EP1779385A4 - Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information - Google Patents

Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information

Info

Publication number
EP1779385A4
EP1779385A4 EP05774399A EP05774399A EP1779385A4 EP 1779385 A4 EP1779385 A4 EP 1779385A4 EP 05774399 A EP05774399 A EP 05774399A EP 05774399 A EP05774399 A EP 05774399A EP 1779385 A4 EP1779385 A4 EP 1779385A4
Authority
EP
European Patent Office
Prior art keywords
audio signal
vector
channel
signal
source location
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Granted
Application number
EP05774399A
Other languages
German (de)
French (fr)
Other versions
EP1779385B1 (en
EP1779385A1 (en
Inventor
Jeong Il Seo
Han Gil Moon
Seung Kwon Beack
Kyeong Ok Kang
In Seon Jang
Koeng Mo Sung
Min Soo Hahn
Jin Woo Hong
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Electronics and Telecommunications Research Institute ETRI
Seoul National University Industry Foundation
Original Assignee
Electronics and Telecommunications Research Institute ETRI
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Priority claimed from KR1020050061425A external-priority patent/KR100663729B1/en
Application filed by Electronics and Telecommunications Research Institute ETRI filed Critical Electronics and Telecommunications Research Institute ETRI
Publication of EP1779385A1 publication Critical patent/EP1779385A1/en
Publication of EP1779385A4 publication Critical patent/EP1779385A4/en
Application granted granted Critical
Publication of EP1779385B1 publication Critical patent/EP1779385B1/en
Not-in-force legal-status Critical Current
Anticipated expiration legal-status Critical

Links

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S3/00—Systems employing more than two channels, e.g. quadraphonic
    • H04S3/008—Systems employing more than two channels, e.g. quadraphonic in which the audio signals are in digital form, i.e. employing more than two discrete digital channels
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; 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
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/03—Application of parametric coding in stereophonic audio systems

