A more general and slightly different point of view of a similar transform is given by J. B. Allen, “Short-term spectral analysis, synthesis and modification by discrete fourier transform, ” IEEE Trans. on Speech and Signal Processing, vol. ASSP-25, pp. 235-238, June 1977, the teachings of which are incorporated herein by reference.
  • Obtaining A Binaural Signal From A Mono Signal
  • In certain implementations, auditory scene synthesizer 704 of FIG. 7 applies different sets of specified level and time differences to the different dominated frequency bands in the combined signal to generate the left and right audio signals of the binaural signal for the synthesized auditory scene. In particular, for each frame k, each dominated frequency band n is associated with a level difference ΔLn[k] and a time difference τn[k]. In preferred embodiments, these level and time differences are applied symmetrically to the spectrum of the combined signal to generate the spectra of the left and right audio signals according to Equations (4) and (5), respectively, as follows: S n L = 10 Δ L n 10 1 + 10 Δ L n 10 S n - 2 π n τ n 2 N and ( 4 ) S n R = 1 1 + 10 2 Δ L n 10 S n - 2 π n τ n 2 N ( 5 )
    where {Sn} are the spectral coefficients of the combined signal and {Sn L} and {Sn R} are the spectral coefficients of the resulting binaural signal. The level differences {ΔLn} are expressed in dB and the time differences {τn} in numbers of samples.
  • For the spectral synthesis of auditory scenes based on HRTFs, the left and right spectra of the binaural signal may be obtained using Equations (6) and (7), respectively, as follows: S n L = m = 1 M w m , n H m , n L S n and ( 6 ) S n R = m = 1 M w m , n H m , n R S n ( 7 )
    where Hm,n L and Hm,n R are the complex frequency responses of the HRTFs corresponding to the sound source m. For each spectral coefficient, a weighted sum of the frequency responses of the HRTFs of all sources is applied with weights wm,n. The level differences ΔLn, time differences τn, and HRTF weights wm,n are preferably smoothed in frequency and time to prevent artifacts.
    Experimental Results
  • To evaluate how useful the present invention is for a desktop conferencing application, twelve participants were given a task which required responding to one of two simultaneous voice messages. This is a variation of the “cocktail party problem” of attending to one voice in the presence of others. The signals were presented to the participants with headphones in an acoustically isolated room. Five different signal kinds were tested for their effect on the ability to respond to one of two simultaneous messages:
      • Test 1: diotic: a mono signal to both ears
      • Test 2: ILDp: an ideally synthesized binaural signal with ILDs
      • Test 3: ITDp: an ideally synthesized binaural signal with ITDs
      • Test 4: ILDp: a binaural signal perceptually synthesized with ILDs using the present invention
      • Test 5: ITDp: a binaural signal perceptually synthesized with ITDs using the present invention
        Each of the participants took all of the tests in randomized order.
  • The tests used the speech corpus introduced in R. S. Bolia, W. T. Nelson, M. A. Ericson, and B. D. Simpson, “A speech corpus for multitalker communications research,” J. Acoust. Soc. Am., vol. 107, no. 2, pp. 1065-1066, February 2000, the teachings of which are incorporated herein by reference. Similar tests have also been conducted by others, such as reported in R. S. Bolia, M. A. Ericson, W. T. McKinley, and B. D. Simpson, “A cocktail party effect in the median plane?,” J. Acoust. Soc. Am., vol. 105, pp. 1390-1391, 1999, and W. Spieth, J. F. Curtis, and J. C. Webster, “Responding to one of two simultaneous messages,” J. Acoust. Soc. Am., vol. 26, no. 3, pp. 391-396, 1954, the teachings of both of which are incorporated herein by reference.
  • A typical sentence of the corpus is “READY LAKER, GO TO BLUE FIVE NOW,” where LAKER is the call sign and BLUE FIVE is a color-number combination. Combinations of the eight different call signs, four different colors, and eight different numbers were chosen randomly with the restriction that the call sign assigned to the participant occurred in 50% of the cases.
  • In the tests, each participant was instructed to respond when his or her call sign was called by indicating the color-number combination by the talker who called the call sign. One out of four female talkers was randomly chosen for each of the two talkers in each test item. One talker was spatially placed at the right side and the other at the left side for Tests 2 and 4 (ILD=±16 dB) and for Tests 3 and 5 (ITD=±500 μsec). Each of the five tests consisted of 20 test items which were preceded by 10 training items.
  • Table I shows the results for the case when the listeners were called by their call signs. The upper row shows the percentage of correct identification of the call sign, and the lower row shows the conditional percentage of the correct color-number combination given that the listener's call sign was correctly identified. These results suggest that the percentages of correct identification of the call sign and of the color and number significantly improve for ideally synthesized binaural signals (Tests 2 and 3) or perceptually synthesized binaural signals (Tests 4 and 5) over the diotic signal (Test 1), with the perceptually synthesized signals of Tests 4 and 5 being almost as good as the ideally synthesized signals of Tests 2 and 3. For the cases when the listeners were not called, the percentages of the listeners responding was below two percent for all five tests.
    TABLE 1
    Test 1 Test 2 Test 3 Test 4 Test 5
    call sign 70% 78% 85% 77% 78%
    color-number 64% 98% 88% 96% 91%

