WO2001046957A1 - Procede et appareil de lissage de flux audio discontinus enchaines - Google Patents

Procede et appareil de lissage de flux audio discontinus enchaines Download PDF

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Publication number
WO2001046957A1
WO2001046957A1 PCT/US2000/033914 US0033914W WO0146957A1 WO 2001046957 A1 WO2001046957 A1 WO 2001046957A1 US 0033914 W US0033914 W US 0033914W WO 0146957 A1 WO0146957 A1 WO 0146957A1
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Prior art keywords
value
audio
audio sample
sample
output
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PCT/US2000/033914
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English (en)
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Smith Freeman
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Sarnoff Corporation
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Priority to JP2001547396A priority Critical patent/JP2003518268A/ja
Publication of WO2001046957A1 publication Critical patent/WO2001046957A1/fr

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    • 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
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
    • G—PHYSICS
    • G11—INFORMATION STORAGE
    • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
    • G11B27/00—Editing; Indexing; Addressing; Timing or synchronising; Monitoring; Measuring tape travel
    • G11B27/02—Editing, e.g. varying the order of information signals recorded on, or reproduced from, record carriers
    • G11B27/031—Electronic editing of digitised analogue information signals, e.g. audio or video signals
    • G11B27/038—Cross-faders therefor

Definitions

  • Embodiments of the present im ention relate to smoothing a sampled audio stream. More particular!) . embodiments of the present ention relate to smoothing a discontinuity of a spliced audio stream .
  • error concealment and audio ⁇ ideo (A Y ) synchronization can require splicing discontinuous audio sample sequences.
  • An error in transmission of a series of audio sample sequences can cause a loss ot an audio sample sequence.
  • Known methods and apparatus to compensate for the loss of an audio sample sequence include copying an adjacent audio sample sequence (e.g.. the preceding audio sample sequence, the subsequent audio sample sequence, etc. ). Such methods and apparatus can result in a discontinuous audio stream that can cause pops and clicks unless the discontinuity is smoothed.
  • FIG. 1 shows a known series of audio sample sequences, e.g. first block 101. second block 102. third block 103.
  • Each audio sample sequence can comprise 100 milliseconds (msecs.) of sampled audio data having a varying audio value (e.g.. decibels (dBs). etc.) over a period of time.
  • the audio value is continuous from first block 101 to second block 102 to third block 103.
  • FIG. 2 illustrates a known method of compensating for a loss of an audio sample sequence.
  • first block 101 can be copied and substituted for the lost second block 1 02 such that a copied first block 202 is positioned betw een first block 101 and third block 103.
  • FIG. 3 show s an illustration of a know n method of joining discontinuous audio sample sequences.
  • the known method includes ramping dow n a preceding sequence to zero and ramping up the subsequent sequence from zero.
  • FIG. 3 show s the trailing edge of first block 301 being ramped down to zero, and the leading edge of copied first block 302 being ramped up from zero.
  • the trailing edge of copied first block 302 is ramped down to zero, and the leading edge of third block 303 is ramped up from zero.
  • This method has disadvantages including 1 ) requiring the need to ramp to zero to be identified soon enough to ramp down the preceding sequence to zero, and 2) the ramp down to zero and back up from zero is not a minimum achievable distortion of the spliced stream of audio sample sequences.
  • a plurality of transition audio sample ⁇ alues can be generated based on a captured value of a final sample of a first audio sample sequence and a received plurality of samples of a second audio sample sequence.
  • Each transition audio sample value can be generated based at least in part on a respective sample of the second audio sample sequence.
  • Each transition audio sample value can represent the ⁇ alue of the captured final sample adjusted with a van ing weighted difference betw een the ⁇ aiue of the final sample and a value of the respective sample of the second audio sample sequence.
