WO2021159782A1 - 数据传输方法、装置、终端、存储介质和系统 - Google Patents

数据传输方法、装置、终端、存储介质和系统 Download PDF

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Publication number
WO2021159782A1
WO2021159782A1 PCT/CN2020/127444 CN2020127444W WO2021159782A1 WO 2021159782 A1 WO2021159782 A1 WO 2021159782A1 CN 2020127444 W CN2020127444 W CN 2020127444W WO 2021159782 A1 WO2021159782 A1 WO 2021159782A1
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Prior art keywords
data
transmission
compression
coefficient
restored
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PCT/CN2020/127444
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English (en)
French (fr)
Inventor
梁俊斌
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腾讯科技（深圳）有限公司
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Publication of WO2021159782A1 publication Critical patent/WO2021159782A1/zh
Priority to US17/675,400 priority Critical patent/US20220172731A1/en

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L69/00—Network arrangements, protocols or services independent of the application payload and not provided for in the other groups of this subclass
    • H04L69/04—Protocols for data compression, e.g. ROHC
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0009—Systems modifying transmission characteristics according to link quality, e.g. power backoff by adapting the channel coding
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0033—Systems modifying transmission characteristics according to link quality, e.g. power backoff arrangements specific to the transmitter
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00—Arrangements for detecting or preventing errors in the information received
    • H04L1/0001—Systems modifying transmission characteristics according to link quality, e.g. power backoff
    • H04L1/0036—Systems modifying transmission characteristics according to link quality, e.g. power backoff arrangements specific to the receiver
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/04—Time compression or expansion
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/03—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters
    • G10L25/06—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the type of extracted parameters the extracted parameters being correlation coefficients
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/90—Pitch determination of speech signals

Definitions

  • This application relates to the field of data transmission, and in particular to a data transmission method, device, terminal, storage medium, and system.
  • the Internet is a transmission network that is prone to network fluctuations and congestion.
  • the current data transmission methods are prone to packet loss caused by network fluctuations, that is, audio data packets. Missed transmissions and mistransmissions; for example, Internet audio applications such as live voice, voice calls, and voice broadcasts have high requirements on network stability and bandwidth, otherwise the audio received by the receiving end may be inconsistent, stuck, etc. Condition.
  • the current data transmission method will repeatedly send a large number of redundant data to reduce the impact of packet loss.
  • this method will take up a lot of network resources and cost a lot of It takes time and computing resources to process and send these redundant data. Therefore, the current data transmission methods are inefficient.
  • this application provides a data transmission method, device, terminal, storage medium, and system.
  • the embodiment of the present application provides a data transmission method, which is suitable for the sending end, and includes:
  • the embodiment of the present application provides a data transmission method, which is suitable for the receiving end, and includes:
  • the transmission data packet including redundant data and a compression factor
  • the embodiment of the present application also provides a data transmission device, which is suitable for the sending end, and includes:
  • the first acquiring unit is used to acquire audio data and transmission status information
  • a coefficient unit configured to determine a scaling coefficient and a redundancy coefficient based on the transmission state information
  • a compression unit configured to perform time-domain compression processing on the audio data according to the scaling factor to obtain compressed data
  • An encoding unit configured to perform channel encoding on the compressed data according to the redundancy coefficient to obtain a transmission data packet
  • the first sending unit is configured to send the transmission data packet.
  • the embodiment of the present application also provides a data transmission device, which is suitable for the receiving end, and includes:
  • the second acquiring unit is configured to acquire a transmission data packet, where the transmission data packet includes redundant data and a scaling factor;
  • the occupying unit is configured to determine the current transmission state information based on the transmission data packet
  • the second sending unit is configured to send the transmission status information at the current moment
  • a decoding unit configured to perform channel decoding on the transmission data packet according to the redundant data to obtain data to be restored
  • the expansion unit is configured to perform time-domain expansion processing on the data to be restored according to the scaling factor to obtain restored data.
  • An embodiment of the present application further provides a terminal, including a memory and a processor.
  • the memory stores computer-readable instructions.
  • the processor executes the implementation of the present application. The steps in any of the data transmission methods provided in the example.
  • the embodiments of the present application also provide one or more non-volatile storage media storing computer-readable instructions.
  • the computer-readable instructions are executed by one or more processors, the one or more processors execute the present application. Steps in any data transmission method provided in the embodiment.
  • An embodiment of the present application also provides a data transmission system.
  • the data transmission system includes a sending end and a receiving end, wherein:
  • the sending end is used to obtain audio data and obtain the transmission status information sent by the receiving end, determine a scaling factor and a redundancy coefficient based on the transmission status information, and perform time-domain compression processing on the audio data according to the scaling factor to obtain Compress the data, perform channel coding on the compressed data according to the redundancy coefficient to obtain a transmission data packet, and send the transmission data packet to the receiving end.
  • the receiving end is used to obtain a transmission data packet sent by the sending end, the transmission data packet includes redundant data and a scaling factor, the transmission state information at the current moment is determined based on the transmission data packet, and the transmission state at the current moment is sent.
  • the information is sent to the sending end, the transmission data packet is channel-decoded to obtain the data to be restored, and the data to be restored is subjected to time-domain expansion processing according to the scaling factor to obtain the restored data.
  • Figure 1a is a schematic diagram of a scenario of a data transmission system provided by an embodiment of the present application.
  • FIG. 1b is a schematic diagram of the first flow of a data transmission method provided by an embodiment of the present application.
  • FIG. 2 is a schematic diagram of a second flow of a data transmission method provided by an embodiment of the present application
  • FIG. 3 is a schematic flowchart of a data transmission system provided by an embodiment of the present application.
  • FIG. 4 is a schematic diagram of the first structure of a data transmission device provided by an embodiment of the present application.
  • FIG. 5 is a schematic diagram of a second structure of a data transmission device provided by an embodiment of the present application.
  • Fig. 6 is a schematic structural diagram of an electronic device provided by an embodiment of the present application.
  • the embodiments of the present application provide a data transmission method, device, terminal, storage medium, and system.
  • the data transmission device can be integrated in an electronic device, and is suitable for the sending end and the receiving end; the sending end and the receiving end can be the same electronic device, or different electronic devices, when the sending end and the receiving end are different In the case of electronic equipment, the sending end and the receiving end may be the same type of electronic equipment or different types of electronic equipment.
  • the electronic device can be a terminal, a server, etc.
  • the terminal can be a mobile phone, a tablet, a Bluetooth smart device, a laptop, or a personal computer (PC)
  • the server can be a single server or A server cluster composed of multiple servers.
  • the data transmission apparatus may also be integrated in multiple electronic devices.
  • the data transmission apparatus may be integrated in multiple servers, and the multiple servers implement the data transmission method of the present application.
  • the server may also be implemented in the form of a terminal.
  • the data transmission system may include a mobile phone A as a sending end and a mobile phone B as a receiving end.
  • mobile phone A can obtain audio data and transmission status information from mobile phone B, and determine the compression coefficient and redundancy coefficient based on the transmission status information, and then perform time-domain compression processing on the audio data according to the compression coefficient to obtain compressed data, and then according to the redundancy
  • the residual coefficient performs channel coding on the compressed data to obtain the transmission data packet, and finally the transmission data packet is sent to the mobile phone B.
  • mobile phone B can obtain the transmission data packet and compression coefficient from mobile phone A, and then determine the current transmission status information based on the transmission data packet, and then send the current transmission status information to mobile phone A, and then channel-decode the transmission data packet. Obtain the data to be restored, and finally perform time-domain expansion processing on the data to be restored according to the compression coefficient to obtain the restored data.
  • a data transmission method based on data transmission is provided. As shown in FIG. 1b, the method is applied to the sender as an example for illustration.
  • the process of the data transmission method may be as follows:
  • audio data refers to electronic data information to be sent, and the electronic data information can be expressed in multiple data types, for example, it can be pure audio data, video data containing audio, and so on.
  • Transmission status information refers to related information that can reflect the status of data transmission.
  • transmission status information can include channel utilization, bandwidth, packet loss rate, redundancy rate of the sender, sending code rate, receiving code rate, transmission rate, and information. Noise ratio, channel gain, noise power, etc.
  • audio data can be obtained through various methods, for example, audio data can be obtained from a database through the network, audio data can also be captured and recorded by sensors, audio data can also be obtained by user input, or read locally Audio data, etc.
  • the audio data of the user can be recorded by using the onboard recording device.
  • the transmission status information can be obtained through various methods, for example, the transmission status information can be obtained from the database through the network, the transmission status information can also be collected through the sensor, and the transmission status information sent by the receiving end can also be obtained through the network. , You can also read the transmission status information locally, and so on.
  • the compression factor is a physical quantity that describes the size of the compressibility
  • the redundancy factor is a physical quantity that describes the size of the transmitted data packet by the redundant data.
  • the receiving end can use redundant data to check and correct the received data information.
  • the redundant data can be error correction codes, error checking codes, data fragments of transmitted data, and so on.
  • the compression factor is 0.9, which can describe that the data is compressed to 0.9 times the original size.
  • the redundancy coefficient is 0.3, it can be described that 30% of the transmitted data is redundant data, and the remaining 70% is valid data.
  • the transmission status information may include the received quantity, so step 102 may include the following steps:
  • the compression factor is determined based on the packet loss rate and the transmission code rate.
  • statistics are performed on the transmission data packets sent within the preset historical time period, so as to obtain the transmission code rate and the transmission quantity.
  • the sender rate refers to the total size of valid data and redundant data sent by the sender in a unit time, in bytes.
  • the sent quantity refers to the number of data packets sent by the sender in a unit time.
  • the received quantity refers to the number of data packets received by the receiving end in a unit time.
  • the Loss Rate refers to the ratio of the number of data packets lost per unit time to the number of packets sent.
