WO2003079330A1 - Method for adaptive codebook pitch-lag computation in audio transcoders - Google Patents
Method for adaptive codebook pitch-lag computation in audio transcodersInfo
- Publication number
- WO2003079330A1 WO2003079330A1 PCT/US2003/007901 US0307901W WO03079330A1 WO 2003079330 A1 WO2003079330 A1 WO 2003079330A1 US 0307901 W US0307901 W US 0307901W WO 03079330 A1 WO03079330 A1 WO 03079330A1
- Authority
- WO
- WIPO (PCT)
- Prior art keywords
- subframe
- pitch lag
- subframes
- destination
- module
- Prior art date
Links
Classifications
-
- 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
- G10L19/04—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 using predictive techniques
- G10L19/16—Vocoder architecture
- G10L19/173—Transcoding, i.e. converting between two coded representations avoiding cascaded coding-decoding
-
- 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
- G10L19/04—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 using predictive techniques
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
-
- 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
- G10L19/04—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 using predictive techniques
- G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
- G10L19/09—Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor
Definitions
- the present invention relates generally to processing telecommunication signals. More particularly, the invention provides a method and apparatus for translating digital speech packets from one code-excited linear prediction (CELP) format to another CELP format. More specifically, it relates to a method and to an apparatus for interpolating an adaptive codebook pitch lag obtained by a first CELP coder as input into another adaptive codebook pitch lag of a second CELP coder.
- CELP code-excited linear prediction
- the invention has been applied to voice transcoding, but it would be recognized that the invention may also include other applications.
- Telecommunication techniques have developed over the years.
- Coding often includes a process of converting a raw signal (voice, image, video, etc) into a format amenable for transmission or storage.
- the coding usually results in a large amount of compression, but generally involves significant signal processing to achieve.
- the outcome of the coding is a bitstream (sequence of frames) of encoded parameters according to a given compression format.
- the compression is achieved by removing statistically and perceptually redundant information using various techniques for modeling the signal.
- the encoded format is referred to as a "compression format" or "parameter space”.
- the decoder takes the compressed bitstream and regenerates the original signal.
- compression typically leads to information loss.
- Coding can be performed using a codec device.
- a CELP-(code excited linear prediction) based codec can be thought of as an algorithm that maps between sampled speech and some parameter space using a model of speech production, i.e. it encodes and decodes the digital speech.
- all CELP-based algorithms operate on frames of speech which are further divided into several subframes.
- the frame parameters used in CELP-based models has linear-predictive coefficients (LPC) used for short-term prediction of the speech signal (and physically relating to the vocal tract, mouth and nasal cavity, and lips), as well as an excitation signal composed from adaptive and fixed codebooks.
- LPC linear-predictive coefficients
- the adaptive codebook is used to model long-term pitch information in the speech.
- Most of the computational effort in analyzing the speech frame is in determining the LPC coefficients and finding the pitch lag (or equivalently adaptive codeword index).
- a lack of inherent interoperability between voice compression standards often means that there may be a need for translation when an end-to-end call traverses network boundaries. Interconnecting these diverse networks and terminals generally requires voice transcoding from one voice standard into another. A need for such transcoding is typically addressed in mobile switching centers, media gateways, multimedia messaging systems, and on the edge of networks.
- voice coding in the context of heterogeneous wireless, mobile and wireline networks illustrate networks that run on different standards.
- voice compression and coding standards used for terminals in different networks - G.729 and G.723.1 for Noice over IP (VoIP), GSM, GSM-AMR, ENRC and a range of other standards used (or emerging) on different wireless networks.
- Figures 1 A, IB and 1C illustrate this diversity of CELP based voice compression standards in a simplified manner. In this case voice transcoding occurs at the edge of every network and between any two networks.
- the computation of adaptive codebook pitch-lag plays an important role in searching the adaptive codebook in voice transcoding.
- the sub-frame size in G.723.1 is 7.5ms (FIG. IB), but it is 5 ms in GSM-AMR (FIG. 1 A) and it is either 6.625 ms or 6.75ms in EVRC (FIG. 1 C).
- the invention provides a method and apparatus for translating digital speech packets from one code-excited linear prediction (CELP) format to another CELP format. More specifically, it relates to a method and to an apparatus for interpolating an adaptive codebook pitch lag obtained by a first CELP coder as input into another adaptive codebook pitch lag of a second CELP coder.
