409 results about "Audio mixing" patented technology

The invention belongs to the technical field of speech translation, and particularly relates to a video speech automatic translation method based on a pre-training model, and the method comprises the following steps: 1, preprocessing video and audio data; step 2, voice recognition and language detection; step 3, machine translation and text post-processing; step 4, speech synthesis and audio mixing; step 5, synchronizing video processing and subtitles; step 6, quality control and multi-dimensional evaluation; 7, carrying out model iteration and data closed loop; and step 8, system deployment and engineering implementation. Through deep fusion of efficient transfer learning of the pre-training model and the multi-modal technology, a high-precision, low-cost and easy-to-expand video speech translation solution is constructed, the time and labor cost of globalized content production is greatly reduced, the cross-language communication efficiency is improved, immersive multi-language experience is provided, and the method is suitable for popularization and application. And a data-driven continuous optimization mechanism is established, so that the system performance is improved along with the increase of the use scale.

The embodiment of the invention provides an audio processing method and device, electronic equipment and a storage medium. The method comprises the following steps: determining a target pulse response of a first audio signal, wherein the target pulse response is a stereo pulse response used for simulating sound reflection and attenuation effects of the first audio signal in an acoustic space; performing convolution processing on the first audio signal and the target pulse response to obtain a second audio signal; and performing audio mixing on the first audio signal and the second audio signal to output a third audio signal. According to the scheme provided by the invention, the stereo pulse response data capable of simulating the reflection and attenuation effects of the sound in the space can be synchronously determined when reverberation is carried out on the audio signal, and reverberation is carried out on the audio signal by using the stereo pulse response data, so that the sound is heard to be generated in different space environments; the reverberation effect of the audio signal is more vivid, the sense of space and the sense of layering of the sound are increased, and the sound is fuller and stereoscopic.

The invention discloses an audio data processing method and device, a storage medium and electronic equipment. The method comprises the steps that to-be-processed audio and at least one track point are acquired, initial audio data are generated based on the to-be-processed audio and the track point, loudspeaker gain corresponding to a target virtual loudspeaker is determined based on the initial audio data, and position information of the track point is used for determining a virtual source position in a virtual playing scene. Virtual loudspeakers are deployed in the virtual playing scene, the target virtual loudspeaker is at least one of the virtual loudspeakers, the target virtual loudspeaker represents the virtual loudspeaker determined based on the virtual source position, and the loudspeaker gain is used for simulating playing of the to-be-processed audio at the virtual source position; and performing sound mixing processing on the audio to be processed and the media information indicated by the source media information identifier based on the loudspeaker gain to generate target audio data. The technical problem that the playing effect of the audio data is poor is solved.

Methods for generating object-based audio programs that are renderable in a personalizable manner and include a speaker channel bed that is renderable without selection of other program content (e.g., to provide a default full-range audio experience). Other embodiments include steps of delivering, decoding, and / or rendering such programs. Rendering of the content of the bed or selected mix of other content of the program can provide an immersive experience. The program can include a plurality of object channels (e.g., object channels indicative of user-selectable and user-configurable objects), a speaker channel bed, and other speaker channels. Another aspect is an audio processing unit (e.g., an encoder or decoder) configured to perform any of the embodiments of the method or that includes a buffer memory storing at least one frame (or other segment) of an object-based audio program (or bitstream thereof) generated according to any of the embodiments of the method.

The invention discloses a display device and an audio control method, and relates to the field of display devices, the audio control method is applied to the display device, the display device comprises a display, an audio output interface and a controller, the controller is in communication connection with the display and the audio output interface, and the audio output interface is configured to be in communication connection with a plurality of target audio devices. The target audio equipment is used for outputting audio data; the method further comprises the following steps: under multi-window display, firstly stopping sound mixing processing on each piece of audio data; and according to the determined target relationship, establishing an audio output channel for audio data transmission between each target window and the target audio equipment, so as to realize transmission of the audio data of different windows to different audio equipment, thereby improving the audio-visual experience of a user under multi-window display.