  • the packet loss rate may increase, that is, the probability of channel loss of data will increase. At this time, it may cause video mosaic phenomenon, partial deformation, blurred images, frequent refresh, asynchronous audio and video, and static images. , Delay, audio interruption and other issues. The higher the packet loss rate, the more obvious the impact on data transmission applications such as audio and video calls.
  • valid data may be repeatedly transmitted, that is, redundant data may be transmitted.
  • redundant data may be transmitted.
  • the redundancy coefficient can be modified so that the sending end sends a certain amount of redundant data to ensure that the information received by the receiving end is complete and correct.
  • the packet loss rate is 25%, that is, 25 transmission data are lost.
  • the redundancy coefficient is 0.25, that is, assuming that 100 transmission data are to be sent again, among which 25 are redundant data and 75 are valid data.
  • the packet loss rate and the transmission code rate in the preset historical time period are positively correlated, that is, the redundant data in the transmitted data packets is getting larger and larger, and the packet loss rate is getting larger and larger, you can It is judged that the channel is currently at the upper working limit.
  • this embodiment can set a compression factor to reduce the size of the transmitted data, so that the transmission code rate is reduced, thereby reducing the pressure on the channel.
  • the step of "determining the compression factor based on the packet loss rate and the transmission code rate” may include the following steps:
  • the compression factor is determined according to the packet loss rate and the transmission code rate.
  • the compression coefficient can be modified to reduce the transmission code rate, thereby reducing the pressure on the channel and maintaining channel stability.
  • the compression factor may be sent when the transmission data packet is sent, for example, the data transmission packet and the compression factor can be simultaneously sent to the receiving end through the network.
  • Time Domain Data Compression refers to deleting and transforming some data in electronic data in the time domain to achieve the effect of compression.
  • time-domain compression methods include Adaptive Differential Pulse Code Modulation (ADPCM), linear predictive coding (LPC), and code excited linear predictive coding (Code Excited).
  • ADPCM Adaptive Differential Pulse Code Modulation
  • LPC linear predictive coding
  • Code Excited code excited linear predictive coding
  • Linear Prediction CELP, Overlap-and-Add (OLA), etc.; among them, the overlap-and-add algorithm can include Synchronized Overlap-Add (SOLA), Pitch Synchronized Overlap -Add, PSOLA), waveform similarity overlap-and-add (WSOLA), etc.
  • SOLA Synchronized Overlap-Add
  • PSOLA Pitch Synchronized Overlap -Add
  • WSOLA waveform similarity overlap-and-add
  • step 103 may include the following steps:
  • pitch is a main time-domain parameter that determines speech prosody.
  • the core of the time-domain pitch synchronization superposition method is pitch synchronization.
  • the pitch in the audio data is marked, for example, voiced sounds are marked; then, the pitch points are marked Audio data is sampled to obtain multiple sub-audio data, a series of insertion, deletion, and modification are performed on the sub-audio data, and compressed data is synthesized.
  • step 103 may include the following steps:
  • the sub audio data is synthesized to obtain compressed data.
  • the coefficients of the sampling window can be determined according to the compression factor, such as the step size, window size, etc.; then, the sampling window is smoothly moved in the audio data, and the sampling window is adjusted every time a certain step is passed. Sampling the data to obtain the sub-audio data; finally, superimpose all the sub-audio data to obtain the compressed data.
  • the compression factor such as the step size, window size, etc.
  • the waveform similarity superposition method can be used to perform time-domain compression.
  • Data can include the following steps:
  • the waveform correlation coefficient can describe the degree of similarity between two waveforms.
  • two sub-audio data with similar waveforms can be superimposed, and finally compressed data is obtained.
  • Digital signals are often transmitted due to various reasons, resulting in errors in the transmitted data stream, which causes the receiving end to produce images such as jumps, distortions, discontinuities, and mosaics.
  • Corresponding processing of data through channel coding can make the channel have a certain error correction ability and anti-interference ability, which greatly avoids the occurrence of bit errors in the data transmission process.
  • channel coding includes a variety of coding methods, such as error correction coding, error checking coding, and so on.
  • the error correction codes may include forward error correction codes (Forward Error Correction, FEC), Reed-solomon codes (RS), convolutional codes, turbo codes, and so on.
  • FEC Forward Error Correction
  • RS Reed-solomon codes
  • convolutional codes turbo codes, and so on.
  • redundant data corresponding to the compressed data can be generated according to the redundancy coefficient, and then these redundant data and compressed data are channel-encoded and packaged to obtain a transmission data packet.
  • the redundant coefficient may be channel-coded to obtain a compression factor identifier, and then the compressed data may be channel-coded according to the compression factor identifier to obtain a transmission data packet.
  • the compression factor identifier is an identifier that carries compression factor information, and can be used to indicate the size of the compression factor.
  • the compression factor identifier corresponding to the compression factor may be used as the header of the transmission data packet.
  • the transmission data packet can be sent to the receiving end.
  • the compression factor When sending a transmission data packet, the compression factor can be sent within a period of time, or a data packet containing the compression factor identifier as the header can be sent, and so on.
  • the data transmission solution provided by the embodiments of this application can be applied to various data transmission scenarios, for example, in audio transmission scenarios, especially VoIP (Voice over Internet Protocol, IP-based voice transmission), voice broadcasting, audio and video live broadcasting, etc.
  • VoIP Voice over Internet Protocol
  • IP-based voice transmission Voice over Internet Protocol
  • voice broadcasting Voice broadcasting
  • audio and video live broadcasting etc.
  • the embodiments of this application can monitor the working conditions of the channel in real time, and control the transmission of redundant data and the effect of audio compression according to the working conditions. Therefore, the implementation of this application is adopted
  • the solution provided by the example can perform data transmission more efficiently while ensuring the audio effect.
  • the embodiment of the present application can obtain audio data and transmission status information; determine the compression coefficient and redundancy coefficient based on the transmission status information; perform time-domain compression processing on the audio data according to the compression coefficient to obtain compressed data; The compressed data is channel-encoded to obtain the transmission data packet; the transmission data packet is sent.
  • this solution can analyze the transmission status information to judge the current working condition of the channel in real time.
  • the channel congestion can be improved by modifying the compression coefficient and the data delay rate and packet loss rate can be reduced.
  • Improve the stability of the channel when the channel packet loss rate is high, the packet loss can be improved by modifying the redundancy coefficient, so as to ensure the correct and complete data transmission.
  • This solution improves the utilization rate of the channel, and makes the data transmission speed faster and more stable. As a result, the solution can improve the efficiency of data transmission.
  • a data transmission method based on data transmission is provided. As shown in FIG. 2, the application of the method to the receiving end is taken as an example for description.
  • the flow of the data transmission method may be as follows:
  • the transmission data packet sent by the sender can be obtained through the channel.
  • the compression coefficient sent by the sending end may be obtained at the same time as the transmission data packet sent by the sending end is obtained.
  • the transmission data packet may include a compression factor identifier, and the compression factor can be obtained by identifying the compression factor identifier.
  • the header of the transmission data packet is the compression factor identifier, and the compression factor can be obtained by reading the header.
  • the state of the channel in addition to analyzing the state of the channel at the transmitting end, the state of the channel may also be analyzed at the receiving end.
  • step 202 may include the following steps:
  • the transmission status information includes the packet loss rate.
  • the received quantity, sent quantity, packet loss rate, and transmission status information can refer to the descriptions in step 102 and step 103.
  • the current transmission status information can be sent to the sending terminal.
  • multiple methods may be used to send the current transmission status information to the sending terminal, for example, sending the current transmission status information to the sending terminal through the network, sending the current transmission status information to the sending terminal through a storage medium, etc. Wait.
  • channel decoding corresponds to channel coding, and the purpose is to restore the data processed by channel coding and restore it to the state when channel coding is not performed.
  • the transmission data packet may include redundant data and transmission data, and the transmission data can be channel-encoded through the redundant data, so as to realize the inspection, error correction, and omission correction of the transmission data, and finally obtain the data to be restored.
  • channel decoding methods there are also various channel decoding methods, such as Forward Error Correction (FEC), Reed-solomon codes (RS), convolutional codes, Turbo codes, etc. .
  • FEC Forward Error Correction
  • RS Reed-solomon codes
  • convolutional codes Turbo codes, etc.
  • Time Domain Data Decompression refers to data modification, addition, and insertion of electronic data in the time domain to achieve the effect of decompression.
  • time domain expansion and time domain compression are similar, such as synchronous waveform superposition method, pitch synchronous superposition method, waveform similarity superposition method, and so on.
  • an overlap and superposition algorithm may be used for time domain expansion.
  • the time domain expansion method is similar to the time domain compression method, and step 103 may include the following steps:
  • the compression coefficient and the decompression coefficient correspond to each other.
  • the decompression coefficient corresponding to the compression coefficient x is 1/x; for example, in some embodiments, the decompression coefficient corresponding to the compression coefficient x is 1-x.
  • the waveform similarity superposition method can be used for time-domain expansion.
  • the step of "synthesizing the sub-data to be restored to obtain the restored data" may include The following steps:
  • the waveform cross-correlation coefficient and the waveform superposition processing can refer to step 103.
  • the data transmission scheme provided by the embodiments of the application can be applied to various data transmission scenarios, for example, in audio transmission scenarios, especially VoIP, voice broadcasting, audio and video live broadcasting, etc., which require high packet loss rate and delay rate.
  • the embodiment of the application can send the working status of the channel to the sending end in real time, so that the sending end can supervise and control the transmission of redundant data and the effect of audio compression on the channel. Therefore, the solution provided by the embodiment of the application can be used Under the circumstance of ensuring the audio effect, data transmission can be carried out more efficiently.
