Low time-delay transform coder, decoder, and encoder/decoder for high-quality audio

• US 5,222,189 A
• Filed: 09/26/1990
• Issued: 06/22/1993
• Est. Priority Date: 01/27/1989
• Status: Expired due to Term

• Current Assignee: Dolby Laboratories Licensing Corporation (Dolby Laboratories Incorporated)
• Original Assignee: Dolby Laboratories Licensing Corporation (Dolby Laboratories Incorporated)
• Inventors: Fielder, Loius D.
• Primary Examiner(s): Knepper, David D.
• Application Number:
  US07/582,956
• Time in Patent Office: 1,000 Days
• Field of Search: 381/29-40, 395/2
• US Class Current: 704/229
• CPC Class Codes: G06T 9/005 Statistical coding, e.g. Huffman, run length coding
• CPC Class Codes: G10L 19/0212 using orthogonal transformation
• CPC Class Codes: H04B 1/665 using psychoacoustic properties of the ear, e.g. masking effect

Abstract

A low bit-rate (192 kBits per second) transform encoder/decoder system (44.1 kHz or 48 kHz sampling rate) for high-quality music applications employs short time-domain sample blocks (128 samples/block) so that the system signal propagation delay is short enough for real-time aural feedback to a human operator. Carefully designed pairs of analysis/synthesis windows are used to achieve sufficient transform frequency selectivity despite the use of short sample blocks. A synthesis window in the decoder has characteristics such that the product of its response and that of an analysis window in the encoder produces a composite response which sums to unity for two adjacent overlapped sample blocks. Adjacent time-domain signal samples blocks are overlapped and added to cancel the effects of the analysis and synthesis windows. A technique is provided for deriving suitable analysis/synthesis window pairs. In the encoder, a discrete transform having a function equivalent to the alternate application of a modified Discrete Cosine Transform and a modified Discrete Sine Transform according to the Time Domain Aliasing Cancellation technique or, alternatively, a Discrete Fourier Transform is used to generate frequency-domain transform coefficients. The transform coefficients are nonuniformly quantized by assigning a fixed number of bits and a variable number of bits determined adaptively based on psychoacoustic masking. A technique is described for assigning the fixed bit and adaptive bit allocations. The transmission of side information regarding adaptively allocated bits is not required. Error codes and protected data may be scattered throughout formatted frame outputs from the encoder in order to reduce sensitivity to noise bursts.

Claims

1. An encoder for the encoding of audio information comprising signal samples, said encoder comprising

means for receiving said signal samples,

subband means, including adaptive bit allocation means, for defining subbands and for generating subband information in response to said signal samples, said subband information for each of said subbands including one or more digital words, each of said digital words comprising an adaptive portion and a non-adaptive portion, wherein coding accuracy of said adaptive portion is established by said adaptive bit allocation means, and

formatting means for assembling digital information including said subband information into a digital output having a format suitable for transmission or storage.

2. An encoder according to claim 1 wherein the coding accuracy of said non-adaptive portion is less than the accuracy required to have no audible quantizing noise.

3. An encoder according to claim 1 wherein said subband means generates said subband information by applying a discrete transform function to blocks of said signal samples.

4. An encoder according to claim 1 wherein said subband means comprises filter bank means and means for storing coding information defining the coding accuracy for said non-adaptive portion, wherein said coding information is preestablished by comparing a representative frequency response for said filter bank means for each of said subbands to a corresponding psychoacoustic masking threshold representative of one or more of said subbands.

5. An encoder according to claim 4 wherein a psychoacoustic masking threshold having a relatively high selectivity for frequencies below a masking tone or narrow band of noise is taken as representative of the psychoacoustic masking threshold in lower frequency subbands and a psychoacoustic masking threshold having a relatively low selectivity for frequencies below a masking tone or narrow band of noise is taken as representative of the psychoacoustic masking threshold in higher frequency subbands.

6. An encoder according to claim 5 wherein a psychoacoustic masking threshold for a single tone or very narrow band of noise at about 1 kHz is taken as representative for subbands within the frequency range of about 500 Hz to 2 kHz and a psychoacoustic masking threshold for a single tone or very narrow band of noise at about 4 kHz is taken as representative for subbands above about 2 kHz.

7. An encoder according to claim 4 wherein said coding information defines said coding accuracy for said non-adaptive portion at a level less than the accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

8. An encoder according to claim 7 wherein said coding information defines said coding accuracy at a level two bits fewer than said accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

9. An encoder according to claim 1 or 4 wherein said subband means represents said subband information in block-floating-point form comprising one or more mantissas and one or more exponents, wherein said coding accuracy of said adaptive portion is based on an effective exponent value for each of said digital words, said effective exponent value derived from the value or values of said one or more exponents.

10. An encoder according to claim 9 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words equal to the value of the corresponding subband exponent.

