CN111986685A - Audio coding and decoding method and system for realizing high sampling rate - Google Patents
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Abstract
The application discloses an audio coding and decoding method and system for realizing high sampling rate, an LC3 audio coder and decoding method and a storage medium, and belongs to the technical field of audio coding and decoding. The audio coding and decoding method for realizing the high sampling rate comprises the following steps: a frequency dividing step, which divides a first audio signal with a certain bandwidth to obtain at least two second audio signals, wherein the frequency bands of the at least two second audio signals are integrated into a certain bandwidth; coding and decoding, namely performing a standard coding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and a synthesizing step of synthesizing the at least two third audio signals to obtain a fourth audio signal. The application of the method and the device for obtaining the high sampling rate in the audio coding and decoding process improves the audio tone quality.
Description
Technical Field
The present application relates to the field of audio encoding and decoding technologies, and in particular, to an audio encoding and decoding method and system, an LC3 audio codec encoding and decoding method, and a storage medium for realizing a high sampling rate.
Background
Currently mainstream bluetooth audio codecs include: SBC audio codecs, which are mandated by the A2DP protocol and are most widely used; the AAC-LC audio codec has good tone quality and wide application range, and is supported by a plurality of mainstream mobile phones; the aptX series audio codec has good tone quality, high code rate, is a unique technology for high pass and is relatively closed; the LDAC audio codec has good tone quality, but high code rate, is a unique Sony technology and is also closed; the LHDC audio codec has good tone quality and high coding rate. For the above reasons, the Bluetooth international association Bluetooth Sig has introduced the LC3 audio codec with many manufacturers, which has the advantages of low delay, high sound quality and coding gain and no special fee in the Bluetooth field, and is receiving attention from many manufacturers.
Along with the improvement of quality of life, people's requirement to audio tone quality is higher and higher, and the demand to supporting high resolution audio frequency is bigger and bigger in the more and more bluetooth audio equipment of well high-end. In the above audio encoder, the maximum sampling rate of the SBC audio codec, the AAC-LC audio codec, the aptX audio codec, and the LC3 audio codec is 48KHz, and the maximum sampling rate of the LDAC audio codec and the LHDC audio codec is 96 KHz. The LDAC Audio codec and the LHDC Audio codec may have sampling rates that meet the High Resolution Audio (High Resolution Audio) sampling rate requirements, while other codecs may not meet the High Resolution Audio sampling rate requirements. The realization of high sampling rate in the audio coding and decoding process is the key to improve the audio quality.
Disclosure of Invention
In view of the above technical problems in the prior art, the present application provides an audio codec method and system, an LC3 audio codec method, and a storage medium for realizing a high sampling rate.
In one embodiment of the present application, an audio encoding and decoding method for achieving a high sampling rate is provided, including: a frequency dividing step, which divides a first audio signal with a certain bandwidth to obtain at least two second audio signals, wherein the union of the frequency bands of the at least two second audio signals is the certain bandwidth; coding and decoding, namely performing a standard coding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and a synthesizing step of synthesizing the at least two third audio signals to obtain a fourth audio signal.
In another aspect of the present application, an audio encoding and decoding system for achieving a high sampling rate is provided, including: the frequency dividing module is used for dividing frequency of a first audio signal with a certain bandwidth to obtain at least two second audio signals, and the union set of frequency bands of the at least two second audio signals is the certain bandwidth; the coding and decoding module is used for carrying out a standard coding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and the synthesis module synthesizes the at least two third audio signals to obtain a fourth audio signal.
In another technical solution of the present application, there is provided an LC3 audio codec method, including: a frequency dividing step, which divides a first audio signal with a certain bandwidth to obtain at least two second audio signals, wherein the union of the frequency bands of the at least two second audio signals is the bandwidth of the first audio signal; coding and decoding, namely performing a standard coding and decoding process on at least two second audio signals by using a plurality of LC3 audio coders according to a preset sampling rate to obtain at least two third audio signals; and a synthesizing step of synthesizing the at least two third audio signals to obtain a fourth audio signal.
