EP1318502A2 - Verfahren zur Audiocodierung - Google Patents
Verfahren zur Audiocodierung Download PDFInfo
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- EP1318502A2 EP1318502A2 EP02024643A EP02024643A EP1318502A2 EP 1318502 A2 EP1318502 A2 EP 1318502A2 EP 02024643 A EP02024643 A EP 02024643A EP 02024643 A EP02024643 A EP 02024643A EP 1318502 A2 EP1318502 A2 EP 1318502A2
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- European Patent Office
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- information
- room
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- sound
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Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H1/00—Details of electrophonic musical instruments
- G10H1/0091—Means for obtaining special acoustic effects
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H7/00—Instruments in which the tones are synthesised from a data store, e.g. computer organs
- G10H7/002—Instruments in which the tones are synthesised from a data store, e.g. computer organs using a common processing for different operations or calculations, and a set of microinstructions (programme) to control the sequence thereof
- G10H7/006—Instruments in which the tones are synthesised from a data store, e.g. computer organs using a common processing for different operations or calculations, and a set of microinstructions (programme) to control the sequence thereof using two or more algorithms of different types to generate tones, e.g. according to tone color or to processor workload
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H7/00—Instruments in which the tones are synthesised from a data store, e.g. computer organs
- G10H7/02—Instruments in which the tones are synthesised from a data store, e.g. computer organs in which amplitudes at successive sample points of a tone waveform are stored in one or more memories
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2210/00—Aspects or methods of musical processing having intrinsic musical character, i.e. involving musical theory or musical parameters or relying on musical knowledge, as applied in electrophonic musical tools or instruments
- G10H2210/155—Musical effects
- G10H2210/265—Acoustic effect simulation, i.e. volume, spatial, resonance or reverberation effects added to a musical sound, usually by appropriate filtering or delays
- G10H2210/295—Spatial effects, musical uses of multiple audio channels, e.g. stereo
- G10H2210/301—Soundscape or sound field simulation, reproduction or control for musical purposes, e.g. surround or 3D sound; Granular synthesis
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2240/00—Data organisation or data communication aspects, specifically adapted for electrophonic musical tools or instruments
- G10H2240/011—Files or data streams containing coded musical information, e.g. for transmission
- G10H2240/046—File format, i.e. specific or non-standard musical file format used in or adapted for electrophonic musical instruments, e.g. in wavetables
- G10H2240/051—AC3, i.e. Audio Codec 3, Dolby Digital
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2240/00—Data organisation or data communication aspects, specifically adapted for electrophonic musical tools or instruments
- G10H2240/011—Files or data streams containing coded musical information, e.g. for transmission
- G10H2240/046—File format, i.e. specific or non-standard musical file format used in or adapted for electrophonic musical instruments, e.g. in wavetables
- G10H2240/061—MP3, i.e. MPEG-1 or MPEG-2 Audio Layer III, lossy audio compression
-
- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
- G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
- G10H2250/541—Details of musical waveform synthesis, i.e. audio waveshape processing from individual wavetable samples, independently of their origin or of the sound they represent
- G10H2250/571—Waveform compression, adapted for music synthesisers, sound banks or wavetables
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
- H04S2400/11—Positioning of individual sound objects, e.g. moving airplane, within a sound field
Definitions
- the invention relates to a method for audio coding according to the features the preamble of claims 1 and 14.
- Dolby Pro Logic is mainly used in film material to influence the localization of the sound. Speakers are usually mapped onto the center channel and noises can only come from the rear speakers. There is also a whole class of methods used to simulate room acoustics. Commonly used names of such methods are "Hall”, “Stadium”, “Jazz”, “Club” etc. With these methods optimized for music signals, it is not desirable to hear voice signals (vocals) only from the center speaker or a music signal only output from the rear speakers, which is possible using the "Dolby Pro Logic” method.
- the Dolby Pro Logic follow-up process which was called Dolby Pro Logic II, has a mode for music that takes these differences into account, in addition to the film mode.
- a method for coding speech is known from EP 0 481 374 B1.
- a discrete reshaping of a speech window is carried out in order to to get a discrete spectrum of coefficients.
- An approximate one Envelopes of the discrete spectrum are found in any of a variety of Subbands calculated and for digitally coding the defined Envelopes of each subband used.
