EP1212923B1 - Verfahren und anordnung zur erzeugung eines zweiten audiosignales von einem ersten audiosignal - Google Patents
Verfahren und anordnung zur erzeugung eines zweiten audiosignales von einem ersten audiosignal Download PDFInfo
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- EP1212923B1 EP1212923B1 EP00956732A EP00956732A EP1212923B1 EP 1212923 B1 EP1212923 B1 EP 1212923B1 EP 00956732 A EP00956732 A EP 00956732A EP 00956732 A EP00956732 A EP 00956732A EP 1212923 B1 EP1212923 B1 EP 1212923B1
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- audio signal
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- 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
- This invention relates to the reproduction of 3D-sound from two-speaker stereo systems, headphones, and multi-speaker audio systems. It relates particularly, though not exclusively, to a method for the creation of one or more virtual sound sources simultaneously from a single, common sound signal which, nevertheless, can be discerned separately from one another by a listener in use.
- the Haas (or Precedence) Effect [M B Gardner, J. Acoust. Soc. Am., 43, (6), pp.1243-1248 (1968)] is the phenomenon that the brain, when presented with several similar pieces of audio information at slightly differing times to process, uses only the first information to arrive from which to compute directional information. The brain then attributes the subsequent, similar information packets with the same directional information. The key to this is that the brain recognises signals which are related to one another (i.e. correlated), and processes them in a particular way.
- This effect is also known under the names "law of the first wavefront”, “auditory suppression effect”, “first-arrival effect” and “threshold of extinction”, and it is used for the basis of sound reinforcement used in Public Address systems.
- the relevance of the precedence effect here is that it can contribute to "spatial fusion", under particular circumstances, during the synthesis of virtual 3D-sound images.
- the "Spatial Fusion” effect is not widely appreciated or known, and it is common both to loudspeaker and headphone listening. It occurs when synthesising several virtual sound sources from a single, common source, such that there is a significant common-mode content present in the left and right channels (and rear-left and rear-right channels in a four-speaker 3D-audio system).
- Example 1 Primary signal + derived reverberation.
- any signals which have been derived directly from another sound source (such as a [secondary] reverberation signal created from a primary source), are perceived to "combine” spatially with the primary signal if they are presented to the listener within a period of about 15 ms. Beyond this time period, they begin to be discernible as separate entities, in the form of an echo.
- the effect of such spatial combination is to inhibit the secondary image and create an imprecise and vaguely positioned spatial image at the location of the primary sound source.
- the HRTF processing decor relates the individual signals sufficiently such that the listener is able to distinguish between them, and hear them as individual sources, rather than "fuse” them spatially into apparently a single sound.
- the HRTF processing decor decorrelates the individual signals sufficiently such that the listener is able to distinguish between them, and hear them as individual sources, rather than "fuse” them spatially into apparently a single sound.
- the symmetry enhances any correlation between the individual sound sources, and the result is that the perceived sounds can become spatially "fused” together into one. For example, if it is required to "virtualise stereo" for headphone listening (i.e.
- a line or area sound-emitter can be considered to be the sum of many individual elemental sound-sources which all possess differing amplitude and phase characteristics.
- Dolby Pro-Logic system or Dolby Surround . It is characterised by the encoding of four channels of analogue audio into two channels of analogue audio, such that it can be recorded on to video tapes and DVDs (and also used for TV broadcast), from which the signals are decoded and used to drive four loudspeakers (left, centre, left-surround and right-surround), together with an optional sub-woofer.
- the bandwidth limitations only allow a single rear-channel "surround” signal. If this signal was fed in parallel to both rear loudspeakers, the Precedence Effect would make the surround channel audio all appear to come from the nearest loudspeaker only.
- the surround signal is fed directly to one of the rear speakers, but it is inverted before being sent to the other rear loudspeaker.
- This is a crude way to decorrelate the signals being emitted from both surround speakers, but it assists the listeners to perceive sounds emanating from both loudspeakers, rather then just one, thus creating a more spacious experience.
- 3D Positional Audio processing for computer games.
- the signals In order to synthesise audio material bearing 3D-sound cues for the listener to perceive, the signals must be convolved with one or more appropriate HRTFs, and then delivered to the listener's ears in such a way that his or her own hearing processes do not interfere with the in-built 3D cues. This can be achieved either by listening through headphones, or via loudspeakers in conjunction with a suitable transaural crosstalk-cancellation scheme (as described in co-pending patent application GB9816059.1).
- a suitable transaural crosstalk-cancellation scheme as described in co-pending patent application GB9816059.1
- we recently devised a method for creating line and area virtual sound-sources GB9813290.5).
