CN112270928A - Method, device and storage medium for reducing code rate of audio encoder - Google Patents
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Abstract
The application discloses a method, a device and a medium for reducing code rate of an audio encoder, belonging to the technical field of wireless communication, wherein the method comprises the following steps: judging whether the original byte number occupied by at least one frame of data in the audio encoder has redundancy or not; recalculating, if the original byte number occupied by at least one frame of data has redundancy, recalculating the byte number actually occupied by the effective value of the data of which the original byte number has redundancy; and an adjusting step, namely reallocating the required byte number of the at least one frame data according to the byte number actually occupied by the effective value of the at least one frame data. By adjusting the coded bit stream output by each frame, the code rate is effectively reduced on the premise of ensuring that the coding result is not changed, so that the aerial bandwidth can be saved.
Description
Technical Field
The present application relates to the field of wireless communications technologies, and in particular, to a method, an apparatus, and a storage medium for reducing a code rate of an audio encoder.
Background
Currently, the mainstream bluetooth audio encoders include the following:
SBC: the standard coding format of the Bluetooth audio, the coding format mandatory by the A2DP protocol, is most widely used, and is required to be supported by all Bluetooth audio equipment, but the tone quality is general;
AAC-LC is a high-compression-ratio coding algorithm, 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 SBC, a plurality of Bluetooth devices are all based on an embedded platform, the battery capacity is limited, the processor has poorer operation capability and the memory is limited;
aptX series: the details of sound reservation are more, the actual listening feeling is better than the former two, the audio coding technology has higher efficiency than the traditional Bluetooth coding, the code rate of aptX is 384kbps, the code rate of aptX-HD is 576kbps, the audio coding technology is a unique technology of high pass, and equipment adopting the APTX technology needs to apply authorization to the high pass and pay authorization fee, and the transmitting and receiving ends need to support the equipment;
LDAC has good sound quality, but the code rate is also very high, which is 330kbps, 660kbps and 990kbps respectively, because the wireless environment of the Bluetooth device is very complicated, the stable support of the high code rate has certain difficulty, and an audio coding technology developed by SONY company is only used on the transmitting and receiving device of SONY company. Therefore, only SONY set of transmitting and receiving equipment supporting LDAC audio coding technology is bought, and Bluetooth audio data transmission of LDAC coding can be supported;
LC 3: the low power consumption, the low delay, the high tone quality and the high coding gain, and no special fee in the bluetooth field, compared with the former four kinds of encoders, the advantages are many, so the LC3 receives the attention of the majority of manufacturers.
The LC3 reduces the code rate at the same sampling rate and improves the sound quality compared with the existing CVSD and WBS, but because of the waveform coding technology, the compression efficiency is very low compared with the traditional vocoder, for example, the recommended code rate of the LC3 at 8k sampling rate is 28kbps, but the maximum code rate of AMR-NB applied to WCDMA is only 12.2kbps, the average code rate is below 10kbps, the maximum code rate of EVRC-a applied to CDMA system is 8kbps, and the average code rate is about 5kbps, while the lowest code rate of EVS applied to VoLTE proposed by 3GPP Release12 is 5.9 kbps. Higher code rate has occupied great bandwidth, needs more transmitting power, and in public occasion, bluetooth equipment is more, can cause the interference each other, has influenced user experience, and the degree of mutual interference has very big relation with bandwidth, transmitting power occupied.
Based on the characteristics of the call, a party of the call usually only has a part of the time to speak, and according to statistics, a party of the call usually only has about 35% of the time to speak, while the LC3 of the current standard uses a fixed code rate, that is, each frame of speech after the encoding is started uses the same available bit number, which causes the party of the call to still use the same code rate encoding when the party does not speak, resulting in waste.
In the prior art, in order to save code rate, the vocoders in 3GPP and 3GPP2 as described above all use very complex voice activity detection and code rate decision algorithms, and although the code rate can be saved to a great extent, the resource consumption is large, such as code space, data space, and computational resource. These methods are not suitable for bluetooth low energy oriented platforms, because the number of bytes and the computing power of the bluetooth low energy platform are limited.
For a voice frame (usually 1 for voice activity detection), the main coded bits are used for encoding side information and arithmetic coding, and the remaining small number of coded bits are used for encoding the residual signal.