  • the embodiment of the application can obtain transmission data packets, which include redundant data and compression coefficient; determine the current transmission status information based on the transmission data packets; send the current transmission status information; according to the redundant data pair
  • the transmission data packet is channel-decoded to obtain the data to be restored; the data to be restored is subjected to time-domain expansion processing according to the compression coefficient to obtain the restored data.
  • this solution can count and send transmission status information so that the sender can analyze the status of the channel, thereby modifying the compression factor to improve channel congestion, and the redundancy factor to improve packet loss, and ultimately ensure data transmission. It is correct and complete, and improves the utilization rate of the channel, which makes the data transmission speed faster and more stable. As a result, this solution can improve the efficiency of the data transmission method.
  • FEC forward error correction
  • PLC Packet Loss Concealment
  • ARQ Automatic Repeat Request
  • FEC technology uses redundant coding to generate redundant information to compensate for packet loss, and its anti-packet ability is proportional to the bandwidth of the channel it uses; optionally, when packet loss occurs at the receiving end, FEC technology can use redundancy Information performs packet loss recovery.
  • redundancy Information performs packet loss recovery. The more redundant information, the stronger the ability to resist packet loss, and the higher the bandwidth occupied. However, the more bandwidth occupied will cause the network quality to deteriorate and cause more packet loss.
  • the data transmission system may include a sending end and a receiving end.
  • the following will take FEC technology for audio transmission between the sending end and the receiving end as an example. The method of the embodiment of the present application will be described in detail.
  • the sending end obtains audio data and the receiving end obtains transmission status information.
  • the sending end determines the compression coefficient and the redundancy coefficient based on the transmission status information.
  • the packet loss rate and the sending code rate of the sender can be counted.
  • the increase of the sending code rate will not necessarily affect the packet loss rate.
  • the packet loss rate will increase after the sending code rate is increased.
  • the sending end performs time-domain compression processing on the audio data according to the compression coefficient to obtain compressed audio.
  • WSOLA can be used to divide the original speech signal into frames with a length of L, and then synthesize in units of frames.
  • the synthesis process may be sampling at the ⁇ (L k ) sampling point of the original signal.
  • the sending end performs channel coding on the compressed audio according to the redundancy coefficient to obtain a transmission data packet.
  • the sending end sends a transmission data packet to the receiving end, and the receiving end obtains the transmission data packet from the sending end, and the transmission data packet includes redundant data and a compression factor.
  • the receiving end determines the current transmission status information based on the transmission data packet.
  • the receiving end sends the current transmission status information to the sending end.
  • the receiving end performs channel decoding on the transmission data packet according to the redundant data to obtain the audio to be restored.
  • the receiving end performs time-domain expansion processing on the audio to be restored according to the compression coefficient to obtain the restored audio.
  • the expansion coefficient can be 1/ ⁇ at this time
  • steps 101 to 105 and steps 201 to 205 please refer to steps 101 to 105 and steps 201 to 205.
  • the data transmission system may include a sending end and a receiving end, where the sending end can obtain audio data and the transmission status information from the receiving end, and determine the compression factor and redundancy based on the transmission status information.
  • the receiving end can Obtain the transmission data packet from the sender, the transmission data packet includes redundant data and compression coefficient, and then determine the current transmission status information based on the transmission data packet, and then send the current transmission status information to the sender, and then according to the redundant data pair
  • the transmission data packet is channel-decoded to obtain the audio to be restored, and finally the audio to be restored is time-domain expanded according to the compression coefficient to obtain the restored audio.
  • this solution can modify the time domain scale of the audio while ensuring that the audio frequency remains unchanged, thereby extending or shortening the audio duration without significantly reducing the audio quality. Due to the process of modifying the time domain scale It can ensure that the pitch frequency of the voice is not destroyed, so the modified voice tone and tone can be well protected.
  • this solution when it is detected that the higher the redundancy rate and the higher the packet loss rate, this solution first compresses the voice signal source in the time domain to reduce the transmission bit rate, and then decodes it at the receiving end before using the same as the sending end.
  • the time-domain scale modification ratio is used for time-domain expansion, and then the original signal is restored. Therefore, this solution not only ensures the audio quality in the case of channel packet loss, but also reduces the working pressure of the channel, and balances the audio quality and channel stability in real time. This makes the channel more stable, and further improves the smoothness of channel transmission, thereby improving the efficiency of data transmission.
  • steps in the embodiments of the present application are not necessarily executed in sequence in the order indicated by the step numbers. Unless specifically stated in this article, the execution of these steps is not strictly limited in order, and these steps can be executed in other orders. Moreover, at least part of the steps in each embodiment may include multiple sub-steps or multiple stages. These sub-steps or stages are not necessarily executed at the same time, but can be executed at different times. The execution of these sub-steps or stages The sequence is not necessarily performed sequentially, but may be performed alternately or alternately with at least a part of other steps or sub-steps or stages of other steps.
  • an embodiment of the present application also provides a data transmission device.
  • the data transmission device may be integrated in an electronic device, and the electronic device may be a terminal, a server, or other devices.
  • the data transmission apparatus may include a first obtaining unit 401, a coefficient unit 402, a compression unit 403, an encoding unit 404, and a first sending unit 405, as follows:
  • the first obtaining unit 401 (1) The first obtaining unit 401:
  • the first obtaining unit 401 may be used to obtain audio data and transmission status information.
  • the coefficient unit 402 may be used to determine the scaling coefficient and the redundancy coefficient based on the transmission state information.
  • the transmission status information includes the received quantity
  • the coefficient unit 402 may include a statistics subunit, a packet loss rate subunit, a redundancy coefficient subunit, and a compression coefficient subunit, as follows:
  • the statistics subunit can be used to count the transmitted transmission data packets to obtain the transmission code rate and the number of transmissions.
  • the packet loss rate subunit can be used to calculate the packet loss rate based on the number of transmissions and the number of receptions.
  • the redundancy coefficient subunit may be used to determine the redundancy coefficient based on the packet loss rate.
  • the compression factor subunit can be used to determine the compression factor based on the packet loss rate and the transmission code rate.
  • the compression factor subunit can be used to:
  • the compression factor is determined according to the packet loss rate and the transmission code rate.
  • the compression unit 403 may be used to perform time-domain compression processing on the audio data according to the scaling factor to obtain compressed data.
  • the compression unit 403 may be used to:
  • the compression unit 403 may include a compression window subunit, a compression sampling subunit, and a compression subunit, as follows:
  • the compression window subunit can be used to determine the sampling window according to the compression factor.
  • the compressed sampling subunit may be used to perform data sampling on the audio data based on the sampling window to obtain sub audio data.
  • the compression subunit can be used to synthesize the sub audio data to obtain compressed data.
  • the compression subunit can be used to:
  • the encoding unit 404 may be used to perform channel encoding on the compressed data according to the redundancy coefficient to obtain a transmission data packet.
  • the encoding unit 404 may be used to:
  • the first sending unit 405 may be used to send a transmission data packet.
  • each of the above units can be implemented as an independent entity, or can be combined arbitrarily, and implemented as the same or several entities.
  • each of the above units please refer to the previous method embodiments, which will not be repeated here.
  • the data transmission device of this embodiment acquires audio data and transmission status information by the first acquisition unit; the coefficient unit determines the scaling factor and redundancy coefficient based on the transmission status information; and the compression unit performs time processing on the audio data according to the scaling factor. Domain compression processing to obtain compressed data; the coding unit performs channel coding on the compressed data according to the redundancy coefficient to obtain a transmission data packet; and the first sending unit sends the transmission data packet.
  • the embodiments of the present application can improve the efficiency of data transmission.
  • an embodiment of the present application also provides a data transmission device, which can be integrated in the receiving end. .
  • the receiving end is a mobile phone as an example, and the method of the embodiment of the present application will be described in detail.
  • the data transmission apparatus may include a second acquiring unit 501, an occupying unit 502, a second sending unit 503, a decoding unit 504, and an expansion unit 505, as follows:
  • the second acquiring unit 501 (1) The second acquiring unit 501:
  • the second obtaining unit 501 may be used to obtain the transmission data packet and the scaling factor.
  • the occupancy unit 502 may be used to determine the current transmission state information based on the transmission data packet.
  • the occupying unit 502 can be used to:
  • the transmission status information includes the packet loss rate.
  • the second sending unit 503 may be used to send the current transmission status information.
  • the decoding unit 504 may be used to perform channel decoding on the transmission data packet to obtain the data to be restored.
  • the expansion unit 505 may be used to perform time-domain expansion processing on the data to be restored according to the scaling factor to obtain the restored data.
  • the expansion unit 505 may include a decompression coefficient subunit, a decompression window subunit, a decompression sampling subunit, and a complex atom unit, as follows:
  • the decompression coefficient subunit may be used to determine the corresponding decompression coefficient according to the compression coefficient.
  • the decompression window subunit may be used to determine the sampling window according to the decompression coefficient.
  • the decompression sampling subunit may be used to perform data sampling on the data to be restored based on the sampling window to obtain the sub-data to be restored.
  • the polyatomic unit can be used to synthesize the sub-data to be restored to obtain the restored data.
  • polyatomic units can be used to:
  • each of the above units can be implemented as an independent entity, or can be combined arbitrarily, and implemented as the same or several entities.
  • each of the above units please refer to the previous method embodiments, which will not be repeated here.
  • the data transmission device of this embodiment acquires the transmission data packet and the scaling factor by the second acquiring unit; the occupying unit determines the current transmission status information based on the transmission data packet; and the second sending unit sends the current transmission status Information; the decoding unit performs channel decoding on the transmission data packet to obtain the data to be restored; the expansion unit performs time-domain expansion processing on the data to be restored according to the scaling factor to obtain the restored data.
  • the embodiments of the present application can improve data transmission.
  • the embodiments of the present application also provide an electronic device, which may be a terminal, a server, or other devices.
