CA2736046A1 - Audio coding system using spectral hole filling - Google Patents

Audio coding system using spectral hole filling Download PDF

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Publication number
CA2736046A1
CA2736046A1 CA2736046A CA2736046A CA2736046A1 CA 2736046 A1 CA2736046 A1 CA 2736046A1 CA 2736046 A CA2736046 A CA 2736046A CA 2736046 A CA2736046 A CA 2736046A CA 2736046 A1 CA2736046 A1 CA 2736046A1
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Prior art keywords
spectral components
subband signals
spectral
scaling
zero
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CA2736046A
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French (fr)
Inventor
Michael Mead Truman
Grant Allen Davidson
Matthew Conrad Fellers
Mark Stuart Vinton
Matthew Aubrey Watson
Charles Quito Robinson
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Dolby Laboratories Licensing Corp
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Dolby Laboratories Licensing Corp
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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/02Speech 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 spectral analysis, e.g. transform vocoders or subband vocoders
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/02Speech 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 spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/032Quantisation or dequantisation of spectral components
    • G10L19/035Scalar quantisation
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Multimedia (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Quality & Reliability (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Optical Elements Other Than Lenses (AREA)
  • Stereophonic System (AREA)
  • Information Retrieval, Db Structures And Fs Structures Therefor (AREA)
  • Optical Communication System (AREA)
  • Optical Recording Or Reproduction (AREA)
  • Adornments (AREA)
  • Fittings On The Vehicle Exterior For Carrying Loads, And Devices For Holding Or Mounting Articles (AREA)
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Abstract

Audio coding processes like quantization can cause spectral components of an encoded audio signal to be set to zero, creating spectral holes in the signal.
These spectral holes can degrade the perceived quality of audio signals that are reproduced by audio coding systems. An improved decoder avoids or reduces the degradation by filling the spectral holes with synthesized spectral components. An improved encoder may also be used to realize further improvements in the decoder.

Claims (45)

1. A method for generating audio information, wherein the method comprises:
receiving an input signal and obtaining therefrom a set of subband signals each having one or more spectral components representing spectral content of an audio signal;
identifying within the set of subband signals a particular subband signal in which one or more spectral components have a non-zero value and are quantized by a quantizer having a minimum quantizing level that corresponds to a threshold, and in which a plurality of spectral components have a zero value;
generating synthesized spectral components that correspond to respective zero-valued spectral components in the particular subband signal and that are scaled according to a scaling envelope less than or equal to the threshold;
generating a modified set of subband signals by substituting the synthesized spectral components for corresponding zero-valued spectral components in the particular subband signal; and generating the audio information by applying a synthesis filterbank to the modified set of subband signals.
2. The method according to claim 1 wherein the scaling envelope is uniform.
3. The method according to claim 1 or 2 wherein the synthesis filterbank is implemented by a block transform that has spectral leakage between adjacent spectral components and the scaling envelope varies at a rate substantially equal to a rate of roll off of the spectral leakage of the block transform.
4. The method according to any one of claims 1 through 3 wherein the synthesis filterbank is implemented by a block transform and the method comprises:
applying a frequency-domain filter to one or more spectral components in the set of subband signals; and deriving the scaling envelope from an output of the frequency-domain filter.
5. The method according to claim 4 that comprises varying the response of the frequency-domain filter as a function of frequency.
6. The method according to any one of claims 1 through 5 that comprises:
obtaining a measure of tonality of the audio signal represented by the set of subband signals; and adapting the scaling envelope in response to the measure of tonality.
7. The method according to claim 6 that obtains the measure of tonality from the input signal.
8. The method according to claim 6 that comprises deriving the measure of tonality from the way in which the zero-valued spectral components are arranged in the particular subband signal.
9. The method according to any one of claims 1 through 8 wherein the synthesis filterbank is implemented by a block transform and the method comprises:
obtaining a sequence of sets of subband signals from the input signal;
identifying a common subband signal in the sequence of sets of subband signals where, for each set in the sequence, one or more spectral components have a non-zero value and a plurality of spectral components have a zero value;
identifying a common spectral component within the common subband signal that has a zero value in a plurality of adjacent sets in the sequence that are either preceded or followed by a set with the common spectral components having a non-zero value;
scaling the synthesized spectral components that correspond to the zero-valued common spectral components according to the scaling envelope that varies from set to set in the sequence according to temporal masking characteristics of the human auditory system;

