CN111429925B - Method and system for reducing audio coding rate - Google Patents
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- CN111429925B CN111429925B CN202010279435.XA CN202010279435A CN111429925B CN 111429925 B CN111429925 B CN 111429925B CN 202010279435 A CN202010279435 A CN 202010279435A CN 111429925 B CN111429925 B CN 111429925B
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- G—PHYSICS
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- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—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
- G10L19/02—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 using spectral analysis, e.g. transform vocoders or subband vocoders
- G10L19/032—Quantisation or dequantisation of spectral components
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- G—PHYSICS
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- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
- G10L19/00—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
- G10L19/04—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 using predictive techniques
- G10L19/16—Vocoder architecture
- G10L19/18—Vocoders using multiple modes
- G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
Abstract
The invention discloses a method and a system for reducing audio coding rate, belonging to the technical field of audio coding and decoding. The method for reducing the audio coding rate comprises the following steps: identifying a sampling frequency; a bandwidth detection step; a code rate conversion step, which is used for converting the coding rate into the first byte number available in each frame during coding; calculating a second byte number, namely subtracting a third byte number of necessary expense of each frame from the first byte number to obtain a second byte number available for quantizing and arithmetically encoding the spectral coefficient of each frame; and calculating a fourth byte number, namely calculating the second byte number and the third byte number to obtain a fourth byte number which is available according to each frame of new bandwidth coding and updating the coding control mark, and updating the coding control mark according to the fourth byte number and the dependent code rate. The application of the invention avoids the waste of the computing power in the audio coding process, so that the coder is in the optimal working state.
Description
Technical Field
The present invention relates to the field of audio coding technology, and in particular, to a method and system for reducing audio coding rate.
Background
In the prior art, the mainstream bluetooth audio encoder includes: the SBC audio encoder is used most widely according to the mandatory requirements of an A2DP protocol, and is supported by all Bluetooth audio equipment, but the tone quality is general; the AAC-LC audio encoder has good tone quality and wide application range, is supported by a plurality of mainstream mobile phones, but has larger memory occupation and high operation complexity compared with the SBC audio encoder, and a plurality of Bluetooth devices are based on an embedded platform, so that the battery capacity is limited, the operation capability of a processor is poor and the memory is limited; the aptX series audio coder has good sound quality but high code rate, wherein the aptX needs the code rate of 384kbps, and the code rate of the aptX-HD is 576kbps, is a unique technology of high pass, and is relatively closed; the LDAC audio encoder has good sound quality, but has high code rates, i.e. 330kbps,660kbps and 990kbps, respectively, because the wireless environment of the bluetooth device is very complex, it is difficult to stably support such high code rate, and the LDAC audio encoder is a unique sony technology and is also very closed.
For the above reasons, the Bluetooth international association Bluetooth Sig has introduced the LC3 audio encoder in conjunction with numerous manufacturers, which has the advantages of low delay, high sound quality, high coding gain, no special fee in the Bluetooth field, and the like, and is paid attention by the manufacturers. The LC3 audio encoder is originally designed to meet the audio application in the field of low-power-consumption Bluetooth, so that the requirement on power consumption is very strict, in the field of low-power-consumption Bluetooth, the power consumption of mobile equipment is expected to be low, the code rate of audio is in direct proportion to occupied air bandwidth, the power consumption of radio frequency is directly influenced, the low code rate not only can reduce the radio frequency power consumption and air interference of a transmitting and receiving end, but also can reduce the operation amount of a coder and a decoder.
Generally, an input signal of an audio encoder has an effective bandwidth close to a Nyquist frequency (Nyquist frequency), but in practical applications, the effective bandwidth of audio of the audio encoder is far smaller than the Nyquist frequency, and at this time, there are situations that some coding modules of the audio codec cannot work normally, and if the audio codec is coded according to the Nyquist frequency, spectral energy leakage is caused; if the original code rate is still used for coding, certain code rate waste is caused, and the coder does not work in an optimal mode; furthermore, the reduction of the effective bandwidth means a reduction in the amount of information, which would cause a problem of a waste of computing power if the nyquist frequency is still followed.
Disclosure of Invention
The technical problem to be solved by the present invention is to provide a method and a system for reducing the audio encoding rate, so that the audio encoder can reduce the unnecessary waste of the computing power in the optimal working mode.
