CN113259710A - Method, system, and medium for improving audio transmission stuck - Google Patents
Method, system, and medium for improving audio transmission stuck Download PDFInfo
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Abstract
The application discloses a method, a system and a medium for improving audio transmission jamming, and belongs to the technical field of Bluetooth communication. The method comprises the following steps: a sending end encodes an audio frame to be sent by utilizing various encoding code rates to obtain corresponding various data frames; and the sending end selects the data frame with the corresponding code rate to send to the receiving end according to the transmission condition of the audio frame. According to the method for improving the audio transmission blocking, the audio frames of the sending end are coded by using various coding code rates, and the data frames obtained by coding with different coding code rates are respectively transmitted according to the transmission condition of the audio frames, so that the receiving effect of the receiving end on the audio data is ensured when the communication environment is poor, the loss of the audio data in the transmission process is reduced, the blocking phenomenon of audio transmission is avoided, and the user experience is improved.
Description
Technical Field
The present application relates to the field of bluetooth communications technologies, and in particular, to a method, a system, and a medium for improving audio transmission stuck.
Background
In the latest bluetooth Low Energy Audio (LE Audio) specification, a point-to-point synchronous stream transmission link technology (CIS) is introduced to implement Low-latency Audio transmission. When a CIS link is established, corresponding Quality of Service (QoS) parameters are configured, where the QoS parameters include a Timeout (FT), a synchronous link time Interval (ISO Interval), a maximum Number of sub-events (NSE) in a time Interval, and a Number of packets (Burst Number, BN) allowed to be sent in a time Interval, and these parameters determine a maximum retransmission Number of each packet, and cannot be modified in a transmission process, but a wireless environment of bluetooth is complicated and variable, for example, when music is listened in a public place, CIS link Quality deteriorates due to complicated interference of the wireless environment, bluetooth transmission fails, and a lost packet needs to be retransmitted at this time. When a certain data packet reaches the retransmission times and cannot be correctly received by the receiving end, the data packet (including the audio packet) is discarded, so that the katon phenomenon of the receiving end is caused.
In the prior art, aiming at the problem of blockage during the transmission of Bluetooth audio data, the invention provides: "data transmission method, apparatus, device, system, and medium", a solution is mentioned in application No. CN 202080001621.5. The method comprises the steps of detecting the packet loss rate of a data packet at a receiving end of data, and sending indication information to a transmitting end when the packet loss rate is larger than a set threshold. The transmitting terminal reduces the audio coding rate of the transmitting terminal according to the indication information, and the audio frames are reduced by reducing the coding rate, so that one data packet can contain more same audio frames, the retransmission times of the audio data are increased, and the frame loss probability is reduced. The method can only reduce the code rate after the transmitting end receives the indication information, avoids the jamming, but cannot solve the problem of jamming caused by the loss of the data packet before the receiving end sends the indication information.
Disclosure of Invention
The method, the system and the medium for improving the audio transmission jam are provided by the application, aiming at the problems that in the prior art, when Bluetooth audio data is transmitted, audio jamming exists because of the loss of a data packet, and the problem that the audio jamming cannot be solved in the whole time period due to the fact that the existing technical means is insufficient when the audio jamming is solved.
In one aspect of the present application, a method for improving audio transmission stuck is provided, including: the method comprises the steps that a sending end encodes an audio frame to be sent by utilizing multiple encoding code rates to obtain corresponding multiple data frames, wherein the multiple encoding code rates comprise a first encoding code rate to an Nth encoding code rate, N is an integer larger than 2, and the value of N is determined according to audio encoding requirements, wherein the first encoding code rate is larger than an ith encoding code rate, the ith encoding code rate is larger than an ith +1 encoding code rate, and i is an integer larger than 1 and smaller than N; the sending end selects a data frame with a corresponding code rate to send to the receiving end according to the transmission condition of the audio frame, and if the audio frame is transmitted for the first time, the sending end sends a first data frame obtained by coding with a first coding code rate to the receiving end; and if the audio frame is not transmitted for the first time, the transmitting end transmits an ith data frame obtained by encoding with an ith encoding rate or an Nth data frame obtained by encoding with an Nth encoding rate to the receiving end, wherein the transmitting times of the audio frame correspond to i or N.
Optionally, if the audio frame is not transmitted for the first time, the transmitting end sends the corresponding ith data frame obtained by encoding at the ith encoding rate or the nth data frame obtained by encoding at the nth encoding rate to the receiving end, including: analyzing and judging the current sending times of the audio frame; and selecting the ith data frame obtained by coding with the ith coding rate or the Nth data frame obtained by coding with the Nth coding rate according to the sending times, and sending the ith data frame or the Nth data frame obtained by coding with the Nth coding rate to the sending end, wherein the number of data frames in a data packet corresponding to the (i + 1) th data frame is more than that of the data packet corresponding to the ith data frame.
Optionally, the sending end reduces the first coding rate, the ith coding rate and/or the nth coding rate according to the first indication information returned by the receiving end to obtain a corresponding first update coding rate, the ith update coding rate and/or the nth update coding rate; or the sending end raises the first updating code rate, the ith updating code rate and/or the Nth updating code rate according to the second indication information returned by the receiving end to obtain the corresponding first code rate, the ith code rate and/or the Nth code rate.
