EP1706951A1 - Microphone numerique - Google Patents

Microphone numerique

Info

Publication number
EP1706951A1
EP1706951A1 EP04804418A EP04804418A EP1706951A1 EP 1706951 A1 EP1706951 A1 EP 1706951A1 EP 04804418 A EP04804418 A EP 04804418A EP 04804418 A EP04804418 A EP 04804418A EP 1706951 A1 EP1706951 A1 EP 1706951A1
Authority
EP
European Patent Office
Prior art keywords
microphone
transmission
audio
audio information
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.)
Withdrawn
Application number
EP04804418A
Other languages
German (de)
English (en)
Inventor
Gerrit Buhe
Axel Haupt
Frank Plath
Jan Watermann
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Sennheiser Electronic GmbH and Co KG
Original Assignee
Sennheiser Electronic GmbH and Co KG
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 Sennheiser Electronic GmbH and Co KG filed Critical Sennheiser Electronic GmbH and Co KG
Priority to EP09015915A priority Critical patent/EP2192788A3/fr
Publication of EP1706951A1 publication Critical patent/EP1706951A1/fr
Withdrawn legal-status Critical Current

Links

Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/005—Details of transducers, loudspeakers or microphones using digitally weighted transducing elements
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B7/00—Radio transmission systems, i.e. using radiation field
    • H04B7/02—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas
    • H04B7/04—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas
    • H04B7/06—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas at the transmitting station
    • H04B7/0602—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas at the transmitting station using antenna switching
    • H04B7/0608—Antenna selection according to transmission parameters
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B7/00—Radio transmission systems, i.e. using radiation field
    • H04B7/02—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas
    • H04B7/04—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas
    • H04B7/06—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas at the transmitting station
    • H04B7/0613—Diversity systems; Multi-antenna system, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas at the transmitting station using simultaneous transmission
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04H—BROADCAST COMMUNICATION
    • H04H60/00—Arrangements for broadcast applications with a direct linking to broadcast information or broadcast space-time; Broadcast-related systems
    • H04H60/02—Arrangements for generating broadcast information; Arrangements for generating broadcast-related information with a direct linking to broadcast information or to broadcast space-time; Arrangements for simultaneous generation of broadcast information and broadcast-related information
    • H04H60/04—Studio equipment; Interconnection of studios
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L7/00—Arrangements for synchronising receiver with transmitter
    • H04L7/0016—Arrangements for synchronising receiver with transmitter correction of synchronization errors
    • H04L7/002—Arrangements for synchronising receiver with transmitter correction of synchronization errors correction by interpolation
    • H04L7/0029—Arrangements for synchronising receiver with transmitter correction of synchronization errors correction by interpolation interpolation of received data signal
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/08—Mouthpieces; Microphones; Attachments therefor
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2420/00—Details of connection covered by H04R, not provided for in its groups
    • H04R2420/07—Applications of wireless loudspeakers or wireless microphones

