CN114095879A - Voice quality monitoring method and device - Google Patents
Voice quality monitoring method and device Download PDFInfo
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- CN114095879A CN114095879A CN202111321158.5A CN202111321158A CN114095879A CN 114095879 A CN114095879 A CN 114095879A CN 202111321158 A CN202111321158 A CN 202111321158A CN 114095879 A CN114095879 A CN 114095879A
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W4/00—Services specially adapted for wireless communication networks; Facilities therefor
- H04W4/06—Selective distribution of broadcast services, e.g. multimedia broadcast multicast service [MBMS]; Services to user groups; One-way selective calling services
- H04W4/10—Push-to-Talk [PTT] or Push-On-Call services
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
- G10L25/48—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use
- G10L25/51—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use for comparison or discrimination
- G10L25/60—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use for comparison or discrimination for measuring the quality of voice signals
Abstract
The invention discloses a voice quality monitoring method and a device, belonging to the following steps that a terminal acquires recorded voice, frames the voice to obtain a plurality of frame packets, and all the frame packets are sequentially coded and respectively packaged into a plurality of data packets; all the data packets are sequentially sent to a server, the server receives the data packets and then reads the frame packet codes, and the intermittent condition of the statistical codes is identified and recorded; the server compares the received frame packet with the missing frame packet to obtain a receiving difference value H, and calculates the voice quality of the current voice frame packet according to the receiving difference value H. The comprehensive judgment method for judging the voice quality by combining the network conditions is realized.
Description
Technical Field
The invention relates to the field of methods for monitoring and optimizing voice quality in a talkback process, in particular to a method and a device for monitoring the voice quality.
Background
The POC talkback technology is a mobile communication system established by a carrier mobile communication network and used for dispatching and commanding, the POC talkback can provide a quick PTT service, and can provide a one-to-many group calling service function of a PMR system, and different from the PMR network with limited channel capacity, a PoC platform supports the establishment of any number of virtual channels and calling groups, and the virtual channels and the calling groups can inevitably pass the processes of framing, packaging, sending and the like in the necessary voice signal processing of the recorded voice in the talkback process;
in the prior art, information such as the sending condition, the playing quality, the packet loss amount and the like after the voice signal processing is finished cannot be effectively summarized and reported in real time, and an effective judgment method is not provided for the above factors in the prior art, or the judgment method does not combine the network quality at the sending time as a reference, so that the voice quality cannot be scientifically and accurately judged.
Disclosure of Invention
The embodiment of the invention provides a voice quality monitoring method and a voice quality monitoring device, which at least solve the problems that the existing judging method is unscientific and cannot effectively consider network environment factors.
The above technical problems are solved by the following methods
There is provided a voice quality monitoring method comprising the steps of,
the terminal acquires the recorded voice and divides the voice into frames to obtain a plurality of frame packets, and all the frame packets are sequentially encoded and respectively packaged into a plurality of data packets;
all the data packets are sequentially sent to a server, the server receives the data packets and then reads the frame packet codes, and the intermittent condition of the statistical codes is identified and recorded;
the server compares the received frame packet with the missing frame packet to obtain a receiving difference value H, and calculates the voice quality of the current voice frame packet according to the receiving difference value H.
Furthermore, the method also comprises the steps of framing the voice into a plurality of frame packets,
all the data packets are sequentially sent and record sending time timestamps, and all the data packets are sequentially sent to a server;
the server receives the data packet and records the receiving time of the corresponding data packet;
and defining a jitter value i according to the time difference between the receiving time and the sending time timestamp, wherein the average number of the jitter values i is an average difference value P.
Further, the calculating and judging the voice quality of the voice frame packet according to the time delay difference comprises
The number of all frame packets is marked as D;
the receiving difference value H is substituted into the following formula
Obtaining a difference ratio R;
when R is less than or equal to 0, recording the voice quality as excellent, or
When R is any one of values between 0 and 100, substituting the difference ratio R into the following formula:
M＝1+0·035R+R(R-60)(100-R)×7×10-6
obtaining a speech score value M, or
When R is 100 or more, it is considered that the voice quality is poor.