Definitions

  • the present invention relates to a method and apparatus for encoding/decoding a multi-channel audio signal, and more particularly, to a method and apparatus for ef ⁇ fectively encoding/decoding a multi-channel audio signal using Virtual Sound Location Information (VLSI).
  • VLSI Virtual Sound Location Information
  • MPEG Moving Picture Experts Group
  • BC Backward Compatibility
  • AAC MPEG-2 Advanced Audio Coding
  • MPEG- 4 AAC has been standardized in the MPEG.
  • multi-channel audio compression technology such as AC-3 and Digital Theater System (DTS)
  • DTS Digital Theater System
  • BCC is technology for effectively compressing a multi-channel audio signal that has been developed on a basis of the fact that people can acoustically perceive space due to a binaural effect. BCC is based on the fact that a pair of ears perceives a location of a specific sound source using interaural level differences and/or interaural time differences.
  • a multi-channel audio signal is downmixed to a monophonic or stereophonic signal and channel information is represented by binaural cue parameters such as Inter-channel Level Difference (ICLD) and Inter-channel Time Difference (ICTD).
  • ICLD Inter-channel Level Difference
  • ICTD Inter-channel Time Difference
  • the present invention is directed to reproduction of a realistic audio signal by encoding/decoding a multi-channel audio signal using only a downmixed audio signal and a small amount of additional information.
  • the present invention is also directed to maximizing transmission efficiency by analyzing a per-channel sound source of a multi-channel audio signal, extracting a small amount of virtual source location information, and transmitting the extracted virtual source location information together with a downmixed audio signal.
  • One aspect of the present invention provides an apparatus for encoding a multi ⁇ channel audio signal, the apparatus including: a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location information from the framed audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.
  • the source location information estimator includes a time-to-frequency converter for converting the framed audio signal into a spectrum; a separator for separating per-band spectrums; an energy vector detector for detecting per-channel energy vectors from the corresponding per- band spectrum; and a VSLI estimator for estimating virtual source location information (VSLI) using the detected per-channel energy vector detected by the energy vector detector.
  • a time-to-frequency converter for converting the framed audio signal into a spectrum
  • a separator for separating per-band spectrums
  • an energy vector detector for detecting per-channel energy vectors from the corresponding per- band spectrum
  • a VSLI estimator for estimating virtual source location information (VSLI) using the detected per-channel energy vector detected by the energy vector detector.
  • Another aspect of the present invention provides an apparatus for decoding a multi ⁇ channel audio signal, the apparatus including: means for receiving the multi-channel audio signal; a signal distributor for separating the received multi-channel audio signal into an encoded downmixed audio signal and a quantized virtual source location vector signal; means for decoding the encoded downmixed audio signal; means for converting the decoded downmixed audio signal into a frequency axis signal; a VSLI extractor for extracting per-band VSLI from the quantized virtual source location vector signal; a channel gain calculator for calculating per-band channel gains using the extracted per- band VSLI; means for synthesizing a multi-channel audio signal spectrum using the converted frequency axis signal and the calculated per-band channel gains; and means for generating a multi-channel audio signal from the synthesized multi-channel spectrum.
  • Yet another aspect of the present invention provides a method of encoding a multi ⁇ channel audio signal, including the steps of: converting the multi-channel audio signal into a framed audio signal; downmixing the framed audio signal; encoding the downmixed audio signal; estimating source location information from the framed audio signal; quantizing the estimated source location information; and multiplexing the encoded downmixed audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.
  • Still another aspect of the present invention provides a method of decoding a multi ⁇ channel audio signal, including the steps of: receiving the multi-channel audio signal; separating the received multi-channel audio signal into an encoded downmixed audio signal and a quantized virtual source location vector signal; decoding the encoded downmixed audio signal; converting the decoded downmixed audio signal into a frequency axis signal; analyzing the quantized virtual source location vector signal and extracting per-band VSLI therefrom; calculating per-band channel gains from the extracted per-band VSLI; synthesizing a multi-channel audio signal spectrum using the converted frequency axis signal and the calculated per-band channel gains; and producing a multi-channel audio signal from the synthesized multi-channel spectrum.
  • FlG. 1 is a block diagram of an apparatus for encoding a multi-channel audio signal according to an exemplary embodiment of the present invention
  • FlG. 2 is a conceptual diagram of a time-to-frequency lattice using an Equivalent
  • FIG. 3 is a conceptual diagram of source location vectors estimated according to the preset invention, in the case where a downmixed multi-channel audio signal is monophonic;
  • FlG. 4 is a conceptual diagram of source location vectors estimated according to the preset invention, in the case where a downmixed multi-channel audio signal is stereophonic;
  • FlG. 5 is a conceptual diagram illustrating a process of estimating virtual source location information according to an exemplary embodiment of the present invention
  • FlG. 6 shows an example of per-channel energy vectors when 5.1 channel speakers are used
  • FlG. 7 is a conceptual diagram illustrating a process of estimating a Left Half-plane
  • LHV Left Half-plane Vector
  • RHV Right Half-plane Vector
  • FlG. 8 is a conceptual diagram illustrating a process of estimating a Left Subsequent Vector (LSV) and a Right Subsequent Vector (RSV) according to the present invention
  • FlG. 9 is a conceptual diagram illustrating a process of estimating a Global Vector
  • FlG. 10 illustrates azimuth angles, each of which represents the corresponding virtual source location information according to the present invention
  • FlG. 11 is a block diagram of an apparatus for decoding an encoded multi-channel audio signal according to an exemplary embodiment of the present invention.
  • FlG. 12 is a block diagram illustrating a process of calculating per-channel gains of a downmixed audio signal using Virtual Source Location Information (VSLI) according to an exemplary embodiment of the present invention.
  • VSLI Virtual Source Location Information
  • FIG. 1 is a block diagram of an apparatus for encoding a multi-channel audio signal according to an exemplary embodiment of the present invention.
  • the multi-channel audio signal encoding apparatus includes a frame converter 100, a downmixer 110, an Advanced Audio Coding (AAC) encoder 120, a multiplexer 130, a quantizer 140, and a Virtual Source Location Information (VSLI) analyzer 150.
  • AAC Advanced Audio Coding
  • VSLI Virtual Source Location Information
  • the frame converter 100 frames the multi-channel audio signal, using a window function such as a sine window, to process the multi-channel audio signal in each block.
  • the downmixer 110 receives the framed multi-channel audio signal from the frame converter 100 and downmixes it into a monophonic signal or a stereophonic signal.
  • the AAC encoder 120 compresses the downmixed audio signal received from the downmixer 110, to generate an AAC encoded signal. It then transmits the AAC encoded signal to the multiplexer 130.
  • the VSLI analyzer 150 extracts Virtual Source Location Information (VSLI) from the framed audio signal.
  • the VSLI analyzer 150 may include a time- to-frequency converter 151, an Equivalent Rectangular Bandwidth (ERB) filter bank 152, an energy vector detector 153, and a location estimator 154.
  • ERP Equivalent Rectangular Bandwidth
  • the time-to-frequency converter 151 performs a plurality of Fast Fourier
  • FFTs Transforms
  • the ERB filter bank 152 divides the converted frequency domain signal (spectrum) into per-band spectrums (for example, 20 bands).
  • FlG. 2 is a conceptual diagram of a time-to-frequency lattice using the ERB filter bank 152.
  • the energy vector extractor 153 estimates per-channel energy vectors from the cor ⁇ responding per-band spectrum.
  • the location estimator 154 estimates virtual source location information (VSLI) using the per-channel energy vectors estimated by the energy vector extractor 153.
  • the VSLI may be represented using azimuth angles between the source location vectors and a center channel.
  • the VSLI estimated by the location estimator 154 can vary depending on whether the downmixed audio signal is monophonic or stereophonic.
  • FIG. 3 is a conceptual diagram illustrating the source location vectors estimated according to the present invention, in the case where the downmixed audio signal is monophonic.