In another aspect of the present application, a computer-readable storage medium is provided, which stores computer instructions, wherein the computer instructions are operable to execute the audio encoding and decoding method in the first aspect to achieve a high sampling rate.
The beneficial effect of this application is: high sampling rate sampling is carried out on variable decoding audio in the audio coding and decoding process, and the tone quality of the coding and decoding audio is improved.
Drawings
FIG. 1 is a flow chart of an embodiment of an audio encoding and decoding method for realizing a high sampling rate according to the present application;
FIG. 2 is a schematic diagram of the operating principle of the quadrature mirror analysis filter of the present application;
FIG. 3 is a flow chart illustrating an application example of the audio encoding and decoding method for realizing high sampling rate according to the present application;
FIG. 4 is a flow chart illustrating an application example of the audio encoding and decoding method for realizing high sampling rate according to the present application;
FIG. 5 is a schematic diagram illustrating an example of an application effect of the audio encoding and decoding method for realizing a high sampling rate according to the present application;
fig. 6 is a schematic structural diagram of an embodiment of an audio codec system for implementing a high sampling rate according to the present application.
Detailed Description
In order to make the objects, technical solutions and advantages of the embodiments of the present application clearer, the technical solutions in the embodiments of the present application will be clearly and completely described below with reference to the drawings in the embodiments of the present application, and it is obvious that the described embodiments are some embodiments of the present application, but not all embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present application.
The terms "first," "second," "third," "fourth," and the like in the description and in the claims of the present application and in the above-described drawings (if any) are used for distinguishing between similar elements and not necessarily for describing a particular sequential or chronological order. It is to be understood that the data so used is interchangeable under appropriate circumstances such that the embodiments of the application described herein are, for example, capable of operation in sequences other than those illustrated or otherwise described herein. Furthermore, the terms "comprises," "comprising," and "having," and any variations thereof, are intended to cover a non-exclusive inclusion, such that a process, method, system, article, or apparatus that comprises a list of steps or elements is not necessarily limited to those steps or elements expressly listed, but may include other steps or elements not expressly listed or inherent to such process, method, article, or apparatus.
Fig. 1 shows a specific embodiment of the audio coding and decoding method for realizing a high sampling rate according to the present application.
In the embodiment shown in fig. 1, the audio coding and decoding method for realizing a high sampling rate of the present application includes: s101, a frequency division step, namely, frequency division is carried out on a first audio signal with a certain bandwidth to obtain at least two second audio signals, and the union of the frequency bands of the at least two second audio signals is a certain bandwidth; s102, coding and decoding, namely performing a standard coding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and S103, a synthesizing step of synthesizing at least two third audio signals to obtain a fourth audio signal.
In the embodiment shown in fig. 1, the audio coding and decoding method for implementing a high sampling rate includes a frequency dividing step S101, which divides a frequency of a first audio signal having a certain bandwidth to obtain at least two second audio signals, where a union of frequency bands of the at least two second audio signals is the bandwidth of the first audio signal. For example, a first audio signal with a bandwidth of 32KHz is obtained, and two second audio signals with frequency bands of 20Hz-16KHz and 16KHz-32KHz are obtained through frequency division processing. In addition, the frequency division processing can be performed on the second audio signal again according to the actual coding and decoding requirements. For example, the second audio signal with the frequency band of 20Hz-16KHz is subjected to frequency division processing, and two audio signals of 20Hz-8KHz and 8KHz-16KHz can be obtained.
In a specific embodiment of the present application, in the dividing step S101, the first audio signal is subjected to frequency division processing by a quadrature image analysis filter.