- each scaled coefficient is divided into a number of bits with at least one converted by a variety of quantizers of different bit lengths.
- the quantizer used for each subband becomes for each speech window by calculating the allocation of bits as a number of bits greater than or determined to be zero, depending on a power density estimate for the subband and a distortion error estimate for the Language window.
- EP 0 587 733 B1 describes a signal analysis system for filtering one or input signals representing several signals.
- the input samples are Analysis window weighted samples.
- means of analysis are Generation of spectral information in response to the time domain signal sample blocks available; where the spectral information Spectral coefficients, which are essentially an even-numbered stacked Time domain aliasing transformation applied to the Time domain signal sample blocks. With the spectral coefficients are essentially coefficients of a modified discrete Cosine transformation or coefficients of a modified discrete sine transformation.
- the analysis means comprise forward-forward transformation means to generate modified sample blocks and Forward transforming means for generating frequency domain transform coefficients.
- EP 0 664 943 B1 is a coding device for adaptive processing of Audio signals for coding, transmission or storage and Recover known, the noise level with the Signal amplitude level fluctuates. It is a processing facility available, which responds to input signals such that they either a first and a second signal or the sum and difference of the first and outputs second signal.
- the first and second signals correspond to the two matrix-coded audio signals of a four to two audio signal matrix, the Processing device also generates a control signal which indicates whether the first and second signal or the sum and difference of the first and second signal is output.
- EP 0 519 055 B1 describes a decoder consisting of a receiving means to receive a plurality of information channels formatted delivery channels, Deformatting means for generating in response to the receiving means, a deformed representation depending on each delivery channel, and Synthesis means for generating output signals depending on the deformed representations, known. Between the means of deforming and the synthesis means are arranged distribution means which on the Deformers address and one or more intermediate signals generate, at least one intermediate signal by combining the Information is generated from two or more of the deformatted representations. The synthesis means generate a respective output signal in response to each of the intermediate signals.
- EP 0 520 068 B1 is an encoder for coding two or more Audio channels known.
- the encoder has a subband device Generation of subband signals, a mixing device for creating a or more composite signals, and means for generating Control information for a corresponding composite signal.
- the encoder has an encoder for generating encoded Information by assigning bits to the one or more composite signals. It is still a formatter for assembling the coded information and the control information an output signal available.
- a speech encoder is known from EP 0 208 712 B1.
- This speech encoder contains a Fourier transformer for executing a discrete one Fourier transform of an incoming speech signal for generation of a discrete transform spectrum of coefficients, one Standardization device for modifying the transformation spectrum for Generation of a standardized, flatter spectrum and for coding one Function that modifies the discrete spectrum.
- one is Device for coding at least a part of the spectrum is available.
- the normalization device has a device (44) for defining the approximated envelopes of the discrete spectrum in each of several Subbands of coefficients and for coding the defined envelope of each subband of coefficients and scaling devices each spectrum coefficient relative to the defined envelope of the relevant subband of coefficients.
- ISO / IEC JTC 1 / SC 29 / WG11 MPEG 4 Structured Audio
- ISO / IEC JTC 1 / SC 29 / WG11 MPEG 4 Structured Audio
- a disadvantage of MPEG 4 Structured Audio is that it is for real-time transmission over a channel with a limited data rate is not suitable because there is no upper one Limitation for the data rate and the computing power is provided. Also the Limitation to samples, which are usually only short pieces of sound correspond and that already corresponds to the playback device at the time need to be completely known when their play starts poses a problem Synchronization is also difficult, should start with the playback at a certain point so it may be necessary to open the file to be calculated completely from the start. An application in film material also contradicts the lack of compression of the samples, which thereby needlessly a lot of data storage.
- Specifying the angle of the Sound sources alone are also not enough to always ensure a good allocation of the Sound sources to reach the image material, because the image material through different projection techniques different sizes and distances can have to the observer. If a speaker can be seen on the left edge of the picture, so its language should also come from the left speakers in a cinema are reproduced in a representation over a 5 meter distance Televisions with 72 cm screen diagonals make more sense, however Output language on the middle speaker, because then there too the speaker appears to be.