Optionally, the receiving end detects the number of times of data packet loss within a preset time range; if the loss times are larger than a preset threshold value, sending first indication information; and if the loss times are not more than the preset threshold value, sending out second indication information.
In one aspect of the present application, a system for improving audio transmission stuck is provided, including: the audio frame transmitting method comprises an encoding module, a transmitting module and a receiving module, wherein the encoding module encodes an audio frame to be transmitted by utilizing a plurality of encoding code rates to obtain a plurality of corresponding data frames, wherein the plurality of encoding code rates comprise a first encoding code rate to an Nth encoding code rate, N is an integer greater than 2, and the value of N is determined according to the audio encoding requirement, wherein the first encoding code rate is greater than an ith encoding code rate, the ith encoding code rate is greater than an (i + 1) th encoding code rate, and i is an integer greater than 1 and less than N; the sending module selects a data frame with a corresponding code rate to send to the receiving module according to the transmission condition of the audio frame, wherein if the audio frame is transmitted for the first time, the sending end sends a first data frame obtained by coding with a first coding code rate to the receiving end; and if the audio frame is not transmitted for the first time, the transmitting end transmits the corresponding ith data frame obtained by encoding with the ith encoding rate or the Nth data frame obtained by encoding with the Nth encoding rate to the receiving end, wherein the transmitting times of the audio frame correspond to i or N.
In one aspect of the present application, a computer-readable storage medium is provided, wherein the storage medium stores computer instructions, and the computer instructions are operated to execute the method in the first aspect.
The beneficial effect of this application is: according to the method for improving the audio transmission stuck, the first audio frame of the sending end is coded by using multiple coding code rates, and the first data frames under different coding code rates are respectively transmitted according to the transmission condition of the first audio frame, so that the audio data are guaranteed to be received at the receiving end, the loss of the audio data in the transmission process is reduced, the stuck phenomenon of audio transmission is avoided, and the user experience is improved.
Drawings
In order to more clearly illustrate the embodiments of the present application or the technical solutions in the prior art, the drawings needed to be used in the description of the embodiments or the prior art will be briefly introduced below, and it is obvious that the drawings in the following description are some embodiments of the present application, and for those skilled in the art, other drawings can be obtained according to these drawings without inventive exercise.
FIG. 1 is a schematic diagram of a Bluetooth audio data transmission process;
FIG. 2 is a schematic flow chart diagram illustrating one embodiment of a method for improving audio transmission stuck according to the present application;
FIG. 3 is a diagram of an example of an audio encoding process of the present application;
FIG. 4 is a schematic flow chart diagram illustrating one embodiment of a method for improving audio transmission stuck;
FIG. 5 is a schematic flow diagram of one example of a method of improving audio transmission stuck according to the present application;
fig. 6 illustrates one embodiment of the present system for improving audio transmission stuck.
With the above figures, there are shown specific embodiments of the present application, which will be described in more detail below. These drawings and written description are not intended to limit the scope of the inventive concepts in any manner, but rather to illustrate the inventive concepts to those skilled in the art by reference to specific embodiments.
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" and "comprising," and any variations thereof, are intended to cover a non-exclusive inclusion, such that a product or apparatus that comprises a list of steps or elements is not necessarily limited to those elements explicitly listed, but may include other elements not expressly listed or inherent to such product or apparatus.
In the prior art, in the latest bluetooth Low Energy Audio (LE Audio for short) specification, a point-to-point synchronous stream transmission link technology (CIS for short) is introduced to implement Low-delay Audio transmission. When a CIS link is established, corresponding Quality of Service (QoS) parameters are configured, where the QoS parameters include a Timeout (FT), a synchronous link time Interval (ISO Interval), a maximum Number of sub-events (NSE) in a time Interval, and a Number of packets (Burst Number, BN) allowed to be sent in a time Interval, and these parameters determine a maximum retransmission Number of each packet, and cannot be modified in a transmission process, but a wireless environment of bluetooth is complicated and variable, for example, when music is listened in a public place, CIS link Quality deteriorates due to complicated interference of the wireless environment, bluetooth transmission fails, and a lost packet needs to be retransmitted at this time. When a certain data packet reaches the retransmission times and cannot be correctly received by the receiving end, the data packet (including the audio packet) is discarded, so that the katon phenomenon of the receiving end is caused.
Fig. 1 is a schematic diagram of a bluetooth audio data transmission process.
As shown in fig. 1, when the bluetooth receiving end detects that a data packet is correct, an ACK signal is returned to the bluetooth transmitting end to indicate that the correct data packet is received; when the Bluetooth receiving end detects a data packet error, a NACK signal is returned to the Bluetooth transmitting end to indicate that a correct data packet is not received; after receiving the ACK signal, the transmitting end sends the next data packet according to the set time sequence; after receiving the NACK signal, the transmitting end retransmits the data packet that failed in transmission, and the number of retransmissions is determined according to the system design. If the transmitting end does not receive the ACK signal or the NACK signal within the specified time, the transmitting end retransmits the corresponding data packet at the moment after the time threshold is exceeded. When a certain data packet reaches the retransmission times and cannot be correctly received by the receiving end, it means that the data packet containing the audio frame data is discarded, thereby causing a pause phenomenon at the receiving end.