In order to achieve the above object, the first technical solution adopted by the present invention is: a method for reducing audio coding rate is provided, which comprises a step of identifying a sampling frequency, wherein when the sampling frequency of an input signal is the lowest sampling frequency, coding is carried out according to a standard flow, and when the sampling frequency of the input signal is not the lowest sampling frequency, the following steps are carried out; a bandwidth detection step, namely, when the effective bandwidth is equal to the Nyquist frequency, coding is carried out according to a standard flow, and when the effective bandwidth is less than the Nyquist frequency, the following steps are carried out; a code rate conversion step, which is used for converting the coding rate into the first byte number available in each frame during coding; calculating a second byte number, namely subtracting a third byte number of necessary expense of each frame from the first byte number to obtain a second byte number available for quantizing and arithmetically encoding the spectral coefficient of each frame; calculating a fourth byte number, namely calculating the available fourth byte number according to the new frame coding of the bandwidth through the second byte number and the third byte number; and updating the coding control mark according to the fourth byte number and the dependent code rate.
In order to achieve the above object, the second technical solution adopted by the present invention is: a system for reducing an audio coding rate is provided, which includes a bandwidth detection module for detecting an effective bandwidth of an input signal; the code rate and coding parameter updating module is used for converting the coding rate into a first byte number available for each frame when a sound channel is coded, subtracting a third byte number of necessary overhead of each frame from the first byte number to obtain a second byte number available for quantizing and arithmetically coding each frame spectral coefficient, calculating through the second byte number and the third byte number to obtain a fourth byte number available for coding each frame according to the new bandwidth, and then updating the coding control mark according to the fourth byte number and the dependent code rate; and a quantization module that replaces the number of encoded spectral lines per frame with a product of the number of maximum spectral lines per frame and the ratio of the effective bandwidth to the nyquist frequency.
The invention has the beneficial effects that: when the method is applied, the audio encoder reduces unnecessary waste of computing power in an optimal working mode, thereby reducing the power consumption of the audio encoder.
Drawings
FIG. 1 is a flow chart illustrating a method for reducing audio coding rate according to the present invention;
FIG. 2 is a schematic diagram of an encoding flow structure of an LC3 audio encoder;
FIG. 3 is a diagram illustrating an example of subjective disparity levels of a method of reducing audio coding rate according to the present invention;
FIG. 4 is a block diagram of a system for reducing the audio coding rate according to the present invention;
FIG. 5 is a spectrogram of a single channel speech with a sampling rate of 48000Hz and an effective bandwidth of 24 kHz;
FIG. 6 is a spectrogram of a single channel speech with a sampling rate of 48000Hz and an effective bandwidth of 12 kHz;
FIG. 7 is an example of a graph of subjective difference levels after encoding of speech spectra having an effective bandwidth of 24kHz and 12 kHz.
Detailed Description
The following detailed description of the preferred embodiments of the present invention, taken in conjunction with the accompanying drawings, will make the advantages and features of the invention easier to understand by those skilled in the art, and thus will clearly and clearly define the scope of the invention.
The current mainstream codecs include: the SBC audio encoder, the AAC-LC audio encoder, the aptX series audio encoder, the LDAC audio encoder and the Bluetooth International Association Bluetooth Sig are combined with a plurality of manufacturers to provide the LC3 audio encoder, because the LC3 encoder has the advantages of low delay, high tone quality and high coding gain, and is favored by the manufacturers, the method and the system for reducing the coding rate of the invention are mainly optimized aiming at the existing LC3 audio encoder, but the method and the system for reducing the audio coding rate of the invention are also suitable for other audio encoders. The method and system for reducing audio coding rate of the present invention are described below by taking an LC3 audio encoder as an example.
In one embodiment of the present invention, fig. 1 is a flow chart illustrating a method for reducing an audio coding rate according to the present invention. In this embodiment, the method for reducing the audio coding rate of the present invention comprises the steps of:
step S101: and identifying a sampling frequency step.
In this embodiment, the audio encoder identifies the sampling frequency of the encoded input signal and determines whether the sampling frequency is the lowest sampling frequency. In one specific example of the present invention, the sampling frequency of 8KHz is the lowest sampling frequency in the LC3 audio encoder specification. When the sampling frequency of the input signal is detected to be 8KHz, encoding is carried out according to the standard encoding flow of the LC3 audio encoder, the encoding rate cannot cause waste, and the operation step of reducing the encoding rate is not needed; when the sampling frequency of the input signal is detected to be other sampling frequencies, the following steps are carried out.
Step S102: and a bandwidth detection step.
