59 results about "Sound mixer" patented technology

The invention provides an audio separation method and related equipment. The audio separation method and the related equipment are suitable for a scene in which background music is separated from songs. A to-be-processed mixed audio signal is obtained, and the mixed audio signal is a unified audio signal formed by mixing a plurality of audio signals together through a sound mixer or a sound console. Wherein the plurality of audio signals includes a human voice signal and a background sound (e.g., music sound) signal. And inputting the mixed audio signal into a diffusion model, extracting a human voice signal in the mixed audio signal, and subtracting the human voice signal from the mixed audio signal to obtain a background sound signal in the mixed audio signal. According to the scheme, the generation type model, namely the diffusion model, which is more advanced compared with a deep neural network is adopted to learn distribution of the human voice signals, and compared with a traditional method, the human voice can be estimated more accurately, so that the human voice included in the mixed audio signals is extracted more accurately; and thus, high-quality background sound is separated from the mixed audio signal.

A wearable electronic device is provided. The electronic device includes a processor operatively connected to a wireless communication circuit, a first audio mixer, a second audio mixer, a first audio output interface, a second audio output interface, and a speaker, the processor being configured to detect output of a plurality of audio signals including a first audio signal having a first property, transmit the first audio signal having the first property to the first audio mixer, transmit remaining audio signals to the second audio mixer, transmit the first audio signal to the first audio output interface to output the first audio signal through the speaker, synthesize the remaining audio signals into a second audio signal using the second audio mixer, and transmit the synthesized second audio signal to the second audio output interface to output the second audio signal through the external device.

A method is provided that is performed at a system including multiple speech collectors to collect speech from a talker to produce corresponding ones of multiple audio signals that each convey speech energy: for each audio signal: separating high-frequency speech energy from low-frequency speech energy; and determining a first energy level of the high-frequency speech energy; and determining a preferred audio signal among the multiple audio signals for subsequent processing at least based on the first energy level of each audio signal.

The invention relates to the technical field of audio signal separation, and provides an audio signal separation method and device, electronic equipment and a storage medium, and the method comprises the steps: obtaining frequency domain signals corresponding to audio signals of different channels of audio equipment; decomposing the frequency domain signal to obtain a coherent sound signal and an environment sound signal of each channel; under the condition that the energy corresponding to the environmental sound signals of the different channels is the same, determining a target environmental sound signal of each channel according to the phase difference between the environmental sound signals of the different channels; and separating the target environment sound signal of the corresponding channel from the audio signal of each channel, and obtaining the target coherent sound signal of the channel. According to the method, the device, the electronic equipment and the storage medium provided by the invention, secondary processing is carried out on the separated target coherent sound signal and the target environment sound signal by a sound mixer or a multi-channel algorithm, an auditory atmosphere with a surrounding sense and a presence sense is created, and the auditory experience of a user is improved.

Disclosed embodiments relate to Multi-Player Playback Devices. In some embodiments, the Multi-Player Playback Device is configured to selectively operate in one of a plurality of operating modes. In each operating mode, the Multi-Player Playback Device is configured to implement a multi-stage audio processing procedure comprising: (i) routing incoming audio streams to mixer inputs of a mixer based on a current operating mode in which the Multi-Player Playback Device is operating; (ii) for one or more audio streams routed to an individual mixer input, (a) generating a mixed stream based on the one or more audio streams routed to the mixer input, and (b) routing the mixed stream to at least one audio output of the Multi-Player Playback Device based at least in part on a channel map associated with the current operating mode in which the Multi-Player Playback Device is operating.

1. Name of the product in this design: Mixer (d2). 2. Purpose of this design: For use in multi-channel audio processors to mix audio signals and modulate sound. 3. The key design feature of this product is its shape. 4. The image or photograph that best illustrates the design's key points: a 3D model.

A method of adapting an automotive audio system to support surround sound comprises: configuring an application program for creation of an audio track object that specifies passthrough mode as a playback mode, wherein the application program is installed to operate under an operating system that is not configured for surround sound; configuring a framework of the operating system with a bypass past a mixer of the operating system, wherein the bypass allows surround-sound data of the audio track object to bypass the mixer; and configuring a hardware abstraction layer with a separate channel for the surround-sound data received from the operating system, wherein the hardware abstraction layer processes the surround-sound data before providing the processed surround-sound data to a decoder.