generating a sequence of modified sets of subband signals by substituting the synthesized spectral components for the corresponding zero-valued common spectral components in the sets; and generating the audio information by applying the synthesis filterbank to the sequence of modified sets of subband signals.
10. The method according to any one of claims 1 through 9 wherein the synthesis filterbank is implemented by a block transform and the method generates the synthesized spectral components by spectral translation of other spectral components in the set of subband signals.
11. The method according to any one of claims 1 through 10 wherein the scaling envelope varies according to temporal masking characteristics of the human auditory system.
12. A method for generating an output signal, wherein the method comprises:
generating a set of subband signals each having one or more spectral components representing spectral content of an audio signal by quantizing information that is obtained by applying an analysis filterbank to audio information;
identifying within the set of subband signals a particular subband signal in which one or more spectral components have a non-zero value and are quantized by a quantizer having a minimum quantizing level that corresponds to a threshold, and in which a plurality of spectral components have a zero value;
deriving scaling control information from the spectral content of the audio signal, wherein the scaling control information controls scaling of synthesized spectral components to be synthesized and substituted for the spectral components having a zero value in a receiver that generates audio information in response to the output signal; and generating the output signal by assembling the scaling control information and information representing the set of subband signals.
13. The method according to claim 12 that comprises:
obtaining a measure of tonality of the audio signal represented by the set of subband signals; and deriving the scaling control information from the measure of tonality.
14. The method according to claim 12 or 13 that comprises:
obtaining an estimated psychoacoustic masking threshold of the audio signal represented by the set of subband signals; and deriving the scaling control information from the estimated psychoacoustic masking threshold.
15. The method according to any one of claims 12 through 14 that comprises:
obtaining two measures of spectral levels for portions of the audio signal represented by the non-zero-valued and the zero-valued spectral components; and deriving the scaling control information from the two measures of spectral levels.
16. An apparatus for generating audio information, wherein the apparatus comprises:
a deformatter that receives an input signal and obtains therefrom a set of subband signals each having one or more spectral components representing spectral content of an audio signal;
a decoder coupled to the deformatter that identifies within the set of subband signals a particular subband signal in which one or more spectral components have a non-zero value and are quantized by a quantizer having a minimum quantizing level that corresponds to a threshold, and in which a plurality of spectral components have a zero value, that generates synthesized spectral components that correspond to respective zero-valued spectral components in the particular subband signal and are scaled according to a scaling envelope less than or equal to the threshold, and that generates a modified set of subband signals by substituting the synthesized spectral components for corresponding zero-valued spectral components in the particular subband signal; and a synthesis filterbank coupled to the decoder that generates the audio information in response to the modified set of subband signals.
17. The apparatus according to claim 16 wherein the scaling envelope is uniform.
18. The apparatus according to claim 16 or 17 wherein the synthesis filterbank is implemented by a block transform that has spectral leakage between adjacent spectral components and the scaling envelope varies at a rate substantially equal to a rate of roll off of the spectral leakage of the block transform.
19. The apparatus according to any one of claims 16 through 18 wherein the synthesis filterbank is implemented by a block transform and the decoder:
applies a frequency-domain filter to one or more spectral components in the set of subband signals; and derives the scaling envelope from an output of the frequency-domain filter.
20. The apparatus according to claim 19 wherein the decoder varies the response of the frequency-domain filter as a function of frequency.
21. The apparatus according to any one of claims 16 through 20 wherein the decoder:
obtains a measure of tonality of the audio signal represented by the set of subband signals; and adapts the scaling envelope in response to the measure of tonality.
22. The apparatus according to claim 21 that obtains the measure of tonality from the input signal.
23. The apparatus according to claim 21 wherein the decoder derives the measure of tonality from the way in which the zero-valued spectral components are arranged in the particular subband signal.
24. The apparatus according to any one of claims 16 through 23 wherein the synthesis filterbank is implemented by a block transform and:
the deformatter obtains a sequence of sets of subband signals from the input signal;
the decoder identifies a common subband signal in the sequence of sets of subband signals where, for each set in the sequence, one or more spectral components have a non-zero value and a plurality of spectral components have a zero value, identifies a common spectral component within the common subband signal that has a zero value in a plurality of adjacent sets in the sequence that are either preceded or followed by a set with the common spectral components having a non-zero value, scales the synthesized spectral components that correspond to the zero-valued common spectral components according to the scaling envelope that varies from set to set in the sequence according to temporal masking characteristics of the human auditory system;
and generates a sequence of modified sets of subband signals by substituting the synthesized spectral components for the corresponding zero-valued common spectral components in the sets; and the synthesis filterbank generates the audio information in response to the sequence of modified sets of subband signals.
25. The apparatus according to any one of claims 16 through 24 wherein the synthesis filterbank is implemented by a block transform and the decoder generates the synthesized spectral components by spectral translation of other spectral components in the set of subband signals.
26. The apparatus according to any one of claims 16 through 25 wherein the scaling envelope varies according to temporal masking characteristics of the human auditory system.
27. An apparatus for generating an output signal, wherein the apparatus comprises:
an analysis filterbank that generates in response to audio information a set of subband signals each having one or more spectral components representing spectral content of an audio signal;
quantizers coupled to the analysis filterbank that quantize the spectral components;
an encoder coupled to the quantizers that identifies within the set of subband signals a particular subband signal in which one or more spectral components have a non-zero value and are quantized by a quantizer having a minimum quantizing level that corresponds to a threshold and in which a plurality of spectral components have a zero value, derives scaling control information from the spectral content of the audio signal, wherein the scaling control information controls scaling of synthesized spectral components to be synthesized and substituted for the spectral components having a zero value in a receiver that generates audio information in response to the output signal;
and a formatter coupled to the encoder that generates the output signal by assembling the scaling control information and information representing the set of subband signals.
28. The apparatus according to claim 27 that:
obtains a measure of tonality of the audio signal represented by the set of subband signals; and derives the scaling control information from the measure of tonality.
29. The apparatus according to claim 27 or 28 comprising a modelling component that:
obtains an estimated psychoacoustic masking threshold of the audio signal represented by the set of subband signals; and derives the scaling control information from the estimated psychoacoustic masking threshold.
30. The apparatus according to any one of claims 27 through 29 that:
obtains two measures of spectral levels for portions of the audio signal represented by the non-zero-valued and the zero-valued spectral components;
and derives the scaling control information from the two measures of spectral levels.
31. A medium that conveys a program of instructions and is readable by a device for executing the program of instructions to perform a method for generating audio information, wherein the method comprises:
receiving an input signal and obtaining therefrom a set of subband signals each having one or more spectral components representing spectral content of an audio signal;
identifying within the set of subband signals a particular subband signal in which one or more spectral components have a non-zero value and are quantized by a quantizer having a minimum quantizing level that corresponds to a threshold, and in which a plurality of spectral components have a zero value;
generating synthesized spectral components that correspond to respective zero-valued spectral components in the particular subband signal and that are scaled according to a scaling envelope less than or equal to the threshold;
generating a modified set of subband signals by substituting the synthesized spectral components for corresponding zero-valued spectral components in the particular subband signal; and generating the audio information by applying a synthesis filterbank to the modified set of subband signals.
32. The medium according to claim 31 wherein the scaling envelope is uniform.
33. The medium according to claim 31 or 32 wherein the synthesis filterbank is implemented by a block transform that has spectral leakage between adjacent spectral components and the scaling envelope varies at a rate substantially equal to a rate of roll off of the spectral leakage of the block transform.
34. The medium according to any one of claims 31 through 33 wherein the synthesis filterbank is implemented by a block transform and the method comprises:
applying a frequency-domain filter to one or more spectral components in the set of subband signals; and deriving the scaling envelope from an output of the frequency-domain filter.
35. The medium according to claim 34 wherein the method comprises varying the response of the frequency-domain filter as a function of frequency.
36. The medium according to any one of claims 31 through 35 wherein the method comprises:
obtaining a measure of tonality of the audio signal represented by the set of subband signals; and adapting the scaling envelope in response to the measure of tonality.
37. The medium according to claim 36 wherein the method obtains the measure of tonality from the input signal.
38. The medium according to claim 36 wherein the method comprises deriving the measure of tonality from the way in which the zero-valued spectral components are arranged in the particular subband signal.
39. The medium according to any one of claims 31 through 38 wherein the synthesis filterbank is implemented by a block transform and the method comprises:
obtaining a sequence of sets of subband signals from the input signal;
identifying a common subband signal in the sequence of sets of subband signals where, for each set in the sequence, one or more spectral components have a non-zero value and a plurality of spectral components have a zero value;