Further, the time difference between the sending and receiving time stamps of each frame packet is n1-nnSubstituting the time difference value of each frame packet into the following formula:
and obtaining an average time delay amount S, judging the network strength according to the average time delay amount S, recording the transmission as network jitter if the average time delay amount S is equal to or exceeds a preset jitter threshold, and recording the transmission as normal if the average time delay amount S is less than the preset jitter threshold.
Further, all the data packets are sequentially sent and recorded with sending time timestamps, all the data packets are sequentially sent to the server, and the steps further comprise,
all data packets comprise media time stamps P and sending time stamps Y;
and sequentially substituting the newly received nth data packet, the media time stamp P corresponding to the nth-1 data packet and the transmission time timestamp Y into the following formula, wherein n is 1 … … n:
D(n,n-1)＝(Pn-Pn-1)-(Yn-Yn-1)
obtain corresponding jitter value D(n,n-1)；
Taking the jitter value D(n,n-1)Substituting the following formula:
obtaining a dynamic network residual jitter value LnWherein L is(n-1)Dynamic network residual jitter value L calculated for the process of the (n-1) th data packet(n-1)
And sequentially calculating the dynamic network residual error jitter values L corresponding to the received 1 st to nth data packets.
There is provided a speech quality monitoring apparatus comprising a speech quality monitoring method as described above, comprising
The encapsulation unit is used for framing the voice after the terminal acquires the recorded voice, sequentially coding a plurality of obtained frame packets and respectively encapsulating the plurality of obtained frame packets into a plurality of data packets;
the uploading unit is used for sequentially sending all the data packets to the server;
the identification unit is used for reading the frame packet codes after the server receives the data packets, identifying and recording the intermittent condition of the statistical codes;
the server compares the received frame packet with the missing frame packet to obtain a receiving difference value H, and calculates the voice quality of the current voice frame packet according to the receiving difference value H.
Furthermore, the method also comprises the steps of framing the voice into a plurality of frame packets,
the sending recording unit is used for sending all the data packets in sequence and recording the sending time timestamps, and all the data packets are sent to the server in sequence;
the receiving and recording unit is used for recording the receiving time of the corresponding data packet by the server after the server receives the data packet;
and the average difference value calculating unit is used for defining a time difference value according to the receiving time and the sending time timestamp as a jitter value i, and the average number of the jitter values i is an average difference value P.
Further, the method further comprises a voice quality judgment unit, which is used for realizing that:
the method for calculating and judging the voice quality of the voice frame packet according to the time delay difference comprises the following steps
The number of all frame packets is marked as D;
the receiving difference value H is substituted into the following formula
Obtaining a difference ratio R;
when R is less than or equal to 0, recording the voice quality as excellent, or
When R is any one of values between 0 and 100, substituting the difference ratio R into the following formula:
M＝1+0.035R+R(R-60)(100-R)×7×10-6
obtaining a speech score value M, or
When R is 100 or more, it is considered that the voice quality is poor.
Further, the system further comprises a network strength judging module, which is used for realizing that:
the time difference value between the time stamp of each frame packet and the time stamp of each frame packet is n1-nnSubstituting the time difference value of each frame packet into the following formula:
and obtaining an average time delay amount S, judging the network strength according to the average time delay amount S, recording the transmission as network jitter if the average time delay amount S is equal to or exceeds a preset jitter threshold, and recording the transmission as normal if the average time delay amount S is less than the preset jitter threshold.
10. A speech quality monitoring device according to claim 7, characterized by:
the device also comprises a network residual jitter value calculation module, which is used for realizing that:
all the data packets are sequentially sent and record sending time timestamps, all the data packets are sequentially sent to a server, the steps also comprise,
all data packets comprise media time stamps P and sending time stamps Y;
and sequentially substituting the newly received nth data packet, the media time stamp P corresponding to the nth-1 data packet and the transmission time timestamp Y into the following formula, wherein n is 1 … … n:
D(n,n-1)＝(Pn-Pn-1)-(Yn-Yn-1)
obtain corresponding jitter value D(n,n-1)；
Taking the jitter value D(n,n-1)Substituting the following formula:
obtaining a dynamic network residual jitter value LnWherein L is(n-1)Dynamic network residual jitter value L calculated for the process of the (n-1) th data packet(n-1)
And sequentially calculating the dynamic network residual error jitter values L corresponding to the received 1 st to nth data packets.