In one embodiment of the invention, the effective bandwidth of the input signal is detected by a bandwidth detection module, when the effective bandwidth of the input signal is equal to the nyquist frequency, the encoding is carried out according to the standard encoding flow of the LC3 audio encoder, no other operation steps for reducing the audio encoding rate are needed, and when the effective bandwidth is less than the nyquist frequency, the following steps are carried out. Wherein in one embodiment of the present invention, the narrow band frequency range is 0-4KHz, the wide band frequency range is 0-8KHz, the semi-ultra-wideband frequency range is 0-12KHz, the ultra-wideband frequency range is 0-16KHz, and the full band frequency range is 0-20KHz.
Step S103: and (5) code rate conversion.
In one example of the invention, the encoding rate is translated into the first number of bytes available per frame for one channel encoding according to the specifications of the LC3 audio encoder. Wherein, the first byte number is represented by nbytes, and the calculation method is as follows:
wherein bitrate represents the coding input parameter "code rate", and N is the frame length according to different input signals ms 7.5 ms or 10 ms, f scal The value of (b) is determined by the following formula:
Step S104: and calculating the second byte number.
In one embodiment of the invention, the second number of bytes is obtained by subtracting the third number of bytes of overhead necessary per frame from the first number of bytes. The second byte number means the number of bytes available for quantization and arithmetic coding of the spectral coefficients of each frame, and is represented by the symbol nbytes _ spec. The third byte number is indicated by the symbol nbytes _ directory. In one example of the invention, the third byte number nbytes _ directory of the necessary overhead per frame comprises two aspects: the number of overhead bytes per frame, which is specified according to the LC3 audio encoder standard, is fixed, e.g., 38 bits are required for the frequency domain noise shaping module and 8 bits are required for the global gain. The bit number required by each frame floating overhead is related to the spectrum distribution characteristics of the input coding signal, and the result can be obtained only after the related module operation is executed, for example, when the code rate and coding parameter updating module is executed, the operation of the long-term post-filter module is finished, so that the bit number required by the long-term post-filter module can be obtained. And the time domain noise shaping module is not operated when the code rate and coding parameter updating module is executed, so that the bit number used by the time domain noise shaping module is uncertain, and at the moment, calculation is carried out according to the maximum possible byte number corresponding to the number of TNS filters in the actually used time domain noise shaping module. Fig. 2 shows the positions of modules in the encoding process of the LC3 audio encoder, wherein the method for reducing the audio encoding rate of the present invention is mainly embodied in the code rate and encoding parameter updating module and the quantization module.
Step S105: and calculating the fourth byte number.
In an embodiment of the invention, the fourth number of bytes is the number of bytes available for encoding per frame according to the bandwidth, indicated by the symbol nbytes _ update. Wherein, the calculation formula of the fourth byte number is as follows:
in an embodiment of the present invention, the adjustment factor of the number of bytes of the spectral coefficient is selected according to the requirement of the sound quality, and the selection range is
The lower the byte number adjustment factor value of the spectral coefficient is, the closer the byte number available for the spectral coefficient is to the effective bandwidth/Nyquist frequency, the greater the actual tone quality is lost, and the lower frequency spectrum is usually higher in energy and higher in importance on the tone quality; the higher the byte number of the spectral coefficient, the better the tone quality, and the lower the code rate reduction amplitude if the value is closer to the Nyquist frequency/effective bandwidth. In addition, values within the above ranges may be selected based on subjective and objective test results in a particular application scenario.
In one embodiment of the present invention, the substrate is,
referred to as the second byte count coefficient. The fourth byte number available for encoding each frame according to the new bandwidth can be calculated by the formula.
Step S106: and updating the coding control mark.
In one embodiment of the present invention, the control flag of the encoder encoding is updated according to the fourth byte count. In one example of the present invention, the encoding control flag is updated according to the code rate specified by the LC3 audio encoder specification and the fourth byte number, wherein the updating step is determined by the LC3 audio encoder standard specification. For example, the temporal domain noise shaping LPC emphasis flag Tns _ LPC _ weighting, the rate flag rateFlag, and the mode flag modeFlag are updated.
In one embodiment of the present invention, the method of reducing an audio coding rate of the present invention includes a quantization step. In this embodiment, the number of spectral lines N is encoded by encoding each frame E Is replaced by
Wherein N is F For the number of samples per frame, and also the maximum spectral line per frame, the following formula is updated:
in this embodiment, N is E Is replaced by
The lastnz calculation is updated where lastnz _ tail refers to the excess of the bandwidth edge, typically the transition from pass-band to stop-band when filtering a wideband signal into a narrowband signal. In one example of the present invention, lastnz _ tail need not be all zeros, and lastnz _ tail may take a smaller value, e.g., 3 to 5. Wherein the lastnz calculation is described as:
lastnz＝N E
while(lastnz>2&&X q [lastnz-1]＝＝0&&X q [lastnz-2]＝＝0)
{lastnz-＝2；}
in one example of the present invention, the quantization step is not performed when the sampling rate of the input signal is 8KHz.
