176 results about "Audio mixing" patented technology

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 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.

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 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 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.

Techniques have been developed to facilitate the capture of performances on handheld or other portable computing devices and, in some cases, the pitch-correction and mixing of such vocal performances with backing tracks for audible rendering on such devices. Captivating visual animations and / or facilities for listener comment and ranking are provided in association with an audible rendering of a performance, e.g., a vocal performance captured and pitch-corrected at another similarly configured mobile device and mixed with backing instrumentals and / or vocals. Geocoding of captured vocal performances and / or listener feedback may facilitate animations or display artifacts in ways that are suggestive of a performance or endorsement emanating from a particular geographic locale on a user manipulable globe. In this way, implementations of the described functionality can transform otherwise mundane mobile devices into social instruments that foster a unique sense of global connectivity and community.

The invention discloses a multichannel audio playing method and device, equipment and a storage medium. The method comprises the following steps: classifying input original audio data according to sound source types to obtain a target sound source data group corresponding to each sound source type; based on a corresponding relationship between the sound source type and a preset sound channel, performing sound mixing virtualization processing on each target sound source data group to obtain independent sound channel audio data corresponding to each target sound source data group; and performing synchronization processing on the independent sound channel audio data, and playing the independent sound channel audio data after synchronization processing. According to the scheme, the target sound source data set is obtained by classifying the original audio data according to the sound source types, and then the sound mixing virtualization processing is independently performed on the target sound source data set, so that the differentiated sound mixing processing of different types of sound sources is realized, meanwhile, the independent multi-sound-channel cooperative playing based on multiple loudspeakers is realized, the multi-sound-channel audio playing effect is improved, and the user experience is improved. Therefore, the auditory experience of the user is effectively improved.

An apparatus for encoding a plurality of audio objects and associated metadata indicating directional information of the plurality of audio objects comprises: a downmixer (400) for downmixing the plurality of audio objects to obtain one or more transport channels; a transport channel encoder (300) for encoding the one or more transport channels to obtain one or more encoded transport channels; and an output interface (200) for outputting an encoded audio signal including the one or more encoded transport channels, wherein the downmixer (400) is configured to downmix the plurality of audio objects in response to directional information regarding the plurality of audio objects.

The invention relates to the technical field of audio processing, and provides a synchronous recording and broadcasting method of a low-delay microphone based on an Android system and related equipment. Acquiring an original audio stream through a microphone and transmitting the original audio stream to the memory buffer area to generate a standby audio stream; performing quantization processing and weighted fusion processing on the acquired network state parameters and the equipment performance parameters to generate a comprehensive evaluation result, and dynamically adjusting a preset audio buffer area; creating a recording thread and a playing thread in the Android system, capturing a standby audio stream in the recording thread, and writing the standby audio stream into an audio buffer area; reading the standby audio stream from the audio buffer area through the playing thread; and carrying out sound mixing processing on the standby audio stream and the audio to be played to generate a sound-mixed audio and playing the sound-mixed audio. The recording and playing threads are created to respectively process the writing and reading of the audio so as to realize parallel operation, and finally, the audio is effectively integrated through audio mixing playing, so that low-delay microphone synchronous recording and playing are realized, and the audio interaction experience is improved.

The invention discloses a sound mixing processing method and device, equipment, a storage medium and a program product, and belongs to the technical field of audio processing. The method is executed by computer equipment and comprises the following steps: acquiring an audio feature of a first song and an audio feature of a second song; obtaining each candidate boundary combination, wherein each candidate boundary combination comprises a candidate time point at which the first song fades out and a candidate time point at which the second song gradually enters; according to the audio characteristics of the first song and the audio characteristics of the second song, obtaining the sound mixing score of each candidate boundary combination; according to the sound mixing score of each candidate boundary combination, selecting a target boundary combination from each candidate boundary combination; and performing sound mixing processing on the first song and the second song according to the target boundary combination.

The invention discloses a gain sharing sound mixing system based on ping-pong timing control, which adopts two maximum value registers to store maximum values in two continuous and overlapped time slices, so that the storage requirement on data of the whole sound source data time slice is avoided, and the use of storage resources can be greatly reduced; meanwhile, when new data exist each time, the maximum value is updated, only the previous maximum value needs to be compared with the current input value, all data of the time slice does not need to be traversed, and the operand can be greatly reduced. The technical problem that an existing gain sharing sound mixing method is large in audio data storage amount and large in calculation amount is solved.

The invention is suitable for the technical field of acoustic processing, and provides a voice propagation method capable of absorbing and reflecting, used equipment comprises a panel installed in a recording studio, the panel is provided with a plurality of conversion grooves arranged in a rectangular array, and the conversion grooves are internally provided with conversion assemblies capable of converting between a sound absorption mode and a reflection mode. A driving assembly for controlling the conversion assemblies to operate is arranged on the outer side of the panel, and the operation steps include S1, when pure'dry sound 'needs to be recorded in the recording studio, the driving assembly is started, the driving assembly drives the conversion assemblies in all the conversion grooves to operate, and the conversion assemblies are converted into sound absorption modes, and S2, when mixed sound needs to be recorded, the driving assembly is started, and the driving assembly drives the conversion assemblies in all the conversion grooves to operate. According to the sound processing method, the static state is converted into the dynamic state, one-key switching of acoustic scenes is achieved, and seamless conversion between the sound absorption mode and the reflection mode can be achieved in the same physical space according to actual requirements.