- CELP code-excited linear prediction
- the invention has been applied to voice transcoding, but it would be recognized that the invention may also include other applications.
- the present invention is a method and apparatus for adaptive codebook pitch-lag computation.
- the apparatus includes (a) a time-base subframe inspection module that stores the adaptive codebook parameters of each subframe from source codec which waits for interpolation or mapping and computes the proportion of subframe overlapping between source codec and destination codec; (b) a decision module that computes the energy of the adaptive codebook among all source subframes which overlap with the destination subframe and searches the maximum energy value as the criterion for the selection of pitch lag; and (c) a selection module that selects the pitch lag of a subframe as an output from all overlapping source subframes based on an output of the decision module.
- the time-base subframe inspection module includes a buffer that stores the pitch lag, pitch gain and number of samples of source subframes which wait for mapping into the destination subframe and a discriminator that determines whether destination subframe is covered by multiple source subframes.
- the method includes the steps of computing the pitch-lag of the destination subframe from source CELP codec parameter space.
- the step of computing the pitch-lags includes the steps of storing the adaptive codebook parameters of each source subframe which overlaps with a destination subframe, deciding whether the destination subframe is wholly covered by one source subframe or multiple source subframes, either outputting the pitch lag of the source subframe if the destination subframe is wholly covered by only one source subframe or outputting the pitch lag of the subframe which has the maximum value of the criterion used by a decision module if the destination subframe is covered by multiple source subframes.
- the step of outputting the pitch lag of a subframe which has the maximum value of the criterion used by a decision module includes steps of searching for the maximum value of the criterion by a decision making module, selecting the pitch lag of a subframe which has the maximum value among all overlapping source subframes, and outputting the pitch lag of that selected subframe.
- the step of searching the maximum value of the criterion by a decision module includes steps of combining the adaptive codebook parameters of overlapped source subframes, computing the proportion of overlap of each source subframe, computing the energy contribution which is used as the criterion value in each overlapped subframe, and indexing the subframe which has the maximum value of the criterion.
- the invention provides an apparatus for processing adaptive codebook pitch lag from one CELP based standard to another CELP based standard.
- the apparatus has various modules that perform at least functionality described herein.
- the apparatus includes a time-base subframe inspection module, which is adapted to associate one or more incoming subframes with an outgoing subframes of a destination codec.
- the apparatus also has a decision module coupled to the time-base subframe inspection module.
- the decision module is adapted to determine a pitch lag parameter of a desired subframe from a plurality of pitch lag parameters among respective two or more incoming subframes.
- the apparatus has a pitch lag selection module coupled to the decision module.
- the pitch lag selection module is adapted to select the desired pitch lag parameter.
- the invention provides a method for processing an adaptive codebook parameter pitch-lag from a source CELP based codec to a destination CELP standard codec.
- the method comprises storing in a memory the more than one adaptive codebook parameters of one or more respective each subframes from a source codec which waits for mapping.
- the method also decides whether the a destination subframe is wholly covered by one source subframe while the one or more subframes wait for mapping.
- the method outputs the a pitch lag of the a source subframe if the destination subframe is wholly covered by a single one source subframe; or output the a desired value of a pitch lag of a source subframe which has maximum value of the based upon a criterion by a decision module if the destination subframe is covered by two or more multiple source subframes.
- the invention provides a computer based system for processing adaptive codebook pitch lag from one CELP based standard to another CELP based standard.
- the system includes computer memory, which may be one or more memories. Various codes are provided on the one or more memories.
- the system includes one or more codes directed to a time-base subframe inspection module, which is adapted to associate one or more incoming subframes with an outgoing subframes of a destination codec.
- the system also includes one or more codes directed to a decision module coupled to the time-base inspection module, which is adapted to determine a desired pitch lag parameter from a plurality of pitch lag parameters among respective the two or more incoming subframes.
- One or more codes are directed to a pitch lag selection module coupled to the decision module.
- the decision module is adapted to select the desired pitch lag parameter.
- computer code or codes can be used in the form of software or firm ware to carryout the functionality described herein.
- An advantage of the present invention is that it provides a fast pitch-lag parameter computation from one codec into another in transcoding without compromising audio quality according to a specific embodiment.
- a fast and correct computation algorithm can improve the audio transcoding, not only in terms of computational performance, but more importantly in terms of maintaining audio quality. Depending upon the embodiment, one or more of these advantages may be achieved.