The method for reducing the audio coding rate can adjust the coding rate of the audio coder according to the difference of the input signals, save the arithmetic coding operation amount of the audio coder and decoder at the coding end and the decoding end by reducing the real-time coding rate, and prolong the service time of the equipment. Fig. 3 is a schematic diagram illustrating subjective difference level changes before and after rate modification according to an application example of the method for reducing audio coding rate of the present invention. Wherein the encoding rate before modification is 128kbps, and the encoding rate after modification is 108.8kbps, wherein the encoding rate is reduced by 15%, and the subjective difference grade change is within the threshold range specified by the Bluetooth International Union. In addition, the method for reducing the audio coding rate can be applied to coding signals with the frame length of 10 milliseconds and the frame length of 7.5 milliseconds and coding signals with all sampling rates except the lowest sampling rate of 8KHz, and has wide application range.
In an embodiment of the present invention, as shown in fig. 4, the system for reducing an audio coding rate of the present invention includes a bandwidth detection module, in an example of the present invention, an effective bandwidth of an input signal is detected by the bandwidth detection module, when the effective bandwidth of the input signal is equal to a nyquist frequency, the input signal is encoded according to a standard encoding flow of an LC3 audio encoder, and when the effective bandwidth is less than the nyquist frequency, no other operation is required to be performed. Wherein in one embodiment of the present invention, the narrow band frequency range is 0-4KHz, the wide band frequency range is 0-8KHz, the semi-ultra-wideband frequency range is 0-12KHz, the ultra-wideband frequency range is 0-16KHz, and the full band frequency range is 0-20KHz.
In this embodiment, the system for reducing audio coding rate of the present invention comprises a code rate and coding parameter updating module, which converts the coding rate into a first number of bytes available per frame when coding a sound channel, and subtracts a third number of bytes of necessary overhead per frame from the first number of bytes to obtain a second number of bytes available for quantizing and arithmetically coding the spectral coefficient per frame, wherein the selection range is selected from the first number of bytes and the third number of bytes of necessary overhead per frame
The coefficient of the second byte number is obtained by multiplying the ratio of the effective bandwidth to the Nyquist frequency by the coefficient of the spectral coefficient byte number, the fourth byte number is obtained by summing the product of the third byte number and the coefficient of the second byte number, and then the coding control mark is updated according to the fourth byte number and the dependent code rate.
In this embodiment, the system for reducing an audio encoding rate of the present invention further comprises a quantization module. Wherein each frame is encoded into a spectrumNumber of lines N E Is replaced by
Wherein N is F Is the number of samples per frame, and is also the number of maximum spectral lines per frame. In one example of the present invention, the quantization module does not operate when the sampling rate of the input signal is 8KHz.
In one example of the present invention, the positions of the bandwidth detection module, the code rate and coding parameter update module, and the quantization module of the system for reducing the audio coding rate of the present invention in the LC3 audio encoder are shown in the gray portion of fig. 2.
Fig. 5 shows a speech spectrogram of a monaural channel with a sampling rate of 48000Hz and an effective bandwidth of 24kHz, fig. 6 shows a speech spectrogram of a monaural channel with a sampling rate of 48000Hz and an effective bandwidth of 12kHz, fig. 7 is a subjective difference level diagram obtained after encoding and decoding two audio using configurations with a bitrate of 125Kbps, fig. 7 shows that the audio with a smaller effective bandwidth has a higher subjective difference level score and most of the audio with a bitrate exceeding 0, however, the improvement of the audio with a smaller effective bandwidth is meaningless because the human ear hardly feels, the reduction of the effective bandwidth means the reduction of the information amount, and if the audio is still encoded according to the nyquist frequency, the computational capability is wasted.
The system for reducing the audio coding rate can adjust the coding rate of the audio coder according to the difference of input signals, saves the arithmetic coding operation amount of the audio coder-decoder at the coding end and the decoding end by reducing the real-time coding rate, avoids the unnecessary operation capacity waste and prolongs the service time of equipment at the same time. In addition, the method for reducing the audio coding rate can be applied to coding signals with the frame length of 10 milliseconds and the frame length of 7.5 milliseconds and coding signals with all sampling rates except the lowest sampling rate of 8KHz, and has wide application range.