An AI-powered multimedia workstation integrates a digital audio workstation (DAW), visual graphics editor, and office productivity tools within a single platform. Operating on an AI / quantum-inspired software system, it performs real-time audio, visual, and data processing. A central AI / Quantum Content DNA Engine enables structural, stylistic, and semantic analysis, separation, and manipulation of multimedia content, including audio, video, images, and text. The workstation features a hybrid control surface with dynamic haptic feedback faders, high-resolution touch displays, and professional audio input / output interfaces. AI functions include automated music composition, intelligent mixing and mastering, cross-modal content generation, and context-aware workflow assistance. Quantum-inspired algorithms accelerate rendering, simulation, and large-scale data analysis. The system supports real-time global collaboration via high-speed network protocols and cross-platform compatibility with external devices and software. By combining AI-assisted and quantum-inspired processing, the workstation enhances creative production, performance, and productivity, providing an integrated environment for professional audio, visual, and office applications.

The invention discloses a vehicle-mounted outdoor concert implementation method, an automobile and a storage medium, and the method is executed by a vehicle-mounted audio system, and comprises the steps: obtaining the type information of a field musical instrument inserted by a user; accompaniment music data is acquired, and a target audio track corresponding to the field musical instrument type is identified and separated; performing audio energy weakening or elimination processing on the extracted target audio track to generate target accompaniment music; obtaining a human voice audio signal sung by a user and a playing audio signal of a field musical instrument in real time, and receiving target accompaniment music; the three paths of audio signals are subjected to real-time audio mixing processing to generate mixed audio signals, and the power amplifier drives the external loudspeaker to play the mixed audio signals, so that the vehicle-mounted outdoor music meeting effect is achieved. According to the invention, the playability and entertainment of the vehicle-mounted audio system are greatly improved, and meanwhile, the individual requirements of music enthusiasts hosting musical instruments on using the musical instruments can be effectively met.

The present application relates to a kind of audio output control method and related equipment.The control method is applied to terminal equipment, comprising: establishing the corresponding relationship of first audio playing device and first target sound source;Obtain all audio track inputs corresponding to all sound sources in terminal equipment, identify all audio track inputs to obtain the first target audio track input corresponding to first target sound source according to the corresponding relationship;According to the corresponding relationship, the first target audio track input is transmitted to first audio playing device, so that the audio data corresponding to first target sound source is output by first audio playing device.Independent processing and transmission of the audio corresponding to multiple sound sources can be realized by implementing the present application, and the mixing interference between sound sources is avoided.

The invention discloses a multi-product intelligent sound mixing method with a wireless microphone as a sound amplification device A, a microphone as a sound amplification device B and a sound amplification device C used for positioning the position of a sound source to realize automatic sound mixing of the three sound amplification devices at the same time, and the method comprises the following specific steps: (1) judging whether the sound amplification device A works or not, and if the sound amplification device A works, starting the sound amplification device C; if yes, the sound amplification equipment B and the sound amplification equipment C do not amplify sound; (2) if the sound reinforcement equipment A does not work, positioning the sound source position based on the sound reinforcement equipment C, and judging whether to start the sound reinforcement equipment B for sound reinforcement; and (3) if the sound reinforcement equipment A does not work and the sound source is also far away from the sound reinforcement equipment B, starting the sound reinforcement equipment C for sound reinforcement. According to the invention, clear audio files can be picked up under the condition of multi-device sound amplification, and clean original sound is provided for subsequent playback.

The invention discloses a hierarchical strategy-planner hybrid architecture for audio-visual navigation (AV-Nav) of a body-equipped agent, and aims to solve the challenge of tight coupling of multi-modal reasoning and precise action control in a traditional end-to-end model. According to the high-level strategy, a Transform-based network is adopted, deep fusion is carried out on real-time visual and acoustic features through a token and self-attention mechanism, and a navigation target point on a 3 * 3 local grid is output. The local target is then passed to a low-level classical planner that calculates the shortest path on the internally maintained dynamic navigation map using the Dijkstra algorithm and determines the atomic action to be performed. The method is characterized in that a planner is integrated with an interactive trial and error obstacle avoidance mechanism: once a collision signal Ct is received from the environment, the planner can immediately remove a corresponding blocked path edge from a navigation map, so that a system is forced to automatically re-plan a collision-free path in the next time step, and the navigation efficiency and robustness are remarkably improved. In addition, the architecture may perform end-to-end training in conjunction with novel audio enhancement strategies, including interfering sound sources, mixing or spectrogram masks.

