A method for automatically controlling loudness of an audio signal to be replayed within a vehicle is provided. The method comprises an automatic loudness measurer measuring a loudness of an incoming audio signal that comprises a sequence of digital audio blocks. Furthermore, a noise level indication signal is received which indicates a noise level within the vehicle and a processor calculates a gain factor based on the measured loudness of the incoming audio signal and the noise level indication signal. If the measured loudness is equal to or above a predefined first target loudness (TLamS), the gain factor is zero. Otherwise, if the noise level indication signal indicates a noise level equal to or above a predefined intermediate noise level, the gain factor is a difference between the measured loudness and a current target loudness, limited to a maximum gain, wherein the maximum gain is a defined absolute maximum gain, and the current target loudness is linearly interpolated, according to the noise level indication signal, between a predefined second target loudness (TLaiS) for the noise level being equal to said intermediate noise level and the predefined first target loudness (TLamS) for the noise level being equal to a predefined maximum noise level. Otherwise, if the noise level indication signal indicates a noise level below the intermediate noise level, the gain is linearly interpolated, according to the noise level, between the gain as defined for the intermediate noise level and zero. Furthermore, the gain factor is applied to the audio signal to be replayed.
Embodiments of the invention provide for methods for automatically controlling loudness of an audio signal. Embodiments of the invention also provide for a device for automatically controlling loudness of an audio signal. In particular, the invention relates to controlling loudness of an audio signal to be replayed within a vehicle, such as a car.
When an OFDM radio system which uses a wide frequency band is interfered with by another narrow-band radio system, the interference can frequently be compensated but the transmission quality decreases drastically. Thus, narrow-band interferers in an OFDM radio system are determined according to the invention whereby none of the subscribers of the radio system transmits in a defined time slot or scan slot but all switch at the same time into the receiving mode. If there is interference (P1, P2), it is detected in this time slot. Countermeasures are taken individually in all the mobile devices, in particular the detection of the frequency and strength of the narrowband interference (P1, P2) and the configuration of a flexible notch filter (140) in the time range to the detected frequency and strength. The scanned received signal (RXS) is then filtered in the time range, i.e. before the FFT (120) and the OFDM channel estimation (130) by the correspondingly configured notch filter (140). The notch (S1, S2) of the notch filter thereby acts in the transmission function like a natural break when receiving data.
A device for processing multi-channel audio signals that include at least a left channel, a right channel, and a center channel, is provided. The device comprises a center extraction unit adapted for extracting a center signal from the left channel and the right channel, wherein a left remainder signal and a right remainder signal remain; a first summation unit for adding the extracted center signal to the center channel of the multi-channel audio signal to obtain an enhanced center channel; a second summation unit adapted for adding the enhanced center channel to the right remainder signal to obtain an enhanced right channel; a third summation unit adapted for adding the enhanced center channel to the left remainder signal to obtain an enhanced left channel; and outputs for providing the enhanced left channel, the enhanced right channel, and enhanced center channel.
MTMT) as a second measurement value (D2). The base station also transmits the first measurement value (D1) to the mobile device (st6). From the first and second measurement values, the mobile device determines the signal transit time and the deviation of its time base and corrects them (st7, st8).
21,37) are arranged on a carrier such that they lie on three similar branches, each with the same number of microphone capsules, which are rotated against each other by 120°. Each of the microphone capsules lies on a corner of a triangle of a grid in a flat isometric coordinate system with three axes rotated by 120° against each other and forming the grid of equilateral triangles.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
H04R 1/04 - Structural association of microphone with electric circuitry therefor
G10L 25/18 - Speech or voice analysis techniques not restricted to a single one of groups characterised by the type of extracted parameters the extracted parameters being spectral information of each sub-band
H04R 5/027 - Spatial or constructional arrangements of microphones, e.g. in dummy heads
H04R 9/02 - Transducers of moving-coil, moving-strip, or moving-wire type Details
15,11 15,35,21,1121,3721,37) are arranged on a support (T, T') such that the microphone capsules lie on three identical branches, each of which is rotated relative to the other branches by 120°, in such a manner that the same number of microphone capsules lies on each branch. In a flat isometric coordinate system with three axes (L0, L1, L2), each of which is rotated relative to the other axes by 120° and which form an L2 grid of equilateral triangles, each of the microphone capsules lies on a corner of a triangle of the L2 grid.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
Embodiments of the invention provide for a device and method for processing multi-channel audio signals, the multi-channel audio signals comprising at least a left channel, a right channel, and a center channel. Embodiments of the invention also provide for a non-transitory computer-readable storage medium having stored thereon instructions that when executed on a computer cause the computer to perform the method.
A microphone array device including microphone capsules and at least one processing unit configured to receive output signals of the microphone capsules, dynamically steer an audio beam based on the received output signal of the microphone capsules, and generate and provide an audio output signal based on the received output signal of the microphone capsules. The processing unit is configured to operate in a dynamic beam mode where at least one focused audio beam is formed that points towards a detected audio source and in a default beam mode where a broader audio beam is formed that covers substantially a default detection area. The microphone array may be incorporated into a conference system.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
To secure earphones in the auditory canal of a user, ear tips are used. In an improved earphone (200), which reduces or avoids pressure sores occurring in the ear of the user, a hard stub of the housing no longer protrudes as hitherto into the ear tip, and instead a soft receiving tube (220) of the ear tip is secured in the housing (250). By means of this inverse receiving principle, the ear tip can better conform to the auditory canal and thus enhances the wearing comfort. For assembly, the tube (220) is plugged into an opening (245) in the housing so that a projection (225) of the tube engages in a depression or inner shoulder (245a) behind the opening and thus secures the ear cushion on the housing.
Microphone capsules for condenser or electret microphones often exhibit individual deviations from a desired ideal behavior, e.g. the frequency response and phase response. Particularly if a plurality of microphone capsules are interconnected to form a microphone array, suitable microphone capsules must be found in a selection process. Some of these deviations can be corrected electronically, e.g. by filtering with a corresponding filter that is individually adapted. An improved microphone capsule, with which an automatic selection and automatic assembly of circuit boards with microphone capsules is facilitated, comprises an electrostatic sound transducer, an amplifier element that outputs an amplified output signal of the electrostatic sound transducer, and at least one electronic memory element. Data obtained by a measurement and relating to the individual frequency response or phase response of the respective microphone capsule can be stored therein. The data can be read out during manufacturing and during operation, which enables automatic sorting of the capsules during production and automatic calibration of the target circuit in operation.
Power supply units e.g. in network operated loudspeakers are tailored to peak values that are reached only relatively rarely and then in pulses. With an intermediate storage of electrical energy in an intermediate circuit energy storage element it is possible to provide a significantly higher amount of power at least for a short period of time. The intermediate circuit energy storage element may be a capacitor or accumulator, for example, which is connected to an intermediate circuit voltage that is higher than the input voltage, and that is generated by an upconverter. A downconverter generates the output voltage of the power supply unit from the energy stored in the intermediate circuit storage element. The output voltage of the power supply unit is used as power supply for an audio amplifier. The power supply unit may provide for a short period of time a higher current or more energy respectively than the actual energy source, for example the network node. Correspondingly, the device operated with the output voltage of the power supply unit, for example the audio amplifier, can have a significantly higher effective power than previously possible for the short period of time.