    Alternative Embodiments
  • In the previous sections, the present invention was described in the context of a desktop conferencing application. The present invention can also be employed for other applications. For example, the present invention can be applied where the input is a binaural signal corresponding to an (actual or synthesized) auditory scene, rather than the input being individual mono source signals as in the previous application. In this latter application, the binaural signal is converted into a single mono signal and auditory scene parameters (e.g., sets of spatial parameters). As in the desktop conferencing application, this application of the present invention can be used to reduce the transmission bandwidth requirements for the auditory scene since, instead of having to transmit the individual left and right audio signals for the binaural signal, only a single mono signal plus the relatively small amount of spatial parameter information need to be transmitted to a receiver, where the receiver performs processing similar to that shown in FIG. 7.
  • FIG. 10 shows a block diagram of transmitter 1000 for such an application, according to one embodiment of the present invention. As shown in FIG. 10, a TF transform 1002 is applied to corresponding frames of each of the left and right audio signals of the input binaural signal to convert the signals to the frequency domain. Auditory scene analyzer 1004 processes the converted left and right audio signals in the frequency domain to generate a set of auditory scene parameters for each of a plurality of different frequency bands in those converted signals. In particular, for each corresponding pair of audio frames, analyzer 1004 divides the converted left and right audio signals into a plurality of frequency bands. Depending on the implementation, each of the left and right audio signals can be divided into the same number of equally sized frequency bands. Alternatively, the size of the frequency bands may vary with frequency, e.g., larger frequency bands for higher frequencies or smaller frequency bands for higher frequencies.
  • For each corresponding pair of frequency bands, analyzer 1004 compares the converted left and right audio signals to generate one or more spatial parameters (e.g., an ILD value, an ITD value, and/or an HRTF). In particular, for each frequency band, the cross-correlation between the converted left and right audio signals is estimated. The maximum value of the cross-correlation, which indicates how much the two signals are correlated, can be used as a measure for the dominance of one source in the band. If there is 100% correlation between the left and right audio signals, then only one source's energy is dominant in that frequency band. The less the cross-correlation maximum is, the less is just one source dominant. The location in time of the maximum of the cross-correlation can be used to correspond to the ITD. The ILD can be obtained by computing the level difference of the power spectral values of the left and right audio signals. In this way, each set of spatial parameters is generated by treating the corresponding frequency range as if it were dominated by a single source signal. For those frequency bands where this assumption is true, the generated set of spatial parameters will be fairly accurate. For those frequency bands where this assumption is not true, the generated set of spatial parameters will have less physical significance to the actual auditory scene. On the other hand, the assumption is that those frequency bands contribute less significantly to the overall perception of the auditory scene. As such, the application of such “less significant” spatial parameters will have little if any adverse affect on the resulting auditory scene. In any case, transmitter 1000 transmits these auditory scene parameters to the receiver for use in reconstructing the auditory scene from the mono audio signal.
  • Auditory scene remover 1006 combines the converted left and right audio signals in the frequency domain to generate the mono audio signal. In a basic implementation, remover 1006 simply averages the left and right audio signals. In preferred implementations, however, more sophisticated processing is performed to generate the mono signal. In particular, for example, the spatial parameters generated by auditory scene analyzer 1004 can be used to modify both the left and right audio signals in the frequency domain as part of the process of generating the mono signal, where each different set of spatial parameters is used to modify a corresponding frequency band in each of the left and right audio signals. For example, if the generated spatial parameters include an ITD value for each frequency band, then the left and right audio signals in each frequency band can be appropriately time shifted using the corresponding ITD value to make the ITD between the left and right audio signals become zero. The power spectra for the time-shifted left and right audio signals can then be added such that the perceived loudness of each frequency band is the same in the resulting mono signal as in the original binaural signal.
  • An inverse TF transform 1008 is then applied to the resulting mono audio signal in the frequency domain to generate the mono audio signal in the time domain. The mono audio signal can then be compressed and/or otherwise processed for transmission to the receiver. Since a receiver having a configuration similar to that in FIG. 7 converts the mono audio signal back into the frequency domain, the possibility exists for omitting inverse TF transform 1008 of FIG. 10 and TF transform 702 of FIG. 7, where the transmitter transmits the mono audio signal to the receiver in the frequency domain.
  • As in the previous application, the receiver applies the received auditory scene parameters to the received mono audio signal to synthesize (or, in this latter case, reconstruct an approximation of) the auditory scene. Note that, in is latter application, there is no need for any a priori knowledge of either the number of sources involved in the original auditory scene or their relative positions. In this latter application, there is no identification of particular sources with particular frequency bands. Rather, the frequency bands are selected in an open-loop manner, but processed with the same underlying assumption as the previous application: that is, that each frequency band can be treated as if it corresponded to a single source using a corresponding set of spatial parameters.
  • Although this latter application has been described in the context of processing in which the input is a binaural signals, this application of the present invention can be extended to (two or multi-channel) stereo signals. Similarly, although the invention has been described in the context of systems that generate binaural signals corresponding to auditory scenes perceived using headphones, the present invention can be extended to apply to the generation of (two or multi-channel) stereo signals for loudspeaker playback.
  • The present invention may be implemented as circuit-based processes, including possible implementation on a single integrated circuit. As would be apparent to one skilled in the art, various functions of circuit elements may also be implemented as processing steps in a software program. Such software may be employed in, for example, a digital signal processor, micro-controller, or general-purpose computer.
  • The present invention can be embodied in the form of methods and apparatuses for practicing those methods. The present invention can also be embodied in the form of program code embodied in tangible media, such as floppy diskettes, CD-ROMs, hard drives, or any other machine-readable storage medium, wherein, when the program code is loaded into and executed by a machine, such as a computer, the machine becomes an apparatus for practicing the invention. The present invention can also be embodied in the form of program code, for example, whether stored in a storage medium, loaded into and/or executed by a machine, or transmitted over some transmission medium or carrier, such as over electrical wiring or cabling, through fiber optics, or via electromagnetic radiation, wherein, when the program code is loaded into and executed by a machine, such as a computer, the machine becomes an apparatus for practicing the invention. When implemented on a general-purpose processor, the program code segments combine with the processor to provide a unique device that operates analogously to specific logic circuits.
  • It will be further understood that various changes in the details, materials, and arrangements of the parts which have been described and illustrated in order to explain the nature of this invention may be made by those skilled in the art without departing from the scope of the invention as expressed in the following claims.

Claims (1)

1. A method for synthesizing an auditory scene, comprising the steps of:
(a) dividing an input audio signal into a plurality of different frequency bands; and
(b) applying two or more different sets of one or more spatial parameters to two or more of the different frequency bands in the input audio signal to generate two or more synthesized audio signals of the auditory scene, wherein for each of the two or more different frequency bands, the corresponding set of one or more spatial parameters is applied to the input audio signal as if the input audio signal corresponded to a single audio source in the auditory scene.
US11/470,314 2001-05-04 2006-09-06 Perceptual synthesis of auditory scenes Abandoned US20070003069A1 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
US11/470,314 US20070003069A1 (en) 2001-05-04 2006-09-06 Perceptual synthesis of auditory scenes

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US09/848,877 US7116787B2 (en) 2001-05-04 2001-05-04 Perceptual synthesis of auditory scenes
US11/470,314 US20070003069A1 (en) 2001-05-04 2006-09-06 Perceptual synthesis of auditory scenes

Related Parent Applications (1)

Application Number Title Priority Date Filing Date
US09/848,877 Continuation US7116787B2 (en) 2001-05-04 2001-05-04 Perceptual synthesis of auditory scenes

Publications (1)

Publication Number Publication Date
US20070003069A1 true US20070003069A1 (en) 2007-01-04