11. An encoder according to claim 9 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, and one or more master exponents, each master exponent associated with a set of subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words derived from a combination of the values of the corresponding subband exponent and the associated master exponent.

12. An encoder according to claim 9, wherein subband information generated in response to an interval of said signal samples constitutes a subband information block, said subband means further comprising means for estimating the relative energy level of each subband represented in a subband information block, wherein said adaptive bit allocation means assigns bits to at least some digital words, said adaptive bit allocation means comprising

means for allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

means for allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subband of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

13. An encoder according to claim 12 wherein said certain number of bits is equal to said maximum number of bits.

14. An encoder according to claim 12 wherein said means for estimating the relative energy level estimates said relative energy level based upon the effective exponent value of each subband represented in a subband information block.

15. An encoder according to claim 14 wherein said means for estimating the relative energy level comprises

means for ascertaining the effective exponent value of the subband which contains the maximum of the values represented by each mantissa in combination with its associated effective exponent value, and

means for assigning a level number to each of all subbands represented in said subband information block, said level number equal to said maximum number of bits reduced by the absolute value of the difference between the ascertained effective exponent value and the effective exponent value corresponding to the subband for which a level is to be assigned, but in no case assigning a level number less than zero.

16. An encoder according to claim 12 wherein said means for allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

17. An encoder according to claim 12 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

18. An encoder according to claim 12 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said means further comprising a means for reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

19. An encoder according to claim 9 wherein said formatting means assembles bits representing said non-adaptive portion of each of said digital words and bits representing said one or more exponents apart from bits representing said adaptive portion of each of said digital words.

20. An encoder according to claim 19 wherein said formatting means assembles said digital information into frames and inserts the bits representing said non-adaptive portion of each of said digital words and the bits representing said one or more exponents into preestablished positions within a respective one of said frames.

21. An encoder according to claim 20 wherein said formatting means inserts into a respective one of said frames the bits representing said non-adaptive portion of each of said digital words and the bits representing said one or more exponents ahead of the bits representing said adaptive portion of each of said digital words.

22. An encoder according to claim 1 or 4, wherein subband information generated in response to an interval of said signal samples constitutes a subband information block, said subband means further comprising means for estimating the relative energy level of each subband represented in a subband information block, wherein said adaptive bit allocation means assigns bits to at least some digital words, said adaptive bit allocation means comprising

means for allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

means for allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subband of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

23. An encoder according to claim 22 wherein said certain number of bits is equal to said maximum number of bits.

24. An encoder according to claim 22 wherein said means for allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

25. An encoder according to claim 22 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

26. An encoder according to claim 22 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said means further comprising a means for reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

27. An encoder according to claim 1 or 4 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

28. An encoder according to claim 1 or 4 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said means further comprising a means for reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

29. An encoder according to claim 1 or 4 wherein said formatting means assembles bits representing said non-adaptive portion of each of said digital words apart from bits representing said adaptive portion of each of said digital words.

30. An encoder according to claim 29 wherein said formatting means assembles said digital information into frames and inserts the bits representing said non-adaptive portion of each of said digital words into pre-established positions within a respective one of said frames.

31. An encoder according to claim 30 wherein said formatting means inserts into a respective one of said frames the bits representing said non-adaptive portion of each of said digital words ahead of the bits representing said adaptive portion of each of said digital words.

32. An encoder for the encoding of audio information comprising signal samples, said encoder having a short signal propagation delay, comprising

means for receiving and grouping said signal samples into overlapping signal sample blocks, the length of the overlap constituting an overlap interval, said signal sample blocks having a time period resulting in a signal propagation delay short enough so that an encoding/decoding system employing the encoder is usable for real-time aural feedback to a human operator,

analysis-window means for weighting each signal sample block by an analysis window, wherein said analysis window constitutes one window of an analysis-synthesis window pair, wherein the product of both windows in said window pair is equal to a product window prederived from an analysis-only window permitting the design of a filter bank in which transform-based digital filters have the ability to trade off steepness of transition band rolloff against depth of stopband rejection in the filter characteristics, and wherein said product window overlapped with itself sums to a constant value across the overlap interval,

means for generating transform coefficients by applying a discrete transform function to each of said analysis-window weighted signal sample blocks,

means for quantizing each of said transform coefficients, and

formatting means for assembling the quantized transform coefficients into a digital output having a format suitable for transmission or storage.

33. An encoder according to claim 32 wherein said product window is derived from an analysis-only window selected from the set of the Kaiser-Bessel window, the Dolph-Chebyshev window, and windows derived from finite impulse filter coefficients using the Parks-McClellan method.

34. An encoder according to claim 32 wherein said means for generating transform coefficients alternately applies a modified Discrete Cosine Transform and a modified Discrete Sine Transform in accordance with the Time-Domain Aliasing Cancellation technique and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of four through seven.