H03F 1/02 - Modifications of amplifiers to raise the efficiency, e.g. gliding Class A stages, use of an auxiliary oscillation
H02M 3/155 - Conversion of DC power input into DC power output without intermediate conversion into AC by static converters using discharge tubes with control electrode or semiconductor devices with control electrode using devices of a triode or transistor type requiring continuous application of a control signal using semiconductor devices only
H02M 3/156 - Conversion of DC power input into DC power output without intermediate conversion into AC by static converters using discharge tubes with control electrode or semiconductor devices with control electrode using devices of a triode or transistor type requiring continuous application of a control signal using semiconductor devices only with automatic control of output voltage or current, e.g. switching regulators
A method and a system with which a primary user, in particular a mobile radio operator, can make a part of the radio spectrum range which is reserved exclusively for the mobile radio operator available in localized and/or time-restricted fashion to a local end user who requested the restricted usage authorization of the radio spectrum range using registered credentials at the mobile radio operator. By way of the credentials the mobile radio operator can allow the local end user limited usage of that radio spectrum range in fee-bearing fashion and can bill same. The local user can then use the assigned frequency band in accordance with his own needs. In particular the user can wirelessly transmit signals in the assigned frequency range between end devices to which no credentials are allocated and in that situation use a transmission protocol which is not defined for the mobile radio network.
For certain application cases, such as e.g., in a sports stadium, a microphone array having a particularly high directivity in the vertical direction and a high, yet in wide limits adjustable directivity in horizontal direction is provided. The microphone array has a plurality of microphones whose output signals are combined into at least one common output signal. The microphones are directional microphones with a preferred direction of high sensitivity and arranged substantially in one plane on a circle or segment of a circle, such that each microphone has a different direction of high directivity. For each of the microphones, the preferred direction of high sensitivity is substantially orthogonal to the circle or segment of the circle. A common output signal of the microphone array is obtained by beamforming. The microphone array has an adjustable preferred direction of high sensitivity, wherein the common output signal comprises the sound recorded from this adjustable direction.
A charging device (100) for charging portable rechargeable electronic devices (200) is provided. The charging device comprising a rechargeable battery (120) and a charging control circuitry for controlling a charging of at least one portable rechargeable device. The charging control circuitry comprises interface circuitry (110) adapted for electrically connecting the portable rechargeable device (200) to the rechargeable battery (120), so as to charge the portable rechargeable device from power of the rechargeable battery; gauging circuitry (150) adapted for automatically detecting a battery charging level of the connected portable rechargeable device, comparing the detected battery charging level to a threshold value and detecting if the threshold value has been reached; and a processor configured for serving as a control unit (140) connected at least to the interface circuitry (110) and the gauging circuitry (150). The control unit comprises a timer (145). The charging control circuitry is adapted for repetitively performing a charging phase by transferring energy from the rechargeable battery (120) to the portable rechargeable device (200) via the interface circuitry (110) until the gauging circuitry (150) indicates that the threshold value has been reached, entering a pause phase wherein the charging of the portable rechargeable device from the rechargeable battery is suspended, and starting the timer; and when the timer (145) has elapsed, returning to a charging phase.
2) that leads to a completely panning-like virtualization, the resulting phase is reduced, or adjusted to the panning phase of 0°. By selecting one or more processing parameters, different audio signals may be binaurally virtualized to different degrees before being superposed to each other.
TR) and a velocity of the moving object from a tracking system, receiving a plurality of microphone signals that comprise sound of a sound event emanating from the moving object from a plurality of microphone capsules, calculating a directional characteristic from the plurality of microphone signals, wherein the directional characteristic is based on beamforming according to the position information and wherein an audio output signal is generated that includes the sound from a preferred direction of high sensitivity, and providing the audio output signal at an output. A beam width or opening angle (α) of the directional characteristic varies over time and depends on the velocity of the moving object, wherein a higher velocity of the moving object results in a larger beam width or larger opening angle respectively.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
H04R 1/20 - Arrangements for obtaining desired frequency or directional characteristics
In radio networks having a plurality of access points, for example WiFi, a wireless audio end device like for example a microphone or headphones is connected to an access point, to which it sends its audio data or from which it receives same. Audio transmission should occur as far as possible in interruption-free manner and with low latency. If the audio end device is moved the connection quality in relation to the previous access point can decrease and require scanning or roaming. In that respect first another base station is sought and then the radio connection is redirected there. In order in that case to avoid disruptive signal interruptions portions of the audio signal which can be particularly well predicted are detected or predicted by means of short-term statistical methods, like for example speech pauses. Scanning and roaming are then carried out during the predicted portions, whereby interruptions which are perceptible to a user are avoided.
H04W 36/36 - Reselection control by user or terminal equipment
G10L 25/60 - Speech or voice analysis techniques not restricted to a single one of groups specially adapted for particular use for comparison or discrimination for measuring the quality of voice signals
A microphone array device including microphone capsules and at least one processing unit configured to receive output signals of the microphone capsules, dynamically steer an audio beam based on the received output signal of the microphone capsules, and generate and provide an audio output signal based on the received output signal of the microphone capsules. The processing unit is configured to operate in a dynamic beam mode where at least one focused audio beam is formed that points towards a detected audio source and in a default beam mode where a broader audio beam is formed that covers substantially a default detection area. The microphone array may be incorporated into a conference system.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
A method of wireless audio transmission between a wireless transmitter and a wireless receiver in a program making special event PMSE system. An audio signal is transmitted from the at least one wireless transmitter to the wireless receiver by way of a first wireless transmission path. The at least one wireless transmitter is coupled to a smart device by way of a second wireless transmission path in order to exchange parameters and/or data. Exchange of parameters and/or data occurs between the smart device and the wireless receiving unit by way of a network, in particular the Internet.
H04W 4/80 - Services using short range communication, e.g. near-field communication [NFC], radio-frequency identification [RFID] or low energy communication
For mobile devices used in a wireless audio transmission system, for example for stage technology, like wireless microphones, body-pack transmitters with an audio input, and body-pack receivers with an audio output, electronic displays are known, on which a respective radio frequency or radio channel in use or other alphanumeric information is displayed. This information can be input at the transmitting end, for example, at a central location like a mixing desk. The displays, however, only function while the mobile devices are switched on so that it is not possible to see the association(s) of the mobile device(s) in their switched-off state. For improved identification of wireless microphones, body-pack transmitters, or body-pack receivers in which an automatic alphanumeric identification of the transmission channel in use is displayed in the switched-on state, according to the invention the display of the identification is retained even after the microphone, body-pack transmitter or body-pack receiver is switched off. In this way, mobile devices can be associated with a radio transmission path, an artist or a stage position even in the switched-off state.
A microphone array device including microphone capsules and at least one processing unit configured to receive output signals of the microphone capsules, dynamically steer an audio beam based on the received output signal of the microphone capsules, and generate and provide an audio output signal based on the received output signal of the microphone capsules. The processing unit is configured to operate in a dynamic beam mode where at least one focused audio beam is formed that points towards a detected audio source and in a default beam mode where a broader audio beam is formed that covers substantially a default detection area. The microphone array may be incorporated into a conference system.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
An audio system comprising a control circuit, at least one loudspeaker and at least one subwoofer for reproducing audio signals at low frequencies. The control circuit comprises an input connector and a bass management circuit that comprises a first filter with a low-pass for pre-filtering audio signals for the subwoofer and a second filter for pre-filtering audio signals for the loudspeaker. The loudspeaker comprises a high-pass filter, wherein in a transition frequency range of the amplitude responses of the first filter and the high-pass filter both filters raise or attenuate the audio signals such that the audio system overall, without considering the second filter, has a wavy amplitude response in the transition frequency range. The second filter however comprises at least one parametric filter that is adapted for flattening the amplitude response in the transition frequency range.
