CN103345376A - Method for monitoring digital audio signal volume - Google Patents
Method for monitoring digital audio signal volume Download PDFInfo
- Publication number
- CN103345376A CN103345376A CN2013102858135A CN201310285813A CN103345376A CN 103345376 A CN103345376 A CN 103345376A CN 2013102858135 A CN2013102858135 A CN 2013102858135A CN 201310285813 A CN201310285813 A CN 201310285813A CN 103345376 A CN103345376 A CN 103345376A
- Authority
- CN
- China
- Prior art keywords
- loudness
- digital audio
- video signals
- input
- data
- Prior art date
- Legal status (The legal status 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 status listed.)
- Granted
Links
Images
Landscapes
- Two-Way Televisions, Distribution Of Moving Picture Or The Like (AREA)
Abstract
The invention provides a method for monitoring digital audio signal volume. The method is achieved by combined action of hardware and programs. A digital audio signal volume monitoring card which comprises a digital audio acquisition unit, a decoding unit, a volume computing unit and a communication unit is adopted for the hardware. A computer accesses the volume monitoring card through the programs, and under the condition of not affecting other producing and playing normal operation, the volume of input audio signals and the dynamic range of the volume are recorded, displayed, stored and analyzed; the digital audio volume monitoring card acquires the digital audio signals and computes the corresponding volume of the audio signals; the digital audio acquisition unit is provided with three sets of input interfaces, and each set of the interface is provided with three input interface circuits suitable for digital audios with AES/EBU signals or S/PDIF signals, the input interface circuits are used for receiving the AES/EBU signals or the S/PDIF signals input in a balanced mode and imbalanced mode respectively; the volume computing unit computes the volume data result.
Description
Technical field
The present invention is a kind of loudness monitoring method of digital audio and video signals, gather multipath audio signal by the digital audio and video signals loudness monitoring card that carries the filtering of sound signal loudness and ranking operation, uses algorithm software calculates loudness size and the dynamic range of sound signal, and signal loudness state is recorded, shows, stores and analyzes.Monitoring by to the recording, make, transmit and play of program audio signal the time improves making and the play quality of audio program.This monitoring method can be used in the middle of all kinds of multimedia equipments and program that contain sound signal.By the mode of digital audio and video signals loudness monitoring card, can be embedded in dissimilar computer based multimedia programmings and make and playback equipment, can be widely used in professional audio fields such as radio and television.The present invention can offer audio frequency and the audio/video program producer uses, and also can be used for independently, sound signal loudness monitoring equipment uses for various types of programs making, transmission and multimedia terminal equipment.
Background technology
AES/EBU DAB agreement is by (world) Audio Engineering Society (Audio Engineering Society, be called for short AES) and (the European Broadcast Union of European Broadcasting Union, be called for short EBU) formulate, be widely used in professional audio fields such as radio and television.The AES/EBU digital audio and video signals can use balance or non-equilibrium mode to transmit, and has adopted two-phase encoding flag technology.S/PDIF DAB agreement is formulated by Sony and PHILIPS Co., and the S/PDIF full name is Sony/Philips Digital Interface, means Sony/philips digital interface, is a kind of DAB agreement of using towards consumer level.S/PDIF uses the transmission of two-phase marker technique equally, with the AES/EBU digital audio and video signals identical frame structure is arranged, as shown in Figure 1.But S/PD IF compares with AES/EBU different electric interfaces specifications and channel status piece are arranged, as shown in Table 1 and Table 2.
PCI express (be called for short PCI-e) has adopted popular point-to-point connected in series, can only realize one-way transmission with respect to traditional PCI bus in the single time cycle, and two single worker of PCI-e connects transfer rate and the quality that can provide higher.
The loudness of sound signal refers to that people's ear is to the sonority of sound institute perception.Loudness not only depends on the amplitude size of sound, also depends on the factors such as frequency height, number of channels, dynamic size of this sound, and concrete numerical value is referring to the Fig. 2 in the description of drawings.
In the collection of audio/video program, making, transmission, broadcast, owing to lack the equipment that can intuitively reflect this sound signal loudness, program making personnel and terminal user can only reach the purpose of regulating the loudness size by observing and regulate the Audio Meter size.But, because loudness depends on amplitude, frequency, the sound channel and dynamic of acoustical signal, only can't satisfy the needs that loudness is adjusted based on the adjustment of level.Therefore, in radio and television and other multimedia programmings, people think that through regular meeting the loudness between the program of each same even (comprising the different broadcasting agencies that make) is very inconsistent, and loudness is suddenly big or suddenly small.For example, insert advertisement in a formal program, the loudness of formal program has obviously different with the loudness of advertising programme mostly.
