WO2006003813A1 - Appareil de codage et de decodage audio - Google Patents

Appareil de codage et de decodage audio Download PDF

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Publication number
WO2006003813A1
WO2006003813A1 PCT/JP2005/011340 JP2005011340W WO2006003813A1 WO 2006003813 A1 WO2006003813 A1 WO 2006003813A1 JP 2005011340 W JP2005011340 W JP 2005011340W WO 2006003813 A1 WO2006003813 A1 WO 2006003813A1
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Prior art keywords
signal
transient
channel
original sound
phase difference
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PCT/JP2005/011340
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English (en)
Japanese (ja)
Inventor
Naoya Tanaka
Kok Seng Chong
Mineo Tsushima
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Matsushita Electric Industrial Co., Ltd.
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Publication of WO2006003813A1 publication Critical patent/WO2006003813A1/fr

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/007—Two-channel systems in which the audio signals are in digital form
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing

Definitions

  • the present invention relates to an apparatus for efficiently encoding an audio signal with a small amount of information, and for decoding the encoded information.
  • audio coding is to compress and transmit digitally encoded audio signals as efficiently as possible, and to reproduce as high quality audio signals as possible by decoding processing at a decoder. .
  • the configuration of the latter is shown in FIG.
  • the information to be input is a monaural time signal 504 and a BCC (Binaural Cue Coding) parameter 506.
  • the mono time signal 504 is typically a downmix signal of the original signal, eg, (L + R) / 2.
  • the TF converter 501 converts the monaural time signal 504 into a frequency parameter 505.
  • any known transformation such as Fourier transformation or cosine transformation may be used.
  • FFT fast Fourier transform
  • Auditory sound field generation means 502 performs processing based on the spatial acoustic information provided by the BCC meter on the input monaural FFT coefficients 505, and stereo FFT coefficients representing a predetermined acoustic space. Generate 507.
  • the inverse TF transformation means 503 inversely transforms the FFT coefficient 50 7 and outputs a stereo time signal 508.
  • the spatial acoustic information given by the BCC parameter is ITD (Inter-Aura or Inter-channel Time Difference), It is expressed as ILD (Inter-aural or Inter- channel Level Difference) and ICC (Inter-aural or Inter- channel Coherence).
  • ITD Inter-Aura or Inter-channel Time Difference
  • ILD Inter-aural or Inter- channel Level Difference
  • ICC Inter-aural or Inter- channel Coherence
  • Auditory sound field generation means 502 generates a stereo signal having a time difference Ts indicated by ITD with respect to monaural FFT coefficient 505. In general, this is realized by performing delay or phase shift operation on the monaural FFT coefficient 505 to generate a time difference of + Ts / 2 and ⁇ Ts / 2, respectively. Subsequently, the gain of each channel is adjusted to reflect the inter-channel level difference indicated by the ILD.
  • ICC is a parameter indicating the spread of acoustic space.
  • Patent Document 2 shows a method of controlling the spread of acoustic space by changing the gain difference between channels in a pseudo-random manner and adjusting the change in gain difference according to the ICC to be transmitted. ing. It also shows how to control the spread of the acoustic space by adjusting the time difference indicated by ITD according to I CC.
  • the stereo signal 508 generated by adjusting the time difference between channels, the level difference and the sense of wideness according to the BCC parameters has the same spatial acoustic characteristics as the reference original sound stereo signal.
  • the sound quality similar to the original sound stereo signal can be realized in terms of hearing.
  • Non-Patent Document l ISO / IEC 14496-3: 2001 AMD2 "Parametric Coding for High Quality Audio
  • Patent Document 1 US Published Patent US 2003/0035553 "Backwards- compatible Perceptual Coding of Spatial Cues
  • Patent Document 2 US Published Patent US2003 / 0219130 "Coherence-based Audio Coding and Synthesis"
  • the attack sound may be a tight sound image or a sound image with a sense of expansion that exists in a narrow range within the sound space. It is necessary to express the power of
  • a measure called coherence (Coherence ce) represented by ICC has been used as an indicator that indicates a sense of sound image spread.
  • This value is a measure that indicates the correlation between channels, and the attack sound is For example, for signals with low inter-channel correlation, there has been a problem that the reliability as a measure decreases.
  • the transient property of the signal is detected at the decoder, and the amount of reverberation component to be added to the signal of transient property. Controls the sense of the sound image's spread.