- a disadvantage of all known inventions with regard to Audio data compression is that due to the fixation of the positions of the Sound sources have a high number of audio channels to emulate the desired sound field is required. This causes a high one Amount of data, both for storage media and for transmission is undesirable. There is also the possibility of replicating Room acoustics limited by fixing the speaker positions.
- Dolby Digital and DTS do not have ceiling speakers provided that could emulate sounds coming from above.
- the object of the invention is to provide a method which is efficient Coding of multiple audio signals taking into account the associated sound field.
- the present invention solves the problem in that the one Sound sources belonging to the sound field first individually with a conventional one Compression methods, e.g. AC3, MP3, AAC, WMA, etc. be compressed. If a maximum allowed is exceeded Data rate becomes a summary of sound sources or a stronger one Compression performed to reduce the data rate.
- a conventional one Compression methods e.g. AC3, MP3, AAC, WMA, etc.
- Data rate becomes a summary of sound sources or a stronger one Compression performed to reduce the data rate.
- Each of these Sound sources are information about their type as well as a Position information added, indicating where the object is in relation to the viewer. Information about the properties of the the room to be reproduced and the current horizontal and vertical Viewing angles are transmitted.
- the information about the properties of the room to be reproduced, the current horizontal and vertical Viewing angles as well as the size and position of the image of the Playback device are then postprocessing and images of the individual sources to the existing speakers or a headphone signal carried out.
- the audio data generated by different sound sources are encoded in such a way that sound sources are combined that have the same characteristics for the hearing and then this with position information, information about the type of channel (vocals, background noise, speech ... ) and information about possible effects (reverb, dynamic compressor).
- This information is transmitted continuously.
- the type of channel can be used for post-processing in which this signal is decorrelated several times for background noise and output to the loudspeakers.
- An individual post-processing algorithm is assigned to each signal type.
- An advantageous implementation results if, in addition to each sound source, information about its radiation characteristics, such as spherical, kidney-shaped, etc., is also transmitted.
- a further advantageous implementation results if additional information about the desired room characteristics (e.g.
- parameters e.g. reverberation time
- directly algorithmic descriptions to achieve these sound impressions as well as properties of the individual sound sources (such as language or music or effect) are also transmitted.
- the room geometry and the room absorption properties are to be transferred as parameters for the description of the room characteristics.
- a living room with curtains and carpets absorbs the sound much more than a bathroom or a church. If a playback device is not powerful enough to evaluate this data, it ignores this additional data during playback.
- a further advantageous embodiment of the invention results if properties are additionally transmitted via the viewing angle, ie that the angles of the sound source and these viewing angles can be used to directly determine whether the associated object is visible in the image.
- the horizontal angle of the object is within the horizontal viewing angle range and the vertical angle of the object is within the vertical viewing angle range.
- the viewing angle like the position of the objects, is related to the viewer of the original scene. The viewing angle can change continuously, which is why it is advantageously transmitted to each individual image.
- a virtual angle is then determined based on the listening position.
- a non-linear mapping is advantageously used there. For example, the angular range shrunk on small screens can be compensated for by stretching the remaining angular range.
- An advantageous embodiment results if the strength of the compression / extension can be adjusted by the user.
- the invention is described below on the basis of a specific exemplary embodiment shown in FIG.
- the exemplary embodiment shows a device according to the invention.
- the device according to the invention has the audio signal inputs A1 to An.
- Information on the position and properties of the corresponding audio signal is transmitted to the n audio signals via the inputs I1 to In.
- the horizontal and vertical viewing angles and the property of the room to be simulated are fed in via the SF input.
- the incoming audio signals are compressed in the devices K1 to Kn in the data rate to the signals C1 to Cn.
- the data streams C, I and SF are then mixed together in the device MX to form a single data stream US.
- the data stream US can then either be transmitted in broadcasting format or can also be temporarily stored on a storage medium.
- this data stream is then sent to a device DX, which regenerates the data streams C1 to Cn and generates ISF1 to ISFn.
- ISFi corresponds to the data stream li, to which SF is added.
- the compressed data streams C1 to Cn are decompressed in the devices D1 to Dn.
- the individual decompressed data streams are then each given to the devices R1 to Rn together with their associated data stream ISFi and the signal BP.
- the signal BP corresponds to information about the image format, the position of the playback device and the parameters selected by the customer for playback.