An earphone and connection cable can be connected to the housing of the earphone with a robust and reliable plug connection. The plug connection includes a coaxial plug having rotationally symmetrical plug contacts and is surrounded with a plastic encapsulation. When the plug connection is inserted the plastic encapsulation forms a first bearing location for receiving mechanical forces acting laterally on the plug while at least one of the plug contacts of the coaxial plug forms a second bearing location for receiving such forces. The distance from the first to the second bearing location in that case is greater than the diameter of the plug contacts of the coaxial plug.
There is set forth a method and a system for recording and synchronizing audio and video signals. The audio signal and the video signal are stored together with time stamps from a respective associated system clock. The invention relates to an adaptation of the duration of the recorded audio sequence to the duration of an associated video sequence in order to level out differences in synchronization of the two system clocks. Alignment of the two system clocks is also introduced, which is based on a data transfer which has variable waiting times for the access to a transmission channel. This thus permits clock alignment with means as are available for example on a smartphone.
Microphone capsules for capacitive or electret microphones often have individual deviations from a desired ideal electro-acoustic response, e.g. the frequency response and the phase response. In particular when multiple microphone capsules are connected together to form a microphone array, suitable microphone capsules have to be found in a selection process. Some of these deviations can be corrected electronically, e.g. by filtering with an appropriately individually adjusted filter. The invention relates to an improved microphone capsule (200), by means of which automatic selection and automatic population of circuit boards with microphone capsules is made easier, and which contains an electrostatic acoustic converter (CT), an amplifier element (Q1) that outputs an amplified output signal (AS, DS) from the electrostatic acoustic converter (CT), and at least one electronic storage element (U1). In the latter, data obtained by a measurement, which relates to the individual frequency response or the phase response of the respective microphone capsule, can be stored. The data can be read during manufacture and during operation, which means that both automatic sorting of the capsules during production and automatic calibration of the target circuit during operation are possible.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
Protection helmets may include integrated communication systems to allow the wearer to communicate with other persons. In a very loud environment, a reduction of ambient noise may be helpful, for example, for improving wireless communication. An improved protection helmet includes a helmet shell, a chin guard, a first microphone mounted on the inside of the chin guard and facing the mouth of the wearer, a second microphone mounted on the outside, the upper side or the lower side of the chin guard and not facing the mouth of the wearer, an electronic noise reduction unit generating a difference signal between signals of the first and the second microphone, and at least one interface for outputting the difference signal. Ambient noise in the signal of the first microphone can be reduced with the signal of the second microphone.
An improved method for configuring an audio reproduction device for detecting sound and providing different output audio signals in a plurality of rooms where at least two wireless microphones connect via a local network to an audio streaming server. Each of the wireless microphones detects room information indicating the room in which it is located, and transmits it to the server, together with an input audio signal. The server compiles at least two different output audio signals according to the respective room information from the input audio signals, and assigns each to a room. The output audio signals are provided via the local network in the rooms such that each of the output audio signals may be received in all rooms, and may be replayed only in the room to which it has been assigned. Each wireless microphone may be used in each of the rooms.
H04S 7/00 - Indicating arrangementsControl arrangements, e.g. balance control
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
A method for receiving audio data streams with mobile devices in different rooms in which sound is emitted via loudspeakers. Each mobile device connects over a local wireless network to a server that controls the sound emitted in the rooms via loudspeakers. Each mobile device detects room information, emitted only in the corresponding room, that indicates the room the respective mobile device is in. The room information is transmitted from each mobile device to the server. The server associates each room information with an output audio data stream that corresponds to the sound emitted via loudspeaker in the respective room in which the mobile device has detected the room information. The server assigns to each of the mobile devices an output audio data stream corresponding to the respective room information, and transmits the output audio data stream over the local network to the mobile device, which receives and replays it.
A conference system with transmitting and receiving sides. The transmitting side has a microphone array unit with microphone capsules, and a processing unit. The processing unit is configured to receive output signals of the microphone capsules and to execute audio beamforming based on the received output signals for acquiring sound coming from an audio source in a first direction. The processing unit has a direction-recognition unit that computes from the output signals of said microphone capsules a score for each of multiple search grid spatial positions and uses a search grid spatial position having a higher score to identify said first direction. The receiving side has an audio reproduction system that reproduces an audio signal detected by the microphone array with directional information of the first direction. The detected audio signal and the directional information regarding the first direction are transmitted from the transmitting side to the receiving side.
A61B 5/00 - Measuring for diagnostic purposes Identification of persons
H04L 12/18 - Arrangements for providing special services to substations for broadcast or conference
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
39.
Method for operating at least one mobile radio device, power supply unit for mobile radio device, charger and audio transmission configuration system
A method for operating at least one mobile radio device that includes a transmitter, a slot for receiving a power supply unit, and a control unit for controlling operation of the transmitter depending on the radio parameters and/or configuration data. Radio parameters and/or configuration data are transferred to a memory of the power supply unit. The power supply unit is inserted into the slot. Operation of the transmitter is controlled by the control unit based on the radio parameters and/or configuration data that are stored in the power supply unit.
A control unit for an audio system including a subwoofer and at least one further loudspeaker. Type information is recorded via the at least one connected loudspeaker, and a configurable filter for the at least one connected loudspeaker is configured by means of appurtenant configuration information from a memory. The configurable filter is configured so that a base phase management is provided for the transition frequency range between the subwoofer and the at least one connected loudspeaker so that the phase of the sound signals of the subwoofer is matched to the phase of the sound signals of the at least one connected loudspeaker.
Loudspeaker systems, which for technical reasons are not suitable for emitting strong bass signals, can use so-called virtual bass systems. Therein, low frequencies are replaced by their harmonics. However, virtual bass cannot always adequately replace real bass, such that tonal discrepancies may result. Methods and systems are disclosed to improve the bass reproduction of virtual bass by mixing the generated harmonics with a reduced original bass component of the input audio signal. The mixing ratio of this blend can be variable and can be determined automatically. For example, the mixing ratio can change when a level threshold is exceeded, when a temperature rises above/drops below a threshold, a calorimetric threshold is exceeded, or at fixed times of day.
G10L 25/21 - Speech or voice analysis techniques not restricted to a single one of groups characterised by the type of extracted parameters the extracted parameters being power information
42.
Automatic identification of a wireless microphone, a body-pack transmitter or a body-pack receiver
For mobile devices used in a wireless audio transmission system, for example for stage technology, like wireless microphones, body-pack transmitters with an audio input, and body-pack receivers with an audio output, electronic displays are known, on which a respective radio frequency or radio channel in use or other alphanumeric information is displayed. This information can be input at the transmitting end, for example, at a central location like a mixing desk. The displays, however, only function while the mobile devices are switched on so that it is not possible to see the association(s) of the mobile device(s) in their switched-off state. For improved identification of wireless microphones, body-pack transmitters, or body-pack receivers in which an automatic alphanumeric identification of the transmission channel in use is displayed in the switched-on state, according to the invention the display of the identification is retained even after the microphone, body-pack transmitter or body-pack receiver is switched off. In this way, mobile devices can be associated with a radio transmission path, an artist or a stage position even in the switched-off state.