  • the electronic device of this embodiment is a terminal as an example for detailed description.
  • FIG. 6 it shows a schematic structural diagram of the terminal involved in the embodiment of the present application, for example:
  • the terminal may include one or more processing core processors 601, one or more memory 602 storing computer-readable storage media, a power supply 603, an input module 604, a communication module 605 and other components.
  • processing core processors 601 one or more memory 602 storing computer-readable storage media
  • memory 602 storing computer-readable storage media
  • input module 604 a communication module 605 and other components.
  • FIG. 4 does not constitute a limitation on the terminal, and may include more or fewer components than shown in the figure, or combine some components, or arrange different components. in:
  • the processor 601 is the control center of the terminal. It uses various interfaces and lines to connect various parts of the entire terminal. By running or executing software programs and/or modules stored in the memory 602, and calling data stored in the memory 602, Perform various functions of the terminal and process data to monitor the terminal as a whole.
  • the processor 601 may include one or more processing cores; in some embodiments, the processor 601 may integrate an application processor and a modem processor, where the application processor mainly processes the operating system, user For interface and application programs, the modem processor mainly deals with wireless communication. It can be understood that the foregoing modem processor may not be integrated into the processor 601.
  • the memory 602 may be used to store software programs and modules.
  • the processor 601 executes various functional applications and data processing by running the software programs and modules stored in the memory 602.
  • the memory 602 may mainly include a program storage area and a data storage area.
  • the program storage area may store an operating system, an application program required by at least one function (such as a sound playback function, an image playback function, etc.), etc.; Data created by the use of the terminal, etc.
  • the memory 602 may include a high-speed random access memory, and may also include a non-volatile memory, such as at least one magnetic disk storage device, a flash memory device, or other volatile solid-state storage devices.
  • the memory 602 may further include a memory controller to provide the processor 601 with access to the memory 602.
  • the terminal also includes a power supply 603 for supplying power to various components.
  • the power supply 603 may be logically connected to the processor 601 through a power management system, so that functions such as charging, discharging, and power consumption management can be managed through the power management system.
  • the power supply 603 may also include any components such as one or more DC or AC power supplies, a recharging system, a power failure detection circuit, a power converter or inverter, and a power status indicator.
  • the terminal may further include an input module 604, which may be used to receive inputted digital or character information, and generate keyboard, mouse, joystick, optical or trackball signal input related to user settings and function control.
  • an input module 604 which may be used to receive inputted digital or character information, and generate keyboard, mouse, joystick, optical or trackball signal input related to user settings and function control.
  • the terminal may also include a communication module 605.
  • the communication module 605 may include a wireless module.
  • the terminal may perform short-distance wireless transmission through the wireless module of the communication module 605, thereby providing users with wireless broadband Internet access.
  • the communication module 605 can be used to help users send and receive emails, browse webpages, and access streaming media.
  • the terminal may also include a display unit and the like.
  • the present application also provides a terminal, including a memory and a processor, and computer-readable instructions are stored in the memory.
  • a terminal including a memory and a processor
  • computer-readable instructions are stored in the memory.
  • the processor executes the implementation of the application. The steps in any of the data transmission methods provided in the example.
  • the present application also provides one or more non-volatile storage media storing computer-readable instructions.
