41results about How to "Improve audio quality" patented technology

The invention relates to the technical field of multi-channel audio synchronization, and discloses a multi-channel audio synchronization method and device for conference terminal equipment, equipment and a storage medium, and the method comprises the steps: collecting an audio data stream of each conference terminal equipment, and obtaining an embedded audio data stream; performing time-frequency domain conversion and peak detection to obtain an audio feature vector; calculating a cross-correlation function between the terminal devices to obtain a global time relation graph; constructing a two-layer cache structure, and adjusting the residence time of the audio data in the cache to obtain a buffer strategy; inputting the buffer strategy as a state into a multi-agent reinforcement learning model for communication resource allocation to obtain a resource allocation strategy; a global synchronization optimization problem is constructed based on a resource allocation strategy, an optimal synchronization parameter is solved through a gradient descent algorithm, time axis adjustment and content alignment are carried out on an embedded audio data stream, and a plurality of paths of synchronized audios are obtained, so that the audio quality and user experience of a multi-party teleconference are improved.

The invention relates to the technical field of audio processing, in particular to an audio optimization method and device based on artificial intelligence, computer equipment and a medium. According to the method, a linear layer is used for mapping a spectrum envelope of a to-be-optimized audio to obtain an envelope feature, an embedding layer is used for embedding a standard audio parameter into a parameter feature, a prediction model is used for predicting a fusion feature of the envelope feature and the parameter feature to obtain a prediction pitch curve, and a noise adding model is used for adding noise to a Mel spectrum of the to-be-optimized audio to obtain a predicted pitch curve. Calculating the noise of the noise addition result by using a noise estimation model to obtain predicted noise, updating the noise estimation model according to the predicted noise, the real noise and the predicted pitch curve, calculating the reference noise of the noise addition result by using the updated noise estimation model, and de-noising the noise addition result according to the reference noise. The noise estimation model is optimized in combination with pitch information, so that the denoising process meets the pitch requirement, and the audio optimization effect is improved.

The invention provides an audio parameter setting method and an electronic device. The method comprises the following steps: acquiring a sound signal played by a loudspeaker device; a frequency response curve of the sound signal is generated. And smoothing the frequency response curve of the sound signal to obtain a smoothed response curve of the sound signal. Determining a target frequency response curve according to the sound signal; and detecting a plurality of turning points of the smoothed response curve of the sound signal by using the neural network model. And determining an equalizer parameter according to a plurality of turning points of the smoothed response curve and the target frequency response curve. Therefore, the audio quality can be improved.

The invention provides a central control method and system for synchronously playing sound equipment in multiple rooms of a hotel, and is applied to the field of data synchronous processing. According to the invention, the audio stream transmission path diagram is constructed based on the network topology preset by the audio routing table, so that the system can visually check the audio stream transmission path of each room and identify and optimize links possibly causing delay, synchronous playing of cross-room sound equipment can be realized, audio signals can be ensured to be transmitted at the lowest delay, the audio quality is improved, and the user experience is improved. And meanwhile, by dynamically adjusting the compression ratio of audio coding, the system can relieve the network bandwidth pressure on the premise of keeping the audio quality, and audio transmission interruption or quality reduction caused by insufficient network bandwidth is avoided.

The invention discloses a sound mixing algorithm test method and device, electronic equipment and a storage medium, and relates to the technical field of audio processing, and the method comprises the steps: processing a sound source audio through a sound mixing algorithm, playing the sound source audio, and measuring the response time of the sound mixing algorithm to the sound source audio; recognizing the test voice by using a voice mixing algorithm, and measuring the recognition accuracy and the recognition time of the voice mixing algorithm on the test voice; analyzing the audio characteristics of the mixed audio processed by the audio mixing algorithm to obtain an analysis result; and processing the test audio by using a sound mixing algorithm to obtain a processing result, and comparing the processing result with an expected result to obtain verification data of the sound mixing algorithm. By testing the response time, the recognition accuracy and the recognition time of the sound mixing algorithm, the analysis result of the sound mixing algorithm on the mixed audio and the verification data of the test audio are obtained, the performance of the sound mixing algorithm can be comprehensively tested, the sound mixing algorithm can be optimized in a targeted mode according to the test result, important information can be conveyed in time, noise interference can be reduced, and the performance of the sound mixing algorithm is improved. And the auditory experience of the user is improved.

