CN111986685B - Audio encoding 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 coding and decoding method and a storage medium, and belongs to the technical field of audio coding and decoding. The audio encoding and decoding method for realizing high sampling rate comprises the following steps: dividing the frequency of the 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 a certain bandwidth; a coding and decoding step, namely performing standard coding and decoding flow on 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 can obtain high sampling rate in the audio encoding and decoding process, and improve the audio tone quality.
Description
Technical Field
The application relates to the technical field of audio coding and decoding, in particular to an audio coding and decoding method and system for realizing high sampling rate, an LC3 audio coder and decoder coding and decoding method and a storage medium.
Background
The bluetooth audio codec of the current mainstream includes: the SBC audio codec, which is mandatory by the A2DP protocol, is most widely used; the AAC-LC audio codec has good tone quality and wide application, and a plurality of mainstream mobile phones are supported; aptX series audio frequency coder-decoder, its tone quality is better, but the code rate is very high, and is the unique technology of the high pass, it is comparatively closed; the LDAC audio codec has better tone quality, but the code rate is also very high, and is a unique technology of Sony and is also very closed; the LHDC audio coder and decoder has better tone quality and higher coding rate. For the above reasons, the Bluetooth international union Bluetooth Sig has been introduced by a number of manufacturers in combination with LC3 audio codecs, which have the advantages of low delay, high sound quality and coding gain, and no patent fee in the Bluetooth field, and are paid attention to by the manufacturers.
With the improvement of life quality, people have higher and higher requirements on audio quality, and more bluetooth audio devices at middle and high ends have higher and higher requirements for supporting high-resolution audio. 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 when performing audio codec is 48KHz, and the maximum sampling rate of the LDAC audio codec and the LHDC audio codec when performing audio codec is 96KHz. The sampling rates of the LDAC audio codec and the LHDC audio codec can meet the sampling rate requirement of high-resolution audio (High Resolution Audio), while other codecs cannot meet the sampling rate requirement of high-resolution audio. Achieving high sampling rates in audio codec is critical to improving audio quality.
Disclosure of Invention
Aiming at the technical problems in the prior art, the application provides an audio encoding and decoding method, an audio encoding and decoding system, an LC3 audio encoding and decoding method and a storage medium for realizing high sampling rate.
In one aspect of the present application, there is provided an audio encoding and decoding method for realizing a high sampling rate, including: dividing the frequency of 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; a coding and decoding step, wherein standard coding and decoding processes are carried out 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, there is provided an audio codec system for realizing a high sampling rate, including: the frequency division module is used for dividing the frequency of the 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 the certain bandwidth; the encoding and decoding module performs a standard encoding and decoding process on the at least two second audio signals to obtain at least two third audio signals; and a synthesis module for synthesizing the at least two third audio signals to obtain a fourth audio signal.
In another aspect of the present application, there is provided a LC3 audio codec method including: dividing the frequency of the 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; a coding and decoding step, namely respectively carrying out standard coding and decoding processes on at least two second audio signals by using a plurality of LC3 audio coder-decoders 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 having computer instructions stored therein, wherein the computer instructions are operative to perform the audio codec method of scheme one that achieves a high sample rate.
The beneficial effects of the application are as follows: the high sampling rate sampling of the variable decoding audio is realized in the audio encoding and decoding process, and the tone quality of the encoding and decoding audio is improved.
Drawings
FIG. 1 is a flow chart of an embodiment of an audio codec method for implementing high sample rates according to the present application;
FIG. 2 is a schematic diagram illustrating the operation of the quadrature mirror analysis filter of the present application;
FIG. 3 is a flow chart of an example of an application of the audio codec method of the present application for implementing a high sampling rate;
FIG. 4 is a flow chart of an example of an application of the audio codec method of the present application for implementing a high sampling rate;
FIG. 5 is a schematic diagram of an example application effect of the audio codec method for implementing high sampling rate according to the present application;
Fig. 6 is a schematic diagram of an embodiment of an audio codec system implementing a high sampling rate according to the present application.