- FIG. 1A, IB and IC are diagrams useful in illustrating the different subframe sizes used in different CELP codecs
- FIG 2 is a simplified function block diagram for performing adaptive codebook pitch lag interpolation according to an embodiment of the present invention
- FIG. 3 is a simplified diagram showing a comparison of different subframe size between source and destination codecs and overlapping according to an embodiment of the present invention
- FIG 4 is a simplified flow diagram illustrating a routine for interpolating pitch lag for different subframe sizes according to an embodiment of the present invention
- FIG. 5 is a simplified block diagram showing the subframe computation in the particular example of transcoding from G.723.1 to GSM-AMR according to an embodiment of the present invention.
- the invention provides a method and apparatus for translating digital speech packets from one code-excited linear prediction (CELP) format to another CELP format. More specifically, it relates to a method and to an apparatus for interpolating an adaptive codebook pitch lag obtained by a first CELP coder as input into another adaptive codebook pitch lag of a second CELP coder.
- CELP code-excited linear prediction
- the invention has been applied to voice transcoding, but it would be recognized that the invention may also include other applications.
- speech signals can be categorized as either voiced or unvoiced signals.
- the adaptive codebook pitch-lag parameter is quite stable during voiced excitation sequences, but it is not stable during unvoiced sounds or at the onset of voiced sounds. Unvoiced sounds are generally weak, random signals, and in such cases the adaptive codebook gain is very small and the selection of adaptive codebook pitch-lag is not as important as for voiced signals.
- Voiced signals on the other hand are generally strong and stable, and the selection of adaptive codebook pitch-lag directly determines the quality of the speech compression.
- the subframe size between source and destination codecs can be different (FIG. 3).
- the subframe in the source codec includes a size of Ns for the first subframe.
- the destination codec (see reference numeral 1) has a first subframe of No, which is smaller in size than the first codec subframe.
- an edge of the first source codec and first destination codec align. Since the first source subframe is large in size and also has a spatial alignment that extends beyond the first destination subframe, the first destination subframe is covered (i.e., wholly covered) by the first source subframe.
- FIG. 2 illustrates a hierarchy of the building blocks used in the pitch lag interpolation according to the present invention.
- a Time-Base Subframe Inspection Module handles the subframe interpolation between the source codec and the destination codec due to the dissimilar subframe sizes of the source and destination codecs; the module handles all cases of source and destination subframe length (i.e. the source subframe length is shorter than the destination subframe, the source subframe length is longer than the destination subframe length and the source subframe length is equal to the destination subframe length).
- the Quick Decision Module computes the criteria of selection function of desired pitch lag for the destination codec.
- the Selection Module handles the computation of the final pitch lag based on the criteria output computed by the Quick Decision Module.
- the Time-Base .Subframe Inspection Module can directly connect to the output (i.e. can bypass the Quick Decision Module and the Selection Module). This is so because the Time-Base Subframe Inspection Module has the ability to map it directly to the output. This is determined by the Time-base Inspection Module based on the position of the destination subframe with relation to the source subframe in time.
- the adaptive codebook gain, adaptive codebook pitch-lag and the sub-frame size in the source codec are g p s , L s , Ns, respectively, and the subframe size in the destination codec is Np.
- the subframe size of the source codec can be different to that of the destination.
- the source and destination frames may not be aligned and they can be overlapped.
- we have described various embodiments list under different case headings, which are merely provided to be illustrating. These embodiments are not intended to be limiting the scope of the claims herein.
- One of ordinary skill in the art would recognize many variations, alternatives, and modifications.
- the adaptive codebook pitch- lag is the pitch- lag of the source subframe for which a function of adaptive codebook gain and overlapping size is the maximum. It can be expressed as: [0032] where E Tha is a function of adaptive gain gp and the portion of overlapping ⁇ in source sub-frame:
- the selected adaptive codebook pitch-lag can be used as adaptive codebook pitch-lag for the destination subframe, or as open-loop adaptive codebook pitch-lag if further tuning is required.
- the adaptive codebook parameters reach the input of the interpolator module of the audio transcoder.
- a check for the current destination subframe alignment in relation to the source subframe is made. If the destination subframe is completely covered by one subframe of the source codec, the pitch lag at the destination subframe is equal to the corresponding pitch lag of the source subframe as specified in Eq. 1.
- the selection module within the audio transcoder searches through the overlapping source subframes for the maximum criteria as specified in equations 2 and 3.