Family

ID=25304522

Family Applications (2)

Application Number Title Priority Date Filing Date
US09/848,877 Expired - Lifetime US7116787B2 (en) 2001-05-04 2001-05-04 Perceptual synthesis of auditory scenes
US11/470,314 Abandoned US20070003069A1 (en) 2001-05-04 2006-09-06 Perceptual synthesis of auditory scenes

Family Applications Before (1)

Application Number Title Priority Date Filing Date
US09/848,877 Expired - Lifetime US7116787B2 (en) 2001-05-04 2001-05-04 Perceptual synthesis of auditory scenes

Country Status (1)

Country Link
US (2) US7116787B2 (en)

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20060235679A1 (en) * 2005-04-13 2006-10-19 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Adaptive grouping of parameters for enhanced coding efficiency
US20060235683A1 (en) * 2005-04-13 2006-10-19 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Lossless encoding of information with guaranteed maximum bitrate
US20090041271A1 (en) * 2007-06-29 2009-02-12 France Telecom Positioning of speakers in a 3D audio conference
US20100169103A1 (en) * 2007-03-21 2010-07-01 Ville Pulkki Method and apparatus for enhancement of audio reconstruction
US20100166191A1 (en) * 2007-03-21 2010-07-01 Juergen Herre Method and Apparatus for Conversion Between Multi-Channel Audio Formats
US20180078815A1 (en) * 2016-09-18 2018-03-22 Foxconn Interconnect Technology Limited Treadmill and monitoring system