Audio reproduction systems include a plurality of loudspeakers that are actuated in accordance with a multi-channel audio format. The loudspeakers can be configuratable by way of a network interface The loudspeakers are registered on the network. A method for automatic configuration of an audio reproduction system includes automatically determining that all loudspeakers are connected to the network, determining the number of loudspeakers which are disposed in the same room and which are part of the loudspeaker arrangement, automatically generating a representation of a virtual loudspeaker arrangement on a display according to the determined number, and sequentially, for each loudspeaker belonging to the loudspeaker arrangement, as an actual loudspeaker, generating a signal by the actual loudspeaker, receiving a user input characterizing a virtual loudspeaker, assigning the position of the characterized virtual loudspeaker to the actual loudspeakers and configuring audio signal processing for the actual loudspeaker according to its assigned position.
G06F 3/0484 - Interaction techniques based on graphical user interfaces [GUI] for the control of specific functions or operations, e.g. selecting or manipulating an object, an image or a displayed text element, setting a parameter value or selecting a range
H04S 3/00 - Systems employing more than two channels, e.g. quadraphonic
A method of low-latency audio transmission in a mobile communications network utilizing first data frames or subframes encoded according to a first format and shorter second data frames encoded in another second format for audio data. An audio transmission system includes a terminal, a base station, and an audio receiver. The terminal transmits via an uplink audio data that are encoded in the second format and other data that are encoded in the first format. The audio receiver directly receives the audio data transmitted via the uplink. The encoding/decoding of the audio data of one of the second data frames is influenced by other audio data of the same second data frame but not by audio data of another second data frame. Audio transmission from the terminal to the audio receiver is effected in the allocated time slots and frequencies in a waveform in conformity with the mobile communications network.
The invention relates to a method and a system by means of which a primary user, in particular a mobile radio operator, can provide a part of the radio spectrum range, which is exclusively reserved for the mobile radio operator, in a localized and/or time-restricted manner to a local end user who has requested the limited usage authorization of the radio spectrum range from the mobile radio operator using registered credentials. The mobile radio operator can allow the local end user limited usage of the radio spectrum range in a billed manner using the credentials. The local user can then use the assigned frequency band according to the user's own needs. In particular, the user can transmit wireless signals between end devices in the assigned frequency range, said end devices not being assigned any credentials, in the process using a transmission protocol which is not defined for the mobile radio network.
For specific applications, such as e.g. in a sports stadium, a microphone array having particularly high directivity in the vertical direction and a high, but broadly adjustable, directivity in the horizontal direction is provided. The microphone array (100) has a plurality of microphones (110), the output signals of which are combined to produce at least one joint output signal (360). The microphones are directional microphones having a preferred direction of high sensitivity (115) and arranged substantially in one plane on a circle (120) or circle segment, so that each microphone has a different preferred direction of high sensitivity. In this case, the preferred direction of high sensitivity (115) for each of the microphones lies substantially orthogonally in relation to the circle or circle segment. A joint output signal (360) of the microphone array is obtained by beamforming (310,..., 350). The microphone array (100) has an adjustable preferred direction of high sensitivity, wherein the joint output signal (360) contains the sound picked up from this adjustable direction.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
A wireless microphone and/or in-ear monitoring system having at least one first mobile device for wirelessly transmitting first audio signals. The system also has at least one base station for wirelessly receiving the first audio signals transmitted by the mobile device. The wireless transmission is based on an orthogonal frequency-division multiplexing transmission (OFDM) during a time-division multiple access (TDMA) time slot. Each wireless microphone occupies at least one slot within 2 ms. Each of the TDMA frames has a plurality of slots which respectively have precisely one OFDM symbol. Accordingly, precisely one OFDM symbol is transmitted in each TDMA slot. During a time slot made available in accordance with the TDMA, a transmission is effected on the basis of an OFDM method. The TDMA frame length is so short as a latency of <4 ms is required for professional audio transmission, for example in the case of wireless microphone systems.
H04H 20/00 - Arrangements for broadcast or for distribution combined with broadcast
H04B 1/00 - Details of transmission systems, not covered by a single one of groups Details of transmission systems not characterised by the medium used for transmission
H03D 7/00 - Transference of modulation from one carrier to another, e.g. frequency-changing
H04B 1/38 - Transceivers, i.e. devices in which transmitter and receiver form a structural unit and in which at least one part is used for functions of transmitting and receiving
A method for generating and providing an audio signal, including receiving a first audio signal via an external microphone of a headphone or earphone, and receiving a second audio signal via a wireless interface. The first audio signal includes a portion reproduced via loudspeakers. The second audio signal corresponds to the portion reproduced via loudspeakers and is received before the corresponding portion of the first audio signal. A propagation time difference is determined between the first audio signal and the second audio signal. The second audio signal is modified by adaptive filtering and temporal shifting such that the propagation time difference between the first and second modified audio signal is substantially compensated. The adaptive filtering models an acoustic transmission of the first audio signal and a modified second audio signal is obtained. The modified second audio signal is inverted, then it is provided via the headphone or earphone.
G10K 11/16 - Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
H03B 29/00 - Generation of noise currents and voltages
G10K 11/178 - Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effectsMasking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
51.
Adaptive filter unit for being used as an echo canceller
n, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.
H04M 9/08 - Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
G06F 17/14 - Fourier, Walsh or analogous domain transformations
H04B 3/23 - Reducing echo effects or singingOpening or closing transmitting pathConditioning for transmission in one direction or the other using a replica of transmitted signal in the time domain, e.g. echo cancellers
H04R 3/02 - Circuits for transducers for preventing acoustic reaction
During telephone calls in a public space, users may be reluctant to provide private or secret information, due to the risk of eavesdropping. A hands-free solution for entering secret information to electronic speech communication devices is based on speech processing. A method for speech processing a voice input data stream comprises steps of scanning the voice input data stream and detecting a spoken delimiter therein, determining a predefined audio sample corresponding to the detected spoken delimiter, inserting the determined predefined audio sample into the voice input data stream at the spoken delimiter, wherein a substituted voice data stream is obtained and wherein speech portions of the voice input data stream at least before the spoken delimiter remain in the substituted voice data stream, and providing the substituted voice data stream for output towards a recipient.
G10L 17/02 - Preprocessing operations, e.g. segment selectionPattern representation or modelling, e.g. based on linear discriminant analysis [LDA] or principal componentsFeature selection or extraction
H04M 3/493 - Interactive information services, e.g. directory enquiries
A shotgun microphone unit which includes a housing, a microphone capsule, a shotgun tube having a longitudinal axis, and a shotgun mounting for mounting the shotgun tube with the microphone capsule within the housing. The shotgun mounting has an axial and a radial mounting, wherein the axial mounting is softer than the radial mounting.
H04R 1/28 - Transducer mountings or enclosures designed for specific frequency responseTransducer enclosures modified by provision of mechanical or acoustic impedances, e.g. resonator, damping means
H04R 1/34 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by using a single transducer with sound reflecting, diffracting, directing or guiding means
55.