  • the source location vectors estimated from the downmixed monophonic signal include a Left Half-plane Vector (LHV), a Right Half- plane Vector (RHV), a Left Subsequent Vector (LSV), a Right Subsequent Vector (RSV), and a Global Vector (GV).
  • LHV Left Half-plane Vector
  • RHV Right Half- plane Vector
  • LSV Left Subsequent Vector
  • RSV Right Subsequent Vector
  • GV Global Vector
  • FIG. 4 is a conceptual diagram illustrating the source location vectors estimated according to the present invention, in the case where the downmixed multi-channel audio signal is stereophonic.
  • the source location vectors estimated from the downmixed monophonic signal include the LHV, the RHV, the LSV, and the RSV, but not the GV.
  • the quantizer 140 quantizes the VSLI (azimuth angles) received from the VSLI analyzer 150 and transmits the quantized VSLI signal to the multiplexer 130.
  • the multiplexer 130 receives the AAC encoded signal from the AAC encoder 120 and the quantized VSLI signal from the quantizer 140 and multiplexes them to generate an encoded multi-channel audio signal (i.e., the AAC encoded signal +the VSLI signal).
  • FIG. 5 is a conceptual diagram illustrating a process of estimating the VSLI according to an exemplary embodiment of the present invention.
  • the input multi-channel audio signal is comprised of five channels including center (C), front left (L), front right (R), left subsequent (LS), and right subsequent (RS)
  • the input signal is converted into the frequency axis signal through the plurality of FFTs and divided into N number of frequency bands (BAND 1, BAND 2,..., and BAND N) in the ERB filter bank 152.
  • the per-channel energy vectors may be detected from the power of each of the five channels for each band (for example, Cl PWR, Ll PWR, Rl PWR, LSI PWR, and RSl PWR).
  • CPP Constant Power Panning
  • the source location vectors may be estimated from the detected per-channel energy vectors and the azimuth angles between the source location vectors and the center channel, which represent VSLI, may be estimated.
  • FlG. 6 to 9 illustrate detailed processes of estimating the VSLI according to the present invention.
  • the per-channel energy vectors estimated using the energy vector estimator are a center channel energy vector (C), a front left channel energy vector (L), a left subsequent channel energy vector (LS), a front right channel energy vector (R), and a right subsequent channel energy vector (RS).
  • the LHV is estimated using the front left channel energy vector (L) and the left subsequent channel energy vector (LS)
  • the RHV is estimated using the front right channel energy vector (R) and the right subsequent channel energy vector (RS) (Refer to HG. 7).
  • the LSV and RSV may be estimated using the LHV, the RHV, and the center channel energy vector (C) (Refer to FlG. 8).
  • the gain of each channel can be calculated using only the LHV, RHV, LSV, and RSV.
  • the GV can be calculated using the LSV and RSV (Refer to HG. 9).
  • the magnitude of the GV is set to the magnitude of the downmixed audio signal.
  • the source location vectors extracted using the above method may be expressed using the azimuth angles between themselves and the center channel.
  • FlG. 10 il ⁇ lustrates the azimuth angles of the source location vectors extracted by the processes shown in FIGS. 6 to 9.
  • the VSLI may be expressed using five azimuth angles, which include a Left Half-plane vector angle (LHa), a Right Half-plane vector angle (RHa), a Left Subsequent vector angle (LSa), and a Right Subsequent vector angle (RSa), and further include a Global vector angle (Ga) in the case where the downmixed audio signal is monophonic. Since each value has a limited dynamic range, quantization can be performed using fewer bits than Inter-Channel Level Difference (ICLD).
  • ICLD Inter-Channel Level Difference
  • a linear quantization method in which quantization is performed in uniform intervals or a nonlinear quantization method in which quantization is performed in non-uniform intervals may be used.
  • the linear quantization method is based on Equation
  • ⁇ represents the magnitude of an angle to be quantized and the cor ⁇ responding quantization index can be obtained from quantization level Q.
  • ⁇ represents the maximal variance level of each angle. i,max
  • ⁇ 180° ⁇ and ⁇ equal 15° and ⁇ and ⁇ l,max 2,max 3, max 4,max 5, max equal 55°.
  • a maximal variance interval of each angle magnitude is limited, and therefore more effective and higher resolution quantization can be provided.
  • the Ga has a generation frequency with a roughly symmetrical distribution centered on a center speaker. In other words, since the Ga varies evenly about the center speaker, it can be assumed that the generation distribution has an average expectation value of 0°. Accordingly, for the Ga, a more effective quantization level can be obtained when quantization is performed using the nonlinear quantization method.
  • the nonlinear quantization method is performed in a general m-law scheme, and m value can be determined depending on a resolution of the quantization level. For example, when the resolution is low, a relatively large m value may be used (15 ⁇ ⁇ 255), and when the resolution is high, a smaller m value (0 ⁇ 5) may be used to perform the nonlinear quantization.
  • FIG. 11 is a block diagram illustrating an apparatus for decoding an encoded multi ⁇ channel audio signal according to an exemplary embodiment of the present invention.
  • the multi-channel audio signal decoding apparatus includes a signal distributor 1110, an AAC decoder 1120, a time-to-frequency converter 1130, an inverse quantizer 1140, a per-band channel gain distributor 1150, a multi-channel spectrum synthesizer 1160, and a frequency-to-time converter 1170.
  • the signal distributor 1110 separates the encoded multi-channel audio signal back into the AAC encoded signal and the VLSI encoded signal, respectively.
  • the AAC decoder 1120 converts the AAC encoded signal back into the downmixed audio signal (monophonic or stereophonic signal).
  • the converted downmixed audio signal can be used to produce monophonic or stereophonic sound.
  • the time-to-frequency converter 1130 converts the downmixed audio signal into a frequency axis signal and transmits it to the multi-channel spectrum synthesizer 1160.
  • the inverse quantizer 1140 receives the separated VSLI encoded signal from the signal distributor 1110 and produces per-band source location vector information from the received VSLI encoded signal.
  • the VSLI includes azimuth angle information (for example, LHa, RHa, LSa, RSa, and Ga in the case where the downmixed audio signal is monophonic), each of which represents the corresponding per-band source location vector.
  • the source location vector is produced from the VSLI.
  • the per-band channel gain distributor 1150 calculates the gain per channel using the per-band VSLI signal converted by the inverse quantizer 1140, and transmits the calculated gain to the multi-channel spectrum synthesizer 1160.
  • the multi-channel spectrum synthesizer 1160 receives a spectrum of the downmixed audio signal from the time-to-frequency converter 1130, separates the received spectrum into per-band spectrums using the ERB filter bank, and restores the spectrum of the multi-channel signal using per-band channel gains output from the per- band channel gain distributor 1150.
  • the frequency-to-time converter 1170 (for example, IFFF) converts the spectrum of the restored multi-channel signal into a time axis signal to generate the multi-channel audio signal.
  • FIG. 12 is a block diagram illustrating a process of calculating the per-channel gain of the downmixed audio signal using the VSLI according to an exemplary embodiment of the present invention.
  • the downmixed audio signal is monophonic is illustrated.
  • block 1210 is omitted.
  • magnitudes of the LSV and the RSV are calculated using the magnitude of the downmixed monophonic signal, which is the magnitude of the GV, and the angle (Ga) of the GV.
  • magnitudes of the LHV and the first gain of the center channel (C) are calculated using the magnitude and angle (LSa) of the LSV (Block 1220).
  • the gain of the center channel (C) is obtained by summing the first gain and the second gain calculated in the above process (block 1240).
  • gains of the front left channel (L) and the left subsequent channel (LS) are calculated using the magnitude of the LHV and the corresponding angle (LHa) (block 1250), and gains of the front right channel (R) and the right subsequent channel (RS) are calculated using the magnitude of the RHV and the corresponding angle (RHa) (block 1260). According to the above processes, the gains of all channels can be calculated.
  • a multi-channel audio signal can be more ef ⁇ fectively encoded/decoded using virtual source location information, and more realistic audio signal reproduction in a multi-channel environment can be realized.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Multimedia (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Mathematical Physics (AREA)
  • Computational Linguistics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Stereophonic System (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