For non-voice frames (the voice activity detection result is 0), such as a mute frame or noise with small energy, after the encoding of the side information is completed, only a few encoding bits are needed for arithmetic encoding and residual encoding, so many bits will remain after the encoding of a frame is completed, and in the current LC3 standard specification, these remaining bits will be filled to 0 and sent to the decoder, resulting in a waste of bandwidth resources.
Disclosure of Invention
The application aims at the problem that an LC3 encoder which meets the standard specification at present is low in efficiency, particularly the problem of the waste of encoding bits, and provides a method, a device and a storage medium for reducing the code rate of an audio encoder.
In order to achieve the purpose, the technical scheme adopted by the application is as follows: a method of reducing a code rate of an audio encoder is provided, comprising: judging whether the original byte number occupied by at least one frame of data in the audio encoder has redundancy or not; recalculating, if the original byte number occupied by at least one frame of data has redundancy, recalculating the byte number actually occupied by the effective value of the data of which the original byte number has redundancy; and an adjusting step, namely reallocating the required byte number of the at least one frame data according to the byte number actually occupied by the effective value of the at least one frame data.
The application adopts another technical scheme that: there is provided a device for reducing a code rate of an audio encoder, comprising: the judgment module is used for judging whether the original byte number occupied by at least one frame of data in the audio encoder has redundancy; a recalculation module for recalculating the byte number actually occupied by the effective value of the data of which the original byte number has redundancy if the original byte number occupied by at least one frame of data has redundancy; and the adjusting module is used for reallocating the required byte number to the at least one frame data according to the byte number actually occupied by the effective value of the at least one frame data.
The application adopts another technical scheme that: a computer-readable storage medium is provided, in which computer-executable instructions are stored, and when executed by a processor, the method for reducing the bitrate of an audio encoder in the first aspect is implemented.
The application adopts another technical scheme that: there is provided a computer apparatus comprising a processor and a memory, the memory storing computer instructions, wherein the processor operates the computer instructions to implement the method of reducing a code rate of an audio encoder in scheme one when executed.
The beneficial effect of this application is: the method, the device and the storage medium for reducing the code rate of the audio encoder are provided, and the code rate is effectively reduced by adjusting the coded bit stream output by each frame on the premise of ensuring that the coding result is not changed, so that the air bandwidth can be saved.
Drawings
FIG. 1 is a flow chart illustrating an embodiment of a method for reducing a bitrate of an audio encoder according to the present application;
FIG. 2 is a diagram of an embodiment of a bitstream output by a current frame of an audio encoder according to a standard encoding of the present application;
FIG. 3 is a diagram illustrating an embodiment of a code stream output by a current frame of an audio encoder according to the present application after a code rate is reduced;
FIG. 4 is a schematic diagram of one embodiment of an audio encoder encoding module of the present application;
fig. 5 is a flowchart illustrating an embodiment of an apparatus for reducing a bitrate of an audio encoder according to the present application.
Detailed Description
The following detailed description of the preferred embodiments of the present application, taken in conjunction with the accompanying drawings, will provide those skilled in the art with a better understanding of the advantages and features of the present application, and will make the scope of the present application more clear and definite.
It is noted that, herein, relational terms such as first and second, and the like may be used solely to distinguish one entity or action from another entity or action without necessarily requiring or implying any actual such relationship or order between such entities or actions. Also, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or apparatus. Without further limitation, an element defined by the phrase "comprising … …" does not exclude the presence of other identical elements in a process, method, article, or apparatus that comprises the element.
In the encoding process of the audio encoder used in the present application, the auxiliary information encoding, the arithmetic encoding, and the residual encoding are all completed according to the standard specification, and are not described herein again.
Fig. 1 is a flowchart illustrating an embodiment of a method for reducing a bitrate of an audio encoder according to the present application.
In the embodiment shown in fig. 1, the method for reducing the bitrate of the audio encoder of the present application includes a process S101, a process S102, and a process S103.
The process S101 shown in fig. 1 is a step of determining whether there is redundancy in the original byte number occupied by at least one frame of data in the audio encoder.