35. An encoder according to claim 32 wherein said means for generating transform coefficients applies a Discrete Fourier Transform and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of one and one-half through three.

36. An encoder according to claim 32 wherein said product window is prederived by

(1) defining an initial window comprising substantially any window in said class of analysis windows having a length equal to one plus the number of samples in the overlap interval,

(2) defining a first unit pulse function, the duration of which is equal to the length of said signal blocks less the overlap interval,

(3) obtaining an interim window by convolving said initial window with said first unit pulse function,

(4) defining a scaling factor by convolving said initial window with a second unit pulse function of duration equal to one, and

(5) obtaining said product window by dividing each element of said interim window by said scaling factor.

37. An encoder according to claim 32 wherein said steepness of transition band rolloff is maximized for a desired depth of stopband rejection.

38. An encoder according to claim 37 wherein the desired depth of stopband rejection is determined empirically by listening tests.

39. An encoder according to claim 37 wherein said transition band rolloff generally follows the lower slope of the human ear'"'"'s psychoacoustic masking curve within a critical band.

40. A decoder for the reproduction of audio information comprising signal samples from a coded signal including digital information, said decoder comprising

deformatting means, including adaptive bit allocation means, for defining subbands and for deriving subband information in response to said coded signal, and for reconstructing digital words using said derived subband information, said digital words comprising an adaptive portion and a non-adaptive portion, wherein coding accuracy of said adaptive portion is established by said adaptive bit allocation means,

inverse subband means for generating signal samples in response to said subband information, and

means for generating said reproduction of audio information in response to said signal samples.

41. A decoder according to claim 40 wherein the coding accuracy of said non-adaptive portion is less than the accuracy required to have no audible quantizing noise.

42. A decoder according to claim 40 wherein said inverse subband means generates said signal samples by applying an inverse discrete transform function to blocks of said subband information.

43. A decoder according to claim 40 wherein said inverse subband means comprises inverse filter bank means and means for storing coding information defining the coding accuracy for said non-adaptive portion, wherein said coding information is preestablished by comparing a representative frequency response for said inverse filter bank means for each of said subbands to a corresponding psychoacoustic masking threshold representative of one or more of said subbands.

44. A decoder according to claim 43 wherein a psychoacoustic masking threshold having a relatively high selectivity for frequencies below a masking tone or narrow band of noise is taken as representative of the psychoacoustic masking threshold in lower frequency subbands and a psychoacoustic masking threshold having a relatively low selectivity for frequencies below a masking tone or narrow band of noise is taken as representative of the psychoacoustic masking threshold in higher frequency subbands.

45. A decoder according to claim 44 wherein a psychoacoustic masking threshold for a single tone or very narrow band of noise at about 1 kHz is taken as representative for subbands within the frequency range of about 500 Hz to 2 kHz and a psychoacoustic masking threshold for a single tone or very narrow band of noise at about 4 kHz is taken as representative for subbands above about 2 kHz.

46. A decoder according to claim 43 wherein said coding information defines said coding accuracy for said non-adaptive portion at a level less than the accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

47. A decoder according to claim 46 wherein said coding information defines said coding accuracy at a level two bits fewer than said accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

48. A decoder according to claim 40 or 43 wherein said subband information is expressed in block-floating-point form comprising one or more mantissas and one or more exponents, wherein said coding accuracy of said adaptive portion is based on an effective exponent value for each of said digital words, said effective exponent value derived from the value or values of said one or more exponents.

49. A decoder according to claim 48 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words equal to the value of the corresponding subband exponent.

50. A decoder according to claim 48 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, and one or more master exponents, each master exponent associated with a set of subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words derived from a combination of the values of the corresponding subband exponent and the associated master exponent.

51. A decoder according to claim 48 wherein said derived subband information generated in response to an interval of said coded signal constitutes a subband information block, said decoder further comprising means for estimating the relative energy level of each subband represented in a subband information block, and wherein said adaptive bit allocation means assigns bits to at least some digital words, said adaptive bit allocation means comprising

means for allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

means for allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subband of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

52. A decoder according to claim 51 wherein said certain number of bits is equal to said maximum number of bits.

53. A decoder according to claim 51 wherein said means for estimating the relative energy level estimates said relative energy level based upon the effective exponent value.

54. A decoder according to claim 53 wherein said means for estimating the relative energy level comprises

means for ascertaining the effective exponent value of the subband which contains the maximum of the values represented by each mantissa in combination with its associated effective exponent value, and

means for assigning a level number to each of all subbands represented in said subband information block, said level number equal to said maximum number of bits reduced by the absolute value of the difference between the ascertained effective exponent value and the effective exponent value corresponding to the subband for which a level is to be assigned, but in no case assigning a level number less than zero.