  • the van ing eighted difference can be based on a varying weighting function that increases from a value of zero to a value of one such that the first transition audio sample value is generated based on the varying weighting function having a value of zero, and each subsequent transition audio sample value is generated based on the varying weighting function having a value incrementally greater until the final transition audio sample value is generated based on the varying weighting function having a value equal to one.
  • FIG. 1 shows a known series of audio sample sequences.
  • FIG. 2 illustrates a known method of compensating for a loss of an audio sample sequence.
  • FIG. 3 shows an illustration of a known method of joining discontinuous audio sampk sequences.
  • FIG. 4 is an illustration of a method in accordance with an embodiment of the present invention to smooth a spliced discontinuous audio stream.
  • FIG. 5 shows an apparatus in accordance w ith an embodiment of the present invention.
  • FIG. 6 show s an apparatus in accordance with another embodiment of the present invention.
  • FIG. 7 illustrates a method in accordance with another embodiment of the present invention.
  • FIG. 8 shows an apparatus in accordance with another embodiment of the present invention.
  • FIG. 9 show s an apparatus in accordance with another embodiment of the present invention.
  • FIG. 4 is an illustration of a method in accordance ith an embodiment of the present invention to smooth a spliced discontinuous audio stream.
  • First block 403 and copied first block 402 can comprise a series of audio sample sequences of an audio stream that have a time-varying value of s(t). Each of first block 403 and copied first block 402 can include 100 milliseconds of audio samples.
  • First block 403 and copied first block 402 can be joined at a juncture (e.g...
  • a transition output audio v alue can be based on interpolating betw een s( t ) and a delayed sequence of audio samples, s(t-x).
  • an interpolation method is implemented that replaces a sample stream s(t) with an output based on s(t-x) - f(t) * [s(t) - s(t-x)].
  • fit) is a function that smoothly increases from zero to one.
  • s(t-x ) can represem a delayed version of sample stream s(t ).
  • x is a period of time ( e.g.. a period of milliseconds, a number of sample periods, etc. !.
  • Plotted lined 41 1 shows the time-varying value of s(t-x) + f(t) * [s(t) - s( t-x)].
  • f( 104 ) 1.0. etc.).
  • x four milliseconds.
  • FIG. 5 shows an apparatus in accordance with an embodiment of the present invention.
  • a multiplexer 450 can select between standard output S 451 and transition output T 452 to provide output o(t).
  • Sample stream s(t). in one embodiment, is communicated to standard output S 45 1 .
  • s(t) can be communicated to dela_ ⁇ 402 to output a delayed samp!.
  • x can be a period of time (e.g.. four milliseconds, one or more sample periods, etc. ).
  • Subtractor 403 can output the difference betw een s( t ) and s( t-x ). e.g.. s( t > - s( i- x).
  • Multiplier 404 can output the product of fi t ), which is a function that can smoothh increase from zero to one over a period of time, and the output of subtractor 403.
  • Adder 40 can output the sum of the outputs of multiplier 404 and delay 402 such that a transition output signal s( t- ⁇ ) - fit) * [s( t ) - s( t-x )] is communicated to transition output T 452.
  • multiplexer 450 can select the sample stream s(t) present at the standard output S 451 as the output o(t).
  • the multiplexor 450 can select the transition output T 452 during a transition period to smooth a discontinuity between the spliced audio sample sequences.
  • the multiplexor 45 1 can select the sample stream s(t) present at the standard output S 451 as the output o(t).
  • multiplexor 450 can output s(t) as output o(t) during the period 0 ⁇ t ⁇ 100 by outputting the signal present at standard output S 451.
  • multiplexor 450 can output the transition output signal s(t-x) + f(t) * [s(t) - s(t-x)] as the output o(t) during the period 100 ⁇ t ⁇ 104 by outputting the signal present at transition output T 452.
  • multiplexer 450 can again output s(t ) as the outptr o(f) by outputting the signal present at standard output S 45 i
  • FIG. 6 show ' s an apparatus in accordance with another embodiment of the present invention.
