WO2012162870A1 - 联合译码装置及方法、必要性判断方法和装置、接收机 - Google Patents

联合译码装置及方法、必要性判断方法和装置、接收机 Download PDF

Info

Publication number
WO2012162870A1
WO2012162870A1 PCT/CN2011/074777 CN2011074777W WO2012162870A1 WO 2012162870 A1 WO2012162870 A1 WO 2012162870A1 CN 2011074777 W CN2011074777 W CN 2011074777W WO 2012162870 A1 WO2012162870 A1 WO 2012162870A1
Authority
WO
WIPO (PCT)
Prior art keywords
unit
decoding
coding rate
source
current frame
Prior art date
Application number
PCT/CN2011/074777
Other languages
English (en)
French (fr)
Inventor
张磊
王昕�
周华
吴建明
韩笑蕾
赵晓群
张楠
方腾龙
Original Assignee
富士通株式会社
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by 富士通株式会社 filed Critical 富士通株式会社
Priority to PCT/CN2011/074777 priority Critical patent/WO2012162870A1/zh
Priority to CN2011800637673A priority patent/CN103283150A/zh
Publication of WO2012162870A1 publication Critical patent/WO2012162870A1/zh
Priority to US14/090,451 priority patent/US20140164002A1/en

Links

Classifications

    • 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
    • H03—ELECTRONIC CIRCUITRY
    • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
    • H03M13/00—Coding, decoding or code conversion, for error detection or error correction; Coding theory basic assumptions; Coding bounds; Error probability evaluation methods; Channel models; Simulation or testing of codes
    • H03M13/29—Coding, decoding or code conversion, for error detection or error correction; Coding theory basic assumptions; Coding bounds; Error probability evaluation methods; Channel models; Simulation or testing of codes combining two or more codes or code structures, e.g. product codes, generalised product codes, concatenated codes, inner and outer codes
    • H03M13/2957—Turbo codes and decoding
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
    • H03M13/00—Coding, decoding or code conversion, for error detection or error correction; Coding theory basic assumptions; Coding bounds; Error probability evaluation methods; Channel models; Simulation or testing of codes
    • H03M13/63—Joint error correction and other techniques
    • H03M13/6312—Error control coding in combination with data compression