- the basis for the criteria in equations 2 and 3 is the strength of the pitch gain in the source codec subframes.
- the adaptive codebook gain is very small and that contrasts with voiced periods, where the pitch gain is strong. Therefore, depending on the portion of overlapping source subframe, as specified by the factor ⁇ from equation 3 and the magnitude of the pitch gain, the decision criteria as specified in equation 3 (E n ) are calculated.
- the pitch lag is then outputted at the destination codec.
- the computed pitch lag should fit within the allowed index range of the pitch lag for the destination codec.
- the pitch lag may be either doubled or halved depending on where it falls, whether at the minimum allowed pitch or at the maximum allowed pitch, respectively.
- G.723.1 GSM-AMR TRANSCODING EXAMPLE
- FIG. 5 we show how the adaptive codebook pitch-lag is interpolated in a G.723.1 to GSM-AMR transcoder (FIG. 5). Again, this diagram is merely an example, which should not unduly limit the scope of the claims herein. One of ordinary skill in the art would recognize many variations, modifications, and alternatives. [0040] It can be seen from FIG. 5 that three GSM-AMR sub-frames are needed to describe the same duration of speech signal as two G.7231 sub-frames. Likewise three GSM-AMR sub-frames are needed for every two G.723.1 sub-frames. If the source codec is G.723.1 and the destination codec is GSM-AMR, the GSM-AMR adaptive codebook pitch-lag after computation is as follows:
- GSM-AMR subframe is 5ms and G.723.1 subframe is 7.5ms.
- the GSM-AMR subframe ⁇ m ⁇ is fully covered by the G723.1 subframe ⁇ n ⁇ . According to the equation (1), its adaptive codebook pitch-lag is
- the invention of adaptive codebook computation described in this document is generic to all CELP based voice codecs, and applies to any voice transcoders between the existing codecs G.723.1, GSM-AMR, EVRC, G.728, G.729, G.729 A, QCELP, MPEG-4 CELP, SMV and all other future CELP based voice codecs that make use of pitch lag information.
Abstract
Description
Claims
Priority Applications (5)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
AU2003214182A AU2003214182A1 (en) | 2002-03-12 | 2003-03-12 | Method for adaptive codebook pitch-lag computation in audio transcoders |
KR10-2004-7014297A KR20040104508A (en) | 2002-03-12 | 2003-03-12 | Method for adaptive codebook pitch-lag computation in audio transcoders |
EP03711590A EP1483758A4 (en) | 2002-03-12 | 2003-03-12 | Method for adaptive codebook pitch-lag computation in audio transcoders |
CN038106450A CN1653521B (en) | 2002-03-12 | 2003-03-12 | Method for adaptive codebook pitch-lag computation in audio transcoders |
JP2003577246A JP2005520206A (en) | 2002-03-12 | 2003-03-12 | Adaptive Codebook, Pitch, and Lag Calculation Method for Audio Transcoder |
Applications Claiming Priority (2)
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US36440302P | 2002-03-12 | 2002-03-12 | |
US60/364,403 | 2002-03-12 |
Publications (1)
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WO2003079330A1 true WO2003079330A1 (en) | 2003-09-25 |
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ID=28041908
Family Applications (1)
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PCT/US2003/007901 WO2003079330A1 (en) | 2002-03-12 | 2003-03-12 | Method for adaptive codebook pitch-lag computation in audio transcoders |
Country Status (7)
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US (2) | US7260524B2 (en) |
EP (1) | EP1483758A4 (en) |
JP (1) | JP2005520206A (en) |
KR (1) | KR20040104508A (en) |
CN (1) | CN1653521B (en) |
AU (1) | AU2003214182A1 (en) |
WO (1) | WO2003079330A1 (en) |
Families Citing this family (24)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN1653521B (en) * | 2002-03-12 | 2010-05-26 | 迪里辛姆网络控股有限公司 | Method for adaptive codebook pitch-lag computation in audio transcoders |
KR100546758B1 (en) * | 2003-06-30 | 2006-01-26 | 한국전자통신연구원 | Apparatus and method for determining transmission rate in speech code transcoding |