- US 5,844,993 discloses use of a complementary comb filter pair to create a pair of rear channels from the single "surround" channel, and shows the first notch and peak features occurring at 100 Hz.
- a signal can be decorrelated by means of comb-filtering, as is known in the prior art.
- Figure 1 shows a simple comb filter, in which the source signal, S, is passed through a time-delay element, and an attenuator element, and then combined with the original signal, S.
- the time-delay corresponds to one half a wavelength
- the two combining waves are exactly 180° out of phase, and cancel each other, whereas when the time delay corresponds to one whole wavelength, the waves combine constructively. If the amplitudes of the two waves are the same, then total nulling and doubling, respectively, of the resultant wave occurs.
- the magnitude of the effect can be controlled.
- the time delay is chosen to be 1 ms
- the first cancellation point exists at 500 Hz.
- the first constructive addition frequency points are at 0 Hz, and 1 kHz, where the signals are in phase. If the attenuation factor is set to 0.5, then the destructive and constructive interference effects are restricted to -3 dB and +3 dB respectively. These characteristics are shown in Figure 1 (lower).
- Inversion can be achieved either by (a) changing the summing node to a "differencing" node (for signal subtraction), or (b) inverting the attenuation coefficient (e.g. from +0.5 to -0.5); the end result is the same in both cases.
- the output of such a pair of complementary filters exhibits maximal amplitude decorrelation within the constraints of the attenuation factors, because the peaks of one correspond to the troughs of the other ( Figure 2), and vice versa.
- a source "triplet" were required, then a convenient method for creating such an arrangement is the creation of a pair of maximally decorrelated sources, which are then used in conjunction with the original source itself, thus providing three sources.
- Audible artefacts As can be seen in Figure 1, the property of a comb filter is to create a series of notches and peaks throughout the spectrum, with the frequency of the lowest feature determined by the time-delay of the filter. Our hearing processes are particularly good at noticing notches in the audio spectrum, and we are also good at detecting tones and notches which are repeated at octave intervals (where the frequencies are multiple values of a fundamental value). Consequently, a comb-filtered signal sounds very artificial, tonally.
- Doppler interaction When more than one comb-filtered signals are subjected to Doppler-effect type processing (as happens in computer game audio applications), then the comb artefacts in the audio become exaggerated, apparently by the interaction between the comb features. Even if one uses complementary comb-filters to make the sources, as described above, the Doppler processing can shift the features in the frequency domain such that they "slide” over each other and become noticeable as artefacts. Notches which are caused to "move” in the frequency domain are especially noticeable: a good example of this is the "flangeing" effect used for music-effects processors, and another is the effect which is heard as a steam train arrives, hissing, at the station platform.
- the hiss sound is, approximately, a form of white noise, and it arrives at the listener both directly and also reflected from the platform surface where it combines with the direct path sound.
- the time-delay difference between the two is small when the train is distant, but increases to correspond to a path length of about twice the listener's ear height above ground when the sound source is above the listener's head and close. For example, if the train were about 4 m distant (with an elevated source), and the ear height were 1.8 m, then the delay would be about 4 ms, and so the first (lowest) notch would occur at about 125 Hz.
- the present invention is a means of decorrelating a sound source so as to provide one or more sound sources which can be used for 3D-sound synthesis such that they can be perceived independently of one another.
- the invention is advantageous over the use of simple comb-filtering, in that: (a) there are no significant audible artefacts present; (b) the derived sources can be Doppler processed without flangeing artefacts, and (c) a plurality of sources can be derived from one single source.
- It includes an audio delay-line (1), which is tapped at two (or more) points within a prescribed range, said points changing frequently and randomly.
- the outputs of the tap nodes are multiplied by predetermined gain factors, one of which is negative, and then added to the original signal. The effect of this is to cause the spectral profile of the derived signal to change, continually, with respect to the original (and, similarly, there are continual changes in relative phase).
- the central feature is an audio delay line (1) in the form of a buffer, as shown at the top of Figure 3, to which audio is written via the "audio write” pointer (2).
- the current data byte is read via the "t 0 " pointer (3).
- the "audio write” pointer (and all the data pointers) moves incrementally towards the right after each sample has been written.
- the audio sampling rate will be 44.1 kHz, and hence the corresponding sampling period is about 22.68 ⁇ s.
- the processing blocks Q and R both comprise a crossfader and a fixed gain amplifier (or attenuator), and the Q block also contains an inverter.
- Each crossfader has two audio inputs and a single audio output. One input to each crossfader is connected so as to receive audio data from a read pointer in the "A" range, and the other input is connected so as to receive audio data from a read pointer in the "B" range.
