Tencent Technology (shenzhen) Company Limited

Huawei Device Co., Ltd.

Huawei Technologies Co., Limited

Tencent Technology Company Limited

Junbin Liang

Hongrui Jiang

Xiyong Wang

Junyang Zhou

Weijun Zheng

Jie Li

Yulong Li

Jingfan Qin

Ningbo Qiao

Fengling Wang

An audio encoding bit rate prediction model training method is performed by a computer device. The method includes: obtaining a sample audio feature parameter corresponding to each of sample audio frames in a first sample audio; performing encoding bit rate prediction on the sample audio feature parameter through an encoding bit rate prediction model, to obtain a sample encoding bit rate for each of the sample audio frames; performing audio encoding on the sample audio frames based on the corresponding sample encoding bit rates to generate sample audio data corresponding to the sample audio frames; performing audio decoding on the sample audio data, to obtain a second sample audio corresponding to the sample audio data; and training the encoding bit rate prediction model based on the first sample audio and the second sample audio until a sample encoding quality score reaches a target encoding quality score.

Audio encoding method, audio decoding method, apparatus, computer device, storage medium, and computer program product

This application relates to a speech coding method performed by a computer device. The method includes: obtaining initial frequency bandwidth feature information corresponding to a speech signal; performing feature compression on initial feature information corresponding to a second band in the initial frequency bandwidth feature information to obtain target feature information corresponding to a compressed band, a frequency interval of the second band being greater than a frequency interval of the compressed band; obtaining, based on the target feature information corresponding to the compressed band, a compressed speech signal corresponding to the speech signal; and coding the compressed speech signal to obtain coded speech data corresponding to the speech signal, a target sampling rate corresponding to the compressed speech signal being less than a sampling rate corresponding to the speech signal.

Speech coding method and apparatus, speech decoding method and apparatus, computer device, and storage medium

A data transmission method and apparatus, a terminal, a storage medium, and a system. The data transmission method includes: obtaining audio data and transmission status information, determining a compression factor and a redundancy factor based on the transmission status information, performing time domain data compression processing on the audio data according to the compression factor to obtain compressed data, performing channel coding on the compressed data according to the redundancy factor to obtain a data transmission packet, and transmitting the data transmission packet.

Data transmission method and apparatus, terminal, storage medium, and system

An electronic device performs sub-band decomposition on a to-be-coded audio to obtain a to-be-coded low frequency signal corresponding to a low frequency band and a to-be-coded high frequency signal corresponding to a high frequency band. The device performs compression coding on the to-be-coded low frequency signal to obtain low frequency coded data of the to-be-coded low frequency signal. The device determines high frequency prediction information according to the to-be-coded low frequency signal. The device performs feature extraction on the to-be-coded high frequency signal to obtain high frequency feature information. The device determines high frequency compensation information of the to-be-coded high frequency signal according to a difference between the high frequency feature information and the high frequency prediction information. The device also performs encapsulation processing on the low frequency coded data and the high frequency compensation information to obtain audio coded data of the to-be-coded audio.

Audio coding and decoding method and apparatus, medium, and electronic device

Embodiments of this application provide an audio data processing method performed by a computer device. The method includes the following steps: acquiring recorded audio; determining prototype audio matching a background reference audio component of the recorded audio from an audio database; extracting candidate speech audio from the recorded audio according to the prototype audio; determining a difference between the recorded audio and the candidate speech audio as the background reference audio component comprised in the recorded audio; performing environmental noise reduction on the candidate speech audio to obtain noise-reduced speech audio corresponding to the candidate speech audio; and combining the noise-reduced speech audio with the background reference audio component to obtain noise-reduced recorded audio.

Audio data processing method and apparatus, device, and medium

This disclosure provides a network call method and apparatus, a computer device, and a storage medium, and belongs to the field of audio data processing. The method includes: performing time-frequency transformation on an acquired audio signal, to obtain a plurality of pieces of frequency domain information of the audio signal; determining a target bit rate corresponding to the audio signal according to the plurality of pieces of frequency domain information; and encoding the audio signal based on the target bit rate, and performing a network call based on the encoded audio signal.

Internet calling method and apparatus, computer device, and storage medium

This application relates to a speech coding method, an electronic device, and a storage medium. The method includes: extracting a first speech frame feature corresponding to a first to-be-encoded speech frame, and obtaining a first speech frame criticality level corresponding to the first to-be-encoded speech frame based on the first speech frame feature; extracting a second speech frame feature corresponding to the subsequent speech frame, and obtaining a second speech frame criticality level corresponding to the subsequent speech frame based on the second speech frame feature; obtaining a criticality trend feature based on the first speech frame criticality level and the second speech frame criticality level, and determining, using the criticality trend feature, an encoding bit rate corresponding to the to-be-encoded speech frame; and encoding the first to-be-encoded speech frame based on the encoding bit rate to obtain an encoding result.