US7433815B2 (en) * | 2003-09-10 | 2008-10-07 | Dilithium Networks Pty Ltd. | Method and apparatus for voice transcoding between variable rate coders |
US7519532B2 (en) * | 2003-09-29 | 2009-04-14 | Texas Instruments Incorporated | Transcoding EVRC to G.729ab |
US9058812B2 (en) * | 2005-07-27 | 2015-06-16 | Google Technology Holdings LLC | Method and system for coding an information signal using pitch delay contour adjustment |
US7602745B2 (en) * | 2005-12-05 | 2009-10-13 | Intel Corporation | Multiple input, multiple output wireless communication system, associated methods and data structures |
KR100900438B1 (en) * | 2006-04-25 | 2009-06-01 | 삼성전자주식회사 | Apparatus and method for voice packet recovery |
US8218529B2 (en) * | 2006-07-07 | 2012-07-10 | Avaya Canada Corp. | Device for and method of terminating a VoIP call |
EP1903559A1 (en) * | 2006-09-20 | 2008-03-26 | Deutsche Thomson-Brandt Gmbh | Method and device for transcoding audio signals |
GB2466672B (en) | 2009-01-06 | 2013-03-13 | Skype | Speech coding |
GB2466673B (en) | 2009-01-06 | 2012-11-07 | Skype | Quantization |
GB2466675B (en) | 2009-01-06 | 2013-03-06 | Skype | Speech coding |
GB2466671B (en) | 2009-01-06 | 2013-03-27 | Skype | Speech encoding |
GB2466669B (en) * | 2009-01-06 | 2013-03-06 | Skype | Speech coding |
GB2466670B (en) | 2009-01-06 | 2012-11-14 | Skype | Speech encoding |
US8243610B2 (en) * | 2009-04-21 | 2012-08-14 | Futurewei Technologies, Inc. | System and method for precoding codebook adaptation with low feedback overhead |
EP2249334A1 (en) * | 2009-05-08 | 2010-11-10 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Audio format transcoder |
US8452606B2 (en) | 2009-09-29 | 2013-05-28 | Skype | Speech encoding using multiple bit rates |
US8521541B2 (en) * | 2010-11-02 | 2013-08-27 | Google Inc. | Adaptive audio transcoding |
CN104243734B (en) * | 2013-06-18 | 2019-03-01 | 深圳市共进电子股份有限公司 | Audio processing system and method |
KR101790901B1 (en) | 2013-06-21 | 2017-10-26 | 프라운호퍼 게젤샤프트 쭈르 푀르데룽 데어 안겐반텐 포르슝 에. 베. | Apparatus and method realizing a fading of an mdct spectrum to white noise prior to fdns application |
PL3011555T3 (en) | 2013-06-21 | 2018-09-28 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Reconstruction of a speech frame |
WO2014202539A1 (en) | 2013-06-21 | 2014-12-24 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Apparatus and method for improved concealment of the adaptive codebook in acelp-like concealment employing improved pitch lag estimation |
EP2980799A1 (en) * | 2014-07-28 | 2016-02-03 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Apparatus and method for processing an audio signal using a harmonic post-filter |
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- 2003-03-12 CN CN038106450A patent/CN1653521B/en not_active Expired - Fee Related
- 2003-03-12 KR KR10-2004-7014297A patent/KR20040104508A/en not_active Application Discontinuation
- 2003-03-12 EP EP03711590A patent/EP1483758A4/en not_active Withdrawn
- 2003-03-12 US US10/350,349 patent/US7260524B2/en not_active Expired - Fee Related
- 2003-03-12 AU AU2003214182A patent/AU2003214182A1/en not_active Abandoned
- 2003-03-12 JP JP2003577246A patent/JP2005520206A/en not_active Withdrawn
- 2003-03-12 WO PCT/US2003/007901 patent/WO2003079330A1/en active Application Filing
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2007
- 2007-07-26 US US11/881,742 patent/US7996217B2/en not_active Expired - Fee Related
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Also Published As
Publication number | Publication date |
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EP1483758A4 (en) | 2007-04-11 |
US20040002855A1 (en) | 2004-01-01 |
JP2005520206A (en) | 2005-07-07 |
EP1483758A1 (en) | 2004-12-08 |
CN1653521A (en) | 2005-08-10 |
US7260524B2 (en) | 2007-08-21 |
CN1653521B (en) | 2010-05-26 |
AU2003214182A1 (en) | 2003-09-29 |
US20080189101A1 (en) | 2008-08-07 |
KR20040104508A (en) | 2004-12-10 |
US7996217B2 (en) | 2011-08-09 |
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