  • the present invention is intended to solve the above-mentioned problems, and an object of the present invention is to provide an audio coding system and decoding technology capable of more accurately expressing the spatial acoustic characteristics to an attack sound.
  • the audio code device of the present invention comprises an original sound signal of m channels (m is a natural number of 2 or more), a downmix signal of n channels (n is a natural number smaller than m), and a channel of the original sound signal.
  • An audio encoding device for generating a spatial acoustic information signal representing a phase difference between the two, and downmixing means for generating the downmix signal by downmixing the original sound signal, and analyzing the original sound signal.
  • the audio decoding device of the present invention is an n-channel (n is a natural number smaller than m) downmix obtained by downmixing m-channel (m is a natural number of 2 or more) source signals.
  • An audio decoding device that generates an m-channel decoded signal based on a phase difference determined between channels from a signal, and is a transient information signal that indicates transientness level indicating the transient property of the original sound signal for each channel. And a signal generation unit for generating the decoded signal for each channel based on the phase difference and the degree of transientness from the downmixing signal, and the generated signal.
  • the present invention can be realized as an audio encoding device and an audio decoding device, and can be realized as an audio transmission system having both capabilities, and the characteristic means provided with these devices.
  • the present invention can also be realized as an audio coding method and an audio decoding method, each of which has a process performed by Effect of the invention
  • the characteristic configuration of using the transient level for each channel which represents the transient property of the original sound signal, enables phase difference, correlation, level difference, etc. between channels when reproducing speech.
  • Spatial acoustic information can be selectively applied on a channel-by-channel basis depending on speech transients. Therefore, if it is indicated by the degree of transientness that the voice is a transient voice (for example, attack sound etc.) for which it is difficult to obtain the space acoustic information accurately, the spatial acoustic information is applied at the time of the reproduction. By suppressing the noise level, the problem that the sound is reproduced according to the inaccurate spatial acoustic information and the sound image of the reproduced sound is unintentionally spread is eliminated.
  • FIG. 1 is a diagram showing an exemplary configuration of an audio transmission system according to a first embodiment of the present invention.
  • FIG. 2 is a diagram showing a configuration example of an audio decoding device according to a first embodiment of the present invention.
  • FIG. 3 is a diagram for explaining a smoothing process of the phase shift amount.
  • FIG. 4 is a diagram showing a configuration example of an audio decoding device according to a second embodiment of the present invention.
  • FIG. 5 is a diagram showing an exemplary configuration of an audio decoding device according to a third embodiment of the present invention.
  • FIG. 6 is a diagram showing a configuration example of a conventional audio decoding device.
  • FIG. 1 is a functional block diagram showing an example of a configuration of an audio transmission system according to a first embodiment of the present invention.
  • the present audio transmission system is an audio transmission system that expresses stereo original sound signals L and R into a monaural downmix signal M and a spatial acoustic information signal representing at least a phase difference between channels of the original sound signals.
  • An audio encoding device 10 and an audio decoding device 20 are included in the audio transmission system.
  • the audio transmission system of the present invention is particularly characterized in that it transmits, together with the downmix signal and the spatial acoustic information signal, a transient information signal representing, for each channel, a transient degree indicating the transientness of the original sound signal. Will be attached. Technical significance of this level of transition Will be described in detail later.
  • the stereo original sound signal is an example when m of the m channel original sound signal is 2, and the monaural downmix signal is when n of the n channel downmix signal is 1.
  • the audio encoding device 10 is a device that multiplexes the downmix signal M obtained from the original sound signals L and R, the spatial sound information signal, and the transient information signal into one bit stream 108 and outputs the multiplexed signal.
  • the downmix means 601 obtains the downmix signal M by downmixing the input original sound signals L and R, and outputs the downmix signal M to the core encoding means 602.
  • this dual mix process may be a process of obtaining the average of the original sound signals L and R and using it as the down mix signal M.
  • the core encoding means 602 obtains a core bit stream by encoding the input downmix signal M, and outputs the core bit stream to the bit stream multiplexing means 605. Specifically, this encoding can be performed using a known encoding technique such as the MPEG-4 AAC method.
  • the spatial acoustic information analysis means 603 analyzes the original sound signals L and R to detect the phase difference between the channels of the original sound signal, and bit stream multiplexes the spatial acoustic information signal representing the detected phase difference. Output to the converting means 605.