Definitions

  • the present invention relates to a receiver and receiver decoding method. Background technique
  • Shannon's separation coding principle is the basis of the current multimedia communication system.
  • the source code is based on the statistical characteristics of the source to achieve effective compression of the source data, while the channel coding is used to increase the robustness of the transmitted data and achieve reliable transmission.
  • the source coding and channel coding of the communication system are designed independently of each other, which reduces the complexity of the system design.
  • the independent design system can obtain the best system performance only when the codeword is infinitely long (that is, the infinite complexity of complexity and delay:) and the point-to-point communication environment, which promotes the joint of the source channel.
  • LTE-A mainly uses R6 Turbo coding as LTE channel coding, and also introduces Tail Biting convolutional codes.
  • many companies are also studying other coding methods, such as low density parity check code LDPC:).
  • the LDPC code and the Turbo code achieve this goal in close proximity to the Shannon limit in their respective ways.
  • the existing idea of joint decoding based on LDPC source channel is combined with hidden Markov source estimation.
  • LDPC coding is performed on the multimedia source containing redundancy, and joint source estimation is adopted at the receiving end. Coding with the decoding method of channel decoding.
  • the storage requirements of the generation matrix are large and the real-time performance is poor, which limits the application of this technology in practical systems.
  • Adaptive Multi Rate (AMR) speech coding is a new speech coding technology developed by 3GPP (3rd Generation Partnership Project) following EFR, FR, and HR. Its core idea is to automatically select the appropriate encoding and decoding algorithm according to the change of the uplink and downlink signal quality, to solve the problem of rate allocation of source and channel coding in a more intelligent way, and make the configuration and utilization of wireless resources more flexible and efficient. Its coding features are:
  • AMR-NB supports multiple coding rates: 12.2 kb/s, 10.2 kb/s, 7.95 kb/s, 7. 40 kb/s, 6.70 kb/s, 5.90 kb/s, 5.15 kb/s and 4.75 Kb/s, in addition, it also includes low rates Background noise coding mode (1.80 kb/s).
  • the speech coding rate depends on the condition of the channel: compared with the fixed coding rate adopted by the current GSM speech coding, AMR speech coding can adaptively select an optimal channel mode according to the radio channel and transmission conditions (all The code rate is transmitted by the rate or half rate and the coding mode (differentiated by bit rate).
  • AMR core frames carry speech or noise-encoded information.
  • the bits generated by the speech encoder are divided into three categories according to their subjective importance: Type A, Type B, and Type C. These three types are suitable for different error protection levels in the network.
  • Type A contains the bits most sensitive to errors. Any bit errors in these bits must be decoded with appropriate error concealment. This class is used to AMR auxiliary information.
  • the CRC in the check is protected; errors in the type B and type C bits will reduce the speech quality, but if the subjective feeling is acceptable, the erroneous speech frame can also be directly decoded, and the bit in the type B is sensitive to the error. To be higher than the bits in type C.
  • the present invention has been made in view of the above problems of the conventional art, and is intended to solve at least one of the disadvantages of the conventional techniques.
  • a source channel joint decoding necessity determining apparatus includes: a source coding rate change determining unit, The source coding rate change determination unit determines whether the source coding rate of the current frame is the same as the source coding rate of the previous frame; the source coding rate criterion determining unit, the source coding rate qualified determining unit determines the current frame Whether the source coding rate is less than a predetermined source coding rate threshold; the current frame signal to interference ratio criterion determining unit, the current frame signal to interference ratio determining unit determines whether the signal to interference ratio of the current frame is lower than a predetermined one a signal-to-interference ratio threshold; and a necessity result determining unit, wherein the source coding rate change determination unit determines that the source coding rate of the current frame is the same as the source coding rate of the previous frame, and the current frame is The necessity result is when the source coding rate is less than
  • a method for determining a source channel joint decoding necessity comprising: determining whether a source coding rate of a current frame is the same as a source coding rate of a previous frame; Whether the source coding rate of the frame is less than a predetermined source coding rate threshold; determining whether the signal to interference ratio of the current frame is lower than a predetermined signal to interference ratio threshold; and when the source coding rate of the current frame is the same as the previous frame.
  • the source coding rate is the same, and the source coding rate of the current frame is less than a predetermined threshold, and when the signal to interference ratio of the current frame is lower than a predetermined signal to interference ratio threshold, it is determined that the source channel joint translation should be performed. code.
  • the coding rate is used as the decision mechanism, and the channel conditions are considered, and the performance of the LTE-Advanced system can be improved.
  • a source channel joint decoding apparatus comprising: a Turbo decoding unit, configured to decode a received signal; a frame unit, the deframing unit is configured to deframe the received signal decoded by the turbo decoding unit, and obtain a source coding rate; the error speech frame determining unit, the error speech frame determining unit is configured to determine the current frame Whether it is an erroneously received speech frame; the source channel joint decoding necessity judging device according to claim 1, determining whether the source channel joint decoding is required according to the source coding rate obtained by the deframing unit; And a joint coding unit, where the joint coding unit performs joint coding of the source channel by using a loop iterative operation of the bit estimation unit and the turbo decoding unit.
  • the source channel joint decoding apparatus further includes a maximum iteration number setting unit and/or a Turbo decoding method setting unit.
  • the maximum iteration number setting unit sets the maximum number of loop iteration operations of the bit estimation unit and the Turbo decoding unit.
  • the deframing unit obtains the type of the bit generated by the transmitting end speech coder, and the maximum iteration number setting unit sets the joint decoding unit according to the type of the bit, the bit estimating unit and The maximum number of loop iteration operations of the Turbo decoding unit.
  • the turbo decoding method setting unit sets a turbo decoding method used by the turbo decoding unit during a loop iterative operation of the bit estimating unit and the turbo decoding unit.
  • the unequal parameter protection strategy can be used to adaptively set the number of iterations of the Turbo decoder and the decoding scheme, reduce the number of decoding iterations, save storage space, and further improve the performance of the LTE-Advanced system.
  • the present invention also relates to a logic component readable program that, when executed by a logic component, can cause the logic component to function as the apparatus described above or cause the logic component to implement the above method.
  • the invention further relates to a logical component readable tangible storage medium storing the above described logical component readable program.
  • FIG. 1 shows a schematic diagram of a mobile phone used as a receiver in accordance with an embodiment of the present invention
  • FIG. 2 is a schematic block diagram of a receiver in accordance with an embodiment of the present invention
  • FIG. 3 is a schematic illustration of a source channel joint decoding apparatus in accordance with an embodiment of the present invention
  • FIG. 4 is a block diagram showing an erroneous speech frame determination unit of an embodiment
  • FIG. 5 is a schematic block diagram showing a joint decoding necessity judging unit according to an embodiment of the present invention.
  • Figure 6 illustrates a joint decoding unit in accordance with an embodiment of the present invention
  • FIG. 7 is a diagram showing a source channel joint decoding apparatus according to another embodiment of the present invention
  • FIG. 8 is a diagram showing a source channel joint decoding apparatus according to still another embodiment of the present invention
  • FIG. 10 is a diagram showing a method for determining a source channel joint decoding necessity according to an embodiment of the present invention
  • FIG. 11 is a flow chart showing a process of joint decoding in accordance with an embodiment of the present invention
  • Figure 12 is a flow chart showing the processing of joint decoding in accordance with another embodiment of the present invention. detailed description
  • Fig. 1 shows a schematic diagram of a mobile phone used as a receiver in accordance with an embodiment of the present invention.
  • the mobile telephone 10 can be a flip type telephone having a flip cover 15 movable between an open position and a closed position. In Fig. 1, the flip cover 15 is shown in an open position. It should be understood that the mobile telephone 10 can be of other construction, such as a "longboard phone" or "slide phone.”
  • Mobile phone 10 can include display 14.
  • the display 14 displays information such as an operation status, time, telephone number, phone book information, various menus, and the like to the user so that the user can utilize various features of the electronic device 10.
  • Display 14 can also be used to visually display content that is received by electronic device 10 and/or retrieved from a memory (not shown) of electronic device 10.
  • Display 14 can be used to present images, videos, and other graphics to the user, such as photos, mobile television content, and video related to the game.
  • keyboard 18 provides a variety of user input operations.
  • keyboard 18 may include alphanumeric keys that allow alphanumeric information (such as phone numbers, phone lists, phone book information, notepads, text, etc.) to be entered.
  • keyboard 18 may include specific function keys 17, such as a "call to send" button for initiating or answering a call, and a "call end” button for ending or “hanging up” the call.
  • the particular function keys may also include menu navigation keys and selection keys that are conveniently navigated through menus displayed on display 14.
  • a pointing device and/or navigation keys can be provided to receive directional inputs from the user.
  • the display 14 and the keyboard 18 can be used in combination with each other to implement the functions of the soft keys.
  • antennas, microcontrollers, and the like that are necessary to perform their functions. It should be noted that the receiver of the present invention is not limited to a mobile phone, but may be any reception.
  • a receiver for AMR speech encoded signals is provided.
  • the receiver 200 includes a signal receiving unit 201, a CP removing unit 202, a fast Fourier transform (FFT) unit 203, a channel estimating unit 204, a multiple input multiple output detecting unit 205, and a letter.
  • the source channel is coupled to the decoding device 206.
  • the channel estimation unit 204 performs channel estimation to obtain a channel estimation value, and also determines a signal to noise ratio estimation value of the received signal, and transmits the signal to noise ratio estimation value to the source channel joint decoding unit 206.
  • the receiving unit 201 Since the functions and implementations of the receiving unit 201, the CP removing unit 202, the fast Fourier transform (FFT) unit 203, the channel estimating unit 204, and the MIMO detecting unit 205 can be known and recognized by those skilled in the art. All methods are known to be carried out, and thus they will not be described in detail in the present invention.
  • FFT fast Fourier transform
  • MIMO detection unit 205 may be omitted.
  • FIG. 3 shows an illustration of a source channel joint decoding apparatus in accordance with an embodiment of the present invention.
  • a source channel joint decoding apparatus includes a turbo decoding unit 301, a deframing unit 302, an error speech frame determining unit 303, a joint decoding necessity determining unit 304, Joint decoding unit 305 and speech decoding unit 306.
  • the turbo decoding unit 301 performs Turbo decoding on the input current frame signal, and outputs a 1-bit bit probability and a decoding hard decision result.
  • the demapping unit 302 performs de-frame according to the decoded hard decision result obtained by the turbo decoding unit, and acquires the source coding rate according to the frame header information obtained by the de-frame.
  • the erroneous speech frame judging unit 303 judges whether or not the current frame is an erroneous speech frame. When it is judged that it is a speech frame and the decoding is not correct, the joint speech necessity indication unit 304 outputs an error speech frame indication signal. When it is judged that it is a non-speech frame or a correct speech frame, a corresponding control signal is output.
  • FIG. 4 shows a block diagram of an erroneous speech frame determination unit of an embodiment.
  • the erroneous speech frame judging unit includes a reception error judging unit 401, a speech frame judging unit 402, and a judgment result determining unit 403.
  • the reception error judging unit 401 may include a CRC Both the verification module and the reception quality determination module or one of them.
  • the CRC check module performs a CRC check on the frame body to determine whether there is a bad voice.
  • the reception quality determining module may determine whether there is voice corruption according to a parameter indicating the quality of reception obtained by deframing the frame by the deframe unit.
  • the speech frame determination unit 402 determines whether the current frame is a speech frame.
  • the judgment result determining unit 403 determines the final judgment result based on the judgment result of the reception error judgment unit 401 and the judgment result of the voice frame judgment unit 402, and outputs a signal indicating the corresponding judgment result.
  • the reception error judging unit 401 and the speech frame judging unit 402 are shown as units operating in parallel in the drawing, their operations may be performed sequentially, or partially in parallel.
  • the speech decoding unit 306 can perform the following processing. For example, when it is determined that the voice frame is correct, the current frame frame body is directly sent to the voice decoding module. When it is determined that it is a background noise frame, it first enters the background noise function selection module and then inputs to the speech decoding module. When it is judged that it is a null frame, it first enters the silent frame replacement module and then enters the speech decoding module. When it is judged that the reception type cannot be judged, the SID (silence description) is bad, etc., first enter the error concealment module and then enter the speech decoding module and the like.
  • the background noise function selection module, the silence frame replacement module, the error concealment module, the speech decoding module, and the like included in the speech decoding unit 306 can be implemented by various methods well known to those skilled in the art, and any method is adopted. The implementation of the present invention is not affected and will not be described in detail herein.
  • the joint decoding necessity judging unit 304 judges whether or not the source channel joint decoding is performed when the erroneous speech frame judging unit 303 judges that the current frame is an incorrectly decoded speech frame.
  • FIG. 5 shows a schematic block diagram of a joint decoding necessity judging unit according to an embodiment of the present invention.
  • the joint decoding necessity judging unit includes a source coding rate change determining unit 501, a source coding rate eligibility determining unit 502, and a current frame signal to interference ratio eligibility.
  • the source coding rate change determination unit 501 determines whether the source coding rate of the current frame is the same as the source coding rate of the previous frame.
