Abstract: A method and a device for improving audio processing performance are provided. In some embodiments, the method includes: acquiring audio data in an audio call in a terminal; processing the audio data by utilizing an audio processing algorithm, to obtain a characteristic parameter of the audio data; analyzing the processing quality of the audio data according to the characteristic parameter of the audio data; and if the processing quality of the audio data does not reach a preset quality standard, optimizing the audio processing algorithm by utilizing optimization data matching the terminal.

Abstract: An echo cancellation data synchronization control method is disclosed. The method includes: estimating a sound card delay value; initializing and accumulating a near-end audio buffer queue and a reference audio buffer queue until a difference obtained by subtracting a length of the near-end audio buffer queue from a length of the reference audio buffer queue is greater than or equal to an audio data length corresponding to the sound card delay value; extracting, in accordance with audio frames, audio data from the head of the reference audio buffer queue and the head of the near-end audio buffer queue for echo cancellation processing; obtaining a relative delay value generated by performing echo cancellation processing; and adjusting the sound card delay value according to the relative delay value.

Abstract: Provided are a sound effect processing method and device. The sound effect processing method includes: invoking a pre-loaded plug-in unit manager to acquire a sound effect processing parameter supported by each preloaded sound effect plug-in unit; acquiring a sound effect configuration file pre-configured by the plug-in unit manager; displaying sound effect index identifiers corresponding to various sound effect modes; according to a selected sound effect index identifier, determining a selected sound effect mode, and according to an adjustment interface, acquiring adjusted parameter control data; sending the adjusted parameter control data to a selected sound effect plug-in unit; and invoking the plug-in unit manager to send data to be processed to the selected sound effect plug-in unit, and according to the adjusted parameter control data, conducting sound effect processing on the data to be processed by the selected sound effect plug-in unit.

Abstract: Audio encoding methods/terminals, audio decoding methods/terminals, and audio codec systems are provided. A plurality of audio signals that are continuous is obtained. it is determined whether each audio signal of the plurality of audio signals includes a designated signal type, according to an audio parameter of each audio signal. A marked audio encoding stream is obtained by performing a marking to each audio signal as having or not having the designated signal type. The marking is used, at a decoding terminal, to perform an enhancement-process to one or more audio signals having the designated signal type. The enhancement-process is not performed to audio signals that do not have the designated signal type.

Abstract: According to an embodiment, a power spectrum iteration factor is determined according to a noisy speech and a background noise, and a moving average power spectrum of the speech is obtained according to the power spectrum iteration factor. A server is able to trace the noisy speech according to the power spectrum iteration factor.

Abstract: Audio encoding methods/terminals, audio decoding methods/terminals, and audio codec systems are pro vided, A plurality of audio signals that are continuous is obtained, it is determined whether each audio signal of the plurality of audio signals includes a designated signal type, according to an audio parameter of each audio signal. A marked audio encoding stream is obtained by performing a marking to each audio signal as having or not having the designated signal type. The marking is used, at a decoding terminal, to perform an enhancement-process to one or more audio signals having the designated signal type. The enhancement-process is not performed to audio signals that do not have the designated signal type.

Abstract: Provided is a method for detecting an audio input interface, comprising: acquiring input signals of every one of audio input interfaces; acquiring energy values of said input signals of every one of said audio input interfaces, and acquiring an interface identification of the audio input interface of which the energy value is the maximum among the energy values; adding said interface identification acquired herein into an identification sequence in an order by acquiring time; and identifying the audio input interface, of which the interface identification is saved most often in said identification sequence, as a valid audio input interface. Also provided is a system for detecting an audio input interface. The present disclosure is capable of effectively identifying the audio input interface connected to the microphone that the user is using, so that the user does not need to switch manually.

Abstract: Systems and methods are provided for data transmission. For example, packet loss is detected based on at least information associated with a first data packet; and in response to packet loss being detected, a packet-retransmission request is sent to a data-transmission terminal and the packet-retransmission request is repeatedly sent to the data-transmission terminal during a retransmission life-cycle associated with the first data packet, the packet-retransmission request including a first identification of the packet-retransmission request.

Abstract: A method and a device for improving audio processing performance are provided. In some embodiments, the method includes: acquiring audio data in an audio call in a terminal; processing the audio data by utilizing an audio processing algorithm, to obtain a characteristic parameter of the audio data; analyzing the processing quality of the audio data according to the characteristic parameter of the audio data; and if the processing quality of the audio data does not reach a preset quality standard, optimizing the audio processing algorithm by utilizing optimization data matching the terminal.

