Adaptive time/frequency-based audio encoding and decoding apparatuses and methods

• US 8,548,801 B2
• Filed: 09/26/2006
• Issued: 10/01/2013
• Est. Priority Date: 11/08/2005
• Status: Expired due to Fees

• Current Assignee: Samsung Electronics Co. Ltd.
• Original Assignee: Samsung Electronics Co. Ltd.
• Inventors: Kim, Junghoe, Oh, Eunmi, Son, Changyong, Choo, Kihyun
• Primary Examiner(s): Godbold, Douglas
• Application Number:
  US11/535,164
• Publication Number:
  US 20070106502A1
• Time in Patent Office: 2,562 Days
• Field of Search: 704/200, 704/201, 704/500, 704/501
• US Class Current: 704/200
• CPC Class Codes: G10L 19/02 using spectral analysis, e.g. transform vocoders or subband vocoders
• CPC Class Codes: G10L 19/12 the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
• CPC Class Codes: G10L 19/20 using sound class specific coding, hybrid encoders or object based coding

Abstract

Adaptive time/frequency-based audio encoding and decoding apparatuses and methods. The encoding apparatus includes a transformation & mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal, an encoding unit to encode each frequency-domain signal in the respective encoding mode, and a bitstream output unit to output encoded data, division information, and encoding mode information for each respective frequency-domain signal. In the apparatuses and methods, acoustic characteristics and a voicing model are simultaneously applied to a frame, which is an audio compression processing unit. As a result, a compression method effective for both music and voice can be produced, and the compression method can be used for mobile terminals that require audio compression at a low bit rate.

Claims

1. An adaptive time/frequency-based audio encoding apparatus, comprising:

a transformation &

mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal;

a time-based encoding unit, implemented by at least one processing device, to perform time-based encoding in a linear prediction coding domain by using at least a long-term prediction on a first frequency-domain signal determined to be encoded in a time-based encoding mode;

a frequency-based encoding unit to perform frequency-based encoding in a frequency domain other than the linear prediction coding domain, on a second frequency-domain signal determined to be encoded in a frequency-based encoding mode; and

a bitstream output unit to output encoded data, division information, and encoding mode information including the time-based encoding mode or the frequency-based encoding mode corresponding to each respective encoded frequency-domain signal.

2. The apparatus of claim 1, wherein the transformation &

mode determination unit comprises;

a frequency-domain transform unit to transform the input audio signal into a full frequency-domain signal; and

an encoding mode determination unit to divide the full frequency-domain signal into the frequency-domain signals according to a preset standard and to determine the time-based encoding mode or the frequency-based encoding mode for each respective frequency-domain signal.

3. The apparatus of claim 2, wherein the full frequency-domain signal is divided into the frequency-domain signals suitable for the time-based encoding mode or the frequency-based encoding mode based on at least one of a spectral tilt, a size of signal energy of each frequency domain, a change in signal energy between sub-frames, and a voicing level determination, and the respective encoding mode for each frequency-domain signal is determined accordingly.

4. The apparatus of claim 2, wherein the frequency-domain transform unit performs the frequency-domain transform using a frequency varying modulated lapped transform (MLT).

5. The apparatus of claim 1, wherein the time-based encoding unit selects the encoding mode for the first input frequency-domain signal based on at least one of a linear coding gain, a spectral change between linear prediction filters of adjacent frames, and a predicted pitch delay, continues to perform the time-based encoding on the first frequency-domain signal when the time-based encoding unit determines that the time-based encoding mode is suitable for the first frequency-domain signal, and stops performing the time-based encoding on the first frequency-domain signal and transmits a mode conversion control signal to the transformation &

mode determination unit when the time-based encoding unit determines that the frequency-based encoding mode is suitable for the first frequency-domain signal, and the transformation &

mode determination unit outputs the first frequency-domain signal again, which was provided to the time-based encoding unit, to the frequency-based encoding unit in response to the mode conversion control signal.

6. The apparatus of claim 1, wherein the time-based encoding unit quantizes a residual signal obtained from linear prediction and dynamically allocates bits to the quantized residual signal according to importance.

7. The apparatus of claim 6, wherein the importance is determined based on a voicing model.

8. The apparatus of claim 1, wherein the time-based encoding unit transforms a residual signal obtained from a linear prediction into a frequency-domain signal, quantizes the frequency-domain signal, and dynamically allocates bits to the quantized signal according to importance.

9. The apparatus of claim 8, wherein the importance is determined based on a voicing model.

10. The apparatus of claim 8, wherein the residual signal is obtained using a code excited linear prediction (CELP) algorithm.

11. The apparatus of claim 1, wherein the frequency-based encoding unit determines a quantization step size of an input frequency-domain signal according to a psychoacoustic model and quantizes the frequency-domain signal.

12. The apparatus of claim 1, wherein the frequency-based encoding unit extracts important frequency components from an input frequency-domain signal according to a psychoacoustic model, encodes the extracted important frequency components, and encodes remaining signals using noise modeling.