Detailed Description
For the purpose of making the objects, technical solutions and advantages of the embodiments of the present application more apparent, the technical solutions of the embodiments of the present application will be clearly and completely described below with reference to the accompanying drawings in the embodiments of the present application, and it is apparent that the described embodiments are some embodiments of the present application, but not all embodiments of the present application. All other embodiments, which can be made by those skilled in the art based on the embodiments of the application without making any inventive effort, are intended to be within the scope of the application.
The terms "first," "second," "third," "fourth" and the like in the description and in the claims and in the above drawings, if any, are used for distinguishing between similar objects and not necessarily for describing a particular sequential or chronological order. It is to be understood that the data so used may be interchanged where appropriate such that the embodiments of the application described herein may be implemented, for example, 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 an embodiment of the audio codec method of the present application for realizing a high sampling rate.
In the specific embodiment shown in fig. 1, the audio encoding and decoding method for realizing high sampling rate of the present application includes: s101, dividing the frequency of the 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 a certain bandwidth; s102, performing standard encoding and decoding processes on at least two second audio signals to obtain at least two third audio signals; and S103, synthesizing the at least two third audio signals to obtain a fourth audio signal.
In the specific embodiment shown in fig. 1, the audio encoding and decoding method for implementing a high sampling rate of the present application includes a step of dividing 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. Wherein for example, a first audio signal with the bandwidth of 32KHz is processed by frequency division to obtain two second audio signals with the frequency bands of 20Hz-16KHz and 16KHz-32 KHz. In addition, according to the actual coding and decoding requirements, the frequency division processing can be performed on the second audio signal again. For example, the second audio signal with the frequency band of 20Hz-16KHz is subjected to frequency division treatment, so that two audio signals of 20Hz-8KHz and 8KHz-16KHz can be obtained.
In one embodiment of the present application, in the step of dividing the frequency of S101, the first audio signal is subjected to frequency division processing by the quadrature image analysis filter.
In this particular embodiment, the quadrature image analysis filter is utilized to divide the first audio signal with a bandwidth. The audio signal with certain bandwidth is passed through the orthogonal mirror image analysis filter to obtain two new audio signals, in which the bandwidth of the audio signal obtained by frequency division is half of that of 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 through signal frequency division of the quadrature mirror analysis filter. Wherein, the audio signal of 20Hz-16KHz is called low-frequency band audio signal, and the audio signal of 16KHz-32KHz is called high-frequency band audio signal. The original audio signal can be divided into a low-band audio signal and a high-band audio signal having the same frequency bandwidth by one quadrature image analysis filter. The filtering characteristics of the low-pass and high-pass filters in the quadrature mirror analysis filter are as shown in fig. 2.
In one embodiment of the present application, the number of quadrature mirror analysis filters is 1 or more.
In this particular embodiment, multiple quadrature mirror analysis filters may be used to divide the audio signal according to the sampling rate requirements in the actual codec. After the audio signal is subjected to frequency division processing by using a quadrature image 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 having a certain bandwidth.
In one example of the application, the bandwidth of the first audio signal is, for example, 32KHz. The low-frequency band audio signal with the frequency band of 20Hz-16KHz and the high-frequency band audio signal with the frequency band of 16KHz-32KHz are obtained after the low-frequency band is passed through a quadrature mirror analysis filter. According to the specific sampling rate requirement of the encoding and decoding process, the quadrature mirror analysis filter can be used for dividing the frequency band of the low-frequency band audio signal of 20Hz-16KHz or the frequency band of the high-frequency band audio signal of 16KHz-32KHz again. The frequency division result of the low-frequency band audio signal of 20Hz-16KHz is to obtain the low-frequency band audio signal of 20Hz-8KHz corresponding to the audio signal, and the high-frequency band audio signal of 8KHz-16 KHz. If the high-frequency band audio signal of 16KHz-32KHz is divided, the low-frequency band audio signal of 16KHz-24KHz and the high-frequency band audio signal of 24KHz-32KHz corresponding to the audio signal are obtained. The plurality of divided signals obtained by the dividing process are noted as second audio signals.