CONFIGURABLE MICROPHONE ARRAY, AND METHOD FOR CONFIGURING A MICROPHONE ARRAY
Microphone arrays with an automatic beam focusing function can detect sound sources within a search region in an autonomous manner, capture the sound of the sound sources, and output same, for example for conference telephones. Said microphone arrays can easily focus on disruptive sound sources and capture disruptive noises. In order to prevent this from happening, a user plays back a defined control audio signal from the direction (137) of a disruptive sound source (130) via a portable electronic device (110), such as a smartphone with a special app. The microphone array (100) detects the defined control audio signal and the direction (137) of the reception of said audio signal and in response thereto configures itself automatically in accordance with the control audio signal and the reception direction. For example, the reception direction can be damped or eliminated from the search region of the microphone array, or a previously configured damping or elimination of a direction can be canceled. Advantageously, the configuration is simple to carry out and does not require a defined orientation of the microphone array in the room. The elimination of parts of the search region can reduce error rates and improve the audio quality.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
A wireless microphone and/or in-ear monitoring system having a clock master prescribing a wordclock, and a clock slave to be synchronized to the wordclock. Between the clock master and the clock slave is a digital wireless transmission link which digitally transmits synchronization signals and audio signals. The clock master has a clock reference prescribing a first sample clock, and a first timer. A first phase of the first clock signal is detected after expiry of the first timer and is wirelessly transmitted to the clock slave, which has a second timer. After expiry of the second timer, a second phase of the second clock signal of the clock slave is detected and compared to the wirelessly transmitted first phase. The difference between the first and second phases is used as an input value as a control unit in the clock slave. The control unit adjusts an adjustable sample clock of the clock slave to correspond to the first clock.
The invention relates to a method for recording and synchronizing audio and video signals. The audio signal and the video signal are stored together with time stamps from a respective associated system clock. The invention relates to an adaption of the duration of the recorded audio sequence to the duration of an associated video sequence in order to level out differences in the synchronization of the two system clocks. The two system clocks are also aligned based on a data transmission which has variable waiting times for accessing a transmission channel. The invention thus allows for a clock synchronization by means such as are for example available on a smartphone.
122) of this kind. The distance (L) between the first and the second bearing point is greater than the diameter (d) of the plug contacts (230, 240) of the coaxial plug in this case.
The present invention relates to a communication system for communication of a plurality of stereo audio signals between a plurality of communication devices, wherein the plurality of communication devices comprises a first communication device, a second communication device and at least a third communication device. Each communication device of the plurality of communication devices may comprise a signal processing unit, an audio interface configured to receive a local voice signal of a user of the communication device, a binaural rendering unit configured to render the local voice signal into a stereo local voice signal based on a first spatial information, an input communication interface configured to receive a first stereo audio signal and a second stereo audio signal of the plurality of stereo audio signals transmitted by the second communication device and the third communication device, respectively. The first stereo audio signal may comprise a second voice signal of a second user of the second communication device, and the second voice signal may include a second spatial information, and wherein the second stereo audio signal may comprise a third voice signal of a third user of the third communication device, and where the third voice signal may include a third spatial information. Furthermore, the communication device may comprise an output communication interface configured to transmit a third stereo audio signal of the plurality of stereo audio signals comprising the local voice signal provided with the first spatial information to the second communication device and the third communication device. The first stereo audio signal and the second stereo audio signal may be transmitted to the audio interface, and the user of the communication device experiences a virtual sound environment, wherein the second voice signal and the third voice signal is positioned in the virtual sound environment based on the second spatial information and the third spatial information, respectively.
A mobile conference system including a first mobile master hands-free unit and a second mobile slave hands-free unit, each having a microphone, a loudspeaker, a battery unit, an operating unit, a lighting unit, and a first transmitting/receiving unit for bidirectional wireless communication between the first and second mobile hands-free units. The first mobile hands-free unit includes a second transmitting/receiving unit for wireless bidirectional audio communication with an external unit. The first mobile hands-free unit is configured to wirelessly transmit audio signals received from the external unit via the second transmitting/receiving unit to the second mobile hands-free unit via the first transmitting/receiving unit, to mix audio signals received from the second mobile hands-free unit via the first transmitting/receiving unit with audio signals recorded via the at least one microphone of the first hands-free unit, and to wirelessly transmit said audio signals to the external unit via the second transmitting/receiving unit.
Microphone arrays with automatic beam focusing may easily focus on disturbing sound sources. In order to prevent this unwanted behavior, a predefined control sound signal is replayed from a direction of a disturbing sound source. The microphone array detects the predefined control sound signal, determines the direction of replay and in response performs a re-configuration according to the control sound signal. The reconfiguration may comprise eliminating the direction from its scanning range or cancel a previously made elimination of a different direction.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
62.
Wireless microphone and/or in-ear monitoring system and method of controlling a wireless microphone and/or in-ear monitoring system
A wireless microphone and/or in-ear monitoring system having at least one first mobile device for wirelessly transmitting first audio signals. The system also has at least one base station for wirelessly receiving the first audio signals transmitted by the mobile device. The wireless transmission is based on an orthogonal frequency-division multiplexing transmission (OFDM) during a time-division multiple access (TDMA) time slot. Each wireless microphone occupies at least one slot within 2 ms. Each of the TDMA frames has a plurality of slots which respectively have precisely one OFDM symbol. Accordingly, precisely one OFDM symbol is transmitted in each TDMA slot. During a time slot made available in accordance with the TDMA, a transmission is effected on the basis of an OFDM method. The TDMA frame length is so short as a latency of <4 ms is required for professional audio transmission, for example in the case of wireless microphone systems.
H04H 20/00 - Arrangements for broadcast or for distribution combined with broadcast
H04B 1/00 - Details of transmission systems, not covered by a single one of groups Details of transmission systems not characterised by the medium used for transmission
H03D 7/00 - Transference of modulation from one carrier to another, e.g. frequency-changing
H04B 1/38 - Transceivers, i.e. devices in which transmitter and receiver form a structural unit and in which at least one part is used for functions of transmitting and receiving
The invention relates to a method for wireless audio transmission between a wireless transmitter (10) and a wireless receiver (50) in a program making special event (PMSE) system. An audio signal is transmitted from the at least one wireless transmitter (10) to the wireless receiver (50) via a first wireless transmission path (80). The at least one wireless transmitter (10) is coupled to a smart device (20) via a second wireless transmission path (30) in order to exchange parameters and/or data. In addition, an exchange of parameters and/or data occurs between the smart device (20) and the wireless receiver unit via a network, in particular the internet.
A wireless microphone or wireless in-ear monitoring system. The system has at least one transmitting and/or receiving unit, which comprises at least two antenna modules, each of the antenna modules having an output plug unit, as well as a combining unit with an input interface. Output signals of the at least two antenna modules are received via the input interface of the combining unit. This is done for those input plug units that are inserted into the input interface of the combining unit. This is carried out in order to execute diversity processing of the signals of the antenna modules.
H04B 7/08 - Diversity systemsMulti-antenna systems, i.e. transmission or reception using multiple antennas using two or more spaced independent antennas at the receiving station
H01Q 1/27 - Adaptation for use in or on movable bodies
65.
APPARATUS AND METHOD FOR PROCESSING AUDIO SIGNALS FOR IMPROVING SPEECH INTELLIGIBILITY
There are numerous applications for streaming audio data to a plurality of receivers at the same time, in which case playback is typically effected using headphones or earphones. A streaming system according to the invention provides a user with the possibility of individually correcting or adjusting the audio data, in particular improving speech intelligibility. This allows particular personal preferences or individual slight hearing damage to be taken into account. An apparatus for processing audio signals with an adjustable improvement in speech intelligibility contains a filter bank (31) for splitting an input signal into at least three signal components of different frequencies, at least three individually adjustable audio compressors (33L, 33M, 33H) which each process one of the signal components from the filter bank, a summation unit (35) for summing the output signals from the audio compressors, and a user interface for inputting at least two independent parameters: one parameter controls the overall volume and the second parameter adjusts an improvement in speech intelligibility by virtue of an increase in the second parameter in the compressor for high frequencies particularly increasing the gain for audio signals with a low volume level. Normal WLAN networks and smartphones can be used.