Provided is a method and apparatus for encoding/decoding a multi-channel audio signal. The apparatus for encoding a multi-channel audio signal includes a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location information from the framed multi-channel audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.

Description

Description
METHOD AND APPARATUS FOR ENCODING AND
DECODING MULTI-CHANNEL AUDIO SIGNAL USING
VIRTUAL SOURCE LOCATION INFORMATION
Background Art
[ 1 ] 1. Field of the Invention
[2] The present invention relates to a method and apparatus for encoding/decoding a multi-channel audio signal, and more particularly, to a method and apparatus for ef¬ fectively encoding/decoding a multi-channel audio signal using Virtual Sound Location Information (VLSI).
[3] 2. Description of Related Art
[4] Throughout the later half of the 1990s, Moving Picture Experts Group (MPEG) has performed research on compressing a multi-channel audio signal. Owing to the remarkable increase in multi-channel contents, increased demand for multi-channel contents, and increased need for a multi-channel audio services in a broadcasting com¬ munications environment, research on the multi-channel audio compression technology has been stepped up.
[5] As a result, multi-channel audio compression technology such as MPEG-2
Backward Compatibility (BC), MPEG-2 Advanced Audio Coding (AAC), and MPEG- 4 AAC, has been standardized in the MPEG. Also, multi-channel audio compression technology, such as AC-3 and Digital Theater System (DTS), has been com¬ mercialized.
[6] In recent years, innovative multi-channel audio signal compression method such as typical Binaural Cue Coding (BCC) has been actively researched (C. Faller, 2002 & 2003; F. Baumgarte, 2001 & 2002). The goal of such research is the transfer of more realistic audio data.
[7] BCC is technology for effectively compressing a multi-channel audio signal that has been developed on a basis of the fact that people can acoustically perceive space due to a binaural effect. BCC is based on the fact that a pair of ears perceives a location of a specific sound source using interaural level differences and/or interaural time differences.
[8] Accordingly, in BCC, a multi-channel audio signal is downmixed to a monophonic or stereophonic signal and channel information is represented by binaural cue parameters such as Inter-channel Level Difference (ICLD) and Inter-channel Time Difference (ICTD).
[9] However, there is a drawback in that a large number of bits are required to quantize the channel information such as ICLD and ICTD, and consequently, a wide bandwidth is required in transmitting the channel information.
[10] SUMMARY OF THE INVENTION
[11] The present invention is directed to reproduction of a realistic audio signal by encoding/decoding a multi-channel audio signal using only a downmixed audio signal and a small amount of additional information.
[12] The present invention is also directed to maximizing transmission efficiency by analyzing a per-channel sound source of a multi-channel audio signal, extracting a small amount of virtual source location information, and transmitting the extracted virtual source location information together with a downmixed audio signal.
[13] One aspect of the present invention provides an apparatus for encoding a multi¬ channel audio signal, the apparatus including: a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location information from the framed audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal. The source location information estimator includes a time-to-frequency converter for converting the framed audio signal into a spectrum; a separator for separating per-band spectrums; an energy vector detector for detecting per-channel energy vectors from the corresponding per- band spectrum; and a VSLI estimator for estimating virtual source location information (VSLI) using the detected per-channel energy vector detected by the energy vector detector.
[14] Another aspect of the present invention provides an apparatus for decoding a multi¬ channel audio signal, the apparatus including: means for receiving the multi-channel audio signal; a signal distributor for separating the received multi-channel audio signal into an encoded downmixed audio signal and a quantized virtual source location vector signal; means for decoding the encoded downmixed audio signal; means for converting the decoded downmixed audio signal into a frequency axis signal; a VSLI extractor for extracting per-band VSLI from the quantized virtual source location vector signal; a channel gain calculator for calculating per-band channel gains using the extracted per- band VSLI; means for synthesizing a multi-channel audio signal spectrum using the converted frequency axis signal and the calculated per-band channel gains; and means for generating a multi-channel audio signal from the synthesized multi-channel spectrum.
[15] Yet another aspect of the present invention provides a method of encoding a multi¬ channel audio signal, including the steps of: converting the multi-channel audio signal into a framed audio signal; downmixing the framed audio signal; encoding the downmixed audio signal; estimating source location information from the framed audio signal; quantizing the estimated source location information; and multiplexing the encoded downmixed audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.
[16] Still another aspect of the present invention provides a method of decoding a multi¬ channel audio signal, including the steps of: receiving the multi-channel audio signal; separating the received multi-channel audio signal into an encoded downmixed audio signal and a quantized virtual source location vector signal; decoding the encoded downmixed audio signal; converting the decoded downmixed audio signal into a frequency axis signal; analyzing the quantized virtual source location vector signal and extracting per-band VSLI therefrom; calculating per-band channel gains from the extracted per-band VSLI; synthesizing a multi-channel audio signal spectrum using the converted frequency axis signal and the calculated per-band channel gains; and producing a multi-channel audio signal from the synthesized multi-channel spectrum. Brief Description of the Drawings
[17] The above and other features and advantages of the present invention will become more apparent to those of ordinary skill in the art by describing in detail exemplary embodiments of the invention with reference to the attached drawings in which:
[18] FlG. 1 is a block diagram of an apparatus for encoding a multi-channel audio signal according to an exemplary embodiment of the present invention;
[19] FlG. 2 is a conceptual diagram of a time-to-frequency lattice using an Equivalent
Rectangular Bandwidth (ERB) filter bank;
[20] FIG. 3 is a conceptual diagram of source location vectors estimated according to the preset invention, in the case where a downmixed multi-channel audio signal is monophonic;
[21] FlG. 4 is a conceptual diagram of source location vectors estimated according to the preset invention, in the case where a downmixed multi-channel audio signal is stereophonic;
[22] FlG. 5 is a conceptual diagram illustrating a process of estimating virtual source location information according to an exemplary embodiment of the present invention;
[23] FlG. 6 shows an example of per-channel energy vectors when 5.1 channel speakers are used;
[24] FlG. 7 is a conceptual diagram illustrating a process of estimating a Left Half-plane
Vector (LHV) and a Right Half-plane Vector (RHV) according to the present invention;
[25] FlG. 8 is a conceptual diagram illustrating a process of estimating a Left Subsequent Vector (LSV) and a Right Subsequent Vector (RSV) according to the present invention;
[26] FlG. 9 is a conceptual diagram illustrating a process of estimating a Global Vector
(GV) according to the present invention;
[27] FlG. 10 illustrates azimuth angles, each of which represents the corresponding virtual source location information according to the present invention;
[28] FlG. 11 is a block diagram of an apparatus for decoding an encoded multi-channel audio signal according to an exemplary embodiment of the present invention; and
[29] FlG. 12 is a block diagram illustrating a process of calculating per-channel gains of a downmixed audio signal using Virtual Source Location Information (VSLI) according to an exemplary embodiment of the present invention.
[30]
[31] DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
[32] The present invention will now be described more fully with reference to the ac¬ companying drawings, in which exemplary embodiments of the invention are shown. This invention may, however, be embodied in different forms and should not be construed as limited to the exemplary embodiments set forth herein. Rather, these exemplary embodiments are provided so that this disclosure will be thorough and complete, and will fully convey the scope of the invention to those skilled in the art.
[33] FIG. 1 is a block diagram of an apparatus for encoding a multi-channel audio signal according to an exemplary embodiment of the present invention. As shown in FIG. 1, the multi-channel audio signal encoding apparatus includes a frame converter 100, a downmixer 110, an Advanced Audio Coding (AAC) encoder 120, a multiplexer 130, a quantizer 140, and a Virtual Source Location Information (VSLI) analyzer 150.
[34] The frame converter 100 frames the multi-channel audio signal, using a window function such as a sine window, to process the multi-channel audio signal in each block. The downmixer 110 receives the framed multi-channel audio signal from the frame converter 100 and downmixes it into a monophonic signal or a stereophonic signal. The AAC encoder 120 compresses the downmixed audio signal received from the downmixer 110, to generate an AAC encoded signal. It then transmits the AAC encoded signal to the multiplexer 130.
[35] The VSLI analyzer 150 extracts Virtual Source Location Information (VSLI) from the framed audio signal. Specifically, the VSLI analyzer 150 may include a time- to-frequency converter 151, an Equivalent Rectangular Bandwidth (ERB) filter bank 152, an energy vector detector 153, and a location estimator 154.
[36] The time-to-frequency converter 151 performs a plurality of Fast Fourier
Transforms (FFTs) to convert the framed audio signal into a frequency domain signal. The ERB filter bank 152 divides the converted frequency domain signal (spectrum) into per-band spectrums (for example, 20 bands). FlG. 2 is a conceptual diagram of a time-to-frequency lattice using the ERB filter bank 152.
[37] The energy vector extractor 153 estimates per-channel energy vectors from the cor¬ responding per-band spectrum.