  • a multiplexer 550 can select between standard output S 551. first transition output Tl 552. and second transition output T2 553 to provide output o(t ).
  • a function fi t) can be a function with a dynamic range between zero and one-half. In one embodiment, fit) smoothh' increases from zero to one-half, and then smoothh' decreases from one-half to zero
  • x cai be a period of time (e.g.. four milliseconds, one or more sample periods, etc. ).
  • Subtractor 503 can output the difference between s(t) and s(t-x). e.g., s(t) - s(t-x).
  • Multiplier 504 can output the product of f(t). which is a function that can increase from zero to one-half and then decrease from one-half to zero er a period of time, and the output of subtractor 503.
  • Adder 505 can output the sum of the outputs of multiplier 504 and delay 502 such that a first transition output signal s(t-x) + f(t) * [s(t) - s(t-x)] is communicated to first transition output Tl 552.
  • Subtractor 513 can output the difference betw een s( t-x ) and s(t). e.g.. s(t-x) - s(t).
  • Multiplier 514 can output the product of fit ), which is a function that can increase from zero to one-half and then decrease from one-half to zero ov er a period of time, and the output of subtractor 51 .
  • Adder 5 1 can output the sum of s(t ) and the output of multiplier 51 sucn that a second transition output signal s(t ) - fit) * [s(t-x ) - s(t)j is communicated to second transition output T2 553.
  • typical operations e.g...
  • multiplexer 550 can select the sample stream s(t ) present at the standard output S 551 as the output o(t ).
  • the multiplexor 550 can select the first transition output signal s(t-x ) - fi t) * [s( t ) - s(t-x)] present at first transition output Tl 552 as the output o(t) during the first half of a transition period.
  • the first half of the transition period can coincide with the period during which f(t) smoothh ' increases from zero to one-half.
  • the multiplexor 550 can select the second transition output signal s(t) + f(t) * [s(t-x ) - s(f)] present at the second transition output T2 553 as the output o(t) during the second half of the transition period.
  • the second half of the transition period can coincide with the period during which f(t) smoothly decreases from one-half to zero.
  • the multiplexor 550 can select the sample stream s(t) present at the standard output S 551 as the output o(t).
  • FIG. 7 illustrates a method in accordance with another embodiment of the present invention.
  • An audio stream can include a series of audio sample sequences, and a first block 703 and copied first block 702 can be spliced together.
  • Each of first block 703 and copied first block 702 can be an audio sample sequence.
  • a discontinuity in an audio value s(t) car. exist at the juncture between first block 703 and copied first block 702.
  • a plurality of transition audio sample values can be based on interpolating between s(t) and the value of s at the juncture ( i.e.. s(tjuncture ).
  • a plurality of transition audio sample values can be generated based on a captured v alue of a final sample of a first audio sample sequence and a received plurality of samples of a second audio sample sequence.
  • Each transition audio sample value can be generated based at least in part on a respect e sample of the second audio sample sequence.
  • Each transition audio sample value can represent the value of the captured final sample adjusted with a varying weighted difference between the value of the final sample and a value of the respective sample of the second audio sample sequence.
  • the varying weighted difference can be based on a varying weighting function that increases from a value of zero to a value of one such that the first transition audio sample value is generated based on the varying weighting function having a v alue of zero, and each subsequent transition audio sample value is generated based on the varying weighting function having a value incrementally greater until the final transition audio sample value is generated based on the varying weighting function having a value equal to one.
  • an interpolation method is implemented that replaces a sample stream s(t ) with an output based on s(t ⁇ unc ⁇ - ) - fi t * ' :' I si u - s(t ⁇ un ur )].
  • s(tmncture ) is the v alue of s(t ) corresponding to the final sample of the first block 703. and fit ) is a function that smooth!) increases from zero to one.
  • Plotted lined 71 1 shows the time-varying value of s( t ⁇ uncturc) - fit) * [s(t) - s(tjun ⁇ ure)].