Drawings
The accompanying drawings, which are included to provide a further understanding of the invention and are incorporated in and constitute a part of this application, illustrate embodiment(s) of the invention and together with the description serve to explain the invention without limiting the invention. In the drawings:
FIG. 1 is a diagram of the execution logic of the present invention.
Detailed Description
In order to make the technical solutions of the present invention better understood, the technical solutions in the embodiments of the present invention will be clearly and completely described below with reference to the drawings in the embodiments of the present invention, and it is obvious that the described embodiments are only a part of the embodiments of the present invention, and not all of the embodiments. All other embodiments, which can be derived by a person skilled in the art from the embodiments given herein without making any creative effort, shall fall within the protection scope of the present invention.
It should be noted that the terms in the description, the claims and the drawings of the present invention are used for distinguishing between similar objects, and the terms first, second and the like are not used for describing a detailed technical solution of the present invention. It is to be understood that the data so used is interchangeable under appropriate circumstances such that the embodiments of the invention described herein are capable of operation in sequences other than those illustrated or described herein. Furthermore, the terms "comprises," "comprising," and "having," and any variations thereof, are intended to cover a non-exclusive inclusion, such that a process, method, system, article, or apparatus that comprises a list of steps or elements is not necessarily limited to those steps or elements expressly listed, but may include other steps or elements not expressly listed or inherent to such process, method, article, or apparatus.
Example one
As shown in fig. 1, there is provided a voice quality monitoring method,
the terminal acquires the recorded voice and divides the voice into frames to obtain a plurality of frame packets, and all the frame packets are sequentially encoded and respectively packaged into a plurality of data packets;
all the data packets are sequentially sent to a server, the server receives the data packets and then reads the frame packet codes, and the intermittent condition of the statistical codes is identified and recorded;
the server compares the received frame packet with the missing frame packet to obtain a receiving difference value H, and calculates the voice quality of the current voice frame packet according to the receiving difference value H.
Specifically, the terminal carries a recording module to record the received voice information, the voice stream information is sequentially divided into a plurality of frame packets after the voice stream is received, each frame packet comprises a packet header, a voice frame and a packet tail which are specified by an agreed format of mutual communication, each frame packet is numbered in sequence and sequentially sent to a receiving end of the server, and the receiving end of the server reads frame packet codes, identifies and counts the intermittent condition of the codes and records the intermittent condition.
Optionally, the framing the speech into a plurality of frame packets further includes,
all the data packets are sequentially sent and record sending time timestamps, and all the data packets are sequentially sent to a server;
the server receives the data packet and records the receiving time of the corresponding data packet;
and defining a jitter value i according to the time difference between the receiving time and the sending time timestamp, wherein the average number of the jitter values i is an average difference value P.
In implementation, the data packet is each frame packet, the frame packet records a time stamp of the sending time, compares the time stamp with the receiving time of the server, compares the difference values to obtain a time difference, and defines the time difference as a jitter value i, the average number of each jitter value i is an average difference value P, and the average difference value P is used for representing the sending time length of the data packet.
Optionally, the step of calculating and judging the voice quality of the voice frame packet according to the time delay difference comprises
The number of all frame packets is marked as D;
the receiving difference value H is substituted into the following formula
Obtaining a difference ratio R;
when R is less than or equal to 0, recording the voice quality as excellent, or
When R is any one of values between 0 and 100, substituting the difference ratio R into the following formula:
M＝1+0.035R+R(R-60)(100-R)×7×10-6
obtaining a speech score value M, or
When R is 100 or more, it is considered that the voice quality is poor.
In the implementation of the method, the first step of the method,
optionally, a time difference between the transmission time of each frame packet and the received time stamp is n1-nnSubstituting the time difference value of each frame packet into the following formula:
and obtaining an average time delay amount S, judging the network strength according to the average time delay amount S, recording the transmission as network jitter if the average time delay amount S is equal to or exceeds a preset jitter threshold, and recording the transmission as normal if the average time delay amount S is less than the preset jitter threshold.