55. A decoder according to claim 51 wherein said means for allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

56. A decoder according to claim 51 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

57. A decoder according to claim 51 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said means further comprising a means for reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

58. A decoder according to claim 48 wherein said deformatting means reconstructs each digital word from bits representing said non-adaptive portion and bits representing said one or more exponents assembled in said coded signal apart from bits representing said adaptive portion.

59. A decoder according to claim 58 wherein said deformatting means reconstructs each digital words from bits representing said non-adaptive portion and bits representing said one or more exponents which occupy pre-established positions within said subband information block.

61. A decoder according to claim 40 or 43 wherein said derived subband information generated in response to an interval of said coded signal constitutes a subband information block, said decoder further comprising means for estimating the relative energy level of each subband represented in a subband information block, and wherein said adaptive bit allocation means assigns bits to at least some digital words, said adaptive bit allocation means comprising

means for allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

means for allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subband of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

62. A decoder according to claim 61 wherein said certain number of bits is equal to said maximum number of bits.

63. A decoder according to claim 61 wherein said means for allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

64. A decoder according to claim 61 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

65. A decoder according to claim 61 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said means further comprising a means for reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

66. A decoder according to claim 40 or 43 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

67. A decoder according to claim 40 or 43 wherein said adaptive bit allocation means stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said means further comprising a means for reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

68. A decoder according to claim 40 or 43 wherein said deformatting means reconstructs each digital word from bits representing said non-adaptive portion assembled in said coded signal apart from bits representing said adaptive portion.

69. A decoder according to claim 68 wherein said deformating means reconstructs each digital word from bits representing said non-adaptive portion which occupy pre-established positions within said subband information block.

60. A decoder according to 59 wherein said deformatting means reconstructs each digital word from bits representing said non-adaptive portion and bits representing said one or more exponents which occupy positions in said subband information block ahead of bits representing said adaptive portion.

70. A decoder according to 69 wherein said deformatting means reconstructs each digital word from bits representing said non-adaptive portion which occupy positions in said subband information block ahead of bits representing said adaptive portion.

71. A decoder for the reproduction of audio information comprising signal samples from a coded signal generated by an encoder that groups said signal samples into overlapping signal sample blocks, the length of the overlap constituting an overlap interval, weights each sample block with an analysis window, generates transform coefficients by applying a discrete transform to the analysis-window weighted signal sample blocks, quantizes each transform coefficient and assembles the quantized transform coefficients into a digital output having a format suitable for transmission or storage, said decoder comprising

means for receiving said digital output for deriving said quantized transform coefficients therefrom,

means for reconstructing decoded transform coefficients from the deformatted quantized transform coefficients,

means for generating signal sample blocks by applying an inverse discrete transform function to said decoded transform coefficients, said inverse discrete transform having characteristics inverse to those of said discrete transform in the encoder, said signal sample blocks having a time period resulting in a signal propagation delay short enough so that an encoding/decoding system employing the decoder is usable for real-time aural feedback to a human operator,

synthesis window means for weighting the signal sample blocks by a synthesis window, wherein a product window equal to the product of said synthesis window and said analysis window is prederived from an analysis-only window permitting the design of a filter bank in which transform-based digital filters have the ability to trade off steepness of transition band rolloff against depth of stopband rejection in the filter characteristics, and wherein said product window overlapped with itself sums to a constant value across the overlap interval, and

means for cancelling the weighting effects of the analysis window means and the synthesis window means to recover said signal samples by adding overlapped signal sample blocks across said overlap interval.

72. A decoder according to claim 71 wherein said product window is derived from an analysis-only window selected from the set of the Kaiser-Bessel window, the Dolph-Chebyshev window, and windows derived from finite impulse filter coefficients using the Parks-McClellan method.

73. A decoder according to claim 71 wherein said means for generating transform coefficients alternately applies an inverse modified Discrete Cosine Transform and an inverse modified Discrete Sine Transform in accordance with the Time-Domain Aliasing Cancellation technique and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of four through seven.

74. A decoder according to claim 71 wherein said means for generating transform coefficients applies an inverse Discrete Fourier Transform and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of one and onehalf through three.

75. A decoder according to claim 71 wherein said product window is prederived by

(1) defining an initial window comprising substantially any window in said class of analysis windows having a length equal to one plus the number of samples in the overlap interval,

(2) defining a first unit pulse function the duration of which is equal to the length of said signal blocks less the overlap interval,

(3) obtaining an interim window by convolving said initial window with said first unit pulse function,

(4) defining a scaling factor by convolving said initial window with a second unit pulse function of duration equal to one, and

(5) obtaining said product window by dividing each element of said interim window by said scaling factor.

76. A decoder according to claim 71 wherein said steepness of transition band rolloff is maximized for a desired depth of stopband rejection.

77. A decoder according to claim 76 wherein the desired depth of stopband rejection is determined empirically by listening tests.