  • the spatial acoustic information analysis means 603 may detect the correlation and level difference between channels of the original sound signal in addition to the phase difference, and may express the detected correlation and level difference in the spatial acoustic information signal together with the phase difference. .
  • phase difference which is mathematically synonymous with time difference
  • the transient detection means 604 detects a transient degree indicating transientness with respect to each of the original sound signals L and R.
  • the transient property detection means 604 is configured to, for example, change value of signal energy or change value of signal amplitude within a predetermined time for each channel of the original sound signal.
  • the degree of transientness in the form of binary information indicating whether the detected change value exceeds a predetermined threshold value or binary information indicating whether the detected change value exceeds a predetermined threshold value, or of quantizing the detected change value.
  • Get The spatial acoustic information analysis means 603 outputs a transient information signal representing the degree of transientness for each channel detected by the transient detection means 604 to the bit stream multiplexing means 605.
  • the degree of transientness of each channel is thus expressed as the degree of transientness itself, or in the form of a deviation of the standard level of transientity (for example, channel average of the degree of transientness) per channel. It is also conceivable to be represented.
  • the bitstream multiplexing means 605 obtains one bitstream 108 by multiplexing the core bitstream, the acoustic space information signal, and the transient information signal, and the bitstream 108 is not shown. Output to communication lines, broadcast lines, recording media, etc.
  • the audio decoding device 20 acquires a bit stream 108 from a communication line, a broadcast line, a recording medium, etc. (not shown), and decodes the decoded signals L ′ and R ′ which simulate the original sound signals L and R. It is an apparatus to be obtained, and is composed of a bit stream separation means 101, a core decoding means 102, and a signal generation means 100.
  • bit stream separation means 101 is an example of a signal acquisition means.
  • the bit stream separation means 101 acquires a bit stream 108, and multiples separates the core bit stream, the spatial acoustic information signal, and the transient information signal from the acquired bit stream 108.
  • the spatial acoustic information signal represents at least a phase difference between channels of the original sound signal, and may further express the correlation and level difference between channels of the original sound signal in addition to the phase difference.
  • the core decoding unit 102 obtains the downmix signal M by decoding the separated core bit stream, and outputs the downmix signal M to the signal generation unit 100.
  • the core decoding means 102 obtains the downmix signal M by inverse conversion of the encoding by the core encoding means 602 described above.
  • the signal generation means 100 obtains from the input downmix signal M the phase difference between the channels represented by the separated spatial environment signal and the separated transient information signal. Decoded signals L 'and R' are generated and output for each channel based on the degree of transientness.
  • the audio decoding device 20 will be described, focusing on the functional detailed configuration of the signal generation means 100 and the processing performed there.
  • FIG. 2 is a diagram showing an example of a functional configuration of the audio decoding device 20 according to the first embodiment of the present invention.
  • the internal functional configuration of the signal generation means 100 is shown in detail in FIG.
  • the signal generation unit 100 includes an FFT unit 103, a signal separation unit 104, a correlation control unit 105, a gain control unit 106, and an IFFT unit 107.
  • the input bit stream 108 is separated into a core bit stream 109, an IPD metric 112, a TF parameter 113, an ICC parameter 114, and a gain parameter 115 in the bit stream separating means 101.
  • the core bit stream 109 is decoded by the core decoding means 102 to generate a monophonic PCM signal 110.
  • This monaural PCM signal 110 power corresponds to the downmix signal M described above.
  • any existing coding scheme such as the MPEG-4 AAC scheme may be used, for example.
  • the decoded monaural PCM signal 110 is converted into monaural FFT coefficients 111 in the FFT means 103.
  • the IPD parameter 112 is a parameter representing inter-channel phase difference, and is mathematically synonymous with the ITD described above.
  • the IPD parameter 112, the ICC parameter 114, and the Gain parameter 115 are represented by a space acoustic information signal and transmitted.
  • the TF parameter 113 is a parameter that represents the degree of transientness described above, and is represented and transmitted by the transientity information signal.
  • the signal separation means 104 applies separation processing based on the IPD parameter 112 and the TF parameter 113 to the monaural FFT coefficient 111 to generate a separated FFT coefficient 116.
  • the correlation control means 105 adjusts the correlation between channels according to the ICC parameter 114 for the separated FFT coefficient 116.
  • the operations of the signal separation means 104 and the correlation control means 105 will be described in detail later.