In this embodiment, the quadrature image analysis filter is used to divide a first audio signal having a certain bandwidth. And obtaining two new audio signals from the audio signal with a certain bandwidth through a quadrature mirror analysis filter, wherein the bandwidth of the audio signal obtained by frequency division is half of the bandwidth of the original signal. For example, the bandwidth of the original signal is 32KHz, and two audio signals with the frequency bands of 20Hz-16KHz and 16KHz-32KHz can be obtained by signal frequency division of the quadrature mirror analysis filter. Wherein, the audio signal of 20Hz-16KHz is called low-band audio signal, and the audio signal of 16KHz-32KHz is called high-band audio signal. The original audio signal can be divided into a low-band audio signal and a high-band audio signal with the same frequency bandwidth by a quadrature image analysis filter. The filter characteristics of the low-pass and high-pass filters in the quadrature mirror analysis filter are shown in fig. 2.
In one embodiment of the present application, the number of quadrature mirror analysis filters is greater than or equal to 1.
In this embodiment, a plurality of quadrature mirror analysis filters may be used to perform frequency division processing on the audio signal according to the sampling rate requirement in the actual encoding and decoding process. After frequency division processing is carried out on the audio signal by using a quadrature mirror analysis filter, a low-frequency band audio signal and a high-frequency band audio signal corresponding to the audio signal are obtained. The low-band audio signal or the high-band audio signal may then be further divided using the quadrature image analysis filter again. Thereby obtaining a plurality of second audio signals from the first audio signal with a certain bandwidth.
In one example of the application, the bandwidth of the first audio signal is 32KHz, for example. After passing through a quadrature mirror analysis filter, low-band audio signals with the frequency band of 20Hz-16KHz and high-band audio signals with the frequency band of 16KHz-32KHz are obtained. According to the requirement of the sampling rate of the specific coding and decoding process, the low-frequency band audio signal with the frequency band of 20Hz-16KHz or the high-frequency band audio signal with the frequency band of 16KHz-32KHz can be divided again by using the quadrature mirror image analysis filter. The frequency division result of the 20Hz-16KHz low-frequency band audio signal is to obtain the 20Hz-8KHz low-frequency band audio signal corresponding to the audio signal and the 8KHz-16KHz high-frequency band audio signal. If the high-frequency band audio signal of 16KHz-32KHz is subjected to frequency division, the low-frequency band audio signal of which the frequency band is 16KHz-24KHz and the high-frequency band audio signal of which the frequency band is 24KHz-32KHz corresponding to the audio signal are obtained. Through frequency division processing, the obtained multiple frequency division signals are marked as second audio signals.
In the embodiment shown in fig. 1, the audio coding and decoding method for realizing a high sampling rate of the present application includes: the process S102 is an encoding and decoding step, in which a standard encoding and decoding process is performed on the at least two second audio signals to obtain at least two third audio signals.
In this embodiment, a first signal with a certain bandwidth is divided by frequency division processing to obtain a plurality of second audio signals, where specific bandwidth information of the second audio signals is determined according to an actual frequency division process. In the coding and decoding step, a standard coder and decoder is used for coding and decoding the obtained second audio signal according to a standard coding and decoding process to obtain a corresponding third audio signal.
In a specific embodiment of the present application, in the codec step, at least two codecs corresponding to the number of the at least two second audio signals are used to perform a codec process on the at least two second audio signals.
In this embodiment, the number of codecs in the codec step is determined according to the number of second audio signals obtained by the frequency division processing. And then, standard encoding and decoding processes are carried out on each second audio signal, and the efficiency of the encoding and decoding processes is improved.
In a specific example of the present application, a first audio signal with a bandwidth of 32KHz is subjected to frequency division processing, and as a result, second audio signals with frequency bands of 20Hz to 16KHz, 16KHz to 24KHz, and 24KHz to 32KHz are obtained. When the encoding and decoding steps are carried out, three standard encoders are used for encoding and decoding the three second audio signals. Wherein the LC3 audio codec may be selected to codec the second audio signal.