[38] The location estimator 154 estimates virtual source location information (VSLI) using the per-channel energy vectors estimated by the energy vector extractor 153. In one exemplary embodiment, the VSLI may be represented using azimuth angles between the source location vectors and a center channel. As described later, the VSLI estimated by the location estimator 154 can vary depending on whether the downmixed audio signal is monophonic or stereophonic.
[39] FIG. 3 is a conceptual diagram illustrating the source location vectors estimated according to the present invention, in the case where the downmixed audio signal is monophonic. As shown in FIG. 3, the source location vectors estimated from the downmixed monophonic signal include a Left Half-plane Vector (LHV), a Right Half- plane Vector (RHV), a Left Subsequent Vector (LSV), a Right Subsequent Vector (RSV), and a Global Vector (GV). In the case where the downmixed multi-channel audio signal is monophonic, since it is not known whether channel gain is higher on the left or on the right, the GV is required.
[40] FIG. 4 is a conceptual diagram illustrating the source location vectors estimated according to the present invention, in the case where the downmixed multi-channel audio signal is stereophonic. As shown in FIG. 4, the source location vectors estimated from the downmixed monophonic signal include the LHV, the RHV, the LSV, and the RSV, but not the GV.
[41] Referring again to FIG. 1, the quantizer 140 quantizes the VSLI (azimuth angles) received from the VSLI analyzer 150 and transmits the quantized VSLI signal to the multiplexer 130. The multiplexer 130 receives the AAC encoded signal from the AAC encoder 120 and the quantized VSLI signal from the quantizer 140 and multiplexes them to generate an encoded multi-channel audio signal (i.e., the AAC encoded signal +the VSLI signal).
[42] FIG. 5 is a conceptual diagram illustrating a process of estimating the VSLI according to an exemplary embodiment of the present invention. As shown in FIG. 5, in the case where the input multi-channel audio signal is comprised of five channels including center (C), front left (L), front right (R), left subsequent (LS), and right subsequent (RS), the input signal is converted into the frequency axis signal through the plurality of FFTs and divided into N number of frequency bands (BAND 1, BAND 2,..., and BAND N) in the ERB filter bank 152.
[43] Next, the per-channel energy vectors may be detected from the power of each of the five channels for each band (for example, Cl PWR, Ll PWR, Rl PWR, LSI PWR, and RSl PWR). Using Constant Power Panning (CPP) in which the magnitudes of signals of neighboring channels are adjusted for sound localization, the source location vectors may be estimated from the detected per-channel energy vectors and the azimuth angles between the source location vectors and the center channel, which represent VSLI, may be estimated.
[44] FlG. 6 to 9 illustrate detailed processes of estimating the VSLI according to the present invention. In detail, as shown in FlG. 6, it is assumed that the per-channel energy vectors estimated using the energy vector estimator are a center channel energy vector (C), a front left channel energy vector (L), a left subsequent channel energy vector (LS), a front right channel energy vector (R), and a right subsequent channel energy vector (RS). The LHV is estimated using the front left channel energy vector (L) and the left subsequent channel energy vector (LS), and the RHV is estimated using the front right channel energy vector (R) and the right subsequent channel energy vector (RS) (Refer to HG. 7).
[45] The LSV and RSV may be estimated using the LHV, the RHV, and the center channel energy vector (C) (Refer to FlG. 8).
[46] In the case where the downmixed audio signal is stereophonic, the gain of each channel can be calculated using only the LHV, RHV, LSV, and RSV. However, in the case where the downmixed audio signal is the monophonic signal, it is not known whether the channel gain is higher on the left or on the right, and therefore the GV is required. The GV can be calculated using the LSV and RSV (Refer to HG. 9). The magnitude of the GV is set to the magnitude of the downmixed audio signal.
[47] The source location vectors extracted using the above method may be expressed using the azimuth angles between themselves and the center channel. FlG. 10 il¬ lustrates the azimuth angles of the source location vectors extracted by the processes shown in FIGS. 6 to 9. As shown, the VSLI may be expressed using five azimuth angles, which include a Left Half-plane vector angle (LHa), a Right Half-plane vector angle (RHa), a Left Subsequent vector angle (LSa), and a Right Subsequent vector angle (RSa), and further include a Global vector angle (Ga) in the case where the downmixed audio signal is monophonic. Since each value has a limited dynamic range, quantization can be performed using fewer bits than Inter-Channel Level Difference (ICLD).
[48] To quantize the VSLI information, a linear quantization method in which quantization is performed in uniform intervals or a nonlinear quantization method in which quantization is performed in non-uniform intervals may be used.
[49] In one exemplary embodiment, the linear quantization method is based on Equation
1 below:
[50] [Equation 1] [51]
[52] , wherein "θ"represents the magnitude of an angle to be quantized and the cor¬ responding quantization index can be obtained from quantization level Q. "i" represents angle index (Ga:i=l, RHa:i=2, LHa:i=3, LSa:i=4, RSa:i=5) and "b" represents sub-band index. "Δθ represents the maximal variance level of each angle. i,max
For example, Δθ equals 180° Δθ and ΔΘ equal 15° and Δθ and ΔΘ l,max 2,max 3, max 4,max 5, max equal 55°. As mentioned above, a maximal variance interval of each angle magnitude is limited, and therefore more effective and higher resolution quantization can be provided.
[53] In general, statistical information on generation frequency with respect to the RHa,
LHa, LSa, and RSa is inconclusive. However, the Ga has a generation frequency with a roughly symmetrical distribution centered on a center speaker. In other words, since the Ga varies evenly about the center speaker, it can be assumed that the generation distribution has an average expectation value of 0°. Accordingly, for the Ga, a more effective quantization level can be obtained when quantization is performed using the nonlinear quantization method.
[54] Typically, the nonlinear quantization method is performed in a general m-law scheme, and m value can be determined depending on a resolution of the quantization level. For example, when the resolution is low, a relatively large m value may be used (15< μ< 255), and when the resolution is high, a smaller m value (0<μ<5) may be used to perform the nonlinear quantization.
[55] FIG. 11 is a block diagram illustrating an apparatus for decoding an encoded multi¬ channel audio signal according to an exemplary embodiment of the present invention. As shown, the multi-channel audio signal decoding apparatus includes a signal distributor 1110, an AAC decoder 1120, a time-to-frequency converter 1130, an inverse quantizer 1140, a per-band channel gain distributor 1150, a multi-channel spectrum synthesizer 1160, and a frequency-to-time converter 1170.
[56] The signal distributor 1110 separates the encoded multi-channel audio signal back into the AAC encoded signal and the VLSI encoded signal, respectively. The AAC decoder 1120 converts the AAC encoded signal back into the downmixed audio signal (monophonic or stereophonic signal). The converted downmixed audio signal can be used to produce monophonic or stereophonic sound. The time-to-frequency converter 1130 converts the downmixed audio signal into a frequency axis signal and transmits it to the multi-channel spectrum synthesizer 1160. [57] The inverse quantizer 1140 receives the separated VSLI encoded signal from the signal distributor 1110 and produces per-band source location vector information from the received VSLI encoded signal. In the encoding process, as described above, the VSLI includes azimuth angle information (for example, LHa, RHa, LSa, RSa, and Ga in the case where the downmixed audio signal is monophonic), each of which represents the corresponding per-band source location vector. The source location vector is produced from the VSLI.
[58] The per-band channel gain distributor 1150 calculates the gain per channel using the per-band VSLI signal converted by the inverse quantizer 1140, and transmits the calculated gain to the multi-channel spectrum synthesizer 1160.
[59] The multi-channel spectrum synthesizer 1160 receives a spectrum of the downmixed audio signal from the time-to-frequency converter 1130, separates the received spectrum into per-band spectrums using the ERB filter bank, and restores the spectrum of the multi-channel signal using per-band channel gains output from the per- band channel gain distributor 1150. The frequency-to-time converter 1170 (for example, IFFF) converts the spectrum of the restored multi-channel signal into a time axis signal to generate the multi-channel audio signal.
[60] FIG. 12 is a block diagram illustrating a process of calculating the per-channel gain of the downmixed audio signal using the VSLI according to an exemplary embodiment of the present invention. Here, the case in which the downmixed audio signal is monophonic is illustrated. In the case where the downmixed audio signal is stereophonic, block 1210 is omitted.
[61] In block 1210, magnitudes of the LSV and the RSV are calculated using the magnitude of the downmixed monophonic signal, which is the magnitude of the GV, and the angle (Ga) of the GV. Next, magnitudes of the LHV and the first gain of the center channel (C) are calculated using the magnitude and angle (LSa) of the LSV (Block 1220). The gain of the center channel (C) is obtained by summing the first gain and the second gain calculated in the above process (block 1240).
[62] Last, gains of the front left channel (L) and the left subsequent channel (LS) are calculated using the magnitude of the LHV and the corresponding angle (LHa) (block 1250), and gains of the front right channel (R) and the right subsequent channel (RS) are calculated using the magnitude of the RHV and the corresponding angle (RHa) (block 1260). According to the above processes, the gains of all channels can be calculated.
[63] According to the present invention, a multi-channel audio signal can be more ef¬ fectively encoded/decoded using virtual source location information, and more realistic audio signal reproduction in a multi-channel environment can be realized.
[64] While the invention has been shown and described with reference to certain exemplary embodiments thereof, it will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the spirit and scope of the invention as defined by the appended claims and their equivalents.
[65]
[66]
[67]
[68]

