SAMPLE RATE SCALABLE LOSSLESS AUDIO CODING

• US 20140214431A1
• Filed: 06/27/2012
• Published: 07/31/2014
• Est. Priority Date: 07/01/2011
• Status: Abandoned Application

• Current Assignee: Dolby Laboratories Licensing Corporation (Dolby Laboratories Incorporated)
• Original Assignee: Dolby Laboratories Licensing Corporation (Dolby Laboratories Incorporated)
• Inventors: Vinton, Mark S., Robinson, Charles Q.
• Application Number:
  US14/126,895
• Publication Number:
  US 20140214431A1
• Time in Patent Office: Days
• US Class Current: 704/500
• CPC Class Codes: G10L 19/0017 Lossless audio signal coding; Perfect reconstruction of coded audio signal by transmission of coding error G10L19/24 takes precedence
• CPC Class Codes: G10L 19/24 Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding

Abstract

A transmitter in an audio coding system generates an encoded audio signal that conveys a losslessly encoded representation of an audio signal at a first sample rate and losslessly encoded representations of related audio information at other sample rates. A companion receiver with limited computational resources can generate a high-quality output audio signal at a desired sample rate by losslessly decoding the encoded representation of the audio signal and possibly other portions of the encoded audio signal as needed to obtain an output signal at one of the other sample rates.

Claims

1-41. -41. (canceled)

42. A method for encoding an audio signal, wherein the method comprises:

obtaining a first signal comprising digital samples that represent the audio signal at a first sample rate;

wherein the first signal is obtained from a delay of an input signal comprising digital samples that represent the audio signal at the first sample rate;

obtaining a second signal comprising digital samples that represent the audio signal at a second sample rate that is higher than the first sample rate;

wherein the second digital signal is obtained from a sample-rate conversion of the input signal;

converting the first signal into a first interim signal comprising digital samples at the second sample rate;

wherein a quality of the sample-rate conversion for obtaining the first interim signal is lower than a quality of the sample-rate conversion for obtaining the second digital signal;

generating a first difference signal comprising samples at the second sample rate that represent a difference between corresponding samples of the second signal and the first interim signal;

applying a first lossless encoder to the first signal to generate a first encoded signal comprising samples representing the first signal at the first sample rate, wherein the first lossless encoder adapts its operation in response to first coding parameters;

applying a second lossless encoder to the first difference signal to generate a second encoded signal comprising samples representing the first difference signal at the second sample rate, wherein the second lossless encoder adapts its operation in response to second coding parameters; and

generating an encoded output signal that conveys the first encoded signal, the second encoded signal, and representations of the first coding parameters and the second coding parameters.

43. The method of claim 42 that comprises:

obtaining a third signal comprising digital samples that represent the audio signal at a third sample rate that is higher than the second sample rate;

converting either the first or the second signal into a second interim signal comprising digital samples at the third sample rate;

generating a second difference signal comprising samples at the third sample rate that represent a difference between corresponding samples of the third signal and the second interim signal;

applying a third lossless encoder to the second difference signal to generate a third encoded signal comprising samples representing the second difference signal at the third sample rate, wherein the third lossless encoder adapts its operation to third coding parameters; and

generating the encoded output signal so that it also conveys the third encoded signal and a representation of the third coding parameters.

44. The method of claim 43 that comprises:

obtaining the third digital signal from a sample-rate conversion of either the input signal or the second signal.

45. The method of claim 44 that comprises:

applying the first lossless encoder to the first signal to generate a first prediction error signal and applying an entropy encoder to the first prediction error signal to generate the first encoded signal, wherein generation of the first prediction error signal comprises applying a first prediction filter to the first signal, and wherein the first lossless encoder adapts its operation by adapting the first prediction filter in response to the first coding parameters that represent prediction filter coefficients;

applying the second lossless encoder to the first difference signal to generate a second prediction error signal and applying an entropy encoder to the second prediction error signal to generate the second encoded signal, wherein generation of the second prediction error signal comprises applying a second prediction filter to the first difference signal, and wherein the second lossless encoder adapts its operation by adapting the second prediction filter in response to the second coding parameters that represent prediction filter coefficients; and

applying the third lossless encoder by to the second difference signal to generate a third prediction error signal and applying an entropy encoder to the third prediction error signal to generate the third encoded signal, wherein generation of the third prediction error signal comprises applying a third prediction filter to the second difference signal, and wherein the third lossless encoder adapts its operation by adapting the third prediction filter in response to the third coding parameters that represent prediction filter coefficients.