Speech coding method and apparatus, computer device, and storage medium

This application provides a packet loss retransmission method, a computer-readable storage medium, and an electronic device. The packet loss retransmission method includes: obtaining a loudness corresponding to a target audio data packet; and in response to receiving a packet loss state indicating that the target audio data packet is lost, in accordance with a determination that the loudness corresponding to the target audio data packet meets a first threshold: retransmitting the target audio data packet. The technical solutions of this application may alleviate the problem of long data retransmission time, and improve data transmission efficiency.

Packet loss retransmission method, system, and apparatus, computer-readable storage medium, and device

A call audio mixing processing method is provided. In the method, call audio streams from terminals of call members participating in a call are obtained. Voice analysis is performed on the call audio streams to determine voice activity corresponding to each of the terminals. The voice activity of the terminals indicate activity levels of the call members participating in the call. According to the voice activity of the terminals, respective voice adjustment parameters corresponding to the terminals are determined. According to the respective voice adjustment parameters corresponding to the terminals, the call audio streams of the terminals are adjusted. Further, mixing processing is performed on the adjusted call audio streams to obtain a mixed audio stream.

Call audio mixing processing

A voice processing method is provided for a terminal. The method includes: performing voice speed detection on a voice obtained from a voice source, to obtain a voice speed value of the voice; obtaining a forward error correction (FEC) redundancy; adjusting the FEC redundancy according to the voice speed value to obtain a target redundancy; performing voice encoding on the voice to obtain a voice encoded packet; performing FEC encoding on the voice encoded packet according to the target redundancy to obtain a redundancy packet; and transmitting the redundancy packet and the voice encoded packet to a receiving end.

Voice processing method and apparatus, computer- readable storage medium, and computer device

The present disclosure discloses a data transmission method performed by a computer device and a non-transitory computer-readable storage medium. According to the present disclosure, voice criticality analysis is performed on a to-be-transmitted audio to obtain a criticality level of each to-be-transmitted audio frame in the to-be-transmitted audio, and a corrected redundancy multiple of each to-be-transmitted audio frame is obtained according to a current redundancy multiple and a redundant transmission factor corresponding to the criticality level of each to-be-transmitted audio frame. Therefore, each to-be-transmitted audio frame is duplicated according to a corrected redundancy multiple of each to-be-transmitted audio frame, to obtain at least one redundancy data packet, and the at least one redundancy data packet is transmitted to a target terminal, which can improve the network anti-packet loss effect without causing network congestion.

Data transmission method and apparatus, terminal, and storage medium

A sampling rate processing method performed by a computer device are disclosed. The method includes: obtaining a first audio signal recorded by a transmitting device, the first audio signal being recorded according to an initial sampling rate of the transmitting device; obtaining a second audio signal recorded by a receiving device during playing of the first audio signal, the second audio signal being recorded according to the initial sampling rate; determining a frequency response gain value of the receiving device according to a power spectrum of the first audio signal and a power spectrum of the second audio signal; determining a target sampling rate of the transmitting device according to the initial sampling rate and the frequency response gain value; and configuring the transmitting device to record audio signals according to the target sampling rate.

Sampling rate processing method, apparatus, and system, storage medium, and computer device

An encoding parameter adjustment method is performed at a computer device. The method includes: obtaining a first audio signal, and determining a psychoacoustic masking threshold within a service frequency band in the first audio signal; obtaining a second audio signal, and determining a background environmental noise estimation value of the frequency within the service frequency band in the second audio signal; determining a masking tag corresponding to the service frequency band according to the psychoacoustic masking threshold of the first audio signal and the background environmental noise estimation value of the second audio signal; determining a masking rate of the service frequency band according to the masking tag corresponding to the frequency within the service frequency band; determining a first reference bit rate according to the masking rate of the service frequency band; and configuring an encoding bit rate of an audio encoder based on the first reference bit rate.

Encoding parameter adjustment method and apparatus, device, and storage medium

A method for improving speech signal intelligibility is performed at a device. A speech signal is obtained. A correspondence between the speech signal and a respective user group among different user groups having distinct voice characteristics is identified. Pre-encoding signal augmentation is performed on the speech signal with a respective pre-augmentation filtering coefficient that corresponds to the respective user group to obtain a group-specific pre-augmented speech signal. The device encodes the pre-augmented speech signal for subsequent transmission through the voice communication channel. An encoded version of the pre-augmented speech signal has reduced loss of signal quality as compared to an encoded version of the speech signal that is obtained without the pre-encoding signal augmentation.

Speech signal cascade processing method, terminal, and computer-readable storage medium

An audio frame loss recovery method and apparatus are disclosed. In one implementation, data from some but not all audio frames is included in a redundant frame. The audio frames whose data is not included in the redundant frame may include multiple audio frames but may not include more than two consecutive audio frames. Because not all audio frames are used in the redundant frame, the amount of information needed to be transmitted in the redundant frame is reduced. A lost audio frame during transmission may be recovered from either the redundant frame when the redundant frame incudes data of the lost frame, or from at least one neighboring frame of the lost frame derived from either the redundant frame or the successfully transmitted audio frames when the redundant frame does not include data of the lost frame.

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Shenzhen, China

Junbin Liang