Families Citing this family (88)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8767969B1 (en) 1999-09-27 2014-07-01 Creative Technology Ltd Process for removing voice from stereo recordings
US7644003B2 (en) * 2001-05-04 2010-01-05 Agere Systems Inc. Cue-based audio coding/decoding
US7116787B2 (en) * 2001-05-04 2006-10-03 Agere Systems Inc. Perceptual synthesis of auditory scenes
US7292901B2 (en) * 2002-06-24 2007-11-06 Agere Systems Inc. Hybrid multi-channel/cue coding/decoding of audio signals
US7583805B2 (en) * 2004-02-12 2009-09-01 Agere Systems Inc. Late reverberation-based synthesis of auditory scenes
US20030035553A1 (en) * 2001-08-10 2003-02-20 Frank Baumgarte Backwards-compatible perceptual coding of spatial cues
US20050180578A1 (en) * 2002-04-26 2005-08-18 Cho Nam I. Apparatus and method for adapting audio signal
US7257231B1 (en) * 2002-06-04 2007-08-14 Creative Technology Ltd. Stream segregation for stereo signals
GB2391439B (en) * 2002-07-30 2006-06-21 Wolfson Ltd Bass compressor
FR2851879A1 (en) * 2003-02-27 2004-09-03 France Telecom PROCESS FOR PROCESSING COMPRESSED SOUND DATA FOR SPATIALIZATION.
DE10330808B4 (en) * 2003-07-08 2005-08-11 Siemens Ag Conference equipment and method for multipoint communication
US7542815B1 (en) 2003-09-04 2009-06-02 Akita Blue, Inc. Extraction of left/center/right information from two-channel stereo sources
ES2291939T3 (en) * 2003-09-29 2008-03-01 Koninklijke Philips Electronics N.V. CODING OF AUDIO SIGNALS.
US7447317B2 (en) 2003-10-02 2008-11-04 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V Compatible multi-channel coding/decoding by weighting the downmix channel
US7970144B1 (en) 2003-12-17 2011-06-28 Creative Technology Ltd Extracting and modifying a panned source for enhancement and upmix of audio signals
US7394903B2 (en) * 2004-01-20 2008-07-01 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Apparatus and method for constructing a multi-channel output signal or for generating a downmix signal
DE102004009628A1 (en) * 2004-02-27 2005-10-06 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus and method for writing an audio CD and an audio CD
US7805313B2 (en) * 2004-03-04 2010-09-28 Agere Systems Inc. Frequency-based coding of channels in parametric multi-channel coding systems
ATE474310T1 (en) * 2004-05-28 2010-07-15 Nokia Corp MULTI-CHANNEL AUDIO EXPANSION
KR100663729B1 (en) 2004-07-09 2007-01-02 한국전자통신연구원 Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information
TWI497485B (en) * 2004-08-25 2015-08-21 Dolby Lab Licensing Corp Method for reshaping the temporal envelope of synthesized output audio signal to approximate more closely the temporal envelope of input audio signal
TWI393121B (en) 2004-08-25 2013-04-11 Dolby Lab Licensing Corp Method and apparatus for processing a set of n audio signals, and computer program associated therewith
EP1784136B1 (en) * 2004-08-30 2011-03-16 Synthes GmbH Hand-held motorized injection device with haptic feedback for highly viscous materials
DE102004042819A1 (en) 2004-09-03 2006-03-23 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus and method for generating a coded multi-channel signal and apparatus and method for decoding a coded multi-channel signal
JP4594681B2 (en) * 2004-09-08 2010-12-08 ソニー株式会社 Audio signal processing apparatus and audio signal processing method
DE102004043521A1 (en) * 2004-09-08 2006-03-23 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Device and method for generating a multi-channel signal or a parameter data set
US7720230B2 (en) * 2004-10-20 2010-05-18 Agere Systems, Inc. Individual channel shaping for BCC schemes and the like
US8204261B2 (en) * 2004-10-20 2012-06-19 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Diffuse sound shaping for BCC schemes and the like
US8340306B2 (en) * 2004-11-30 2012-12-25 Agere Systems Llc Parametric coding of spatial audio with object-based side information
US7787631B2 (en) * 2004-11-30 2010-08-31 Agere Systems Inc. Parametric coding of spatial audio with cues based on transmitted channels
WO2006060278A1 (en) * 2004-11-30 2006-06-08 Agere Systems Inc. Synchronizing parametric coding of spatial audio with externally provided downmix
US7903824B2 (en) * 2005-01-10 2011-03-08 Agere Systems Inc. Compact side information for parametric coding of spatial audio
EP1691348A1 (en) * 2005-02-14 2006-08-16 Ecole Polytechnique Federale De Lausanne Parametric joint-coding of audio sources
DE102005010057A1 (en) * 2005-03-04 2006-09-07 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus and method for generating a coded stereo signal of an audio piece or audio data stream
DE102005014477A1 (en) * 2005-03-30 2006-10-12 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus and method for generating a data stream and generating a multi-channel representation
KR100588218B1 (en) * 2005-03-31 2006-06-08 엘지전자 주식회사 Mono compensation stereo system and signal processing method thereof
US7983922B2 (en) * 2005-04-15 2011-07-19 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Apparatus and method for generating multi-channel synthesizer control signal and apparatus and method for multi-channel synthesizing
TWI396188B (en) 2005-08-02 2013-05-11 Dolby Lab Licensing Corp Controlling spatial audio coding parameters as a function of auditory events
RU2419249C2 (en) * 2005-09-13 2011-05-20 Кониклейке Филипс Электроникс Н.В. Audio coding
KR101315070B1 (en) 2005-09-13 2013-10-08 코닌클리케 필립스 일렉트로닉스 엔.브이. A method of and a device for generating 3D sound
WO2007080225A1 (en) * 2006-01-09 2007-07-19 Nokia Corporation Decoding of binaural audio signals