identifying a common spectral component within the common subband signal that has a zero value in a plurality of adjacent sets in the sequence that are either preceded or followed by a set with the common spectral components having a non-zero value;
scaling the synthesized spectral components that correspond to the zero-valued common spectral components according to the scaling envelope that varies from set to set in the sequence according to temporal masking characteristics of the human auditory system;
generating a sequence of modified sets of subband signals by substituting the synthesized spectral components for the corresponding zero-valued common spectral components in the sets; and generating the audio information by applying the synthesis filterbank to the sequence of modified sets of subband signals.
40. The medium according to any one of claims 31 through 39 wherein the synthesis filterbank is implemented by a block transform and the method generates the synthesized spectral components by spectral translation of other spectral components in the set of subband signals.
41. The medium according to any one of claims 31 through 40 wherein the scaling envelope varies according to temporal masking characteristics of the human auditory system.
42. A medium that conveys a program of instructions and is readable by a device for executing the program of instructions to perform a method for generating an output signal, wherein the method comprises:
generating a set of subband signals each having one or more spectral components representing spectral content of an audio signal by quantizing information that is obtained by applying an analysis filterbank to audio information;
identifying within the set of subband signals a particular subband signal in which one or more spectral components have a non-zero value and are quantized by a quantizer having a minimum quantizing level that corresponds to a threshold, and in which a plurality of spectral components have a zero value;
deriving scaling control information from the spectral content of the audio signal, wherein the scaling control information controls scaling of synthesized spectral components to be synthesized and substituted for the spectral components having a zero value in a receiver that generates audio information in response to the output signal; and generating the output signal by assembling the scaling control information and information representing the set of subband signals.
43. The medium according to claim 42 wherein the method comprises:
obtaining a measure of tonality of the audio signal represented by the set of subband signals; and deriving the scaling control information from the measure of tonality.
44. The medium according to claim 42 or 43 wherein the method comprises:
obtaining an estimated psychoacoustic masking threshold of the audio signal represented by the set of subband signals; and deriving the scaling control information from the estimated psychoacoustic masking threshold.
45. The medium according to any one of claims 42 through 44 wherein the method comprises:
obtaining two measures of spectral levels for portions of the audio signal represented by the non-zero-valued and the zero-valued spectral components; and deriving the scaling control information from the two measures of spectral levels.
CA2736046A 2002-06-17 2003-05-30 Audio coding system using spectral hole filling Abandoned CA2736046A1 (en)