- the crossfader is set to transfer signal to the output from one of the inputs with a gain of unity, and from the other input with a gain of zero.
- a crossfade cycle rate of greater than 0.5 Hz, preferably 5 - 100 cycles per second is chosen, although much higher cycle rates can, in principle, be used.
- the Q and R processing block gain stages have fixed gain (or attenuation). It is convenient, but not essential, that they are set equal to one another, because the decorrelation contributions from both cells would be equally weighted. It is also convenient that the sum of all the gain stages (G P , G Q and G R ) is unity, because this corresponds to a maximum overall gain of unity (0 dB) through the system with respect to the original audio signal written to the buffer.
- the output from the Q and R sections are fed into a summing node, together with the output from the t 0 "read" pointer, which is transferred to the node via a fixed gain stage, Gp (4).
- the output of the summing node is the final system output: the dynamically decorrelated signal.
- the system is initialised prior to use: (a) the "write" and "t 0 " pointers are allocated; (b) the R 1 , R 2 , Q 1 and Q 2 pointers are allocated to random locations within their respective ranges (the R 1 and Q 2 pointers always lie in the "B" range of the buffer, and the R 2 and Q 1 pointers always lie in the "A” range of the buffer); (c) the gain of the Q and R gain stages is set; and (d) the Q and R crossfaders are configured such that, initially, the Q 1 and R 1 pointers transfer data to the Q and R gain blocks with unity-gain, and from the Q 2 and R 2 pointers with zero-gain.
- the first audio sample is written into the buffer. Data is read from all "read” taps, processed by the associated crossfaders and gain stages, and then summed by the summing (output) node.
- the pointers are all shifted by one sample (to the right in Figure 3), ready for the next read/write event, and the crossfaders are incremented.
- the gain of the zero-gain crossfade path i.e. Q 2 and R 2 at this point
- the gain of the zero-gain crossfade path is increased by a factor of: ⁇ crossfade cycle period ⁇ x ⁇ sample frequency ⁇ .
- the gain factor for the zero-gain crossfade path would be increased from 0 to 1/8192.
- the gain factor for the unit gain (at this point) crossfade path would be decreased from 1 to 8191/8192. Items 2 and 3 are repeated until the crossfade from Q 1 to Q 2 (and R 1 to R 2 ) has fully occurred, after 8192 samples (see Figure 4). At this point, the gain contribution from pointers Q 1 and R 1 is zero, and they are reallocated to new, random positions within their specified ranges.
- the crossfade process is now reversed so as to fade progressively, sample by sample, to these newly allocated taps, such that after another 8192 samples (16384 in all), the unity-gain path is, once again, from pointers Q 1 and R 1 , and the zero-gain path from Q 2 and R 2 , at which point they are reallocated to new, random positions within their specified ranges. This cyclic process is repeated continually.
- the output signal at this point in time is the sum of three vectors (in contrast to the comb filter described earlier, which is the sum of only two vectors), although it is the sum of five vectors almost all of the time.
- An amplitude spectrum of the resultant signal created by the parameters used in the above example is shown in Figure 5.
- the maximum gain is unity (0 dB), which occurs when all three contributions are effectively in phase (taking account of the inverter) and because the gains are 0.50, 0.25 and 0.25.
- the spectral profile is somewhat pseudo-randomly aperiodic (albeit not perfectly so), unlike that of a comb filter, which is perfectly regular and periodic. This feature is important because the profiling is much less audible as an artefact, making the overall effect "tone-neutral".
- the output signal is the sum of the three vectors, but now the pseudo-random modifications of the amplitude and phase spectra are different because of the changed tap locations, as is indicated by the amplitude spectrum shown in Figure 6, which also includes the previous data of Figure 5 to show the differences. (The phase spectra have not been shown here because they are relatively meaningless owing to the "wrap-around" effect which happens when phase differences exceed 2 ⁇ .)
- the decorrelated spectral profile of Figure 5 has been gradually transformed into the spectral profile of Figure 6 (solid line) in about one fifth of a second, and it continues to change, smoothly, continuously and randomly, within the constraints of the specified parameters.
- the main applications are related to (a) Sensaura ZoomFX; (b) the virtualisation of Dolby Digital (creating several right-surround and left-surround sources, rather than a single pair, thus creating a "diffuse" sound-field effect as is important for THX -specified systems), and (c) the virtualisation of Dolby Surround for headphones, creating a pair of decorrelated rear channels from the single, provided surround channel.