- the decoded audio signals are then mapped in the devices Ri to the sum signals S1 to Sk, taking into account the parameters present in BP and ISFi.
- the number of sum signals k corresponds to the number of loudspeakers installed by the customer.
- the sum signals S generated by the individual devices Ri are summed and then output to the loudspeakers L1 to Lk.
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- Engineering & Computer Science (AREA)
- Physics & Mathematics (AREA)
- Acoustics & Sound (AREA)
- Multimedia (AREA)
- General Engineering & Computer Science (AREA)
- Signal Processing (AREA)
- Stereophonic System (AREA)
- Compression, Expansion, Code Conversion, And Decoders (AREA)
Abstract
Description
Weiterhin gibt es eine ganze Klasse von Verfahren, die zur Nachbildung von Raumakustiken verwendet werden. Häufig anzutreffende Bezeichnungen derartiger Verfahren lauten "Hall", "Stadium", "Jazz", "Club" etc.. Bei diesen auf Musiksignale optimierten Verfahren ist es nicht erwünscht, Sprachsignale (Gesang) nur aus dem Center-Lautsprecher zu vernehmen oder ein Musiksignal nur aus den hinteren Lautsprechern auszugeben, welches unter Verwendung des "Dolby Pro Logic" Verfahrens möglich ist.
Bei dem Nachfolgeverfahren von Dolby Pro Logic, welches Dolby Pro Logic II genannt wurde, ist abgesehen von dem Filmmodus ein Modus für Musik vorgesehen, der diese Unterschiede berücksichtigt.
Ist ein Wiedergabegerät nicht leistungsfähig genug, diese Daten auszuwerten, so ignoriert es eben diese Zusatzdaten bei der Wiedergabe.
Eine weitere vorteilhafte Ausgestaltung der Erfindung ergibt sich, wenn zusätzlich noch Eigenschaften über den Sichtwinkel übertragen werden, d.h. dass man aus den Winkeln der Klangquelle und diesen Sichtwinkeln direkt schließen kann, ob das dazugehörige Objekt im Bild sichtbar ist. Dies ist dann der Fall, wenn der horizontale Winkel des Objektes innerhalb des horizontalen Sichtwinkelbereiches und der vertikale Winkel des Objektes innerhalb des vertikalen Sichtwinkelbereiches liegt. Der Sichtwinkel ist dabei wie auch die Position der Objekte auf den Betrachter der Orginalszene bezogen. Der Sichtwinkel kann sich laufend ändern, daher wird dieser vorteilhafterweise zu jedem einzelnen Bild übertragen. Basierend auf dem Winkel des Objektes, des Sichtwinkels sowie der Position und Größe des projezierten Bildes wird dann ein virtueller Winkel bezogen auf die Abhörposition ermittelt. Damit beim Verlassen von Objekten aus dem Sichtwinkelbereich kein abrupter Sprung stattfindet, wird dort vorteilhafterweise eine nichtlineare Abbildung verwendet. So kann z.B. der bei kleinen Bildschirmen geschrumpfte Winkelbereich durch eine Streckung des restlichen Winkelbereiches ausgeglichen werden. Eine vorteilhafte Ausgestaltungung ergibt sich, wenn die Stärke der Stauchung/Streckung durch den Anwender einstellbar ist.
- eine wesentlich realistischere Audiodarbietung erreicht;
- die Ausnutzung der vorhandenen Kanäle wesentlich verbessert, somit wird bei DVDs Kapazität und bei Broadcast-Verfahren Bandbreite gespart;
- bei Hinzufügung der Klangeffekte im Endgerät bei der Codierung der unverfälschten Originalsignale eine wesentlich geringere Datenrate benötigt, dies gibt eine zusätzliche Datenratenersparnis;
- die Kanalabbildung wesentlich verbessert, ein überfliegender Hubschrauber kann mit nur einem Kanal perfekt nachgebildet werden;
- das System abwärtskompatibel gehalten, die Wiedergabe anderer Standards kann durch Annahme der Positionen der Klangquellen auf die dabei üblichen Aufstellungspositionen der Lautsprecher erfolgen; es muss die vom Kunden gewählte Lautsprecherkonstellation nicht modifiziert werden;
- die Aufstellungspositionen der Lautsprecher nicht vorgegeben und kann beliebig sein, die Lautsprecher müssen dem System nur bekannt gemacht werden; das System berechnet dann, auf welchen Lautsprechern es bestimmte Kanäle am geeignetesten ausgibt; so ist es möglich, die räumlichen Gegebenheiten perfekt auszunutzen;
- eine bessere Anpassung an verschiedene Bildformate bei der Wiedergabe erreicht, Klangquellen zu im Bildbereich gehörenden Objekten können klanglich auch dorthin fokussiert werden.