WO2007080211A1 (en) * 2006-01-09 2007-07-19 Nokia Corporation Decoding of binaural audio signals
KR100803212B1 (en) * 2006-01-11 2008-02-14 삼성전자주식회사 Method and apparatus for scalable channel decoding
PL1989920T3 (en) * 2006-02-21 2010-07-30 Koninl Philips Electronics Nv Audio encoding and decoding
KR100773560B1 (en) 2006-03-06 2007-11-05 삼성전자주식회사 Method and apparatus for synthesizing stereo signal
KR100763920B1 (en) * 2006-08-09 2007-10-05 삼성전자주식회사 Method and apparatus for decoding input signal which encoding multi-channel to mono or stereo signal to 2 channel binaural signal
RU2460155C2 (en) * 2006-09-18 2012-08-27 Конинклейке Филипс Электроникс Н.В. Encoding and decoding of audio objects
WO2008039042A1 (en) * 2006-09-29 2008-04-03 Lg Electronics Inc. Methods and apparatuses for encoding and decoding object-based audio signals
WO2008060111A1 (en) * 2006-11-15 2008-05-22 Lg Electronics Inc. A method and an apparatus for decoding an audio signal
KR101062353B1 (en) * 2006-12-07 2011-09-05 엘지전자 주식회사 Method for decoding audio signal and apparatus therefor
KR101370354B1 (en) * 2007-02-06 2014-03-06 코닌클리케 필립스 엔.브이. Low complexity parametric stereo decoder
ATE526663T1 (en) * 2007-03-09 2011-10-15 Lg Electronics Inc METHOD AND DEVICE FOR PROCESSING AN AUDIO SIGNAL
KR20080082916A (en) * 2007-03-09 2008-09-12 엘지전자 주식회사 A method and an apparatus for processing an audio signal
JP4449998B2 (en) * 2007-03-12 2010-04-14 ヤマハ株式会社 Array speaker device
JP4488036B2 (en) * 2007-07-23 2010-06-23 ヤマハ株式会社 Speaker array device
WO2009031870A1 (en) * 2007-09-06 2009-03-12 Lg Electronics Inc. A method and an apparatus of decoding an audio signal
US8559611B2 (en) * 2008-04-07 2013-10-15 Polycom, Inc. Audio signal routing
US8355921B2 (en) * 2008-06-13 2013-01-15 Nokia Corporation Method, apparatus and computer program product for providing improved audio processing
US9025775B2 (en) * 2008-07-01 2015-05-05 Nokia Corporation Apparatus and method for adjusting spatial cue information of a multichannel audio signal
JP5577597B2 (en) * 2009-01-28 2014-08-27 ヤマハ株式会社 Speaker array device, signal processing method and program
CN101510988B (en) * 2009-02-19 2012-03-21 华为终端有限公司 Method and apparatus for processing and playing voice signal
US20120076305A1 (en) * 2009-05-27 2012-03-29 Nokia Corporation Spatial Audio Mixing Arrangement
CA2765116C (en) 2009-06-23 2020-06-16 Nokia Corporation Method and apparatus for processing audio signals
US20100324915A1 (en) * 2009-06-23 2010-12-23 Electronic And Telecommunications Research Institute Encoding and decoding apparatuses for high quality multi-channel audio codec
WO2011000409A1 (en) 2009-06-30 2011-01-06 Nokia Corporation Positional disambiguation in spatial audio
US20110026745A1 (en) * 2009-07-31 2011-02-03 Amir Said Distributed signal processing of immersive three-dimensional sound for audio conferences
CN102576533B (en) * 2009-08-14 2014-09-17 Dts有限责任公司 Object-oriented audio streaming system
WO2011029984A1 (en) * 2009-09-11 2011-03-17 Nokia Corporation Method, apparatus and computer program product for audio coding
US20110103624A1 (en) * 2009-11-03 2011-05-05 Bran Ferren Systems and Methods for Providing Directional Audio in a Video Teleconference Meeting
US8380333B2 (en) * 2009-12-21 2013-02-19 Nokia Corporation Methods, apparatuses and computer program products for facilitating efficient browsing and selection of media content and lowering computational load for processing audio data
CN102770913B (en) 2009-12-23 2015-10-07 诺基亚公司 Sparse audio
US8463414B2 (en) * 2010-08-09 2013-06-11 Motorola Mobility Llc Method and apparatus for estimating a parameter for low bit rate stereo transmission
US8908874B2 (en) 2010-09-08 2014-12-09 Dts, Inc. Spatial audio encoding and reproduction
JP5582027B2 (en) * 2010-12-28 2014-09-03 富士通株式会社 Encoder, encoding method, and encoding program
US9026450B2 (en) 2011-03-09 2015-05-05 Dts Llc System for dynamically creating and rendering audio objects
US9794678B2 (en) * 2011-05-13 2017-10-17 Plantronics, Inc. Psycho-acoustic noise suppression
JP5870819B2 (en) * 2012-03-30 2016-03-01 ブラザー工業株式会社 Voice control device, voice control method, and voice control program
US9558785B2 (en) 2013-04-05 2017-01-31 Dts, Inc. Layered audio coding and transmission
CN105075294B (en) * 2013-04-30 2018-03-09 华为技术有限公司 Audio signal processor
CN104540073B (en) * 2014-12-16 2017-11-21 湖南科技大学 A kind of multichannel electronic firecrackers and its implementation
WO2016169591A1 (en) 2015-04-22 2016-10-27 Huawei Technologies Co., Ltd. An audio signal processing apparatus and method
CN107358960B (en) * 2016-05-10 2021-10-26 华为技术有限公司 Coding method and coder for multi-channel signal
CN107347173A (en) * 2017-06-01 2017-11-14 华南理工大学 The implementation method of multi-path surround sound dynamic ears playback system based on mobile phone
US10504529B2 (en) * 2017-11-09 2019-12-10 Cisco Technology, Inc. Binaural audio encoding/decoding and rendering for a headset
EP3985665A1 (en) * 2018-04-05 2022-04-20 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Apparatus, method or computer program for estimating an inter-channel time difference
US10667072B2 (en) 2018-06-12 2020-05-26 Magic Leap, Inc. Efficient rendering of virtual soundfields
JP7384162B2 (en) * 2018-08-17 2023-11-21 ソニーグループ株式会社 Signal processing device, signal processing method, and program
US10972835B2 (en) * 2018-11-01 2021-04-06 Sennheiser Electronic Gmbh & Co. Kg Conference system with a microphone array system and a method of speech acquisition in a conference system