In the specific embodiment shown in fig. 1, the audio encoding and decoding method for realizing high sampling rate of the present application includes: and S102, performing a standard encoding and decoding process on the at least two second audio signals to obtain at least two third audio signals.
In this embodiment, the first signal having a certain bandwidth is divided by the frequency division process, so as 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 encoding and decoding step, the obtained second audio signal is encoded and decoded by using a standard codec according to a standard encoding and decoding flow to obtain a corresponding third audio signal.
In a specific embodiment of the present application, in the encoding and decoding step, at least two second audio signals are subjected to a codec procedure using at least two codecs corresponding to the number of 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 coding and decoding processes are carried out on each second audio signal, so that the efficiency of the coding and decoding process is improved.
In one embodiment of the present application, the first audio signal with the bandwidth of 32KHz is used for frequency division processing, and the result is that the second audio signal with the frequency bands of 20Hz-16KHz, 16KHz-24KHz and 24KHz-32KHz is obtained. In the encoding and decoding step, the three second audio signals are encoded and decoded using three standard encoders. Wherein the LC3 audio codec may be selected for the codec of the second audio signal.
In a specific embodiment of the application, the high sampling rate is the sum of the sampling rates of the respective at least two codecs. Each of the second audio is encoded and decoded by using a plurality of codecs. In the overall codec process of the present application, the high sampling rate achieved by the present application is the sum of the sampling rates of the individual codecs. For example, in the encoding and decoding step, two LC3 audio codecs with a sampling rate of 16KHz and an LC3 audio codec with a sampling rate of 32KHz are used to encode and decode 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+16khz+32khz=64 KHz, that is, the present application can realize a high sampling rate of 64 KHz.
In the actual encoding and decoding steps, according to the actual audio frequency division result, audio frequency codecs with different models and different sampling rates can be selected for combination, and the encoding 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 sampling rate selection of the audio encoder, reasonable settings may be made according to the desired sampling rate target. For example, if it is desired to achieve a sampling rate of 64KHz, if there are two second audio signals after the crossover process, two LC3 audio codecs with a sampling rate of 32KHz may be set; if there are three second audio signals after the crossover processing, two LC3 audio codecs with a sampling rate of 16KHz and one LC3 audio codec with a sampling rate of 32KHz can be set. The selection and combination modes of specific codecs can be reasonably designed according to the sampling rate requirements in the actual codec process.
In the specific embodiment shown in fig. 1, the audio encoding and decoding method for realizing high sampling rate of the present application includes: and step S103, synthesizing the at least two third audio signals to obtain a fourth audio signal.
In this embodiment, a plurality of corresponding third audio signals are obtained through the encoding and decoding steps, and the plurality of third audio signals are synthesized to obtain a fourth audio signal. The fourth audio signal corresponds to an audio signal obtained by encoding and decoding the first audio signal.
In a specific embodiment of the application, in the synthesizing step, at least two third audio signals are synthesized by means of orthogonal mirror synthesis filters. After the audio signal is subjected to frequency division processing through the orthogonal image analysis filter, the frequency-divided second audio signal is subjected to coding and decoding processing to obtain a corresponding third audio signal, and a plurality of third audio signals are synthesized through the orthogonal image synthesis filter to obtain a fourth audio signal serving 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. In the step of synthesis, orthogonal image synthesis filters with the same number as the orthogonal image analysis filters are selected for synthesis, and the positions of the orthogonal image synthesis filters and the orthogonal image analysis filters are in one-to-one correspondence, so that the obtained fourth audio signal and the first audio signal are ensured to be the same type of audio signal as the result of the direct encoding and decoding process.
In one example of the present application, the frequency division process is performed with a first audio signal having a bandwidth of 32KHz to obtain a second audio signal having a frequency band of 20Hz-16KHz, 16KHz-24KHz, and 24KHz-32 KHz. The third audio signals with the corresponding frequency bands of 20Hz-16KHz, 16KHz-24KHz and 24KHz-32KHz are obtained through the encoding and decoding steps. And when the synthesizing step is carried out, the third audio signals with the frequency bands of 16KHz-24KHz and 24KHz-32KHz are synthesized by using an orthogonal mirror image synthesis filter 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 signal with the final frequency band of 32 KHz. In the synthesizing step, the orthogonal image synthesizing filter is in one-to-one correspondence with the orthogonal image analyzing filter in the frequency dividing step, and the corresponding synthesis is needed for the frequency-divided audio signals.