In an embodiment of the present application, taking the LC3 encoder as an example, for the process of calculating the original byte number redundancy occupied by each frame data in the audio encoder, the number of bytes occupied by the auxiliary information encoding, the arithmetic encoding and the residual information encoding is subtracted from the total byte number available for the current frame data in the audio encoder, and the remaining is the redundancy of the original byte number occupied by the current frame data in the audio encoder. In this embodiment, if the number of original bytes occupied by the current frame data of the audio encoder has redundancy, it indicates that the number of original bytes occupied by the current frame data of the audio encoder has a bandwidth waste phenomenon, and if the number of original bytes occupied by the current frame data of the audio encoder has no redundancy, the next operation is performed according to the standard specification.
In a specific example of the present application, nbits is the total number of bits available for the current frame, nbits _ side is the number of bits of the auxiliary information code used by the current frame, nbits _ residual _ enc is the number of bits of the residual information code used by the current frame, nbits _ ari is the number of bits of the arithmetic code used by the current frame, given _ bits is the number of bits that can be saved by the data of the current frame, and given _ bytes is the number of bytes that can be saved by the current frame; the following '8' refers to the number of protection bits, and if the number of bits of a certain part is not an integral multiple of 8, a part is lost after right shifting by 3 bits; therefore, protection is required. For simplicity, the output bit number of each module increases protection; the bit number that can be saved by the current frame of the audio encoder can be obtained as follows: saved _ bytes ═ nbits > >3) - ((nbits _ side +8) > >3) - ((nbits _ ari +8) > >3) - ((nbits _ residual _ enc +8) > > 3).
The process S102 shown in fig. 1 is a recalculation step, in which if the original byte number occupied by at least one frame of data has redundancy, the byte number actually occupied by the effective value of the data whose original byte number has redundancy is recalculated.
In one embodiment of the present application, the number of bytes occupied by the valid value of the redundant frame data in the audio encoder is the total number of bytes occupied by the current frame data of the audio encoder minus the number of bytes occupied by the current frame data of the audio encoder. The current step provides for the next re-allocation of bytes to the frame data for which the original bytes are redundant.
In one specific example of the present application, preferably, in an LC3 audio encoder with a frame length of 10ms, a sampling rate of 16k, and a code rate of 64 kbps; nbytes is 80, i.e. the number of bytes output per frame is 80. The number of bytes required to be output by the current frame of the audio encoder is as follows: nbytes-saved _ bytes.
The process S103 shown in fig. 1 is an adjustment step of reallocating the required number of bytes of the at least one frame of data according to the number of bytes actually occupied by the effective value of the at least one frame of data.
In an embodiment of the present application, if the number of bytes actually occupied by the valid value of any frame data is smaller than the minimum required number of bytes for encoding a frame specified by the audio encoder standard, the number of bytes reallocated to the data is the number of bytes specified by the encoder standard.
In one embodiment of the present application, preferably, the standard specifies that the minimum required number of bytes for a frame to be encoded is 20; therefore, the 20 bytes of storage space is reallocated for the frame data of which the effective value actually occupies less bytes than the minimum required byte number for encoding one frame specified by the audio encoder standard. FIG. 2 is a code stream encoded and outputted according to the standard, which occupies 80 bytes, and it can be seen that most of the data in the front is 0, which wastes valuable bandwidth; fig. 3 shows a codestream output according to the present application, which occupies 20 bytes and saves 60 bytes.
In an embodiment of the present application, if the number of bytes actually occupied by the effective value of any frame of data is not less than the minimum required number of bytes for encoding a frame specified by the audio encoder standard, the number of bytes reallocated to the data is the number of bytes actually occupied by the effective value. The current step effectively utilizes the code rate of the current frame data, saves the number of bytes of the unused original bytes of the current frame of the audio encoder, and saves the air bandwidth.
In one embodiment of the present application, the adjusting step comprises: copying the arithmetically coded data, the auxiliary information data and the residual information data from the number of original bytes occupied by at least one frame of data in the audio encoder; and putting the copied arithmetically encoded data, the data of the auxiliary information, and the data of the residual information in the number of bytes reallocated thereto, respectively. The effective utilization of the byte number required by the current frame data is realized.
In one embodiment of the present application, the process of copying the arithmetically encoded data from the number of original bytes occupied by at least one frame of data in the audio encoder includes: the length of arithmetically encoded data in at least one frame data in an audio encoder is calculated, and a start point of the arithmetically encoded data is recorded.