46. The method of claim 42 that comprises:

applying the first lossless encoder to the first signal to generate a first prediction error signal and applying an entropy encoder to the first prediction error signal to generate the first encoded signal, wherein generation of the first prediction error signal comprises applying a first prediction filter to the first signal, and wherein the first lossless encoder adapts its operation by adapting the first prediction filter in response to the first coding parameters that represent prediction filter coefficients; and

applying the second lossless encoder to the first difference signal to generate a second prediction error signal and applying an entropy encoder to the second prediction error signal to generate the second encoded signal, wherein generation of the second prediction error signal comprises applying a second prediction filter to the first difference signal, and wherein the second lossless encoder adapts its operation by adapting the second prediction filter in response to the second coding parameters that represent prediction filter coefficients.

47. An apparatus (100) for encoding an audio signal, wherein the apparatus comprises:

a first terminal (111) that receives a first signal comprising digital samples that represent the audio signal at a first sample rate;

a delay (102) that provides the first signal to the first terminal (111) as a delayed version of an input signal, wherein the input signal comprises digital samples that represent the audio signal at the first sample rate;

a second terminal (121) that receives a second signal comprising digital samples that represent the audio signal at a second sample rate that is higher than the first sample rate;

a sample-rate converter (103) that provides the second signal to the second terminal by converting the input signal to the second sample rate;

a first sample-rate converter (112) coupled to the first terminal (111) to convert the first signal into a first interim signal comprising digital samples at the second sample rate;

wherein a quality of the sample-rate conversion for obtaining the first interim signal is lower than a quality of the sample-rate conversion for obtaining the second signal;

a first difference calculator (122) coupled to the first sample-rate converter (112) and the second terminal (121) to generate a first difference signal comprising samples at the second sample rate that represent a difference between corresponding samples of the second signal and the first interim signal;

a first lossless encoder (116) coupled to the first terminal (111) to be applied to the first signal to generate a first encoded signal comprising samples representing the first signal at the first sample rate, wherein the first lossless encoder (111) adapts its operation in response to first coding parameters;

a second lossless encoder (126) coupled to the first difference calculator (122) to be applied to the first difference signal to generate a second encoded signal comprising samples representing the first difference signal at the second sample rate, wherein the second lossless encoder adapts its operation in response to second coding parameters; and

a formatter (108) coupled to the first lossless encoder (116) and the second lossless encoder (126) to generate an encoded output signal that conveys encoded representations of the first encoded signal, the second encoded signal, the first coding parameters and the second coding parameters.

48. The apparatus (100) of claim 47 that comprises:

a third terminal (131) that receives a third signal comprising digital samples that represent the audio signal at a third sample rate that is higher than the second sample rate;

a second sample-rate converter (114) that either is coupled to the first terminal (111) to convert the first signal into a second interim signal comprising digital samples at the third sample rate or is coupled to the second terminal (121) to convert the second signal into the second interim signal;

a second difference calculator (132) coupled to second sample-rate converter (114) and the third terminal (131) to generate a second difference signal comprising samples at the third sample rate that represent a difference between corresponding samples of the third signal and the second interim signal;

a third lossless encoder (136) coupled to the second difference calculator (132) to be applied to the second difference signal to generate a third encoded signal comprising samples representing the second difference signal at the third sample rate, wherein the third lossless encoder adapts its operation in response to third coding parameters; and

the formatter (108) coupled to the third lossless encoder (136) to generate the encoded output signal so that it also conveys encoded representations of the third encoded signal and the third coding parameters.