An audio system with a base station connected to multiple mobile devices. The base station and the mobile devices are configured to exchange data. The base station is connected to an audio mixer configured to receive multiple input signals from the mobile devices, mix them into a single signal, and provide it as an in-ear monitoring signal. This signal is then transmitted to a mobile device that includes an output amplifier and connected transducers. The mobile device receiving the in-ear monitoring signal has a volume control that allows the user to adjust the playback volume of the in-ear monitoring signal. The mobile device also has an audio compressor that compresses the input signal level of the in-ear monitoring signal as a function of the volume setting, ensuring that the output level of the output amplifier does not exceed a predefined limit.

A rendering method of an object-based audio signal and an apparatus for performing the same are provided. The rendering method of an object-based audio signal includes obtaining a rendered audio signal, performing clipping prevention on the rendered audio signal using a first limiter, mixing a signal output by the first limiter using a mixer, and performing clipping prevention on the mixed signal using a second limiter.

The invention provides a sound adjusting device which comprises a background sound source, an external sound source receiver, a frequency adjuster, a sound mixer and an output port. The background sound source is used for providing a background audio. The background audio is monaural audio. The external sound source receiver is used for receiving an external audio frequency from the outside. The frequency adjuster is used for receiving the background audio and adjusting the background audio into a left ear background audio and a right ear background audio. A first frequency of the left ear background audio and a second frequency of the right ear background audio have a frequency difference. The audio mixer is used for mixing the external audio and the left-ear background audio into a left-ear audio and mixing the external audio and the right-ear background audio into a right-ear audio. The output port is used for outputting the left-ear audio frequency to a left-ear horn and outputting the right-ear audio frequency to a right-ear horn.

A microphone assembly includes a microphone array and a mixer. The microphone array includes first, second, and third microphones. The mixer includes an analog switch assembly and an amplifier assembly configured to mix recording signals of the microphones in different manners to realize different recording patterns of the microphone array. The different recording patterns realizable by the microphone array may include the cardioid pattern, the omnidirectional pattern, the stereo pattern, and the bidirectional pattern.

A voice capture system including a first and second voice beamformer, a voice mixer, a voice rejected noise beamformer, a noise beamformer adjustor, a jammer suppressor, and a speech enhancer is provided. The first and second voice beamformer and the voice mixer generate a voice enhanced reference signal based on a first and second frequency domain microphone signal. The voice rejected noise beamformer includes filter weights and generates a noise reference signal based on the first and second frequency domain microphone signal. The noise beamformer adjustor adjusts the one or more filter weights of the voice rejected noise beamformer to account for fit variation. The jammer suppressor generates a jammer suppressed signal based on the voice enhanced reference signal and the noise reference signal. The speech enhancer dynamically generates an output voice signal by applying a dynamic noise suppression signal to each frequency bin of the jammer suppressed signal.

Passive IQ Mixer with Improved Conversion Gain This description relates to an IQ mixer (12) comprising a mixing stage (18) including a first mixer (MixI) and a second mixer (MixQ), and a first inductive block (BI) connected to the first mixer (MixI) and including at least one first inductive component, and a second inductive block (BQ) connected to the second mixer (MixQ) and including at least one second inductive component. Figure for the abbreviation: Fig. 4

The present application relates to the technical field of mixing sound, in particular to a kind of automatic mixing method, mixer, storage medium and electronic equipment, comprising the following steps: step S1, obtain the i-th sample value;Step S2, obtain input value;Step S3, compare input value and storage value, judge the size of input value and storage value;Step S4, obtain the first output value, and replace storage value with the first output value;Step S5, obtain the second output value, and replace storage value with the second output value;Step S6, the first output value or the second output value is output to external sound box to make external sound box output sound, control acquisition module to collect the i+1-th sample value of next moment;Square is carried out directly to single sample value and is output, reduce the loss to memory, repeatedly call output value in the process of running, make it as storage value in the calculation of next sample value, recursive process is realized, and noise interference is eliminated by controlling the activation and release of sample value.

Systems and methods are disclosed for operating a wireless audio network including a plurality of wireless microphone units (e.g., wireless delegate units) and a central access point having a mixer. The wireless microphone units may perform voice detection and level sensing, and make a preliminary gating decision. The central access point may make a final gating decision, determine the granting of wireless communications channels, and generate a final mixed audio output signal.

