Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S5/00—Pseudo-stereo systems, e.g. in which additional channel signals are derived from monophonic signals by means of phase shifting, time delay or reverberation 

Definitions

• the invention relates to audio signals and to devices or methods for their generation
• audio signals which are emitted via two or more loudspeakers give the listener a spatial impression, provided that they have different amplitudes, frequencies, transit time or phase differences or are correspondingly reverberated.
• Such decorrelated signals can be placed on the one hand by differently
• pseudo-stereophonic techniques that produce such a suitable decorrelation - starting from a mono signal.
• EP2124486 and EP1850639 describe a method for methodically evaluating the angle of incidence for the sound event to be imaged, which is included by the microphone main axis and the sounding axis for the sound source
• Amplitude correction as well as the time correction are selected independently of the recording situation.
• R (M - S) * l / V ⁇ 2), which is not the same as for intensity-stereophonic signals, ie for stereo signals that differ only in their levels, but not in transit time or phase differences or in different frequency spectra one restricting the imaging width or shifting the imaging direction of the acquired stereo signals as intended, but rather increasing or decreasing the image quality
• Audio signal are optimal, but is not trivial.
• the adjustment of the parameters also often has an influence on the degree of correlation between the left and the right channel.
• pseudostereophonic signals It is also an aim to provide a method and a device for optimally and automatically determining the parameters ( ⁇ , ⁇ , p or f (or n), ⁇ , ⁇ ) in this extraction.
• this method and this device are to be selected from several decorrelated, in particular pseudostereophonic, signal variants those whose decorrelation proves to be particularly favorable.
• the selection criteria should themselves be able to be influenced in the most efficient and compact way, in order to obtain signals of different types
• an apparatus and a method for obtaining pseudo-stereophonic output signals x (t) and y (t) is thus based on a
• the extraction is iteratively optimized until a part of ⁇ x (t), y (t)> within the predetermined
• the desired domain of definition is preferably determined by a single numerical parameter a, where preferably 0 ⁇ a ⁇ 1.
• This parameter and thus the domain of definition, can be determined, for example, by the inequalities
• Definition range starting from the unit circle of the complex number plane or the imaginary axis (if the maximum level of the output signal x (t), y (t) was normalized on the unit circle), using the parameter a, 0 ⁇ a ⁇ 1, set arbitrarily.
• Unit circle of the complex number plane is selected, and another new definition area is defined.
• the degree of correlation of the output signals (x (t) and y (t)) is normalized.
• the level of the maximum of the resulting left and right channels is normalized. In this way, certain
• Parameters can be iteratively optimized to achieve the desired domain of definition without affecting the degree of correlation or the level of the maximum of the resulting left and right channels.
• x (t) represents the function value of the left output channel at time t
• f * [x (t)] [x (t) / V 2] * (-1 + i )
• g * [y (t)] [y (t) / V 2] * (l + i) in which the recovery is iteratively optimized until the following criteria are met:
• Standardization can preferably be targeted by the
• first the modulation for the maximum of the left signal L and of the right signal R is uniformly set to, for example, 0 dB by means of a first logic element.
• This method proves to be particularly favorable, as with a single parameter, namely a, in particular the different nature of
• Output signals of a device or a method according to EP2124486 or EP1850639 is optimally taken into account.
• the parameter may preferably be dependent on the type of audio signal, for example to manually or automatically edit speech or music differently. In the case of language, this is determined by a
• each optimum can be derived from the unit circle or the imaginary axis
• R * and ⁇ are directly related to the loudness of the output signal to be obtained (ie those parameters according to which the listener also judges the validity of a stereophonic mapping).
• the degree of correlation r, the parameter a defining the desired respective definition range and the limit value R * and its deviation ⁇ can be optimized for the respective nature of the input signals optimal systems for the respective field of application (for example speech or music reproduction).