Claims (8)
- Verfahren zur Erzeugung eines zweiten Audiosignals (DDC out) aus einem ersten Audiosignal (1) zur Verwendung beim Synthetisieren eines dreidimensionalen Klangfeldes, wobei das zweite Audiosignal derart aus dem ersten Audiosignal dekorreliert wird, dass es im Betrieb von einem Zuhörer als von dem ersten Audiosignal unabhängig wahrgenommen werden kann, wobei das Verfahren die folgenden Schritte umfasst oder aus ihnen besteht: a) Ableiten eines ersten verzögerten Audiosignals (Q) aus dem ersten Signal; b) Multiplizieren dieses ersten verzögerten Audiosignals mit einem Verstärkungsfaktor zwischen null und minus eins zum Erhalt eines ersten verzögerten, Pegel-angepassten Audiosignals; c) Ableiten eines zweiten verzögerten Audiosignals (R) mit einer von dem ersten verzögerten Audiosignal abweichenden Verzögerungszeit aus dem ersten Signal; d) Multiplizieren dieses zweiten verzögerten Audiosignals mit einem Verstärkungsfaktor zwischen null und plus eins zum Erhalt eines zweiten verzögerten, Pegel-angepassten Audiosignals; und e) Kombinieren dieser ersten und zweiten verzögerten, Pegel-angepassten Signale mit dem ersten Audiosignal oder einer zeitlich verzögerten (to) Version des ersten Audiosignals, um das zweite Audiosignal zu erzeugen, dadurch gekennzeichnet, dass die ersten und zweiten verzögerten Audiosignale um Zeitperioden verzögert werden, die dazu veranlasst werden, sich in einer zufälligen oder pseudo-zufälligen oder quasi zufälligen Weise zu verändern.
- Verfahren nach Anspruch 1, bei dem sich die Verstärkungsfaktoren der ersten und zweiten verzögerten, Pegel-angepassten Signale auf null summieren.
- Verfahren nach Anspruch 1 oder 2, bei dem die Verzögerungszeiten dazu veranlasst werden, sich mit einer Frequenz von größer als 0,5 Hz zu verändern.
- Vorrichtung zum Erzeugen eines zweiten Audiosignals (DDC out) aus einem ersten Audiosignal zur Verwendung beim Synthetisieren eines dreidimensionalen Klangfeldes, wobei das zweite Audiosignal derart aus dem ersten Audiosignal dekorreliert wird, dass es im Betrieb von einem Zuhörer als von dem ersten Audiosignal unabhängig wahrgenommen werden kann, wobei die Vorrichtung eine Audiosignal-Laufzeitschaltung (1) umfasst oder aus ihr besteht, mit dem ersten Audiosignal als Eingangssignal und einer Mehrzahl von Ausgangs-Abgriffpunkten (2) mit jeweils einer entsprechend unterschiedlichen Verzögerungszeit innerhalb eines vorbestimmten Bereichs von Verzögerungszeiten, wobei die Ausgangssignale von jedem Abgriffpunkt dieser Mehrzahl mit einem ausgewählten Verstärkungsfaktor (4) vervielfältigt werden, wobei einer der ausgewählten Verstärkungsfaktoren negativ und einer der ausgewählten Verstärkungsfaktoren positiv ist, wobei die Mehrzahl der Pegel-angepassten Ausgangssignale von den Ausgangs-Abgriffpunkten mit dem ersten Audiosignal oder einer zeitlich verzögerten Version des ersten Audiosignals kombiniert werden, um das zweite Audiosignal zu erzeugen, dadurch gekennzeichnet, dass den jeweiligen unterschiedlichen Verzögerungszeiten, die jedem Ausgangs-Abgriffpunkt entsprechen, ausgewählte feste Werte zugeordnet sind, welche so angeordnet sind, dass sie sich in andere feste Werte innerhalb des vorbestimmten Bereichs von Verzögerungszeiten umwandeln.
- Vorrichtung nach Anspruch 4, wobei die Mehrzahl ausgewählter Verstärkungsfaktoren sich auf null summiert.
- Vorrichtung nach Anspruch 4 oder 5, wobei die ausgewählten festen Werte so angeordnet sind, dass sie sich bei einer Frequenz größer als 0,5 Hz in andere feste Werte umwandeln.
- Vorrichtung nach Anspruch 4 bis 6, wobei die ausgewählten festen Werte der jeweiligen abweichenden Verzögerungszeiten in einer zufälligen oder pseudo-zufälligen oder quasi zufälligen Weise umgewandelt werden.