5. A speech quality monitoring method according to claim 2, characterized by: all the data packets are sequentially sent and record sending time timestamps, all the data packets are sequentially sent to a server, the steps also comprise,
all data packets comprise media time stamps P and sending time stamps Y;
and sequentially substituting the newly received nth data packet, the media time stamp P corresponding to the nth-1 data packet and the transmission time timestamp Y into the following formula, wherein n is 1 … … n:
D(n,n-1)＝(Pn-Pn-1)-(Yn-Yn-1)
obtain corresponding jitter value D(n,n-1)；
Taking the jitter value D(n,n-1)Substituting the following formula:
obtaining a dynamic network residual jitter value LnWherein L is(n-1)Dynamic network residual jitter value L calculated for the process of the (n-1) th data packet(n-1)
And sequentially calculating the dynamic network residual error jitter values L corresponding to the received 1 st to nth data packets.
A voice quality monitoring device comprises the voice quality monitoring method, and comprises
The encapsulation unit is used for framing the voice after the terminal acquires the recorded voice, sequentially coding a plurality of obtained frame packets and respectively encapsulating the plurality of obtained frame packets into a plurality of data packets;
the uploading unit is used for sequentially sending all the data packets to the server;
the identification unit is used for reading the frame packet codes after the server receives the data packets, identifying and recording the intermittent condition of the statistical codes;
the server compares the received frame packet with the missing frame packet to obtain a receiving difference value H, and calculates the voice quality of the current voice frame packet according to the receiving difference value H.
Optionally, the method further comprises framing the speech into a plurality of frame packets,
the sending recording unit is used for sending all the data packets in sequence and recording the sending time timestamps, and all the data packets are sent to the server in sequence;
the receiving and recording unit is used for recording the receiving time of the corresponding data packet by the server after the server receives the data packet;
and the average difference value calculating unit is used for defining a time difference value according to the receiving time and the sending time timestamp as a jitter value i, and the average number of the jitter values i is an average difference value P. Optionally, the method further includes a voice quality determination unit, configured to implement:
the method for calculating and judging the voice quality of the voice frame packet according to the time delay difference comprises the following steps
The number of all frame packets is marked as D;
the receiving difference value H is substituted into the following formula
Obtaining a difference ratio R;
when R is less than or equal to 0, recording the voice quality as excellent, or
When R is any one of values between 0 and 100, substituting the difference ratio R into the following formula:
M＝1+0.035R+R(R-60)(l00-R)×7×10-6
obtaining a speech score value M, or
When R is 100 or more, it is considered that the voice quality is poor.
Optionally, the system further includes a network strength determining module, configured to implement:
the time difference value between the time stamp of each frame packet and the time stamp of each frame packet isn1-nnSubstituting the time difference value of each frame packet into the following formula:
and obtaining an average time delay amount S, judging the network strength according to the average time delay amount S, recording the transmission as network jitter if the average time delay amount S is equal to or exceeds a preset jitter threshold, and recording the transmission as normal if the average time delay amount S is less than the preset jitter threshold.
Optionally, the method further includes a network residual jitter value calculation module, configured to implement:
all the data packets are sequentially sent and record sending time timestamps, all the data packets are sequentially sent to a server, the steps also comprise,
all data packets comprise media time stamps P and sending time stamps Y;
and sequentially substituting the newly received nth data packet, the media time stamp P corresponding to the nth-1 data packet and the transmission time timestamp Y into the following formula, wherein n is 1 … … n:
D(n,n-1)＝(Pn-Pn-1)-(Yn-Yn-1)
obtain corresponding jitter value D(n,n-1)；
Taking the jitter value D(n,n-1)Substituting the following formula:
obtaining a dynamic network residual jitter value LnWherein L is(n-1)Dynamic network residual jitter value L calculated for the process of the (n-1) th data packet(n-1)
And sequentially calculating the dynamic network residual error jitter values L corresponding to the received 1 st to nth data packets.