  • the source coding rate qualification determining unit 502 determines whether the source coding rate of the current frame is less than a predetermined source coding rate threshold.
  • the current frame signal to interference ratio judging unit 503 determines whether the signal to interference ratio of the current frame is lower than a predetermined signal to interference ratio threshold. In an embodiment, when the source coding rate change determining unit 501 determines that the source coding rate of the current frame is the same as the source coding rate of the previous frame, and the source coding rate of the current frame When less than the predetermined threshold and the signal to interference ratio of the current frame is lower than the predetermined signal to interference ratio threshold, the necessity result determining unit 504 determines that the source channel joint decoding should be performed. Otherwise, it is determined that joint decoding is not performed, and the corresponding signal is output to the speech decoding unit 306 (e.g., the error concealment module in the speech decoding unit 306).
  • the speech decoding unit 306 e.g., the error concealment module in the speech decoding unit 306
  • signal to interference ratio in the present invention should be understood broadly, including signal interference to noise ratio, signal to interference ratio, and signal to noise ratio.
  • the joint decoding unit 305 and the turbo decoding unit together decode the current frame.
  • FIG. 6 illustrates a joint decoding unit in accordance with an embodiment of the present invention.
  • the Turbo decoding unit 301 is also shown.
  • joint decoding unit 305 includes prior probability estimation unit 601, bit estimation unit 602, a posteriori probability estimation unit 603, end determination unit 604, and parameter estimation unit 605.
  • the prior probability estimation unit 601 calculates the index value prior probability based on the index value posterior probability.
  • the index value posterior probability is derived, for example, from an a posteriori index value estimate obtained from Turbo decoding of the previous frame.
  • the bit estimation unit 602 calculates a new index value bit probability based on the index value prior probability from the prior probability estimation unit 601 and the index value bit probability from the turbo coding unit, and outputs the new index value bit probability to the Turbo translation. Code unit.
  • the turbo decoding unit 301 re-synchronizes the current frame and outputs the index value bit probability based on the index value bit probability from the bit estimation unit 602 and the received information of the current frame, and will be under the control of the joint decoding end judging unit 604.
  • the bit probability is output to the bit estimation unit 602 to effect joint decoding of the source channel.
  • the joint decoding end judging unit 604 judges whether or not the decoding result of the turbo decoding unit 301 is correct, or whether the predetermined number of loops has been reached. If the predetermined number of cycles has been reached or the decoding result is correct, the joint decoding end judging unit 604 judges that the joint decoding ends, disconnects the connection between the bit estimating unit 602 and the turbo decoding unit 301, and causes the Turbo decoding unit to output The index value bit probability is output to the posterior probability estimation unit.
  • the posterior probability estimation unit 602 estimates the index value posterior probability based on the index value bit probability from the turbo decoding unit, and outputs it to the parameter estimation unit 605.
  • the parameter estimation unit 605 performs parameter estimation according to the MMSE criterion or the MAP criterion according to the posterior probability estimated by the posterior probability estimation unit (for example, estimating the LSF (Line Spectral Frequency) sub-vector index, the adaptive codebook index, the pulse position, and the pulse. Symbols and other parameters).
  • the parameter estimate is passed to the speech decoding module.
  • the process of cooperative decoding between the joint coding unit and the turbo coding unit is not limited to the process described above.
  • the calculation of the posterior probability and the calculation of the prior probability can be performed at each iteration.
  • the process of joint decoding between the joint coding unit and the Turbo decoding unit can also be referred to the following documents:
  • joint coding unit of the present invention may further comprise a storage unit operable to store the a posteriori probability value and the source coding rate of the previous frame.
  • the joint decoding of the source channel divides the calculation of the iterative channel decoding and the posterior probability into two processes. First, the iterative channel decoding is performed, and then the posterior probability is calculated by using the bit likelihood value decoded by the channel, and the posterior information and the prior information are not calculated every iteration, thereby reducing the computation amount and simplifying the structure and calculation of the joint decoding. The process, and reduces the performance penalty caused by the index value bit independence assumption.
  • joint decoding unit shown in Figure 6 is merely illustrative.
  • a joint decoding unit that employs a joint decoding method such as that shown in Fig. 12 together with the turbo decoding unit may also be employed.
  • FIG. 7 shows a source channel joint decoding apparatus according to another embodiment of the present invention. versus In the source channel joint decoding apparatus shown in FIG. 3, the source channel joint decoding apparatus shown in FIG. 7 further includes a maximum iteration number setting unit 307.
  • the deframing unit 302 also acquires the parameter type.
  • the parameter type is the type A, type B and type C of the bits generated by the transmitting speech coder. These three types of bits apply different error protection levels in the network, which have been described above and will not be described again.
  • the maximum iteration number setting unit 307 sets the number of iterations as follows:
  • Bits of parameter type A Set the number of iterations greater than the first predetermined number of times
  • (2;) Bits of the parameter type B and type C Set the number of iterations less than the second predetermined number of times.
  • the first predetermined number of times may be the same as the second predetermined number of times, for example, both are two. However, it may be different, for example, the first predetermined number of times may be set to 4, and the second predetermined number of times may be set to 2.
  • Different predetermined times can also be set for type B and type C.
  • the end judging unit 604 sets the number of iterations set by the unit according to the maximum number of iterations when judging whether or not to end the iteration.
  • the setting of the number of loops according to the type of bits generated by the voice coder of the transmitting end is only an implementation manner, and the Turbo decoding unit, the bit estimating unit, etc. may be implemented according to requirements for performance as needed. Determine the appropriate number of cycles by calculating the speed of the hardware, etc. In this case, the deframing unit is not required to obtain the type of bits generated by the transmitting end speech coder.
  • FIG. 8 shows a source channel joint decoding apparatus according to still another embodiment of the present invention.
  • the source channel joint decoding apparatus shown in Fig. 8 further includes a turbo decoding method setting unit 308 as compared with the source channel joint decoding apparatus shown in Fig. 3.
  • the turbo decoding method setting unit 308 sets a decoding algorithm used in the Turbo decoding unit at each iteration.
  • the decoding algorithm used by Turbo decoding can be set as follows:
  • Turbo code decoding in the iterative process uses a joint algorithm consisting of any two or more of the above three algorithms.
  • the SOVA algorithm can be used in the first loop and the LOG-MAP algorithm in the second loop.
  • the Max-Log-MAP algorithm has the lowest complexity and relatively poor performance.
  • the SOVA algorithm has a fast calculation speed but a slightly poor performance.
  • the Log-MAP algorithm is reliable. High and accurate, but the calculation is more complicated. Selecting the Log-MAP algorithm in the iterative process can improve the reliability of decoding.
  • a more accurate algorithm such as the Log-MAP algorithm may be used first, and a simple algorithm such as a Max-Log-MAP algorithm or a SOVA algorithm may be selected in the subsequent iterative process;
  • a relatively simple algorithm is selected, and in the subsequent iterative process, the external information of the bit estimation is used, and a more accurate algorithm is used, thereby shortening the convergence time of the algorithm.
  • the flow of an exemplary process performed by the above-described receiver and its components in accordance with the present invention is described below.
  • the description of the device can be used to assist in understanding the method performed by the device, and a description of the method performed by the device can be used to assist in understanding the device.
  • FIG. 9 shows a method for judging the necessity of joint decoding of a source channel according to an embodiment of the present invention.
  • step S901 it is determined whether the source coding rate of the current frame is the same as the source coding rate of the previous frame. Then, in step S902, it is determined whether the source coding rate of the current frame is less than a predetermined source coding rate threshold.
  • step S903 it is determined whether the signal to interference ratio of the current frame is lower than a predetermined signal to interference ratio threshold.
  • step S904 when the source coding rate of the current frame is the same as the source coding rate of the previous frame, and the source coding rate of the current frame is less than a predetermined threshold, and the current frame is dry When the ratio is lower than the predetermined signal-to-interference ratio threshold, it is determined that the source channel joint decoding should be performed.
  • FIG. 10 illustrates a source channel joint decoding and determining method according to an embodiment of the present invention.
  • the source channel joint decoding method first, in step S1001, the received signal is decoded by the turbo decoding unit. Then, in step S1002, the received signal decoded by the turbo decoding unit is deframed, and the parameter type and the source coding rate are obtained.
  • step S1003 it is determined whether the current frame is a voice frame that receives an error. When it is determined that the voice frame is received incorrectly (S1003, YES), it is determined in step S1004 whether or not source channel joint decoding is required.
  • step S1004 When it is determined that joint decoding is required (step S1004, YES), the bit estimation is set in step S1005.
  • step S1005 The maximum number of cyclic iteration operations (joint decoding operations) of the unit and the turbo decoding unit.
  • the maximum number of loop iteration operations of the bit estimation unit and the Turbo decoding unit is set by deblocking in S1002. It is also possible to set the number of loop operations based on time (for example, not busy hours), performance requirements, customer type (which requires a user with high communication quality, the maximum number of times can be larger). Then, in step S1006, a Turbo decoding method used by the Turbo decoding is set during a loop iterative operation of the bit estimating unit and the Turbo decoding unit. Then, in step S1007, joint decoding is performed, that is, according to the set maximum number of loop iteration operations and the set Turbo decoding method, the bit estimating unit performs cooperative operation with the Turbo decoding unit to perform current frame processing. decoding.
  • step S1008 The parameter estimated after decoding is transmitted to step S1008 for speech decoding processing.
  • step S1003 if it is determined in step S1003 that the speech frame is not received, or if it is determined in step S1004 that joint decoding is not necessary, the process proceeds directly to step S1008 to perform speech decoding processing.
  • Step S1008 includes performing speech decoding using a speech decoder, and necessary operations such as silent frame replacement, background noise function selection, and error concealment.
  • FIG. 11 is a flow chart showing the processing of joint decoding in accordance with an embodiment of the present invention.
  • the prior probability calculation may be first performed in step S1101 to estimate the prior probability.
  • the prior probability estimate can be based on the posterior probability, ie the posterior probability of the previous frame.
  • bit estimation is performed based on the prior probability and the index value bit probability obtained by Turbo decoding.
  • turbo decoding is performed again based on the result of the bit estimation.
  • step S1104 based on the result of the turbo decoding, it is judged whether or not joint decoding should be ended.
  • step S1104 when the number of loops has been greater than a predetermined number of times (e.g., a set number of times), or if, for example, the result of Turbo decoding is correct, it is judged that joint decoding should be ended.
  • a predetermined number of times e.g., a set number of times
  • step S1106 the posterior probability estimation is performed (step S1105), and parameter estimation is performed (step S1106).
  • step S1104 NO when it is judged that the joint decoding should not be ended (step S1104, NO), the processing returns to step S1102, and the bit estimation and subsequent processing are continued.
  • the bit type for example, 4 times of the maximum number of cycles are set for the A type bit, 3 times of the maximum number of cycles are set for the B type bit, and 2 times of the maximum number of cycles are set for the C type bit.
  • joint decoding can first decode the type A bits (most More than four times), then the B type bits are decoded (up to 3 times), and finally the C type bits are decoded (maximum 2 times) for a total of 9 times. That is, decoding different types of bits one by one.
  • the loop iterations may be totaled 4 times, the first and second times decode the three types of bits of the AC, and the third time decode the two types of bits of B and A.
  • the A-type bit is only decoded for the fourth time. That is, the generalization and the intermediate jump decoding.
  • Figure 12 is a flow chart showing the processing of joint decoding in accordance with another embodiment of the present invention.
  • the prior probability calculation may be first performed in step S1201 to estimate the prior probability.
  • the prior probability estimate is based on the posterior probability, ie the posterior probability of the previous frame.
  • bit estimation is performed based on the prior probability and the index value bit probability obtained by Turbo decoding.
  • Turbo decoding is performed again based on the result of the bit estimation.
  • step S1104 a posterior probability estimation is performed, and then, in step S1205, based on the result of the Turbo estimation, it is judged whether or not joint decoding should be ended. For example, when the number of loops has been greater than a predetermined number of times (e.g., a set number of times), or if the result of Turbo decoding, for example, is correct, it is judged that joint decoding should be ended.
  • step S1205, YES parameter estimation is performed (step S1206), and the processing is ended.
  • step S1205 when it is judged that the joint decoding should not be ended (step S1205, NO), the processing returns to step S1201, the prior probability estimation is continued, and the subsequent processing is continued.
  • the above apparatus and method of the present invention may be implemented by hardware or by hardware in combination with software.
  • the present invention relates to a logic component readable program that, when executed by a logic component, enables the logic component to implement the apparatus or components described above, or to implement the various components described above Method or step.
  • Logic components such as field programmable logic components, microprocessors, processors used in computers, and the like.
  • the present invention also relates to a storage medium for storing the above program, such as a hard disk, a magnetic disk, an optical disk, a DVD, a flash, a magneto-optical disk, a memory card, a memory stick, and the like.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Probability & Statistics with Applications (AREA)
  • Theoretical Computer Science (AREA)
  • Health & Medical Sciences (AREA)
  • Signal Processing (AREA)
  • Computational Linguistics (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Error Detection And Correction (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