Definitions

  • the present invention preferably relates to a microphone, in particular a digital microphone, that is to say a microphone in which the signal coming out of the microphone converter is already digitized in the microphone and the corresponding microphone signal digitally from the microphone, be it via cable or wirelessly, e.g. B. is delivered in a microphone receiver, preamplifier, etc.
  • a microphone in particular a digital microphone, that is to say a microphone in which the signal coming out of the microphone converter is already digitized in the microphone and the corresponding microphone signal digitally from the microphone, be it via cable or wirelessly, e.g. B. is delivered in a microphone receiver, preamplifier, etc.
  • High-quality audio data generate large amounts of data after AD conversion. If you feel - such as B. common in studio technology - an audio signal at 48 kHz with a word length of z. B. 16 bits from, there is a data stream of 768000 bits per channel, which already means more than 1.5 Mbit / s for a stereo channel.
  • redundancy is added in the form of additional bits that are
  • start page can be used for error detection and error correction.
  • the amount of data to be transferred increases again considerably, depending on the desired correction capacity.
  • audio data compression methods e.g. B. used according to the MPEG standard. These reduce the redundancy contained in the audio signal by processing longer blocks of audio data using various methods. As the block size increases, the compression rate that can be achieved increases and the amount of data decreases. Lossy and lossless methods are known, the former achieving a higher compression rate.
  • This block formation creates a delay in the transmission system, since a data block can only be processed when it has completely run in. This applies to both the sending and the receiving side. This effect is undesirable in some applications, in particular in studio applications in which microphones of the present type are to be used in particular.
  • Compressed audio signals are more sensitive to transmission errors. Interfering with a few bits within a compressed data block can render the entire block unusable. Therefore, increased precautions for error correction must be taken, which means that the load of data increases again. In certain error scenarios, the reduction in the amount of data due to the necessary error protection can be completely consumed and even reversed.
  • Today's systems for digital wireless radio transmission of audio data which have to do without compression processes or only with low compression factors, operate in frequency ranges that allow high occupied bandwidths.
  • the ISM bands at 900 MHz and 2400 MHz are popular.
  • the bandwidth used there can be several MHz.
  • these frequency ranges are also used by many other radio systems.
  • a prioritization of certain applications is not possible due to approval regulations in these frequency ranges.
  • the transmission of audio data from microphones to the receiver is at risk of being undesirably disturbed at any time, because other transmitters in the vicinity may be transmitting on the same frequency band.
  • a special property of the transmission method according to the invention or of the microphone according to the invention is that transmission is possible in real time at a high data rate with a low bandwidth.
  • a high-quality transmission of the audio signals of the microphone can be achieved.
  • the focus here is not only on wireless audio microphone transmission, but also on wired audio microphone transmission.
  • FIG. 1 components of a transmitting device
  • FIG. 2 components of a receiving device
  • FIG. 3 shows a block diagram of the audio source
  • FIG. 4 shows a special non-linear characteristic
  • Figure 5 shows the asynchronous clocking of an A / D converter device.
  • an audio source 10 supplies a digital data stream which is pending transmission.
  • This audio source can contain a nonlinear characteristic curve which transcodes the signal for shorter word lengths.
  • the frame generation 20 inserts additional data 22 into the data stream at regular intervals. These are used to synchronize the data stream on the receiver side and / or to transmit status data of the transmitter and / or to transmit a checksum or other data for controlling or monitoring the transmitter or the receiver.
  • the frame generation is able to output a data stream that is sorted according to the relevance of the data. Bits with high importance and bits with low importance can be sorted at certain positions in the frame.
  • redundancy is added to the user data for error correction on the receiver side.
  • the incoming data are grouped into short packets of length N. Code words of length M> N are formed from these.
  • the packets of length N contain data of different relevance levels, which can be supplied to self-sufficient error correction systems. The relevance levels are thus retained in the packages of length M.
  • a subsequent symbol generator 35 forms a complex modulation symbol from a code word. He uses a set of 2 M symbols. The individual modulation symbols differ in terms of amplitude and / or phase. Each symbol represents M bits of the redundant data stream. The Different levels of relevance can be assigned to corresponding bit positions within the symbol. High Rele data. vanz are assigned to safer positions and vice versa.
  • the symbol generator can be designed such that the difference to the previous code word is converted into a modulation symbol.
  • a subsequent channel filter 40 limits the spectrum of the symbol current and thus forms the transmission spectrum.
  • the filtered symbol data stream is converted into an analog signal in an RF transmitter 45 and modulated onto an RF carrier, amplified and radiated via an antenna arrangement 46.
  • individual antennas can also be used, such as antenna arrangements from several segments which automatically control the segment which has an optimal radiation.
  • a single-channel or multi-channel antenna arrangement 86 receives the RF signal.
  • the analog / digital conversion takes place in each case in a converter device which is operated with a clock which is asynchronous, autonomous and fixed to the transmitter. This minimizes the clock jitter.
  • the subsequent digital signal processing also works in parts asynchronously to the transmitter and is finally only synchronized in the audio sink 50.
  • a single or multi-channel channel filter 80 selects the transmission spectrum and reconstructs the transmitted symbol stream.
  • a symbol decoder 75 reconstructed from one or more symbol streams, the complex modulation symbols from the symbol stock M 2.