For various dissimilar programs, the dynamic range of loudness has very big difference.Generally speaking, classical music, film audio frequency have bigger dynamic range, and programs such as pop music, dynamic range is less.Audio program finally will broadcast at different broadcast media, as televisor, radio, home theater sound equipment, mobile phone, MP3 player etc.Be subjected to the restriction of sound equipment hardware performance, different broadcast media has different sound broadcast qualities.For example, the war subject matter film that bigger loudness dynamic range is arranged, sound clips such as blast have very big loudness, and the dialogue loudness in the film just can not be very big.When this film when household audio and video system broadcasts, because home theater has the good sound output performance, the user can very clearly hear this two kinds of sound, if and this film just broadcasts by televisor without the audio frequency processing, general televisor is because speaker system functional limitation, when the user can not hear dialogue, because the loudness of blast sound is excessive, sound defects such as distorsion will appear; And when the user reduced volume and makes that explosive sound is unlikely to distortion, dialogue sound was too little again, makes the user not hear.Relative, if dynamic very big classical music program of loudness, do the compression of loudness at the TV broadcast, when in televisor, broadcasting, the user can think that loudness is comparatively balanced, excessive and the too small situation of sound can not occur, and when the user uses household audio and video system to broadcast, will perceive and the sound quality variation of bringing after the music loudness dynamic compression.For example, the French horn in the symphony and violin have different tone colors and loudness, and this loudness differences itself is exactly the part of music aesthetic feeling, and have passed through after the compression, and it is consistent that their loudness becomes, and this aesthetic feeling has just lacked.
So, when the program making personnel when making audio program, must consider the broadcast medium that program is faced, and with this as the program making foundation, the loudness of suitably adjusting audio program is dynamic, guarantees user's listening quality.The loudness maximal value of program, minimum value, mean value, and the dynamic range of loudness are weighed the objective indicator of broadcast quality exactly.The producer of audio/video program, if the foundation with these loudness measurements result during as program making, adjust the size of the sound signal of program, the loudness that just can envoy's purpose audio frequency reaches hope, for all kinds of multimedia terminal users, also data are understood the loudness situation of the program of listening to accordingly, obtain better appreciation effect.
The research of equal loudness contour indicates that also the difference of loudness also has considerable influence to the quality of sound.
In addition, different playing environments and playback terminal also have different demands to the dynamic range of program loudness.For example: when playing when film audio is play at the cinema and on TV, just need dynamically adjust the loudness of sound, to adapt to the demand of different audiomonitors and environment.
So, take more accurately, method is monitored the loudness power of sound signal easily, in order to instruct program making personnel and terminal user to adjust, can instruct making, the result of broadcast of all kinds of audio/video programs better, improve user's satisfaction.
Offshore company has developed measuring equipment at the sound signal loudness measurement, the more representational two big classes that are divided into: based on the loudness table of pure hardware and loudness table based on pure software.
Based on the loudness table of pure hardware, the Related product of the RTW of representative is Germany, the companies such as Dolby of the U.S..The sound signal loudness table of these companies all is with the pure example, in hardware work of special measurement instrument.The advantage of this series products is that the real-time of measuring is very high, and the result is accurate, and shortcoming is complex structure, and is expensive, can only show in real time in playing process, can't recall and finish program loudness situation, is observing and is using very inconvenient.Current, audio/video program great majority is made on computers by the producer, is independent of the loudness table of the pure example, in hardware outside the computing machine, and when being connected with the program making software of various computing machine platform, cost is bigger.
Loudness table based on pure software can be divided into two classes.The first kind is the software of independent operating, and it is independent of audio/video program and makes software work, the Pinguin loudness measurement software of representative is German Pinguin company.It utilizes computer sound card to gather sound signal, and calculates the loudness of sound signal on computers.Second class is to make the card format work of software as audio/video program, and representative is the VST of ToneBooster company loudness plug-in unit.It directly carries out loudness calculating to the audio file that audio frequency and video are made in the software, can not directly carry out real-time loudness to the sound signal that is input to sound card and calculate.
Advantage based on the loudness table of pure software is that price is relatively low, and measurement result digitizing, documentation, is conducive to cooperate on computers the producer to the making of audio/video program.But this series products also has very big weakness, and the measurement computation process of loudness is finished by the CPU of computing machine fully, when the loudness table exists as the plug-in unit of program making software, takies a large amount of system resources.When loudness software independent operating, also can monopolize the sound card resource, make other softwares not operate sound card simultaneously, can't carry out normal program simultaneously and make.In addition, the loudness software of card format operation is subjected to the restriction that audio/video program is made software version, and versatility is relatively poor, and can not directly carry out real-time loudness to the sound signal that is input to sound card and calculate.
Summary of the invention
The purpose of this invention is to provide a kind of loudness size and the dynamic method that can gather and calculate sound signal.This method realizes by the acting in conjunction of hardware and software two parts.Hardware components especially proposes digital audio and video signals loudness monitoring card a kind of loudness of calculating input digital audio signal, that use at computer PCI express interface.This integrated circuit board is gathered digital audio and video signals, and the loudness of sound signal correspondence is calculated; Computing machine is this integrated circuit board of routine access by software section, just can record, shows, stores and analyze the loudness of the digital audio and video signals of importing.The present invention can offer audio frequency and the audio/video program producer uses, and also can be used for independently, the sound signal monitoring equipment uses for various types of programs making, transmission and multimedia terminal equipment.