  • the computer-readable instructions are executed by one or more processors, one or more The processor executes the steps in any data transmission method provided in the embodiments of the present application.
  • Non-volatile memory may include read-only memory (Read-Only Memory, ROM), magnetic tape, floppy disk, flash memory, or optical storage.
  • Volatile memory may include random access memory (RAM) or external cache memory.
  • RAM can be in various forms, such as static random access memory (Static Random Access Memory, SRAM) or dynamic random access memory (Dynamic Random Access Memory, DRAM), etc.

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Abstract

本申请实施例公开了一种数据传输方法、装置、终端、存储介质和系统；本申请实施例可以获取音频数据以及传输状态信息；基于传输状态信息确定压缩系数和冗余系数；根据压缩系数对音频数据进行时域压缩处理，得到压缩数据；根据冗余系数对压缩数据进行信道编码，得到传输数据包；发送传输数据包。

Description

数据传输方法、装置、终端、存储介质和系统
本申请要求于2020年02月10日提交中国专利局，申请号为2020100852933，申请名称为“数据传输方法、装置、终端、存储介质和系统”的中国专利申请的优先权，其全部内容通过引用结合在本申请中。
技术领域
本申请涉及数据传输领域，具体涉及一种数据传输方法、装置、终端、存储介质和系统。
背景技术
互联网是一种很容易出现网络波动和堵塞的传输网络，对于对网络要求较高的应用，特别是互联网音频应用，目前数据传输的方法很容易出现网络波动导致的丢包现象，即音频数据包漏传、误传现象；比如，语音直播、语音通话、语音广播等互联网音频应用对网络的稳定性和带宽具有较高的要求，否则接收端接收到的音频可能会出现不连贯、卡顿等情况。
目前的数据传输方法为了解决丢包所导致的数据漏传、误传，会大量地重复发送冗余数据来减少丢包带来的影响，然而该方法会占用大量的网络资源，并且需要耗费大量时间和计算资源去处理、发送这些冗余的数据，因此，目前数据传输的方法效率低下。
发明内容
根据本申请提供的各种实施例，本申请提供一种数据传输方法、装置、终端、存储介质和系统。
本申请实施例提供一种数据传输方法，适用于发送端，包括：
获取音频数据以及传输状态信息；
基于所述传输状态信息确定压缩系数和冗余系数；
根据所述压缩系数对所述音频数据进行时域压缩处理，得到压缩数据；
根据所述冗余系数对所述压缩数据进行信道编码，得到传输数据包；
发送所述传输数据包。
本申请实施例提供一种数据传输方法，适用于接收端，包括：
获取传输数据包，所述传输数据包包括冗余数据和压缩系数；
基于所述传输数据包确定当前时刻的传输状态信息；
发送所述当前时刻的传输状态信息；
根据所述冗余数据对所述传输数据包进行信道解码，得到待复原数据；
根据所述压缩系数对所述待复原数据进行时域扩张处理，得到复原数据。
本申请实施例还提供一种数据传输装置，适用于发送端，包括：
第一获取单元，用于获取音频数据以及传输状态信息；
系数单元，用于基于所述传输状态信息确定缩放系数和冗余系数；
压缩单元，用于根据所述缩放系数对所述音频数据进行时域压缩处理，得到压缩数据；
编码单元，用于根据所述冗余系数对所述压缩数据进行信道编码，得到传输数据包；
第一发送单元，用于发送所述传输数据包。
本申请实施例还提供一种数据传输装置，适用于接收端，包括：
第二获取单元，用于获取传输数据包，所述传输数据包包括冗余数据和缩放系数；
占用单元，用于基于所述传输数据包确定当前时刻的传输状态信息；
第二发送单元，用于发送所述当前时刻的传输状态信息；
解码单元，用于根据所述冗余数据对所述传输数据包进行信道解码，得到待复原数据；
扩张单元，用于根据所述缩放系数对所述待复原数据进行时域扩张处理，得到复原数据。
本申请实施例还提供一种终端，包括存储器和处理器，所述存储器中存储有计算机可读指令，所述计算机可读指令被所述处理器执行时，使得所述处理器执行本申请实施例所提供的任一种数据传输方法中的步骤。
本申请实施例还提供一个或多个存储有计算机可读指令的非易失性存储介质，所述计算机可读指令被一个或多个处理器执行时，使得一个或多个处理器执行本申请实施例所提供的任一种数据传输方法中的步骤。
本申请实施例还提供一种数据传输系统，所述数据传输系统包括发送端和接收端，其中：
所述发送端用于获取音频数据以及获取接收端发送的传输状态信息，基于所述传输状态信息确定缩放系数和冗余系数，根据所述缩放系数对所述音频数据进行时域压缩处理，得到压缩数据，根据所述冗余系数对所述压缩数据进行信道编码，得到传输数据包，发送所述传输数据包至接收端。
所述接收端用于获取发送端发送的传输数据包，所述传输数据包包括冗余数据和缩放系数，基于所述传输数据包确定当前时刻的传输状态信息，发送所述当前时刻的传输状态信息到发送端，对所述传输数据包进行信道解码，得到待复原数据，根据所述缩放系数对所述待复原数据进行时域扩张处理，得到复原数据。
本申请的一个或多个实施例的细节在下面的附图和描述中提出。本申请的其它特征、目的和优点将从说明书、附图以及权利要求书变得明显。
附图说明
为了更清楚地说明本申请实施例中的技术方案，下面将对实施例描述中所需要使用的附图作简单地介绍，显而易见地，下面描述中的附图仅仅是本申请的一些实施例，对于本领域技术人员来讲，在不付出创造性劳动的前提下，还可以根据这些附图获得其他的附图。
图1a是本申请实施例提供的数据传输系统的场景示意图；
图1b是本申请实施例提供的数据传输方法的第一种流程示意图；
图2是本申请实施例提供的数据传输方法的第二种流程示意图；
图3是本申请实施例提供的数据传输系统的流程示意图
图4是本申请实施例提供的数据传输装置的第一种结构示意图；
图5是本申请实施例提供的数据传输装置的第二种结构示意图；
图6是本申请实施例提供的电子设备的结构示意图。
具体实施方式
下面将结合本申请实施例中的附图，对本申请实施例中的技术方案进行清楚、完整地描述，显然，所描述的实施例仅仅是本申请一部分实施例，而不是全部的实施例。基于本申请中的实施例，本领域技术人员在没有作出创造性劳动前提下所获得的所有其他实施例，都属于本申请保护的范围。
本申请实施例提供一种数据传输方法、装置、终端、存储介质和系统。
其中，该数据传输装置可以集成在电子设备中，适用于发送端和接收端；该发送端和接收端可以为同一电子设备，也可以为不同的电子设备，当发送端和接收端为不同的电子设备时，发送端和接收端可以为同一种类型的电子设备，也可以为不同类型的电子设备。
比如，该电子设备可以为终端、服务器等设备；终端可以为手机、平板电脑、智能蓝牙设备、笔记本电脑、或者个人电脑(Personal Computer，PC)等设备；服务器可以是单一服务器，也可以是由多个服务器组成的服务器集群。
在一些实施例中，该数据传输装置还可以集成在多个电子设备中，比如，数据传输装置可以集成在多个服务器中，由多个服务器来实现本申请的数据传输方法。
在一些实施例中，服务器也可以以终端的形式来实现。
例如，参考图1a，该数据传输系统中可以包括作为发送端的手机A，和作为接收端的手机B。
其中，手机A可以获取音频数据以及从手机B获取传输状态信息，并基于传输状态信息确定压缩系数和冗余系数，再根据压缩系数对音频数据进行时域压缩处理，得到压缩数据，然后根据冗余系数对压缩数据进行信道编码，得到传输数据包，最后发送传输数据包至手机B。
其中，手机B可以从手机A获取传输数据包和压缩系数，然后基于传输数据包确定当前时刻的传输状态信息，再发送当前时刻的传输状态信息至手机A，然后对传输数据包进行信道解码，得到待复原数据，最后根据压缩系数对待复原数据进行时域扩张处理，得到复原数据。
以下分别进行详细说明。需说明的是，以下实施例的序号不作为对实施例优选顺序的限定。
在一些实施例中，提供了一种基于数据传输的数据传输方法，如图1b所示，以该方法应用于发送端为例进行说明，该数据传输方法的流程可以如下：
101、获取音频数据以及传输状态信息。
其中，音频数据是指待发送的电子数据信息，该电子数据信息可以表现为多种数据类型，比如，可以是纯音频数据、包含音频的视频数据等等。
传输状态信息是指可以反映数据传输状态的相关信息，比如，传输状态信息可以包括信道的利用率、带宽、丢包率、发送端的冗余率、发送码率、接收码率、传输速率、信噪比、信道增益、噪声功率，等等。
在本实施例中，可以通过多种方法获取音频数据，比如，通过网络从数据库中获取音频数据，也可以通过传感器捕捉录制音频数据，还可以由用户输入得到音频数据，还可以在本地读取音频数据，等等。
比如，在一些实施例中，可以利用搭载的录音设备录制得到用户的音频数据。
在本实施例中，可以通过多种方法获取传输状态信息，比如，通过网络从数据库中获取传输状态信息，也可以通过传感器采集传输状态信息，还可以通过网络从获取接收端发送的传输状态信息，还可以在本地读取传输状态信息，等等。
比如，在一些实施例中，可以通过网络与接收端通信，从而获取接收端发送的传输状态信息。
102、基于传输状态信息确定压缩系数和冗余系数。
其中，压缩系数是是描述压缩性大小的物理量；冗余系数是描述冗余数据占发送数据包大小的物理量。
接收端可以利用冗余数据来检查、纠正其接收到的数据信息，冗余数据可以是纠错码、查错码、传输数据的数据片段，等等。
比如，在一些实施例中，压缩系数是0.9，则可以描述数据被压缩至原有大小的0.9倍。
比如，在一些实施例中，冗余系数是0.3，则可以描述发送数据中，有30％是冗余的数据，其余70％是有效的数据。
在一些实施例中，为了更快、更容易地分析信道当前的工作状态，并在丢包率升高时提高冗余率，在信道出现拥塞时降低压缩系数以便减小发送数据包的大小，传输状态信息中可以包括接收数量，那么步骤102可以包括以下步骤：
对发送的传输数据包进行统计，得到发送码率、发送数量；
根据发送数量和接收数量计算丢包率；
基于丢包率确定冗余系数；
基于丢包率和发送码率确定压缩系数。
可选地，本实施例是对预设历史时间端内所发送的传输数据包进行统计，从而得到发送码率、发送数量。
其中，发送码率(Sender Bit Rate)是指发送端在单位时间内发送的有效数据和冗余数据的总大小，以比特(Byte)为单位。
发送数量是指发送端在单位时间内发送的数据包的数量。
接收数量是指接收端在单位时间内接收的数据包的数量。