The application realizes the audio encoding and decoding method with high sampling rate, and carries out frequency division processing on a first audio signal with a certain bandwidth through the orthogonal mirror image analysis filter to obtain a plurality of second audio signals. And respectively carrying out encoding and decoding processes on each second audio signal through a standard encoder to obtain a third audio signal of a corresponding encoding and decoding result. The encoding and decoding processes are participated in by the encoding and decoding devices, so that the sampling rate of the whole encoding and decoding process is the sum of the sampling rates of a plurality of encoding and decoding devices, the sampling rate of the first audio signal encoding and decoding process is improved, the encoding of the audio signal with high sampling rate is realized, the tone quality of the audio signal after the encoding and decoding processes is improved, and better use experience is provided for users. The audio coding and decoding method for realizing high sampling rate can realize reasonable design of high sampling process according to the actual model of the coder and decoder and the target sampling rate. In addition, the audio codec method for realizing high sampling rate of the present application is further applied to various codecs including LC3 audio codec and SBC audio codec. The method is suitable for the audio encoding and decoding process in the Bluetooth technical field, is also suitable for other technical fields related to the audio encoding and decoding process, and improves the tone quality of audio.
Fig. 3 shows an example of an application of the audio codec method of the present application for realizing a high sampling rate.
As shown in fig. 3, an audio signal with a bandwidth of 32KHz of the PCM audio signal is input, and the sampling rate of the PCM audio signal is 64KHz. Because the maximum sampling rate of the LC3 audio codec is 48KHz, it is necessary to perform the codec using the audio codec method of the present application for realizing a high sampling rate. Firstly, the input PCM audio signal is subjected to frequency division processing through a quadrature mirror analysis filter to respectively obtain a low-frequency band PCM audio signal with the frequency band of 20Hz-16KHz and a high-frequency band PCM audio signal with the frequency band of 16KHz-32 KHz. Subsequently, the LC3 audio encoder with the sampling rate of 32KHz is utilized at the Bluetooth transmitting end to encode the low-frequency band PCM audio signal with the frequency band of 20Hz-16KHz and the high-frequency band PCM audio signal with the frequency band of 16KHz-32 KHz. After the encoding is finished, at the Bluetooth receiving end, an LC3 audio decoder with the sampling rate of 32KHz is utilized to decode the audio signals, and after the decoding is finished, an orthogonal mirror image synthesis filter is utilized to synthesize the two decoded audio signals, and finally the final PCM audio signals are output. Through carrying out frequency division processing on the input PCM audio signal, utilizing the LC3 codec to carry out the coding and decoding process, the sampling rate of the input PCM audio signal is the sum value of the sampling rates of the two LC3 codecs, and then the sampling rate of the audio signal is improved, and the tone quality is improved. As shown in fig. 3, in the quadrature mirror synthesis filterAnd/>Which is identical to/>, in a quadrature mirror analysis filterAnd/>The following relationship is satisfied:
fig. 4 shows an example of an application of the audio codec method of the present application for realizing a high sampling rate.