  • the inter-channel correlation adjusted FFT coefficients 117 are gain adjusted in gain control means 106 to produce stereo FFT coefficients 118.
  • the inverse FFT means 107 inverse FFT transforms the stereo FFT coefficients 118 and outputs a stereo PCM signal 119.
  • the operation of the signal separation unit 104 will be described in detail.
  • the IPD is phase difference information between channels of the original sound stereo signal analyzed and encoded in an encoder (the audio encoding device 10 is an example of this encoder).
  • IPD is encoded as a representative value for a plurality of FFT coefficients included in each subband, with respect to the FFT coefficients divided into a plurality of subbands. Therefore, signal separation processing is performed in units of subbands.
  • the inter-channel phase difference in the sub-band is represented by 0.
  • the signal separation means 104 separates the monaural FFT coefficient 111 into two signals having an inter-channel phase difference ⁇ .
  • two FFT coefficients having phases of ⁇ / 2 and + ⁇ / 2 with respect to monaural FFT coefficient 111 are calculated as follows.
  • s is a monaural FFT coefficient 111
  • k is a suffix of the FFT coefficient
  • N is the number of FFT points to be used.
  • the separated stereo FFT coefficient d and the original monaural FFT coefficient si 16 are input to the correlation control means 105, the inter-channel correlation is adjusted according to the ICC parameter 114, and the stereo FFT coefficient 105 having a desired sense of spread is generated. Be done.
  • the adjustment of the interchannel correlation can be realized by adding and subtracting the separated stereo FFT coefficient and the original monaural FFT coefficient, and is performed, for example, according to the equation shown below.
  • R '(k) cos (a) s (k) + sin (a) d R (k)
  • the embodiment of the present invention uses a TF parameter 113 that represents the degree of transientness of the input signal.
  • the TF parameter 113 is calculated and transmitted for each channel of the input original sound stereo signal in the encoder.
  • a change value of signal energy within a predetermined time frame, a change value of signal amplitude, etc. can be used, and binary information with or without the transient property or the degree of the transient property It is transmitted as multi-value information representing multiple steps.
  • TF parameters are calculated and transmitted as one value representing all frequency bands for each channel, or the frequency bands are divided into a plurality of subbands, and calculated and transmitted for each subband. .
  • the signal 113 is input to the signal separation unit 104.
  • the signal separation processing in the signal separation means 104 represented by the equation (1) is transformed, for example, as follows by using the TF parameter 113.
  • subscripts L and R indicate that the parameters are L channel and R channel, respectively.
  • FIG. 3 is a diagram showing an example of phase shift amount smoothing. Assuming that the phase shift amount of frequency subband 202 is 0 1 and the phase shift amounts of adjacent subbands 201 and 203 are 0 0 and 0 2 respectively, the phase shift amounts for FFT coefficients included in each subband are It is controlled to change stepwise from ⁇ 0 to ⁇ 1 and ⁇ 2 based on the position on the frequency. By smoothing the phase shift amount like this, it is possible to prevent the sound quality deterioration due to the rapid phase change at the sub-band boundary. In the example shown in FIG. 3, although the smoothing is performed by linear interpolation, any interpolation method such as curve interpolation using a polynomial can be used.
  • the inter-channel phase difference information by the IPD parameter is the force transmitted as the phase difference information between L and R channels, and the down-mixed monaural signal (M
  • the phase difference information ⁇ LM between L and M channels and the phase difference information ⁇ RM between R and M channels may be transmitted.
  • the signal separation process in this case can be realized, for example, as the following equation.
  • phase difference between two channels arbitrarily selected from the three channels L, R, and M is an arbitrary two pairs of inter-channel phase differences (in the above example, a combination of L and M, Any combination of R and M) can be reproduced. Therefore, the inter-channel phase difference information may be transmitted for any two sets of channel combinations.
  • FIG. 4 is a diagram showing an example of a functional configuration of the audio decoding device 20a according to the second embodiment of the present invention.
  • the audio decoding device 20a adds a transientity detection means 301 in the signal generation means 100a to the audio decoding device 20 of the first embodiment described above, and the transientness degree The difference is that, instead of the TF parameter 113, which is shown, a transient channel information TC (Transient Channel) parameter 303 is used.
  • the transient detection unit 301 analyzes the output monaural PCM signal 110 of the core decoding unit 102 to calculate a TF parameter 302 indicating the degree of transient of the signal.
  • TC Transient Channel
  • the degree of transientity for each channel can be calculated as follows.
