# US 12,112,761 B2 · Audio signal encoding method and apparatus

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- Number: US 12,112,761 B2
- Document ID: US12112761B2
- Status: Granted
- Kind: B2
- Application number: 18/451,975
- Filed: 2023-08-18
- Published: 2023-12-07
- Granted: 2024-10-08
- Earliest claimed: 2018-06-29
- Assignees: Huawei Technologies Co., Ltd.
- Applicants: Huawei Technologies Co., Ltd.
- Inventors: Eyal Shlomot; Jonathan Alastair Gibbs; Haiting Li
- Web page: https://patentgrep.com/patent/US12112761B2

## CPC

- G10L 19/008: Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
- G10L 19/038: Vector quantisation, e.g. TwinVQ audio
- G10L 19/06: Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
- G10L 19/032: Quantisation or dequantisation of spectral components
- G10L 19/07: Line spectrum pair [LSP] vocoders
- G10L: SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING

## Abstract

An encoding method includes determining an adaptive broadening factor based on a quantized line spectral frequency (LSF) vector of a first channel of a current frame of an audio signal and an LSF vector of a second channel of the current frame, and writing the quantized LSF vector and the adaptive broadening factor into a bitstream.

## Claims

### Claim 1

An audio signal encoding method, comprising:
obtaining a current frame of an audio signal, wherein the current frame comprises a first channel signal and a second channel signal;
obtaining an inter-channel time difference (ITD) between the first channel signal and the second channel signal;
performing time alignment on the first channel signal and the second channel signal based on the ITD to respectively obtain a time-aligned first channel signal and a time-aligned second channel signal;
performing a time-domain downmixing on the time-aligned first channel signal and the time-aligned second channel signal to respectively obtain a third channel signal and a fourth channel signal;
obtaining a first quantized line spectral frequency (LSF) vector of the third channel signal;
obtaining a second LSF vector of the fourth channel signal;
calculating a second adaptive broadening factor based on the first quantized LSF vector and the second LSF vector;
obtaining a first adaptive broadening factor by quantizing the second adaptive broadening factor; and
writing the first quantized LSF vector and the first adaptive broadening factor into a bitstream,
wherein the first quantized LSF vector, the second LSF vector, and the second adaptive broadening factor satisfy a first equation comprising:
β
=
∑
i
=
1
M
w
i
[
-
LSF
_
S
2
⁢
(
i
)
+
LSF
S
⁢
(
i
)
⁢
LSF
_
S
⁢
(
i
)
-
LSF
S
⁢
(
i
)
⁢
LSF
P
⁢
(
i
)
+
LSF
_
S
⁢
(
i
)
⁢
LSF
P
⁢
(
i
)
]
∑
i
=
1
M
w
i
[
-
LSF
_
S
2
(
i
)
-
LSF
P
2
(
i
)
+
2
⁢
LSF
_
S
(
i
)
⁢
LSF
P
(
i
)
]
,
wherein β represents the second adaptive broadening factor, wherein LSFS represents the second LSF vector, wherein LSFP represents the first quantized LSF vector, wherein LSFS represents a mean vector associated with the second LSF vector, wherein i represents a vector index and is an integer, wherein 1≤i≤M, wherein M is a linear prediction order, and wherein w is a weighting coefficient.

### Claim 2 (depends on 1)

The audio signal encoding method of claim 1, further comprising obtaining a second quantized LSF vector of the fourth channel signal based on the first adaptive broadening factor and the first quantized LSF vector.

### Claim 3 (depends on 2)

The audio signal encoding method of claim 2, further comprising:
performing pull-to-average processing on the first quantized LSF vector based on the first adaptive broadening factor to obtain a broadened LSF vector of the third channel signal; and
further obtaining the second quantized LSF vector based on the broadened LSF vector.

### Claim 4 (depends on 3)

The audio signal encoding method of claim 3, further comprising performing the pull-to-average processing according to a second equation comprising:
LSFSB(i)=βq·LSFP(i)+(1−βq)·LSFS(i)
wherein LSFSB represents the broadened LSF vector, and wherein βq represents the first adaptive broadening factor.