The invention conception of the application is as follows: at the audio data sending end, after receiving the NACK signal or the timeout signal returned by the receiving end, the sending end retransmits the corresponding audio frame data until the upper limit of the number of retransmissions is satisfied. When the receiving end does not successfully receive the data packet, the transmitting end retransmits the corresponding data packet, the transmitting end directly uses various audio coding code rates to respectively code audio frame data to be transmitted, and when the audio frame data are transmitted for the first time, the data frame data coded by the audio frame data are transmitted by using a higher coding code rate; when the audio frame data is not correctly received at the receiving end and the transmitting end needs to retransmit the audio frame data, it indicates that there is a possibility that the currently transmitted data packet may be lost. In order to avoid the loss of the transmitted data packet, when the sending end retransmits the audio frame data, the data frame data after the audio frame data are coded by using a lower coding rate for transmission, because the coding rate is reduced, the data packet sent by the sending end contains a plurality of same audio frame data, which is equivalent to increasing the retransmission times, thereby reducing the frame loss probability. In addition, along with the increase of the retransmission times, the frame loss probability is higher and higher in the transmission process, so that when the audio frame data is retransmitted, the data frame data coded by the audio frame data is transmitted by selecting a lower coding rate along with the increase of the retransmission times, the data loss is avoided, and the blockage problem is avoided.
Aiming at the technical problems in the prior art, the application provides a method for improving the audio transmission blockage. Firstly, a sending end utilizes various coding code rates to code an audio frame to be sent, and obtains corresponding various data frames. The method comprises the steps of coding an audio frame to be sent according to a plurality of preset coding rates to obtain a plurality of corresponding data frames, wherein the plurality of coding rates comprise a first coding rate to an Nth coding rate, N is an integer larger than 2, and the value of N is determined according to audio coding requirements, wherein the first coding rate is larger than an ith coding rate, the ith coding rate is larger than an ith +1 coding rate, and i is an integer larger than 1 and smaller than N. Then, the sending end selects the data frame with the corresponding code rate to send to the receiving end according to the transmission condition of the audio frame. If the audio frame is transmitted for the first time, sending a first data frame corresponding to the first coding rate to a receiving end; and if the audio frame is not transmitted for the first time, sending the ith data frame obtained by coding with the ith coding rate or the Nth data frame obtained by coding with the Nth coding rate to a receiving end. In the process, the sending times of the current sending of the audio frame is analyzed and judged, and then the corresponding ith data frame obtained through coding with the ith coding rate or the Nth data frame obtained through coding with the Nth coding rate is selected according to the sending times to be sent to the sending end, wherein the number of data frames in a data packet corresponding to the (i + 1) th data frame is more than that of data frames in a data packet corresponding to the ith data frame.
According to the method for improving the audio transmission stuck, the first audio frame of the sending end is coded by using multiple coding code rates, and the first data frames under different coding code rates are respectively transmitted according to the transmission condition of the first audio frame, so that the audio data are guaranteed to be received at the receiving end, the loss of the audio data in the transmission process is reduced, the stuck phenomenon of audio transmission is avoided, and the user experience is improved.
The following describes the technical solutions of the present application and how to solve the above technical problems with specific embodiments. The following several specific embodiments may be combined with each other, and details of the same or similar concepts or processes may not be repeated in some embodiments. Embodiments of the present application will be described below with reference to the accompanying drawings.
FIG. 2 is a schematic flow chart diagram illustrating one embodiment of a method for improving audio transmission stuck.
In the embodiment shown in fig. 2, the method for improving audio transmission blockage of the present application includes a process S201, in which a sending end encodes an audio frame to be sent by using multiple encoding code rates to obtain multiple corresponding data frames.
In this embodiment, when a transmitting end of audio transmission performs encoding transmission on an audio frame, the transmitting end first encodes the audio frame using multiple code rates with different sizes, and correspondingly obtains multiple encoded data frames. And then, selecting the data frame with the corresponding code rate to transmit according to different transmission conditions.
Optionally, the sending end encodes the data frame to be sent by using multiple coding rates to obtain multiple corresponding data frames, including: the method comprises the steps that a sending end encodes an audio frame according to multiple preset encoding rates to obtain multiple corresponding data frames, wherein the multiple encoding rates comprise a first encoding rate to an Nth encoding rate, N is an integer larger than 2, and the value of N is determined according to audio encoding requirements, wherein the first encoding rate is larger than an ith encoding rate, the ith encoding rate is larger than an ith +1 encoding rate, and i is an integer larger than 1 and smaller than N.
In this embodiment, in the sending end device, multiple coding rates are preset in the sending end device by adding corresponding control codes, and the audio frames to be sent are respectively coded, so as to obtain multiple corresponding data frames. The multiple coding rates include a first coding rate to an Nth coding rate. The specific value of N can be determined according to actual audio coding requirements, the first coding rate is greater than the ith coding rate, the ith coding rate is greater than the (i + 1) th coding rate, and i is an integer greater than 1 and less than N.