Claims

Claims
[1] An apparatus for encoding a multi-channel audio signal, the apparatus comprising: a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location in¬ formation from the framed audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.
[2] The apparatus according to claim 1, wherein said downmixing means downmixes the framed audio signal as either one of monophonic or stereophonic signal.
[3] The apparatus according to claim 1, wherein when the downmixed audio signal is the monophonic signal, the source location information estimator estimates an LHV (Left Half-plane Vector), an RHV (Right Half-plane Vector), an LSV (Left Subsequent Vector), an RSV (Right Subsequent Vector), and a GV (Global Vector).
[4] The apparatus according to claim 1, wherein when the downmixed audio signal is the stereophonic signal, the source location information estimator estimates an LHV (Left Half-plane Vector), an RHV (Right Half-plane Vector), an LSV (Left Subsequent Vector), and an RSV (Right Subsequent Vector).
[5] The apparatus according to claim 1, wherein said source location information estimator comprises: a time-to-frequency converter for converting the framed audio signal into a spectrum; a separator for separating per-band spectrums; an energy vector detector for detecting per-channel energy vectors from the cor¬ responding per-band spectrum; and a VSLI estimator for estimating virtual source location information (VSLI) using the detected per-channel energy vector detected by the energy vector detector.
[6] The apparatus according to claim 5, wherein said time-to-frequency converter converts the framed audio signal into the spectrum using a plurality of FFTs (Fast Fourier Transforms).
[7] The apparatus according to claim 5, wherein the separator separates the spectrum using an ERB (Equivalent Rectangular Bandwidth) filter bank. [8] The apparatus according to claim 5, wherein the detected per-channel energy vector includes a center channel energy vector (C), a front left channel energy vector (L), a left subsequent channel energy vector (LS), a front right channel energy vector (R), and a right subsequent channel energy vector (RS).
[9] The apparatus according to claim 5, wherein the VSLI is represented as azimuth angle information based on a center channel, and the azimuth angle information includes an LHa (Left Half-plane vector angle), an RHa (Right Half-plane vector angle), an LSa (Left Subsequent vector angle), and an RSa (Right Subsequent vector angle).
[10] The apparatus according to claim 9, wherein when the downmixed audio signal is the monophonic signal, the azimuth angle information further includes a Ga (Global vector angle).
[11] An apparatus for decoding a multi-channel audio signal, the apparatus comprising: means for receiving the multi-channel audio signal; a signal distributor for separating the received multi-channel audio signal into an encoded downmixed audio signal and a quantized virtual source location vector signal; means for decoding the encoded downmixed audio signal; means for converting the decoded downmixed audio signal into a frequency axis signal; a VSLI extractor for extracting per-band VSLI from the quantized virtual source location vector signal; a channel gain calculator for calculating per-band channel gains using the extracted per-band VSLI; means for synthesizing a multi-channel audio signal spectrum using the converted frequency axis signal and the calculated per-band channel gains; and means for generating a multi-channel audio signal from the synthesized multi¬ channel spectrum.
[12] The apparatus according to claim 11, wherein the VSLI extractor extracts per- band virtual source azimuth angle information from the quantized virtual source location vector signal and produces VSLI from the extracted azimuth angle in¬ formation.
[13] The apparatus according to claim 12, wherein the virtual source azimuth angle information includes an LHa (Left Half-plane vector angle), an RHa (Right Half- plane vector angle), an LSa (Left Subsequent vector angle), and an RSa (Right Subsequent vector angle) for each band, and the produced VSLI vectors include an LHV (Left Half-plane Vector), an RHV (Right Half-plane Vector), an LSV (Left Subsequent Vector), and an RSV (Right Subsequent Vector).
[14] The apparatus according to claim 13, wherein when the encoded downmixed audio signal is monophonic, and the virtual source azimuth angle information further includes a Ga (Global vector angle), and a GV (Global Vector) is produced from the Ga.
[15] A method of encoding a multi-channel audio signal, comprising the steps of: converting the multi-channel audio signal into a framed audio signal; downmixing the framed audio signal; encoding the downmixed audio signal; estimating source location information from the framed audio signal; quantizing the estimated source location information; and multiplexing the encoded downmixed audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.
[16] The method according to claim 15, wherein the framed audio signal is downmixed into either one of a monophonic signal and a stereophonic signal.
[17] The method according to claim 15, wherein when the downmixed audio signal is the monophonic signal, the estimated source location information includes an LHV (Left Half-plane Vector), an RHV (Right Half-plane Vector), an LSV (Left Subsequent Vector), an RSV (Right Subsequent Vector), and a GV (Global Vector).
[18] The method according to claim 15, wherein when the downmixed audio signal is the stereophonic signal, the estimated source location information includes an LHV (Left Half-plane Vector), an RHV (Right Half-plane Vector), an LSV (Left Subsequent Vector), and an RSV (Right Subsequent Vector).
[19] The method according to claim 15, wherein the step of estimating the source location information comprises the steps of: converting the framed audio signal into a spectrum; separating the spectrum into per-band spectrums; detecting per-channel energy vectors from the per-band spectrums; and estimating VSLI using the detected per-channel energy vectors.
[20] The method according to claim 19, wherein the detected per-channel energy vectors include a center channel energy vector (C), a front left channel energy vector (L), a left subsequent channel energy vector (LS), a front right channel energy vector (R), and a right subsequent channel energy vector (RS).
[21] The method according to claim 19, wherein the step of estimating the VSLI comprises the steps of: estimating an LHV using the front left channel energy vector (L) and the left subsequent channel energy vector (LS); estimating an RHV using the front right channel energy vector (R) and the right subsequent channel energy vector (RS); estimating an LSV using the estimated LHV and the center channel energy vector (C); and estimating an RSV using the estimated RHV and the center channel energy vector (C).
[22] The method according to claim 21, wherein when the downmixed audio signal is the monophonic signal, the estimated VLSI further includes a GV, and the estimating of the VSLI further comprises the step of estimating the GV using the estimated LSV and RSV.
[23] The method according to claim 19, wherein when the downmixed audio signal is the stereophonic signal, the VSLI is expressed using an LHa, an RHa, an LSa, and an RSa based on a center channel.
[24] The method according to claim 19, wherein when the downmixed audio signal is the monophonic signal, the VSLI is expressed using a Ga, an LHa, an RHa, an LSa, and an RSa.
[25] A method of decoding a multi-channel audio signal, comprising the steps of: receiving the multi-channel audio signal; separating the received multi-channel audio signal into an encoded downmixed audio signal and a quantized virtual source location vector signal; decoding the encoded downmixed audio signal; converting the decoded downmixed audio signal into a frequency axis signal; analyzing the quantized virtual source location vector signal and extracting per- band VSLI therefrom; calculating per-band channel gains from the extracted per-band VSLI; synthesizing a multi-channel audio signal spectrum using the converted frequency axis signal and the calculated per-band channel gains; and producing a multi-channel audio signal from the synthesized multi-channel spectrum.
[26] The method according to claim 25, wherein said step of extracting the per-band
VSLI extracts per-band virtual source azimuth angle information from the quantized virtual source location vector signal, and VSLI is produced from the extracted azimuth angle information.
[27] The method according to claim 26, wherein the virtual source azimuth angle in¬ formation includes an LHa (Left Half-plane vector angle), an RHa (Right Half- plane vector angle), an LSa (Left Subsequent vector angle), and an RSa (Right Subsequent vector angle) , for each band, and the produced VSLI includes an LHV (Left Half-plane Vector), an RHV (Right Half-plane Vector), an LSV (Left Subsequent Vector), and an RSV (Right Subsequent Vector). [28] The method according to claim 27, wherein when the encoded downmixed audio signal is monophonic, the virtual source azimuth angle information further includes a Ga (Global vector angle), and a GV (Global Vector) is produced from the Ga. [29] The method according to claim 27, wherein said step of calculating the channel gain comprises, for each band, the steps of: calculating magnitudes of the LSV and the RSV using a magnitude of the downmixed audio signal; calculating a first gain of a center channel (C) and a magnitude of the LHV using the magnitude of the LSV and the LSa; calculating a second gain of a center channel (C) and a magnitude of the RHV using the magnitude of the RSV and the RSa; summing the first and second gains of the center channel (C) to produce a gain of the center channel (C); calculating gains of a front left channel (L) and a left subsequent channel (LS) using the magnitude of the LHV and the LHa; and calculating gains of a front right channel (R) and a right subsequent channel (RS) using the magnitude of the RHV and the RHa. [30] A computer-readable recording medium storing a computer program for performing the method for encoding a multi-channel audio signal according to any one of claims 15 to 24. [31] A computer-readable recording medium storing a computer program for performing the method for decoding a multi-channel audio signal according to any one of claims 25 to 29.
EP05774399A 2004-07-09 2005-07-08 Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information Not-in-force EP1779385B1 (en)