As shown in fig. 4, an audio signal with a bandwidth of 48KHz of the PCM audio signal is input, and the sampling rate of the PCM audio signal is 96KHz. Because the maximum sampling rate of the LC3 audio codec is 48KHz, the sampling rate requirement cannot be met, and therefore, the audio codec method for realizing high sampling rate needs to be utilized for performing the codec process. Firstly, the input PCM audio signal is subjected to frequency division processing through a quadrature mirror analysis filter to respectively obtain a low-frequency band PCM audio signal with the frequency band of 20Hz-24KHz and a high-frequency band PCM audio signal with the frequency band of 24KHz-48 KHz. Then, the quadrature mirror analysis filtering is utilized to carry out frequency division processing on the low-frequency band PCM audio signal with the frequency band of 20Hz-24KHz to obtain a frequency band 1 audio signal with the frequency band of 20Hz-12KHz and a frequency band 2 audio signal with the frequency band of 12KHz-24 KHz; the orthogonal image analysis filtering is utilized to carry out frequency division processing on the high-frequency band PCM audio signal with the frequency band of 24KHz-48KHz, thus obtaining the band 3 audio signal with the frequency band of 24KHz-36KHz and the band 4 audio signal with the frequency band of 36KHz-48 KHz. The audio signals of the frequency band 1, the frequency band 2, the frequency band 3 and the frequency band 4 are respectively encoded and decoded by using 4 LC3 audio codecs with 24KHz sampling rate. And finally, the two orthogonal image synthesis filters synthesize decoding results corresponding to the audio signals of the frequency band 1 and the frequency band 2, and synthesize decoding results corresponding to the audio signals of the frequency band 3 and the frequency band 4. The synthesized audio signal is 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, a higher sampling rate is achieved with the standard codec with a lower sampling rate by the audio codec method of the present application that achieves a high sampling rate. Higher sample rates including, but not limited to, 64KHz, 96KHz, 128KHz, and 192KHz may be achieved with a cooperating arrangement of quadrature image analysis filters and quadrature image synthesis filters. For example, for a sample rate of 64KHz, the divide by quadrature mirror analysis filter processing and standard codec coordination may be designed as a 32khz+32khz combination, or a 32khz+16khz+16khz combination; for a sample rate of 128KHz, a combination of 32KHz+32KHz+32KHz+32KHz may be designed. The low sampling rate in the existing codec is utilized, the frequency division processing of the input audio is carried out through the orthogonal mirror image analysis filter, and then the codec is combined, so that the high sampling rate in the process of encoding and decoding is realized.
Fig. 5 shows an example of the application effect of the audio codec method of the present application for realizing a high sampling rate.
Fig. 5 is a diagram showing an application of the audio codec method for realizing the high sampling rate according to the present application. The LC3 audio coder is used for coding the test audio under the condition of the same code rate, wherein the upper curve in the figure is the subjective difference grade change curve applying the application, and the lower curve is the subjective difference grade change curve according to the standard coding and decoding method. As shown in fig. 5, the subjective difference score of the present application is significantly higher than the standard LC3 encoding process, indicating that the sound quality using the method of the present application is significantly better than the standard LC3 codec flow.
Fig. 6 illustrates one embodiment of an audio codec system implementing high sample rates in accordance with the present application.
In this embodiment, the audio codec system implementing a high sampling rate of the present application includes: the frequency division module is used for dividing the frequency of the 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; the encoding and decoding module performs a standard encoding and decoding process on at least two second audio signals to obtain at least two third audio signals; and a synthesis module for synthesizing 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: dividing the frequency of the 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; a coding and decoding step, namely respectively carrying out standard coding and decoding processes on at least two second audio signals by using a plurality of LC3 audio coder-decoders 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 highest sample rate of the LC3 audio codec is 48KHz. In order to improve the sampling rate of the LC3 audio codec and improve the tone quality of the encoded and decoded audio, in the method for encoding and decoding the LC3 audio codec, at least two second audio signals are obtained by carrying out frequency division processing on a first audio signal, a plurality of LC3 codecs are utilized to respectively encode and decode the second audio signals, and finally the obtained encoding and decoding results are combined to obtain the encoding and decoding results of the first audio signal. The sampling rate of the encoding and decoding process is the sum of the sampling rates of a plurality of LC3 audio codecs, so that the encoding and decoding process with high sampling rate is realized.
In one example of the application, the highest sample rate of the LC3 audio codec is 48KHz. If the high sampling rate of 96KHz is realized, the first audio signal can be divided into two second audio signals, then the two second audio coding signals are respectively encoded and decoded by using two LC3 audio codecs with the sampling rate of 48KHz, and the obtained encoding and decoding results are combined. Thereby implementing a codec process for the first audio signal at a high sampling rate of 96 KHz. In the actual encoding and decoding process, a suitable LC3 audio encoder and decoder with a certain sampling rate can be selected according to the requirement of the actual encoding and decoding sampling rate to be combined, so that the requirements of different sampling rates in the audio encoding and decoding process are further realized.