丢包率(Loss Rate)是指单位时间内丢失的数据包数量与发送数量的比值。当网络不稳定时，丢包率可能会升高，即信道丢失数据的概率会升高，此时，可能会造成视频马赛克现象、局部变形、图像模糊、频繁刷新、音视频不同步、图像静止、延迟、音频中断等问题。丢包率越高，对音视频通话等数据传输应用的影响效果越明显。
在本实施例中，为了降低丢包率，可以重复传输有效数据，即，传输冗余数据。本实施例可以修改冗余系数，使得发送端发送一定数量的冗余数据，以保证接收端接收到的信息完整、正确。
比如，在本实施例中，若发送数量为100，接收数量为75，则丢包率为25％，即丢失25个发送数据，此时，判断信道的丢包率比较严重，则修改当前的冗余系数为0.25，即，假设要再次发送100个发送数据，其中冗余数据占25个，有效数据占75个。在一些实施例中，当预设的历史时间段内丢包率和发送码率呈正相关，即发送的数据包中冗余数据越来越大，且丢包率也越来越 大，则可以判断信道当前处于工作上限，为了防止继续重复发送数据对信道造成网络崩溃的情况，本实施例可以设置压缩系数来降低发送数据的大小，使得发送码率降低，从而减小对信道的压力。
可选地，在一些实施例中，步骤“基于丢包率和发送码率确定压缩系数”可以包括以下步骤：
a.分别对丢包率和发送码率进行统计，得到丢包率对应的第一变化趋势和发送码率对应的第二变化趋势，以及丢包率和发送码率之间的相关性；
b.当第一变化趋势和第二变化趋势均为上升趋势，且丢包率和发送码率之间的相关性呈正相关时，根据丢包率和发送码率确定压缩系数。
即当信道的网络带宽达到上限时，丢包率和发送码率呈上升趋势且相互正相关，此时可以通过修改压缩系数来降低发送码率，从而减轻信道的压力，维持信道稳定性。
在一些实施例中，可以在发送传输数据包时发送该压缩系数，比如，通过网络同时向接收端发送数据传输包和压缩系数。
103、根据压缩系数对音频数据进行时域压缩处理，得到压缩数据。
时域压缩(Time Domain Data Compression)是指在时间域对电子数据中的一些数据进行删除、变换等，从而达到压缩的效果。
时域压缩方法具有多种，比如，时域压缩方法包括自适应差分脉冲编码调制(Adaptive Differential Pulse Code Modulation，ADPCM)、线性预测编码(linear predictive coding，LPC)、码激励线性预测编码(Code Excited Linear Prediction，CELP)、重叠叠加算法(Overlap-and-Add，OLA)、等等；其中，重叠叠加算法可以包括同步波形叠加法(Synchronized Overlap-Add，SOLA)、基音同步叠加法(Pitch Synchronized Overlap-Add，PSOLA)、波形相似叠加法(waveform similarity overlap-and-add，WSOLA)，等等。
比如，在一些实施例中，为了解决压缩数据不连续，基音断裂等情况，可以采用基音同步叠加法进行时域压缩，可选地，步骤103可以包括以下步骤：
对音频数据进行基音分析，确定音频数据对应的基音点；
根据基音点对音频数据进行数据采样，得到多个子音频数据；
根据压缩系数从多个子音频数据中筛选出目标子音频数据；
对目标子音频数据进行合成处理，得到压缩数据。
其中，基音是一种决定语音韵律的主要时域参数，时域基音同步叠加法的核心是基音同步，首先对音频数据中的基音进行标注，例如，对浊音进行标注；然后，根据基音点对音频数据进行数据采样，得到多个子音频数据，对子音频数据进行一系列的插入、删除和修改，合成得到压缩数据。
比如，在一些实施例中，可以采用重叠叠加算法进行时域压缩，可选地，步骤103可以包括以下步骤：
根据压缩系数确定采样窗口；
基于采样窗口对音频数据进行数据采样，得到子音频数据；
对子音频数据进行合成处理，得到压缩数据。
在本实施例中，可以根据压缩系数决定采样窗口的系数，比如，步长、窗口大小，等等；然后，在音频数据中平滑移动该采样窗口，每经过一定的步长则对采样窗口中的数据进行采样，得到子音频数据；最后，叠加所有的子音频数据，得到压缩数据。
其中，在一些实施例中，为了解决压缩数据不连续等情况导致的语音质量较差，可以采用波形相似叠加法进行时域压缩，可选地，步骤“对子音频数据进行合成处理，得到压缩数据”可以包括以下步骤：
计算子音频数据之间的波形互相关系数；
根据波形互相关系数确定波形相似的子音频数据；
对波形相似的子音频数据进行波形叠加处理，得到压缩数据。
其中，波形互相关系数可以描述两个波形之间的相似程度，在本实施例中，对于波形相似的两个子音频数据可以进行叠加，最后得到压缩数据。
104、根据冗余系数对压缩数据进行信道编码，得到传输数据包。
数字信号在传输中往往由于各种原因，使得在传送的数据流中产生误码，从而使接收端产生图象跳跃、扭曲、不连续、出现马赛克等现象。通过信道编码对数据进行相应的处理，可以使信道具有一定的纠错能力和抗干扰能力，极大地避免了数据传输过程中误码的发生。
其中，信道编码包括多种编码方式，比如，纠错编码、查错编码，等等。其中，纠错码可以包括前向纠错编码(Forward Error Correction，FEC)、里所码(Reed-solomon codes，RS)、卷积码、Turbo码等等。
在本实施例中，可以根据冗余系数生成压缩数据对应的冗余数据，再将这些冗余数据和压缩数据进行信道编码、打包，得到传输数据包。
在一些实施例中，可以对所述冗余系数进行信道编码，得到压缩系数标识符，再根据所述压缩系数标识符对所述压缩数据进行信道编码，得到传输数据包。
其中，压缩系数标识符是携带压缩系数信息的标识符，可以用于表示压缩系数的大小。
比如，在一些实施例中，可以将压缩系数对应的压缩系数标识符作为该传输数据包的包头。
105、发送传输数据包。
最后，可以将传输数据包发送到接收端。
在发送传输数据包时，可以在一段时间内发送压缩系数，也可以发送含有以压缩系数标识符为包头的数据包，等等。本申请实施例提供的数据传输方案可以应用在各种数据传输场景中，比如，在音频传输场景中，特别是VoIP(Voice over Internet Protocol，基于IP的语音传输)、语音广播、音视频直播等对丢包率、延时率要求较高的场景中，本申请实施例可以实时地监督信道的工作情况，并根据工作情况控制冗余数据的发送以及控制音频压缩的效果，故采用本申请实施例提供的方案，能够在保证音频效果的情况下，更高效地进行数据传输。
由上可知，本申请实施例可以获取音频数据以及传输状态信息；基于传输状态信息确定压缩系数和冗余系数；根据压缩系数对音频数据进行时域压缩处理，得到压缩数据；根据冗余系数对压缩数据进行信道编码，得到传输数据包；发送传输数据包。
由此，本方案可以通过对传输状态信息进行分析，实时地判断信道当前的工作情况，当信道拥塞时，通过修改压缩系数可以改善信道拥塞等情况可以降 低数据的延时率和丢包率，提高信道的稳定性；当信道丢包率高时，通过修改冗余系数可以改善丢包等情况，从而在保证数据传输的正确、完整。本方案提高了信道的利用率，使得数据传输的速度更快、更稳定，由此，本方案可以提升数据传输的效率。
在一些实施例中，提供了一种基于数据传输的数据传输方法，如图2所示，以该方法应用于接收端为例进行说明，该数据传输方法的流程可以如下：
201、获取传输数据包和压缩系数。
本实施例可以通过信道获取发送端发送的传输数据包。
在一些实施例中，可以在获取发送端发送的传输数据包的同时，获取发送端发送的压缩系数。
在一些实施例中，传输数据包中可以包括压缩系数标识符，通过对该压缩系数标识符进行识别，可以获得压缩系数。
比如，在在一些实施例中，传输数据包的包头为压缩系数标识符，通过读取该包头，即可得到压缩系数。
202、基于传输数据包确定当前时刻的传输状态信息。
在一些实施例中，除了在发送端分析信道的状态，还可以在接收端分析信道的状态。
故在一些实施例中，步骤202可以包括以下步骤：
对接收的传输数据包进行统计，得到接收数量；
根据发送数量和接收数量计算丢包率；
确定当前时刻的传输状态信息，传输状态信息包括丢包率。
其中，接受数量、发送数量、丢包率、传输状态信息可以参考步骤102和步骤103中的叙述。
203、发送当前时刻的传输状态信息。
在本实施例中，可以向发送终端发送当前时刻的传输状态信息。
可选地，可以通过多种方法向发送终端发送当前时刻的传输状态信息，例如，通过网络向发送终端发送当前时刻的传输状态信息、通过存储介质向发送终端发送当前时刻的传输状态信息，等等。
204、对传输数据包进行信道解码，得到待复原数据。
其中，信道解码对应信道编码，目的是还原由信道编码处理的数据，使其恢复至未进行信道编码时的状态。
在一些实施例中，传输数据包中可以包括冗余数据和传输数据，通过冗余数据可以对传输数据进行信道编码，从而实现对传输数据的检查、纠错、补漏等，最终得到待复原数据。
对应信道编码的方法，信道解码的方法也具有多种，比如，前向纠错编码(Forward Error Correction，FEC)、里所码(Reed-solomon codes，RS)、卷积码、Turbo码等等。
205、根据压缩系数对待复原数据进行时域扩张处理，得到复原数据。
时域扩张(Time Domain Data Decompression)是指在时间域内对电子数据进行数据修改、增加、插入等，从而达到解压缩的效果。
时域扩张和时域压缩的方法类似，比如，同步波形叠加法、基音同步叠加法、波形相似叠加法，等等。
比如，在一些实施例中，为了解决扩张数据不连续，可以采用重叠叠加算法进行时域扩张，可选地，时域扩张的方法和时域压缩的方法类类似，步骤103可以包括以下步骤：
根据压缩系数确定对应的解压系数；
根据解压系数确定采样窗口；
基于采样窗口对待复原数据进行数据采样，得到子待复原数据；
对子待复原数据进行合成处理，得到复原数据。
其中，压缩系数和解压系数相互对应，比如，在一些实施例中，压缩系数x对应的解压系数为1/x；比如，在一些实施例中，压缩系数x对应的解压系数为1-x。
类似地，在一些实施例中，为了解决压缩数据不连续、语音质量差等情况，可以采用波形相似叠加法进行时域扩张，步骤“对子待复原数据进行合成处理，得到复原数据”可以包括以下步骤：
计算子待复原数据之间的波形互相关系数；
根据波形互相关系数确定波形相似的子待复原数据；
对波形相似的子待复原数据进行波形叠加处理，得到复原数据。
其中，波形互相关系数、波形叠加处理可以参考步骤103。
本申请实施例提供的数据传输方案可以应用在各种数据传输场景中，比如，在音频传输场景中，特别是VoIP、语音广播、音视频直播等对丢包率、延时率要求较高的场景中，本申请实施例可以实时地向发送端发送信道的工作情况，以便发送端对信道进行监督控制冗余数据的发送以及控制音频压缩的效果，故采用本申请实施例提供的方案，能够在保证音频效果的情况下，更高效地进行数据传输。
由上可知，本申请实施例可以获取传输数据包，传输数据包包括冗余数据和压缩系数；基于传输数据包确定当前时刻的传输状态信息；发送当前时刻的传输状态信息；根据冗余数据对传输数据包进行信道解码，得到待复原数据；根据压缩系数对待复原数据进行时域扩张处理，得到复原数据。