The invention relates to the field of data processing, in particular to an audio data dynamic transmission processing method and audio playing equipment. The method comprises the following steps: identifying key transmission nodes in an audio transmission network, and marking a plurality of transmission network nodes; carrying out multi-dimensional state sensing on a plurality of transmission network nodes, carrying out multi-dimensional environment index reconstruction, and constructing a dynamic twin network environment; obtaining audio data to be transmitted; performing low-frequency filtering optimization processing on the to-be-transmitted audio data, and performing feature segmentation processing to obtain a plurality of audio feature segments; performing data stream encapsulation processing on the plurality of audio feature segments, and performing lossless compression coding to generate a plurality of audio lossless compression codes; and based on a dynamic twin network environment, performing intelligent segmented transmission processing on the plurality of audio lossless compression codes, and extracting audio segmented transmission response data. According to the invention, efficient audio data transmission and relatively low audio error are realized.

The present invention discloses a non-invasive adaptive speaker anonymity system and method based on adversarial samples. The system includes a diverse target speaker generation module, an any-to-any adversarial perturbation construction module and an imperceptible adversarial perturbation optimization module. The diverse target speaker generation module generates target speaker embedding codes on demand according to input target labels, the any-to-any adversarial perturbation construction module constructs adversarial perturbations according to input target speaker embedding codes, and the imperceptible adversarial perturbation optimization module further optimizes the input adversarial perturbations to obtain adversarial perturbations that are imperceptible to human ears. The present invention can realize any-to-any speaker identity conversion, hide the original user's voiceprint information among a group of target speakers to adaptively avoid identity detection by an automatic speaker recognition system, and sample target embedding codes on demand through a pre-trained conditional variational autoencoder, which greatly releases the computing and storage resource requirements for actual deployment.

This invention provides an audio dynamic range control circuit and an audio device. The audio dynamic range control circuit includes an audio zero-crossing detection circuit, a preprocessing circuit, a first control circuit, a first gain selection circuit, a gain multiplication processing circuit, a second control circuit, a third control circuit, and a second gain selection circuit. The audio dynamic range control circuit provided by this invention achieves an effect similar to the existing RMS algorithm by performing absolute value processing on the linear domain sampling data of the audio signal, while simplifying the hardware circuit implementation. Simultaneously, by detecting the zero-crossing position of the audio signal through the audio zero-crossing detection circuit, and updating the gain value at subsequent zero-crossing positions when a sampling point with high energy is detected, the nonlinearity of the output signal can be greatly improved and auditory discomfort can be eliminated, effectively improving audio quality.

The utility model discloses an integrated horn and microphone structure, which comprises a horn and microphone integrated structure and a ring stand, the horn and microphone integrated structure integrates a horn and a microphone to facilitate assembly, and the horn and microphone integrated structure is fixedly arranged on the ring stand; and the PCB is arranged on the back surface of the horn and microphone integrated structure, and the horn and microphone integrated structure is controlled through the PCB. According to the utility model, the loudspeaker and the microphone are integrated, so that the assembly process is simplified; the audio quality and consistency are improved; the earphone can be used for a unique snail shell-shaped earphone shell design, so that the wearing comfort and the sound quality are improved.

The utility model provides a voice broadcast system. The voice broadcast system comprises a transmitting terminal and a plurality of receiving terminals, the transmitting end comprises an antenna, a frequency band dial switch, an audio interface, a first power interface, a plurality of denoising modules and a modulation module; the frequency band dial switch is used for selecting a target frequency band of the initial audio signal, and the target frequency band is 655 MHz to 690 MHz; the de-noising module is used for filtering the initial power supply signal and is also used for de-noising the initial audio signal to obtain a first audio signal; the modulation module is used for converting the first audio signal into a wireless signal; the antenna is used for sending the wireless signal to a receiving end; a receiving control panel and a loudspeaker are arranged in each receiving end, and the receiving control panel is used for receiving and demodulating the wireless signal to obtain a second audio signal; and the loudspeaker is used for playing the second audio signal. According to the technology, noise can be reduced, the audio quality is improved, and the phenomenon of channel crossing under multiple channels is avoided.