78. A decoder according to claim 76 wherein said transition band rolloff generally follows the lower slope of the human ear'"'"'s psychoacoustic masking curve within a critical band.

79. An encoding method for the encoding of audio information comprising signal samples, said encoding method comprising

receiving said signal samples,

defining subbands and generating subband information in response to said signal samples, said subband information for each of said subbands including one or more digital words, each of said digital words comprising an adaptive portion and a non-adaptive portion, wherein coding accuracy of said adaptive portion is established by adaptive bit allocating, and

assembling digital information including said subband information into a digital output having a format suitable for transmission or storage.

80. An encoding method according to claim 79 wherein the coding accuracy of said non-adaptive portion is less than the accuracy required to have no audible quantizing noise.

81. An encoding method according to claim 79 wherein said generating subband information applies a discrete transform function to blocks of said signal samples.

82. An encoding method according to claim 79 wherein said generating subband information comprises filtering and storing coding information defining the coding accuracy for said non-adaptive portion, wherein said coding information is preestablished by comparing a representative frequency response for said filtering for each of said subbands to a corresponding psychoacoustic masking threshold representative of one or more of said subbands.

83. An encoding method according to claim 82 wherein a psychoacoustic masking threshold having a relatively high selectivity for frequencies below a masking tone or narrow band of noise is taken as representative of the psychoacoustic masking threshold in lower frequency subbands and a psychoacoustic masking threshold having a relatively low selectivity for frequencies below a masking tone or narrow band of noise is taken as representative of the psychoacoustic masking threshold in higher frequency subbands.

84. An encoding method according to claim 83 wherein a psychoacoustic masking threshold for a single tone or very narrow band of noise at about 1 kHz is taken as representative for subbands within the frequency range of about 500 Hz to 2 kHz and a psychoacoustic masking threshold for a single tone or very narrow band of noise at about 4 kHz is taken as representative for subbands above about 2 kHz.

85. An encoding method according to claim 82 wherein said coding information defines said coding accuracy for said non-adaptive portion at a level less than the accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

86. An encoding method according to claim 85 wherein said coding information defines said coding accuracy at a level two bits fewer than said accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

87. An encoding method according to claim 79 or 82 wherein said generating subband information represents said subband information in block-floating-point form comprising one or more mantissas and one or more exponents, wherein said coding accuracy of said adaptive portion is based on an effective exponent value for each of said digital words, said effective exponent value derived from the value or values of said one or more exponents.

88. An encoding method according to claim 87 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words equal to the value of the corresponding subband exponent.

89. An encoding method according to claim 87 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, and one or more master exponents, each master exponent associated with a set of subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words derived from a combination of the values of the corresponding subband exponent and the associated master exponent.

90. An encoding method according to claim 87, wherein subband information generated in response to an interval of said signal samples constitutes a subband information block, said generating subband information further comprising estimating the relative energy level of each subband represented in a subband information block, wherein said adaptive bit allocating assigns bits to at least some digital words, said adaptive bit allocating comprising

allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subband of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

91. An encoding method according to claim 90 wherein said certain number of bits is equal to said maximum number of bits.

92. An encoding method according to claim 90 wherein said estimating the relative energy level estimates said relative energy level based upon the effective exponent value of each subband represented in a subband information block.

93. An encoding method according to claim 92 wherein said estimating the relative energy level comprises

ascertaining the effective exponent value of the subband which contains the maximum of the values represented by each mantissa in combination with its associated effective exponent value, and

assigning a level number to each of all subbands represented in said subband information block, said level number equal to said maximum number of bits reduced by the absolute value of the difference between the ascertained effective exponent value and the effective exponent value corresponding to the subband for which a level is to be assigned, but in no case assigning a level number less than zero.

94. An encoding method according to claim 90 wherein said allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

95. An encoding method according to claim 90 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

96. An encoding method according to claim 90 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said adaptive bit allocating further comprising reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

97. An encoding method according to claim 87 wherein said assembling digital information assembles bits representing said non-adaptive portion of each of said digital words and bits representing said one or more exponents apart from bits representing said adaptive portion of each of said digital words.

98. An encoding method according to claim 97 wherein said assembling digital information assembles said digital information into frames and inserts the bits representing said non-adaptive portion of each of said digital words and the bits representing said one or more exponents into pre-established positions within a respective one of said frames.

99. An encoding method according to claim 98 wherein said assembling digital information inserts into a respective one of said frames the bits representing said non-adaptive portion of each of said digital words and the bits representing said one or more exponents ahead of the bits representing said adaptive portion of each of said digital words.

100. An encoding method according to claim 79 or 82, wherein subband information generated in response to an interval of said signal samples constitutes a subband information block, said generating subband information further comprising estimating the relative energy level of each subband represented in a subband information block, wherein said adaptive bit allocating assigns bits to at least some digital words, said adaptive bit allocating comprising

allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subband of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

101. An encoding method according to claim 100 wherein said certain number of bits is equal to said maximum number of bits.