由此，本方案可以统计并发送传输状态信息，以便发送端对信道的状态进行分析，从而修改压缩系数以改善信道拥塞等情况、修改冗余系数以改善丢包等情况，最终保证了数据传输的正确、完整的，并提高了信道的利用率，使得数据传输的速度更快、更稳定，由此，本方案可以提升数据传输方法的效率。
为了抵抗网络不稳定导致的丢包，减轻接收端声音的卡顿和不连贯的问题，可以采用多种信道编码方法进行丢包补偿，比如，前向纠错法(forward error correction，FEC)、丢包隐藏法(Packet Loss Concealment，PLC)、自动重传请求法(Automatic Repeat Request，ARQ)，等等。
其中，FEC技术是通过冗余编码生成冗余信息来进行丢包补偿，其抗丢包能力与其所用信道的带宽呈正比；可选地，当接收端出现丢包时，FEC技术可以利用冗余信息进行丢包恢复，冗余信息越多抗丢包能力越强，同时占用带宽越高，然而带宽占用越多会引发网络质量变差从而引发更多丢包。
在本实施例中，提供了一种基于数据传输的数据传输系统，该数据传输系统中可以包括发送端和接收端，以下将以FEC技术进行发送端和接收端之间的音频传输为例，对本申请实施例的方法进行详细说明。
如图3所示，一种数据传输系统流程如下：
301、发送端获取音频数据以及从接收端获取传输状态信息。
302、发送端基于传输状态信息确定压缩系数和冗余系数。
在本实施例中，可以分析音频带宽占用增加后的网络丢包率状况，即冗余率提升后的网络丢包率状况。
可选地，可以统计丢包率和发送端的发送码率。
在网络带宽没达到上限时，发送码率的提升不会对丢包率有必然影响，而当网络带宽达到上限时，发送码率提升后丢包率会随着提升，当该现象比较稳定时则可判为网络带宽达到上限状态，则可启动本方案进行数据传输。
303、发送端根据压缩系数对音频数据进行时域压缩处理，得到压缩音频。
在本实施例中，可以采用WSOLA来将原始语音信号以长度为L进行分帧，再以帧为单位进行合成。
其中，假设压缩系数为α，为了克服帧合成过程中出现频谱断裂、相位不连续等导致的杂音问题，合成过程可以是在原始信号的τ(L k)采样点处采样。
其中，采样点τ(L k)的计算方法如下：
L k＝kL,τ(L k)＝αL k
在该采样点的邻域[-Δmax,Δmax]内移动，其中，邻域[-Δmax,Δmax]可以由技术人员设置。
然后，寻找与分解后的第k帧信号波形最相关的波形，将其确定为合成帧的起始位置，最后通过汉宁窗加窗后进行叠加处理，得到时域压缩或扩张的新语音信号。
由于语音的时域压缩对带宽节省的效果比较明显，所以能有效改善FEC冗余度提升对信道带宽的影响。
304、发送端根据冗余系数对压缩音频进行信道编码，得到传输数据包。
305、发送端向接收端发送传输数据包，以及，接收端从发送端获取传输数据包，传输数据包包括冗余数据和压缩系数。
306、接收端基于传输数据包确定当前时刻的传输状态信息。
307、接收端向发送端发送当前时刻的传输状态信息。
308、接收端根据冗余数据对传输数据包进行信道解码，得到待复原音频。
309、接收端根据压缩系数对待复原音频进行时域扩张处理，得到复原音频。
在本实施例中，假设压缩系数为α，则此时扩张系数可以为1/α，
以上流程可以参考步骤101～105，以及步骤201～205。
由上可知，在本申请实施例中，数据传输系统可以包括发送端和接收端，其中，发送端可以获取音频数据以及从接收端获取传输状态信息，并基于传输状态信息确定压缩系数和冗余系数，再根据压缩系数对音频数据进行时域压缩处理，得到压缩音频，然后根据冗余系数对压缩音频进行信道编码，得到传输数据包，最后向接收端发送传输数据包；其中，接收端可以从发送端获取传输数据包，传输数据包包括冗余数据和压缩系数，再基于传输数据包确定当前时刻的传输状态信息，然后向发送端发送当前时刻的传输状态信息，再根据冗余数据对传输数据包进行信道解码，得到待复原音频，最后根据压缩系数对待复原音频进行时域扩张处理，得到复原音频。
由此，本方案可以在保证音频频率不变的前提下，对音频的时域尺度进行修改，从而在不明显降低音频质量的同时延长或缩短音频持续时间，由于在时域尺度进行修改的过程中能够保证语音的基音频率不被破坏，因此被修改的语音音色和音调都能很好地得到保护。
此外，当检测到冗余率越高而丢包率越大时，本方案先对语音信号源做时域尺度的压缩，从而降低发送码率，然后在接收端解码后再使用与发送端相同的时域尺度修改比率进行时域扩张，进而恢复原有信号，故本方案不仅在信道丢包的情况下保证了音频质量，还可以降低信道的工作压力，实时地平衡了音频质量和信道稳定性，使得信道更加稳定，进一步使得信道传输的流畅度，从而提升数据传输的效率。
应该理解的是，本申请各实施例中的各个步骤并不是必然按照步骤标号指示的顺序依次执行。除非本文中有明确的说明，这些步骤的执行并没有严格的顺序限制，这些步骤可以以其它的顺序执行。而且，各实施例中至少一部分步骤可以包括多个子步骤或者多个阶段，这些子步骤或者阶段并不必然是在同一 时刻执行完成，而是可以在不同的时刻执行，这些子步骤或者阶段的执行顺序也不必然是依次进行，而是可以与其它步骤或者其它步骤的子步骤或者阶段的至少一部分轮流或者交替地执行。
为了更好地实施以上方法，本申请实施例还提供一种数据传输装置，该数据传输装置可以集成在电子设备中，该电子设备可以为终端、服务器等设备。
比如，在本实施例中，将以数据传输装置集成在数据传输端为例，对本申请实施例的方法进行详细说明。
例如，如图4所示，该数据传输装置可以包括第一获取单元401、系数单元402、压缩单元403、编码单元404以及第一发送单元405，如下：
(一)第一获取单元401：
第一获取单元401可以用于获取音频数据以及传输状态信息。
(二)系数单元402：
系数单元402可以用于基于传输状态信息确定缩放系数和冗余系数。
在一些实施例中，传输状态信息包括接收数量，故系数单元402可以包括统计子单元、丢包率子单元、冗余系数子单元以及压缩系数子单元，如下：
(1)统计子单元：
统计子单元可以用于对发送的传输数据包进行统计，得到发送码率、发送数量。
(2)丢包率子单元：
丢包率子单元可以用于根据发送数量和接收数量计算丢包率。
(3)冗余系数子单元：
冗余系数子单元可以用于基于丢包率确定冗余系数。
(4)压缩系数子单元：
压缩系数子单元可以用于基于丢包率和发送码率确定压缩系数。
在一些实施例中，压缩系数子单元可以用于：
分别对丢包率和发送码率进行统计，得到丢包率对应的第一变化趋势和发送码率对应的第二变化趋势，以及丢包率和发送码率之间的相关性；
当第一变化趋势和第二变化趋势均为上升趋势，且丢包率和发送码率之间 的相关性呈正相关时，根据丢包率和发送码率确定压缩系数。
(三)压缩单元403：
压缩单元403可以用于根据缩放系数对音频数据进行时域压缩处理，得到压缩数据。
在一些实施例中，压缩单元403可以用于：
对音频数据进行基音分析，确定音频数据对应的基音点；
根据基音点对音频数据进行数据采样，得到多个子音频数据；
根据压缩系数从多个子音频数据中筛选出目标子音频数据；
对目标子音频数据进行合成处理，得到压缩数据。
在一些实施例中，压缩单元403可以包括压缩窗口子单元、压缩采样子单元以及压缩子单元，如下：
(1)压缩窗口子单元：
压缩窗口子单元可以用于根据压缩系数确定采样窗口。
(2)压缩采样子单元：
压缩采样子单元可以用于基于采样窗口对音频数据进行数据采样，得到子音频数据。
(3)压缩子单元：
压缩子单元可以用于对子音频数据进行合成处理，得到压缩数据。
在一些实施例中，压缩子单元可以用于：
计算子音频数据之间的波形互相关系数；
根据波形互相关系数确定波形相似的子音频数据；
对波形相似的子音频数据进行波形叠加处理，得到压缩数据。
(四)编码单元404：
编码单元404可以用于根据冗余系数对压缩数据进行信道编码，得到传输数据包。
在一些实施例中，编码单元404可以用于：
对所述冗余系数进行信道编码，得到压缩系数标识符；
根据所述压缩系数标识符对所述压缩数据进行信道编码，得到传输数据 包。
(五)第一发送单元405：
第一发送单元405可以用于发送传输数据包。
实施时，以上各个单元可以作为独立的实体来实现，也可以进行任意组合，作为同一或若干个实体来实现，以上各个单元的实施可参见前面的方法实施例，在此不再赘述。
由上可知，本实施例的数据传输装置由第一获取单元获取音频数据以及传输状态信息；由系数单元基于传输状态信息确定缩放系数和冗余系数；由压缩单元根据缩放系数对音频数据进行时域压缩处理，得到压缩数据；由编码单元根据冗余系数对压缩数据进行信道编码，得到传输数据包；由第一发送单元发送传输数据包。
由此，本申请实施例可以提高数据传输的效率。
为了更好地实施以上方法，本申请实施例还提供一种数据传输装置，该数据传输装置可以集成在接收端中。。
比如，在本实施例中，将以接收端为手机为例，对本申请实施例的方法进行详细说明。
例如，如图5所示，该数据传输装置可以包括第二获取单元501、占用单元502、第二发送单元503、解码单元504以及扩张单元505，如下：
(一)第二获取单元501：
第二获取单元501可以用于获取传输数据包和缩放系数。
(二)占用单元502：
占用单元502可以用于基于传输数据包确定当前时刻的传输状态信息。
在一些实施例中，占用单元502可以用于：
对接收的传输数据包进行统计，得到接收数量；
根据发送数量和接收数量计算丢包率；
确定当前时刻的传输状态信息，传输状态信息包括丢包率。
(三)第二发送单元503：
第二发送单元503可以用于发送当前时刻的传输状态信息。
(四)解码单元504：
解码单元504可以用于对传输数据包进行信道解码，得到待复原数据。
(五)扩张单元505：
扩张单元505可以用于根据缩放系数对待复原数据进行时域扩张处理，得到复原数据。
在一些实施例中，扩张单元505可以包括解压系数子单元、解压窗口子单元、解压采样子单元以及复原子单元，如下：
(1)解压系数子单元：
解压系数子单元可以用于根据压缩系数确定对应的解压系数。
(2)解压窗口子单元：
解压窗口子单元可以用于根据解压系数确定采样窗口。
(3)解压采样子单元：
解压采样子单元可以用于基于采样窗口对待复原数据进行数据采样，得到子待复原数据。
(4)复原子单元：
复原子单元可以用于对子待复原数据进行合成处理，得到复原数据。
在一些实施例中，复原子单元可以用于：
计算子待复原数据之间的波形互相关系数；
根据波形互相关系数确定波形相似的子待复原数据；
对波形相似的子待复原数据进行波形叠加处理，得到复原数据。
实施时，以上各个单元可以作为独立的实体来实现，也可以进行任意组合，作为同一或若干个实体来实现，以上各个单元的实施可参见前面的方法实施例，在此不再赘述。
由上可知，本实施例的数据传输装置由第二获取单元获取传输数据包和缩放系数；由占用单元基于传输数据包确定当前时刻的传输状态信息；由第二发送单元发送当前时刻的传输状态信息；由解码单元对传输数据包进行信道解码，得到待复原数据；由扩张单元根据缩放系数对待复原数据进行时域扩张处理，得到复原数据。
由此，本申请实施例可以提升数据传输。