  • side information from the respective reception levels, the respective distances of the symbol from the ideal position and / or the distortions of the transmission channel can be used.
  • the codewords of length M determined in this way are checked for errors in the channel decoder 70 and possibly corrected.
  • the redundancy added in the transmitter is removed and the packets of length N are put together to form an overall frame.
  • a frame decomposition 60 the additional data 62 added on the transmission side are split off, possibly evaluated and the transmitted audio data stream is output.
  • the bits previously sorted according to relevance are returned to their original position.
  • An audio sink 50 contains a method for the detection and concealment of residual errors which does not contain any secondary information from the data stream. Only a checksum tests the data stream for existing residual errors. The correction values determined by this method are only used if a received frame has been reliably identified as faulty.
  • FIG. 3 An audio source, specifically for the transmission of digital audio data, is shown in FIG. 3. Its task is to provide the digital data stream to be transmitted.
  • the arrangement therefore contains means for receiving an analog input signal and a digitizing device.
  • the arrangement contains either a plug connection 102 for a microphone and / or a directly connected microphone 101 and / or a general analog high-level input 103.
  • the input signal is adapted to the modulation range of an A / D converter 107 via an adjustable amplifier 105.
  • a limiting device 106 is included to avoid overdriving.
  • the A / D converter delivers digital samples of width K. In studio technology, these are usually 24 bits.
  • a downstream non-linear characteristic curve 109 reduces the word width of the sample value to the width L. A deterioration in the signal-to-noise ratio is consciously accepted.
  • a characteristic curve according to the invention in accordance with FIG. 4 avoids the disadvantages described above.
  • a linear relationship between input and output signal is provided for small input signals 120.
  • a non-linear relationship between input and output signal is provided for larger input signals 121. This could e.g. B. be a logarithmic relationship.
  • segment 121 is calculated in such a way that it passes into segment 120 continuously, ie without a kink. In relation to the joint A, segment 120 forms the tangent of segment 121.
  • the properties of the characteristic curve can be varied within wide limits by the choice of the joint A, the slope of the segment 120 and the curvature of the segment 121.
  • the segments 122 and 123 with the joint A ' apply correspondingly symmetrically as shown in FIG. 4.
  • segment 121 or 123 it is possible to subdivide segment 121 or 123 into sub-segments with different curvatures, the curvatures being selected such that a constant transition is ensured.
  • the word width can be reduced by cutting off the least significant bits, a dithering algorithm for white noise shaping preferably being used.
  • the audio source can have a digital input 104.
  • the already digital signal is fed directly to the non-linear characteristic curve 109.
  • a switch 108 is necessary to switch between different inputs, be they analog or digital.
  • the channel encoder and the symbol generator it is essential that you are free to choose the frame length, the number of redundant bits for an error. correction and the number of bits to be transferred per symbol.
  • a major limitation here is the low bandwidth in the RF transmission.
  • the number of bits that have to be transmitted must therefore be kept as low as possible. It is therefore not possible to add a large amount of redundant bits. It is also necessary to transfer several bits per symbol. The number of additional bits that are required to organize or synchronize the data stream on the receiver side also has a significant influence.
  • M bits are transmitted simultaneously in a modulation symbol.
  • a channel coder 30 is selected such that it generates output words of the same length M as the modulation symbol on the basis of data blocks of length N.
  • the frame length is selected as an integer multiple of the word length L of the samples and at the same time as an integer multiple of the length N.
  • bits can be exempted from this protection if they are considered less relevant or less worth protecting.
  • a particularly advantageous embodiment is obtained if the number of bits per modulation symbol is selected as an integral multiple of the output word width of the channel coder.
  • a digital receiver for audio signals all components are usually operated with a clock which is derived from a common mother clock by division.
  • this also includes the clock for the A / D converter device, which digitizes the analog signals arriving from the HF part.
  • This master clock is adjusted by a clock recovery circuit so that it is exactly matched to the transmission signal.
  • This arrangement always has a certain clock jitter, which leads to a suboptimal result in the A / D conversion.
  • the incoming analog signals which represent a filtered symbol data stream, are fed to an A / D converter device 210.
  • the sampling clock 215 operates independently of the entire remaining clocking of the system. It is fixed at a frequency that is a multiple of the necessary baseband sampling rate.
  • the digitized symbols enter an interpolation device 240 and a symbol clock recovery device 220.
  • the symbol clock recovery calculates the true symbol rate on the basis of the digitized symbol stream and also sends it to the interpolation device 240.
  • the symbols are resampled in the interpolation device 240 at the true symbol rate and thus reduced to the true symbol rate.
  • the symbols obtained in this way then pass through the further receiver circuit 250, namely the channel decoder and the frame decomposition, and then enter a short FIFO memory 260.
  • the read-out clock for the FIFO memory is based on the true symbol clock, which is obtained by a clock multiplier and / or clock divider 225 and a jitter suppression 230.
  • the present application also relates to a touch-sensitive antenna diversity for a radio transmission microphone.
  • This antenna diversity can be carried out together with a digital microphone (as described above) or independently of it.
  • a radio microphone with an antenna connected emits a modulated RF signal. If the task then is to increase the transmission security of the transmitted RF signal and thus to achieve high-quality signal transmission, it must be pointed out in this connection that antenna diversity is common in microphone receivers in order to reduce field strength drops caused by multipath propagation to minimize.