The invention relates to the technical field of audio processing, in particular to an airborne high-reliability intelligent voice call squelch reduction method, which comprises the following steps: performing ADC conversion on an acquired analog audio signal to obtain a digital audio signal; carrying out framing and FFT conversion on the digital audio signal, carrying out phase optimization, calculating a dynamic spectrum entropy weight, combining an auditory masking compensation factor, carrying out spectrum synthesis, and carrying out IFFT conversion and overlap elimination to obtain an audio signal after sound mixing processing; noise reduction processing is carried out on the audio signal after sound mixing processing through multi-mode feature extraction, and an audio signal after noise reduction processing is obtained; performing squelch judgment of Bayesian decision optimization on the audio signal subjected to noise reduction processing through energy feature extraction to obtain an audio signal subjected to squelch processing; performing automatic gain processing on the audio signal after the squelch processing to obtain an audio signal after automatic gain; and performing DAC conversion on the audio signal after automatic gain, and outputting a noise-reduced analog audio signal.

Aspects of the present disclosure provide techniques, including devices and systems implementing the techniques, to output an adjusted audio mix to a wearable audio device in real-time using at least one pre-trained machined learned (ML) models. The pre-trained ML models enable adjustment of audio output with ultra-low latency. One example technique for adjusting audio output generally includes detecting, by one or more feedforward microphones, one or more spatialized sounds proximate a user, separating, using at least one pre-trained ML model, at least a portion of the spatialized sounds, wherein the separation is based on content of the sound, processing at least a portion of the spatialized sounds, and outputting, by the audio output device, an adaptive mix of the processed spatialized sounds. Such techniques may enable a user to filter out or reduce unwanted noises based on their sound content.

This disclosure provides methods, devices, and systems for audio signal mixing. The present implementations more specifically relate to mixing audio signals from a microphone array by performing fixed beamforming to generate beams, reducing noise on the beams, and mixing the beams to generate a final audio signal for playback. In some aspects, an audio mixing system includes a fixed beamformer to generate beams from audio signals from a microphone array and noise reduction units (NRUs) to reduce a noise component of each audio beam. The system also includes logic to calculate a signal characteristic of each reduced noise audio beam to determine, based on the signal characteristics, the reduced noise audio beams that include a speech component. The logic also generates a gain for each audio beam based on the selection, with the gains used in beam mixing. In some aspects, the NRU includes a neural network noise reduction unit.

The invention provides an audio processing method. The audio processing method comprises the following steps: determining a target sound effect level of a target scene sound effect; according to a preset weight range of each sound effect level in a plurality of sound effect levels and the target sound effect level, determining a target level weight of each sound effect level, the target sound effect level being any one of the plurality of sound effect levels; according to the target hierarchy weight of each sound effect hierarchy, performing audio track mixing on the target sound effect of each sound effect hierarchy to obtain a target scene sound effect of a target scene corresponding to the target sound effect hierarchy; the audio effect can be adjusted by adjusting the weight proportion of different sound effect levels, audio mixing is performed according to the target level weight of each sound effect level, dynamic audio mixing can be realized, immersion and dynamic adaptability can be enhanced, and user experience can be improved.

The invention provides a broadcast audio processing method and device for a broadcast intelligent audio mixing technology, and the method comprises the steps: collecting an audio signal through an audio interface, carrying out the preprocessing of the audio signal to obtain a first audio signal, and carrying out the feature extraction of the first audio signal; integrating the collected feature vectors to generate a vector matrix as the input of a preset parameter optimization prediction model; receiving a first parameter group output by the parameter optimization prediction model, processing the audio signal through an anti-distortion processing chain, and avoiding distortion caused by audio dynamic problems due to amplitude limiting processing; the audio output quality is monitored in real time, the output quality and the preset ideal audio quality are evaluated, the first parameter group is dynamically adjusted and fed back to the parameter optimization prediction model, and the problems that manual adjustment of multiple paths of audio tracks is time-consuming and prone to errors are solved.

The invention provides a method, a device and a system for audio signal mixing. The present implementation more particularly relates to mixing audio signals from a microphone array to generate beams by performing fixed beamforming, reducing noise on the beams, and mixing the beams to generate a final audio signal for playback. In some aspects, an audio mixing system includes a fixed beamformer to generate beams from audio signals from a microphone array and a noise reduction unit (NRU) to reduce noise components of each audio beam. The system also includes logic to compute a signal characteristic of each de-noised audio beam to determine a de-noised audio beam including a speech component based on the signal characteristic. The logic also generates a gain for each audio beam based on the selection, wherein the gains are used in beam mixing. In some aspects, the NRU includes a neural network noise reduction unit.