### Claim 5 (depends on 1)

The audio signal encoding method of claim 1, wherein the second LSF vector meets a reusing condition when a distance between an LSF vector of the third channel signal and the second LSF vector of the fourth channel signal is less than or equal to a threshold.

### Claim 6

An audio signal encoding apparatus, comprising:
a processor; and
a memory coupled to the processor and configured to store programming instructions for execution by the processor to cause the audio signal encoding apparatus to:
obtain a current frame of an audio signal, wherein the current frame comprises a first channel signal and a second channel signal;
obtain an inter-channel time difference (ITD) between the first channel signal and the second channel signal;
perform time alignment on the first channel signal and the second channel signal based on the ITD to respectively obtain a time-aligned first channel signal and a time-aligned second channel signal;
perform a time-domain downmixing on the time-aligned first channel signal and the time-aligned second channel signal to respectively obtain a third channel signal and a fourth channel signal;
obtain a first quantized line spectral frequency (LSF) vector of the third channel signal;
obtain a second LSF vector of the fourth channel signal;
calculate a second adaptive broadening factor based on the first quantized LSF vector and the second LSF vector;
obtain a first adaptive broadening factor by quantizing the second adaptive broadening factor; and
write the first quantized LSF vector and the first adaptive broadening factor into a bitstream,
wherein the first quantized LSF vector, the second LSF vector, and the second adaptive broadening factor satisfy a first equation comprising:
β
=
∑
i
=
1
M
w
i
[
-
LSF
_
S
2
⁢
(
i
)
+
LSF
S
⁢
(
i
)
⁢
LSF
_
S
⁢
(
i
)
-
LSF
S
⁢
(
i
)
⁢
LSF
P
⁢
(
i
)
+
LSF
_
S
⁢
(
i
)
⁢
LSF
P
⁢
(
i
)
]
∑
i
=
1
M
w
i
[
-
LSF
_
S
2
(
i
)
-
LSF
P
2
(
i
)
+
2
⁢
LSF
_
S
(
i
)
⁢
LSF
P
(
i
)
]
,
wherein β represents the second adaptive broadening factor, wherein LSFS represents the second LSF vector, wherein LSFP represents the first quantized LSF vector, wherein LSFS represents a mean vector associated with the second LSF vector, wherein i is a vector index and is an integer, wherein 1≤i≤M, wherein M is a linear prediction order, and wherein w is a weighting coefficient.

### Claim 7 (depends on 6)

The audio signal encoding apparatus of claim 6, wherein the programming instructions for execution by the processor further cause the audio signal encoding apparatus to obtain a second quantized LSF vector of the fourth channel signal based on the first adaptive broadening factor and the first quantized LSF vector.

### Claim 8 (depends on 7)

The audio signal encoding apparatus of claim 7, wherein the programming instructions for execution by the processor further cause the audio signal encoding apparatus to:
perform pull-to-average processing on the first quantized LSF vector based on the first adaptive broadening factor to obtain a broadened LSF vector of the third channel signal; and
further obtain the second quantized LSF vector based on the broadened LSF vector.

### Claim 9 (depends on 8)

The audio signal encoding apparatus of claim 8, wherein the programming instructions for execution by the processor further cause the audio signal encoding apparatus to:
perform the pull-to-average processing according to a second equation comprising:
LSFSB(1)=βq·LSFP(i)+(1−βq)·LSFS(i),
wherein LSFSB represents the broadened LSF vector, and wherein βq represents the first adaptive broadening factor.

### Claim 10 (depends on 6)

The audio signal encoding apparatus of claim 6, wherein the second LSF vector meets a reusing condition when a distance between an LSF vector of the third channel signal and the second LSF vector of the fourth channel signal is less than or equal to a threshold.