  • Wireless microphones usually consist of a microphone, a housing with an internal transmitter unit and a transmitter antenna. These functional units are usually grouped together in such a way that the exact position of the transmitting antenna cannot necessarily be recognized. As a result, it is common for a hand-held microphone to be held not only on its housing, but also on the antenna or on the microphone basket.
  • the invention proposes to equip a hand-held microphone transmitter with two antennas, which are optionally or weightedly connected to the transmitter unit. These antennas are located at different positions of the microphone, as far apart as possible, e.g. B. on the microphone basket on the one hand and at the end of the microphone handle on the other.
  • the antennas must be electrically isolated from the microphone housing and connected galvanically or capacitively to a switchover unit.
  • This switching unit which is attached between the transmitter and the antenna, can consist of an RF switch or a circulator, which is connected to the diversity antennas.
  • the switching criterion in the detector is obtained from the reflected power detected in the line coupler. Complete shadowing of both diversity antennas at the same time is very unlikely due to the large distance between the two antennas. If this happens anyway, it can be assumed that one of the two antennas can be shielded less than the other antenna due to its larger geometric dimensions.
  • the switching criterion can also be taken from a touch sensor, which evaluates a change in resistance of the sensor to the housing or increase in ripple voltage of the sensor and triggers the switching process.
  • a method for relevance-adapted error protection coding of digital audio data can be used in order to improve the overall quality of the audio data transmission.
  • the relevance-adapted error protection coding method is based on the exploitation of system-inherent differences in the uncorrected bit error rate between the individual output bit positions of higher-quality, digital modulation types. By assigning these different quality levels to self-sufficient error correction systems, these differences remain and thus enable a relevant distribution of the user data. Higher quality levels with a lower bit error rate preserve the data with the highest information content, while data with lower information content are distributed to lower quality levels.
  • FIG. 10 structural diagram for a transmission system without relevance-adapted error protection coding
  • Fig. 11 Structure diagram for a transmission system with relevance-adapted error protection coding.
  • 11 has disadvantages in addition to the described advantages, in particular in terms of implementation costs, since several encoders and decoders have to be operated in parallel. On the one hand, this requires increased hardware expenditure compared to simple coding, on the other hand, The data may also be delayed. Many decoders work with a fixed layout (calculated in bits). If the total data rate is now divided into several parallel data streams, the data rate per decoder drops and the entire layout (measured in time) increases. The overall delay is determined by the decoder with the lowest data rate.
  • the delay is (the delay) at a decoder of t De c 0- of 400 bits / 1 MBit / s, 400 microseconds ,
  • the result is Delay with two decoders from
  • a puncturing is carried out in order to reduce the redundancy back to a desired level.
  • this puncturing is not constant, but variable and adapted to the data to be processed. This puncturing is reversed accordingly in the receiver and the decoding can then be carried out in a common decoder. The delay remains as small as possible.
  • an artificial noise signal can be added to the analog signal depending on the residual error rate or other quality criteria in order to mask residual errors. This enhances the usual auditory impression of analogue transmission.
  • Error masking and / or admixing of noise can be used independently on different outputs of the receiver. This allows z. B. the production line and a listening line treated differently with different methods and / or different parameters.
  • the present inventions are particularly preferably applicable not only for microphones, but also in the opposite transmission direction, that is to say for loudspeaker systems, headphone systems and in particular for in-ear monitor systems.
  • the described wireless transmission method is used for headphones or earphones of any kind and stereo transmission is required, a low audio data compression is used for the analog input of the transmission system, which is then decompressed accordingly on the receiving side.
  • a reasonable range of parameters can be specified as follows: BW ⁇ 200 kHz, a pulse shaping rolloff factor a: 0.1 ⁇ a ⁇ 0.5 and a symbol rate in the range between 130 kHz and 185 kHz.
  • a concrete value example would be a bandwidth BW of 200 kHz, an a of 0.3 and an f sym of 150 kHz.
  • the coded bit rate (f ⁇ ded ) follows according to the formula
  • a meaningful parameter range is specified by M> 1 bit and ⁇ 8 bit, a symbol set of 2 M > 1 and ⁇ 256 and a coded bit rate ("gross") of fcoded ⁇ 130 kBit / s and ⁇ 1, 48 MBit / s.
  • gross coded bit rate
  • a reasonable parameter range for the code rate is> 1/2 and ⁇ 1 and for the uncoded bit rate ("net")> 65 kBit / s and ⁇ 1, 48 MBit / s.
  • An excellent parameter example for C is 3/4, i.e. 0.75 and for f uncoded 450 kBit / s.
  • a useful parameter range for the frame rate FR is> 1 and ⁇ 1, 5 and the audio data rate f Aud i o should normally be 43 kbit / s and ⁇ 1, 48 Mbit / s.
  • a concrete numerical example can be given for FR with 1, 125 and f AUd i o with 400 kbit / s.
  • a reasonable parameter range for the PCM resolution ⁇ is> 12 and ⁇ 24 and for the audio sampling rate> 1, 8 kHz and ⁇ 124 kHz.
  • the described digital receiver according to the invention can, in addition to various digital modulations, also conventional analog, such as, for. Legs
  • FM Frequency Modulation
  • pre-emphasis or compression is undone on a digital level.
  • the receiver according to the invention is thus fully backwards compatible with the various systems already established.
  • the receiver according to the invention can have a frequency scan function which graphically and / or in tabular form shows the band occupancy to the user.
  • the compander used on an FM route can be determined and displayed.