The invention discloses a high-signal-to-noise-ratio directional sound collection processing method, system and platform suitable for remote monitoring and a storage medium, and the method comprises the steps: obtaining to-be-processed audio original signal data in an environment in real time based on remote monitoring, processing the audio original signal data at a time, and generating corresponding first audio signal data; filtering the first audio signal data to generate corresponding second audio signal data, and converting the second audio signal data to generate third audio signal data corresponding to the second audio signal data; in combination with a deep learning model, performing noise reduction processing on the third audio signal data, and generating and backing up fourth audio signal data corresponding to the third audio signal data; according to the invention, noise can be effectively suppressed, the signal-to-noise ratio of an audio signal is improved, and the sound collection quality is improved; data tampering is prevented, and authenticity and integrity of audio data and safety of sound collection are ensured.

The invention provides a method and system for improving voice communication audio quality of a remote user, and belongs to the technical field of audio processing, and the method comprises the steps: determining target factors which influence the quality of a first audio signal, and the target factors comprise environment noise and network bandwidth; in response to the fact that the target factor is environment noise, the environment noise in the first audio signal is suppressed based on a second audio signal, a target audio signal is obtained, and the second audio signal is a signal which is equal to the environment noise in audio amplitude and opposite to the environment noise in phase; and in response to the target factor being the network bandwidth, adjusting the first audio signal based on the adjusted transmission rate to obtain a target audio signal, the adjusted transmission rate being obtained based on the network bandwidth. According to the method and the system for improving the voice communication audio quality of the remote user, the voice communication audio quality of the remote user can be improved.

The present invention relates to the field of speech recognition processing technology, and is particularly suitable for obtaining training data for machine learning models used in speech recognition transcription. In view of the high cost of data acquisition, high consumption, poor data quality / accuracy, and poor screening effect of existing pseudo-label data in different scenarios, a method, system, device, and medium for screening training data of the present invention are proposed, aiming to solve the technical problem of how to screen high-quality training data for models such as speech recognition, search, and transcription based on the accuracy of pseudo-labels of semi-supervised learning. To this end, the method of the present invention uses the mean number of node links of the decoding path in decoding to sort the decoding results to screen the pseudo-label speech data with the highest sorting as model training data. The screening efficiency and data quality are improved, and the cost and consumption are reduced.

The invention discloses a method for audio frame dynamic recombination and audio quality analysis, which comprises the following steps of: receiving an audio data packet to be detected, sorting audio data, judging whether the audio data packet has packet loss or not, if the packet loss exists, recovering the packet loss data, and simultaneously, judging the audio frame type of the audio data packet without packet loss so as to realize the audio frame dynamic recombination and the audio quality analysis. And determining an audio signal generated by the output zero-damage audio frame. And if the to-be-detected audio data packet is the interfered audio frame or the damaged audio frame, the audio signal is not generated, so that the normal work of the vocal music equipment is ensured, and meanwhile, better audio quality experience is provided for consumers.

The application provides a model training method, a speech conversion method and device, equipment and a storage medium, and belongs to the technical field of artificial intelligence. The method comprises the following steps: obtaining sample audio data of a sample speaker object; inputting the sample audio data into a neural network model comprising an encoding network and a decoding network; performing reconstruction processing on the sample audio data by using the encoding network to obtain initial audio data; performing speech alignment on the initial audio data to obtain a sample audio embedding vector; performing decoupling processing on the sample audio embedding vector, pre-acquired sample pitch parameters and a sample timbre feature vector by using the decoding network to obtain synthesized audio data; performing loss calculation on the synthesized audio data and sample speech data by using a loss function to obtain a model loss value; and performing parameter updating on the neural network model according to the model loss value to train the neural network model and obtain a speech conversion model. The application can improve the speech conversion effect.