  • signal separation processing may be performed based on the equation (3) described above. According to this configuration, it is necessary to newly provide a means for detecting transientity in the decoder, but in general, the amount of information necessary for code transmission of TC parameters is determined by coding TF parameters for each channel. ⁇ ⁇ ⁇ Less than the amount of information required to transmit. Therefore, high-quality coded transmission can be realized with a smaller amount of information.
  • transientness detection means 301 calculates the transientness degree using monaural PCM signal 110 output from core decoding means.
  • the configuration is such that the change of the monaural FFT coefficient 111 output to the series is used. It is good.
  • FIG. 5 is a diagram showing a functional configuration of an audio decoding device 20b according to a third embodiment of the present invention.
  • the audio decoding device 20b removes the signal separation means 104 and the correlation control means 105 in the signal generation means 100b from the audio decoding device 20 of the first embodiment described above, and newly
  • the second embodiment differs in that a second signal separation unit 401 is provided, and an ICC parameter 114 is input to the second signal separation unit 401.
  • signal separation processing and inter-channel correlation control processing are integrated and performed simultaneously according to TF parameter 113 and ICC parameter 114.
  • the processing in the second signal separation means 401 differs depending on whether the inter-channel phase difference information by the IPD parameter 112 is transmitted or not.
  • inter-channel phase difference and inter-channel correlation are both measures of signal separation between channels, and generally, inter-channel correlation between signals processed to have correct inter-channel phase difference. Is consistent with the desired inter-channel correlation. Therefore, the phase shift amount is controlled in accordance with the phase difference information for the sub-band in which the IPD parameter 112 is transmitted.
  • processing in the frequency sub-band is expressed by the following equation.
  • ⁇ ⁇ a - (l.0- TF R)
  • a is the inter-channel phase difference between which the inter-channel correlation ICC force is also estimated.
  • the ICD parameter is not required for the frequency subband in which the IPD parameter is transmitted, and the transmission can be omitted. Sound quality coded transmission can be realized.
  • the inter-channel phase difference information transmitted by the IPD parameter is a value representative of a certain frequency sub-band, there is a possibility that correct inter-channel correlation can not be obtained. Therefore, a correction term by the ICC parameter may be introduced also in the processing in the frequency subband in which the IPD parameter is transmitted. As the correction term, correction is performed such that the phase shift amount increases as the inter-channel correlation force S decreases as the inter-channel correlation indicated by the ICC parameter increases, for example, equation (6) The process indicated by can be corrected as in the following equation.
  • DCT DCT, filter bank, etc. are known instead of force FFT using an example of performing decoding processing using FFT coefficients. It is easy to realize the same configuration using any time-frequency conversion.
  • down-mixed monaural signals for the stereo original sound signal and spatial acoustic parameters for representing the stereo acoustic space are represented by symbols.
  • This invention is applicable to multichannel original sound signals of any number of channels, and the structure described for the structure of transmitting and decoding a stereo signal based on the transmitted information. That is, an n-channel downmix signal of m by n is generated and encoded with respect to the n-channel input original sound signal, and transmitted as a core bit stream. The remaining nm channels are represented by spatial acoustic parameters and coded and transmitted.
  • the m-channel signal is decoded in the core decoding means, and the decoding process is performed according to the construction of the present invention to generate a ⁇ nm-channel signal.
  • the n-channel signal is the same as the input original sound signal. Can be generated.
  • the audio encoding device, the audio decoding device, and the audio transmission system of the present invention it is characterized by using the transition degree of each channel of the original sound signal. Phase difference, correlation and level difference between channels! It is difficult to accurately obtain space acoustic information by using conventional powers, and the degree to which the space acoustic information is applied during reproduction selectively for a transient sound (for example, attack sound etc.) Can be suppressed. [0069] By this, the inconvenience that the sound is expected to be transient and the transient sound that is expected to be sound image is reproduced according to the inaccurate spatial acoustic information, and the sound image of the reproduced sound is spread as a result is eliminated. Ru.
  • the present invention provides an audio encoding device and an audio encoding method including steps performed by characteristic means included in these devices that can be realized as an audio decoding device as described above. And as an audio decoding method.
  • An audio coding method that includes is included in the present invention.
  • Audio decoding including a signal generation step of generating the decoded signal for each channel based on the phase difference and the transition degree from the downmix signal, and outputting the generated decoded signal.