In a specific embodiment of the present application, the high sampling rate is a sum of the sampling rates of the respective at least two codecs. The respective second audios are codec by using a plurality of codecs. In the overall coding and decoding process of the present application, the high sampling rate realized by the present application is the sum of the sampling rates of the respective codecs. For example, in the encoding and decoding step, two LC3 audio codecs with a sampling rate of 16KHz and a LC3 audio codec with a sampling rate of 32KHz are used to encode and decode the three second audio signals, and in the audio encoding and decoding method for realizing a high sampling rate of the present application, the high sampling rate is 16KHz +32KHz =64KHz, that is, the present application can realize a high sampling rate of 64 KHz.
In the actual coding and decoding steps, according to the actual audio frequency division result, audio codecs of different models and different sampling rates can be selected to be combined, and the coding and decoding process with high sampling rate is realized. For example, the type of audio codec may be an LC3 audio codec, an SBC audio codec, or other commonly used audio codecs. Regarding the selection of the sampling rate of the audio encoder, it can be set appropriately according to the desired target of the sampling rate. For example, if it is desired to achieve a sampling rate of 64KHz, if there are two second audio signals after the frequency division process, two LC3 audio codecs with a sampling rate of 32KHz may be set; if there are three second audio signals after the frequency division process, two LC3 audio codecs with a sampling rate of 16KHz and one LC3 audio codec with a sampling rate of 32KHz may be provided. The selection and combination mode of the specific codec can be reasonably designed according to the sampling rate requirement in the actual coding and decoding process.
In the embodiment shown in fig. 1, the audio coding and decoding method for realizing a high sampling rate of the present application includes: a process S103 of synthesizing at least two third audio signals to obtain a fourth audio signal.
In this embodiment, a plurality of third audio signals are obtained through the encoding and decoding step, and the plurality of third audio signals are synthesized to obtain a fourth audio signal. The fourth audio signal is equivalent to an audio signal obtained by encoding and decoding the first audio signal.
In a specific embodiment of the present application, in the synthesizing step, at least two third audio signals are synthesized by a quadrature mirror synthesis filter. And after the frequency division processing is carried out on the audio signals by the orthogonal mirror image analysis filter, the coding and decoding processing is carried out on the second audio signals after the frequency division to obtain corresponding third audio signals, and the plurality of third audio signals are synthesized by the orthogonal mirror image synthesis filter to obtain a fourth audio signal as a final coding and decoding result.
In one embodiment of the present application, the number of quadrature mirror synthesis filters is the same as the number of quadrature mirror analysis filters. And in the frequency division step, orthogonal mirror image analysis filters are selected for frequency division processing, in the synthesis step, the same number of orthogonal mirror image synthesis filters as the orthogonal mirror image analysis filters are selected for synthesis, and the positions of the orthogonal mirror image synthesis filters and the positions of the orthogonal mirror image analysis filters are in one-to-one correspondence, so that the finally obtained fourth audio signal and the result of directly carrying out the coding and decoding process on the first audio signal are the same type of audio signal.
In one example of the application, the first audio signal with the bandwidth of 32KHz is subjected to frequency division processing, and the result of the frequency division processing is that second audio signals with the frequency bands of 20Hz-16KHz, 16KHz-24KHz and 24KHz-32KHz are obtained. And obtaining third audio signals with corresponding frequency bands of 20Hz-16KHz, 16KHz-24KHz and 24KHz-32KHz through the encoding and decoding steps. In the synthesizing step, the orthogonal mirror image synthesizing filter is used for synthesizing the third audio signals with the frequency bands of 16KHz-24KHz and 24KHz-32KHz to obtain the audio signals with the frequency bands of 16KHz-32KHz, and then the signals with the frequency bands of 20Hz-16KHz and 16KHz-32KHz are synthesized to obtain the fourth audio signals with the frequency bands of 32KHz finally. That is, in the synthesis step, the quadrature mirror synthesis filter and the quadrature mirror analysis filter in the frequency division step are in one-to-one correspondence, and the audio signal to be subjected to frequency division needs to be correspondingly synthesized.