• the invention resides in cascading a plurality of partially adjustable parameters (eg, logic elements) in a stereo converter (for example, according to EP2124486 or EP1850639), with feedback regarding such devices or methods being that there is an optimized change the parameter ⁇ or ⁇ or p or f (or n) or CC or ß is carried out until all
• FIG. 1 shows an example of a circuit for two logic elements for normalizing the level and for normalizing the degree of correlation of the output signals of a stereo converter (for example, a
• FIG. 2 shows an example of a circuit which outputs given signals x (t), y (t) by means of
• FIG. 3 shows a first example of a
• FIG. 3a shows a second example of a circuit for the selection of a new definition range, which is advantageous to the person skilled in the art, by means of the parameter a.
• FIG. 4 shows a first example of a circuit for a third logic element, which has the circuits shown in FIG. 1 generated, as shown in FIG. 2 on the complex
• FIG. FIG. 4a shows a second example of a circuit for a third logic element that is advantageous for the person skilled in the art, which has the circuits shown in FIG. 1 generated, as shown in FIG. 2 mapped on the complex number plane
• FIG. 5 shows an example of a circuit for a fourth logic element, which concludes the relief of the function f * [x (t)] + g * [y (t)] in the sense one
• FIG. 5a shows a second example of a circuit for a fourth logic element that is advantageous for the person skilled in the art, which finally considers the relief of the function f * [x (t)] + g * [y (t)] in order to maximize its functional values the user also sets the limit value R * (or the inequality (8a) defined by the inequality (8a))
• FIG. 6a shows an input circuit for an already existing stereo signal prior to transfer to a circuit according to FIG. 6b for the determination of
• FIG. 6 b shows a circuit for determining the localization of the signal whose inputs are connected to the outputs of FIG. 5 or FIG. 5a and the outputs of FIG. 6a are connected.
• FIG. FIG. 7 shows another example of a stereophonic or standardization circuit
• pseudo-stereophonic signals which, if shown in FIG. 6b is activated, as soon as the parameter z is present as an input signal.
• Amplification factor ⁇ corresponds to the final value of the amplification factor ⁇ of FIG. 1 upon delivery of the
• FIG. 8 shows an example of a circuit which outputs given signals x (t), y (t) by means of the
• FIG. Fig. 9 shows an example of a circuit for adjusting the image width of an audio signal.
• Damping ⁇ and p optimized parameters ⁇ , ⁇ , f (or the simplifying parameter n), CC, ß are determined to convert a mono signal into corresponding pseudo-stereophonic signals, which is an optimal
• FIG. 1 shows the circuit principle for the first two logic elements described above
• Stereo converter with an MS matrix 110 for example, a stereo converter according to EP2124486 or EP1850639
• the input signal M and S before passing through one of the MS matrix upstream amplifier
• a circuit according to FIG. 7 can be supplied, which optionally and ideally of FIG. 6b is followed, and is activated as soon as the from FIG. 6b
• the first logic element 120 for normalizing the level is coupled to two identical amplifiers with the gain factor p * and provides for a maximization of the left channel L and right channel R maximized to 0 dB. From the assembly 110 (for example, an MS matrix according to EP2124486 or EP1850639)
• Downstream of a logic element 120 reaches, via the feedbacks 121 and 122 and variation or correction of the gain factor p * of the amplifier 118 and 119 causes a modulation of the maximum value of L and R to 0 dB.
• r can be set by user in the range -1 ⁇ r ⁇ 1 and ideally moves in the range of 0.2 ⁇ r ⁇ 0.7. Any deviation from r leads over the
• the resulting signals L and R pass through the amplifiers 118 and 119 as well as the
• Logic element 120 which in turn via the feedbacks 121 and 122 a renewed modulation of the Maximum value of L and R to 0 dB, and are then supplied again to the logic element 125.
• FIG. 2 clarifies the circuit principle, which the input signals x (t), y (t) on the
• Amplification factor -1 passes through. This results in the transfer functions
• Element 232 determines the argument of f * [x (t)] + g * [y (t)].
• FIG. 3 illustrates the circuit principle for the selection of the definition range, whereby a stepless regulation via the parameter 0 ⁇ a ⁇ 1, starting from the unit circle of the complex number plane or the imaginary axis, is made possible.