49. The apparatus (100) of claim 48 that comprises:

a second sample-rate converter (104) that provides the third signal to the third terminal (131) by converting either the input signal or the second signal to the third sample rate.

50. The apparatus (100) of claim 48, wherein:

the first lossless encoder (116) comprises a first prediction filter (45) coupled to the first terminal (111) and a first entropy encoder (47) coupled to the first prediction filter (45) to generate the first encoded signal, wherein the first prediction filter (45) adapts its operation in response to the first coding parameters that represent prediction filter coefficients;

the second lossless encoder (126) comprises a second prediction filter coupled to the first difference calculator (122) and a second entropy encoder coupled to the second prediction filter to generate the second encoded signal, wherein the second prediction filter adapts its operation in response to the second coding parameters that represent prediction filter coefficients; and

the third lossless encoder (136) comprises a third prediction filter coupled to the second difference calculator (132) and a third entropy encoder coupled to the third prediction filter to generate the third encoded signal, wherein the third prediction filter adapts its operation in response to the third coding parameters that represent prediction filter coefficients.

51. The apparatus (100) of claim 47 that comprises:

the first lossless encoder (116) comprises a first prediction filter (45) coupled to the first terminal (111) and a first entropy encoder (47) coupled to the first prediction filter (45) to generate the first encoded signal, wherein the first prediction filter (45) adapts its operation in response to the first coding parameters that represent prediction filter coefficients; and

the second lossless encoder (126) comprises a second prediction filter coupled to the first difference calculator (122) and a second entropy encoder coupled to the second prediction filter to generate the second encoded signal, wherein the second prediction filter adapts its operation in response to the second coding parameters that represent prediction filter coefficients.

52. A method for decoding an encoded audio signal, wherein the method comprises:

receiving the encoded audio signal that conveys a first encoded signal comprising samples representing audio information at a first sample rate, a second encoded signal comprising samples that represent audio information at a second sample rate that is higher than the first sample rate, wherein the audio information at the second sample rate represents a difference between a first audio signal generated by a first sample rate conversion of an audio input signal from the first sample rate to the second sample rate and a second audio signal generated by a second sample rate conversion of a delayed version of the audio input signal from the first sample rate to the second sample rate, wherein a quality of the first sample rate conversion is higher than a quality of the second sample rate conversion, and representations of first coding parameters of a first lossless encoder that was used to generate the first encoded signal and second coding parameters of a second lossless encoder that was used to generate the second encoded signal;

processing the encoded audio signal to obtain the first encoded signal, the second encoded signal, and the representations of the first coding parameters and the second coding parameters;

generating a first decoded signal by applying a first lossless decoder to the first encoded signal, wherein the first lossless decoder adapts its operation in response to the first coding parameters;

generating a second decoded signal by applying a second lossless decoder to the second encoded signal, wherein the second lossless decoder adapts its operation in response to the second coding parameters;

converting the first decoded signal into a first interim signal comprising digital samples at the second sample rate;

generating a first summation signal comprising samples at the second sample rate that represent a sum of corresponding samples of the first interim signal and the second decoded signal; and

generating an output signal from at least one signal in a set of signals comprising the first decoded signal and the first summation signal, wherein the first decoded signal comprises digital samples that represent an audio signal at the first sample rate and the first summation signal comprises digital samples that represent the audio signal at the second sample rate.

53. An apparatus for decoding an encoded audio signal, wherein the apparatus comprises:

means for receiving the encoded audio signal that conveys a first encoded signal comprising samples representing audio information at a first sample rate, a second encoded signal comprising samples that represent audio information at a second sample rate that is higher than the first sample rate, wherein the audio information at the second sample rate represents a difference between a first audio signal generated by a first sample rate conversion of an audio input signal from the first sample rate to the second sample rate and a second audio signal generated by a second sample rate conversion of a delayed version of the audio input signal from the first sample rate to the second sample rate, wherein a quality of the first sample rate conversion is higher than a quality of the second sample rate conversion, and representations of first coding parameters of a first lossless encoder that was used to generate the first encoded signal and second coding parameters of a second lossless encoder that was used to generate the second encoded signal;