Technical scheme of the present invention is: a kind of digital audio and video signals loudness monitoring method, and this method realizes by hardware and the acting in conjunction of program two parts; Hardware using comprises the digital audio and video signals loudness monitoring card of DAB collecting unit, decoding unit, loudness computing unit and 4 parts of communication unit; Described program is that computing machine passes through the described loudness monitoring card of this routine access, is not influencing other making, is playing under the situation of operate as normal, and loudness size and the dynamic range thereof of the sound signal of (or claiming to gather) imported in record, demonstration, storage and analysis; This DAB loudness monitoring card is gathered digital audio and video signals, and the program meter of sound signal correspondence is calculated.
The DAB collecting unit has 3 groups of input interfaces of as many as, every group interface all has 3 kinds of input interface circuits of as many as (Fig. 7) of the DAB that is adapted to AES/EBU or S/PDIF signal, is respectively applied to AES/EBU signal and the S/PDIF signal (referring to table 1 and table 2) of receiving balance and non-equilibrium input; Every group of DAB input interface circuit uses a multi-channel switch to select a signal as input of above-mentioned 3 digital audio input interface circuit; The digital audio decoding code element is converted into data stream with the digital audio and video signals of AES/EBU or S/PDIF form; Above-mentioned every group of DAB input interface circuit uses a multi-channel switch to select input signal; Above-mentioned 3 group interfaces can constitute the array mode of the multiple sound signal from monophony to 5.1 surround sounds;
The digital audio decoding code element is converted into data stream with the digital audio and video signals of AES/EBU or S/PDIF form;
The flow process of digital audio decoding unit decodes: by detecting the saltus step of digital audio and video signals, recover bit clock, calculate the voice-grade channel sample frequency according to corresponding channel bit rate; The voice-grade channel bit rate is 64 times of sample frequency;
The digital audio decoding code element is according to the channel status data that read, and judgement will be read voice data on each subframe by several sampling precision;
AES/EBU has identical frame form with the S/PDIF digital audio and video signals, and the subframe preamble has identified the beginning of a subframe; The digital audio decoding code element can detect the digital audio and video signals of input after having locked the channel bit rate, find preamble, then the voice data in each subframe is read, and draws sample frequency and be 1/64 data stream of channel bit rate;
The loudness of loudness computing unit is calculated: the loudness computing unit comprises that filtering, integral mean and three subelements of weighted calculation of working in order draw the loudness data result; The loudness computing unit carries out filtering with the data stream that the digital audio decoding code element provides by K weighted wave filter, draw filtering data after, in the integral time of determining, the filtering data of all passages is carried out integral mean separately; Weighted calculation: the difference of the data based passage of integral mean that draws is given different weights, and weight numerical value is imported by computing machine; Remake the average of each integral mean result then, draw the loudness data result at last;
Communication unit sends to the loudness data result of loudness computing unit in the computing machine by PCI express interface, is recorded, stores, shows and analyzed by computing machine.
Further, the digital audio and video signals collecting unit collects the digital audio and video signals of 3 AES/EBU ports or S/PDIF port, these signals are imported into AES/EBU in decoding unit and the S/PDIF Decoder bank is decoded, obtain 6 passages, each passage is the 48kHz sampling rate, the channel bit rate is the data stream of 3.072Mbit/s16 position sampling precision, and the data stream with 6 passages is input to the loudness computing unit then.
Further, after the loudness computing unit receives the data stream of 6 passages of as many as, 6 data stream of as many as are input in the K weighted bank of filters, respectively 6 data stream of this as many as are carried out filtering by frequency content, drawn 6 data stream of new as many as.
Further, next the loudness computing unit began according to certain integral time, and 6 data flow points are not input to integral mean device group, and integrated square is average by carrying out this integral time, had obtained 6 average datas.
Further, next the loudness computing unit with these 6 data input weight calculator, at first multiply by corresponding weight, last again with the results added after the weighting and ask denary logarithm, and this logarithm be multiply by 20, draws the loudness result calculated.
Further, according to the sound channel constituted mode of program, import 1～6 channel digital audio signal, calculate monophony, stereo, until the loudness of 5.1 surround sound programs; According to different channel combinations, can calculate the loudness of 6 monophony programs, 3 stereophonic programs or 1 5.1 surround sound program simultaneously.
Further, computing machine just can record, shows, stores and analyze the loudness of the digital audio and video signals of importing by this integrated circuit board of routine access.
Further, the precision of sound signal and sampling rate determine that at output terminal the loudness monitoring card belongs to the input end of signal, and monitoring card can only come solution to read this signals sampling precision and sampling rate according to the coding of input signal, audio frequency is the 48kHz sample frequency, sampling precision 16 bits; Or sample frequency is 32kHz, 44.1kHz, and corresponding sampling precision is 8 bits or 24 bits.