### Claim 11

A computer program product comprising computer-executable instructions that are stored on a non-transitory computer-readable medium and that, when executed by a processor, cause an audio signal encoding apparatus to:
obtain a current frame of an audio signal, wherein the current frame comprises a first channel signal and a second channel signal;
obtain an inter-channel time difference (ITD) between the first channel signal and the second channel signal;
perform time alignment on the first channel signal and the second channel signal based on the ITD to respectively obtain a time-aligned first channel signal and a time-aligned second channel signal;
perform a time-domain downmixing on the time-aligned first channel signal and the time-aligned second channel signal to respectively obtain a third channel signal and a fourth channel signal;
obtain a first quantized line spectral frequency (LSF) vector of the third channel signal;
obtain a second LSF vector of the fourth channel signal;
calculate a second adaptive broadening factor based on the first quantized LSF vector and the second LSF vector;
obtain a first adaptive broadening factor by quantizing the second adaptive broadening factor; and
write the first quantized LSF vector and the first adaptive broadening factor into a bitstream;
wherein the first quantized LSF vector, the second LSF vector, and the second adaptive broadening factor satisfy a first equation comprising:
β
=
∑
i
=
1
M
w
i
[
-
LSF
_
S
2
⁢
(
i
)
+
LSF
S
⁢
(
i
)
⁢
LSF
_
S
⁢
(
i
)
-
LSF
S
⁢
(
i
)
⁢
LSF
P
⁢
(
i
)
+
LSF
_
S
⁢
(
i
)
⁢
LSF
P
⁢
(
i
)
]
∑
i
=
1
M
w
i
[
-
LSF
_
S
2
(
i
)
-
LSF
P
2
(
i
)
+
2
⁢
LSF
_
S
(
i
)
⁢
LSF
P
(
i
)
]
,
wherein β represents the second adaptive broadening factor, wherein LSFS represents the second LSF vector, wherein LSFP represents the first quantized LSF vector, wherein LSFS represents a mean vector associated with the second LSF vector, wherein i is a vector index and is an integer, wherein 1≤i≤M, wherein M is a linear prediction order, and wherein w is a weighting coefficient.

### Claim 12 (depends on 11)

The computer program product of claim 11, wherein the computer-executable instructions, when executed by the processor, further cause the audio signal encoding apparatus to obtain a second quantized LSF vector of the fourth channel signal based on the first adaptive broadening factor and the first quantized LSF vector.

### Claim 13 (depends on 12)

The computer program product of claim 12, wherein the computer-executable instructions, when executed by the processor, further cause the audio signal encoding apparatus to:
perform pull-to-average processing on the first quantized LSF vector based on the first adaptive broadening factor to obtain a broadened LSF vector of the third channel signal; and
further obtain the second quantized LSF vector based on the broadened LSF vector.

### Claim 14 (depends on 13)

The computer program product of claim 13, wherein the computer-executable instructions, when executed by the processor, further cause the audio signal encoding apparatus to perform the pull-to-average processing according to a second equation comprising:
LSFSB(i)=βq·LSFP(i)+(1−βq)·LSFS(i)
wherein LSFSB represents the broadened LSF vector, LSFP and wherein βq represents the first adaptive broadening factor LSFS.

### Claim 15 (depends on 11)

The computer program product of claim 11, wherein the second LSF vector meets a reusing condition when a distance between an LSF vector of the third channel signal and the second LSF vector of the fourth channel signal is less than a threshold.

### Claim 16 (depends on 15)

The computer program product of claim 15, wherein the computer-executable instructions, when executed by the processor, further cause the audio signal encoding apparatus to further write an indication of not to perform quantization encoding on the second LSF vector when the distance is less than the threshold.

### Claim 17 (depends on 11)

The computer program product of claim 11, wherein the second LSF vector meets a reusing condition when a distance between an LSF vector of the third channel signal and the second LSF vector of the fourth channel signal is equal to a threshold.

### Claim 18 (depends on 17)

The computer program product of claim 17, wherein the computer-executable instructions, when executed by the processor, further cause the audio signal encoding apparatus to further write an indication of not to perform quantization encoding on the second LSF vector when the distance is equal to the threshold.

### Claim 19 (depends on 5)

The audio signal encoding method of claim 5, further comprising further writing an indication of not to perform quantization encoding on the second LSF vector when the distance is less than or equal to the threshold.

### Claim 20 (depends on 10)

The audio signal encoding apparatus of claim 10, wherein the programming instructions for execution by the processor further cause the audio signal encoding apparatus to further write an indication of not to perform quantization encoding on the second LSF vector when the distance is less than or equal to the threshold.