- Vorrichtung nach Anspruch 4 bis 7, wobei das zweite Audiosignal aus dem ersten Audiosignal oder einer zeitlich verzögerten Version des ersten Audiosignals und zwei Ausgangs-Abgriffpunkten oder zwei Gruppen von überblendeten Ausgangs-Abgriffpunkten erzeugt wird, wobei die Amplitude des Ausgangssignals von einem beliebigen Abgriffpunkt null ist, wenn die Verzögerungszeit des betreffenden Anzapfungspunktes von einem festen Wert zu einem anderen verändert wird.
Applications Claiming Priority (3)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
GB9920811A GB2353926B (en) | 1999-09-04 | 1999-09-04 | Method and apparatus for generating a second audio signal from a first audio signal |
GB9920811 | 1999-09-04 | ||
PCT/GB2000/003393 WO2001019138A2 (en) | 1999-09-04 | 2000-09-04 | Method and apparatus for generating a second audio signal from a first audio signal |
Publications (2)
Publication Number | Publication Date |
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EP1212923A2 EP1212923A2 (de) | 2002-06-12 |
EP1212923B1 true EP1212923B1 (de) | 2004-04-21 |
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Application Number | Title | Priority Date | Filing Date |
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EP00956732A Expired - Lifetime EP1212923B1 (de) | 1999-09-04 | 2000-09-04 | Verfahren und anordnung zur erzeugung eines zweiten audiosignales von einem ersten audiosignal |
Country Status (5)
Country | Link |
---|---|
EP (1) | EP1212923B1 (de) |
AT (1) | ATE265128T1 (de) |
DE (1) | DE60010100D1 (de) |
GB (1) | GB2353926B (de) |
WO (1) | WO2001019138A2 (de) |
Families Citing this family (5)
Publication number | Priority date | Publication date | Assignee | Title |
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ATE426235T1 (de) | 2002-04-22 | 2009-04-15 | Koninkl Philips Electronics Nv | Dekodiervorrichtung mit dekorreliereinheit |
SE0301273D0 (sv) * | 2003-04-30 | 2003-04-30 | Coding Technologies Sweden Ab | Advanced processing based on a complex-exponential-modulated filterbank and adaptive time signalling methods |
FR2853804A1 (fr) * | 2003-07-11 | 2004-10-15 | France Telecom | Procede de decodage d'un signal permettant de reconstituer une scene sonore et dispositif de decodage correspondant |
TWI393121B (zh) | 2004-08-25 | 2013-04-11 | Dolby Lab Licensing Corp | 處理一組n個聲音信號之方法與裝置及與其相關聯之電腦程式 |
US8238560B2 (en) | 2006-09-14 | 2012-08-07 | Lg Electronics Inc. | Dialogue enhancements techniques |
Family Cites Families (6)
Publication number | Priority date | Publication date | Assignee | Title |
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NL8303945A (nl) * | 1983-11-17 | 1985-06-17 | Philips Nv | Inrichting voor het realiseren van een pseudo-stereo signaal. |
GB9107011D0 (en) * | 1991-04-04 | 1991-05-22 | Gerzon Michael A | Illusory sound distance control method |
WO1994022278A1 (en) * | 1993-03-18 | 1994-09-29 | Central Research Laboratories Limited | Plural-channel sound processing |
JP2959361B2 (ja) * | 1993-10-21 | 1999-10-06 | ヤマハ株式会社 | 残響音付加装置 |
US5774560A (en) * | 1996-05-30 | 1998-06-30 | Industrial Technology Research Institute | Digital acoustic reverberation filter network |
GB9627015D0 (en) * | 1996-12-28 | 1997-02-19 | Central Research Lab Ltd | Processing audio signals |
-
1999
- 1999-09-04 GB GB9920811A patent/GB2353926B/en not_active Expired - Lifetime
-
2000
- 2000-09-04 AT AT00956732T patent/ATE265128T1/de not_active IP Right Cessation
- 2000-09-04 DE DE60010100T patent/DE60010100D1/de not_active Expired - Lifetime
- 2000-09-04 EP EP00956732A patent/EP1212923B1/de not_active Expired - Lifetime
- 2000-09-04 WO PCT/GB2000/003393 patent/WO2001019138A2/en active IP Right Grant
Also Published As
Publication number | Publication date |
---|---|
WO2001019138A3 (en) | 2001-11-15 |
ATE265128T1 (de) | 2004-05-15 |
EP1212923A2 (de) | 2002-06-12 |
WO2001019138A2 (en) | 2001-03-15 |
DE60010100D1 (de) | 2004-05-27 |
GB9920811D0 (en) | 1999-11-10 |
GB2353926A (en) | 2001-03-07 |
GB2353926B (en) | 2003-10-29 |
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