Various embodiments are directed to a method for adjusting cross point gains of a matrix mixer. For example, the method may receive one or more characteristics associated with an audio system comprising a plurality of microphone arrays each having one or more lobes, a plurality of loudspeakers, and a matrix mixer comprising one or more cross points each configured to connect each of the one or more lobes of the plurality of microphone arrays with each of the plurality of loudspeakers; calculate a potential gain and a desired achievable gain for each of the one or more cross points based on the one or more characteristics and a desired reinforcement level; and apply the desired achievable gain as a gain of the one or more cross points when the potential gain is greater than or equal to the desired achievable gain.

Wireless microphone or bodypack transmitter (400) with an audio input, wherein the microphone or bodypack transmitter has a transmitter and a receiver for wirelessly exchanging data with a remote station (131-135), and wherein an alphanumeric identifier of a radio transmission channel used for this purpose is displayed on the switched-on microphone or bodypack transmitter (410), wherein the identifier contains a name or an IP address, was entered on the remote station or on a mixer (140) connected thereto and was wirelessly transmitted to the switched-on microphone or bodypack transmitter, and wherein the display of the identifier is retained even when the microphone or bodypack transmitter is switched off.

Various embodiments are directed to a method for adjusting cross point gains of a matrix mixer. For example, the method may receive one or more characteristics associated with an audio system comprising a plurality of microphone arrays each having one or more lobes, a plurality of loudspeakers, and a matrix mixer comprising one or more cross points each configured to connect each of the one or more lobes of the plurality of microphone arrays with each of the plurality of loudspeakers; calculate a potential gain and a desired achievable gain for each of the one or more cross points based on the one or more characteristics and a desired reinforcement level; and apply the desired achievable gain as a gain of the one or more cross points when the potential gain is greater than or equal to the desired achievable gain.

The invention provides a conference message notification method and electronic equipment. The conference message notification method comprises the following steps: receiving a first text message from a first application; based on the first character message, obtaining a first PCM data stream corresponding to the first character message; writing the first PCM data stream into a sound mixer of the electronic equipment; and transmitting an audio data stream through the second application, wherein the audio data stream comprises the first PCM data stream.

The application discloses an audio quality inspection method, which comprises the following steps: editing a script in a script editor and generating a mix script according to the script; performing a mix operation by a mixer according to the mix script to obtain a mix result; performing a quality inspection operation on the mix result by a quality inspection device to determine whether the audio has quality problems and the type of the problems, and to propose corresponding quality inspection modification suggestions; recording quality inspection information of the problematic paragraphs to generate a quality inspection work order; and checking the quality inspection work order by a work order manager to determine whether the problems are true and whether the quality inspection modification suggestions are correct, and performing corresponding quality inspection processing operations. In addition, the application also discloses an audio quality inspection system, which comprises a script editor, a material manager, a mixer, a quality inspection device and a work order manager. The audio quality inspection method and system can efficiently and high-qualityly realize quality inspection feedback and modification processing of the produced audio in the process of generating audio books converted from texts to audios.

1. Name of the designed product: digital sound mixing console (SM). 2. Use of the designed product: for sound effect processing. 3. Design points of the designed product: in the overall shape. 4. Picture or photo that best shows the design points: perspective view. 5. No design points, omit the top view.

The invention provides a multichannel noise simulation system based on XAudio 2, and the system comprises an XAudio 2 engine initialization module which is used for initializing a COM assembly and creating an XAudio 2 engine instance; the main audio frequency sound mixer creating module is used for calling a CreateMasteringVoice method to create a main audio frequency sound mixer; the audio data loading module is used for reading an audio stream from an audio file and filling the audio stream into the XAUDIO2BUFFER structure; the source voice creating and playing module is used for calling a CreateSourceVoice method to create source voice, submitting the source voice to XAUDIO2BUFFER and starting playing; and the volume control module is used for independently adjusting the volume of each sound channel through a SetChannelVolumes function and muting other sound channels when the target sound channel is played. According to the invention, the accuracy of sound positioning and the user experience can be improved, and the system cost is reduced.