Citations (67)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4236039A (en) * 1976-07-19 1980-11-25 National Research Development Corporation Signal matrixing for directional reproduction of sound
US4815132A (en) * 1985-08-30 1989-03-21 Kabushiki Kaisha Toshiba Stereophonic voice signal transmission system
US4972484A (en) * 1986-11-21 1990-11-20 Bayerische Rundfunkwerbung Gmbh Method of transmitting or storing masked sub-band coded audio signals
US5371799A (en) * 1993-06-01 1994-12-06 Qsound Labs, Inc. Stereo headphone sound source localization system
US5463424A (en) * 1993-08-03 1995-10-31 Dolby Laboratories Licensing Corporation Multi-channel transmitter/receiver system providing matrix-decoding compatible signals
US5579430A (en) * 1989-04-17 1996-11-26 Fraunhofer Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Digital encoding process
US5583962A (en) * 1991-01-08 1996-12-10 Dolby Laboratories Licensing Corporation Encoder/decoder for multidimensional sound fields
US5677994A (en) * 1994-04-15 1997-10-14 Sony Corporation High-efficiency encoding method and apparatus and high-efficiency decoding method and apparatus
US5682461A (en) * 1992-03-24 1997-10-28 Institut Fuer Rundfunktechnik Gmbh Method of transmitting or storing digitalized, multi-channel audio signals
US5701346A (en) * 1994-03-18 1997-12-23 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Method of coding a plurality of audio signals
US5703999A (en) * 1992-05-25 1997-12-30 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Process for reducing data in the transmission and/or storage of digital signals from several interdependent channels
US5706309A (en) * 1992-11-02 1998-01-06 Fraunhofer Geselleschaft Zur Forderung Der Angewandten Forschung E.V. Process for transmitting and/or storing digital signals of multiple channels
US5771295A (en) * 1995-12-26 1998-06-23 Rocktron Corporation 5-2-5 matrix system
US5812971A (en) * 1996-03-22 1998-09-22 Lucent Technologies Inc. Enhanced joint stereo coding method using temporal envelope shaping
US5825776A (en) * 1996-02-27 1998-10-20 Ericsson Inc. Circuitry and method for transmitting voice and data signals upon a wireless communication channel
US5850496A (en) * 1997-07-02 1998-12-15 Stryker Corporation Endoscope with integrated, self-regulating light source
US5860060A (en) * 1997-05-02 1999-01-12 Texas Instruments Incorporated Method for left/right channel self-alignment
US5878080A (en) * 1996-02-08 1999-03-02 U.S. Philips Corporation N-channel transmission, compatible with 2-channel transmission and 1-channel transmission
US5889843A (en) * 1996-03-04 1999-03-30 Interval Research Corporation Methods and systems for creating a spatial auditory environment in an audio conference system
US5890125A (en) * 1997-07-16 1999-03-30 Dolby Laboratories Licensing Corporation Method and apparatus for encoding and decoding multiple audio channels at low bit rates using adaptive selection of encoding method
US5912976A (en) * 1996-11-07 1999-06-15 Srs Labs, Inc. Multi-channel audio enhancement system for use in recording and playback and methods for providing same
US5930733A (en) * 1996-04-15 1999-07-27 Samsung Electronics Co., Ltd. Stereophonic image enhancement devices and methods using lookup tables
US5946352A (en) * 1997-05-02 1999-08-31 Texas Instruments Incorporated Method and apparatus for downmixing decoded data streams in the frequency domain prior to conversion to the time domain
US5956674A (en) * 1995-12-01 1999-09-21 Digital Theater Systems, Inc. Multi-channel predictive subband audio coder using psychoacoustic adaptive bit allocation in frequency, time and over the multiple channels
US6016473A (en) * 1998-04-07 2000-01-18 Dolby; Ray M. Low bit-rate spatial coding method and system
US6021389A (en) * 1998-03-20 2000-02-01 Scientific Learning Corp. Method and apparatus that exaggerates differences between sounds to train listener to recognize and identify similar sounds
US6108584A (en) * 1997-07-09 2000-08-22 Sony Corporation Multichannel digital audio decoding method and apparatus
US6111958A (en) * 1997-03-21 2000-08-29 Euphonics, Incorporated Audio spatial enhancement apparatus and methods
US6131084A (en) * 1997-03-14 2000-10-10 Digital Voice Systems, Inc. Dual subframe quantization of spectral magnitudes
US6205430B1 (en) * 1996-10-24 2001-03-20 Stmicroelectronics Asia Pacific Pte Limited Audio decoder with an adaptive frequency domain downmixer
US6236731B1 (en) * 1997-04-16 2001-05-22 Dspfactory Ltd. Filterbank structure and method for filtering and separating an information signal into different bands, particularly for audio signal in hearing aids
US6282631B1 (en) * 1998-12-23 2001-08-28 National Semiconductor Corporation Programmable RISC-DSP architecture
US20010031054A1 (en) * 1999-12-07 2001-10-18 Anthony Grimani Automatic life audio signal derivation system
US20010031055A1 (en) * 1999-12-24 2001-10-18 Aarts Ronaldus Maria Multichannel audio signal processing device
US6323018B1 (en) * 1996-06-11 2001-11-27 Protein Technologies Int'l Lnc. Recovery of isoflavones from soy molasses
US6356870B1 (en) * 1996-10-31 2002-03-12 Stmicroelectronics Asia Pacific Pte Limited Method and apparatus for decoding multi-channel audio data
US20020055796A1 (en) * 2000-08-29 2002-05-09 Takashi Katayama Signal processing apparatus, signal processing method, program and recording medium
US6408327B1 (en) * 1998-12-22 2002-06-18 Nortel Networks Limited Synthetic stereo conferencing over LAN/WAN
US6424939B1 (en) * 1997-07-14 2002-07-23 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Method for coding an audio signal
US6434191B1 (en) * 1999-09-30 2002-08-13 Telcordia Technologies, Inc. Adaptive layered coding for voice over wireless IP applications
US20030035553A1 (en) * 2001-08-10 2003-02-20 Frank Baumgarte Backwards-compatible perceptual coding of spatial cues
US6539367B1 (en) * 2000-05-26 2003-03-25 Agere Systems Inc. Methods and apparatus for decoding of general codes on probability dependency graphs
US20030081115A1 (en) * 1996-02-08 2003-05-01 James E. Curry Spatial sound conference system and apparatus
US20030161479A1 (en) * 2001-05-30 2003-08-28 Sony Corporation Audio post processing in DVD, DTV and other audio visual products
US6614936B1 (en) * 1999-12-03 2003-09-02 Microsoft Corporation System and method for robust video coding using progressive fine-granularity scalable (PFGS) coding
US20030187663A1 (en) * 2002-03-28 2003-10-02 Truman Michael Mead Broadband frequency translation for high frequency regeneration
US20030219130A1 (en) * 2002-05-24 2003-11-27 Frank Baumgarte Coherence-based audio coding and synthesis
US6658117B2 (en) * 1998-11-12 2003-12-02 Yamaha Corporation Sound field effect control apparatus and method
US20030236583A1 (en) * 2002-06-24 2003-12-25 Frank Baumgarte Hybrid multi-channel/cue coding/decoding of audio signals
US20040091118A1 (en) * 1996-07-19 2004-05-13 Harman International Industries, Incorporated 5-2-5 Matrix encoder and decoder system
US6763115B1 (en) * 1998-07-30 2004-07-13 Openheart Ltd. Processing method for localization of acoustic image for audio signals for the left and right ears
US6782366B1 (en) * 2000-05-15 2004-08-24 Lsi Logic Corporation Method for independent dynamic range control
US6845163B1 (en) * 1999-12-21 2005-01-18 At&T Corp Microphone array for preserving soundfield perceptual cues
US20050053242A1 (en) * 2001-07-10 2005-03-10 Fredrik Henn Efficient and scalable parametric stereo coding for low bitrate applications
US20050069143A1 (en) * 2003-09-30 2005-03-31 Budnikov Dmitry N. Filtering for spatial audio rendering
US6885992B2 (en) * 2001-01-26 2005-04-26 Cirrus Logic, Inc. Efficient PCM buffer
US20050157883A1 (en) * 2004-01-20 2005-07-21 Jurgen Herre Apparatus and method for constructing a multi-channel output signal or for generating a downmix signal
US6934676B2 (en) * 2001-05-11 2005-08-23 Nokia Mobile Phones Ltd. Method and system for inter-channel signal redundancy removal in perceptual audio coding
US6940540B2 (en) * 2002-06-27 2005-09-06 Microsoft Corporation Speaker detection and tracking using audiovisual data
US20050226426A1 (en) * 2002-04-22 2005-10-13 Koninklijke Philips Electronics N.V. Parametric multi-channel audio representation
US6973184B1 (en) * 2000-07-11 2005-12-06 Cisco Technology, Inc. System and method for stereo conferencing over low-bandwidth links
US6987856B1 (en) * 1996-06-19 2006-01-17 Board Of Trustees Of The University Of Illinois Binaural signal processing techniques
US20060206323A1 (en) * 2002-07-12 2006-09-14 Koninklijke Philips Electronics N.V. Audio coding
US7116787B2 (en) * 2001-05-04 2006-10-03 Agere Systems Inc. Perceptual synthesis of auditory scenes
US7181019B2 (en) * 2003-02-11 2007-02-20 Koninklijke Philips Electronics N. V. Audio coding
US20070094012A1 (en) * 2005-10-24 2007-04-26 Pang Hee S Removing time delays in signal paths
US7516066B2 (en) * 2002-07-16 2009-04-07 Koninklijke Philips Electronics N.V. Audio coding