The audio/video program of total digitalization is made basically, basically be to finish all making on computers, can be embedded into the loudness monitoring card that moves in the computing machine, not only easy to use than pure hardware loudness table, can record and analyze historical data, and have cost advantage.With first kind pure software loudness epiphase ratio, the loudness monitoring card is occupying system resources not, and is not made the restriction of software type and version.Compare with embedded loudness plug-in unit, then can not made the limitation of software type and version, do not take cpu resource.Accordingly, for all kinds of terminal users, can obtain the Various types of data of program loudness easily, be used for instructing the adjustment to all kinds of listening equipments, thereby improve final audibility.
Beneficial effect of the present invention: method and the instant-plugging computer hardware of the loudness size by a kind of digital audio and video signals that can calculate input is provided, and can in time show and analyze the loudness numerical value of current audio signals; The present invention is a kind of loudness measurement method with hardware technology and software engineering combination.At first, the present invention gathers digital audio and video signals with a kind of loudness monitoring card of example, in hardware, and finishes the overall process that audio loudness is calculated in card, and measurement result is accurate, real-time is high, and does not take computer resource.The loudness result of calculation of sound signal, can record, store and show by installed software on the computing machine, accomplished documentation and the digitizing of measurement result, the loudness monitoring card among the present invention can be installed in the audio frequency and video workstation easily, but also independent operating.This sound signal loudness measurement method can be embedded among the digitized audio/video program making link easily, and the present invention also can independently carry out the loudness monitoring, is used for the loudness status analysis to the finished product program.
Description of drawings
Figure 1A ES/EBU frame structure;
Fig. 2 K weighted wave filter amplitude-frequency response;
Fig. 3 AES/EBU24 byte lane status block constitutes;
Fig. 4 S/PDIF24 byte lane status block constitutes;
The system chart of Fig. 5 digital audio and video signals loudness monitoring card;
Fig. 6 loudness comfort zone and tolerance district;
Fig. 7 DAB input interface circuit;
Fig. 8 is digital audio decoding flow process of the present invention.
Specific embodiments
The invention will be further described below in conjunction with accompanying drawing and subordinate list.
Digital audio and video signals loudness monitoring card comprises DAB collecting unit, decoding unit, loudness computing unit and 4 parts of communication unit.
The DAB collecting unit has 3 groups of input interfaces, every group interface all has 3 kinds of input interface circuits (Fig. 7) of the DAB that is adapted to AES/EBU or S/PDIF signal, is respectively applied to AES/EBU signal and the S/PDIF signal (referring to table 1 and table 2) of receiving balance and non-equilibrium input; Every group of DAB input interface circuit uses a multi-channel switch to select a signal as input of above-mentioned 3 digital audio input interface circuit.
The digital audio decoding code element is converted into the digital audio and video signals of AES/EBU or S/PDIF form the data stream that can use for the loudness computing unit.
Table 1AES/EBU home block
Table 2S/PDIF home block
The flow process of digital audio decoding unit decodes: by detecting the saltus step of digital audio and video signals, recover bit clock, calculate the voice-grade channel sample frequency according to corresponding channel bit rate; The voice-grade channel bit rate is 64 times of sample frequency; Be example as the sound signal with the 48kHz sample frequency, its corresponding channel bit rate is 3.072Mbit/s.
According to the digital frame structure of AES/EBU or S/PDIF signal, the 30th of each subframe is the channel status data, and 192 subframes have 192 channel status data, totally 24 bytes, as shown in Figure 3 and Figure 4.
AES/EBU has different channel status data with S/PDIF, is AES/EBU signal or S/PDIF with the 0th description of the 0th byte, as shown in Table 1 and Table 2.If this position is 1, represent that this is the signal of an AES/EBU digital audio encoding, its sampling precision will provide 3～5 of the 2nd byte of channel status data; If this position is 0, represent that this is the signal of a S/PDIF digital audio encoding, its sampling precision will provide 1～3 of the 4th byte of channel status data.
The digital audio decoding code element is according to the channel status data that read, and judgement will be read voice data on each subframe by several sampling precision.
The precision of sound signal and sampling rate are (as the DAB output of digital sound-track engraving apparatus and computing machine) in the output terminal decision of digital audio and video signals, the loudness monitoring card belongs to the input end of digital audio and video signals, it can only come solution to read this signals sampling precision and sampling rate according to the coding of input signal, the TV programme sound intermediate frequency generally is the 48kHz sample frequency, sampling precision 16 bits, but for other program also 32kHz, 44.1kHz etc., also can be that 8 bits, 24 compare top grade.
AES/EBU has identical frame form with the S/PDIF digital audio and video signals, and the subframe preamble has identified the beginning of a subframe; The digital audio decoding code element can detect the digital audio and video signals of input after having locked the channel bit rate, find preamble, then the voice data in each subframe is read, and draws sample frequency and be 1/64 data stream of channel bit rate.