102. An encoding method according to claim 100 wherein said allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

103. An encoding method according to claim 100 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

104. An encoding method according to claim 100 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said adaptive bit allocating further comprising reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

105. An encoding method according to claim 79 or 82 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

106. An encoding method according to claim 79 or 82 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said adaptive bit allocating further comprising reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

107. An encoding method according to claim 79 or 82 wherein said assembling digital information assembles bits representing said non-adaptive portion of each of said digital words apart from bits representing said adaptive portion of each of said digital words.

108. An encoding method according to claim 107 wherein said assembling digital information assembles said digital information into frames and inserts the bits representing said non-adaptive portion of each of said digital words into pre-established positions within a respective one of said frames.

109. An encoding method according to claim 108 wherein said assembling digital information inserts into a respective one of said frames the bits representing said non-adaptive portion of each of said digital words ahead of the bits representing said adaptive portion of each of said digital words.

110. An encoding method for the encoding of audio information comprising signal samples, said encoding method having a short signal propagation delay, comprising

receiving and grouping said signal samples into overlapping signal sample blocks, the length of the overlap constituting an overlap interval, said signal sample blocks having a time period resulting in a signal propagation delay short enough so that an encoding/decoding method employing the encoding method is usable for real-time aural feedback to a human operator,

weighting each signal sample block by an analysis window, wherein said analysis window constitutes one window of an analysis-synthesis window pair, wherein the product of both windows in said window pair is equal to a product window prederived from an analysis-only window permitting the design of a filter bank in which transform-based digital filters have the ability to trade off steepness of transition band rolloff against depth of stopband rejection in the filter characteristics, and wherein said product window overlapped with itself sums to a constant value across the overlap interval,

generating transform coefficients by applying a discrete transform function to each of said analysis-window weighted signal sample blocks,

quantizing each of said transform coefficients, and

assembling the quantized transform coefficients into a digital output having a format suitable for transmission or storage.

111. An encoding method according to claim 110 wherein said product window is derived from an analysis-only window selected from the set of the Kaiser-Bessel window, the Dolph-Chebyshev window, and windows derived from finite impulse filter coefficients using the Parks-McClellan method.

112. An encoding method according to claim 110 wherein said generating transform coefficients alternately applies a modified Discrete Cosine Transform and a modified Discrete Sine Transform in accordance with the Time-Domain Aliasing Cancellation technique and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of four through seven.

113. An encoding method according to claim 110 wherein said generating transform coefficients applies a Discrete Fourier Transform and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of one and one-half through three.

114. An encoding method according to claim 110 wherein said product window is prederived by

(1) defining an initial window comprising substantially any window in said class of analysis windows having a length equal to one plus the number of samples in the overlap interval,

(2) defining a first unit pulse function, the duration of which is equal to the length of said signal blocks less the overlap interval,

(3) obtaining an interim window by convolving said initial window with said first unit pulse function,

(4) defining a scaling factor by convolving said initial window with a second unit pulse function of duration equal to one, and

(5) obtaining said product window by dividing each element of said interim window by said scaling factor.

115. An encoding method according to claim 110 wherein said steepness of transition band rolloff is maximized for a desired depth of stopband rejection.

116. An encoding method according to claim 115 wherein the desired depth of stopband rejection is determined empirically by listening tests.

117. An encoding method according to claim 115 wherein said transition band rolloff generally follows the lower slope of the human ear'"'"'s psychoacoustic masking curve within a critical band.

118. A decoding method for the reproduction of audio information comprising signal samples from a coded signal including digital information, said decoding method comprising

defining subbands and deriving subband information in response to said coded signal, and reconstructing digital words using said derived subband information, said digital words comprising an adaptive portion and a non-adaptive portion, wherein coding accuracy of said adaptive portion is established by adaptive bit allocating,

generating signal samples in response to said subband information, and

generating said reproduction of audio information in response to said signal samples.

119. A decoding method according to claim 118 wherein the coding accuracy of said non-adaptive portion is less than the accuracy required to have no audible quantizing noise.

120. A decoding method according to claim 118 wherein said generating signal samples applies an inverse discrete transform function to blocks of said subband information.

121. A decoding method according to claim 118 wherein said generating signal sample blocks comprises inverse filtering and storing coding information defining the coding accuracy for said non-adaptive portion, wherein said coding information is preestablished by comparing a representative frequency response for said inverse filter bank for each of said subbands to a corresponding psychoacoustic masking threshold representative of one or more of said subbands.

