Speech coding system and method using silence enhancement

• US 10,204,628 B2
• Filed: 12/30/2013
• Issued: 02/12/2019
• Est. Priority Date: 09/22/1999
• Status: Expired due to Fees

• Current Assignee: DigiMedia Tech, LLC (IP Investments Group LLC)
• Original Assignee: Nytell Software LLC (Intellectual Ventures LLC)
• Inventors: Gao, Yang
• Primary Examiner(s): Opsasnick, Michael N
• Application Number:
  US14/143,831
• Publication Number:
  US 20140119572A1
• Time in Patent Office: 1,870 Days
• Field of Search: 704225, 704226
• CPC Class Codes: G10L 19/00 Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis in musical instruments G10H
• CPC Class Codes: G10L 19/167 Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
• CPC Class Codes: G10L 19/20 using sound class specific coding, hybrid encoders or object based coding
• CPC Class Codes: G10L 19/22 Mode decision, i.e. based on audio signal content versus external parameters
• 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
• CPC Class Codes: G10L 2019/0001 Codebooks
• CPC Class Codes: H03G 3/00 Gain control in amplifiers or frequency changers

Abstract

Various techniques for speech coding and decoding are disclosed. For example, speech data generated from a speech signal may be decoded by receiving the speech data in a format that has at least one main pulse in a subframe of the speech data, and generating a first predicted pulse that has a lower gain than the main pulse. A second predicted pulse may also be generated as a mirror image of the first predicted pulse on a reverse time scale, on the other side of the main pulse in the subframe of the speech data. The the speech signal may be reconstructed using the first predicted pulse and the second predicted pulse.

Claims

1. A method, comprising:

receiving, at a computing device, an input audio signal including a zero level, wherein the input audio signal comprises a plurality of audio frames;

determining, by the computing device, that at least one frame of the plurality of audio frames has a volume level within a selected range of the zero level;

modifying, by the computing device, the volume level of the at least one frame to correspond to the zero level of the input audio signal to generate an enhanced audio signal;

filtering the enhanced audio signal by a high-pass filter to generate a filtered audio signal;

attenuating noise included in the filtered audio signal to generate a pre-processed speech signal;

encoding, by the computing device, the pre-processed speech signal to generate an encoded plurality of audio frames, wherein the encoded plurality of audio frames are decodable by a speech decoding system to generate a reproduced version of the input audio signal; and

transmitting, to the speech decoding system via a communication link, a signal that includes the encoded plurality of audio frames.

2. The method of claim 1, wherein the determining includes comparing the volume level of the at least one frame to one or more quantization levels.

3. The method of claim 1, wherein the selected range includes one or more quantization levels, wherein the one or more quantization levels correspond to quantization levels of pulse code modulation (PCM).

4. The method of claim 1, further comprising:

tracking, by the computing device, the zero level adaptively by using the modified volume level of the at least one frame as a feedback.

5. The method of claim 4, wherein the tracking the zero level adaptively includes using a minimum resolution.

6. The method of claim 1, wherein the determining includes determining whether the volume level is within two quantization levels of the zero level.

7. The method of claim 1, wherein the modifying the volume level of the at least one frame includes ramping the volume level to the zero level.

8. The method of claim 1, further comprising:

composing, by the computing device, a modified audio signal using the at least one frame and the input audio signal, wherein the modified audio signal corresponds to a preprocessed audio signal.

9. The method of claim 1, wherein the receiving the input audio signal further comprises:

the computing device reading, from the input audio signal, one or more audio samples; and

the computing device buffering the one or more audio samples.

10. A method, comprising:

receiving, at a computing device, a speech signal including a zero level and one or more samples, wherein the one or more samples have corresponding magnitude levels;

determining, by the computing device, that at least one sample of the one or more samples corresponds to silence noise;

in response to the at least one sample corresponding to silence noise, the computing device adjusting the corresponding magnitude levels of the at least one sample to the zero level to generate an enhanced audio signal;

filtering the enhanced audio signal by a high-pass filter to generate a filtered audio signal;

attenuating noise included in the filtered audio signal to generate a pre-processed speech signal;

encoding, by the computing device, the pre-processed speech signal to generate encoded one or more samples, wherein the encoded one or more samples are decodable by a speech decoding system to generate a reproduced version of the speech signal; and

transmitting, to the speech decoding system via a communication link, a signal that includes the encoded one or more samples.

11. The method of claim 10, wherein the silence noise corresponds to a portion of the speech signal that does not contain voiced content.

12. The method of claim 10, wherein the determining includes comparing the magnitude levels of the one or more samples to a quantization level.

13. The method of claim 10, wherein the encoding includes encoding the pre-processed speech signal according to an A-law speech coding algorithm.

14. The method of claim 10, wherein the adjusting the corresponding magnitude levels includes ramping the at least one sample by determining a quantization range based on a speech coding algorithm.

15. The method of claim 14, wherein the speech coding algorithm is A-law, and wherein the quantization range includes +8 and −

8.

16. The method of claim 10, wherein the adjusting the corresponding magnitude levels includes adjusting the at least one sample by ramping a magnitude level of the at least one sample.