The embodiment of the invention relates to the technical field of vehicle-mounted cabins, and discloses a cabin talkback integrated control method, system and device and a medium. When a user carries out talkback operation, the first cabin domain controller issues a PTT instruction for activating the first interphone; after receiving the PPT instruction, the first interphone collects a voice signal and transmits the voice signal to a first cabin area controller through a BLE Bluetooth link, and the first cabin area controller receives and stores the voice signal; when a user carries out receiving operation, the second interphone receives a voice signal of the first interphone and transmits the voice signal to the second cockpit domain controller through the BLE Bluetooth link; and the second cockpit domain controller receives and stores the voice signal and controls a loudspeaker to play the voice signal. The technical problem that a cockpit is low in control capability of intercom equipment can be at least solved.

Embodiments of the present application provide an audio data processing method, a server and a program product. The method is applied to a server, and the method comprises: performing redundant coding processing on to-be-processed audio data to obtain coded audio data, in a case where it is determined based on network state information that the redundant coding process is to be performed; and performing a first redundancy processing operation on the coded audio data to obtain processed audio data. The method significantly improves the quality of audio transmission in a weak network environment, reduces audio delay, freezing and distortion, and improves user experience.

The utility model discloses a sound acquisition device for short video production, and relates to the technical field of sound acquisition devices, the sound acquisition device comprises a bottom plate, the top of the bottom plate is fixedly connected with a support shell, the sound acquisition device also comprises a protection mechanism, the front end of a first threaded rod is fixedly connected with a first knob, the first knob drives the first threaded rod to rotate, and the first threaded rod is fixedly connected with the support shell. A first threaded rod drives a movable shell to move forwards on the inner wall of a fixed shell, the fixed shell drives tow rods on the two sides to move, the tow rods drive sliding plates on the two sides to slide on the inner wall of a limiting shell and move towards the center of the inner wall of the limiting shell, and the two sliding plates drive two protective shells to slide on the inner wall of a supporting shell. The two protective shells are close to each other to form a closed space to protect the sound receiving microphone, so that dust in the air is effectively prevented from being attached to and accumulated on a pickup head of the microphone to cause blockage, and the sound definition of recording and the recording quality are not affected.

The invention discloses an audio transmission method and device, equipment and a storage medium, and relates to the field of digital communication, and the method comprises the steps: obtaining a digital audio signal; encoding the digital audio signal according to a preset audio encoding format to obtain an encoded digital audio signal; partitioning the coded digital audio signal to obtain a first data block; performing interleaving processing on the first data block to obtain a second data block; and sending the second data block to a second terminal, so that the second terminal recovers the second data block to obtain a fourth data block, and performs audio reconstruction based on the fourth data block to obtain a reconstructed digital audio signal.

The invention provides an audio channel parameter adjusting method and device, a medium and a program product, and relates to the technical field of audio engineering. The audio channel parameter adjusting method comprises the following steps: detecting parameters of an audio channel; if it is detected that the parameters change, marshalling corresponding to the audio channels is obtained, the changed parameters are applied to the audio channels in the marshalling, the marshalling comprises at least two audio channels, and the marshalling is created based on a received marshalling instruction. According to the embodiment of the invention, unified adjustment of multiple audio channel parameters can be realized in an audio channel marshalling mode, the adjustment speed is high, the efficiency is high, the consistency of audio signals is effectively ensured, and the audio quality and the user experience are improved.

An audio processing method and apparatus, and an electronic device, a computer-readable storage medium and a computer program product. The method comprises: performing filtering processing on an audio signal, so as to obtain a low-frequency signal and a high-frequency signal (101); coding the low-frequency signal to obtain a code stream of the low-frequency signal (102); performing frequency-domain transformation processing on the low-frequency signal, so as to obtain a low-frequency spectrum, and performing frequency-domain transformation processing on the high-frequency signal, so as to obtain a high-frequency spectrum (103); performing spectral-envelope extraction processing on the low-frequency spectrum and the high-frequency spectrum, so as to obtain spectral-envelope information, and performing spectral-flatness extraction processing on the high-frequency spectrum, so as to obtain spectral-flatness information (104); and performing quantization coding processing on the spectral-flatness information and the spectral-envelope information, so as to obtain a frequency-band expanded code stream, and forming a coded code stream by combining the frequency-band expanded code stream and the code stream of the low-frequency signal (105)..