The invention discloses a pickup acquisition and playing implementation method and system, and intelligent equipment. The method comprises the following steps: acquiring first audio data and second audio data; the second audio data and the first audio data after reverberation reduction processing enter an annular buffer queue in sequence according to a time sequence; synchronously performing audio preprocessing operation on the first audio data and the second audio data; performing sound mixing processing to obtain target audio data; obtaining target audio data after noise reduction; and playing the target audio data in the playing buffer area according to the audio sampling rate. According to the invention, the audio data in each audio data processing step is allocated, and timing processing of the data is carried out through sound card acquisition, so that the audio data can be acquired and output according to a stable time interval; and finally, storing the target audio data into a playing buffer area to ensure that the playing speed of the plurality of sound cards can be stably maintained.

The application provides a method and system for generating personalized music based on AI, and related equipment, the method comprising receiving a multi-modal creation request of a user, the request including one or more of a text description, an image, a video, and a humming melody; preprocessing and cross-modal semantic analysis of the multi-modal creation request to generate structured music semantic parameters; then using a structure generation model to generate a symbolic intermediate representation of information defining the structure of the musical form, harmony, and melody based on the music semantic parameters, and then rendering a multi-track audio containing specific instrument timbre and performance details through a texture rendering model conditioned on the intermediate representation; finally, adaptive mixing and mastering of the audio to output a final music file. The method of the application realizes diversified user input, controllable music structure, and automatic post-processing, thereby improving the intelligent level of personalized music production.

A method for generating music on the basis of artificial intelligence, according to an embodiment of the present invention, may comprise the steps of: receiving at least one of text information, image information, music information, and video information; extracting an embedding vector on the basis of the received information; calculating a similarity between the extracted input embedding vector and an embedding vector of a component block included in a database; on the basis of the similarity, selecting a reference component block from the database; filtering the database on the basis of the selected reference component block; selecting one or more component blocks from the filtered database; generating a reference section on the basis of the reference component and the selected one or more component blocks; and generating one or more audio sections on the basis of the generated reference section, and listing the reference section and the one or more audio sections in a time series.

This application discloses an audio processing method, apparatus, terminal, and computer program product, belonging to the field of audio technology. The method includes: determining the loudness difference between at least two audio streams, wherein the at least two audio streams are synchronously played; when the loudness difference is outside a certain range, performing loudness amplification and loudness compression processing on the at least two audio streams; and performing mixing processing on the loudness-processed at least two audio streams. The solution provided by this application maintains a reasonable loudness difference between at least two synchronously played audio streams, improving audio playback quality in scenarios with simultaneous playback of multiple audio streams and enhancing the user's audio listening experience.

This invention discloses a context-aware adaptive tourism audio narrative method and system, relating to the field of interactive entertainment technology, including the following steps: collecting multi-dimensional context data through a multi-dimensional perception module; calculating team aggregation degree, average team movement speed, and environmental status based on the multi-dimensional context data; constructing a preset script through a dynamic plot engine module, processing multi-dimensional context data, and making dynamic plot decisions; generating and broadcasting adaptive audio through an adaptive mixing module; and achieving real-time adaptive linkage between plot content and real-world scenes through deep integration of multi-dimensional environmental perception, dynamic plot decisions, and intelligent mixing technology. It can automatically select and broadcast matching narrative branches based on multiple context data such as weather, crowd flow, and team status. Simultaneously, through spatial audio rendering and intelligent fusion of ambient sound, it breaks down the auditory boundaries between the plot and reality, significantly enhancing tourists' immersion and emotional resonance.

Provided in the embodiments of the present application are an audio processing method, an electronic device, a computer storage medium, and a computer program product. The audio processing method comprises: acquiring shared audio data from shared audio and video, and remote-end audio data received from a remote end; acquiring first audio data from an audio conversion apparatus, and collecting second audio data by means of an audio collection apparatus, wherein the first audio data is obtained by inputting the shared audio data and the remote-end audio data into the audio conversion apparatus, and the second audio data includes the first audio data, which is played by an audio playback apparatus; using the first audio data as reference data to perform echo cancellation processing on the second audio data, so as to obtain primary echo-canceled data; and on the basis of the primary echo-canceled data and the shared audio data in the first audio data, performing an audio mixing operation, so as to obtain near-end audio data. By means of the embodiments of the present application, users can be prevented from hearing the superimposed sound of shared audio.

The invention discloses a resource optimization method and system for decision making of a main processor and scheduling in an audio processor. The method comprises the following steps: acquiring processor load information when the audio processor performs audio processing on multiple audio input channels and channel feature information of each audio input channel; when the processor load information meets a preset load condition, the main processor determines an adjustment result of audio processing configuration according to a preset resource optimization rule based on the processor load information and the channel feature information; the main processor generates and sends a control instruction to the audio processor according to the adjustment result; and the audio processor updates the sound mixing routing relationship in the internal scheduling according to the control instruction, and carries out switching or intensity adjustment on the processing link of each channel according to the processing algorithm configuration. According to the technical scheme, the continuity and the stability of conference audio processing are improved on the premise of not changing a hardware structure.