Specifically, the first coding rate may be set as a standard coding rate in the sending end device. In multiple coding rates of the sending end device, for example, when the value of N is 3, the multiple coding rates include a first coding rate, a second coding rate, and a third coding rate, and correspondingly, the first coding rate is greater than the second coding rate, and the second coding rate is greater than the third coding rate; when the value of N is 4, the multiple coding rates include a first coding rate, a second coding rate, a third coding rate, and a fourth coding rate, and correspondingly, the first coding rate is greater than the second coding rate, the second coding rate is greater than the third coding rate, and the third coding rate is greater than the fourth coding rate.
It should be noted that the value of N can be set reasonably according to the coding requirement of the sending end device or the empirical evaluation of the audio frequency stuck phenomenon; similarly, the specific values of various coding rates can be determined according to the actual coding requirements. The present application is not particularly limited with respect to the selection of specific N and the specific setting of the code rate values for the various coding rates.
Optionally, the sending end encodes the audio frame according to multiple preset encoding rates, including: the method comprises the steps of coding an audio frame according to a preset coding process and multiple coding code rates, wherein the preset coding process comprises a low-complexity coding process and a high-sound-quality coding process, and in the low-complexity coding process, a standard coding process is carried out on the audio frame according to a first coding code rate to obtain a corresponding data frame; and coding the audio frame according to the ith coding rate or the Nth coding rate, deleting high-frequency spectral coefficient data, residual error data and/or LSB data in the coding process to obtain a corresponding data frame, wherein the LSB is fully called as a Least Significant Bit and represents the lowest Significant Bit, namely the lowest Bit of the quantized spectral coefficient, the contribution to the whole tone quality is small, the code rate can be reduced by discarding the LSB, and the loss to the tone quality is small. In the high-sound-quality coding process, the audio frame is coded according to the first coding code rate or the Nth coding code rate by sharing the corresponding coding module to obtain the corresponding data frame, wherein the coding module comprises a bandwidth detection module and/or a time domain noise shaping module.
In this optional embodiment, after encoding an input frame of audio frames, a plurality of data frames corresponding to different encoding rates are output, which may be implemented by an encoding method including a low-complexity encoding process and a high-quality encoding process. When the audio frame is coded by using the first coding rate in the low-complexity coding process, the audio frame can be coded according to a standard coding process, and when the audio frame is coded by using the ith coding rate or the Nth coding rate, particularly when the spectral coefficients are subjected to arithmetic coding and residual coding, because the code rate is low, the available bit number is insufficient, so that part of high-frequency spectral coefficients and all residual data or LSB data are removed in the coding process; or when the bit number is enough to carry out the spectral coefficient coding, part of residual data or LSB data is removed, and the complexity of coding and the operation cost are reduced through reducing the data quantity.
In the optional embodiment, in the high sound quality encoding process, when the audio frame is encoded according to the first encoding code rate, the ith encoding code rate or the nth encoding code rate, the audio frame is encoded according to the standard encoding process, and most encoding modules are shared in the encoding process with different code rates. For example, in the LC3 audio encoder, a low-delay modified discrete cosine transform module, a bandwidth detection module, an impulse detection module, a transform domain noise shaping and time domain noise shaping module, and the like are shared. In different audio encoders, the common encoding module is correspondingly replaced according to different structures of the audio encoders. In addition, in the high-sound-quality coding process, in the coding process of different coding code rates, the different coding code rates correspond to a set of pedigree quantization, noise level detection, arithmetic coding and residual coding process, and compared with the low-complexity coding process, the process has higher complexity and computing power, but the sound quality can be ensured, and still has lower computing power loss compared with the condition that a plurality of data frames corresponding to coding are coded by using the corresponding coding code rates through different encoders.
In addition, the audio frame can be encoded by using the first encoding code rate, the ith encoding code rate or the Nth encoding code rate through a standard encoding method, so as to obtain the corresponding data frame. For example, when the value of N is 3, the audio frame may be encoded according to the standard encoding process by using three standard encoders and using the corresponding first encoding rate, second encoding rate, and third encoding rate, respectively, to obtain the corresponding data frame. Since the standard encoding process is a common encoding means in the prior art, the present application does not describe much, and the following describes the low complexity encoding process and the high sound quality encoding process in detail.
Low complexity encoding procedure:
the method is suitable for low-complexity versions, has low computational requirement, and is mainly applied to equipment sensitive to power consumption, such as Bluetooth earphones and other equipment. The method has the basic idea that when arithmetic coding and residual coding are carried out on the spectral coefficients by utilizing the ith coding rate or the Nth coding rate, because the code rate is lower, the available bit number is insufficient, partial high spectral coefficients and all residual or LSB data are removed, or the available bit number is enough for all the spectral coefficients to be coded, and partial residual or LSB data are removed. When encoding spectral coefficients, encoding is performed from low to high according to the spectral coefficient index, and components from low to high frequencies are mapped. For convenience of description, a first encoding rate and a second encoding rate are taken as examples, where the first encoding rate is greater than the second encoding rate, and an LC3 audio encoder is taken as an example for explanation. The specific process of the method is described as follows:
and coding the audio frame to be transmitted according to the coding specification until finishing the time domain noise shaping. And then, carrying out a pedigree number quantization process, and simultaneously calculating the bit budget of the coding spectral coefficient corresponding to the second coding rate when calculating the bit budget of the coding spectral coefficient corresponding to the first coding rate according to the specification in the spectral coefficient quantization process.