Multimedia devices today, and smartphones, tablets, etc., often have both a telephone function and an audio recording and/or audio reproduction function. An improved set of earphones that has stereophonic microphones and that is better suited to connection to such a multimedia device contains two ear units (271, 272), intended to be worn in or on the ear, each containing at least one sound generator (271S, 272S) and a microphone (271M, 272M) for the binaural recording of ambient sound, a microphone unit (25) containing at least one further microphone, an operator control unit (24) connected to the two ear units and the microphone unit by cable, and, as an interface, a connection unit suited to connecting the earpiece to the multimedia device. To allow easy handling, the set of earphones has at least two different function modes (11, 12) between which it is possible to change either automatically or manually.
The Higher Order Ambisonics (HOA) format is usually used for approximated sound field descriptions. It cannot, however, be directly reproduced. The invention relates to a method (100) for processing a digital audio signal, which is present as a 3D sound field description in HOA format of the order N, into a binaural reproduction format containing: filtering of the rotated 3D sound field description with rendering filters (21 L, 21 R) according to head-related impulse responses (HRIR) for a left channel and a right channel, a summation (25L, 25R) of the output signals from the rendering filters into intermediate signals (ZLi, ZR) for the left channel and the right channel, and a correction of the intermediate signals with an equalisation filter (30L, 30R) in each case to modify the timbre, wherein the equalisation filters are automatically adjusted in accordance with the order N. This generates the left and right output signals (40L, 40R) for binaural reproduction, through headphones (50), for instance. An optional upstream step comprises a rotation (10) of the 3D sound field description in accordance with a rotation control signal (16).
The invention relates to a planar dynamic sound transducer comprising a magnet arrangement, which consists of bar magnets and a mounting frame for example, and a diaphragm arrangement. The diaphragm arrangement includes a tensioned diaphragm film (diaphragm for short), a tensioning device for the diaphragm and a conductor structure applied to the diaphragm. When the conductor structure is conventionally provided with electrical connections, mechanically sensitive connections and/or high transition resistances often occur. An improved diaphragm arrangement for a planar dynamic sound transducer comprises a support frame (130), which has at least one contacting surface (200), and a diaphragm (110) tensioned on the support frame, onto which diaphragm at least one electrically conductive conductor track (120) is applied by coating. At least one end of the conductor track applied by coating extends onto the contacting surface (200) of the support frame. The connecting line (170) can be connected to the contacting surface (200) of the support frame by means of a soldered connection (160).
A method for a low-latency audio transmission in a mobile communications network is used to transmit not just first data frames or sub-frames (PRB1) coded in a first format but also alternatively shorter second data frames coded in a different second format for audio data. An audio transmission system comprises a terminal (100), a base station (200), and an audio receiver (300). The terminal (100) transmits audio data coded in the second format and other data coded in the first format via an uplink (UL1). The audio receiver according to the invention directly receives the audio data transmitted via the uplink. The coding/decoding of the audio data of one of the second data frames (LLF0) is influenced by other audio data of the same second data frame but not by the audio data of a different second data frame. Thus, the decoding of the audio data can be started earlier and the latency time is shortened in an audio receiver according to the invention. The audio transmission from the terminal to the audio receiver is carried out in the allocated time slots and frequencies in a waveform which conforms to the mobile communications network.
There is provided a method of detecting and synchronizing audio/video signals. At least one audio signal is detected by means of at least one microphone unit. Timestamps are generated and stored together with the detected audio signal in the microphone unit. An optical synchronization signal is output by the microphone unit, wherein the optical synchronization signal contains optical timestamps which are respectively associated with one of the generated timestamps. At least one video signal is detected by means of at least one camera unit. The video signal at least partially has the optical synchronization signal output by the microphone unit. The optical timestamps contained in the optical synchronization signal are extracted. The video signal and the audio signal are synchronized on the basis of the timestamps in the audio signal and the optical timestamps extracted from the detected optical synchronization signal.
H04N 21/43 - Processing of content or additional data, e.g. demultiplexing additional data from a digital video streamElementary client operations, e.g. monitoring of home network or synchronizing decoder's clockClient middleware
H04N 21/41 - Structure of clientStructure of client peripherals
H04N 21/8547 - Content authoring involving timestamps for synchronizing content
72.
Hearing device with flat flexible electric connection
The invention relates to a hearing device that comprises a flexible electric connection between device components that are pivotably connected to each other. The electric connection comprises an originally flat flexible circuit comprising one or more electric conductors embedded in a flexible substrate. The flexible circuit comprises a centre portion that is curved in its original, flat state and that in its mounted stage is bent to form a turn such that two curved sections of the curved centre portion emerge that are arranged opposite to each other in two spaced apart planes that essentially are parallel to each other.
The disclosure relates to a microphone arrangement comprising at least three groups of microphones that are mounted on a head-wearable support structure. The at least three groups of microphones comprising a first group of microphones with one or more microphones, a second group of microphones with one or more microphones, and a third group of microphones with one or more microphones, wherein the first group is mounted to a casing that accommodates signal transmission circuitry, the second group is mounted to slide with respect to the casing and the first group is mounted in a direction of a first axis. Furthermore, the third group comprises either at least one microphone that is arranged on the support structure so as to exhibit less sensitivity for sound coming from a user's mouth than for sound coming from a user's environment when the microphone arrangement is head-worn; or at least two microphones that are arranged symmetrically with respect to a user's head when the microphone arrangement is head-worn and that provide for a directionality that is orientated to the direction of a user's vision; or both.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
2; wherein the foam portion (40) has pores which are sufficiently small so that the foam portion (40) can suck water residues away from the holes (31) in the head (30) like a sponge. At the same time however the pores are of such a size that they cannot independently retain water so that water residues, after removal of the microphone unit (1) from the water, run for the greatest part out of the foam portion (40) downwardly under the effect of the force of gravity; wherein the foam portion (40) has a predominant pore density of 15 ppi to 80 ppi.
H04R 1/44 - Special adaptations for subaqueous use, e.g. for hydrophone
H04R 1/28 - Transducer mountings or enclosures designed for specific frequency responseTransducer enclosures modified by provision of mechanical or acoustic impedances, e.g. resonator, damping means
77.
Electrodynamic transducer and method for manufacturing an electrodynamic transducer
A method for manufacturing an electrodynamic transducer from a membrane system module and a magnet system module is disclosed. The membrane system module comprises an annular first chassis unit made by injection-molding and a diaphragm fixed thereon. A coil is fixed on a coil seat of the diaphragm. The magnet system module comprises a pole piece, a magnet and a yoke that each have a central hole, and a second chassis unit that is made by injection-molding and that fills the central holes in the pole piece, the magnet and the yoke. It surrounds the yoke at least partially, so that the pole piece, the magnet and the yoke are held. In the transducer, the membrane system module and the magnet system module are plugged together, wherein an annular recess on the lower side of the first chassis unit is arranged on a circumferential shoulder of the second chassis unit. This defines a position of the coil relative to the pole piece, to the magnet and to the yoke of the magnet system module.