The present invention discloses a Bluetooth audio wall plug codec that can be connected to a Dante audio network, and belongs to the technical field of codecs. The Bluetooth audio wall plug codec that can be connected to a Dante audio network includes an audio acquisition unit, an audio processing unit, a network transmission unit and a codec. The present invention solves the problem that the transmission capacity and delay of the prior art cannot meet the existing needs, and the wiring is complicated and the cost is high in audio transmission. The present invention collects audio data from an audio input device, and performs filtering, noise reduction and enhancement on the collected audio data, thereby improving the audio quality of the audio data, thereby increasing the audibility of the audio. The processed audio data is encoded and decoded using a Dante protocol transmission codec, and the decoded audio data is converted into analog audio data and output through an external audio receiving device. In addition, the codec is trained and evaluated, which can improve the quality and encoding and decoding speed of the codec.

The embodiment of the invention provides an audio correction method and device, electronic equipment and a storage medium, and belongs to the technical field of voice recognition. The method comprises the following steps: acquiring a sample audio, and acquiring an original audio correction model; performing pitch feature extraction on the sample audio according to a pitch encoder; performing pitch correction processing on the original pitch vector according to the generator; performing content feature extraction on the sample audio according to a content encoder; performing content correction processing on the original content vector according to the generator; performing audio fusion processing on the corrected pitch vector and the corrected content vector according to a generator; performing audio quality discrimination on the corrected audio according to a target discriminator; performing parameter adjustment on the original audio correction model according to the target discrimination data to obtain a target audio correction model; and performing audio correction on a pre-acquired target audio according to the target audio correction model. The embodiment of the invention can be applied to financial science and technology, the audio quality is improved, and the accuracy of speech recognition is improved.

The application discloses a phoneme duration prediction method based on VITS neural spline flow improvement, introduces a method based on neural spline flow by improving the random duration predictor in the VITS model, effectively improves the accuracy of phoneme duration prediction, through detailed analysis of input data distribution and block interval division, the model can perform more effective coupling transformation in each block interval, which enables the model to predict the duration of phonemes with higher accuracy in actual speech synthesis applications. In addition, the application can speed up the training process and focus on optimizing the performance of the duration prediction module by freezing the model, this method not only improves the training efficiency, but also ensures the accuracy of the model in predicting the duration of phonemes, and the mean and variance of the error are used as evaluation indexes to further verify the prediction accuracy of the model, and the transformation interval boundary value is optimized by adjusting the hyperparameter B, realizing the optimization of the performance of the model.

The utility model discloses a sounding unit, which comprises a vibration assembly, a basin stand, and a magnetic circuit assembly and a shaping support chip which are arranged on the basin stand, the vibration assembly comprises a voice coil, the magnetic circuit assembly comprises a central magnetic steel and a plurality of side magnetic steels, the central magnetic steel is located in an area surrounded by the voice coil, and the plurality of side magnetic steels are wound on the periphery of the voice coil; the multiple pieces of edge magnetic steel comprise two pieces of first edge magnetic steel and two pieces of second edge magnetic steel, a gap is formed between the adjacent edges of the two pieces of second edge magnetic steel, the gap and the adjacent edges of the two pieces of second edge magnetic steel define a containing area, the two pieces of first edge magnetic steel are arranged in the containing area, and the two pieces of second edge magnetic steel are arranged in the containing area. First gaps are formed between the two ends of the first side magnetic steel and the adjacent second side magnetic steel; the shaping support chip is provided with a connecting arm, one part of the connecting arm is bonded with the bottom of the voice coil, and the other part extends outwards from the first gap. The side magnetic steel is of a surrounding voice coil structure, the whole voice coil is almost completely surrounded by a magnetic circuit, and the performance is improved.