According to the audio coding and decoding method for realizing the high sampling rate, the frequency division processing is carried out on the first audio signals with certain bandwidth through the orthogonal mirror image analysis filter, and a plurality of second audio signals are obtained. And respectively carrying out coding and decoding processes on each second audio signal through a standard coder to obtain a third audio signal of a corresponding coding and decoding result. The coding and decoding process is participated in by the coder and the decoder, so that the sampling rate of the whole coding and decoding process is the sum of the sampling rates of the plurality of coders and the decoder, the sampling rate of the first audio signal coding and decoding process is further improved, the coding of the high sampling rate to the audio signal is realized, the tone quality of the audio signal after the coding and decoding process is further improved, and better use experience is realized for a user. The audio coding and decoding method for realizing the high sampling rate can realize the rational design of the high sampling process according to the actual codec model and the target sampling rate. In addition, the audio coding and decoding method for realizing the high sampling rate is further applied to various codecs including an LC3 audio codec and an SBC audio codec. The method is suitable for the audio coding and decoding process in the technical field of Bluetooth and is also suitable for other technical fields related to the audio coding and decoding process, and the tone quality of the audio is improved.
Fig. 3 shows an application example of the audio coding and decoding method for realizing high sampling rate.
As shown in FIG. 3, the input PCM audio signal has a bandwidth of 32KHz and the sampling rate for the PCM audio signal is 64 KHz. Since the maximum sampling rate of the LC3 audio codec is 48KHz, it is necessary to perform codec by using the audio codec method for realizing a high sampling rate of the present application. Firstly, the input PCM audio signal is subjected to frequency division processing through an orthogonal mirror image analysis filter, and a low-frequency PCM audio signal with the frequency band of 20Hz-16KHz and a high-frequency PCM audio signal with the frequency band of 16KHz-32KHz are respectively obtained. Subsequently, a low-band PCM audio signal with a frequency band of 20Hz-16KHz and a high-band PCM audio signal with a frequency band of 16KHz-32KHz are encoded at the Bluetooth transmitting end using an LC3 audio encoder with a sampling rate of 32 KHz. After the encoding is finished, decoding the audio signals at a Bluetooth receiving end by using an LC3 audio decoder with the sampling rate of 32KHz, synthesizing the two decoded audio signals by using an orthogonal mirror image synthesis filter after the decoding is finished, and finally outputting the final PCM audio signals. By carrying out frequency division processing on the input PCM audio signal and utilizing the LC3 codec to carry out the coding and decoding process, the sampling rate of the input PCM audio signal is the sum of the sampling rates of the two LC3 codecs, so that the sampling rate of the audio signal is improved, and the tone quality is improved. In quadrature mirror synthesis filters, as shown in figure 3
And
in a quadrature mirror analysis filter
And
the following relationship is satisfied:
fig. 4 shows an application example of the audio coding and decoding method for realizing high sampling rate.