本发明涉及信源信道联合译码的装置及方法，以及必要性判断的方法、装置和接收机。该信源信道联合译码必要性判断装置包括：信源编码速率改变判断单元，判断当前帧的信源编码速率是否与前一帧的信源编码速率相同；信源编码速率适格性判断单元，判断当前帧的信源编码速率是否小于预定的信源编码速率阈值；当前帧信号干扰比（SIR）适格性判断单元，判断当前帧的信干比是否低于预定的信干比门限值；以及必要性结果确定单元，当该当前帧的信源编码速率与前一帧的信源编码速率相同，该当前帧的信源编码速率小于预定阈值，并且该当前帧的信干比低于预定的信干比门限值时，确定进行信源信道联合译码。

Description

联合译码装置及方法、 必要性判断方法和装置、 接收机 技术领域
本发明涉及接收机以及接收机译码方法。 背景技术
香农的分离编码原理是目前多媒体通信系统的设计基础， 信源编码 基于信源的统计特性实现信源数据的有效压缩， 而信道编码则用于增加 传输数据的鲁棒性， 实现可靠传输。 通信系统的信源编码和信道编码在 结构上分别独立地进行设计， 降低了系统设计的复杂性。 然而根据香农 的分离原理，只有在码字无限长 (即意味着复杂度和延迟的无限大：)和点对 点通信环境下， 独立设计系统才能获得最佳系统性能， 这就促使了信源 信道联合编译码算法的研究和发展。
目前在信道编码方面， LTE-A主要沿用 R6的 Turbo编码作为 LTE 信道编码，此外还引入了 Tail Biting卷积码。同时很多公司也在研究其他 的编码方式， 如低密度奇偶校验码LDPC：)。 LDPC码和 Turbo码以各自 的方式实现接近 Shannon限的这一目标。 现有的基于 LDPC的信源信道 联合译码的思想是与隐马尔可夫信源估计相结合， 在发送端， 对含有冗 余的多媒体信源进行 LDPC编码， 在接收端采用联合信源估计和信道译 码的译码方法进行联合译码。但是， 在联合迭代译码过程中， 由于 LDPC 译码迭代次数高， 对生成矩阵的存储需求大、 实时性差， 限制了这项技 术在实际系统中的应用。
另外， 自适应多速率（AMR, Adaptive Multi Rate)语音编码是由 3GPP(3rd Generation Partnership Project)制定的继 EFR、 FR、 HR之后的一 种新的语音编码技术。 它的核心思想是根据上下行信号质量的变化情况， 自动选择合适的编译码算法， 以更加智能的方式解决信源和信道编码的 速率分配问题， 使得无线资源的配置和利用更加灵活和高效。 其编码特 点为：
(l)AMR-NB支持多种编码速率： 12.2 kb/s， 10.2 kb/s, 7.95 kb/s, 7. 40 kb/s, 6.70 kb/s, 5.90 kb/s, 5.15 kb/s和 4.75 kb/s， 此外， 它还包括低速率 (1.80 kb/s)的背景噪声编码模式。
(2)语音编码速率取决于信道的条件：与现在的 GSM语音编码采用的 固定的编码速率相比， AMR语音编码则可根据无线信道和传输情况来自 适应地选择一种最佳信道模式 (全速率或半速率)和编码模式 (以比特率来 区分)进行编码传输。
不等参数保护策略： AMR核心帧携带了语音或噪声编码后的信 息， 语音编码器产生的比特根据它们的主观重要性分成三类： 类型 A、 类型 B和类型 C。 这三类适合网络中不同的错误保护级别， 类型 A包含 对错误最敏感的比特位， 这些比特中任何位出错， 都必须采用合适的差 错隐藏后才可译码，这一类通过 AMR辅助信息中的 CRC进行校验保护； 类型 B和类型 C级别的比特出现错误将降低语音质量， 但如果主观感觉 可以接受， 错误语音帧也是可以直接译码的， 类型 B中的比特对错误的 敏感程度要高于类型 C中的比特。
在目前的译码方式中， AMR变速率语音编码的特点没有得到充分利 用。 发明内容
本发明鉴于常规技术的上述问题做出， 用于解决因常规技术的局限 而具有的一种或更多种缺点， 至少提供一种有益的选择。
为了实现上述目的， 根据本发明的一个方面， 提供了一种信源信道 联合译码必要性判断装置， 所述信源信道联合译码必要性判断装置包括: 信源编码速率改变判断单元， 所述信源编码速率改变判断单元判断当前 帧的信源编码速率是否与前一帧的信源编码速率相同； 信源编码速率适 格判断单元， 所述信源编码速率适格判断单元判断当前帧的信源编码速 率是否小于预定的信源编码速率阈值； 当前帧信干比适格性判断单元， 所述当前帧信干比适格性判断单元判断当前帧的信干比是否低于预定的 信干比门限值； 以及必要性结果确定单元， 当所述信源编码速率改变判 断单元判断出当前帧的信源编码速率与前一帧的信源编码速率相同， 并 且所述当前帧的信源编码速率小于预定阈值， 同时所述当前帧的信干比 低于预定的信干比门限值时， 所述必要性结果确定单元确定应该进行信 源信道联合译码。
根据本发明的另一个方面， 提供了一种信源信道联合译码必要性判 断方法， 所述方法包括： 判断当前帧的信源编码速率是否与前一帧的信 源编码速率相同； 判断当前帧的信源编码速率是否小于预定的信源编码 速率阈值； 判断当前帧的信干比是否低于预定的信干比门限值； 以及当 所述当前帧的信源编码速率与前一帧的信源编码速率相同， 并且所述当 前帧的信源编码速率小于预定阈值， 同时所述当前帧的信干比低于预定 的信干比门限值时， 确定应该进行信源信道联合译码。
根据本发明的这种实施方式， 以编码速率作为判决机制， 并考虑了 信道条件， 能够提高 LTE-Advanced系统性能。
根据本发明的另一方面， 提供了一种信源信道联合译码装置及其方 法， 所述信源信道联合译码装置包括： Turbo 译码单元， 用于对接收的 信号进行译码； 解帧单元， 所述解帧单元用于对经 Turbo译码单元译码的 接收信号进行解帧， 并获得信源编码速率； 错误语音帧判断单元， 所述 错误语音帧判断单元用于确定当前帧是否是错误接收的语音帧； 根据权 利要求 1所述的信源信道联合译码必要性判断装置， 根据所述解帧单元所 获得的信源编码速率判断是否需要进行信源信道联合译码； 以及联合译 码单元， 所述联合译码单元利用比特估计单元与所述 Turbo译码单元的循 环迭代操作而进行信源信道联合译码。
在一种实施方式中， 所述信源信道联合译码装置还包括最大迭代次 数设定单元和 /或 Turbo译码方法设定单元。该最大迭代次数设定单元设定 所述比特估计单元与所述 Turbo译码单元的循环迭代操作的最大次数。 在 优选的实施方式中， 解帧单元获得发送端语音编码器产生的比特的类型， 最大迭代次数设定单元根据所述比特的类型设定所述联合译码单元中， 所述比特估计单元与所述 Turbo译码单元的循环迭代操作的最大次数。 该 Turbo译码方法设定单元设定在所述比特估计单元与所述 Turbo译码单元 的循环迭代操作过程中， 所述 Turbo译码所使用的 turbo译码方法。
根据这些实施方式， 可以采用不等参数保护策略， 自适应设置 Turbo 译码器的迭代次数及译码方案， 减少译码迭代次数， 节省存储空间， 进 一歩提高 LTE-Advanced系统性能。 本发明还涉及一种逻辑部件可读程序， 当所述逻辑部件可读程序被 逻辑部件执行时， 能够使所述逻辑部件作为上述的装置工作或使所述逻 辑部件实现上述方法。
本发明还涉及一种逻辑部件可读有形存储介质， 所述有形存储介质 存储有上述的逻辑部件可读程序。
应该注意， 术语"包括 /包含 /具有"在本文使用时指特征、 要件、 歩骤 或组件的存在， 但并不排除一个或更多个其它特征、 要件、 歩骤或组件 的存在或附加。
以上的一般说明和以下结合附图的详细说明都是示意性的， 不是对 本发明的保护范围的限制。 附图说明
从以下参照附图对本发明的详细描述中， 将更清楚地理解本发明的 以上和其它目的、 特征和优点。 在整个附图中， 相同或类似的标记标识 相同或类似的元素。
图 1 示出了作为根据本发明的实施方式的接收机使用的移动电话的 示意图；
图 2示出了依据本发明的一种实施方式的接收机的示意性框图； 图 3 示出了依据本发明的一种实施方式的信源信道联合译码装置的 示意性；
图 4示出了一种实施方式的错误语音帧判断单元的框图；
图 5示出了依据本发明的一种实施方式的联合译码必要性判断单元 的示意性框图；
图 6示出了依据本发明的一种实施方式的联合译码单元；
图 7示出了依据本发明另一实施方式的信源信道联合译码装置； 图 8示出了依据本发明又一实施方式的信源信道联合译码装置； 图 9示出了依据本发明一种实施方式的信源信道联合译码必要性判 断方法；
图 10示出了依据本发明一种实施方式的信源信道联合译码必要性判 断方法； 图 11示出了依据本发明一种实施方式的联合译码的处理过程的流程 图； 以及
图 12示出了依据本发明另一种实施方式的联合译码的处理过程的流 程图。 具体实施方式
下面参照附图详细说明本发明具体实施方式。 这些实施方式都是示 例性的， 不是对本发明保护范围的限制。 应该说明， 为了说明的清楚， 在以下的描述中省略了本领域技术人员所知的并且如果进行描述则有可 能对本发明的理解造成困扰的技术的说明。
图 1 示出了作为根据本发明的实施方式的接收机使用的移动电话的 示意图。 如图 1所示， 移动电话 10可以是具有可在打开位置与闭合位置 之间移动的翻盖 15的翻盖型电话。 图 1中， 翻盖 15被示出为处于打开 位置。 应了解的是， 移动电话 10可以为其它结构， 诸如 "长板型电话" 或 "滑盖型电话" 的结构。
移动电话 10可包括显示器 14。显示器 14向用户显示诸如操作状态、 时间、 电话号码、 电话簿信息、 各种菜单等的信息， 使得用户能利用电 子设备 10的各种特征。显示器 14还可以用于可视地显示电子设备 10接 收到的和 /或从电子设备 10 的存储器 (未示出)检索到的内容。 显示器 14 可用于向用户呈现图像、 视频和其他图形， 诸如相片、 移动电视内容以 及与游戏相关的视频。