Schließlich ist die zur Übertragung der Zusatzinformationen benötigte Datenmenge gegenüber der Datenmenge zur Codierung der einzelnen Audiokanäle vernachlässigbar gering.
Das Ausführungsbeispiel zeigt eine erfindungsgemäße Vorrichtung. Die erfindungsgemäße Vorrichtung weist die Audiosignaleingänge A1 bis An auf. Über die Eingänge I1 bis In werden zu den n Audiosignalen Informationen zu der Position und die Eigenschaften des entsprechenden Audiosignales übertragen. Über den Eingang SF werden die horizontalen und vertikalen Sichtwinkel sowie die Eigenschaft des nachzubildenden Raumes eingespeist. Die eingehenden Audiosignale werden in den Vorrichtungen K1 bis Kn in der Datenrate zu den Signalen C1 bis Cn komprimiert. Die Datenströme C, I und SF werden dann in der Vorrichtung MX zu einem einzelnen Datenstrom US zusammengemischt. Der Datenstrom US kann dann entweder im Broadcasting-Format übertragen werden oder auch auf einem Speichermedium zwischengespeichert werden. Zur Ausgabe wird dieser Datenstrom dann auf eine Vorrichtung DX gegeben, die daraus wieder die Datenströme C1 bis Cn regeneriert sowie ISF1 bis ISFn erzeugt. ISFi entspricht dabei dem Datenstrom li, dem noch SF hinzugefügt ist. Die komprimierten Datenströme C1 bis Cn werden in den Vorrichtungen D1 bis Dn dekomprimiert. Die einzelnen dekomprimierten Datenströme werden dann jeweils zusammen mit ihrem zugehörigen Datenstrom ISFi sowie dem Signal BP auf die Vorrichtungen R1 bis Rn gegeben. Das Signal BP entspricht dabei einer Information über das Bildformat, der Position des Wiedergabegerätes sowie den vom Kunden gewählten Parametern zur Wiedergabe. In den Vorrichtungen Ri erfolgt dann eine Abbildung der dekodierten Audiosignale auf die Summensignale S1 bis Sk unter Berücksichtigung der in BP und ISFi vorhandenen Parameter. Die Anzahl der Summensignale k entspricht dabei der vom Kunden installierten Anzahl an Lautsprechern. Die von den einzelnen Vorrichtungen Ri erzeugten Summensignale S werden summiert und dann auf die Lautsprecher L1 bis Lk ausgegeben.
- A
- Audiosignaleingang
- BP
- Information über Bildformat, Position der Klangquelle und Kundenparameter
- C
- Komprimierte Audiosignale
- D
- Vorrichtung zur Dekompression einzelner Audiosignale
- DX
- Vorrichtung zur Aufspaltung des Datenstromes
- I
- Eingang für Informationen zu einem Audiosignal
- ISF
- Information zu einem Audiosignal sowie Informationen über die Sichtwinkel der Klangquelle
- K
- Vorrichtung zur Kompression einzelner Audiosignale
- L
- Lautsprecherausgangssignal
- MX
- Vorrichtung zur Zusammenmischung einzelner Datenströme
- R
- Vorrichtung zur Abbildung eines Audiosignales auf die Lautsprechersignale
- S
- Summenkanäle für die Lautsprecher
- SF
- Informationen über die Sichtwinkel der Klangquellen
- US
- Übertragungsstrecke
Claims (14)
- Verfahren zur Codierung von Audiosignalen, wobei mehrere Audiosignale mit einem an und für sich bekannten Kompressionsverfahren komprimiert werden,
dadurch gekennzeichnet, dass
ähnliche zu einem Klangfeld gehörende Klangquellen so zusammengefasst und die Kompressionsfaktoren so gewählt werden, dass die maximal zulässige Datenrate nicht überschritten wird,