H04R 31/00 - Apparatus or processes specially adapted for the manufacture of transducers or diaphragms therefor
B29C 45/14 - Injection moulding, i.e. forcing the required volume of moulding material through a nozzle into a closed mouldApparatus therefor incorporating preformed parts or layers, e.g. injection moulding around inserts or for coating articles
B29C 65/48 - Joining of preformed partsApparatus therefor using adhesives
H04R 9/02 - Transducers of moving-coil, moving-strip, or moving-wire type Details
n, B(t)), formed by a transformation of the second audio signal B(t) into the frequency domain; and to calculate a filtered audio signal by subtracting delayed audio estimation data from the transformed second audio signal, wherein the delayed audio estimation data is provided by a memory unit of the adaptive filter unit, which is arranged to provide a data exchange with the processor, and wherein the delayed audio estimation data comprises a frequency dependent time delay compared to the transformed second audio signal.
H04M 9/08 - Two-way loud-speaking telephone systems with means for conditioning the signal, e.g. for suppressing echoes for one or both directions of traffic
G06F 17/14 - Fourier, Walsh or analogous domain transformations
H04B 3/23 - Reducing echo effects or singingOpening or closing transmitting pathConditioning for transmission in one direction or the other using a replica of transmitted signal in the time domain, e.g. echo cancellers
H04R 3/02 - Circuits for transducers for preventing acoustic reaction
A method for transmission and low-latency real-time output and/or processing of an audio data stream that is transmitted from at least one transmitter to at least one receiver over a jittering transmission path. The method includes a calibration for determining a distribution of latencies in transmission of packets of the audio data stream, whereby a group of packets of the audio data stream is used as calibration packets and wherein a reference time grid and an offset of a fastest calibration packet are determined. Then, a shift of an output time grid for audio output and/or processing, based on the reference time grid and the determined offset of the fastest calibration packet, and the audio packets of the audio data stream are provided according to the output time grid for audio output and/or processing.
There is provided a wireless camera receiver that includes a housing, a wireless receiving unit for receiving a wirelessly transmitted audio signal, and an XLR connection. The XLR connection is adapted to be rotatable with respect to the housing.
H04N 5/60 - Receiver circuitry for the sound signals
H04N 9/82 - Transformation of the television signal for recording, e.g. modulation, frequency changingInverse transformation for playback the individual colour picture signal components being recorded simultaneously only
A method for wirelessly transmitting audio signals based on the Bluetooth protocol from a Bluetooth audio source to a computer device (audio sink). An audio signal is converted in the Bluetooth audio source into an audio data packet. The audio data packets are converted into L2CAP data packets in the Bluetooth audio source based on a protocol with access to the L2CAP layer, and wirelessly transmitted. The Bluetooth audio source suppresses renewed transmission of L2CAP data packets which were erroneous or which were not received by the sink. Real-time transmission or reproduction of the audio stream or the audio signal can thus be effected. The audio stream or an audio signal on the L2CAP layer may be transmitted with a reduced repetition rate in respect of erroneous data packets. Access to RFCOMM or another data transport protocol with access to the L2CAP layer can be provided from the application layer.
H04L 29/06 - Communication control; Communication processing characterised by a protocol
H04L 1/08 - Arrangements for detecting or preventing errors in the information received by repeating transmission, e.g. Verdan system
H04L 1/16 - Arrangements for detecting or preventing errors in the information received by using return channel in which the return channel carries supervisory signals, e.g. repetition request signals
H04L 1/20 - Arrangements for detecting or preventing errors in the information received using signal-quality detector
H04L 29/08 - Transmission control procedure, e.g. data link level control procedure
H04W 4/80 - Services using short range communication, e.g. near-field communication [NFC], radio-frequency identification [RFID] or low energy communication
82.
Wireless pocket transmitter, rechargeable battery unit for a wireless pocket transmitter, wireless microphone, rechargeable battery for a wireless microphone and charging unit for a pocket transmitter and/or a microphone
A wireless pocket transmitter having a rear side, a front side and a receiving compartment for a rechargeable battery. The receiving compartment has a rear wall which at least partially forms a part of the rear side, two side surfaces and a connecting portion with electrical contacts. The two side surfaces are each coupled with a first side to the rear wall and with a first end to the connecting portion. The second ends of the side surfaces each have a respective guide for the battery. The guides do not extend along the entire length of the side surface and the guide has two projections and a passage therebetween.
A wireless microphone and/or in-ear monitor system is proposed that has at least one clock master (TM) for prescribing a word clock and at least one clock slave (TS) that can be synchronised to the word clock prescribed by the clock master (TM). Between the clock master (TM) and the at least one clock slave (TS) there is a digital wireless transmission link that digitally transmits both synchronisation signals and audio signals. The clock master (TM) has a clock reference in order to prescribe a first sample clock (S1). The clock master further has a synchronisation interface (SY) for wirelessly transmitting a synchronisation word (S). The clock master (TM) has a first timer (T1). A first phase (P1) of the first clock signal (S1) is detected after expiry of the first timer (T1) and the first phase (P1) is wirelessly transmitted to the at least one clock slave (TS). The at least one first clock slave (TS) has a second timer (T2). After expiry of the second timer (T2), a second phase (P2) of the second clock signal (S2) of the clock slave (TS) is detected and is compared with the wirelessly transmitted first phase (P1). The difference between the first and second phases (P1, P2) is used as an input variable for a control unit (R) in the at least one clock slave (TS). The control unit (R) adjusts an adjustable sample clock of the at least one clock slave (TS) such that it corresponds to the first clock (S1) of the clock master (TM).
The invention relates to a shotgun microphone unit (1000) which comprises a housing (1400), a microphone capsule (1110), a shotgun tube (1120) having a longitudinal axis (1120a) and a shotgun mounting for mounting the shotgun tube (1120) with the microphone capsule (1110) within the housing. The shotgun mounting has an axial and a radial mounting, wherein the axial mounting is designed in a softer manner than the radial mounting.
H04R 1/34 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by using a single transducer with sound reflecting, diffracting, directing or guiding means
H04R 1/28 - Transducer mountings or enclosures designed for specific frequency responseTransducer enclosures modified by provision of mechanical or acoustic impedances, e.g. resonator, damping means
The invention relates to a microphone unit for a mobile device, in particular a smartphone or a tablet computer. The microphone unit comprises a microphone mounting (120), a microphone (130) with a first end (135) and a second end (136), and a first clamping jaw (124) with a first inner wall (121) which defines a first plane (121a) and a second inner wall (122) which defines a second plane (122a). The first inner wall (121) and the second inner wall (122) are arranged substantially perpendicularly to each other, and the first plane (121a) and the second plane (122a) form an intersection (127). The microphone mounting (120) further comprises a second clamping jaw (126) with a third inner wall (123), said second inner wall (122) and third inner wall (123) lying opposite each other in a parallel manner and being designed to rest against opposing lateral surfaces (102, 103) of a mobile device (100). In addition, the microphone mounting (120) comprises a connection piece (125) for connecting the first clamping jaw (124) to the second clamping jaw (126), said connection piece (125) and inner walls (121, 122, 123) of the clamping jaws (124, 126) delimiting a clamping region lying therebetween for clamping a mobile device (100). Lastly, the microphone unit comprises a folding device (133) which is connected to the first end (135) of the microphone (130) and is designed to pivot the microphone (130) about a pivot axis (134). The pivot axis (134) lies parallel to the intersection (127), and the folding device lies outside of the clamping region.
There is provided an electrodynamic sound transducer comprising a chassis and at least one diaphragm which is capable of vibrating and which at its edge has at least two oppositely disposed fixing portions for fixing the diaphragm to the chassis. The edge of the diaphragm is not connected to the chassis between the fixing portions so that the diaphragm can vibrate freely at those locations.