  • f(t) increases from zero to one as follows: 0. .125. .25. .50. 1 .
  • One of skill in art can appreciate different embodiments of f(t) based on differences in the length of the transition period, the rate of increase from zero to one. the smoothness of the increase from zero to one. differences in logic to implement f(t). etc.
  • FIG. 8 shows an apparatus in accordance with another embodiment of the present invention.
  • a multiplexer 850 can select between standard output S 851 and transition output T 852 to provide output o(t).
  • Sample stream s(t). in one embodiment, is communicated to standard output S 851.
  • s(t) can be communicated to register 802. which can store the value of the final audio sample immediately prior to the juncture, i.e.. s(tmnc ⁇ urc ).
  • register 802 is a latch register.
  • Subtractor 803 can output the difference between s(t) and s(tjuncturc). e.g.. s(t ) - s(tjunctur .
  • Multiplier 804 can output the product of f(t).
  • w hich is a function that can smooth! ⁇ ' vary from zero to one over a period of time, and the output of subtractor 803.
  • Adder 805 can output the sum of the outputs of multiplier 804 and register 802 such that a transition output signal s(tmncture) + fit) * [si t ) - s(t ⁇ un ⁇ ure)] is communicated to transition output T 852.
  • multiplexer 850 can select the sample stream s(t ) present at the standard output S 851 as the output oi t j.
  • the multiplexor 850 can select the transition output signal s(tjuncture ) -t- f(t) * [s(t) - s(t ⁇ uncture )] present at the transition output T 852 during a transition period to smooth a discontinuity between the spliced audio sample sequences. Subsequent to the transition period, the multiplexor 850 can select the sample stream s(t) present at the standard output S 85 1 as the output o(t).
  • multiplexor 850 can output s(t) as output o(t during the period 0 ⁇ t ⁇ 100 by outputting the signal present at standard output S 851.
  • multiplexor 850 can output the transition output signal s(tjuncmre ) - ⁇ - fit) * [s( t > - s(t ⁇ unc ⁇ ure)j as the output o(t ) during the period 100 ⁇ t ⁇ 1 4 b) outputting the signal present at transition output T 852.
  • multiplexer 850 can again output s(t) as output o(t) by outputting the signal present at standard output S 851.
  • FIG. 9 shows an apparatus in accordance with another embodiment of the present invention.
  • An output multiplexer 950 can select between standard output S 951. and transition output T 952 to provide output o(t).
  • function fi t) can be a function with a dynamic range between zero and one-half. In one embodiment, fi t) smoothh' increases from zero to one-half, and then smoothly decreases from one-half to zero.
  • Sample stream s(t). in one embodiment, is communicated to standard output S 951 .
  • s(t) can be communicated to register 902. which can store the value of the final audio sample immediately prior to the juncture, i.e.. s(tjuncmre).
  • Subtractor 903 can output the difference between s(t) and s(tjuncture). e.g.. s(t) - s(tjuncture).
  • Invertor 91 1 can receive and invert the output of subtractor 903. i.e.. invertor 91 1 can output s(tjuncturc) - s(t).
  • First multiplexer 921 can receive each of the outputs of subtractor 903 and inverter
  • 91 1. can thereby selectively output one of s(t) - s(tjuncture) and s(tjuncture) - s(t) based on a multiplexer control signal m(t).
  • the value of m(t) can be based at least in part on the value of f(t).
  • m(t) can have a value of logical one (e.g.. five volts. 3.3 volts. 2.5 volts, etc.) when f(t) smoothly increases from zero to one-half and m(t) can have a value of logical zero (e.g.. approximately zero volts, etc. ) when f(t) smoothly decreases from one-half to zero.
  • First multiplexer 922 can output s(t ) - s(tjunc ⁇ ure) when fit) smooth!)' increases from zero to on - half, and output s(tjuncture ) - s t ) when f(t) smoothly decreases from one-half to zero.
  • Multiplier 904 can output the product of f(t) and the output of first multiplexer 921 .
  • Second multiplexer 922 can receive selectively output one of s(t ncture) and s(t) based on multiplexer control signal m(t). Second multiplexer 922 can output s(tjuncture) when fit ) smoothly increases from zero to one-half, and output s(t ) when fit ) smoothly decreases from one-half to zero.