In the above embodiments of the present invention, the descriptions of the respective embodiments have respective emphasis, and for parts that are not described in detail in a certain embodiment, reference may be made to related descriptions of other embodiments.
In the embodiments provided in the present application, it should be understood that the disclosed technology can be implemented in other ways. The above-described embodiments of the apparatus are merely illustrative, and for example, the division of the units may be a logical division, and in actual implementation, there may be another division, for example, multiple units or components may be combined or integrated into another system, or some features may be omitted, or not executed. In addition, the shown or discussed mutual coupling or direct coupling or communication connection may be an indirect coupling or communication connection through some interfaces, units or modules, and may be in an electrical or other form.
The units described as separate parts may or may not be physically separate, and parts displayed as units may or may not be physical units, may be located in one place, or may be distributed on a plurality of units. Some or all of the units can be selected according to actual needs to achieve the purpose of the solution of the embodiment.
In addition, functional units in the embodiments of the present invention may be integrated into one processing unit, or each unit may exist alone physically, or two or more units are integrated into one unit. The integrated unit can be realized in a form of hardware, and can also be realized in a form of a software functional unit.
The integrated unit, if implemented in the form of a software functional unit and sold or used as a stand-alone product, may be stored in a computer readable storage medium. Based on such understanding, the technical solution of the present invention may be embodied in the form of a software product, which is stored in a storage medium and includes instructions for causing a computer device (which may be a personal computer, a server, or a network device) to execute all or part of the steps of the method according to the embodiments of the present invention. And the aforementioned storage medium includes: a U-disk, a Read-Only Memory (ROM), a Random Access Memory (RAM), a removable hard disk, a magnetic or optical disk, and other various media capable of storing program codes.
The foregoing is only a preferred embodiment of the present invention, and it should be noted that, for those skilled in the art, various modifications and decorations can be made without departing from the principle of the present invention, and these modifications and decorations should also be regarded as the protection scope of the present invention.
Claims (10)
1. A voice quality monitoring method is characterized in that: comprises the following steps of (a) carrying out,
the terminal acquires the recorded voice and divides the voice into frames to obtain a plurality of frame packets, and all the frame packets are sequentially encoded and respectively packaged into a plurality of data packets;
all the data packets are sequentially sent to a server, the server receives the data packets and then reads the frame packet codes, and the intermittent condition of the statistical codes is identified and recorded;
the server compares the received frame packet with the missing frame packet to obtain a receiving difference value H, and calculates the voice quality of the current voice frame packet according to the receiving difference value H.
2. A speech quality monitoring method according to claim 1, characterized by: and framing the speech into a plurality of frame packets,
all the data packets are sequentially sent and record sending time timestamps, and all the data packets are sequentially sent to a server;
the server receives the data packet and records the receiving time of the corresponding data packet;
and defining a jitter value i according to the time difference between the receiving time and the sending time timestamp, wherein the average number of the jitter values i is an average difference value P.
3. A speech quality monitoring method according to claim 1, characterized by: the method for calculating and judging the voice quality of the voice frame packet according to the time delay difference comprises the following steps
The number of all frame packets is marked as D;
the receiving difference value H is substituted into the following formula
Obtaining a difference ratio R;
when R is less than or equal to 0, recording the voice quality as excellent, or
When R is any one of values between 0 and 100, substituting the difference ratio R into the following formula:
M＝1+0.035R+R(R-60)(100-R)×7×10-6
obtaining a speech score value M, or
When R is 100 or more, it is considered that the voice quality is poor.
4. A speech quality monitoring method according to claim 2, characterized by:
the time difference value between the time stamp of each frame packet and the time stamp of each frame packet is n1-nnSubstituting the time difference value of each frame packet into the following formula:
and obtaining an average time delay amount S, judging the network strength according to the average time delay amount S, recording the transmission as network jitter if the average time delay amount S is equal to or exceeds a preset jitter threshold, and recording the transmission as normal if the average time delay amount S is less than the preset jitter threshold.