Family Cites Families (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP3227942B2 (en) 1993-10-26 2001-11-12 ソニー株式会社 High efficiency coding device
US6539357B1 (en) * 1999-04-29 2003-03-25 Agere Systems Inc. Technique for parametric coding of a signal containing information
US6823018B1 (en) * 1999-07-28 2004-11-23 At&T Corp. Multiple description coding communication system
US6850496B1 (en) * 2000-06-09 2005-02-01 Cisco Technology, Inc. Virtual conference room for voice conferencing
BR0305555A (en) 2002-07-16 2004-09-28 Koninkl Philips Electronics Nv Method and encoder for encoding an audio signal, apparatus for providing an audio signal, encoded audio signal, storage medium, and method and decoder for decoding an encoded audio signal
JP2006521577A (en) 2003-03-24 2006-09-21 コーニンクレッカ フィリップス エレクトロニクス エヌ ヴィ Encoding main and sub-signals representing multi-channel signals

Patent Citations (69)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4236039A (en) * 1976-07-19 1980-11-25 National Research Development Corporation Signal matrixing for directional reproduction of sound
US4815132A (en) * 1985-08-30 1989-03-21 Kabushiki Kaisha Toshiba Stereophonic voice signal transmission system
US4972484A (en) * 1986-11-21 1990-11-20 Bayerische Rundfunkwerbung Gmbh Method of transmitting or storing masked sub-band coded audio signals
US5579430A (en) * 1989-04-17 1996-11-26 Fraunhofer Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Digital encoding process
US6021386A (en) * 1991-01-08 2000-02-01 Dolby Laboratories Licensing Corporation Coding method and apparatus for multiple channels of audio information representing three-dimensional sound fields
US5583962A (en) * 1991-01-08 1996-12-10 Dolby Laboratories Licensing Corporation Encoder/decoder for multidimensional sound fields
US5682461A (en) * 1992-03-24 1997-10-28 Institut Fuer Rundfunktechnik Gmbh Method of transmitting or storing digitalized, multi-channel audio signals
US5703999A (en) * 1992-05-25 1997-12-30 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Process for reducing data in the transmission and/or storage of digital signals from several interdependent channels
US5706309A (en) * 1992-11-02 1998-01-06 Fraunhofer Geselleschaft Zur Forderung Der Angewandten Forschung E.V. Process for transmitting and/or storing digital signals of multiple channels
US5371799A (en) * 1993-06-01 1994-12-06 Qsound Labs, Inc. Stereo headphone sound source localization system
US5463424A (en) * 1993-08-03 1995-10-31 Dolby Laboratories Licensing Corporation Multi-channel transmitter/receiver system providing matrix-decoding compatible signals
US5701346A (en) * 1994-03-18 1997-12-23 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Method of coding a plurality of audio signals
US5677994A (en) * 1994-04-15 1997-10-14 Sony Corporation High-efficiency encoding method and apparatus and high-efficiency decoding method and apparatus
US5956674A (en) * 1995-12-01 1999-09-21 Digital Theater Systems, Inc. Multi-channel predictive subband audio coder using psychoacoustic adaptive bit allocation in frequency, time and over the multiple channels
US5771295A (en) * 1995-12-26 1998-06-23 Rocktron Corporation 5-2-5 matrix system
US20030081115A1 (en) * 1996-02-08 2003-05-01 James E. Curry Spatial sound conference system and apparatus
US5878080A (en) * 1996-02-08 1999-03-02 U.S. Philips Corporation N-channel transmission, compatible with 2-channel transmission and 1-channel transmission
US5825776A (en) * 1996-02-27 1998-10-20 Ericsson Inc. Circuitry and method for transmitting voice and data signals upon a wireless communication channel
US5889843A (en) * 1996-03-04 1999-03-30 Interval Research Corporation Methods and systems for creating a spatial auditory environment in an audio conference system
US5812971A (en) * 1996-03-22 1998-09-22 Lucent Technologies Inc. Enhanced joint stereo coding method using temporal envelope shaping
US5930733A (en) * 1996-04-15 1999-07-27 Samsung Electronics Co., Ltd. Stereophonic image enhancement devices and methods using lookup tables
US6323018B1 (en) * 1996-06-11 2001-11-27 Protein Technologies Int'l Lnc. Recovery of isoflavones from soy molasses
US6987856B1 (en) * 1996-06-19 2006-01-17 Board Of Trustees Of The University Of Illinois Binaural signal processing techniques
US20040091118A1 (en) * 1996-07-19 2004-05-13 Harman International Industries, Incorporated 5-2-5 Matrix encoder and decoder system
US6205430B1 (en) * 1996-10-24 2001-03-20 Stmicroelectronics Asia Pacific Pte Limited Audio decoder with an adaptive frequency domain downmixer
US6356870B1 (en) * 1996-10-31 2002-03-12 Stmicroelectronics Asia Pacific Pte Limited Method and apparatus for decoding multi-channel audio data
US5912976A (en) * 1996-11-07 1999-06-15 Srs Labs, Inc. Multi-channel audio enhancement system for use in recording and playback and methods for providing same
US6131084A (en) * 1997-03-14 2000-10-10 Digital Voice Systems, Inc. Dual subframe quantization of spectral magnitudes
US6111958A (en) * 1997-03-21 2000-08-29 Euphonics, Incorporated Audio spatial enhancement apparatus and methods
US6236731B1 (en) * 1997-04-16 2001-05-22 Dspfactory Ltd. Filterbank structure and method for filtering and separating an information signal into different bands, particularly for audio signal in hearing aids
US5860060A (en) * 1997-05-02 1999-01-12 Texas Instruments Incorporated Method for left/right channel self-alignment
US5946352A (en) * 1997-05-02 1999-08-31 Texas Instruments Incorporated Method and apparatus for downmixing decoded data streams in the frequency domain prior to conversion to the time domain
US5850496A (en) * 1997-07-02 1998-12-15 Stryker Corporation Endoscope with integrated, self-regulating light source
US6108584A (en) * 1997-07-09 2000-08-22 Sony Corporation Multichannel digital audio decoding method and apparatus
US6424939B1 (en) * 1997-07-14 2002-07-23 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Method for coding an audio signal
US5890125A (en) * 1997-07-16 1999-03-30 Dolby Laboratories Licensing Corporation Method and apparatus for encoding and decoding multiple audio channels at low bit rates using adaptive selection of encoding method
US6021389A (en) * 1998-03-20 2000-02-01 Scientific Learning Corp. Method and apparatus that exaggerates differences between sounds to train listener to recognize and identify similar sounds
US6016473A (en) * 1998-04-07 2000-01-18 Dolby; Ray M. Low bit-rate spatial coding method and system
US6763115B1 (en) * 1998-07-30 2004-07-13 Openheart Ltd. Processing method for localization of acoustic image for audio signals for the left and right ears
US6658117B2 (en) * 1998-11-12 2003-12-02 Yamaha Corporation Sound field effect control apparatus and method
US6408327B1 (en) * 1998-12-22 2002-06-18 Nortel Networks Limited Synthetic stereo conferencing over LAN/WAN
US6282631B1 (en) * 1998-12-23 2001-08-28 National Semiconductor Corporation Programmable RISC-DSP architecture
US6434191B1 (en) * 1999-09-30 2002-08-13 Telcordia Technologies, Inc. Adaptive layered coding for voice over wireless IP applications
US6614936B1 (en) * 1999-12-03 2003-09-02 Microsoft Corporation System and method for robust video coding using progressive fine-granularity scalable (PFGS) coding
US20010031054A1 (en) * 1999-12-07 2001-10-18 Anthony Grimani Automatic life audio signal derivation system
US6845163B1 (en) * 1999-12-21 2005-01-18 At&T Corp Microphone array for preserving soundfield perceptual cues
US20010031055A1 (en) * 1999-12-24 2001-10-18 Aarts Ronaldus Maria Multichannel audio signal processing device
US6782366B1 (en) * 2000-05-15 2004-08-24 Lsi Logic Corporation Method for independent dynamic range control
US6539367B1 (en) * 2000-05-26 2003-03-25 Agere Systems Inc. Methods and apparatus for decoding of general codes on probability dependency graphs
US6973184B1 (en) * 2000-07-11 2005-12-06 Cisco Technology, Inc. System and method for stereo conferencing over low-bandwidth links
US20020055796A1 (en) * 2000-08-29 2002-05-09 Takashi Katayama Signal processing apparatus, signal processing method, program and recording medium
US6885992B2 (en) * 2001-01-26 2005-04-26 Cirrus Logic, Inc. Efficient PCM buffer
US7116787B2 (en) * 2001-05-04 2006-10-03 Agere Systems Inc. Perceptual synthesis of auditory scenes
US6934676B2 (en) * 2001-05-11 2005-08-23 Nokia Mobile Phones Ltd. Method and system for inter-channel signal redundancy removal in perceptual audio coding
US20030161479A1 (en) * 2001-05-30 2003-08-28 Sony Corporation Audio post processing in DVD, DTV and other audio visual products
US7382886B2 (en) * 2001-07-10 2008-06-03 Coding Technologies Ab Efficient and scalable parametric stereo coding for low bitrate audio coding applications
US20050053242A1 (en) * 2001-07-10 2005-03-10 Fredrik Henn Efficient and scalable parametric stereo coding for low bitrate applications
US20030035553A1 (en) * 2001-08-10 2003-02-20 Frank Baumgarte Backwards-compatible perceptual coding of spatial cues
US20030187663A1 (en) * 2002-03-28 2003-10-02 Truman Michael Mead Broadband frequency translation for high frequency regeneration
US20050226426A1 (en) * 2002-04-22 2005-10-13 Koninklijke Philips Electronics N.V. Parametric multi-channel audio representation
US20030219130A1 (en) * 2002-05-24 2003-11-27 Frank Baumgarte Coherence-based audio coding and synthesis
US20030236583A1 (en) * 2002-06-24 2003-12-25 Frank Baumgarte Hybrid multi-channel/cue coding/decoding of audio signals
US6940540B2 (en) * 2002-06-27 2005-09-06 Microsoft Corporation Speaker detection and tracking using audiovisual data
US20060206323A1 (en) * 2002-07-12 2006-09-14 Koninklijke Philips Electronics N.V. Audio coding
US7516066B2 (en) * 2002-07-16 2009-04-07 Koninklijke Philips Electronics N.V. Audio coding
US7181019B2 (en) * 2003-02-11 2007-02-20 Koninklijke Philips Electronics N. V. Audio coding
US20050069143A1 (en) * 2003-09-30 2005-03-31 Budnikov Dmitry N. Filtering for spatial audio rendering
US20050157883A1 (en) * 2004-01-20 2005-07-21 Jurgen Herre Apparatus and method for constructing a multi-channel output signal or for generating a downmix signal
US20070094012A1 (en) * 2005-10-24 2007-04-26 Pang Hee S Removing time delays in signal paths