The loudness calculation process: by the loudness computing unit data stream that the digital audio decoding code element provides is carried out filtering by K weighted wave filter, draw filtering data after, in the integral time of determining, the filtering data of all passages is carried out integral mean separately; Weighted calculation: the difference of the data based passage of integral mean that draws is given different weights, and weight numerical value is imported by computing machine; Remake the average of each integral mean result then, draw the loudness data result at last;
At first, digital audio decoding code element (part 2 among Fig. 5) is with 3 digital audio and video signals, be input to AES/EBU and the S/PDIF Decoder bank is decoded, obtain 6 passages, each passage is the 48kHz sampling rate, the data stream of 16 sampling precisions, the data stream with 6 passages is input to loudness computing unit (the 3rd part among Fig. 5) then.
After the loudness computing unit receives the data stream of 6 passages of as many as (1～6 passage), 6 data stream of this as many as are input in the K weighted bank of filters, respectively 6 data stream of this as many as are carried out filtering by frequency content (seeing the K wave filter of Fig. 2), drawn 6 data stream of new as many as.
According to the constituted mode of program sound sound channel, also can only import 1 passage, or the digital audio and video signals of 2,3,4,5,6 passages, calculate the loudness of these passages; Perhaps calculate 6 monophonys simultaneously, or 3 stereo, or the loudness of 1 surround sound program
Next the loudness computing unit begins according to (meeting the loudness calculation requirement certain integral time, can be by computer settings, as 100-400ms), 6 data flow points are not input to integral mean device group, integrated square is average by carrying out this integral time, has obtained 6 average datas.
The loudness computing unit is next with these 6 data input weight calculator, at first multiply by corresponding weight (can by computer settings), again with the results added after the weighting and ask denary logarithm, last, this logarithm be multiply by 20, draw the loudness result calculated.
Communication unit (3 parts among Fig. 5) sends to computing machine with loudness result of calculation by PCI express interface, and computing machine can various forms shows and store this result, and does further to analyze, and the content of analysis mainly contains:
1) maximal value of loudness, minimum value, mean value;
2) loudness dynamic range, program loudness dynamic range refers to the scope that fluctuates of the loudness numerical value centered by the loudness desired value.The loudness dynamic range should fluctuate in the sense of hearing comfort zone centered by the loudness desired value, does not exceed sense of hearing tolerance district.In television program designing, generally with-24dB LKFS as the loudness desired value.The comfort zone of loudness is seen Fig. 6 with tolerance district synoptic diagram; According to user's demand, also can only import 1 passage, or the digital audio and video signals of 2,3,4,5 passages, calculate the loudness of these passages.
3pin interface among Fig. 7 is generally the XLR interface, also can use TSR etc. to have 3 or 3 interfaces with upper terminal.2pin interface among Fig. 7 is generally bnc interface, also can use TS, RCA etc. to have the interface of two or more terminals.
The employed two-phase encoding flag of AES/EBU and S/PDIF is a kind of scale-of-two frequency modulation (PFM) coding.Saltus step when a bit time interval begins for binary one, can saltus step occur at this time period middle part, and for binary zero, saltus step not take place then.The digital audio decoding code element recovers bit clock by detecting saltus step, calculates the channel sample frequency according to corresponding channel bit rate.The channel bit rate always is 64 times of sample frequency, is example with the sound signal of 48kHz sample frequency, and its corresponding channel bit rate is 3.072Mbit/s.
The 30th of each subframe is the channel status data, and 192 subframes have 192 channel status data, totally 24 bytes, as shown in Figure 3 and Figure 4.AES/EBU has different channel status data with S/PDIF, is AES/EBU signal or S/PDIF with the 0th description of the 0th byte, as shown in Table 1 and Table 2.If this position is 1, represent that this is the signal of an AES/EBU digital audio encoding, its sampling precision will provide 3～5 of the 2nd byte of channel status data; If this position is 0, represent that this is the signal of a S/PDIF digital audio encoding, its sampling precision will provide 1～3 of the 4th byte of channel status data.
The digital audio decoding code element is according to the channel status data that read, and judgement will be read voice data on each subframe by several sampling precision.
AES/EBU has identical frame form with the S/PDIF digital audio and video signals, and as shown in Figure 1, the subframe preamble has identified the beginning of a subframe.The digital audio decoding code element can detect the digital audio and video signals of input after having locked the channel bit rate, find preamble, then the voice data in each subframe is read, and draws sample frequency and be 1/64 data stream of channel bit rate.
The loudness calculation process is (standard program) as shown in Figure 8.
The loudness computing unit carries out filtering with the data stream that the digital audio decoding code element provides by K weighted wave filter, the frequency response of wave filter as shown in Figure 2, after drawing filtering data, at computing machine in specified integral time, the filtering data of all passages is carried out integral mean separately, the difference of the data based passage of integral mean that draws is given different weights, and weight numerical value is imported by computing machine.Remake the average of each integral mean result then, draw the loudness data result at last.