键盘 18提供了多种用户输入操作。 例如， 键盘 18可包括允许输入 字母数字信息 （诸如， 电话号码、 电话列表、 电话簿信息、 记事本、 文 本等） 的字母数字键。 此外， 键盘 18可包括特定的功能键 17， 诸如用于 启动或应答电话的 "呼叫发送"键、 以及用于结束或者 "挂断" 电话的 "呼叫结束"键。 特定的功能键还可以包括在显示在显示器 14上的菜单 来方便地进行导航的菜单导航键和选择键。 例如， 可以提供指点设备和 / 或导航键以接收来自用户的方向性输入。 此外， 显示器 14和键盘 18可 以彼此结合起来使用以实现软键的功能。 接收机 10中还包括天线、 微控 制器等实现其功能所必须的部件。 应该注意， 本发明的接收机不限于移动电话， 而可以是任何接收
AMR语音编码信号的接收机。
图 2示出了依据本发明的一种实施方式的接收机的示意性框图。 如 图 2所示， 依据本发明的接收机 200包括信号接收单元 201、 CP去除单 元 202、 快速傅里叶变换 （FFT) 单元 203、 信道估计单元 204、 多输入 多输出检测单元 205、以及信源信道联合译码装置 206。信道估计单元 204 进行信道估计得到信道估计值， 并且还确定接收的信号的信噪比估计值， 并将该信噪比估计值传送给信源信道联合译码单元 206。 由于接收单元 201、 CP去除单元 202、 快速傅里叶变换 （FFT) 单元 203、 信道估计单 元 204、多输入多输出检测单元 205的功能和实现可以采用本领域的技术 人员现在所知的和未来所知的所有方法来进行， 因而本发明不再对它们 进行详细的描述。
在另选的实施方式中， MIMO检测单元 205可以省略。
应该注意， 该图仅是示例性的， 并不是对本发明的保护范围的限制。 下面详细说明信源信道联合译码装置 206。
图 3 示出了依据本发明的一种实施方式的信源信道联合译码装置的 示意性。 如图 3所示， 依据本发明的一种实施方式， 信源信道联合译码 装置包括 Turbo译码单元 301、解帧单元 302、错误语音帧判断单元 303、 联合译码必要性判断单元 304、联合译码单元 305以及语音译码单元 306。
Turbo译码单元 301对输入的当前帧信号进行 Turbo译码， 并输出索 弓 1值比特概率和译码硬判决结果。
解帧单元 302根据 Turbo译码单元得到的译码硬判决结果进行解帧， 根据解帧得到的帧头信息获取信源编码速率。
错误语音帧判断单元 303判断当前帧是否是错误的语音帧。 在判断 出是语音帧， 并且解码不正确时， 向联合译码必要性判断单元 304输出 错误语音帧指示信号。 当判断出是非语音帧或正确的语音帧时， 输出相 应的控制信号。
图 4示出了一种实施方式的错误语音帧判断单元的框图。 如图 4所 示， 错误语音帧判断单元包括接收错误判断单元 401、 语音帧判断单元 402 以及判断结果确定单元 403。 接收错误判断单元 401可以包括 CRC 校验模块和接收质量确定模块两者或它们中的一个。 CRC校验模块对帧 体进行 CRC校验， 确定是否存在语音坏。 接收质量确定模块可以根据解 帧单元通过解帧获得的指示接收质量的参数来确定是否存在语音坏。 语 音帧判断单元 402判断当前帧是否是语音帧。 判断结果确定单元 403根 据接收错误判断单元 401 的判断结果和语音帧判断单元 402的判断结果 确定最终判断结果， 并输出指示相应的判断结果的信号。 虽然在图中， 将接收错误判断单元 401和语音帧判断单元 402示出为并行操作的单元， 但它们的操作也可以顺序地进行， 或部分并行的进行。
当判断出是非语音帧或正确的语音帧时， 语音译码单元 306可以进 行如下的处理。 例如当判断出是正确的语音帧时， 直接将当前帧帧体发 送给语音译码模块。 当判断出是背景噪声帧时， 先进入背景噪声函数选 择模块再输入到语音译码模块。 当判断出是空帧时， 先进入静音帧替换 模块再进入语音译码模块。 当判断出是接收类型无法判断、 SID ( silence description) 坏等时， 先进入错误隐藏模块再进入语音译码模块等等。 由 于包括在语音译码单元 306 中的背景噪声函数选择模块、 静音帧替换模 块、 错误隐藏模块、 语音译码模块等都可以采用本领域技术人员所熟知 的各种方法实现， 并且采用任何方法都不影响本发明的实现， 因而这里 不再详细叙述。
联合译码必要性判断单元 304在错误语音帧判断单元 303判断出当 前帧是解码不正确的语音帧时， 判断是否进行信源信道联合译码。
图 5示出了依据本发明的一种实施方式的联合译码必要性判断单元 的示意性框图。 如图 5所示， 依据本发明的一种实施方式的联合译码必 要性判断单元包括信源编码速率改变判断单元 501、信源编码速率适格判 断单元 502、当前帧信干比适格性判断单元 503以及必要性结果确定单元 504。信源编码速率改变判断单元 501判断当前帧的信源编码速率是否与 前一帧的信源编码速率相同。 信源编码速率适格判断单元 502判断当前 帧的信源编码速率是否小于预定的信源编码速率阈值。 当前帧信干比适 格性判断单元 503判断当前帧的信干比是否低于预定的信干比门限值。 在一种实施方式中， 当信源编码速率改变判断单元 501 判断当前帧的信 源编码速率与前一帧的信源编码速率相同， 并且当前帧的信源编码速率 小于预定阈值， 同时当前帧的信干比低于预定的信干比门限值时， 必要 性结果确定单元 504确定应该进行信源信道联合译码。 否则判断出不进 行联合译码， 并将相应信号输出给语音译码单元 306 (例如语音译码单元 306中的错误隐藏模块）。
应该注意， 在本发明中信干比的含义应做广义的理解， 其包括信号 干扰噪声比、 信号干扰比以及信号噪声比等。
当联合译码必要性判断单元 304确定应进行联合译码时， 由联合译 码单元 305与 Turbo译码单元一起对当前帧进行译码。
图 6示出了依据本发明的一种实施方式的联合译码单元。 为了说明 的方便， 图中也示出了 Turbo译码单元 301。
如图 6所示， 依据本发明的一种实施方式， 联合译码单元 305包括 先验概率估计单元 601、 比特估计单元 602、 后验概率估计单元 603、 结 束判断单元 604以及参数估计单元 605。
先验概率估计单元 601 根据索引值后验概率计算索引值先验概率。 索引值后验概率例如来自于基于前一帧的 Turbo解码获得的后验索引值 估计。
比特估计单元 602基于来自先验概率估计单元 601 的索引值先验概 率和来自 Turbo译码单元的索引值比特概率计算新的索引值比特概率， 并将该新的索引值比特概率输出到 Turbo译码单元。
Turbo译码单元 301根据来自比特估计单元 602的索引值比特概率和 当前帧的接收信息，重新对当前帧进行 turbo解码并输出索引值比特概率， 并在联合译码结束判断单元 604 的控制下将该比特概率输出到比特估计 单元 602， 从而实现信源信道联合译码。
联合译码结束判断单元 604判断 Turbo译码单元 301的译码结果是 否正确， 或者是否达到预定循环次数。 如果已经达到预定循环次数或者 译码结果正确， 则联合译码结束判断单元 604判定联合译码结束， 断开 比特估计单元 602和 Turbo译码单元 301之间的连接，并使 Turbo译码单 元输出的索引值比特概率输出到后验概率估计单元。
后验概率估计单元 602根据来自 Turbo译码单元的索引值比特概率， 估计索引值后验概率， 并输出到参数估计单元 605。 参数估计单元 605根据后验概率估计单元估计出的后验概率， 依据 MMSE准则或 MAP准则进行参数估计（例如估计 LSF (线谱频率）子矢 量索引、 自适应码本索引、 脉冲位置及脉冲的符号等参数）。 将参数估计 值传递至语音译码模块。
联合译码单元与 Turbo译码单元之间协作进行联合译码的过程可以 参见《电子与信息学报》 2008年 11期 公开的《基于参数冗余量分配方 案和可变参数估计准则的 GSM EFR信源信道联合解码算法》 （作者为周 琳 吴镇扬）。 通过引用， 将该文并入本文中， 如同在本文中完全阐述了 一样。
但是应该注意， 联合译码单元与 Turbo译码单元之间协作进行联合 译码的过程并不限于以上说明的过程。 例如可以在每次迭代时都进行后 验概率的计算以及先验概率的计算。 关于联合译码单元与 Turbo译码单 元之间协作进行联合译码的过程还可以参照以下文献进行：
N. Gortz. On the iterative approximation of optimal joint source-channel decoding [J]. IEEE transaction on selected areas in communications 2001, 19(9): 1662-1670;
M.Adrat, P.Vary, J.Spittka. Iterative source-channel decoder using extrinstic information from softbit-source decoding [C] . In the proceedings of ICASSP' 01. Salt Lake City, USA. 2001 :2653-2656.
很显然， 本发明的联合译码单元还可以包括存储单元， 该存储单元 可以用来存储后验概率值和前一帧的信源编码速率。
根据本发明该实施方式的信源信道联合解码， 将迭代信道解码和后 验概率的计算分成两个过程。 首先进行迭代的信道解码， 而后利用信道 解码出的比特似然值计算后验概率， 不必每次迭代都计算后验信息和先 验信息， 因而减少了运算量， 简化了联合解码的结构和计算过程， 并减 少了索引值比特独立性假设造成的性能损失。