Firstly, the total bit number available for a frame of LC3 code stream is calculatednbits, where bitrate is the code rate of the first code stream, and frame _ duration is one frame duration, and is divided into 7.5ms or 10 ms. Wherein, the total bit number of the first coding rate: nbits _1= bitrate _1 frame _ duration; total bit number of second coding rate: nbits _2= bitrate _2 frame _ duration. Then calculating the available bit number of a frame of spectral coefficient
And the available bit number of the spectral coefficient of the first coding rate is as follows:
；
available bit number of spectral coefficient of second coding rate:
。
wherein
The number of bits occupied for the bandwidth information,
the number of bits occupied for the TNS-related information,
the number of bits of the LTPF-related information,
the number of bits for the SNS-related information,
the number of bits for the global gain is,
the number of bits is filled for the noise,
for arithmetic coding of the number of bits, the variables are all in the coding processAnd (4) calculating.
Based on the available bit number of the first code rate spectral coefficient and the standard specification, global gain estimation is executed to obtain a public global gain index
The pedigree number is quantized to obtain public quantized spectral coefficients
，
Calculating the number of bits consumed by the spectral coefficient to obtain the number of common consumed bits
Spectral coefficient truncation and global gain adjustment. Truncation of spectral coefficients to updated public quantized spectral coefficients
，
The main thing is to set the non-coding part of high frequency spectrum coefficient to zero. Finally, whether the global gain needs to be adjusted is judged, if so, the global gain is adjusted according to the updated public global gain
The spectral coefficients are re-quantized,
， 
otherwise, the quantization is finished and the re-quantization is performed at most once. Then, side information encoding, arithmetic encoding and residual encoding are performed, wherein the specific flow is shown in fig. 3. The quantization process is to simplify the operation, the quantization is performed only for the first code rate, and the second code rate is encoded according to the available bit numberTaking the number from the quantization result of a code rate, taking the number according to the importance, and performing corresponding processing, wherein the priority sequence from high to low is as follows: spectral coefficients (from low to high frequency), residuals, and LSBs.
Fig. 3 is a diagram of an example of an audio encoding process according to the present application. As shown in fig. 3, the auxiliary information encoding is performed first, wherein the auxiliary information encoding process includes: writing the auxiliary information into a code stream with a first coding rate and writing the auxiliary information into a code stream with a second coding rate; then, an arithmetic coding process is performed, wherein the arithmetic coding process includes: the first step is as follows: writing the TNS information into a code stream of a first coding code rate; the second step is that: writing the TNS information into a code stream of a second coding rate; the third step: writing all the spectrum data into a code stream with a first coding code rate; the fourth step: and writing all the spectrum data into a code stream with a second coding rate, wherein the ending condition is that the residual available bit number is not enough to write the next pair of spectrum coefficients. And finally, residual coding is carried out, wherein the process of residual coding comprises the following steps: if the bit number of the first code stream is remained after the arithmetic coding, writing the residual error or LSB into the code stream of the first coding code rate; and if the bit number of the second code stream is remained after the arithmetic coding, writing the residual error or the LSB into the code stream of the second coding code rate.
High-sound quality coding process:
the method is suitable for high-quality sound versions, has slightly high computational power requirements, and is mainly applied to equipment insensitive to power consumption, such as mobile phones, PCs and the like. The method is implemented by having two encoders at the transmitting end, but the two encoders share most of the encoding modules, such as low-delay modified discrete cosine transform, bandwidth detection, impulse detection, transform domain noise shaping, and time domain noise shaping, and then have a set of pedigree number quantization, noise level estimation, arithmetic coding, and residual coding, although the complexity is slightly higher than that of method 1, but still much lower than that of the two standard encoders. The second method is specifically described as follows:
firstly coding according to a coding specification, and then calculating the bit budget of a coding spectrum coefficient corresponding to a first coding code rate according to the coding specification, and simultaneously calculating the bit budget of a coding spectrum coefficient corresponding to a second coding code rateA bit budget for coding spectral coefficients. First, the total bit number of a frame of LC3 code stream, i.e., the total bit number nbits _1 of the first coding rate and the total bit number nbits _2 of the first coding rate, is calculated, and the calculation process is similar to that described in the first method, and is not described herein again. And calculating the available bit number of the spectral coefficient of the frame by the same method to obtain the available bit number of the spectral coefficient of the first coding rate
And the number of bits available for spectral coefficients of the second coding rate
. Then based on the available bit number of the first coding code rate spectrum coefficient
And a standard specification for performing global gain estimation to obtain a first global gain index
Obtaining a first quantized spectral coefficient by quantizing the pedigree number
，
. Calculating the number of bits consumed by the spectral coefficient to obtain a first number of bits consumed by the spectral coefficient
And truncating the spectral coefficients to obtain updated first quantized spectral coefficients
，
The main thing is to set the non-coding part of high frequency spectrum coefficient to zero. Then, global gain adjustment is carried out, whether global gain needs to be adjusted or not is judged, and if the global gain needs to be adjusted, the global gain is adjusted according to the resultUpdated first global gain
The re-quantization of the spectral coefficients is started, otherwise the quantization is finished and the re-quantization is performed at most once. And the number of available bits of spectral coefficients based on the second coding rate
And the standard specification using the same method as described above, a second global gain is obtained
Second quantized spectral coefficients
， 
Second spectral coefficient consumption bit number
。
The flow described in the second method is the same as the flow chart in fig. 3. The difference is that in the arithmetic coding part, the fourth step is modified to write all the spectrum data into the code stream of the second coding code rate; in addition, in the second method, when a plurality of code streams are generated by encoding, for example, two code streams, the quantization operations are performed on the plurality of code streams, that is, the quantization operations are performed on the first encoding code rate and the second encoding code rate, respectively, because the available bits are enough to write all the spectral coefficients according to the quantization performed by the standard specification.