To provide a technology for realizing voice training, voice acting lessons, etc., avoiding the Three Cs (closed spaces, crowded places, and close-contact settings).SOLUTION: In a connecting booth system of the present invention in which a lecturer and a student are separately located in a plurality of physically isolated booths and a predetermined lesson can be performed, a first booth device 1A is arranged in one booth, and a second booth device 1B is arranged in the other booth; the first booth device is provided with a camera 2, a display unit 3, a microphone 4, a speaker 5, an electronic musical instrument 6, and an information terminal 7; the second booth device 1B is provided with a camera 12, a display unit 11, a mixer 13, a microphone 16, a speaker 17, an information terminal 18, and an electronic musical instrument 19; when a plurality of first booth devices are connected to the second booth device, images of the plurality of first booth devices are displayed by divided screens.SELECTED DRAWING: Figure 1

The embodiment of the invention discloses an audio processing method, system and device and a readable storage medium, and the method comprises the steps: obtaining a target audio signal and an ultrasonic signal, carrying out the gain processing of the target audio signal and the ultrasonic signal, inputting the processed signals into a sound mixer, obtaining a first audio signal, and obtaining a second audio signal; sequentially carrying out amplitude limiting and filtering processing on the basis of the first audio signal to obtain a second audio signal, carrying out up-mixing processing on the second audio signal to obtain a third audio signal, playing the third audio signal through a loudspeaker of a target sound area, and playing the third audio signal through a loudspeaker of the target sound area, and each candidate sound area is provided with a corresponding loudspeaker. Therefore, the independent playing of the audio is realized, newly added hardware is avoided, and the cost consumption during the independent playing of the audio is reduced.

A device determines estimated playback latency data that may then be used to synchronize audio presentation with other devices. Playback audio data is stored in a buffer by an application processor. A digital signal processor (DSP) retrieves frames of audio data from the buffer. The DSP may generate ultrasonic audio data and mix the playback audio data with the ultrasonic audio data. The mixed data is then sent to an audio output front end the presents the audio data that is audible to a user and the ultrasonic audio that is not. The estimated playback latency data is determined based on a buffer data quantity and a time since a last frame was consumed by the DSP. Given the estimated playback latency data, presentation of subsequent frames may be synchronized by advancing or delaying their write to the buffer, resulting in playback at a specified time.

The invention is classified in the field of audio player engineering, namely multimedia system, and is applicable to audio file reproduction. The technical effect of the invention is in expanding its functionality and versatility, interactivity and user engagement. The mechanical multimedia audio player is characterized by having a housing, which houses a mechanically driven generator connected to each other, a curved handle rigidly connected to the shaft of the generator, a voltage limiter, a power converter, a charging port, a memory card reader, an amplifier, an audio speaker, a display, a control board, a microcontroller, a switchboard, an audio player and a mixer.

A system 3 synchronises a plurality of AV devices 2 to corresponding multichannel audio. Each device 2 has preloaded images, and a distribution device 4 wirelessly connected AV device. The server 4 triggers preloaded images on each device. A cinema server 6 is connected to server 4 and configured to send time codes signals. A sound mixer 7 is cable connected to the cinema server 6 and receives audio data corresponding to VR images simultaneously with the server 4. The server 4 is cable connected to a wireless access point 5 to simultaneously message each device 2. A speaker element is provided for each channel of audio connected to mixer 7 such that upon receipt of the audio file, the mixer 7 relays each channel audio signal to respective speakers 8 in synchronisation with the VR device images.

Systems and methods for routing audio and / or video streams and remote rendered video layout for conference side group conversations and voice assistant providers are disclosed. During a main conference comprising a plurality of users, a request is received to create a side group conversation comprising a subgroup of the plurality of users. A voice assistant provider is invited to the side group conversation. Users in the subgroup may be granted different permissions to query or receive responses from the voice assistant. Incoming audio and video streams from each user are demultiplexed and decoded. Decoded audio streams are processed by an appropriate audio mixer of the group and voice assistant. Decoded video streams are processed and rendered based on a video grouping render policies. Processed audio streams for each group and voice assistant and / or rendered videos are encoded and routed to each user through their dedicated multiplexer.

Systems and methods are described herein for receiving, at a plurality of audio channels, respective audio signals captured by one or more microphones; based on a speech quality determination for each signal, identifying, in real time, a first subset of the audio channels as capturing speech audio, and a second subset of the audio channels as capturing noise audio, wherein the first subset comprises one or more audio channels and the second subset comprises one or more other audio channels; generating, using a first mixer, a mixed audio output that includes the signals received at the one or more audio channels; generating, using a second mixer, a noise mix that includes the signals received at the one or more other audio channels; and removing off-axis noise from the mixed audio output by applying, to that output, a mask determined based on the noise mix.