As shown in FIG. 4, the input PCM audio signal has a bandwidth of 48KHz of audio signal and a sampling rate of 96KHz for the PCM audio signal. Because the maximum sampling rate of the LC3 audio codec is 48KHz, which cannot meet the requirement of the sampling rate, the audio codec method for realizing the high sampling rate of the present application needs to be used for the codec process. Firstly, the input PCM audio signal is subjected to frequency division processing through an orthogonal mirror image analysis filter, and a low-frequency PCM audio signal with the frequency band of 20Hz-24KHz and a high-frequency PCM audio signal with the frequency band of 24KHz-48KHz are respectively obtained. Then, carrying out frequency division processing on the low-frequency-band PCM audio signal with the frequency band of 20Hz-24KHz by utilizing orthogonal mirror image analysis filtering to obtain a frequency band 1 audio signal with 20Hz-12KHz and a frequency band 2 audio signal with 12KHz-24 KHz; and carrying out frequency division processing on the high-frequency PCM audio signal with the frequency band of 24KHz-48KHz by utilizing quadrature mirror image analysis filtering to obtain a frequency band 3 audio signal with 24KHz-36KHz and a frequency band 4 audio signal with 36KHz-48 KHz. The 4 LC3 audio codecs with 24KHz sampling rate are used to perform the codec process on the audio signals of the band 1, the band 2, the band 3 and the band 4. The last two quadrature mirror synthesis filters synthesize the decoding results corresponding to the audio signals of band 1 and band 2, and synthesize the decoding results corresponding to the audio signals of band 3 and band 4. The synthesized audio signal is then synthesized by a quadrature mirror synthesis filter to obtain a final output audio signal, and the specific process is shown in fig. 4.
In one example of the present application, with the audio codec method for implementing a high sampling rate of the present application, a higher sampling rate is implemented with a standard codec having a lower sampling rate. Higher sampling rates, including but not limited to 64KHz, 96KHz, 128KHz, and 192KHz, may be achieved with a coordinated arrangement of quadrature image analysis filters and quadrature image synthesis filters. For example, for realizing a sampling rate of 64KHz, the frequency division processing by the quadrature mirror analysis filter and the standard codec cooperation can be designed as a combination of 32KHz +32KHz, or a combination of 32KHz +16KHz +16 KHz; for realizing a sampling rate of 128KHz, a combination of 32KHz +32KHz +32KHz +32KHz can be designed. The low sampling rate of the existing coder and decoder is utilized, the frequency division processing of the input audio is carried out through the orthogonal mirror image analysis filter, then the coder and decoder are combined, and the high sampling rate in the coding and decoding process is further realized.
Fig. 5 shows an example of the application effect of the audio coding and decoding method for realizing a high sampling rate according to the present application.
Fig. 5 is a diagram illustrating an application of the audio coding and decoding method for realizing a high sampling rate according to the present application. The test audio is encoded by using an LC3 audio codec under the condition of the same code rate, wherein an upper curve in the graph is a subjective difference grade change curve applying the application, and a lower curve is a subjective difference grade change curve according to a standard encoding and decoding method. As shown in fig. 5, the subjective difference score of the present application is significantly higher than that of the standard LC3 encoding process, indicating that the sound quality using the method of the present application is significantly better than that of the standard LC3 codec flow.
Fig. 6 shows a specific embodiment of the audio codec system for implementing high sampling rate.
In this embodiment, the audio encoding and decoding system for realizing a high sampling rate of the present application includes: the frequency division module is used for carrying out frequency division on the first audio signal with a certain bandwidth to obtain at least two second audio signals, and the frequency bands of the at least two second audio signals are combined into a certain bandwidth; the encoding and decoding module is used for carrying out a standard encoding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and the synthesis module synthesizes the at least two third audio signals to obtain a fourth audio signal.
In one embodiment of the present application, an LC3 audio codec method includes: a frequency dividing step, which divides a first audio signal with a certain bandwidth to obtain at least two second audio signals, wherein the union of the frequency bands of the at least two second audio signals is the bandwidth of the first audio signal; coding and decoding, namely performing a standard coding and decoding process on at least two second audio signals by using a plurality of LC3 audio coders according to a preset sampling rate to obtain at least two third audio signals; and a synthesizing step of synthesizing the at least two third audio signals to obtain a fourth audio signal.
In this particular embodiment, the maximum sampling rate of the LC3 audio codec is 48 KHz. In order to improve the sampling rate of the LC3 audio codec and improve the sound quality of the codec audio, in the LC3 audio codec method of the present application, at least two second audio signals are obtained by performing frequency division processing on a first audio signal, then a plurality of LC3 codecs are used to perform a codec process on the second audio signals, and finally, the obtained codec results are combined to obtain a codec result of the first audio signal. Wherein the sampling rate of the coding and decoding process is the sum of the sampling rates of the LC3 audio coders, thereby realizing the coding and decoding process with high sampling rate.