In practical application, different methods can be selected according to different specific product requirements, and the selection method is preset when the product leaves a factory so as to perform corresponding coding processing.
In the embodiment shown in fig. 2, the method for improving audio transmission stuck includes a process S202, where a sending end selects a data frame with a corresponding code rate according to a transmission condition of an audio frame and sends the data frame to a receiving end.
In this embodiment, for the transmission of audio frames at the transmitting end, data frames with corresponding code rates are selected and transmitted to the receiving end. The transmission condition of the audio frame includes a first transmission, a second transmission, a third transmission, and the like of the audio frame. The data frames obtained by coding with different code rates are correspondingly transmitted according to different transmission conditions of the audio frames, so that the occurrence of the pause phenomenon is reduced in the audio transmission process.
Optionally, the sending end selects a data frame with a corresponding code rate to send to the receiving end according to the transmission condition of the audio frame, including: if the audio frame is transmitted for the first time, the transmitting end transmits a first data frame corresponding to the first coding rate to the receiving end; and if the audio frame is not transmitted for the first time, the transmitting end transmits an ith data frame obtained by encoding with an ith encoding rate or an Nth data frame obtained by encoding with an Nth encoding rate to the receiving end, wherein the transmitting times of the audio frame correspond to i or N.
In this optional embodiment, when the audio frame is transmitted for the first time, the risk of data loss is low, and therefore the first data frame corresponding to the first coding rate is sent to the receiving end. The first coding rate is a standard coding rate of the sending end device, i.e. a higher coding rate. And sending the data frame obtained after the coding is carried out through the first coding rate to a receiving end, so as to ensure the tone quality of the audio coding. If the first transmission fails, for example, the receiving end returns a NACK signal or the transmission time is out, the transmitting end needs to retransmit the audio frame, so that the possibility of data loss is increased in the data transmission process due to the first transmission failure, and in order to avoid the data loss and the packet loss in the transmission process, when the audio frame is not transmitted for the first time, the ith data frame obtained by coding the ith coding rate or the nth data frame obtained by coding the nth coding rate is transmitted to the receiving end.
Optionally, if the audio frame is not transmitted for the first time, the sending end sends the corresponding ith data frame obtained by coding at the ith coding rate or the nth data frame obtained by coding at the nth coding rate to the receiving end, where the sending end includes: analyzing and judging the current sending times of the audio frame; and selecting the ith data frame obtained by coding with the ith coding rate or the Nth data frame obtained by coding with the Nth coding rate according to the sending times, and sending the ith data frame or the Nth data frame obtained by coding with the Nth coding rate to the sending end, wherein the number of data frames in a data packet corresponding to the (i + 1) th data frame is more than that of the data packet corresponding to the ith data frame.
In this optional embodiment, in the sending end device, the sending times of the audio frame data to be sent may be analyzed, where the sending end device sends the audio frame data for the first time, and the sending end device receives a NACK signal or sends the transmission timeout information of the receiving end device for the second time, sends the transmission timeout information for the third time, and so on. The retransmission times are determined according to the actual transmission requirements and coding requirements of the transmitting end device. For example, if the number of retransmissions is set to 4, the corresponding N may be set to 4. During first transmission, a sending end sends data frame data obtained after coding through a first coding rate to a receiving end; during the second transmission, the sending end sends data frame data obtained after the data frame data are coded through the second coding rate to the receiving end; during the third transmission, the sending end sends data frame data obtained after the data frame data are coded through the third coding rate to the receiving end; and in the fourth transmission, the sending end sends the data frame data obtained after the data frame data is coded by the fourth coding rate to the receiving end. The relationship between the first coding rate and the fourth coding rate is sequentially reduced, for example, the first coding rate is greater than the second coding rate. By sequentially reducing the coding rate, the number of data frames in a data packet corresponding to the second data frame obtained after coding is more than that in a data packet corresponding to the first data frame, and the number of data frames in a data packet corresponding to the (i + 1) th data frame is more than that in a data packet corresponding to the (i) th data frame, which is equivalent to increasing the retransmission times and avoiding the loss of data.
Specifically, when the second coding rate is half of the first coding rate, the original data packet contains one audio frame, and the current coding rate is reduced to half, so that one data packet can contain two identical audio frames; although the whole code rate is not changed, the sending end sends the same frame more times, which is equivalent to increase the retransmission times, thereby reducing the frame loss probability.