The invention provides a conference audio processing method and system based on multichannel broadcast and a medium. The method comprises the steps of obtaining conference reservation information, determining corresponding conference spokesman associated data and a semantic feature library according to the conference reservation information, obtaining a real-time conference single-path audio stream, and performing voice signal processing in combination with the semantic feature library and the conference spokesman associated data to obtain an optimized conference single-path audio stream. The method comprises the following steps: acquiring a speaking video and a speaking document image of a conference spokesman, performing spokesman separation and dynamic sound mixing processing according to the speaking video and the speaking document image in combination with an optimized conference single-path audio stream to obtain a multi-channel mixed audio stream, and performing multi-layer sound field reconstruction processing according to the multi-channel mixed audio stream to obtain an adaptive audio; according to the invention, through dynamic beam adjustment, spokesman separation, dynamic sound mixing processing and multi-layer sound field reconstruction, conference audio intelligent processing based on multi-channel broadcast is realized.

An audio processing method and an electronic device are provided. In a dual-view recording mode, the electronic device perform filtering, azimuth virtualization, and remixing on collected audio based on focal lengths of pictures in two display regions, relative positions of the two display regions, and values of areas of the two display regions, so that audio-picture presentation synchronization of the picture and the sound is achieved, and a user has synchronized three-dimensional experience in terms of hearing and vision.

Surround sound systems can dramatically expand the size of a user's sound field. Much surround sound content is mixed in a simplistic way where the front audio is copied to the rear, at a lower volume. It can be difficult for users to appreciate the value proposition of a surround sound system without more compelling spatially complex content. Quantifying surround sound complexity of various content items based on an audio spatial complexity scoring system can address this issue. Algorithms can be implemented to determine an audio spatial complexity score based on audio channels of a content item. Large catalog of content items can be analyzed, and audio spatial complexity scores can be associated with various content items. If a user has a surround sound system, content items with a high audio spatial complexity can be retrieved or recommended to the user to demonstrate the surround sound system's value better.

The invention relates to the technical field of artificial intelligence and digital audio processing, and discloses a music element dynamic recombination and automatic sound mixing data processing and making system. Comprising a music information deep representation and semantic understanding module, a music element dynamic recombination module, an intelligent adaptive sound mixing module, a multi-agent collaborative optimization and evaluation learning module, a user interaction and collaborative creation interface module and a knowledge base and learning result storage module. According to the method, deep understanding of music content is realized through multi-modal music feature extraction, structure analysis and emotion semantic dynamic modeling, sound mixing parameters are adaptively adjusted in real time according to context factors by utilizing a dynamic weight matrix mechanism, and a creation closed loop is formed by combining multi-agent collaborative optimization; the problems that in the prior art, music element recombination adaptability is poor, the sound mixing effect is homogenized, and cross-module collaboration is insufficient are solved, and the music creation efficiency and the emotion expression accuracy can be improved.

The invention provides a vehicle-mounted audio processing method, a vehicle-mounted audio processing system and a vehicle, the method is applied to a vehicle-mounted controller, and the method comprises the following steps: obtaining at least two audio signals corresponding to at least two target audio sources in a plurality of vehicle-mounted audio sources in a target vehicle, and according to the at least two target audio sources, obtaining at least two audio signals corresponding to at least two target audio sources in the target vehicle; determining a target audio mixing scene from a plurality of preset audio mixing scenes, obtaining audio mixing parameters of at least two target audio sources in the target audio mixing scene according to the target audio mixing scene, and sending an audio mixing instruction to a digital signal processing unit, so that the digital signal processing unit adopts a preset audio mixing algorithm to carry out audio mixing according to the audio mixing parameters of the at least two target audio sources. And carrying out sound mixing processing on the at least two audio signals. Therefore, sound mixing parameters are automatically adjusted based on different sound mixing scenes and audio sources, clear transmission of key information is ensured, sound mixing processing is carried out on audio signals based on the strong processing capacity of the digital signal processing unit, the response speed is optimized, and the processing delay is reduced.

The present invention provides an audio data mixing method, a terminal, and a computer-readable storage medium. The audio data mixing method includes: obtaining the same audio information transmitted by at least two audio channels respectively, where the audio information includes a standard synchronization sound source; determining the delay duration of the audio channels according to the time of receiving the standard synchronization sound source transmitted by the at least two audio channels respectively; and determining the buffer lengths of the queues respectively corresponding to the at least two audio channels according to the delay durations corresponding to the at least two audio channels. This application detects the delay times of different audio channels, and determines the buffer lengths of the queues of other audio channels based on the delay durations of all audio channels, so that the mixing module can receive audio data synchronously transmitted by different audio channels, thereby improving the mixing effect.