Applications Claiming Priority (4)

Application Number Priority Date Filing Date Title
KR20040053665 2004-07-09
KR20040081303 2004-10-12
KR1020050061425A KR100663729B1 (en) 2004-07-09 2005-07-07 Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information
PCT/KR2005/002213 WO2006006809A1 (en) 2004-07-09 2005-07-08 Method and apparatus for encoding and cecoding multi-channel audio signal using virtual source location information

Publications (3)

Publication Number Publication Date
EP1779385A1 EP1779385A1 (en) 2007-05-02
EP1779385A4 true EP1779385A4 (en) 2007-07-25
EP1779385B1 EP1779385B1 (en) 2010-09-22

Family

ID=35784122

Family Applications (1)

Application Number Title Priority Date Filing Date
EP05774399A Not-in-force EP1779385B1 (en) 2004-07-09 2005-07-08 Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information

Country Status (2)

Country Link
EP (1) EP1779385B1 (en)
WO (1) WO2006006809A1 (en)

Families Citing this family (10)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8180067B2 (en) 2006-04-28 2012-05-15 Harman International Industries, Incorporated System for selectively extracting components of an audio input signal
US7876904B2 (en) 2006-07-08 2011-01-25 Nokia Corporation Dynamic decoding of binaural audio signals
KR101111520B1 (en) 2006-12-07 2012-05-24 엘지전자 주식회사 A method an apparatus for processing an audio signal
WO2011044064A1 (en) * 2009-10-05 2011-04-14 Harman International Industries, Incorporated System for spatial extraction of audio signals
EP2541547A1 (en) 2011-06-30 2013-01-02 Thomson Licensing Method and apparatus for changing the relative positions of sound objects contained within a higher-order ambisonics representation
JP6174129B2 (en) 2012-05-18 2017-08-02 ドルビー ラボラトリーズ ライセンシング コーポレイション System for maintaining reversible dynamic range control information related to parametric audio coders
US10844689B1 (en) 2019-12-19 2020-11-24 Saudi Arabian Oil Company Downhole ultrasonic actuator system for mitigating lost circulation
WO2015105775A1 (en) * 2014-01-07 2015-07-16 Harman International Industries, Incorporated Signal quality-based enhancement and compensation of compressed audio signals
CA2999393C (en) * 2016-03-15 2020-10-27 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Apparatus, method or computer program for generating a sound field description
EP3297298B1 (en) * 2016-09-19 2020-05-06 A-Volute Method for reproducing spatially distributed sounds

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1999052326A1 (en) * 1998-04-07 1999-10-14 Ray Milton Dolby Low bit-rate spatial coding method and system
WO2003090208A1 (en) * 2002-04-22 2003-10-30 Koninklijke Philips Electronics N.V. pARAMETRIC REPRESENTATION OF SPATIAL AUDIO
WO2005101905A1 (en) * 2004-04-16 2005-10-27 Coding Technologies Ab Scheme for generating a parametric representation for low-bit rate applications

Family Cites Families (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6128597A (en) * 1996-05-03 2000-10-03 Lsi Logic Corporation Audio decoder with a reconfigurable downmixing/windowing pipeline and method therefor
US5946352A (en) * 1997-05-02 1999-08-31 Texas Instruments Incorporated Method and apparatus for downmixing decoded data streams in the frequency domain prior to conversion to the time domain
US7292901B2 (en) * 2002-06-24 2007-11-06 Agere Systems Inc. Hybrid multi-channel/cue coding/decoding of audio signals
US20030014243A1 (en) * 2001-07-09 2003-01-16 Lapicque Olivier D. System and method for virtual localization of audio signals

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1999052326A1 (en) * 1998-04-07 1999-10-14 Ray Milton Dolby Low bit-rate spatial coding method and system
WO2003090208A1 (en) * 2002-04-22 2003-10-30 Koninklijke Philips Electronics N.V. pARAMETRIC REPRESENTATION OF SPATIAL AUDIO
WO2005101905A1 (en) * 2004-04-16 2005-10-27 Coding Technologies Ab Scheme for generating a parametric representation for low-bit rate applications

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See also references of WO2006006809A1 *