A method of low latency group-addressed audio/video streaming in an IEEE 802.11 wireless network is provided. A data stream from at least one access point is transmitted to a plurality of wireless receiving stations as multicast traffic. The data stream comprises beacon frames at beacon intervals. Data packets of the data stream from the at least one access point are transmitted to a plurality of wireless receiving stations as multicast traffic as soon as available for transmission. The transmission of data packets that were transmitted during a previous beacon interval is repeated during a subsequent beacon interval as multicast traffic.
There is provided an electrodynamic sound transducer having a diaphragm capable of vibrating, a vibrating coil coupled to the diaphragm, and a magnet system. The magnet system has a first and a second magnet ring, which are arranged above and below the diaphragm and are radially magnetized. The vibrating coil is arranged between the first and second magnet rings.
H04R 9/02 - Transducers of moving-coil, moving-strip, or moving-wire type Details
H04R 1/28 - Transducer mountings or enclosures designed for specific frequency responseTransducer enclosures modified by provision of mechanical or acoustic impedances, e.g. resonator, damping means
Disclosed is a method for joining a first component (100) which has a first thread (110) on or in a second component (200) which has a second thread (210). A second component (200) which has the second thread (210) is held upwards in a receiving unit (310) of a joining unit (300). A first component (100) which has a first thread (110) is placed downwards on the second component (200) such that the first component (100) rests on the second component (200) on the basis of its force of gravity or weight force. The second component (200) is rotationally jolted by the joining unit (300) until the first and the second threads (110, 210) engage at least partially in each other.
B25B 21/00 - Portable power-driven screw or nut setting or loosening toolsAttachments for drilling apparatus serving the same purpose
B25B 21/02 - Portable power-driven screw or nut setting or loosening toolsAttachments for drilling apparatus serving the same purpose with means for imparting impact to screwdriver blade or nut socket
G10L 21/00 - Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
H02K 7/06 - Means for converting reciprocating motion into rotary motion or vice versa
The invention relates to a mobile conference system, which has a first mobile master hands-free unit (1100) and at least one second mobile slave hands-free unit (1200), which each have at least one microphone (1130, 1230) for recording audio signals, a loudspeaker (1140, 1240) for playing back audio signals to be played back, a battery unit (1010) for supplying energy, an operating unit (1110, 1210), at least one lighting unit (1121, 1221), and a first transmitting/receiving unit (1150, 1250) for bidirectional wireless communication between the first and the second mobile hands-free units (1100, 1200). The first hands-free unit (1100) has a second transmitting/receiving unit (1160) for wireless bidirectional audio communication with an external unit. The first hands-free unit (1100) is designed to wirelessly transmit audio signals that were received from the external unit by means of the second transmitting/receiving unit (1160) to the second mobile hands-free unit (1200) by means of the first transmitting/receiving unit (1150). The first hands-free unit (1100) is designed to mix audio signals that were received from the second mobile hands-free unit (1200) by means of the first transmitting/receiving unit (1150) with audio signals recorded by means of the at least one microphone (1130) of the first hands-free unit (1100) and to wirelessly transmit said audio signals to the external unit by means of the second transmitting/receiving unit (1160).
Method and devices for providing surround audio signals are provided. Surround audio signals are received and are binaurally filtered by at least one filter unit. In some embodiments, the input surround audio signals are also processed by at least one equalizing unit. In those embodiments, the binaurally filtered signals and the equalized signals are combined to form output signals.
There is provided a stereo microphone unit having a first and a second interference tube which are arranged at an angle relative to each other and which respectively have a first end and a second end. The stereo microphone unit has a first and a second microphone capsule for detecting an audio signal. The microphone capsules are respectively provided at an end of the first and second interference tubes. The microphone unit further has a securing unit for jointly securing the first and second interference tubes, in particular to a mobile device or to a stand. There is also a cover as a wind protector which is connected to the securing unit and completely encloses the first and second interference tubes.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
A communication system for authenticate a second communication device to a first communication device, wherein the communication system comprises a physical connection between a first communication device and a second communication device, where a first message may be transmitted from the first communication device, via the physical connection, to the second communication device. Furthermore, the communication system comprises a non-physical connection between the first communication device and the second communication device, where a second message may be transmitted from the first communication device, via the non-physical connection, to the second communication device, and wherein the second communication device may be configured to evaluate the first message and the second message based on a matching criteria, and if the evaluation of the first message and the second message fulfills the matching criteria then the second communication device may be configured to transmit an acceptance to the first communication device.
H04W 12/04 - Key management, e.g. using generic bootstrapping architecture [GBA]
H04L 9/14 - Arrangements for secret or secure communicationsNetwork security protocols using a plurality of keys or algorithms
H04W 84/18 - Self-organising networks, e.g. ad hoc networks or sensor networks
H04W 4/80 - Services using short range communication, e.g. near-field communication [NFC], radio-frequency identification [RFID] or low energy communication
Thus there is provided a guitar amplifier microphone unit including at least one microphone capsule having a respective microphone capsule holder and a frame for holding the at least one microphone capsule holder. The at least one microphone capsule holder is arranged displaceably and/or rotatably on the frame.
A conference system (1000) is provided, comprising: a microphone array unit (2000) having a plurality of microphone capsules (2001 - 2017) arranged in or on a board (2020) mountable on or in a ceiling of a conference room (1001). The microphone array unit (2000) has a steerable beam (2000b) and a maximum detection angle range (2730). The conference system comprises a processing unit (2400) which is configured to receive the output signals of the microphone capsules (2001 - 2017) and to steer the beam based on the received output signal of the microphone array unit (2000). The processing unit (2400) is configured to control the microphone array (2000) to limit the detection angle range (2730) to exclude at least one predetermined exclusion sector (2731) in which a noise source is located.
H04R 1/40 - Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
The invention relates to an electrodynamic sound transducer comprising a chassis (130), a membrane (110) with a hole (150) in the center of the membrane (110), a moving coil (120), a magnetic system (140) and a resonator (200) which is placed in the hole (150) in the center of the membrane.
H04R 1/28 - Transducer mountings or enclosures designed for specific frequency responseTransducer enclosures modified by provision of mechanical or acoustic impedances, e.g. resonator, damping means
H04R 9/02 - Transducers of moving-coil, moving-strip, or moving-wire type Details
H04R 5/033 - Headphones for stereophonic communication
H04R 7/18 - Mounting or tensioning of diaphragms or cones at the periphery
According to the invention, an earpiece cable (100) is provided with a first and second end (110, 120), wherein the second end (120) has a plug (121). The cable also has at least one electrical line (140) with at least two wires (140a, 140b) and a sheath (150), and at least one eyelet unit (300, 300a, 300b) for the securing an auxiliary unit or an accessory (600). The sheath (150) is opened at one point and the eyelet unit (300, 300a, 300b) is provided there in such a way that at least one wire (140a, 140b) contacts the eyelet unit (300, 300a, 300b).
An electrodynamic sound transducer has a diaphragm, a dome and a surround and a voice coil. The sound transducer further has a first magnet ring and a second magnet ring as part of the magnet system, the first magnet ring and second magnet ring being arranged on opposite sides of the diaphragm. The voice coil is coupled with the diaphragm and is arranged approximately on or outside of the circumference of the first magnet ring and second magnet ring.
The invention relates to a hi-fi audio tube amplifier (1000) comprising a housing (500), a tube housing section (300), and a storage box (400) for headphones (700) which is integrated on the housing (500); the tube housing section (300) includes a plurality of tubes (100). Also disclosed are headphones for a hi-fi audio tube amplifier.
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