Cited By (12)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20060235679A1 (en) * 2005-04-13 2006-10-19 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Adaptive grouping of parameters for enhanced coding efficiency
US20060235683A1 (en) * 2005-04-13 2006-10-19 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Lossless encoding of information with guaranteed maximum bitrate
US20110060598A1 (en) * 2005-04-13 2011-03-10 Fraunhofer-Gesellschaft Zur Forderung Der Angewandten Forschung E.V. Adaptive grouping of parameters for enhanced coding efficiency
US7991610B2 (en) 2005-04-13 2011-08-02 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Adaptive grouping of parameters for enhanced coding efficiency
US9043200B2 (en) 2005-04-13 2015-05-26 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Adaptive grouping of parameters for enhanced coding efficiency
US20100169103A1 (en) * 2007-03-21 2010-07-01 Ville Pulkki Method and apparatus for enhancement of audio reconstruction
US20100166191A1 (en) * 2007-03-21 2010-07-01 Juergen Herre Method and Apparatus for Conversion Between Multi-Channel Audio Formats
US8908873B2 (en) 2007-03-21 2014-12-09 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Method and apparatus for conversion between multi-channel audio formats
US9015051B2 (en) * 2007-03-21 2015-04-21 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Reconstruction of audio channels with direction parameters indicating direction of origin
US20090041271A1 (en) * 2007-06-29 2009-02-12 France Telecom Positioning of speakers in a 3D audio conference
US8280083B2 (en) * 2007-06-29 2012-10-02 France Telecom Positioning of speakers in a 3D audio conference
US20180078815A1 (en) * 2016-09-18 2018-03-22 Foxconn Interconnect Technology Limited Treadmill and monitoring system

Also Published As

Publication number Publication date
US7116787B2 (en) 2006-10-03
US20030026441A1 (en) 2003-02-06

Similar Documents

Publication Publication Date Title
US7116787B2 (en) Perceptual synthesis of auditory scenes
US20030035553A1 (en) Backwards-compatible perceptual coding of spatial cues
Faller et al. Efficient representation of spatial audio using perceptual parametrization
Faller et al. Binaural cue coding-Part II: Schemes and applications
Pulkki et al. Spatial impulse response rendering II: Reproduction of diffuse sound and listening tests
KR101184568B1 (en) Late reverberation-base synthesis of auditory scenes
Majdak et al. Multiple exponential sweep method for fast measurement of head-related transfer functions
EP2329661B1 (en) Binaural filters for monophonic compatibility and loudspeaker compatibility
US8751029B2 (en) System for extraction of reverberant content of an audio signal
US7006636B2 (en) Coherence-based audio coding and synthesis
US8243969B2 (en) Method of and device for generating and processing parameters representing HRTFs
Gardner Transaural 3-D audio
US11750995B2 (en) Method and apparatus for processing a stereo signal
KR20080078882A (en) Decoding of binaural audio signals
CN105684465B (en) Sound spatialization with interior Effect
Begault Virtual Acoustic Displays for Teleconferencing: Intelligibility Advantage for'Telephone-Grade'Audio
Jinzai et al. Microphone position realignment by extrapolation of virtual microphone
Pulkki et al. Efficient spatial sound synthesis for virtual worlds
Salmon et al. The influence of vision on perceived differences between sound spaces
Schneiderwind et al. Effects of Modified Late Reverberation on Audio-Visual Plausibility and Externalization in AR
Evans et al. Perceived performance of loudspeaker-spatialized speech for teleconferencing
Hupke et al. Perceptual evaluation of an augmented audience service under realistic live conditions
Shabtai et al. Spherical array processing with binaural sound reproduction for improved speech intelligibility
Chen et al. Highly realistic audio spatialization for multiparty conferencing using headphones
JP2023503140A (en) Converting binaural signals to stereo audio signals

Legal Events

Date Code Title Description
AS Assignment

Owner name: AGERE SYSTEMS INC., PENNSYLVANIA

Free format text: ASSIGNMENT OF ASSIGNORS INTEREST;ASSIGNOR:FALLER, CHRISTOF;REEL/FRAME:018209/0898

Effective date: 20010502

STCB Information on status: application discontinuation

Free format text: ABANDONED -- FAILURE TO RESPOND TO AN OFFICE ACTION