13. An adaptive time/frequency-based audio decoding apparatus, comprising:

a bitstream sorting unit to extract encoded data of at least one frequency band, and encoding mode information including a time-based encoding mode or a frequency-based encoding mode, of the at least one frequency band from an input bitstream;

a time-based decoding unit, implemented by at least one processing device, to perform a time-based decoding in a linear prediction coding domain by using at least a long-term prediction, on first encoded data based on the time-based encoding mode;

a frequency-based decoding unit to perform a frequency-based decoding in a frequency domain other than the linear prediction coding domain, on second encoded data based on the frequency-based encoding mode; and

a collection &

inverse transform unit to collect decoded data and to perform an inverse frequency-domain transform on the collected data.

14. The apparatus of claim 13, wherein the time-based decoding unit decodes the first encoded data using a CELP algorithm.

15. The apparatus of claim 13, wherein the collection &

inverse transform unit performs envelope smoothing on the decoded data in the frequency domain and then performs the inverse frequency-domain transform on the decoded data such that the decoded data maintains continuity in the frequency domain.

16. The apparatus of claim 13, wherein a final audio signal is generated using a frequency-varying MLT after the decoded data is collected in the frequency domain.

17. An adaptive time/frequency-based audio encoding method, comprising:

dividing an input audio signal into a plurality of frequency-domain signals and selecting a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal;

performing a time-based encoding in a linear prediction coding domain by using at least a long-term prediction on a first frequency-domain signal determined to be encoded in the time-based encoding mode;

performing a frequency-based encoding in a frequency domain other than the linear prediction coding domain, on a second frequency-domain signal determined to be encoded in the frequency-based encoding mode; and

outputting encoded data, division information, and encoding mode information including the time-based encoding mode or the frequency-based encoding mode of each respective frequency-domain signal.

18. The method of claim 17, wherein the division of the input audio signal comprises:

transforming the input audio signal into a full frequency-domain signal; and

dividing the full frequency-domain signal into the frequency-domain signals according to a preset standard and selecting the time-based encoding mode or the frequency-based encoding mode for each respective frequency-domain signal.

19. The method of claim 18, wherein the division of the full frequency-domain signal comprises:

dividing the full frequency-domain signal into the frequency-domain signals suitable for the time-based encoding mode or the frequency-based encoding mode based on at least one of a spectral tilt, a size of signal energy of each frequency domain, a change in signal energy between sub-frames and a voicing level determination; and

selecting the encoding mode for each respective frequency-domain signal.

20. An adaptive time/frequency-based audio decoding method, comprising:

extracting encoded data of at least one frequency band from an input bitstream, and encoding mode information including a time-based encoding mode or a frequency-based encoding mode, of the at least one frequency band;

performing a time-based decoding in a linear prediction coding domain by using at least a long-term prediction, on first encoded data based on the time-based encoding mode;

performing a frequency-based decoding in a frequency domain other than the linear prediction coding domain, on second encoded data based on the frequency-based encoding mode; and

collecting decoded data and performing an inverse frequency-domain transform on the collected data.

21. A non-transitory computer-readable recording medium having a software program to execute an adaptive time/frequency-based audio encoding method, the method comprising:

dividing an input audio signal into a plurality of frequency-domain signals and selecting a time-based encoding mode or a frequency-based encoding mode of each respective frequency-domain signal;

performing a time-based encoding in a linear prediction coding domain by using at least a long-term prediction on a first frequency-domain signal determined to be encoded in the time-based encoding mode;

performing a frequency-based encoding in a frequency domain other than the linear prediction coding domain, on a second frequency-domain signal determined to be encoded in the frequency-based encoding mode; and

outputting encoded data, division information, and encoding mode information including the time-based encoding mode or the frequency-based encoding mode, of each respective frequency-domain signal.

22. A method of decoding a bitstream including encoded data and encoding mode information for at least one frequency band, comprising:

extracting the encoded data of the at least one frequency band from the bitstream, and encoding mode information including a time-based encoding mode or a frequency-based encoding mode, of the at least one frequency band;

decoding the encoded data of the at least one frequency band in a linear prediction coding domain, by using at least a long-term prediction, based on the time-based encoding mode;

decoding the encoded data of the at least one frequency band in a frequency domain other than the linear prediction coding domain, based on the frequency-based encoding mode; and

performing an inverse frequency-domain transform on the decoded data of the at least one frequency band.

23. An audio decoding method, comprising:

extracting encoded data from an input bitstream;

decoding first encoded data, by using a code excited linear prediction (CELP) with at least a long-term prediction, in a first domain based on a mode information of the encoded data;

decoding second encoded data by using an advanced audio coding (AAC), in a second domain based on the mode information of the encoded data;

inverse-transforming data decoded in the second domain; and

generating a signal including the inverse-transformed data and the result of decoding in the first domain.

24. The method of claim 23, wherein the first and second domains comprise a frequency domain.

25. The method of claim 23, wherein the first and second domains are different from each other.

26. An audio decoding method, comprising:

extracting encoded data from an input bitstream;

decoding first encoded data, by using at least a long-term prediction, in a linear prediction coding domain based on a mode information of the encoded data;

decoding second encoded data in a frequency domain other than the linear prediction coding domain based on the mode information of the encoded data;

inverse-transforming data decoded in the frequency domain; and

generating a signal including the inverse-transformed data and the result of decoding in the linear prediction coding domain.

27. The method of claim 26, wherein the second encoded data is decoded by using an advanced audio coding (AAC).