Also Published As

Publication number Publication date
EP1779385B1 (en) 2010-09-22
WO2006006809A1 (en) 2006-01-19
EP1779385A1 (en) 2007-05-02

Similar Documents

Publication Publication Date Title
US7783495B2 (en) Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information
EP1779385B1 (en) Method and apparatus for encoding and decoding multi-channel audio signal using virtual source location information
EP2612322B1 (en) Method and device for decoding a multichannel audio signal
KR101117336B1 (en) Audio signal encoder and audio signal decoder
US9058803B2 (en) Multichannel audio stream compression
US8332229B2 (en) Low complexity MPEG encoding for surround sound recordings
US8706508B2 (en) Audio decoding apparatus and audio decoding method performing weighted addition on signals
US20240185869A1 (en) Combining spatial audio streams
EP4082010A1 (en) Combining of spatial audio parameters
CN112997248A (en) Encoding and associated decoding to determine spatial audio parameters
WO2021130404A1 (en) The merging of spatial audio parameters
US20210250717A1 (en) Spatial audio Capture, Transmission and Reproduction
US20240046939A1 (en) Quantizing spatial audio parameters
US20230335143A1 (en) Quantizing spatial audio parameters
CN116547749A (en) Quantization of audio parameters
US20240079014A1 (en) Transforming spatial audio parameters
WO2020201619A1 (en) Spatial audio representation and associated rendering
EP3948861A1 (en) Determination of the significance of spatial audio parameters and associated encoding
Jang et al. Sound source location cue coding system for compact representation of multi-channel audio

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

17P Request for examination filed

Effective date: 20070205

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HU IE IS IT LI LT LU LV MC NL PL PT RO SE SI SK TR

A4 Supplementary search report drawn up and despatched

Effective date: 20070621

RIC1 Information provided on ipc code assigned before grant

Ipc: G11B 20/10 20060101AFI20060216BHEP

Ipc: G10L 19/00 20060101ALI20070615BHEP

Ipc: H04S 3/00 20060101ALI20070615BHEP

RAP1 Party data changed (applicant data changed or rights of an application transferred)

Owner name: SEOUL NATIONAL UNIVERSITY INDUSTRY FOUNDATION

Owner name: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTIT

DAX Request for extension of the european patent (deleted)
17Q First examination report despatched

Effective date: 20080125

GRAP Despatch of communication of intention to grant a patent

Free format text: ORIGINAL CODE: EPIDOSNIGR1

GRAS Grant fee paid

Free format text: ORIGINAL CODE: EPIDOSNIGR3

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HU IE IS IT LI LT LU LV MC NL PL PT RO SE SI SK TR

REG Reference to a national code

Ref country code: GB

Ref legal event code: FG4D

REG Reference to a national code

Ref country code: CH

Ref legal event code: EP

REG Reference to a national code

Ref country code: IE

Ref legal event code: FG4D

REF Corresponds to:

Ref document number: 602005023738

Country of ref document: DE

Date of ref document: 20101104

Kind code of ref document: P

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: FI

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: AT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

REG Reference to a national code

Ref country code: NL

Ref legal event code: VDEP

Effective date: 20100922

LTIE Lt: invalidation of european patent or patent extension

Effective date: 20100922

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: SI

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: PL

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LV

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: SE

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: GR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20101223

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: CZ

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: RO

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: IT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: SK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: PT

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20110124

Ref country code: EE

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: IS

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20110122

Ref country code: NL

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: BE

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: ES

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20110102

PLBE No opposition filed within time limit

Free format text: ORIGINAL CODE: 0009261

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT

26N No opposition filed

Effective date: 20110623

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DK

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

REG Reference to a national code

Ref country code: DE

Ref legal event code: R097

Ref document number: 602005023738

Country of ref document: DE

Effective date: 20110623

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: MC

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20110731

REG Reference to a national code

Ref country code: CH

Ref legal event code: PL

GBPC Gb: european patent ceased through non-payment of renewal fee

Effective date: 20110708

REG Reference to a national code

Ref country code: FR

Ref legal event code: ST

Effective date: 20120330

REG Reference to a national code

Ref country code: IE

Ref legal event code: MM4A

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: FR

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20110801

Ref country code: CH

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20110731

Ref country code: LI

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20110731

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: GB

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20110708

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: IE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20110708

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: LU

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20110708

Ref country code: CY

Free format text: LAPSE BECAUSE OF EXPIRATION OF PROTECTION

Effective date: 20100922

REG Reference to a national code

Ref country code: DE

Ref legal event code: R082

Ref document number: 602005023738

Country of ref document: DE

Representative=s name: PATENTANWAELTE BETTEN & RESCH, DE

REG Reference to a national code

Ref country code: DE

Ref legal event code: R088

Ref document number: 602005023738

Country of ref document: DE

REG Reference to a national code

Ref country code: DE

Ref legal event code: R082

Ref document number: 602005023738

Country of ref document: DE

Representative=s name: BETTEN & RESCH PATENT- UND RECHTSANWAELTE PART, DE

Effective date: 20130717

Ref country code: DE

Ref legal event code: R081

Ref document number: 602005023738

Country of ref document: DE

Owner name: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH IN, KR

Free format text: FORMER OWNERS: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE, DAEJEON, KR; SEOUL NATIONAL UNIVERSITY INDUSTRY FOUNDATION, SEOUL/SOUL, KR

Effective date: 20130717

Ref country code: DE

Ref legal event code: R081

Ref document number: 602005023738

Country of ref document: DE

Owner name: SNU R&DB FOUNDATION, KR

Free format text: FORMER OWNERS: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE, DAEJEON, KR; SEOUL NATIONAL UNIVERSITY INDUSTRY FOUNDATION, SEOUL/SOUL, KR

Effective date: 20130717

Ref country code: DE

Ref legal event code: R081

Ref document number: 602005023738

Country of ref document: DE

Owner name: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH IN, KR

Free format text: FORMER OWNER: ELECTRONICS AND TELECOMMUNICATI, SEOUL NATIONAL UNIVERSITY INDUS, , KR

Effective date: 20130717

Ref country code: DE

Ref legal event code: R082

Ref document number: 602005023738

Country of ref document: DE

Representative=s name: PATENTANWAELTE BETTEN & RESCH, DE

Effective date: 20130717

Ref country code: DE

Ref legal event code: R081

Ref document number: 602005023738

Country of ref document: DE

Owner name: SNU R&DB FOUNDATION, KR

Free format text: FORMER OWNER: ELECTRONICS AND TELECOMMUNICATI, SEOUL NATIONAL UNIVERSITY INDUS, , KR

Effective date: 20130717

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: TR

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

Ref country code: BG

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20101222

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: HU

Free format text: LAPSE BECAUSE OF FAILURE TO SUBMIT A TRANSLATION OF THE DESCRIPTION OR TO PAY THE FEE WITHIN THE PRESCRIBED TIME-LIMIT

Effective date: 20100922

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: DE

Payment date: 20160614

Year of fee payment: 12

REG Reference to a national code

Ref country code: DE

Ref legal event code: R119

Ref document number: 602005023738

Country of ref document: DE

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20180201