• the user can determine the domain of definition a on the complex number plane.
• the cosine (333) or sine (334) of the just determined argument of f * [x (t)] + g * [y (t)] is calculated.
• the resulting 333 signal is then an amplifier 335th
• FIG. FIG. 4 shows the circuit principle for the third logic element, which has the structure shown in FIG. 1 generated, as shown in FIG. 2 on the complex number level
• Transfer functions f * [x (t)] + g * [y (t)] and the signals resulting from 334 and 335 are here supplied to a further logic element 436, which checks whether the criteria (4) and (5) are fulfilled. thus, whether the values of the sum of the transfer functions f * [x (t)] + g * [y (t)] within the user defined by means of a
• Feedback 437 new optimized values ⁇ or f (or n) or CC or ß determined, and will run through the entire system so far again until the values of the sum of the transfer functions f * [x (t)] + g * [y (t)] within the user's name using a
• Deviation ⁇ can freely choose for this maximization. Overall, the condition needs
• FIG. 3a, 4a and 5a An alternative circuit principle which is advantageous to the person skilled in the art is shown in FIG. 3a, 4a and 5a, which replace the corresponding figures 3, 4 and 5 in a preferred variant:
• FIG. 3a again allows, via the parameter a, 0 ⁇ a ⁇ 1, the selection of a new domain of definition, where a is used to provide stepless regulation based on the unit circle of the complex number plane or the imaginary axis.
• the squared real part (333a) or squared imaginary part (334a) of f * [x (t)] + g * [y (t)] is calculated.
• the signal resulting from 333a is subsequently supplied to an amplifier 335a and amplified by the user-selectable amplification factor l / a 2 .
• the squared sine of the argument of the sum of the transfer functions f * [x (t] + g * [y (t)] is calculated.
• FIG. 4a at the output of Figure 3a
• the imaginary part of the sum of the transfer functions f * [x (t)] + g * [y (t)] and the signals resulting from 334a and 335a are here supplied to a further logic element 436a, which checks whether the above criterion is fulfilled, thus whether the values of the sum of the transfer functions f * [x (t)] + g * [y (t)] lie within the new value range defined by the user by means of a. If this is not true, are over a
• Feedback 437a new optimized values ⁇ or f (or n) or CC or ß determined, and will run through the entire system so far again until the values of the sum of the transfer functions f * [x (t)] + g * [y (t)] within the user's name using a
• Deviation ⁇ can freely choose for this maximization. Overall, the condition must be new
• Pseudo-stereo converter for example according to one of the embodiments in EP2124486 or EP1850639 (vice versa here assuming the case
• proportional reductions ⁇ and p) iteratively determines new parameters ⁇ or f (or n) or CC or ⁇ until x (t) and y (t) satisfy the above-mentioned conditions (4), (5) and (8 ) or (4a) and (8a) meet.
• Amplification factor a) and loudness (determined by the selectable limit R * or the selectable deviation ⁇ ) according to the user and set the Output signals L and R of the described arrangement.
• Directional characteristic to reflect for example, there is a mirror image with respect to the main axis. This can be done manually by swapping the left and right channels.
• the correct imaging direction can be determined by means of
• Phantom sound sources formed pseudo stereophonic method also shown, for example, according to FIG. 6b (FIG. 5 or FIG.
• f * (l (t ⁇ )) + g * CrCt 1 )) may be equal to zero in at least one case) which are already shown in FIG. 2 determined transfer function f * (x Ct 1 )) + g * (y (ti)) with the
• Number are the left channel x (t) and the right channel y (t) of the approximately from an arrangement according to FIG. 1-5 or FIG. 1, 2, 3a to 5a resulting stereo signal reversed.