zu jeder dieser zusammengefassten Klangquellen Informationen über die Art der Quelle sowie Positionsinformation, die angibt, wo sich das Objekt in Bezug auf den Betrachter befindet, beigefügt wird. - Verfahren nach Anspruch 1,
dadurch gekennzeichnet, dass
Informationen über die Eigenschaften des nachzubildenden Raumes sowie Informationen über die aktuellen horizontalen und vertikalen Sichtwinkel in den erzeugten Datenstrom eingefügt werden. - Verfahren nach Anspruch 1 oder 2,
dadurch gekennzeichnet, dass
im Wiedergabegerät die Größe und Position der Bildprojektion ausgewertet und basierend darauf auf den Eigenschaften des nachzubildenden Raumes, auf den Informationen über die aktuellen horizontalen und vertikalen Sichtwinkeln sowie den Positionsinformationen und Informationen über die Arten und Abstrakteigenschaften der Quellen eine Abbildung der Klangquellen auf die vorhandenen Lautsprecher durchgeführt wird. - Verfahren nach einem der Ansprüche 1 bis 3,
dadurch gekennzeichnet, dass
zur Codierung das AC3-Verfahren, das DTS - Verfahren oder das MP3-Verfahren, oder das AAC-Verfahren, das WMA-Verfahren oder ein ähnliches Verfahren angewendet wird. - Verfahren nach einem der Ansprüche 1 bis 4,
dadurch gekennzeichnet, dass
die Informationen über die Art der Quelle in Form von Algorithmen oder Parametern zu vorgegebenen Algorithmen angegeben wird. - Verfahren nach einem der Ansprüche 1 bis 4,
dadurch gekennzeichnet, dass
die Informationen über die Eigenschaften des nachzubildenden Raumes in Form von Algorithmen oder Parametern zu vorgegebenen Algorithmen angegeben wird. - Verfahren nach einem der Ansprüche 1 bis 6,
dadurch gekennzeichnet, dass
eine Wiedergabe über Kopfhörer oder über Lautsprecher erfolgt. - Verfahren nach einem der Ansprüche 1 bis 7,
dadurch gekennzeichnet, dass
jeder Klangquelle eine Information über deren Abstrahleigenschaften zugewiesen werden kann. - Verfahren nach Anspruch 8,
dadurch gekennzeichnet, dass
die Abstrahleigenschaften kugelförmig, kegelförmig, flächenförmig oder nierenförmig sind. - Verfahren nach einem der Ansprüche 1 bis 9,
dadurch gekennzeichnet, dass
im Wiedergabegerät die Eigenschaften des Abhörraumes, d.h. des Raumes in dem die Lautsprecher aufgestellt werden, einstellbar sind und/oder durch Messungen bestimmt werden können und diese bei der Wiedergabe berücksichtigt werden. - Verfahren nach Anspruch 10,
dadurch gekennzeichnet, dass
die Eigenschaften des Abhörraumes durch die Geometrie des Raumes und Beschaffenheit der Wände und des Bodens vorgegeben werden. - Verfahren nach einem der Ansprüche 1 bis 11,
dadurch gekennzeichnet, dass
eine beliebige Anzahl an Lautsprechern zur Wiedergabe verwendbar ist. - Verfahren nach einem der Ansprüche 1 bis 9,
dadurch gekennzeichnet, dass
die Lautsprecherpositionen dem Wiedergabegerät bekannt gemacht werden, so dass die bestmögliche Abbildung der einzelnen Kanäle auf die Lautsprecher vorgenommen wird. - Vorrichtung zur Durchführung des Verfahren nach einem oder mehreren der vorangehenden Ansprüche.