In one embodiment of the application, a computer readable storage medium stores computer instructions operable to perform the audio codec method described in any of the embodiments that achieves a high sample rate. 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 (English: central Processing Unit, CPU for short), other general purpose Processor, digital signal Processor (English: DIGITAL SIGNAL Processor, DSP for short), application specific integrated Circuit (Application SPECIFIC INTEGRATED Circuit, ASIC for short), field programmable gate array (English: field Programmable GATE ARRAY, FPGA for short), or other programmable logic device, 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 manners. For example, the apparatus embodiments described above are merely illustrative, e.g., the division of elements is merely a logical functional division, and there may be additional divisions of actual implementation, e.g., multiple elements or components may be combined or integrated into another system, or some features may be omitted, or not performed. Alternatively, the coupling or direct coupling or communication connection shown or discussed with each other may be an indirect coupling or communication connection via some interfaces, devices or units, which may be in electrical, mechanical or other form.
The units described as separate units may or may not be physically separate, and units shown as units may or may not be physical units, may be located in one place, or may be distributed over a plurality of network units. Some or all of the units may be selected according to actual needs to achieve the purpose of the solution of this embodiment.
The foregoing is only illustrative of the present application and is not to be construed as limiting the scope of the application, and all equivalent structural changes made by the present application and the accompanying drawings, or direct or indirect application in other related technical fields, are included in the scope of the present application.
Claims (6)
1. An audio codec method for realizing a high sampling rate, comprising:
Dividing the frequency of a first audio signal with a certain bandwidth through orthogonal image analysis filters 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, and the number of the orthogonal image analysis filters is more than or equal to 1;
a coding and decoding step, wherein standard coding and decoding processes are carried out on the at least two second audio signals to obtain at least two third audio signals; and
And synthesizing the at least two third audio signals through orthogonal image synthesis filters to obtain a fourth audio signal, wherein the number of the orthogonal image synthesis filters is the same as that of the orthogonal image analysis filters, and the positions of the orthogonal image synthesis filters and the orthogonal image analysis filters are in one-to-one correspondence.
2. The audio codec method according to 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 at least two second audio signal numbers.
3. The audio codec method of claim 1, wherein the high sampling rate is a sum of sampling rates of the respective at least two codecs.
4. An audio codec system for achieving a high sampling rate, comprising:
The frequency division module is used for dividing the frequency of the first audio signal with a certain bandwidth through the orthogonal image analysis filter to obtain at least two second audio signals, the union of the frequency bands of the at least two second audio signals is the certain bandwidth, and the number of the orthogonal image analysis filters is more than or equal to 1;
the encoding and decoding module performs a standard encoding 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 through orthogonal image synthesis filters to obtain a fourth audio signal, wherein the number of the orthogonal image synthesis filters is the same as that of the orthogonal image analysis filters, and the positions of the orthogonal image synthesis filters and the positions of the orthogonal image analysis filters are in one-to-one correspondence.
5. A method of LC3 audio codec, comprising:
Dividing the frequency of a first audio signal with a certain bandwidth through orthogonal image analysis filters 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, and the number of the orthogonal image analysis filters is more than or equal to 1;
A coding and decoding step, namely respectively carrying out standard coding and decoding processes on the at least two second audio signals by using a plurality of LC3 audio coder-decoders according to a preset sampling rate to obtain at least two third audio signals; and
And synthesizing the at least two third audio signals through orthogonal image synthesis filters to obtain a fourth audio signal, wherein the number of the orthogonal image synthesis filters is the same as that of the orthogonal image analysis filters, and the positions of the orthogonal image synthesis filters and the orthogonal image analysis filters are in one-to-one correspondence.
6. A computer readable storage medium storing computer instructions, wherein the computer instructions are operative to perform the audio codec method of any one of claims 1-3 that achieves a high sampling rate.
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