• an originally stereophonic signal is to be recoded into a mono signal plus the function f (or its simplifying parameter n) describing the directional characteristic and the parameter ⁇ , ⁇ , ⁇ , ⁇ or p (for example for the purpose of data compression) (example for an output 640a) , which can be extended by the parameter z, see below)
• Image width of the obtained stereo signal on the basis of the targeted variation of the degree of correlation r of the resulting stereo signal or the attenuation ⁇ or p (for the formation of the resulting
• Opening angle ⁇ can be maintained, and it makes sense only a final amplitude correction about according to the logic element 120 of Figure 1 is necessary, if this limitation or extension of the image width is done manually.
• FIG. 7 shows a further example of a circuit for standardization stereophonic or pseudostereophonous
• Amplification factor ⁇ corresponds to the final value of the amplification factor ⁇ of FIG. 1 upon delivery of the
• FIGS. 7 to 9 can also be used autonomously in other circuits or algorithms.
• a logic element 110a (the
• this MS matrix activated) the left and the right channel swapped as long as the parameter z is equal to 1, otherwise there is no such interchanging.
• the resulting output signals L and R of the MS matrix 110 are now uniformly amplified by the factor p * (amplifiers 118, 119) so that the maximum of both signals has a level of exactly 0 dB (normalization on the unit circle of the complex
• Downstream of a logic element 120 reaches, via the feedbacks 121 and 122 and variation or correction of the gain factor p * of the amplifier 118 and 119 causes a modulation of the maximum value of L and R to 0 dB.
• Image width of the stereo signal to be obtained suitably selected threshold value S * or a suitably chosen deviation ⁇ , both defined by the
• Stereo signal Stereo signal
• the previous steps just described as shown in FIG. 7 to 9, as long as pass until the above condition (9) is satisfied.
• the output signals for the logic element 640 are now to an arrangement approximately according to the
• An already existing stereo signal can with respect to r or a or R * or ⁇ or the
• Imaging direction (or the parameters S * or ⁇ or U * or K described below) and then also newly evaluated as a mono signal on the basis of the parameters ⁇ , f (or n) with respect to a device or a method according to EP2124486 or EP1850639, CC, ß, ⁇ and p are coded.
• Imaging direction (expressed for example by the parameter z, which can take the value 0 or 1), such a decoder is reduced to an arrangement according to EP2124486 or EP1850639.
• Satellite broadcasting equipment professional audio equipment, television, film and radio and electronic consumer goods.
• the invention is also of particular importance in the context of obtaining stable FM stereo signals under adverse reception conditions (such as in automobiles).
• a stable stereophonic sound can be used with the pure help of the main channel signal (L + R) as the input signal
• the complete or incomplete sub-channel signal (L-R) representing the result of the subtraction of the left-right channel of the original stereo signal can be used to form a usable S signal or parameters f (or n), which describe the directional characteristic of the signal to be stereophonized, the manual or metrological to
• determining angle ⁇ , the main axis and sound source include, the fictitious left opening angle CC, the fictitious right opening angle ß, the attenuations ⁇ or p for the formation of the resulting
• Amplification factor p * of FIG. 1 for the normalization of the resulting from the MS matrix or from any other arrangement according to the invention left and right channel on the unit circle (1 corresponds to, for example, the mediated by p * maximum level of 0 dB, where x (t) that from this normalization
• Amplification factor a for the definition of the permissible value range for the sum of the transfer functions of the resulting output signals (for example, the complex transfer functions
• reproduced sound sources for example by determining the associated quadrants for the function values of the above transfer functions (2) and (3) for the original stereo signal (the approximately by

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• Engineering & Computer Science (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• Stereophonic System (AREA)
• Stereo-Broadcasting Methods (AREA)

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• 2010
  • 2010-04-29 KR KR1020127004584A patent/KR20120062727A/ko not_active Application Discontinuation
  • 2010-04-29 CN CN201080032967.8A patent/CN102577440B/zh not_active Expired - Fee Related
  • 2010-04-29 WO PCT/EP2010/055876 patent/WO2011009649A1/de active Application Filing
  • 2010-04-29 WO PCT/EP2010/055877 patent/WO2011009650A1/de active Application Filing
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  • 2010-04-29 EP EP10716543A patent/EP2457390A1/de not_active Withdrawn
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• 2012
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• 2016
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