In one specific example of the present application, preferably, in an LC3 audio encoder with a frame length of 10ms, a sampling rate of 16k, and a code rate of 64 kbps; calculating a first Copy length Copy _ length1 and a first Copy start point Copy start1, which correspond to the length of the arithmetically encoded data and Copy the arithmetically encoded data from the original byte number start point to the new byte number; the operation steps are as follows:
Copy_length1＝((nbits_ari+8)>>3)；
Copy_start1＝0；
in a specific embodiment of the present application, the copying the data of the auxiliary information and the data of the residual information from the number of original bytes occupied by at least one frame of data in the audio encoder includes: calculating the length of the auxiliary information data and the residual information data in at least one frame of data in the audio encoder, and recording the starting point of the auxiliary information data and the residual information data.
In one specific example of the present application, preferably, in an LC3 audio encoder with a frame length of 10ms, a sampling rate of 16k, and a code rate of 64 kbps; calculating a second Copy length2 and a second Copy start point Copy start2, which correspond to the data of the auxiliary information and the data of the residual information; the operation steps are as follows: copy _ length2 ═ ((nbits _ side +8) > >3) + ((nbits _ residual _ enc +8) > > 3); copy _ start2 ═ (nbits > >3) -Copy _ length 2;
after the steps are finished, the next operation is continued according to the requirements related to the LC3 codec standard specification. After the relevant steps of the present application are completed, a new overall operational flow diagram for audio encoder encoding is shown in fig. 4; the byte number of the data frame is reduced in the encoding process, so that the code rate is reduced, and the condition of air bandwidth waste is solved.
Fig. 5 is a flowchart illustrating an embodiment of an apparatus for reducing a bitrate of an audio encoder according to the present application.
In the embodiment shown in fig. 5, the apparatus for reducing the bitrate of the audio encoder of the present application includes: module 501, module 502, and module 503.
The block 501 shown in fig. 5 is a determining block: and the module is used for judging whether the original byte number occupied by at least one frame of data in the audio encoder has redundancy. The module 502 shown in FIG. 5 is a recalculation module: and the module is used for recalculating the byte number actually occupied by the effective value of the data with the redundancy original byte number if the original byte number occupied by the data of at least one frame has the redundancy. The block 503 shown in fig. 5 is an adjustment block: and the module is used for reallocating the required byte number of the at least one frame data according to the byte number actually occupied by the effective value of the at least one frame data.
In one embodiment of the present application, the modules 501, 502, and 503 in the apparatus for reducing a bitrate of an audio encoder of the present application may be directly in hardware, in a software module executed by a processor, or in a combination of the two.
A software module may reside in RAM memory, flash memory, ROM memory, EPROM memory, EEPROM memory, registers, hard disk, a removable disk, a CD-ROM, or any other form of storage medium known in the art. An exemplary storage medium is coupled to the processor such the processor can read information from, and write information to, the storage medium.
The Processor may be a Central Processing Unit (CPU), other general-purpose Processor, a Digital Signal Processor (DSP), an Application Specific Integrated Circuit (ASIC), a Field Programmable Gate Array (FPGA), other Programmable logic devices, discrete Gate or transistor logic, discrete hardware components, or any combination thereof. A general purpose processor may be a microprocessor, but in the alternative, the processor may be any conventional processor, controller, microcontroller, or state machine. A processor may also be implemented as a combination of computing devices, e.g., a combination of a DSP and a microprocessor, a plurality of microprocessors, one or more microprocessors in conjunction with a DSP core, or any other such configuration. In the alternative, the storage medium may be integral to the processor. The processor and the storage medium may reside in an ASIC. The ASIC may reside in a user terminal. In the alternative, the processor and the storage medium may reside as discrete components in a user terminal.
The apparatus for optimizing the memory of the audio/video codec of the present application may be configured to execute the method for reducing the code rate of the audio encoder described in any of the above embodiments, and the implementation principle and the technical effect are similar, which are not described herein again.
In a specific embodiment of the present application, a computer-readable storage medium stores computer instructions, which are operable to perform a method for reducing a bitrate of an audio encoder as described in any of the embodiments.
In a specific embodiment of the present application, a computer device includes a processor and a memory, the memory storing computer instructions, wherein the processor operates the computer instructions to perform a method for reducing a code rate of an audio encoder as described in any of the embodiments.
