Method and system for reduction of quantization-induced block-discontinuities and general purpose audio codec
A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms.
1. A method comprising:
receiving audio data;
splitting, using at least one processor, a frame of the received audio data into a subframe;
applying a cosine packet transform to the subframe; and
determining optimal transform coefficients for the subframe.
2. The method as recited in claim 1 , wherein the optimal transform coefficients capture one or more characteristics of the received audio data.
3. The method as recited in claim 1 , further comprising performing a boundary analysis on the received audio data, wherein the boundary analysis comprises applying boundary exclusion and interpolation to the received audio data.
4. The method as recited in claim 3 , wherein applying boundary exclusion and interpolation on the received audio data comprises applying residue quantization to the received audio data.
5. The method as recited in claim 3 , further comprising normalizing output of the boundary analysis on the received audio data.
6. The method as recited in claim 1 , further comprising applying rate control to achieve a target bit rate.
7. The method as recited in claim 6 , further comprising modifying one or more parameters used by a signal and residue classifier.
8. The method as recited in claim 6 , further comprising modifying one or more parameters used by a quantization function.
9. The method as recited in claim 1 , further comprising:
applying a quantization function to strong signal components of the received audio data;
applying a stochastic noise analysis to weak signal components of the received audio data; and
formatting output of the quantization function and output of the stochastic noise analysis into a bit-stream format.
10. A non-transitory computer-readable medium including a set of instructions that, when executed by at least one processor, cause a computer system to perform steps comprising:
receiving audio data;
splitting a frame of the received audio data into a subframe;
applying a cosine packet transform to the subframe; and
determining optimal transform coefficients for the subframe.
11. The computer-readable storage medium as recited in claim 10 , further comprising instructions that, when executed, cause at least one processor to perform steps comprising:
performing a boundary analysis on the received audio data;
applying a signal and residue classifier to identify strong signal components and weak signal components of the received audio data;
applying a quantization function to strong signal components of the received audio data;
applying a stochastic noise analysis to weak signal components of the received audio data; and
formatting output of the quantization function and output of the stochastic noise analysis into a bit-stream format.
12. A method comprising:
receiving a bit stream;
generating, using at least one processor, cosine packet coefficients based on the received bit stream;
synthesizing a time-domain signal from the cosine packet coefficients; and
generating audio data based on the time-domain signal.
13. The method as recited in claim 12 , further comprising separating the received bit stream into signal components and noise components.
14. The method as recited in claim 13 , further comprising applying a stochastic noise synthesis to the noise components.
15. The method as recited in claim 13 , wherein generating cosine packet coefficients based on the received bit stream comprises applying an inverse quantization function to the signal components.
16. The method as recited in claim 15 , wherein the inverse quantization function comprises an adaptive sparse vector quantization type function.
17. The method as recited in claim 15 , wherein synthesizing the time-domain signal from the cosine packet coefficients comprises applying an inverse transform function to the cosine packet coefficients.
18. The method as recited in claim 12 , further comprising renormalizing the audio data, wherein the audio data comprises a combined signal that consists of the time-domain signal and a noise signal.
19. The method as recited in claim 12 , further comprising applying a boundary synthesis function to the audio data, wherein the audio data comprises a combined signal that consists of the time-domain signal and a noise signal.
20. The method as recited in claim 19 , further comprising clipping the audio data using one of a soft clipping technique or a hard clipping technique.