means for processing the encoded audio signal to obtain the first encoded signal, the second encoded signal, and the representations of the first coding parameters and the second coding parameters;

means for generating a first decoded signal by applying a first lossless decoder to the first encoded signal, wherein the first lossless decoder adapts its operation in response to the first coding parameters;

means for generating a second decoded signal by applying a second lossless decoder to the second encoded signal, wherein the second lossless decoder adapts its operation in response to the second coding parameters;

means for converting the first decoded signal into a first interim signal comprising digital samples at the second sample rate;

means for generating a first summation signal comprising samples at the second sample rate that represent a sum of corresponding samples of the first interim signal and the second decoded signal; and

means for generating an output signal from at least one signal in a set of signals comprising the first decoded signal and the first summation signal, wherein the first decoded signal comprises digital samples that represent an audio signal at the first sample rate and the first summation signal comprises digital samples that represent the audio signal at the second sample rate.

54. A non-transitory computer readable storage medium that is readable by a device and that records a program of instructions executable by the device to perform a method for encoding an audio signal, where the method comprises:

obtaining a first signal comprising digital samples that represent the audio signal at a first sample rate;

wherein the first signal is obtained from a delay of an input signal comprising digital samples that represent the audio signal at the first sample rate;

obtaining a second signal comprising digital samples that represent the audio signal at a second sample rate that is higher than the first sample rate;

wherein the second digital signal is obtained from a sample-rate conversion of the input signal;

converting the first signal into a first interim signal comprising digital samples at the second sample rate;

wherein a quality of the sample-rate conversion for obtaining the first interim signal is lower than a quality of the sample-rate conversion for obtaining the second digital signal;

generating a first difference signal comprising samples at the second sample rate that represent a difference between corresponding samples of the second signal and the first interim signal;

applying a first lossless encoder to the first signal to generate a first encoded signal comprising samples representing the first signal at the first sample rate, wherein the first lossless encoder adapts its operation in response to first coding parameters;

applying a second lossless encoder to the first difference signal to generate a second encoded signal comprising samples representing the first difference signal at the second sample rate, wherein the second lossless encoder adapts its operation in response to second coding parameters; and

generating an encoded output signal that conveys the first encoded signal, the second encoded signal, and representations of the first coding parameters and the second coding parameters.

55. A non-transitory computer readable storage medium that is readable by a device and that records a program of instructions executable by the device to perform a method for decoding an encoded audio signal, where the method comprises:

receiving the encoded audio signal that conveys a first encoded signal comprising samples representing audio information at a first sample rate, a second encoded signal comprising samples that represent audio information at a second sample rate that is higher than the first sample rate, wherein the audio information at the second sample rate represents a difference between a first audio signal generated by a first sample rate conversion of an audio input signal from the first sample rate to the second sample rate and a second audio signal generated by a second sample rate conversion of a delayed version of the audio input signal from the first sample rate to the second sample rate, wherein a quality of the first sample rate conversion is higher than a quality of the second sample rate conversion, and representations of first coding parameters of a first lossless encoder that was used to generate the first encoded signal and second coding parameters of a second lossless encoder that was used to generate the second encoded signal;

processing the encoded audio signal to obtain the first encoded signal, the second encoded signal, and the representations of the first coding parameters and the second coding parameters;

generating a first decoded signal by applying a first lossless decoder to the first encoded signal, wherein the first lossless decoder adapts its operation in response to the first coding parameters;

generating a second decoded signal by applying a second lossless decoder to the second encoded signal, wherein the second lossless decoder adapts its operation in response to the second coding parameters;

converting the first decoded signal into a first interim signal comprising digital samples at the second sample rate;

generating a first summation signal comprising samples at the second sample rate that represent a sum of corresponding samples of the first interim signal and the second decoded signal; and

generating an output signal from at least one signal in a set of signals comprising the first decoded signal and the first summation signal, wherein the first decoded signal comprises digital samples that represent an audio signal at the first sample rate and the first summation signal comprises digital samples that represent the audio signal at the second sample rate.