应该注意， 图 6所示的联合译码单元只是示意性的。 也可以采用是 与 Turbo译码单元一起采用例如图 12所示的联合译码方法的联合译码单 元。
图 7示出了依据本发明另一实施方式的信源信道联合译码装置。 与 图 3所示的信源信道联合译码装置相比， 图 7所示的信源信道联合译码 装置还包括最大迭代次数设定单元 307。
在这种实施方式中， 解帧单元 302还获取参数类型。 参数类型即发 送端语音编码器产生的比特的类型 A、 类型 B和类型 C。 这三类比特在 网络中适用不同的错误保护级别， 这已经在前文描述， 不再赘述。
最大迭代次数设定单元 307如下地设定迭代次数：
(l)参数类型为 A的比特： 设定大于第一预定次数的迭代次数； (2；)参数类型为 B和类型 C级别的比特： 设定小于第二预定次数的迭 代次数。
第一预定次数可以与第二预定次数相同，例如都为 2。但也可以不同， 例如第一预定次数可以设定为 4， 而第二预定次数可以设定为 2。
针对类型 B和类型 C也可以设定不同的预定次数。
结束判断单元 604在判断是否结束迭代时根据该最大迭代次数设定 单元设定的迭代次数。
应该注意， 根据发送端语音编码器产生的比特的类型而进行循环次 数的设定仅仅是一种实施方式， 可以按照需要， 根据对性能的要求、 实 现 Turbo译码单元、 比特估计单元等的相应硬件的运算速度等， 确定合 适的循环次数。 在这种情况下， 不需要解帧单元获得发送端语音编码器 产生的比特的类型。
图 8示出了依据本发明又一实施方式的信源信道联合译码装置。 与 图 3所示的信源信道联合译码装置相比， 图 8所示的信源信道联合译码 装置还包括 Turbo译码方法设定单元 308。
Turbo译码方法设定单元 308设定每次迭代时在 Turbo译码单元中使 用的译码算法。 例如可以如下地设定 Turbo译码使用的译码算法：
(l)迭代过程中 Turbo码译码全部使用 Max-Log-MAP算法；
(2；)迭代过程中 Turbo码译码全部使用 SOVA算法；
(3；)迭代过程中 Turbo码译码都使用 Log-MAP算法；
(4；)迭代过程中 Turbo码译码使用上述 3种算法中的任意两种或两种 以上组成的联合算法。 例如可以在第一次循环时使用 SOVA算法， 而在 第二次循环时使用 LOG-MAP算法。 Max-Log-MAP算法复杂度最低， 性能也相对较差； SOVA算法计算 速度快， 但性能稍差， 当对实时性要求较高时， 在迭代过程中全部选择 SOVA算法； Log-MAP算法可靠性高、 精确， 但计算较复杂， 在迭代过 程中全部选择 Log-MAP 算法可提高译码的可靠性。 对于迭代过程中 Turbo码译码使用联合算法的情况，可首先使用较精确的算法如 Log-MAP 算法，在随后的迭代过程中选择计算简单的算法如 Max-Log-MAP算法或 者 SOVA算法； 或者在第一次迭代过程中选择比较简单的算法， 而在随 后的迭代过程， 利用比特估计的外信息， 使用比较精确算法， 从而缩短 算法的收敛时间。
很显然， 图 7的实施方式和图 8的实施方式可以组合使用。
下面描述依据本发明的上述接收机及其部件执行的示例性处理的流 程。 对装置的描述可以用来帮助对该装置执行的方法进行理解， 对装置 执行的方法的说明可以用来帮助对装置进行理解。
图 9示出了依据本发明一种实施方式的信源信道联合译码必要性判 断方法。 如图 9所示， 首先， 在歩骤 S901 , 判断当前帧的信源编码速率 是否与前一帧的信源编码速率相同。 然后， 在歩骤 S902, 判断当前帧的 信源编码速率是否小于预定的信源编码速率阈值。 接着， 在歩骤 S903 , 判断当前帧的信干比是否低于预定的信干比门限值。最后，在歩骤 S904, 当所述当前帧的信源编码速率与前一帧的信源编码速率相同， 并且所述 当前帧的信源编码速率小于预定阈值， 同时所述当前帧的信干比低于预 定的信干比门限值时， 确定应该进行信源信道联合译码。
显然， 歩骤 S901到歩骤 S903的顺序可以改变， 也可以并行执行。 图 10示出了依据本发明一种实施方式的信源信道联合译码判断方 法。 如图 10所示， 根据该信源信道联合译码方法， 首先， 在歩骤 S1001 , 利用 Turbo译码单元对接收的信号进行译码。 然后， 在歩骤 S1002, 对经 Turbo译码单元译码的接收信号进行解帧， 并获得参数类型和信源编码速 率。 接着， 在歩骤 S1003 , 确定当前帧是否是接收错误的语音帧。 当确定 出是接收错误的语音帧时 （S1003 , 是）， 在歩骤 S1004判断是否需要进 行信源信道联合译码。 在该歩骤中， 可以采用图 9所示的方法。 当判断出 需要进行联合译码时（歩骤 S1004, 是）， 在歩骤 S1005设定所述比特估计 单元与所述 Turbo译码单元的循环迭代操作（联合译码操作）的最大次数。 在一种实施方式中， 根据发送端语音编码器产生的比特的类型 （在歩骤
S1002中通过解帧获得）， 设定比特估计单元与所述 Turbo译码单元的循环 迭代操作的最大次数。 也可以根据时间 （例如不繁忙时段）、 性能要求、 客户类型 （要求通信质量高的用户则该最大次数可以大一些） 等等设定 循环操作的次数。 然后， 在歩骤 S1006 , 设定在所述比特估计单元与所 述 Turbo译码单元的循环迭代操作过程中， 所述 Turbo译码所使用的 Turbo 译码方法。 接着在歩骤 S1007中进行联合译码， 即根据设定的循环迭代操 作的最大次数以及设定的 Turbo解码方法，所述比特估计单元与所述 Turbo 译码单元进行协同操作， 对当前帧进行解码。 解码后估计出的参数传送 到歩骤 S1008进行语音译码处理。 另一方面， 在歩骤 S1003判断出不是接 收错误的语音帧时， 或在歩骤 S1004判断出不必进行联合译码时， 直接进 入歩骤 S1008进行语音译码处理。 歩骤 S1008包括利用语音译码器进行语 音译码， 以及必要的静音帧替换、 背景噪声函数选择、 错误隐藏等操作。
图 11示出了依据本发明一种实施方式的联合译码的处理过程的流程 图。 如图 11所示， 依据本发明的一种实施方式， 在比特估计单元与 Turbo 译码单元进行联合译码时， 可以首先在歩骤 S1101进行先验概率计算， 估 计出先验概率。 先验概率估计可以基于后验概率， 即前一帧的后验概率。 然后在歩骤 S1102 , 根据该先验概率以及 Turbo译码得出的索引值比特概 率进行比特估计。 接着， 在歩骤 S1103 , 根据比特估计的结果， 再次进 行 Turbo译码。 在歩骤 S1104, 根据该 Turbo译码的结果， 判断是否应该结 束联合译码。 例如当循环次数已经大于预定次数 （例如设定的次数） 时， 或者例如 Turbo译码的结果正确时， 判断出应该结束联合译码。 当判断出 应该结束联合译码时 （歩骤 S1104 , 是）， 则进行后验概率估计 （歩骤 S1105 ) , 并进行参数估计 （歩骤 S1106)。 另一方面， 当判断出不应结束 联合译码时 （歩骤 S1104, 否）， 处理返回到歩骤 S1102, 继续进行比特估 计以及后续处理。
在根据比特类型进行译码时， 例如对于为 A类型比特设定了 4次最大 循环次数， 为 B类型比特设定了 3次最大循环次数、 为 C类型比特设定了 2 次最大循环次数的情况下， 联合译码可以先针对 A类型比特进行译码（最 多四次）， 然后针对 B类型比特进行译码（最多 3次）， 最后针对 C类型比特 进行译码（最大 2次）， 总计 9次。即逐一地针对不同类型的比特进行译码。 另外， 在一种实施方式中， 可以总计循环迭代 4次， 第一、 二次对 A-C三 种类型的比特进行译码， 而第三次对 B、 A两种类型的比特进行译码。 第 四次仅对 A类型比特进行译码。 即总括进行、 中间跳跃式的译码。
图 12示出了依据本发明另一种实施方式的联合译码的处理过程的流 程图。 如图 12所示， 依据本发明的另一种实施方式， 在比特估计单元与 Turbo译码单元进行译码时， 可以首先在歩骤 S1201进行先验概率计算， 估计出先验概率。 先验概率估计以基于后验概率， 即前一帧的后验概率。 然后在歩骤 S1202, 根据该先验概率以及 Turbo译码得出的索引值比特概 率进行比特估计。 接着， 在歩骤 S1203 , 根据比特估计的结果， 再次进 行 Turbo译码。 在歩骤 S1104, 进行后验概率估计， 然后， 在歩骤 S1205, 根据该 Turbo估计的结果， 判断是否应该结束联合译码。 例如当循环次数 已经大于预定次数 （例如设定的次数） 时， 或者例如 Turbo译码的结果正 确时， 判断出应该结束联合译码。 当判断出应该结束联合译码时 （歩骤 S1205 , 是）， 则进行参数估计 （歩骤 S1206), 并结束处理。 另一方面， 当判断出不应结束联合译码时（歩骤 S1205,否），处理返回到歩骤 S1201 , 继续先验概率估计， 并继续进行后续的处理。
本发明以上的装置和方法可以由硬件实现， 也可以由硬件结合软件 实现。 本发明涉及这样的逻辑部件可读程序， 当该程序被逻辑部件所执 行时， 能够使该逻辑部件实现上文所述的装置或构成部件， 或使该逻辑 部件实现上文所述的各种方法或歩骤。 逻辑部件例如现场可编程逻辑部 件、 微处理器、 计算机中使用的处理器等。 本发明还涉及用于存储以上 程序的存储介质， 如硬盘、 磁盘、 光盘、 DVD、 flash, 磁光盘、 存储卡、 存储棒等等。
以上结合具体的实施方式对本发明进行了描述， 但本领域技术人员 应该清楚， 这些描述都是示例性的， 并不是对本发明保护范围的限制。 本领域技术人员可以根据本发明的精神和原理对本发明做出各种变型和 修改， 这些变型和修改也在本发明的范围内。