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
DE10154932A DE10154932B4 (de) | 2001-11-08 | 2001-11-08 | Verfahren zur Audiocodierung |
DE10154932 | 2001-11-08 |
Publications (3)
Publication Number | Publication Date |
---|---|
EP1318502A2 true EP1318502A2 (de) | 2003-06-11 |
EP1318502A3 EP1318502A3 (de) | 2009-10-07 |
EP1318502B1 EP1318502B1 (de) | 2010-06-09 |
Family
ID=7705086
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP02024643A Expired - Lifetime EP1318502B1 (de) | 2001-11-08 | 2002-11-05 | Verfahren zur Audiocodierung |
Country Status (4)
Country | Link |
---|---|
EP (1) | EP1318502B1 (de) |
AT (1) | ATE470925T1 (de) |
DE (2) | DE10154932B4 (de) |
ES (1) | ES2347221T3 (de) |
Cited By (2)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US9451363B2 (en) | 2012-03-06 | 2016-09-20 | Dolby Laboratories Licensing Corporation | Method and apparatus for playback of a higher-order ambisonics audio signal |
US10192563B2 (en) | 2014-03-26 | 2019-01-29 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for screen related audio object remapping |
Citations (3)
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WO1998057436A2 (en) * | 1997-06-10 | 1998-12-17 | Lars Gustaf Liljeryd | Source coding enhancement using spectral-band replication |
EP1098298A2 (de) * | 1999-11-08 | 2001-05-09 | Mitsubishi Denki Kabushiki Kaisha | Sprachkodierung mittels merfachen Langzeitprädiktion-Kandidaten |
EP1107232A2 (de) * | 1999-12-03 | 2001-06-13 | Lucent Technologies Inc. | Kombinierte Stereokodierung von Audiosignalen |
-
2001
- 2001-11-08 DE DE10154932A patent/DE10154932B4/de not_active Expired - Lifetime
-
2002
- 2002-11-05 AT AT02024643T patent/ATE470925T1/de not_active IP Right Cessation
- 2002-11-05 EP EP02024643A patent/EP1318502B1/de not_active Expired - Lifetime
- 2002-11-05 DE DE50214481T patent/DE50214481D1/de not_active Expired - Lifetime
- 2002-11-05 ES ES02024643T patent/ES2347221T3/es not_active Expired - Lifetime
Patent Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
WO1998057436A2 (en) * | 1997-06-10 | 1998-12-17 | Lars Gustaf Liljeryd | Source coding enhancement using spectral-band replication |
EP1098298A2 (de) * | 1999-11-08 | 2001-05-09 | Mitsubishi Denki Kabushiki Kaisha | Sprachkodierung mittels merfachen Langzeitprädiktion-Kandidaten |
EP1107232A2 (de) * | 1999-12-03 | 2001-06-13 | Lucent Technologies Inc. | Kombinierte Stereokodierung von Audiosignalen |
Cited By (10)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US9451363B2 (en) | 2012-03-06 | 2016-09-20 | Dolby Laboratories Licensing Corporation | Method and apparatus for playback of a higher-order ambisonics audio signal |
US10299062B2 (en) | 2012-03-06 | 2019-05-21 | Dolby Laboratories Licensing Corporation | Method and apparatus for playback of a higher-order ambisonics audio signal |
US10771912B2 (en) | 2012-03-06 | 2020-09-08 | Dolby Laboratories Licensing Corporation | Method and apparatus for screen related adaptation of a higher-order ambisonics audio signal |
US11228856B2 (en) | 2012-03-06 | 2022-01-18 | Dolby Laboratories Licensing Corporation | Method and apparatus for screen related adaptation of a higher-order ambisonics audio signal |
US11570566B2 (en) | 2012-03-06 | 2023-01-31 | Dolby Laboratories Licensing Corporation | Method and apparatus for screen related adaptation of a Higher-Order Ambisonics audio signal |
US11895482B2 (en) | 2012-03-06 | 2024-02-06 | Dolby Laboratories Licensing Corporation | Method and apparatus for screen related adaptation of a Higher-Order Ambisonics audio signal |
US10192563B2 (en) | 2014-03-26 | 2019-01-29 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for screen related audio object remapping |
US10854213B2 (en) | 2014-03-26 | 2020-12-01 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for screen related audio object remapping |
US11527254B2 (en) | 2014-03-26 | 2022-12-13 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for screen related audio object remapping |
US11900955B2 (en) | 2014-03-26 | 2024-02-13 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for screen related audio object remapping |
Also Published As
Publication number | Publication date |
---|---|
DE10154932B4 (de) | 2008-01-03 |
EP1318502B1 (de) | 2010-06-09 |
DE10154932A1 (de) | 2003-06-12 |
ES2347221T3 (es) | 2010-10-27 |
DE50214481D1 (de) | 2010-07-22 |
EP1318502A3 (de) | 2009-10-07 |
ATE470925T1 (de) | 2010-06-15 |
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