Companding apparatus and method to reduce quantization noise using advanced spectral extension

Patent metadata

Field	Value
Publication number	US-9947335-B2
Application number	US-201414762690-A
Country	US
Kind code	B2
Filing date	Apr 1, 2014
Priority date	Apr 5, 2013
Publication date	Apr 17, 2018
Grant date	Apr 17, 2018

Abstract

Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

First claim

The invention claimed is:

1. A method of expanding an audio signal comprising:
receiving an audio signal; and
expanding the audio signal to an expanded dynamic range through an expansion process comprising: dividing the received audio signal into a plurality of time-segments using a defined window shape, calculating a wideband gain for each time-segment in the frequency domain using a p-norm of spectral magnitudes of each time-segment of a frequency domain representation of the audio signal, wherein the p-norm value is selected to emphasize weak spectral content of the audio signal versus strong spectral content of the audio signal, and applying individual gain values to each time-segment to obtain the expanded dynamic range audio signal, wherein the application of the individual gain values amplifies segments of relatively high intensity and attenuates segments of relatively low intensity.

2. The method of claim 1 , wherein a value of p in the p-norm is less than 2.

3. The method of claim 2 , wherein a first filterbank is used to analyze the audio signal to obtain a frequency domain representation, and the defined window shape corresponds to a prototype filter for the first filterbank, and wherein the prototype filter is shaped to smooth the individual gain values to eliminate discontinuities in an envelope of the audio signal.

4. The method of claim 3 , wherein the first filterbank is one of a quadrature modulated filter (QMF) bank or a short-time Fourier transform.

5. The method of claim 3 , wherein the wideband gain for each time segment is calculated using the subband samples in a subset of subbands in the respective time segment.

6. A method of compressing an audio signal comprising:
receiving an initial audio signal; and
compressing the initial audio signal to substantially reduce an original dynamic range of the initial audio signal through a compression process comprising dividing the initial audio signal into a plurality of segments using a defined window shape, calculating a wideband gain in the frequency domain using a p-norm of spectral magnitudes of each segment of the plurality of segments of frequency domain samples of the initial audio signal, wherein the p-norm value is selected to emphasize weak spectral content of the audio signal versus strong spectral content of the audio signal, and applying individual gain value to each segment of the plurality of segments to amplify segments of relatively low intensity and attenuate segments of relatively high intensity.

7. The method of claim 6 , wherein the segments are overlapping and wherein a first filterbank is used to analyze the audio signal to obtain a frequency domain representation and the defined window shape corresponds to a prototype filter for the first filterbank, and wherein the prototype filter is shaped to smooth the individual gain values to eliminate discontinuities in an envelope of the audio signal.

8. The method of claim 7 , wherein the first filterbank is one of a quadrature modulated filter (QMF) bank or a short-time Fourier transform, and wherein the value of p in the p-norm is less than 2.

9. The method of claim 7 , wherein each individual gain value is calculated using the subband samples in a subset of subbands in a respective time segment.

10. The method of claim 9 , wherein the subset of subbands corresponds to the entire frequency range spanned by the first filterbank, and wherein the gain is applied in the domain of the first filterbank.

11. An apparatus for compressing an audio signal comprising:
a first interface receiving an initial audio signal; and
a compressor compressing the initial audio signal to substantially reduce an original dynamic range of the initial audio signal by dividing the initial audio signal into a plurality of segments using a defined window shape, calculating a wideband gain in the frequency domain using a p-norm of spectral magnitudes of each segment of the plurality of segments of frequency domain samples of the initial audio signal, wherein the p-norm value is selected to emphasize weak spectral content of the audio signal versus strong spectral content of the audio signal, and applying individual gain values to each segment of the plurality of segments to amplify segments of relatively low intensity and attenuate segments of relatively high intensity.

12. The apparatus of claim 11 , further comprising a first filterbank analyzing the audio signal to obtain a frequency domain representation and wherein the defined window shape corresponds to a prototype filter for the first filterbank, and further wherein the first filterbank is one of a quadrature modulated filter (QMF) bank or a short-time Fourier transform, and wherein the prototype filter is shaped to smooth the individual gain values to eliminate discontinuities in an envelope of the audio signal.

13. The apparatus of claim 12 , wherein the individual gain values are calculated using the subband samples in a subset of subbands in each respective time segment, and wherein the value of p in the p-norm is less than 2.

14. The apparatus of claim 13 , wherein the subset of subbands corresponds to the entire frequency range spanned by the first filterbank, and wherein the gain is applied in the domain of the first filterbank.

15. The apparatus of claim 12 , further comprising a second interface transmitting a compressed version of the initial audio signal to an expander that receives the compressed version of audio signal, and expands the compressed version of the audio signal to substantially restore it to an original dynamic range of the initial audio signal by dividing the initial audio signal into a plurality of segments using the defined window shape, calculating a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal; and applying a respective gain value to each segment of the plurality of segments to amplify segments of relatively high intensity and attenuate segments of relatively low intensity.

16. An apparatus for expanding an audio signal comprising:
a first interface receiving a compressed audio signal; and
an expander expanding the compressed audio signal to substantially restore its original uncompressed dynamic range by dividing the initial audio signal into a plurality of segments using a defined window shape, calculating a wideband gain in the frequency domain using a p-norm of spectral magnitudes of each segment of the plurality of segments of frequency domain samples of the initial audio signal, wherein the p-norm value is selected to emphasize weak spectral content of the audio signal versus strong spectral content of the audio signal, and applying individual gain values to each segment of the plurality of segments to amplify segments of relatively high intensity and attenuate segments of relatively low intensity.

17. The apparatus of claim 16 , further comprising a first filterbank analyzing the audio signal to obtain a frequency domain representation and wherein the defined window shape corresponds to a prototype filter for the first filterbank, and further wherein the first filterbank is one of a quadrature modulated filter (QMF) bank or a short-time Fourier transform, and wherein the prototype filter is shaped to smooth the individual gain values to eliminate discontinuities in an envelope of the audio signal.

18. The apparatus of claim 17 , wherein the wideband gain comprises an individual gain value for each time segment, and wherein each individual gain value is calculated using the subband samples in

Assignees

• Dolby Int Ab
• Dolby Laboratories Licensing Corp

Inventors

• Hedelin Per
• Biswas Arijit
• Schug Michael
• Melkote Vinay

Classifications

• G10L21/034 Primary

  Automatic adjustment · CPC title

• G10L19/008 Primary

  Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing · CPC title

• G10L19/032

  Quantisation or dequantisation of spectral components · CPC title

• G10L25/45

  characterised by the type of analysis window · CPC title

• H04B1/66 Primary

  for reducing bandwidth of signals; for improving efficiency of transmission (H04B1/68 takes precedence) · CPC title