Communication unit sends to the loudness data result of loudness computing unit in the computing machine, is stored and is shown by computing machine.
In this example, use the EP4CGX50FPGA chip to realize digital audio decoding code element, loudness computing unit and communication unit, the digital audio and video signals of 6 passages is carried out loudness measurement calculate.
More than disclosed only be a specific embodiment of the present invention; but the present invention is not limited thereto; those skilled in the art is not breaking away under the scope and spirit that the present invention protects; can make various concrete variations according to different actual needs, but still belong to the protection domain of this patent.
Claims (9)
1. a digital audio and video signals loudness monitoring method is characterized in that realizing by hardware and the acting in conjunction of program two parts; Hardware using comprises the digital audio and video signals loudness monitoring card of DAB collecting unit, decoding unit, loudness computing unit and 4 parts of communication unit; Described program is that computing machine passes through the described loudness monitoring card of this routine access, is not influencing other making, is playing under the situation of operate as normal, loudness size and the dynamic range thereof of the sound signal that record, demonstration, storage and analysis are imported; This DAB loudness monitoring card is gathered digital audio and video signals, and the program meter of sound signal correspondence is calculated.
The DAB collecting unit has 3 groups of input interfaces of as many as, every group interface all has 3 kinds of input interface circuits of as many as of the DAB that is adapted to AES/EBU or S/PDIF signal, is respectively applied to AES/EBU signal and the S/PDIF signal of receiving balance and non-equilibrium input; Every group of DAB input interface circuit uses a multi-channel switch to select a signal as input of above-mentioned 3 digital audio input interface circuit; The digital audio decoding code element is converted into data stream with the digital audio and video signals of AES/EBU or S/PDIF form; Above-mentioned every group of DAB input interface circuit uses a multi-channel switch to select input signal;
The digital audio decoding code element is converted into data stream with the digital audio and video signals of AES/EBU or S/PDIF form;
The flow process of digital audio decoding unit decodes: by detecting the saltus step of digital audio and video signals, recover bit clock, calculate the voice-grade channel sample frequency according to corresponding channel bit rate; The voice-grade channel bit rate is 64 times of sample frequency;
The digital audio decoding code element is according to the channel status data that read, and judgement will be read voice data on each subframe by several sampling precision;
AES/EBU has identical frame form with the S/PDIF digital audio and video signals, and the subframe preamble has identified the beginning of a subframe; The digital audio decoding code element can detect the digital audio and video signals of input after having locked the channel bit rate, find preamble, then the voice data in each subframe is read, and draws sample frequency and be 1/64 data stream of channel bit rate;
The loudness of loudness computing unit is calculated: the loudness computing unit comprises that filtering, integral mean and three subelements of weighted calculation of working in order draw the loudness data result; The loudness computing unit carries out filtering with the data stream that the digital audio decoding code element provides by K weighted wave filter, draw filtering data after, in the integral time of determining, the filtering data of all passages is carried out integral mean separately; Weighted calculation: the difference of the data based passage of integral mean that draws is given different weights, and weight numerical value is imported by computing machine; Remake the average of each integral mean result then, draw the loudness data result at last;
Communication unit sends to the loudness data result of loudness computing unit in the computing machine by PCI express interface, is recorded, stores, shows and analyzed by computing machine.
2. digital audio and video signals loudness monitoring method according to claim 1, it is characterized in that digital audio and video signals collecting unit and decoding unit are with the digital audio and video signals of 3 AES/EBU ports or S/PDIF port, be input to AES/EBU and the S/PDIF Decoder bank is decoded, obtain 6 passages, each passage is the 48kHz sampling rate, the channel bit rate is the data stream of 3.072Mbit/s16 position sampling precision, and the data stream with 6 passages is input to the loudness computing unit then.
3. digital audio and video signals loudness monitoring method according to claim 1, after it is characterized in that the loudness computing unit receives the data stream of 6 passages of as many as, 6 data stream of as many as are input in the K weighted bank of filters, respectively 6 data stream of this as many as are carried out filtering by frequency content, drawn 6 data stream of new as many as.
4. digital audio and video signals loudness monitoring method according to claim 1, it is characterized in that next the loudness computing unit began according to certain integral time, 6 data flow points are not input to integral mean device group, and integrated square is average by carrying out this integral time, has obtained 6 average datas.
5. digital audio and video signals loudness monitoring method according to claim 1, it is characterized in that the loudness computing unit is next with these 6 data input weight calculator, at first multiply by corresponding weight, again with the results added after the weighting and ask denary logarithm, at last, this logarithm be multiply by 20, draw the loudness result calculated.
6. according to the described digital audio and video signals loudness of one of claim 1～5 monitoring method, it is characterized in that the demand according to the user, import 1 passage, or the digital audio and video signals of 2,3,4,5 passages, calculate the loudness of these passages.