The invention provides a vehicle-mounted audio processing system and method and a vehicle machine system, and relates to the technical field of audio processing, the vehicle-mounted audio processing system comprises a first operating system module, a second operating system module and a power amplifier module; the first operating system module is used for processing a first input audio signal into a first output audio signal; the second operating system module is used for inputting a second input audio signal to the power amplifier module; and the power amplifier module is used for carrying out sound mixing on the second input audio signal and the first output audio signal to obtain an output audio, and is used for reducing the ear return delay of the microphone-free vehicle-mounted karaoke.

The utility model discloses an audio control system which comprises a plurality of wireless audio input devices, an audio encoding and decoding device and an audio processing device. The audio encoding and decoding equipment comprises at least two audio encoders and decoders; the plurality of wireless audio input devices are wirelessly connected with the audio encoding and decoding device and are used for transmitting audio signals to corresponding audio encoders and decoders in the audio encoding and decoding device; the number of the plurality of wireless audio input devices is twice that of the at least two audio codec, and one audio codec is connected with the two wireless audio input devices; the audio coding and decoding equipment is connected with the audio processing equipment through a first I2S bus, and each audio coding and decoding device in the audio coding and decoding equipment is used for carrying out sound mixing processing on audio signals transmitted by the corresponding wireless audio input equipment and then transmitting the audio signals to the audio processing equipment through the first I2S bus; and the audio processing equipment is connected with the audio encoding and decoding equipment through an I2C bus and is used for controlling each audio encoding and decoding device.

The invention relates to the technical field of sound therapy relaxation and sleep aiding, in particular to a staged sound scene guided multi-sound-source relaxation and sleep aiding method. A multi-sound-source sound scene output model is constructed by taking a sound bowl as a core sound source, and a parameterized control structure of a sound source set and sound mixing gain is established. A time sequence model of four stages of entering, transition, maintaining and quitting resetting is constructed according to the respiration and pulse frequency of a user, and stable stage switching is achieved by combining continuous meeting of time criteria. Structured breathing guidance is introduced in the entering stage, a sound bowl is used as a sensing anchor point in the transition and maintenance stage, paragraph arrangement is adopted, binaural rhythms are selectively superposed, and sound sources are decreased progressively according to the priority and reset prompts are added in the exiting stage. And a personalized scheme is generated by recording user data self-adaptive adjustment parameters. According to the invention, stage switching is realized to fit the physiological state, individual differences are adapted, the relaxed state is ensured to be continuous and stable, the reset quit is stable and smooth, the application scene is diversified and wide, and the relaxed sleep-aiding effect and the overall use experience are remarkably improved.

The disclosure relates to a method of rendering audio for a virtual environment. The method comprises obtaining input of one or more audio objects and corresponding positional metadata for the one or more audio objects, the one or more audio objects and corresponding positional metadata characterizing an audio mix; obtaining superseding pose-dependent modification metadata describing how the audio mix is intended to vary based on listener pose of a listener in the virtual environment; obtaining listener pose information indicative of the listener pose in the virtual environment; and adjusting levels of the one or more audio objects of the audio mix based on the superseding pose-dependent modification metadata and the listener pose information. The disclosure further relates to corresponding apparatus, computer programs, and computer-readable storage media.

The application provides a recording method, device, electronic equipment, system and storage medium, and relates to the technical field of voice processing. The method comprises the following steps: acquiring voice signals collected by at least one voice collection device; performing speaker separation on the voice signals, and adding target voice signals of each speaker separated out into different audio tracks, wherein the target voice signals and the audio tracks are in one-to-one correspondence; respectively performing optimization processing on the target voice signals in each audio track to obtain target audio track signals of each audio track; and performing mixing processing on the target audio track signals of each audio track to obtain a recording audio file. The technical scheme provided by the application can improve the quality of a recording audio in a multi-speaker recording scene.