Claims

权利要求书
1、 一种信源信道联合译码必要性判断方法， 所述方法包括： 判断当前帧的信源编码速率是否与前一帧的信源编码速率相同； 判断当前帧的信源编码速率是否小于预定的信源编码速率阈值； 判断当前帧的信干比是否低于预定的信干比门限值； 以及
当所述当前帧的信源编码速率与前一帧的信源编码速率相同， 并且 所述当前帧的信源编码速率小于所述信源编码速率阈值， 同时所述当前 帧的信干比低于所述信干比门限值时， 确定应该进行信源信道联合译码。
2、 一种信源信道联合译码方法， 所述信源信道联合译码方法包括： 利用 Turbo译码单元对接收的信号进行译码；
对经 Turbo译码单元译码的接收信号进行解帧， 并获得信源编码速 率；
确定当前帧是否是错误接收的语音帧；
当确定出当前帧时接收错误的语音帧时， 利用权利要求 1所述的方法 判断是否需要进行信源信道联合译码； 以及
当判断出需要进行联合译码时， 禾 I」用比特估计单元与所述 Turbo译码 单元的循环迭代译码操作而进行信源信道联合译码。
3、 依据权利要求 2所述的方法， 其中， 所述方法还包括：
设定所述比特估计单元与所述 Turbo译码单元的循环迭代译码操作 的最大次数。
4、 根据权利要求 3所述的方法， 其中， 所述设定包括：
获得发送端语音编码器产生的比特的类型；
根据所述比特的类型设定所述比特估计单元与所述 Turbo译码单元 的循环迭代译码操作的最大次数或对各类型的比特分别设定进行迭代循 环译码操作的最大次数。
5、 根据权利要求 2所述的方法， 其中， 所述方法还包括：
设定在所述比特估计单元与所述 Turbo译码单元的循环迭代操作过 程中， 所述 Turbo译码所使用的 Turbo译码方法。
6、 一种信源信道联合译码必要性判断装置， 所述信源信道联合译码 必要性判断装置包括：
信源编码速率改变判断单元， 所述信源编码速率改变判断单元判断 当前帧的信源编码速率是否与前一帧的信源编码速率相同；
信源编码速率适格性判断单元， 所述信源编码速率适格性判断单元 判断当前帧的信源编码速率是否小于预定的信源编码速率阈值；
当前帧信干比适格性判断单元， 所述当前帧信干比适格性判断单元 判断当前帧的信干比是否低于预定的信干比门限值； 以及
必要性结果确定单元， 当所述信源编码速率改变判断单元判断出当 前帧的信源编码速率与前一帧的信源编码速率相同， 并且所述当前帧的 信源编码速率小于所述信源编码速率阈值， 同时所述当前帧的信干比低 于预定的信干比门限值时， 所述必要性结果确定单元确定应该进行信源 信道联合译码。
7、 一种信源信道联合译码装置， 所述信源信道联合译码装置包括：
Turbo 译码单元， 用于对接收的信号进行译码；
解帧单元， 所述解帧单元用于对经 Turbo译码单元译码的接收信号进 行解帧而获得信源编码速率；
错误语音帧判断单元， 所述错误语音帧判断单元用于确定当前帧是 否是错误接收的语音帧；
利用权利要求 6所述的信源信道联合译码必要性判断装置， 根据所述 解帧单元所获得的信源编码速率判断是否需要进行信源信道联合译码； 以及
联合译码单元， 所述联合译码单元利用比特估计单元与所述 Turbo译 码单元的循环迭代操作而进行信源信道联合译码。
8、 根据权利要求 7所述的信源信道联合译码装置， 其中， 所述信源 信道联合译码装置还包括：
最大迭代次数设定单元， 设定所述比特估计单元与所述 Turbo译码单 元的循环迭代译码操作的最大次数。
9、 根据权利要求 8所述的信源信道联合译码装置， 其中，
所述解帧单元获得发送端语音编码器产生的比特的类型，
最大迭代次数设定单元根据所述比特的类型设定所述比特估计单元 与所述 Turbo译码单元进行的循环迭代译码操作的最大次数， 或对各类型 的比特分别设定进行迭代循环译码操作的最大次数。
10、 根据权利要求 7所述的信源信道联合译码装置， 其中， 所述信源 信道联合译码装置还包括：
Turbo译码方法设定单元，设定在所述比特估计单元与所述 Turbo译码 单元的循环迭代操作过程中， 所述 Turbo译码所使用的 Turbo译码方法。
11、一种接收机，所述接收机使用权利要求 1-5中任一项所述的方法， 或包括权利要求 6-10中任一项所述的装置。
12、 根据权利要求 11所述的接收机， 所述接收机包括信道估计单元， 所述信道估计单元估计信干比， 所述信源信道联合译码装置或所述信源 信道联合译码方法在进行是否需要进行信源信道联合译码的判断时使用 所述信干比。
13、 一种逻辑部件可读程序， 当所述逻辑部件可读程序被逻辑部件 执行时， 能够使所述逻辑部件作为权利要求 6-10中任一项所述的装置工 作或使所述逻辑部件实现权利要求 1-5中任一项所述的方法。
14、 一种逻辑部件可读有形存储介质， 所述有形存储介质存储有权 利要求 13所述的逻辑部件可读程序。
PCT/CN2011/074777 2011-05-27 2011-05-27 联合译码装置及方法、必要性判断方法和装置、接收机 WO2012162870A1 (zh)