122. A decoding method according to claim 121 wherein a psychoacoustic masking threshold having a relatively high selectivity for frequencies below a masking tone or narrow band of noise is taken as representative of the psychoacoustic masking threshold in lower frequency subbands and a psychoacoustic masking threshold having a relatively low selectively for frequencies below a masking tone or narrow band of noise is taken as representative of the pyschoacoustic masking threshold in higher frequency subbands.

123. A decoding method according to claim 122 wherein a psychoacoustic masking threshold for a single tone or very narrow band of noise of about 1 kHz is taken as representative for subbands within the frequency range of about 500 Hz to 2 kHz and a psychoacoustic masking threshold for a single tone or very narrow band of noise at about 4 kHz is taken as representative for subbands above about 2 kHz.

124. A decoding method according to claim 121 wherein said coding information defines said coding accuracy for said non-adaptive portion at a level less than the accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

125. A decoding method according to claim 124 wherein said coding information defines said coding accuracy at a level two bits fewer than said accuracy required to have no quantizing noise in excess of said corresponding psychoacoustic masking threshold.

126. A decoding method according to claim 118 or 121 wherein said subband information is expressed in block-floating-point form comprising one or more mantissas and one or more exponents, wherein said coding accuracy of said adaptive portion is based on an effective exponent value for each of said digital words, said effective exponent value derived from the value or values of said one or more exponents.

127. A decoding method according to claim 126 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words equal to the value of the corresponding subband exponent.

128. A decoding method according to claim 126 wherein said subband information comprises one or more mantissas and a subband exponent for each of said subbands, and one or more master exponents, each master exponent associated with a set of subbands, each of said mantissas corresponding to a respective one of said digital words, said effective exponent value for each of said digital words derived from a combination of the values of the corresponding subband exponent and the associated master exponent.

129. A decoding method according to claim 126 wherein said derived subband information generated in response to an interval of said coded signal constitutes a subband information block, said decoding method further comprising estimating the relative energy level of each subband represented in a subband information block, and wherein said adaptive bit allocating assigns bits to at least some digital words, said adaptive bit allocating comprising

allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subbands of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

130. A decoding method according to claim 129 wherein said certain number of bits is equal to said maximum number of bits.

131. A decoding method according to claim 129 wherein said estimating the relative energy level estimates said relative energy level based upon the effective exponent value.

132. A decoding method according to claim 131 wherein said estimating the relative energy level comprises

ascertaining the effective exponent value of the subband which contains the maximum of the values represented by each mantissa in combination with its associated effective exponent value, and

assigning a level number to each of all subbands represented in said subband information block, said level number equal to said maximum number of bits reduced by the absolute value of the difference between the ascertained effective exponent value and the effective exponent value corresponding to the subband for which a level is to be assigned, but in no case assigning a level number less than zero.

133. A decoding method according to claim 129 wherein said allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

134. A decoding method according to claim 129 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

135. A decoding method according to claim 129 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said adaptive bit allocating further comprising reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

136. A decoding method according to claim 126 wherein said reconstructing digital words reconstructs each digital word from bits representing said non-adaptive portion and bits representing said one or more exponents assembled in said coded signal apart from bits representing said adaptive portion.

137. A decoding method according to claim 136 wherein said reconstructing digital words reconstructs each digital words from bits representing said non-adaptive portion and bits representing said one or more exponents which occupy pre-established positions within said subband information block.

139. A decoding method according to claim 118 or 121 wherein said derived subband information generated in response to an interval of said coded signal constitutes a subband information block, said decoding method further comprising estimating the relative energy level of each subband represented in a subband information block, and wherein said adaptive bit allocating assigns bits to at least some digital words, said adaptive bit allocating comprising

allocating at most a maximum number of bits to each of the digital words of a first group of subbands possessing the greatest energy levels and stopping when a certain number of bits has been allocated to each of the digital words of said first group of subbands, and

allocating bits to the digital words of a second group of subbands adjoining subbands in which each of the digital words have been allocated said certain number of bits, each of the subbands of said second group of subbands constituting one subband of a pair of subbands immediately adjacent to said subbands in which digital words have been allocated said certain number of bits.

140. A decoding method according to claim 139 wherein said certain number of bits is equal to said maximum number of bits.

141. A decoding method according to claim 139 wherein said allocating bits to the digital words constituting said second group of subbands allocates bits to the digital words of said adjacent subbands on the low-frequency side before bits are allocated to the digital words of said adjacent subbands on the high-frequency side.

142. A decoding method according to claim 139 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

143. A decoding method according to claim 139 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said adaptive bit allocating further comprising reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

144. A decoding method according to claim 118 or 121 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals a limited number of adaptively allocatable bits.

145. A decoding method according to claim 118 or 121 wherein said adaptive bit allocating stops allocating bits when the number of bits allocated equals or exceeds a limited number of adaptively allocatable bits, said adaptive bit allocating further comprising reducing the number of bits adaptively allocated to selected digital words until the number of bits adaptively allocated equals said limited number of adaptively allocatable bits.