The method has good effect on LC3 voice coding, and the average code rate of the test vector shown at the beginning of the method can be reduced from 64kbps to 42.6kbps, and the sound quality is the same.
The present application exemplifies speech coding, and the principle can be applied to speech coding with a 10ms frame length and a 7.5 ms frame length, and all sampling rates; it can also be used for music coding of 10ms frame length and 7.5 ms frame length, and all sampling rates. The method has the advantages that the code rate during voice communication is reduced, so that the air bandwidth can be saved; the implementation of the method is very simple, and only a small amount of control codes need to be added for updating the code rate.
In the embodiments provided in the present application, it should be understood that the disclosed apparatus and method may be implemented in other ways. For example, the above-described apparatus embodiments are merely illustrative, and for example, the division of the units is only one logical division, and other divisions may be realized in practice, for example, a plurality of units or components may be combined or integrated into another system, or some features may be omitted, or not executed. In addition, the shown or discussed mutual coupling or direct coupling or communication connection may be an indirect coupling or communication connection through some interfaces, devices or units, and may be in an electrical, mechanical or other form.
The units described as separate parts may or may not be physically separate, and parts displayed as units may or may not be physical units, may be located in one place, or may be distributed on a plurality of network units. Some or all of the units can be selected according to actual needs to achieve the purpose of the solution of the embodiment.
The above description is only an example of the present application and is not intended to limit the scope of the present application, and all equivalent structural changes made by using the contents of the specification and the drawings, which are directly or indirectly applied to other related technical fields, are included in the scope of the present application.
Claims (8)
1. A method for reducing a code rate of an audio encoder, comprising:
judging whether the original byte number occupied by at least one frame of data in the audio encoder has redundancy or not;
recalculating, if the original byte number occupied by the at least one frame of data has redundancy, recalculating the byte number actually occupied by the effective value of the data of which the original byte number has redundancy; and
and an adjusting step, namely reallocating the required byte number of the at least one frame of data according to the actually occupied byte number of the effective value of the at least one frame of data.
2. The method for reducing the bitrate of an audio encoder of claim 1, further comprising:
if the number of bytes actually occupied by the effective value of the data of any frame is smaller than the minimum required number of bytes of a frame encoding specified by the audio encoder standard, the number of bytes reallocated to the data is the number of bytes specified by the encoder standard.
3. The method for reducing the bitrate of an audio encoder of claim, wherein the adjusting step comprises:
copying the arithmetically coded data, the auxiliary information data and the residual information data from the original byte number occupied by at least one frame of data in the audio encoder; and
and respectively putting the copied data of the arithmetic coding, the copied data of the auxiliary information and the copied data of the residual error information into the number of bytes which are redistributed to the arithmetic coding, the auxiliary information and the residual error information.
4. The method for reducing the bitrate of an audio encoder according to claim 3, wherein the copying of the arithmetically encoded data from the number of original bytes occupied by at least one frame of data in the audio encoder comprises:
the length of arithmetically encoded data in at least one frame data in the audio encoder is calculated, and a start point of the arithmetically encoded data is recorded.
5. The method for reducing the bitrate of the audio encoder according to claim 3, wherein the copying the data of the auxiliary information and the data of the residual information from the number of original bytes occupied by at least one frame of data in the audio encoder comprises:
and calculating the lengths of the auxiliary information data and the residual information data in at least one frame of data in the audio encoder, and recording the starting points of the auxiliary information data and the residual information data.
6. An apparatus for reducing a code rate of an audio encoder, comprising:
the judgment module is used for judging whether the original byte number occupied by at least one frame of data in the audio encoder has redundancy;
a recalculation module for recalculating the byte number actually occupied by the effective value of the data of which the original byte number has redundancy if the original byte number occupied by the at least one frame of data has redundancy; and
and the adjusting module is used for reallocating the required byte number of the at least one frame of data according to the actually occupied byte number of the effective value of the at least one frame of data.
7. A computer readable storage medium storing computer instructions, wherein the computer instructions are operable to perform the method of reducing a code rate of an audio encoder of any of claims 1-5.
8. A computer device comprising a processor and a memory, the memory storing computer instructions, wherein the processor operates the computer instructions to perform the method of reducing a code rate of an audio encoder of any of claims 1-5.
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