  • the method of conversion is included in the present invention.
  • a computer-executable program for executing these methods using a computer and a program recording medium storing the program are also included in the present invention.
  • the present invention is a technology that enables high-quality transmission and reproduction of stereo or multi-channel audio signals with a small amount of information, and can be performed with less bandwidth in broadcasting, communication, music distribution including the Internet, etc. It enables high-quality services, and enables recording and storage of longer-term high-quality audio signals in media such as CDs, DVDs and hard disks.

Abstract

Appareil de décodage audio destiné à améliorer le processus de séparation de signal en se basant sur des informations acoustiques spatiales pour améliorer la qualité du son, comportant des moyens de séparation de signal (104) qui effectuent, en se basant à la fois sur un paramètre IPD d’informations de différence de phases entre canaux (112) et un paramètre TF de degré de transition de signal (113), un processus de séparation de signal pour un coefficient FFT monaural entré (111). Des coefficients FFT (116) tels que séparés pour une pluralité de canaux sont ajustés en corrélation entre canaux par des moyens de contrôle de correction (105) et ensuite ajustés en gain par des moyens de commande de gain (106). Le processus de séparation de signal est réalisé en décalant, en se basant sur une quantité de décalage de phase telle que calculée à partir du paramètre IPD, la phase du coefficient monaural FFT. La quantité de décalage pour chaque canal est ajustée selon le paramètre TF, améliorant de ce fait la qualité du son par rapport à un signal passager, tout particulièrement un son d’attaque ou similaire.
PCT/JP2005/011340 2004-07-02 2005-06-21 Appareil de codage et de decodage audio WO2006003813A1 (fr)

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WO2007116809A1 (fr) * 2006-03-31 2007-10-18 Matsushita Electric Industrial Co., Ltd. Dispositif de codage audio stereo, dispositif de decodage audio stereo et leur procede
JP2009524339A (ja) * 2006-01-19 2009-06-25 エルジー エレクトロニクス インコーポレイティド メディア信号の処理方法及び装置
JP2009543112A (ja) * 2006-06-29 2009-12-03 エヌエックスピー ビー ヴィ 音声パラメータの復号化
WO2010036059A2 (fr) * 2008-09-25 2010-04-01 Lg Electronics Inc. Procédé et appareil pour traiter un signal
JP2010521909A (ja) * 2007-03-21 2010-06-24 フラウンホファー・ゲゼルシャフト・ツール・フォルデルング・デル・アンゲバンテン・フォルシュング・アインゲトラーゲネル・フェライン 音声の再現を高めるための方法および装置
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EP2467851A2 (fr) * 2009-08-18 2012-06-27 Samsung Electronics Co., Ltd. Procédé de décodage audio multicanal et appareil correspondant
US8258849B2 (en) 2008-09-25 2012-09-04 Lg Electronics Inc. Method and an apparatus for processing a signal
US8290167B2 (en) 2007-03-21 2012-10-16 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Method and apparatus for conversion between multi-channel audio formats
US8346380B2 (en) 2008-09-25 2013-01-01 Lg Electronics Inc. Method and an apparatus for processing a signal
EP2612322A1 (fr) * 2010-10-05 2013-07-10 Huawei Technologies Co., Ltd. Procédé et appareil d'encodage/de décodage de signal audio multicanal
JP5340378B2 (ja) * 2009-02-26 2013-11-13 パナソニック株式会社 チャネル信号生成装置、音響信号符号化装置、音響信号復号装置、音響信号符号化方法及び音響信号復号方法
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
US8917874B2 (en) 2005-05-26 2014-12-23 Lg Electronics Inc. Method and apparatus for decoding an audio signal
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
US9595267B2 (en) 2005-05-26 2017-03-14 Lg Electronics Inc. Method and apparatus for decoding an audio signal
US9626976B2 (en) 2006-02-07 2017-04-18 Lg Electronics Inc. Apparatus and method for encoding/decoding signal
RU2782168C1 (ru) * 2010-07-19 2022-10-21 Долби Интернешнл Аб Система и способ для генерирования ряда сигналов высокочастотных поддиапазонов
US11568880B2 (en) 2010-07-19 2023-01-31 Dolby International Ab Processing of audio signals during high frequency reconstruction
WO2024023108A1 (fr) * 2022-07-28 2024-02-01 Dolby International Ab Amélioration d'image acoustique pour audio stéréo

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