146. A decoding method according to claim 118 or 121 wherein said reconstructing digital words reconstructs each digital word from bits representing said non-adaptive portion assembled in said coded signal apart from bits representing said adaptive portion.

147. A decoding method according to claim 146 wherein said reconstructing digital words reconstructs each digital word from bits representing said non-adaptive portion which occupy pre-established positions within said subband information block.

138. A decoding method according to 137 wherein said reconstructing digital words reconstructs each digital word from bits representing said non-adaptive portion and bits representing said one or more exponents which occupy positions in said subband information block ahead of bits representing said adaptive portion.

148. A decoding method according to 147 wherein said reconstructing digital words reconstructs each digital word from bits representing said non-adaptive portion which occupy positions in said subband information block ahead of bits representing said adaptive portion.

149. A decoding method for the reproduction of audio information comprising signal samples from a coded signal generated by an encoding method that groups said signal samples into overlapping signal sample blocks, the length of the overlap constituting an overlap interval, weights each sample block with an analysis window, generates transform coefficients by applying a discrete transform to the analysis-window weighted signal sample blocks, quantizes each transform coefficient and assembles the quantized transform coefficients into a digital output having a format suitable for transmission or storage, said decoding method comprising

receiving said digital output for deriving said quantized transform coefficients therefrom,

reconstructing decoded transform coefficients from the deformatted quantized transform coefficients,

generating signal sample blocks by applying an inverse discrete transform function to said decoded transform coefficients, said inverse discrete transform having characteristics inverse to those of said discrete transform in the encoding method, said signal sample blocks having a time period resulting in a signal propagation delay short enough so that an encoding/decoding method employing the decoding method is usable for real-time aural feedback to a human operator,

weighting the signal sample blocks by a synthesis window, wherein a product window equal to the product of said synthesis window and said analysis window is prederived from an analysis-only window permitting the design of a filter bank in which transform-based digital filters have the ability to trade off steepness of transition band rolloff against depth of stopband rejection in the filter characteristics, and wherein said product window overlapped with itself sums to a constant value across the overlap interval, and

cancelling the weighting effects of the analysis window and the synthesis window to recover said signal samples by adding overlapped signal sample blocks across said overlap interval.

150. A decoding method according to claim 149 wherein said product window is derived from an analysis-only window selected from the set of the Kaiser-Bessel window, the Dolph-Chebyshev window, and windows derived from finite impulse filter coefficients using the Parks-McClellan method.

151. A decoding method according to claim 149 wherein said generating transform coefficients alternately applies an inverse modified Discrete Cosine Transform and an inverse modified Discrete Sine Transform in accordance with the Time-Domain Aliasing Cancellation technique and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of four through seven.

152. A decoding method according to claim 151 wherein said generating transform coefficients applies an inverse Discrete Fourier Transform and wherein said product window is derived from a Kaiser-Bessel window having an alpha value in the range of one and one-half through three.

153. A decoding method according to claim 149 wherein said product window is prederived by

(1) defining an initial window comprising substantially any window in said class of analysis windows having a length equal to one plus the number of samples in the overlap interval,

(2) defining a first unit pulse function the duration of which is equal to the length of said signal blocks less the overlap interval,

(3) obtaining an interim window by convolving said initial window with said first unit pulse function,

(4) defining a scaling factor by convolving said initial window with a second unit pulse function of duration equal to one, and

(5) obtaining said product window by dividing each element of said interim window by said scaling factor.

154. A decoding method according to claim 149 wherein said steepness of transition band rolloff is maximized for a desired depth of stopband rejection.

155. A decoding method according to claim 154 wherein the desired depth of stopband rejection is determined empirically by listening tests.

156. A decoding method according to claim 154 wherein said transition band rolloff generally follows the lower slope of the human ear'"'"'s psychoacoustic masking curve within a critical band.

157. A method for defining coding information which defines the coding accuracy of digital words representing spectral information in a plurality of frequency subbands, said digital words generated in response to an input signal by a split-band encoder comprising a filter bank, wherein said coding information comprises a nonadaptive coding accuracy, said method comprising

(1) obtaining a predicted quantizing noise spectrum of said split-band encoder for a frequency subband based upon a representative frequency response of said filter bank for said frequency subband,

(2) generating a subband value equal to the number of bits required to quantize spectral energy within said frequency subband such that said predicted quantizing noise spectrum does not exceed a representative psychoacoustic masking threshold for spectral energy within said frequency subband,

(3) setting said nonadaptive coding accuracy for said frequency subband equal to or less than said subband value, and

(4) reiterating the previous steps for each of said plurality of frequency subbands.

158. A method according to claim 157 wherein said nonadaptive coding accuracy for at least one of said plurality of frequency subbands is set equal to a value less than the respective subband value.