According to the method for improving the audio transmission stuck, the first audio frame of the sending end is coded by using multiple coding code rates, and the first data frames under different coding code rates are respectively transmitted according to the transmission condition of the first audio frame, so that the audio data are guaranteed to be received at the receiving end, the loss of the audio data in the transmission process is reduced, the stuck phenomenon of audio transmission is avoided, and the user experience is improved.
FIG. 4 is a schematic flow chart diagram illustrating one embodiment of a method for improving audio transmission stuck.
In the embodiment shown in fig. 4, the method for improving audio transmission stuck includes a process S401, where the sending end reduces the first coding rate, the ith coding rate, and/or the nth coding rate according to the first indication information returned by the receiving end, and obtains a corresponding first updated coding rate, an ith updated coding rate, and/or an nth updated coding rate.
Optionally, the receiving end detects the number of times of data packet loss within a preset time range, and if the number of times of data packet loss is greater than a preset threshold, sends out the first indication information.
Specifically, the receiving end detects the number of times of loss of the received data packet within a preset time range, and if the number of times of loss is greater than a preset threshold value, it indicates that the CIS link quality is deteriorated due to increased complex interference in the wireless environment, and the audio jamming phenomenon occurs more frequently. At this time, the receiving end feeds back the first indication information to the transmitting end. After the sending end receives the first indication information, the sending end can reduce the first coding code rate, the ith coding code rate and/or the Nth coding code rate, and correspondingly increase the sending times of data in the sending process of the data by reducing the code rate for coding the audio frame, so that the probability of frame loss is reduced, and the pause phenomenon of audio transmission is avoided.
In the embodiment shown in fig. 4, the method for improving audio transmission blockage includes a process S402, where the sending end increases the first update coding rate, the ith update coding rate, and/or the nth update coding rate according to the second indication information returned by the receiving end, so as to obtain the corresponding first coding rate, the ith coding rate, and/or the nth coding rate.
Optionally, the receiving end detects the number of times of data packet loss within a preset time range, and if the number of times of data packet loss is not greater than a preset threshold, sends out the second indication information.
Specifically, the receiving end detects the loss times of the received data packets within a preset time range, and if the loss times are not greater than a preset threshold value, the wireless environment interference is less, the CIS link quality is enhanced, the data transmission is smooth, and the probability of the blocking phenomenon is low. At this time, the receiving end feeds back the second indication information to the transmitting end. And after the sending end receives the second indication information, the sending end can increase the reduced first coding rate, the ith coding rate and/or the Nth coding rate. Under the environment of good communication, the quality of audio coding can be improved by increasing the code rate of coding the audio frames, and meanwhile, the phenomenon of audio blockage can not occur.
In the present embodiment shown in fig. 4, in step S403, according to the first coding rate, the ith coding rate, or the nth coding rate determined in step S401 or step S402; or the first update coding rate, the ith update coding rate, or the nth update coding rate, where the subsequent transmission process of the specific coding process is similar to the processes S201 and S202 in fig. 2. And will not be described in detail herein.
Fig. 5 is a flow chart illustrating an example of the method for improving audio transmission stuck according to the present application.
As shown in fig. 5, at a transmitting end of data transmission, it is determined whether first indication information sent from a receiving end is received, where the first indication information is used to reduce a coding rate of the transmitting end. If the first indication information is received, the sending end reduces and updates the first code rate of the code; and meanwhile, judging whether second indication information sent by the receiving end is received or not, wherein the second indication information is used for increasing the coding code rate of the sending end. And if the second indication information is received, the transmitting end raises and updates the second code rate of the coding. At the receiving end, when the number of times of loss of the transmitted data packets is greater than a preset threshold value within a preset time, it indicates that the communication environment is poor at the moment, and data loss can occur, resulting in a stuck phenomenon. At the moment, first indication information is sent out, the coding code rate of a sending end is reduced, and then the packet loss rate is reduced; at the receiving end, when in the preset time, the number of times that the transmitted data packet is lost is not more than the preset threshold value, which indicates that the communication environment is good at the moment, the probability of data loss is low, and the coding code rate of the transmitting end is increased at the moment, so that the tone quality is improved, and the use experience of the user is improved.
At a sending end, for the audio data of the current frame, except for outputting the frame data corresponding to the encoded first code rate, the frame data obtained by a plurality of lower code rate codes lower than the first code rate is output at the same time. When the current audio data is transmitted for the first time, the probability of data loss is low, and the transmitting end transmits frame data obtained by first code rate coding; if the current audio data is transmitted for the first time, the data is transmitted for at least one time and the data loss probability is high, then the frame data obtained after the data is encoded by the lower code rate is transmitted according to the preset transmission rule, the loss of the data packet in the transmission process of the audio data is avoided, and the phenomenon of transmission blockage is avoided.
Fig. 6 illustrates one embodiment of the present system for improving audio transmission stuck.