  • Adder 905 can output the sum of the outputs of multiplier 904 and second multiplexer 922. When fit) smoothh' increases from zero to one-half. m(t) has a value of logical one and adder 905 can output s(tjuncture) + f(t) * [s(t ) - s(tjuncture)]. When f(t) smoothly decreases from one-half to zero, m(t) has a value of logical zero and adder 905 can
  • multiplexer 950 can select the sample stream s(t) present at the standard output S 951 as the output o(t).
  • the multiplexor 950 can select the transition output signal present at transition output T 952 as the output o(t ) during both halv es of tlie transition period
  • the first half of tnc transition period can coincide it the period during w hich fi t ) smooth!) increases lrom zero to one-hall and m( t ) na> a v alue o logical one.
  • the transition output signal present at transition output T 952 is sUiuncuti - > - f(t) * [sit ) - sltiunciuie ik
  • the second half of the transition period can coincide w ith the perkv.
  • a discontinuitv m . spliced audio stream is smoothed b ⁇ generating an output signal based on interpolating witn . time-varying weighting function and a reference v alue based on at least the final sample of th. audio sample sequence preceding the juncture having a discontinuity .
  • the interpolation is based at least in part on a delayed audio stream, s(t-x).
  • the interpolation is based at least in part on the value of the last sample of the audio sample sequence preceding the juncture having a discontinuity.
  • an interpolation based at least in part on an average of the last four samples of the audio sample sequence preceding the juncture having a discontinuity an interpolation based at least in part on a mean of the final samples ( e.g.. the last five samples, the number of samples corresponding to the length of the transition period, etc. ) of the audio sample sequence preceding the juncture having a diseontinuit) . etc.
  • Embodiments of the present invention include s) stems incorporating at least one of the apparatus, or performing at least one of the methods, described herein.
  • Examples of systems in accordance with embodiments of the present invention include a digital audio system of a computer, a television, a High Definition Telev ision (HDTV ) receiv er decode: . compact disc ( CD ) player, digital versatile disc (DYD) piaver. portable audio player, hon stereo S ⁇ stein, digital radio receiv er, etc
  • the splice. discontinuous audio streams can be smoothed to avoid audible artifacts (e.g.. pops, clickt . etc.) without modifying the audio sample sequence preceding the juncture having a discontinuity .
  • the points in the audio stream at which splices ma) occur can be identified in advance (e.g.. at or after a copied audio sample sequence is inserted into the audio stream, etc.).
  • the value(s) of the last sample( s ) of the audio sample sequence preceding the juncture hav ing a discontinuit) can be captured, stores, av erageu.
  • instructions adapted ⁇ . be executed b) a processor to perform a method m accordance w ith an embodiment of th_ present invention are stored on a computer-readable medium and distributed as software I Ik computer-readable medium can be a dev ice adapted to store digital mtormation. f or examp .
  • a computer-readable medium includes a portable magnetic disk, such as a flopp) disk: or a ZipCB disk, manufactured the Iomega Corporation of Rov . I 'tan: or a Compact Disk Reac Onh Memon (CD-ROM ) as is know n m the art for distributing sottw are.
  • the compute - readable medium can be distributed to a user that has a processor suitable for executing instructions adapted to be executed.
  • T he term "" adapted to be executed “” is meant to encompass am instructions that are read) to be executed in their present form ( e.g.. machine code) b) a processor, or require further manipulation (e.g.. compilation, decryption, or provided w ith an access code, etc.) to be ready to be executed b) a processor.
  • Embodiments of the present invention advantageoush allow smoothing of a spliced discontinuous audio stream.
  • the audio sample sequence preceding a discontinuity may not need to be modified (e.g., ramped down to zero, etc.).
  • a reduced amount of distortion of the audio stream can be provided while smoothing the discontinuit ⁇ '.
  • the last sample of the audio sample sequence preceding the discontinuity need be captured. That last sample (e.g.. a point at which a splice occurs ) typically can be identified easily in advance.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing For Digital Recording And Reproducing (AREA)