7. according to the described digital audio and video signals loudness of one of claim 1～5 monitoring method, it is characterized in that computing machine is by this integrated circuit board of routine access, just can record, show, store and analyze the loudness of the digital audio and video signals of importing, difference according to the program audio sound channel is formed, and carries out the loudness of 6 monophony programs or 3 stereophonic programs or 1 5.1 surround sound program simultaneously and calculates.
8. according to the described digital audio and video signals loudness of one of claim 1～5 monitoring method, the precision and the sampling rate that it is characterized in that sound signal determine at output terminal, the loudness monitoring card belongs to the input end of signal, monitoring card can only come solution to read this signals sampling precision and sampling rate according to the coding of input signal, audio frequency is the 48kHz sample frequency, sampling precision 16 bits; Or sample frequency is 32kHz, 44.1kHz, and corresponding sampling precision is 8 bits or 24 bits.
9. digital audio and video signals loudness monitoring method according to claim 5, it is characterized in that communication unit sends to computing machine with loudness unit result of calculation by PCI express interface, computing machine shows with various forms and stores this result, and performs an analysis the content of analysis:
1) maximal value of loudness, minimum value, mean value;
2) loudness dynamic range, program loudness dynamic range refers to the scope that fluctuates of the loudness numerical value centered by the loudness desired value; The loudness dynamic range should fluctuate in the sense of hearing comfort zone centered by the loudness desired value, does not exceed sense of hearing tolerance district.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201310285813.5A CN103345376B (en) | 2013-07-08 | 2013-07-08 | A kind of digital audio signal volume monitoring method |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201310285813.5A CN103345376B (en) | 2013-07-08 | 2013-07-08 | A kind of digital audio signal volume monitoring method |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN103345376A true CN103345376A (en) | 2013-10-09 |
| CN103345376B CN103345376B (en) | 2015-12-23 |
Family
ID=49280175
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201310285813.5A Active CN103345376B (en) | 2013-07-08 | 2013-07-08 | A kind of digital audio signal volume monitoring method |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN103345376B (en) |
Cited By (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN103942048A (en) * | 2014-04-09 | 2014-07-23 | Tcl集团股份有限公司 | Method and device for displaying voice volume in cartoon mode |
| CN104053054A (en) * | 2014-06-23 | 2014-09-17 | 中兴通讯股份有限公司 | Data collection method and device |
| CN104142879A (en) * | 2014-04-08 | 2014-11-12 | 腾讯科技（深圳）有限公司 | Audio loudness prompting method and device as well as user terminal |
| CN105100787A (en) * | 2014-05-20 | 2015-11-25 | 南京视威电子科技股份有限公司 | Loudness display device and display method |
| CN105681876A (en) * | 2015-12-30 | 2016-06-15 | 惠州市伟乐科技股份有限公司 | Audio loudness uniformity control method and system of radio and television broadcasting system |
| CN108182047A (en) * | 2016-12-08 | 2018-06-19 | 武汉斗鱼网络科技有限公司 | The display methods and device of a kind of information volume |
| CN108337606A (en) * | 2015-09-13 | 2018-07-27 | 国光电器股份有限公司 | System, method and storage medium for the audio signal compensation based on loudness |
| CN109697047A (en) * | 2017-10-23 | 2019-04-30 | 北京淳中科技股份有限公司 | Audio signal monitors method, apparatus, system, array display unit and split screen system |
| CN111045633A (en) * | 2018-10-12 | 2020-04-21 | 北京微播视界科技有限公司 | Method and apparatus for detecting loudness of audio signal |
| CN111327944A (en) * | 2020-03-09 | 2020-06-23 | 上海电力大学 | Broadcast voice loudness detection method based on embedded CPU |
Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1973434A (en) * | 2004-07-01 | 2007-05-30 | 杜比实验室特许公司 | Method for correcting metadata affecting the playback loudness and dynamic range of audio information |
| CN101079296A (en) * | 2006-05-22 | 2007-11-28 | 上海奇码数字信息有限公司 | Audio frequency decoder and audio frequency decoding method |
| US20100109926A1 (en) * | 2008-10-31 | 2010-05-06 | At&T Intellectual Property I, L.P. | System and Method to Modify a Metadata Parameter |
-
2013
- 2013-07-08 CN CN201310285813.5A patent/CN103345376B/en active Active
Patent Citations (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1973434A (en) * | 2004-07-01 | 2007-05-30 | 杜比实验室特许公司 | Method for correcting metadata affecting the playback loudness and dynamic range of audio information |
| CN101079296A (en) * | 2006-05-22 | 2007-11-28 | 上海奇码数字信息有限公司 | Audio frequency decoder and audio frequency decoding method |