In one example of the present application, the maximum sampling rate of the LC3 audio codec is 48 KHz. If the high sampling rate of 96KHz is realized, the first audio signal can be divided into two second audio signals, then two LC3 audio codecs with sampling rates of 48KHz are used for respectively carrying out coding and decoding processes on the two second audio coded signals, and then obtained coding and decoding results are combined. Thereby realizing a coding and decoding process of the first audio signal with a high sampling rate of 96 KHz. In the actual coding and decoding process, a proper LC3 audio coder with a certain sampling rate can be selected according to the actual coding and decoding sampling rate requirement for combination, and then different sampling rate requirements in the audio coding and decoding process are realized.
In a specific embodiment of the present application, a computer-readable storage medium stores computer instructions, wherein the computer instructions are operable to perform the audio codec method for realizing a high sampling rate described in any one of the embodiments. Wherein the storage medium 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.
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, a division of a unit is merely a logical division, and an actual implementation may have another division, 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.
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 embodiments are merely examples, which are not intended to limit the scope of the present disclosure, and all equivalent structural changes made by using the contents of the specification and the drawings, or any other related technical fields, are also included in the scope of the present disclosure.
Claims (10)
1. An audio encoding and decoding method for achieving a high sampling rate, comprising:
a frequency dividing step, which divides a first audio signal with a certain bandwidth to obtain at least two second audio signals, wherein the union of the frequency bands of the at least two second audio signals is the bandwidth of the first audio signal;
coding and decoding, namely performing a standard coding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and
and a synthesizing step of synthesizing the at least two third audio signals to obtain a fourth audio signal.
2. The audio coding and decoding method for achieving a high sampling rate according to claim 1, wherein in the dividing step, the first audio signal is divided by a quadrature image analysis filter.
3. The method as claimed in claim 2, wherein the number of the quadrature mirror analysis filters is 1 or more.
4. The audio coding and decoding method for achieving high sampling rate according to claim 1, wherein in the synthesizing step, the at least two third audio signals are synthesized by a quadrature mirror synthesis filter.
5. The method of claim 1, wherein the number of the quadrature mirror synthesis filters is the same as the number of the quadrature mirror analysis filters.
6. The audio codec method for realizing a high sampling rate as claimed in claim 1, wherein in the codec step, the codec flow is performed on the at least two second audio signals using at least two codecs corresponding to the number of the at least two second audio signals.
7. The method of claim 1, wherein the high sampling rate is a sum of sampling rates of the at least two codecs.
8. An audio coding/decoding system for achieving a high sampling rate, comprising:
the frequency dividing module is used for dividing frequency of a first audio signal with a certain bandwidth to obtain at least two second audio signals, and the union set of frequency bands of the at least two second audio signals is the certain bandwidth;
the coding and decoding module is used for carrying out a standard coding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and
and the synthesis module synthesizes the at least two third audio signals to obtain a fourth audio signal.
9. An LC3 audio codec coding and decoding method, comprising:
a frequency dividing step, which divides a first audio signal with a certain bandwidth to obtain at least two second audio signals, wherein the union of the frequency bands of the at least two second audio signals is the bandwidth of the first audio signal;
coding and decoding, namely performing a standard coding and decoding process on the at least two second audio signals by using a plurality of LC3 audio coders according to a preset sampling rate to obtain at least two third audio signals; and
and a synthesizing step of synthesizing the at least two third audio signals to obtain a fourth audio signal.
10. A computer-readable storage medium storing computer instructions, wherein the computer instructions are operable to perform the audio codec method for achieving a high sampling rate according to any one of claims 1 to 7.
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