The utility model relates to the technical field of wireless earphones, in particular to an ear clip type wireless earphone, which comprises a sounding piece provided with a sounding hole. The ear hook is connected with the sound production part; the battery compartment is arranged at one end, far away from the sound production part, of the ear hook; in a wearing state, the ear hooks are configured to bypass auricles from the auricular conchae of a user, so that the battery bins are located between the head of the user and the auricles, and the sound production holes face the ear canals of the user; one end, far away from the ear hook, of the sounding piece is provided with a first tangent plane, one end, far away from the ear hook, of the battery compartment is provided with a second tangent plane, the first tangent plane is parallel to the second tangent plane, and the distance between the first tangent plane and the second tangent plane is 1-3mm. The earphone provided by the utility model improves the sound quality definition and immersion under the condition that the wearing comfort is ensured, and reduces the sound leakage of the earphone.

The utility model discloses a sounding unit, which comprises a vibration assembly, a basin stand, and a shaping support chip and a magnetic circuit assembly which are arranged on the basin stand, the vibration assembly comprises a voice coil, the magnetic circuit assembly comprises a central magnetic steel and two side magnetic steels, the central magnetic steel is located in an area surrounded by the voice coil, and the two side magnetic steels are located in an area surrounded by the voice coil. The voice coil is located in an area defined by the two pieces of side magnetic steel, and a first gap is formed between the two pieces of side magnetic steel; the shaping support chip is provided with a connecting arm, one part of the connecting arm is bonded with the voice coil, and the other part of the connecting arm extends outwards from the first gap so as to extend out of an area defined by the two pieces of edge magnetic steel. According to the utility model, the voice coil is arranged in the area encircled by the two pieces of side magnetic steel, only the first gap is arranged between the two pieces of side magnetic steel to avoid the connecting arm of the shaped support chip, and the avoiding space is small, so that the side magnetic steel can be of a structure encircling the voice coil, the whole voice coil is almost fully surrounded by a magnetic circuit, and the performance is improved.

The present invention relates to a noise reduction type environmentally friendly Bluetooth headset and a noise reduction method thereof, the noise reduction type environmentally friendly Bluetooth headset comprises a main body shell, an ear hook, an earphone cover plate and an earplug, the earplug has a sponge ring, a silicone ring is embedded in the sponge ring, and the silicone ring is sleeved on the outside of a speaker housing, characterized in that a control panel, a power supply device, a data storage device, a Bluetooth communication device and a Beidou positioning device are arranged on the inner side of the main body shell, an earphone switch button, a volume increase button, a volume decrease button and a positioning button are arranged on the side of the main body shell, a display screen is arranged on one side wall of the main body shell, a noise reduction device and an audio collection device are arranged in the cavity of the earphone cover plate, a speaker is arranged on the inner side of the speaker housing, the noise reduction device, the audio collection device and the speaker are respectively connected to the control panel, and the main body shell, the earphone cover plate and the speaker housing are all made of degradable materials. The Bluetooth headset can not only effectively reduce noise, but also has rich functions and meets the requirements for environmental protection.

The invention discloses a debugging method of an audio processor, which relates to the technical field of audio processors, and is characterized in that the output end of an audio signal generator is connected with the input end of the audio processor through a high-quality audio cable, and the input end of an audio analyzer is connected to the output end of the audio processor; and the oscilloscope is used for monitoring and analyzing the quality of the signal output by the audio processor in real time, connecting the oscilloscope to a key node so as to observe the waveform of the audio signal and the characteristics of the electric signal, carrying out initialization setting on all connected equipment, and setting proper parameters according to debugging requirements. According to the design scheme, customized debugging and optimization are carried out according to different audio processor models, application scenes and user requirements, through scientific and reasonable debugging steps and methods, the performance potential of the audio processor can be brought into full play, the audio quality is improved, better audio experience is provided for users, and the method has wide applicability and flexibility.

The invention relates to a vehicle-mounted call noise reduction method and device, equipment and a readable storage medium. The method comprises the steps that under the condition that it is detected that a target vehicle is powered on, a noise reduction activation file corresponding to the target vehicle is acquired; according to the existence condition of the noise reduction activation file, a noise reduction engine of the target vehicle is controlled to be started; and under the condition that the noise reduction engine is successfully started, the vehicle-mounted call of the target vehicle is sent to the noise reduction engine, so that the noise reduction engine carries out noise reduction processing on the vehicle-mounted call. By adopting the method, the audio quality in the call process can be improved while the call delay is reduced.