Abstract: Audio encoding methods/terminals, audio decoding methods/terminals, and audio codec systems are provided. A plurality of audio signals that are continuous is obtained. It is determined whether each audio signal of the plurality of audio signals includes a designated signal type, according to an audio parameter of each audio signal. A marked audio encoding stream is obtained by performing a marking to each audio signal as having or not having the designated signal type. The marking is used, at a decoding terminal, to perform an enhancement-process to one or more audio signals having the designated signal type. The enhancement-process is not performed to audio signals that do not have the designated signal type.

Abstract: An echo cancellation data synchronization control method is disclosed. The method includes: estimating a sound card delay value; initializing and accumulating a near-end audio buffer queue and a reference audio buffer queue until a difference obtained by subtracting a length of the near-end audio buffer queue from a length of the reference audio buffer queue is greater than or equal to an audio data length corresponding to the sound card delay value; extracting, in accordance with audio frames, audio data from the head of the reference audio buffer queue and the head of the near-end audio buffer queue for echo cancellation processing; obtaining a relative delay value generated by performing echo cancellation processing; and adjusting the sound card delay value according to the relative delay value.

Abstract: Systems and methods are provided for audio encoding. For example, first quantization encoding is performed on audio data associated with a current frame of an audio data stream to obtain first quantization-encoded data; second quantization encoding is performed on the audio data to obtain second quantization-encoded data; the first quantization-encoded data is coupled to the current frame of the audio data stream; and the second quantization-encoded data is coupled to a next frame of the audio data stream.

Abstract: Provided are a sound effect processing method and device, a plug-in unit manager and a sound effect plug-in unit, which belong to the field of information technology processing.

Abstract: According to an embodiment, a power spectrum iteration factor is determined according to a noisy speech and a background noise, and a moving average power spectrum of the speech is obtained according to the power spectrum iteration factor. A server is able to trace the noisy speech according to the power spectrum iteration factor.

Abstract: The present disclosure discloses a method and system for sending media data in telepresence technology. By collecting audio or video data corresponding to a spatial area, the collected data is processed respectively to acquire multiple streams of data distinguished by a collecting spatial area and corresponding collecting spatial information; the multiple streams of data are encoded, encoded multiple streams of audio data and the collecting spatial information are packed together, and/or encoded multiple streams of video data and the collecting spatial information are packed together, and a packet including multiple streams of audio or video data is sent, respectively. The present disclosure also discloses a method and system for playing media data in telepresence technology. With the methods and systems, it is possible to directly identify the corresponding collecting spatial area, i.e.

Abstract: The present disclosure discloses a method and system for sending media data in telepresence technology. By collecting audio or video data corresponding to a spatial area, the collected data is processed respectively to acquire multiple streams of data distinguished by a collecting spatial area and corresponding collecting spatial information; the multiple streams of data are encoded, encoded multiple streams of audio data and the collecting spatial information are packed together, and/or encoded multiple streams of video data and the collecting spatial information are packed together, and a packet including multiple streams of audio or video data is sent, respectively. The present disclosure also discloses a method and system for playing media data in telepresence technology. With the methods and systems, it is possible to directly identify the corresponding collecting spatial area, i.e.

Abstract: The present disclosure discloses a method and system for sending media data in telepresence technology. By collecting audio or video data corresponding to a spatial area, the collected data is processed respectively to acquire multiple streams of data distinguished by a collecting spatial area and corresponding collecting spatial information; the multiple streams of data are encoded, encoded multiple streams of audio data and the collecting spatial information are packed together, and/or encoded multiple streams of video data and the collecting spatial information are packed together, and a packet including multiple streams of audio or video data is sent, respectively. The present disclosure also discloses a method and system for playing media data in telepresence technology. With the methods and systems, it is possible to directly identify the corresponding collecting spatial area, i.e.

Abstract: Methods, apparatus, and systems for voice processing are provided herein. An exemplary method can be implemented by a terminal. A voice bit stream to be sent can be obtained. Voice control information corresponding to the voice bit stream to be sent can be obtained. The voice control information can be used for a voice server to determine a voice-mixing strategy. The voice bit stream and the voice control information can be sent to the voice server. At least one voice bit stream, returned by the voice server based on the voice-mixing strategy, can be received. The at least one voice bit stream can be outputted.

Abstract: Systems and methods are provided for data transmission. For example, packet loss is detected based on at least information associated with a first data packet; and in response to packet loss being detected, a packet-retransmission request is sent to a data-transmission terminal and the packet-retransmission request is repeatedly sent to the data-transmission terminal during a retransmission life-cycle associated with the first data packet, the packet-retransmission request including a first identification of the packet-retransmission request.

Abstract: Systems and methods are provided for audio encoding. For example, first quantization encoding is performed on audio data associated with a current frame of an audio data stream to obtain first quantization-encoded data; second quantization encoding is performed on the audio data to obtain second quantization-encoded data; the first quantization-encoded data is coupled to the current frame of the audio data stream; and the second quantization-encoded data is coupled to a next frame of the audio data stream.