  • the result of the scan function can be used to develop suggestions for the selection of frequencies for your own audio transmission system. This enables the system to configure itself.
  • AD conversion By scanning (AD conversion) a higher reception bandwidth in the receiver than is required in a single transmission path, more than one transmitter can be received and demodulated simultaneously by the following channel selection in digital. To do this, it is necessary to increase the IF bandwidth in the analog as well as the dynamics of the AD conversion.
  • the channels are then separated on the digital level by digital filters or other necessary computing operations.
  • Each channel can have its own parameters, even different modulation or different preprocessing.
  • Audio signals from the various channels can be mixed locally in the receiver, in particular in a bodypack receiver.
  • This mixture is controlled locally in the bodypack by suitable operating means or is carried out automatically locally according to predetermined parameters or externally controlled by a sound engineer using a remote control console (similar to a mixing console). In the latter case, only control signals are transmitted to the bodypack.
  • a remote control console similar to a mixing console. In the latter case, only control signals are transmitted to the bodypack.
  • the processing variants specified above are not based on a Bo- dypack receivers are limited, but also apply to stationary receivers accordingly.
  • the properties of the digital transmission can be adapted very flexibly to the requirements of different applications (flexibility through scalability). For example, it may be useful to adequately scale down the audio quality in favor of longer uptime or longer transmission range of the application, e.g. B. for language / reportage. On the other hand, the audio quality, dynamics and / or sampling rate can be increased if only a short distance has to be covered and / or a limited operating time is acceptable.
  • Mono, stereo and double mono audio transmission can also be selected as options for various applications.
  • the digital data stream can be transmitted encrypted, e.g. B. by the previously known encryption mechanisms and procedures, whereby eavesdropping on the radio link can be effectively prevented. Audio transmission, especially reception, is only possible if the sender and receiver have the same key.
  • Encryption has the advantage that the audio signal of the microphone cannot be easily received by a third party, but only by the receiver assigned to the microphone and thus not anyone can do this even when presenting on stage, in a television studio or anywhere else Receive the decoded microphone signal and thus take unauthorized recordings.
  • the digital receiver in contrast to the previous diversity methods, in the digital receiver according to the invention it is not only possible to switch between the individual reception channels, but all incoming data streams can be processed simultaneously. Since the bits of the data stream with a Quality information (soft decision) are provided, for each bit can be decided anew, from which reception branch it is selected.
  • the selection algorithm can have a switching effect as well as perform a weighted merge. This enables the most advantageous weighting of the data streams from several channels and the information content is optimally used.
  • the selection algorithm can be supported by further page information, such as. B. field strength of the antenna signal or current bit error rate.
  • the algorithm can be implemented adaptively or switchable to take into account the RF characteristics of the room.
  • a display For the display of a continuous weighting, a display is chosen that represents this channel balance for the user in a meaningful way.
  • a bar starting from the center is preferably selected, which represents the tendency to one or the other channel on both sides.
  • the receiver according to the invention thus has a diversity weight display.
  • a process optimized for efficient processing is used to conceal remaining errors.
  • This essentially consists of a number N of fixed filters with a filter depth k. These are optimized for typical audio signals and are very easy to implement algorithmically.
  • An adaptive filter is introduced to the currently optimal filter by an adaptation algorithm. This ensures continuous adaptation to the audio signal.
  • the CRC check sum is used to decide whether the adaptive filter is currently being taught in or used for an audio sample replacement. A replacement is only carried out if the current prediction error lies significantly above the mean prediction error and the CRC sum indicates an error.
  • This threshold or the underlying algorithm can also be adapted. In order to avoid error propagation, a feedback of the replaced samples is also necessary.
  • FIG. 14 A digital implementation of microphone transmitters and / or receivers makes it possible in a simple manner to integrate additional functions which make it possible to set up or check the radio link and other, possibly wired, connection paths for the audio signal.
  • An audio test signal of known amplitude e.g. sinus tone, noise, etc.
  • a modulation signal of known symbol sequences eg PRBS sequence, sequential sequence of all symbols etc.
  • a microphone receiver it is also very advantageous for a microphone receiver if it is provided with a switchable (pluggable) filter at the receiver input. It is also particularly advantageous if the filter at the receiver input is integrated in the receiver antenna, e.g. B. can also be screwed or plugged into the antenna of the receiver.
  • the input filter of the receiver with the antenna is a unit and such an electronically shieldable antenna is particularly advantageous because the filter properties of the receiver can also be adjusted by attaching the antenna.
  • the antenna and the receiver contain devices which enable the receiver to adjust to antenna-specific parameters or to indicate that the antenna does not match the desired reception frequency range.
  • a microphone system according to the invention if means for changing the frequency response or sound character are provided, so that the microphone can have a desired characteristic through a corresponding sound design, e.g. B. the characteristics of a microphone, such as the famous U87 by Georg Neumann, Berlin, or MD 421 by Sennheiser, and others.
  • a desired characteristic e.g. B. the characteristics of a microphone, such as the famous U87 by Georg Neumann, Berlin, or MD 421 by Sennheiser, and others.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Detection And Prevention Of Errors In Transmission (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