The invention provides a real-time call translation method and device, relates to the technical field of data processing, and is applied to a Bluetooth headset, and the Bluetooth headset establishes communication connection with a mobile terminal through a Bluetooth hands-free protocol and establishes communication connection with a translation application running in the mobile terminal through a Bluetooth low-power-consumption general attribute protocol. The method comprises the following steps: acquiring an original sound to be translated, and performing audio compression coding on the original sound to be translated to obtain a compressed audio; transmitting the compressed audio to the translation application through a Bluetooth low-power-consumption general attribute protocol; receiving a translated audio returned by the translation application through a Bluetooth low-power universal attribute protocol, wherein the translated audio is an audio generated after the translation application translates the compressed audio; and carrying out sound mixing processing on the original sound to be translated and the translated audio to obtain a mixed audio, and outputting the mixed audio.

This invention discloses an adjustable frequency response microphone integrating XLR and USB outputs, comprising a microphone body, a dynamic microphone driver, a passive filter, a pass-through / preamp circuit, an XLR XLR interface, a USB module, and a function control circuit. The passive filter employs an LC circuit structure, supporting three frequency modes: low-frequency cutoff, mid-frequency boost, and flat response, effectively filtering out ambient noise and enhancing vocal quality. The pass-through / preamp circuit offers both pass-through and preamp modes, with adjustable preamp gain to accommodate dynamic microphone drivers of varying sensitivities, and can be externally or internally powered by phantom power. The USB module integrates a Type-C interface and an audio chip, supporting analog-to-digital / digital-to-analog conversion, real-time headphone monitoring, and external power supply. The function control circuit integrates an encoder and a touch panel, enabling adjustments such as volume, mixing, and mute, and enhances the interactive experience with an RGB lighting module. This invention combines the advantages of professional XLR output with convenient USB output, making it suitable for professional recording, live streaming, gaming, and other scenarios.

The invention provides a multi-room audio distribution method and system based on voice recognition, and the method comprises the steps: receiving a voice instruction of a user, and converting the voice instruction into text information; semantic analysis and intention recognition are conducted on the text information, and specific operation instructions and operation parameters are extracted; the operation instructions and the operation parameters are distributed to at least one function module related to the multi-room audio system, and the function module comprises an input source selection module, an input source volume control module, a multi-path output volume control module, a multi-path input and output routing module and a sound mixing module; the at least one function module executes corresponding operation according to the received operation instruction and the operation parameter so as to realize distribution of multi-room audios; and generating voice feedback information corresponding to the operation, and returning the voice feedback information to the user.

This disclosure provides methods, components, devices and systems for multirate audio mixing. Some aspects more specifically relate to mixing audio streams with different sample rates. In some examples, an audio source device may convert audio streams with different sample rates to the frequency domain using a modified discrete cosine transform (MDCT), and the audio source device may mix the audio streams with different sample rates in the frequency domain. The audio source device may apply a pre-emphasis filter after mixing the audio streams in the frequency domain. An audio stream with a higher sample rate may be down-sampled by dropping frequency bins after converting the audio stream to the frequency domain. Additionally, or alternatively, an audio stream with a lower sample rate may be up-sampled by padding frequency bins of the frequency domain-converted audio stream.

The application discloses a kind of self-adapting array microphone cascade mixing methods, applied to microphone cascade mixing system, belong to communication technical field.Microphone cascade mixing system includes synchronization module, channel gain equalization estimation module and adaptive mixing module, synchronization module is connected with channel gain equalization estimation module by multiple channel one communication, channel gain equalization estimation module is connected with adaptive mixing module by multiple channel two communication, synchronization module is connected with adaptive mixing module by multiple channel three communication;First microphone array device is connected with synchronization module by Ch0 channel, …, Nth microphone array device is connected with synchronization module by Ch (N-1) channel.By setting synchronization module, channel gain equalization estimation module and adaptive mixing module, and using two-stage phase level synchronization processing, mixing synchronization processing, so that the application can realize volume equalization, target speech enhancement, signal quality optimization and other multiple beneficial effects.

An electronic device comprising circuity configured to determine, for a track selected from a set of tracks, a perceptual information loss related to removing the selected track from the set of tracks.

Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators.In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators.In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

The present technology relates to an information processing apparatus, an information processing method, and a program capable of reproducing a sound pressure of a playback sound in a measurement environment. An information processing apparatus according to one aspect of the present technology records sound pressure reproduction data used to reproduce sound pressure of a speaker as a sound pressure when outputting a playback sound obtained by playback processing using transfer characteristics of sound in a measurement environment where the speaker is installed from an output apparatus prepared in a playback environment together with transfer characteristic data representing the transfer characteristics. The present technology can be applied to a system that performs audio mixing of contents such as movies.