Priority Applications (3)

Application Number Priority Date Filing Date Title
PCT/CN2011/074777 WO2012162870A1 (zh) 2011-05-27 2011-05-27 联合译码装置及方法、必要性判断方法和装置、接收机
CN2011800637673A CN103283150A (zh) 2011-05-27 2011-05-27 联合译码装置及方法、必要性判断方法和装置、接收机
US14/090,451 US20140164002A1 (en) 2011-05-27 2013-11-26 Joint decoding apparatus and method, necessity judging method and apparatus, and receiver

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
PCT/CN2011/074777 WO2012162870A1 (zh) 2011-05-27 2011-05-27 联合译码装置及方法、必要性判断方法和装置、接收机

Related Child Applications (1)

Application Number Title Priority Date Filing Date
US14/090,451 Continuation US20140164002A1 (en) 2011-05-27 2013-11-26 Joint decoding apparatus and method, necessity judging method and apparatus, and receiver

Publications (1)

Publication Number Publication Date
WO2012162870A1 true WO2012162870A1 (zh) 2012-12-06

Family

ID=47258263

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/CN2011/074777 WO2012162870A1 (zh) 2011-05-27 2011-05-27 联合译码装置及方法、必要性判断方法和装置、接收机

Country Status (3)

Country Link
US (1) US20140164002A1 (zh)
CN (1) CN103283150A (zh)
WO (1) WO2012162870A1 (zh)

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US8972254B2 (en) * 2011-06-28 2015-03-03 Utah State University Turbo processing for speech recognition with local-scale and broad-scale decoders
CN111081226B (zh) * 2018-10-18 2024-02-13 北京搜狗科技发展有限公司 语音识别解码优化方法及装置
CN113450809B (zh) * 2021-08-30 2021-11-30 北京百瑞互联技术有限公司 语音数据处理方法、系统及介质

Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2004066546A1 (en) * 2003-01-21 2004-08-05 Sony Ericsson Mobile Communications Ab Speed data receiver with detection of channel coding rate
CN1805565A (zh) * 2005-01-14 2006-07-19 华为技术有限公司 对不连续发送制式下的自适应多速率语音编码的译码方法
US20090041087A1 (en) * 2006-03-03 2009-02-12 Hiroyuki Yamasuge Wireless communication apparatus and wireless communication method
CN101656542A (zh) * 2009-09-08 2010-02-24 广州杰赛科技股份有限公司 双二元卷积Turbo码译码方法和装置
CN101777969A (zh) * 2010-01-12 2010-07-14 浙江大学 一种基于四发射天线准正交空时分组码的编译码方法

Family Cites Families (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6625776B1 (en) * 1998-09-30 2003-09-23 Northrop Grumman Corporation Adaptive coding scheme for a processing communications satellite
CN1225854C (zh) * 2002-06-13 2005-11-02 华为技术有限公司 一种自适应调制与编码方法
US7639749B2 (en) * 2004-12-14 2009-12-29 Ittiam Systems (P) Ltd. System and method for improving the performance of OFDM systems
FR2910200A1 (fr) * 2006-12-18 2008-06-20 Commissariat Energie Atomique Recepteur a decodage conditionnel

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2004066546A1 (en) * 2003-01-21 2004-08-05 Sony Ericsson Mobile Communications Ab Speed data receiver with detection of channel coding rate
CN1805565A (zh) * 2005-01-14 2006-07-19 华为技术有限公司 对不连续发送制式下的自适应多速率语音编码的译码方法
US20090041087A1 (en) * 2006-03-03 2009-02-12 Hiroyuki Yamasuge Wireless communication apparatus and wireless communication method
CN101656542A (zh) * 2009-09-08 2010-02-24 广州杰赛科技股份有限公司 双二元卷积Turbo码译码方法和装置
CN101777969A (zh) * 2010-01-12 2010-07-14 浙江大学 一种基于四发射天线准正交空时分组码的编译码方法

Also Published As

Publication number Publication date
US20140164002A1 (en) 2014-06-12
CN103283150A (zh) 2013-09-04

Similar Documents

Publication Publication Date Title
Fingscheidt et al. Softbit speech decoding: A new approach to error concealment
TWI351657B (en) Method and apparatus for scaling and quantizing so
KR102308579B1 (ko) 오디오 대역폭 선택
JP3943127B2 (ja) Ｔｄｍａ無線システムにおけるソフト誤り補正
JP6077011B2 (ja) 冗長フレーム符号化および復号のためのデバイス
KR101422379B1 (ko) 서브 밴드 코딩 디코더에서 손실 패킷들의 은닉
CN106411458B (zh) 用于对接收数据进行控制的设备和方法
JP2001513598A (ja) 可変レート通信システムにて受信データのレートを決定する方法及び装置
JP4875249B2 (ja) 自動音声認識実行方法
US8578247B2 (en) Bit error management methods for wireless audio communication channels
US8036323B2 (en) Method and system for decoding single antenna interference cancellation (SAIC) and redundancy processing adaptation using frame process
JP2008535323A (ja) 信号変調フォーマットタイプの検出
EP3533147A1 (en) Iterative decoding of polar code with bit-flipping of unreliable bits
KR20150023858A (ko) 반복 응답 결합 메커니즘을 가진 통신 시스템 및 그 동작 방법
WO2012034100A1 (en) Data classification based on error detection check and decoder metrics in a wireless communication system
KR20150003874A (ko) 블라인드 디코딩 메카니즘을 사용하는 통신 시스템 및 그 동작 방법
WO2012162870A1 (zh) 联合译码装置及方法、必要性判断方法和装置、接收机
US6004028A (en) Device and method for receiving and reconstructing signals with improved perceived signal quality
KR20160096191A (ko) 음성/오디오 비트스트림 디코딩 방법 및 장치
WO2009117877A1 (en) Estimation of error propagation probability to improve performance of decision-feedback based systems
US9037942B2 (en) Modified joint source channel decoder
WO2014032578A1 (zh) 一种确定软比特信息的方法及装置
US8612825B2 (en) Data classification based on cyclic redundancy check and decoder metric
US5406562A (en) Bit error rate estimation
CN114362763A (zh) 联合译码方法及装置、存储介质及电子设备

Legal Events

Date Code Title Description
121 Ep: the epo has been informed by wipo that ep was designated in this application

Ref document number: 11866711

Country of ref document: EP

Kind code of ref document: A1

NENP Non-entry into the national phase

Ref country code: DE

122 Ep: pct application non-entry in european phase

Ref document number: 11866711

Country of ref document: EP

Kind code of ref document: A1