L'invention concerne un procédé permettant de transmettre des informations audio numérisées de haute qualité et à temporisation réduite, notamment un procédé pour transmettre des informations audio numérisées dans un enregistrement audio (microphone) et/ou une voie de restitution. Selon l'invention, il est prévu un filtre de canal, utilisé pour former le spectre de haute fréquence de la transmission, ledit spectre ne présentant pas d'atténuation dans une première plage d'approximativement 100 à 300 kHz de bande passante utile et présentant une atténuation de bande affaiblie, notamment de préférence de plus de 60 dB ou de plus de 80 dB (figure 13), dans une plage située en dehors de la première plage.
EP04804418A 2003-12-30 2004-12-30 Microphone numerique Withdrawn EP1706951A1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
EP09015915A EP2192788A3 (fr) 2003-12-30 2004-12-30 Microphone numérique

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE10361817 2003-12-30
PCT/EP2004/014833 WO2005064828A1 (fr) 2003-12-30 2004-12-30 Microphone numerique

Publications (1)

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EP1706951A1 true EP1706951A1 (fr) 2006-10-04

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EP04804418A Withdrawn EP1706951A1 (fr) 2003-12-30 2004-12-30 Microphone numerique
EP09015915A Withdrawn EP2192788A3 (fr) 2003-12-30 2004-12-30 Microphone numérique

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US (1) US8332058B2 (fr)
EP (2) EP1706951A1 (fr)
JP (1) JP2007517441A (fr)
CN (2) CN102395063B (fr)
WO (1) WO2005064828A1 (fr)

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JP2007517441A (ja) 2007-06-28
EP2192788A3 (fr) 2010-08-04
CN1902845A (zh) 2007-01-24
US8332058B2 (en) 2012-12-11
CN102395063A (zh) 2012-03-28
US20070168819A1 (en) 2007-07-19
CN102395063B (zh) 2016-05-18
WO2005064828A1 (fr) 2005-07-14

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