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US10/174,493 US7447631B2 (en) 2002-06-17 2002-06-17 Audio coding system using spectral hole filling
US10/174,493 2002-06-17
CA2489441A CA2489441C (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling

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CA2736046A Abandoned CA2736046A1 (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling
CA2736055A Expired - Lifetime CA2736055C (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling
CA2489441A Expired - Lifetime CA2489441C (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling
CA2735830A Expired - Lifetime CA2735830C (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling
CA2736065A Expired - Lifetime CA2736065C (en) 2002-06-17 2003-06-09 Audio coding system using characteristics of a decoded signal to adapt synthesized spectral components
CA2736060A Expired - Lifetime CA2736060C (en) 2002-06-17 2003-06-09 Audio coding system using characteristics of a decoded signal to adapt synthesized spectral components

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CA2736055A Expired - Lifetime CA2736055C (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling
CA2489441A Expired - Lifetime CA2489441C (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling
CA2735830A Expired - Lifetime CA2735830C (en) 2002-06-17 2003-05-30 Audio coding system using spectral hole filling
CA2736065A Expired - Lifetime CA2736065C (en) 2002-06-17 2003-06-09 Audio coding system using characteristics of a decoded signal to adapt synthesized spectral components
CA2736060A Expired - Lifetime CA2736060C (en) 2002-06-17 2003-06-09 Audio coding system using characteristics of a decoded signal to adapt synthesized spectral components

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US (4) US7447631B2 (en)
EP (6) EP1736966B1 (en)
JP (6) JP4486496B2 (en)
KR (5) KR100991448B1 (en)
CN (1) CN100369109C (en)
AT (7) ATE536615T1 (en)
CA (6) CA2736046A1 (en)
DE (3) DE60310716T8 (en)
DK (3) DK1514261T3 (en)
ES (1) ES2275098T3 (en)
HK (6) HK1070729A1 (en)
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PL (1) PL208344B1 (en)
PT (1) PT2216777E (en)
SG (3) SG177013A1 (en)
SI (2) SI2209115T1 (en)
TW (1) TWI352969B (en)
WO (1) WO2003107328A1 (en)

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DK2207169T3 (en) 2012-02-06
SG10201702049SA (en) 2017-04-27
CA2736060C (en) 2015-02-17
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CN1662958A (en) 2005-08-31
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JP2012103718A (en) 2012-05-31
DE60332833D1 (en) 2010-07-15
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ATE536615T1 (en) 2011-12-15
US7447631B2 (en) 2008-11-04
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PL208344B1 (en) 2011-04-29
DE60333316D1 (en) 2010-08-19
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US7337118B2 (en) 2008-02-26
IL216069A (en) 2015-11-30
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EP2207170A1 (en) 2010-07-14
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EP1514261A1 (en) 2005-03-16
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SI2207169T1 (en) 2012-05-31
JP4486496B2 (en) 2010-06-23
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US20090144055A1 (en) 2009-06-04
AU2003237295A1 (en) 2003-12-31
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