本申请实施例还提供一种电子设备，该电子设备可以为终端、服务器等设备。
在本实施例中，将以本实施例的电子设备是终端为例进行详细描述，比如，如图6所示，其示出了本申请实施例所涉及的终端的结构示意图，例如：
该终端可以包括一个或者一个以上处理核心的处理器601、一个或一个以上存储有计算机可读存储介质的存储器602、电源603、输入模块604以及通信模块605等部件。本领域技术人员可以理解，图4中示出的终端结构并不构成对终端的限定，可以包括比图示更多或更少的部件，或者组合某些部件，或者不同的部件布置。其中：
处理器601是该终端的控制中心，利用各种接口和线路连接整个终端的各个部分，通过运行或执行存储在存储器602内的软件程序和/或模块，以及调用存储在存储器602内的数据，执行终端的各种功能和处理数据，从而对终端进行整体监控。在一些实施例中，处理器601可包括一个或多个处理核心；在一些实施例中，处理器601可集成应用处理器和调制解调处理器，其中，应用处理器主要处理操作系统、用户界面和应用程序等，调制解调处理器主要处理无线通信。可以理解的是，上述调制解调处理器也可以不集成到处理器601中。
存储器602可用于存储软件程序以及模块，处理器601通过运行存储在存储器602的软件程序以及模块，从而执行各种功能应用以及数据处理。存储器602可主要包括存储程序区和存储数据区，其中，存储程序区可存储操作系统、至少一个功能所需的应用程序(比如声音播放功能、图像播放功能等)等；存储数据区可存储根据终端的使用所创建的数据等。此外，存储器602可以包括高速随机存取存储器，还可以包括非易失性存储器，例如至少一个磁盘存储器件、闪存器件、或其他易失性固态存储器件。相应地，存储器602还可以包括存储器控制器，以提供处理器601对存储器602的访问。
终端还包括给各个部件供电的电源603，在一些实施例中，电源603可以通过电源管理系统与处理器601逻辑相连，从而通过电源管理系统实现管理充 电、放电、以及功耗管理等功能。电源603还可以包括一个或一个以上的直流或交流电源、再充电系统、电源故障检测电路、电源转换器或者逆变器、电源状态指示器等任意组件。
该终端还可包括输入模块604，该输入模块604可用于接收输入的数字或字符信息，以及产生与用户设置以及功能控制有关的键盘、鼠标、操作杆、光学或者轨迹球信号输入。
该终端还可包括通信模块605，在一些实施例中通信模块605可以包括无线模块，终端可以通过该通信模块605的无线模块进行短距离无线传输，从而为用户提供了无线的宽带互联网访问。比如，该通信模块605可以用于帮助用户收发电子邮件、浏览网页和访问流式媒体等。
尽管未示出，终端还可以包括显示单元等。
在一些实施例中，本申请还提供一种终端，包括存储器和处理器，存储器中存储有计算机可读指令，计算机可读指令被所述处理器执行时，使得所述处理器执行本申请实施例所提供的任一种数据传输方法中的步骤。
在一些实施例中，本申请还提供了一个或多个存储有计算机可读指令的非易失性存储介质，所述计算机可读指令被一个或多个处理器执行时，使得一个或多个处理器执行本申请实施例所提供的任一种数据传输方法中的步骤。
本领域普通技术人员可以理解实现上述实施例方法中的全部或部分流程，是可以通过计算机可读指令来指令相关的硬件来完成，所述的计算机可读指令可存储于一非易失性计算机可读取存储介质中，该计算机可读指令在执行时，可包括如上述各方法的实施例的流程。其中，本申请所提供的各实施例中所使用的对存储器、存储、数据库或其它介质的任何引用，均可包括非易失性和易失性存储器中的至少一种。非易失性存储器可包括只读存储器(Read-Only Memory，ROM)、磁带、软盘、闪存或光存储器等。易失性存储器可包括随机存取存储器(Random Access Memory，RAM)或外部高速缓冲存储器。作为说明而非局限，RAM可以是多种形式，比如静态随机存取存储器(Static Random Access Memory，SRAM)或动态随机存取存储器(Dynamic Random Access Memory，DRAM)等。
以上对本申请实施例所提供的一种数据传输方法、装置、终端和计算机可读存储介质进行了详细介绍，本文中应用了具体个例对本申请的原理及实施方式进行了阐述，以上实施例的说明只是用于帮助理解本申请的方法及其核心思想；同时，对于本领域的技术人员，依据本申请的思想，在具体实施方式及应用范围上均会有改变之处，综上所述，本说明书内容不应理解为对本申请的限制。

Claims (16)

  1. 一种数据传输方法，用于发送端，包括：
    获取音频数据以及传输状态信息；
    基于所述传输状态信息确定压缩系数和冗余系数；
    根据所述压缩系数对所述音频数据进行时域压缩处理，得到压缩数据；
    根据所述冗余系数对所述压缩数据进行信道编码，得到传输数据包；及
    发送所述传输数据包。
  2. 如权利要求1所述的数据传输方法，其特征在于，所述传输状态信息包括接收数量，所述基于所述传输状态信息确定压缩系数和冗余系数，包括：
    对发送的传输数据包进行统计，得到发送码率、发送数量；
    根据所述发送数量和所述接收数量计算丢包率；
    基于所述丢包率确定冗余系数；
    基于所述丢包率和所述发送码率确定压缩系数；及
    所述发送所述传输数据包，包括：
    发送所述传输数据包和所述冗余系数。
  3. 如权利要求2所述的数据传输方法，其特征在于，所述基于所述丢包率和所述发送码率确定压缩系数，包括：
    分别对所述丢包率和所述发送码率进行统计，得到所述丢包率对应的第一变化趋势和所述发送码率对应的第二变化趋势，以及所述丢包率和所述发送码率之间的相关性；及
    当所述第一变化趋势和所述第二变化趋势均为上升趋势，且所述丢包率和所述发送码率之间的相关性呈正相关时，根据所述丢包率和所述发送码率确定压缩系数。
  4. 如权利要求1所述的数据传输方法，其特征在于，所述根据所述压缩系数对所述音频数据进行时域压缩处理，得到压缩数据，包括：
    根据所述压缩系数确定采样窗口；
    基于所述采样窗口对所述音频数据进行数据采样，得到子音频数据；及
    对所述子音频数据进行合成处理，得到压缩数据。
  5. 如权利要求4所述的数据传输方法，其特征在于，所述对所述子音频数据进行合成处理，得到压缩数据，包括：
    计算子音频数据之间的波形互相关系数；
    根据所述波形互相关系数确定波形相似的子音频数据；及
    对所述波形相似的子音频数据进行波形叠加处理，得到压缩数据。
  6. 如权利要求1所述的数据传输方法，其特征在于，所述根据所述压缩系数对所述音频数据进行时域压缩处理，得到压缩数据，包括：
    对所述音频数据进行基音分析，确定所述音频数据对应的基音点；
    根据所述基音点对所述音频数据进行数据采样，得到多个子音频数据；
    根据所述压缩系数从所述多个子音频数据中筛选出目标子音频数据；及
    对所述目标子音频数据进行合成处理，得到压缩数据。
  7. 如权利要求1所述的数据传输方法，其特征在于，所述根据所述冗余系数对所述压缩数据进行信道编码，得到传输数据包，包括
    对所述冗余系数进行信道编码，得到压缩系数标识符；及
    根据所述压缩系数标识符对所述压缩数据进行信道编码，得到传输数据包。
  8. 一种数据传输方法，用于接收端，包括：
    获取传输数据包和压缩系数；
    基于所述传输数据包确定当前时刻的传输状态信息；
    发送所述当前时刻的传输状态信息；
    对所述传输数据包进行信道解码，得到待复原数据；及
    根据所述压缩系数对所述待复原数据进行时域扩张处理，得到复原数据。
  9. 如权利要求8所述的数据传输方法，其特征在于，所述传输数据包还包括发送码率、发送数量，所述基于所述传输数据包确定当前时刻的传输状态信息，包括：
    对接收的传输数据包进行统计，得到接收数量；
    根据所述发送数量和所述接收数量计算丢包率；及
    确定当前时刻的传输状态信息，所述传输状态信息包括丢包率。
  10. 如权利要求8所述的数据传输方法，其特征在于，所述根据所述压缩系数对所述待复原数据进行时域扩张处理，得到复原数据，包括：
    根据所述压缩系数确定对应的解压系数；
    根据所述解压系数确定采样窗口；
    基于所述采样窗口对所述待复原数据进行数据采样，得到子待复原数据；及
    对所述子待复原数据进行合成处理，得到复原数据。
  11. 如权利要求10所述的数据传输方法，其特征在于，所述对所述子待复原数据进行合成处理，得到复原数据，包括：
    计算所述子待复原数据之间的波形互相关系数；
    根据所述波形互相关系数确定波形相似的子待复原数据；及
    对所述波形相似的子待复原数据进行波形叠加处理，得到复原数据。
  12. 一种数据传输装置，适用于发送端，包括：
    第一获取单元，用于获取音频数据以及传输状态信息；
    系数单元，用于基于所述传输状态信息确定缩放系数和冗余系数；
    压缩单元，用于根据所述缩放系数对所述音频数据进行时域压缩处理，得到压缩数据；
    编码单元，用于根据所述冗余系数对所述压缩数据进行信道编码，得到传输数据包；及
    第一发送单元，用于发送所述传输数据包。
  13. 一种数据传输装置，适用于接收端，包括：
    第二获取单元，用于获取传输数据包，所述传输数据包包括冗余数据和缩放系数；
    占用单元，用于基于所述传输数据包确定当前时刻的传输状态信息；
    第二发送单元，用于发送所述当前时刻的传输状态信息；
    解码单元，用于根据所述冗余数据对所述传输数据包进行信道解码，得到待复原数据；及
    扩张单元，用于根据所述缩放系数对所述待复原数据进行时域扩张处理， 得到复原数据。
  14. 一种终端，其特征在于，包括存储器和处理器，所述存储器中存储有计算机可读指令，所述计算机可读指令被所述处理器执行时，使得所述处理器执行如权利要求1～10任一项所述的数据传输方法中的步骤。
  15. 一个或多个存储有计算机可读指令的非易失性存储介质，所述计算机可读指令被一个或多个处理器执行时，使得一个或多个处理器执行权利要求1～10任一项所述的数据传输方法中的步骤。
  16. 一种数据传输系统，所述数据传输系统包括发送端和接收端，其中：
    所述发送端用于获取音频数据以及获取接收端发送的传输状态信息，基于所述传输状态信息确定压缩系数和冗余系数，根据所述压缩系数对所述音频数据进行时域压缩处理，得到压缩数据，根据所述冗余系数对所述压缩数据进行信道编码，得到传输数据包，发送所述传输数据包至接收端；
    所述接收端用于获取发送端发送的传输数据包，所述传输数据包包括冗余数据和缩放系数，基于所述传输数据包确定当前时刻的传输状态信息，发送所述当前时刻的传输状态信息到发送端，对所述传输数据包进行信道解码，得到待复原数据，根据所述缩放系数对所述待复原数据进行时域扩张处理，得到复原数据。
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