Claims (9)
1. A method for reducing an audio coding rate, comprising:
a sampling frequency identification step of performing encoding according to a standard flow when a sampling frequency of an input signal is a lowest sampling frequency, and performing the following steps when the sampling frequency of the input signal is not the lowest sampling frequency;
a bandwidth detection step, when the effective bandwidth is equal to the Nyquist frequency, the coding is carried out according to the standard flow, and when the effective bandwidth is less than the Nyquist frequency, the following steps are carried out;
a code rate conversion step, which is used for converting the coding rate into the first byte number available in each frame during coding;
calculating a second byte number, namely subtracting a third byte number of necessary expense of each frame from the first byte number to obtain a spectrum coefficient of each frame, quantizing and arithmetically encoding the available second byte number;
calculating a fourth byte number, namely calculating the available fourth byte number according to the new frame coding of the bandwidth through the second byte number and the third byte number; and
and updating the coding control mark according to the fourth byte number and the dependent code rate, wherein the step of calculating the fourth byte number comprises the following steps:
selecting a spectral coefficient byte number adjusting factor according to the requirement of tone quality, wherein the selection range is between 1 and the ratio of the Nyquist frequency to the effective bandwidth;
calculating the second byte count coefficient that multiplies the ratio of the effective bandwidth to the Nyquist frequency by the spectral coefficient byte count adjustment factor;
calculating the fourth byte count by summing the third byte count multiplied by the second byte count and the second byte count coefficient.
2. A method of reducing an audio coding rate as defined in claim 1, further comprising a quantization step that replaces the number of encoded spectral lines per frame with a product of the number of maximum spectral lines per frame and the ratio of the effective bandwidth to the nyquist frequency.
3. A method of reducing an audio encoding rate as defined in claim 1 wherein the lowest sampling frequency is the sampling frequency of the narrowband speech at 8KHz.
4. A method for reducing an audio encoding rate as defined in claim 1 wherein the third number of bytes of overhead necessary per frame comprises a fixed number of bytes of overhead fixed per frame and a number of bytes of overhead floating relative to a signal spectral distribution characteristic.
5. A method for reducing an audio encoding rate as defined in claim 2 wherein, in the quantizing step, when the input sampling frequency is the lowest sampling frequency, this step is not performed.
6. A system for reducing an audio encoding rate, comprising:
the bandwidth detection module is used for detecting the effective bandwidth of the input signal;
the code rate and coding parameter updating module is used for converting a coding rate into a first byte number available for each frame when a sound channel is coded, subtracting a third byte number of necessary overhead of each frame from the first byte number to obtain a second byte number available for quantizing and arithmetically coding a spectral coefficient of each frame, calculating through the second byte number and the third byte number to obtain a fourth byte number available for coding each frame according to new bandwidth, and then updating a coding control mark according to the fourth byte number and a dependent code rate; and
a quantization module that replaces the number of encoded spectral lines per frame with a product of the number of maximum spectral lines per frame and the ratio of the effective bandwidth to the nyquist frequency, wherein the step of calculating a fourth number of bytes comprises:
selecting a spectral coefficient byte number adjusting factor according to the requirement of tone quality, wherein the selection range is between 1 and the ratio of the Nyquist frequency to the effective bandwidth;
calculating the second byte count coefficient that multiplies the ratio of the effective bandwidth to the Nyquist frequency by the spectral coefficient byte count adjustment factor;
calculating the fourth byte count by summing the third byte count multiplied by the second byte count and the second byte count coefficient.
7. The system of claim 6, wherein in the code rate and coding parameter update module, when calculating the fourth byte number, a spectral coefficient byte number adjustment factor ranging from 1 to a ratio of the nyquist frequency to the effective bandwidth is selected, the ratio of the effective bandwidth to the nyquist frequency is multiplied by the spectral coefficient byte number adjustment factor to obtain the second byte number coefficient, and the product of the third byte number and the second byte number coefficient is summed to obtain the fourth byte number.
8. The system for audio coding rate reduction according to claim 6, wherein the processing of the system for audio coding rate reduction is not required when the input signal sampling frequency is 8KHz.
9. The system for reducing an audio coding rate of claim 6 wherein processing by the system for reducing an audio coding rate is not required when the effective bandwidth is equal to the Nyquist frequency.
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| CN112365897A (en) * | 2020-11-26 | 2021-02-12 | 北京百瑞互联技术有限公司 | Method, device and medium for self-adaptively adjusting interframe transmission code rate of LC3 encoder |
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