5. A speech quality monitoring method according to claim 2, characterized by: all the data packets are sequentially sent and record sending time timestamps, all the data packets are sequentially sent to a server, the steps also comprise,
all data packets comprise media time stamps P and sending time stamps Y;
and sequentially substituting the newly received nth data packet, the media time stamp P corresponding to the nth-1 data packet and the transmission time timestamp Y into the following formula, wherein n is 1 … … n:
D(n,n-1)＝(Pn-Pn-1)-(Yn-Yn-1)
obtain corresponding jitter value D(n,n-1)；
Taking the jitter value D(n,n-1)Substituting the following formula:
obtaining a dynamic network residual jitter value LnWherein L is(n-1)Dynamic network residual jitter value L calculated for the process of the (n-1) th data packet(n-1)
And sequentially calculating the dynamic network residual error jitter values L corresponding to the received 1 st to nth data packets.
6. A voice quality monitoring apparatus comprising a voice quality monitoring method according to claims 1 to 5, characterized in that: comprises that
The encapsulation unit is used for framing the voice after the terminal acquires the recorded voice, sequentially coding a plurality of obtained frame packets and respectively encapsulating the plurality of obtained frame packets into a plurality of data packets;
the uploading unit is used for sequentially sending all the data packets to the server;
the identification unit is used for reading the frame packet codes after the server receives the data packets, identifying and recording the intermittent condition of the statistical codes;
the server compares the received frame packet with the missing frame packet to obtain a receiving difference value H, and calculates the voice quality of the current voice frame packet according to the receiving difference value H.
7. A speech quality monitoring device according to claim 6, characterized in that: and framing the speech into a plurality of frame packets,
the sending recording unit is used for sending all the data packets in sequence and recording the sending time timestamps, and all the data packets are sent to the server in sequence;
the receiving and recording unit is used for recording the receiving time of the corresponding data packet by the server after the server receives the data packet;
and the average difference value calculating unit is used for defining a time difference value according to the receiving time and the sending time timestamp as a jitter value i, and the average number of the jitter values i is an average difference value P.
8. A speech quality monitoring device according to claim 6, characterized in that:
the voice quality judging unit is used for realizing that:
the method for calculating and judging the voice quality of the voice frame packet according to the time delay difference comprises the following steps
The number of all frame packets is marked as D;
the receiving difference value H is substituted into the following formula
Obtaining a difference ratio R;
when R is less than or equal to 0, recording the voice quality as excellent, or
When R is any one of values between 0 and 100, substituting the difference ratio R into the following formula:
M＝1+0.035R+R(R-60)(100-R)×7×10-6
obtaining a speech score value M, or
When R is 100 or more, it is considered that the voice quality is poor.
9. A speech quality monitoring device according to claim 7, characterized by:
the system also comprises a network intensity judging module used for realizing that:
the time difference value between the time stamp of each frame packet and the time stamp of each frame packet is n1-nnSubstituting the time difference value of each frame packet into the following formula:
and obtaining an average time delay amount S, judging the network strength according to the average time delay amount S, recording the transmission as network jitter if the average time delay amount S is equal to or exceeds a preset jitter threshold, and recording the transmission as normal if the average time delay amount S is less than the preset jitter threshold.
10. A speech quality monitoring device according to claim 7, characterized by:
the device also comprises a network residual jitter value calculation module, which is used for realizing that:
all the data packets are sequentially sent and record sending time timestamps, all the data packets are sequentially sent to a server, the steps also comprise,
all data packets comprise media time stamps P and sending time stamps Y;
and sequentially substituting the newly received nth data packet, the media time stamp P corresponding to the nth-1 data packet and the transmission time timestamp Y into the following formula, wherein n is 1 … … n:
D(n,n-1)＝(Pn-Pn-1)-(Yn-Yn-1)
obtain corresponding jitter value D(n,n-1)；
Taking the jitter value D(n,n-1)Substituting the following formula:
obtaining a dynamic network residual jitter value LnWherein L is(n-1)Dynamic network residual jitter value L calculated for the process of the (n-1) th data packet(n-1)
And sequentially calculating the dynamic network residual error jitter values L corresponding to the received 1 st to nth data packets.
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