In the embodiment shown in fig. 6, the system for improving audio transmission blockage of the application includes an encoding module 601, which encodes an audio frame to be transmitted by using multiple encoding rates to obtain multiple corresponding data frames, where the multiple encoding rates include a first encoding rate to an nth encoding rate, N is an integer greater than 2, and a value of N is determined according to an audio encoding requirement, where the first encoding rate is greater than an ith encoding rate, the ith encoding rate is greater than an ith +1 encoding rate, and i is an integer greater than 1 and less than N;
a sending module 602, which selects a data frame with a corresponding code rate to send to a receiving module according to the transmission condition of the audio frame, wherein
If the audio frame is transmitted for the first time, the transmitting end transmits a first data frame obtained by encoding with a first encoding rate to the receiving end;
and if the audio frame is not transmitted for the first time, the transmitting end transmits the corresponding ith data frame obtained by encoding with the ith encoding rate or the Nth data frame obtained by encoding with the Nth encoding rate to the receiving end, wherein the transmitting times of the audio frame correspond to i or N.
In one embodiment of the present application, a computer-readable storage medium stores computer instructions, wherein the computer instructions are operable to perform the method for improving audio transmission stuck 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 one 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 the method of improving audio transmission stuck described in any of the embodiments.
In the embodiments provided in the present application, it should be understood that the disclosed apparatus may be implemented in other manners. 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 (7)
1. A method for improving audio transmission stuck, comprising:
a sending end encodes an audio frame to be sent by using multiple encoding code rates to obtain corresponding multiple data frames, wherein the multiple encoding code rates comprise a first encoding code rate to an Nth encoding code rate, N is an integer greater than 2, and the value of N is determined according to audio encoding requirements, wherein the first encoding code rate is greater than an ith encoding code rate, the ith encoding code rate is greater than an ith +1 encoding code rate, and i is an integer greater than 1 and less than N;
the sending end selects the data frame with the corresponding code rate to send to the receiving end according to the transmission condition of the audio frame, wherein,
if the audio frame is transmitted for the first time, the sending end sends a first data frame obtained by the first coding rate coding to the receiving end;
and if the audio frame is not transmitted for the first time, the sending end sends an ith data frame obtained by coding with the ith coding rate or an Nth data frame obtained by coding with the Nth coding rate to the receiving end, wherein the sending times of the audio frame correspond to the i or the N.
2. The method of claim 1, wherein the transmitting end encodes the audio frames to be transmitted with a plurality of coding rates, comprising:
encoding the audio frame according to a preset encoding process and the plurality of encoding rates, the preset encoding process comprising a low complexity encoding process and a high sound quality encoding process, wherein,
in the low-complexity encoding process, performing a standard encoding process on the audio frame according to the first encoding code rate to obtain the corresponding data frame;
coding the audio frame according to the ith coding rate or the Nth coding rate, and deleting high-frequency spectrum coefficient data, residual error data and/or LSB data in the coding process to obtain the corresponding data frame;
in the high-sound-quality encoding process, the audio frame is encoded according to the first encoding code rate, the ith encoding code rate or the Nth encoding code rate by sharing a corresponding encoding module to obtain the corresponding data frame, wherein the encoding module comprises a bandwidth detection module and/or a time domain noise shaping module.
3. The method of claim 1, wherein if the audio frame is not transmitted for the first time, the sending end sends a corresponding ith data frame encoded by the ith coding rate or an nth data frame encoded by the nth coding rate to the receiving end, and the method includes:
analyzing and judging the current sending times of the audio frame;
and selecting the corresponding ith data frame obtained by coding with the ith coding rate or the N data frame obtained by coding with the N coding rate according to the sending times to send the ith data frame or the N data frame obtained by coding with the N coding rate to the receiving end, wherein the number of data frames in a data packet corresponding to the (i + 1) th data frame is more than that of the data packets corresponding to the ith data frame.
4. The method of claim 1 or 2, further comprising:
the sending end reduces the first coding rate, the ith coding rate and/or the Nth coding rate according to the first indication information returned by the receiving end to obtain a corresponding first updating coding rate, the ith updating coding rate and/or the Nth updating coding rate; or
And the sending end raises the first updating code rate, the ith updating code rate and/or the Nth updating code rate according to second indication information returned by the receiving end to obtain the corresponding first code rate, the ith code rate and/or the Nth code rate.
5. The method of claim 4, further comprising:
the receiving end detects the data packet loss times within a preset time range;
if the loss times are larger than a preset threshold value, sending the first indication information;
and if the lost times are not greater than the preset threshold value, sending out the second indication information.
6. A system for improving audio transmission stuck, comprising: an encoding module, a transmitting module and a receiving module, wherein,
the encoding module encodes an audio frame to be transmitted by using multiple encoding rates to obtain corresponding multiple data frames, wherein the multiple encoding rates include a first encoding rate to an Nth encoding rate, N is an integer greater than 2, and the value of N is determined according to the audio encoding requirement, wherein the first encoding rate is greater than an ith encoding rate, the ith encoding rate is greater than an (i + 1) th encoding rate, and i is an integer greater than 1 and less than N;
the sending module selects the data frame with the corresponding code rate to send to the receiving module according to the transmission condition of the audio frame, wherein,
if the audio frame is transmitted for the first time, the transmitting end transmits a first data frame obtained by the first coding rate coding to the receiving end;
and if the audio frame is not transmitted for the first time, the sending end sends an ith data frame obtained by coding with the ith coding rate or an Nth data frame obtained by coding with the Nth coding rate to the receiving end, wherein the sending times of the audio frame correspond to the i or the N.
7. A computer-readable storage medium having stored thereon computer instructions operable to perform the method of any one of claims 1-5.
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