Abstract

Dans l'une des exécutions, on peut obtenir des valeurs d'échantillons de transition sur la base de la valeur prélevée de l'échantillon final d'une première suite d'échantillons audio et d'échantillons provenant d'une deuxième suite d'échantillons audio. Chacune des valeurs des échantillons de transition peut être obtenue au moins en partie en fonction d'un échantillon correspondant de la deuxième suite d'échantillons audio. Chacune des valeurs d'échantillons de transition peut représenter la valeur de l'échantillon final prélevé, ajustée à l'aide de la différence pondérée variable entre la valeur de l'échantillon final et la valeur de l'échantillon correspondant de la deuxième suite d'échantillons audio. La différence pondérée variable peut se baser sur une fonction variable de pondération passant de 0 à 1, la valeur du premier échantillon audio de transition étant obtenue sur la base de la fonction de pondération variable de valeur nulle, et chacune des valeurs suivantes de l'échantillon audio de transition est obtenue sur la base de la fonction variable de pondération présentant une valeur grandissant par incréments jusqu'à ce que la valeur de l'échantillon audio final de transition soit obtenue sur la base d'une fonction de pondération variable d'une valeur égale à 1.
PCT/US2000/033914 1999-12-22 2000-12-14 Procede et appareil de lissage de flux audio discontinus enchaines WO2001046957A1 (fr)

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JP2001547396A JP2003518268A (ja) 1999-12-22 2000-12-14 不連続な繋ぎ音声ストリームの平滑化方法および装置

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Cited By (4)

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JP2004004440A (ja) * 2002-03-22 2004-01-08 Yamaha Corp 歌唱合成装置、歌唱合成用プログラム及び歌唱合成用プログラムを記録したコンピュータで読み取り可能な記録媒体
EP1754393A1 (fr) * 2004-04-16 2007-02-21 Smart Internet Technology Crc Pty Limited Appareils et procedes destines a etre utilises pour creer une scene audio
US8069051B2 (en) * 2007-09-25 2011-11-29 Apple Inc. Zero-gap playback using predictive mixing
US20230074155A1 (en) * 2014-09-09 2023-03-09 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Audio splicing concept

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CN104780438A (zh) * 2015-03-20 2015-07-15 广东欧珀移动通信有限公司 一种视频与歌曲音频拼接的方法及装置

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Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2004004440A (ja) * 2002-03-22 2004-01-08 Yamaha Corp 歌唱合成装置、歌唱合成用プログラム及び歌唱合成用プログラムを記録したコンピュータで読み取り可能な記録媒体
EP1754393A1 (fr) * 2004-04-16 2007-02-21 Smart Internet Technology Crc Pty Limited Appareils et procedes destines a etre utilises pour creer une scene audio
EP1754393A4 (fr) * 2004-04-16 2011-02-23 Dolby Lab Licensing Corp Appareils et procedes destines a etre utilises pour creer une scene audio
US8069051B2 (en) * 2007-09-25 2011-11-29 Apple Inc. Zero-gap playback using predictive mixing
US20230074155A1 (en) * 2014-09-09 2023-03-09 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Audio splicing concept
US11882323B2 (en) * 2014-09-09 2024-01-23 Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. Audio splicing concept

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CN1413347A (zh) 2003-04-23

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