| US20100109926A1 (en) * | 2008-10-31 | 2010-05-06 | At&T Intellectual Property I, L.P. | System and Method to Modify a Metadata Parameter |
Non-Patent Citations (1)
| Title |
|---|
| 薛兵: "电视新闻数字音频直播系统", 《电声技术》, no. 6, 30 June 2004 (2004-06-30) * |
Cited By (16)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN104142879A (en) * | 2014-04-08 | 2014-11-12 | 腾讯科技（深圳）有限公司 | Audio loudness prompting method and device as well as user terminal |
| US9870198B2 (en) | 2014-04-08 | 2018-01-16 | Tencent Technology (Shenzhen) Company Limited | Method and system of illustrating audio signal strength |
| CN103942048B (en) * | 2014-04-09 | 2018-10-09 | Tcl集团股份有限公司 | A kind of method and device that speech volume animation is shown |
| CN103942048A (en) * | 2014-04-09 | 2014-07-23 | Tcl集团股份有限公司 | Method and device for displaying voice volume in cartoon mode |
| CN105100787A (en) * | 2014-05-20 | 2015-11-25 | 南京视威电子科技股份有限公司 | Loudness display device and display method |
| CN105100787B (en) * | 2014-05-20 | 2017-06-30 | 南京视威电子科技股份有限公司 | Loudness display device and display methods |
| CN104053054A (en) * | 2014-06-23 | 2014-09-17 | 中兴通讯股份有限公司 | Data collection method and device |
| CN108337606B (en) * | 2015-09-13 | 2020-11-03 | 国光电器股份有限公司 | Systems, methods, and storage media for loudness-based audio signal compensation |
| CN108337606A (en) * | 2015-09-13 | 2018-07-27 | 国光电器股份有限公司 | System, method and storage medium for the audio signal compensation based on loudness |
| CN105681876A (en) * | 2015-12-30 | 2016-06-15 | 惠州市伟乐科技股份有限公司 | Audio loudness uniformity control method and system of radio and television broadcasting system |
| CN108182047B (en) * | 2016-12-08 | 2020-10-16 | 武汉斗鱼网络科技有限公司 | Volume information display method and device |
| CN108182047A (en) * | 2016-12-08 | 2018-06-19 | 武汉斗鱼网络科技有限公司 | The display methods and device of a kind of information volume |
| CN109697047A (en) * | 2017-10-23 | 2019-04-30 | 北京淳中科技股份有限公司 | Audio signal monitors method, apparatus, system, array display unit and split screen system |
| CN109697047B (en) * | 2017-10-23 | 2022-08-26 | 北京淳中科技股份有限公司 | Audio signal monitoring method, device and system, array imager and split screen system |
| CN111045633A (en) * | 2018-10-12 | 2020-04-21 | 北京微播视界科技有限公司 | Method and apparatus for detecting loudness of audio signal |
| CN111327944A (en) * | 2020-03-09 | 2020-06-23 | 上海电力大学 | Broadcast voice loudness detection method based on embedded CPU |
Also Published As
| Publication number | Publication date |
|---|---|
| CN103345376B (en) | 2015-12-23 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| CN103345376B (en) | A kind of digital audio signal volume monitoring method | |
| Francombe et al. | Evaluation of spatial audio reproduction methods (part 2): analysis of listener preference | |
| CN1708186B (en) | Method for processing audio signals from two input sound channels and creating a plurality of output sound channels | |
| CN101228582B (en) | Audio reproduction method and apparatus supporting audio thumbnail function | |
| US8073146B2 (en) | Audio test apparatus and test method thereof | |
| CN101202087B (en) | Device and method for testing audio sound-recording | |
| CN105578379B (en) | Device and method for generating the output signal at least two output channels | |
| CN111316353B (en) | Determining spatial audio parameter coding and associated decoding | |
| RU2008132156A (en) | PERSONALIZED DECODING OF MULTI-CHANNEL VOLUME SOUND | |
| CN104469603B (en) | Electronic equipment, dividing method and frequency divider | |
| CN107251009A (en) | Digital audio frequency filter for variable sample rates | |
| WO2019129350A1 (en) | Determination of spatial audio parameter encoding and associated decoding | |
| CN103188595A (en) | Method and system of processing multichannel audio signals | |
| CN101131816A (en) | Audio file generation method, device and digital player | |
| CN101192182B (en) | Audio- playback test device and method | |
| CN103999154A (en) | Apparatus and method for audio encoding | |
| WO2019105575A1 (en) | Determination of spatial audio parameter encoding and associated decoding | |
| CN104981866A (en) | Method for determining a stereo signal | |
| CN103399724B (en) | A kind of DAB loudness measurement card | |
| CN101600132B (en) | Method and device for adjusting audio file play effect on portable handheld device | |
| Wilson et al. | Perception & evaluation of audio quality in music production | |
| CN103297896A (en) | Audio output method and electronic equipment | |
| Feiten et al. | Audio adaptation according to usage environment and perceptual quality metrics | |
| CN101243488A (en) | Apparatus for encoding and decoding audio signal and method thereof | |
| CN103299657A (en) | Audio signal processing |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant |