Methods and devices for controlling speech quality

    公开(公告)号:US10212552B2

    公开(公告)日:2019-02-19

    申请号:US15007338

    申请日:2016-01-27

    Abstract: The disclosure relates to methods and devices for controlling speech quality, in particular by controlling end-to-end latency and by improving speech quality in case of mobility scenarios. A method 200 for controlling end-to-end latency between receiving and processing audio frames includes: receiving 201 a data packet comprising at least one coded audio frame; storing 202 the received data packet in a packet buffer; retrieving 203 the received data packet from the packet buffer and decoding the at least one coded audio frame into audio samples; and processing 204 the audio samples, wherein a scheduling of retrieving 203 the received data packet from the packet buffer and decoding the at least one coded audio frame is based on a target criterion with respect to audio quality of the audio samples and latency between receiving the data packet and processing the audio samples, and wherein the scheduling is dynamically and smoothly shifted in time in order to avoid audio distortions.

    OPTIMIZING SYNCHRONIZATION OF AUDIO AND NETWORK TASKS IN VOICE OVER PACKET SWITCHED NETWORKS
    2.
    发明申请
    OPTIMIZING SYNCHRONIZATION OF AUDIO AND NETWORK TASKS IN VOICE OVER PACKET SWITCHED NETWORKS 有权
    通过分组交换网络优化音频和网络任务在语音中的同步

    公开(公告)号:US20160094605A1

    公开(公告)日:2016-03-31

    申请号:US14500119

    申请日:2014-09-29

    Abstract: A user equipment device (UE) comprises physical layer circuitry configured to transmit and receive radio frequency electrical signals with one or more nodes of a radio access network, an audio subsystem configured to generate frames of audio data, and processing circuitry. The processing circuitry is configured to calculate a time delay from generation of an audio data frame by the audio subsystem of the UE device to transmission of an audio data packet by the physical layer circuitry during a voice call, and decrease the time delay to a delay value that preserves a specified minimum time for delivery of the generated audio data frame to the physical layer circuitry to meet a scheduled transmission time of the audio data packet.

    Abstract translation: 用户设备设备(UE)包括物理层电路,其被配置为用无线电接入网络的一个或多个节点发送和接收射频电信号,配置成生成音频数据帧的音频子系统和处理电路。 处理电路被配置为计算由UE设备的音频子系统产生音频数据帧到语音呼叫期间由物理层电路传输音频数据分组的时间延迟,并将时延延迟到延迟 值,其保留用于将生成的音频数据帧传送到物理层电路的指定最小时间,以满足音频数据分组的调度传输时间。

    Communication terminal and method for switching a call between radio access technologies

    公开(公告)号:US10341904B2

    公开(公告)日:2019-07-02

    申请号:US14670543

    申请日:2015-03-27

    Abstract: A communication terminal comprising a first communication component configured for communication using a first radio access technology, a second communication component configured for communication using a second radio access technology, wherein the first communication component is configured to communicate speech data of a call using the first radio access technology; and a controller configured to control the second communication component to generate transport protocol packets comprising speech data of the call for uplink transmission of the speech data using the second radio access technology; andto switch the call from the first radio access technology to the second radio access technology access technology after the second communication component begins generation of transport protocol packets for uplink transmission.

    METHODS AND DEVICES FOR CONTROLLING SPEECH QUALITY
    4.
    发明申请
    METHODS AND DEVICES FOR CONTROLLING SPEECH QUALITY 审中-公开
    用于控制语音质量的方法和设备

    公开(公告)号:US20160286575A1

    公开(公告)日:2016-09-29

    申请号:US15007338

    申请日:2016-01-27

    Abstract: The disclosure relates to methods and devices for controlling speech quality, in particular by controlling end-to-end latency and by improving speech quality in case of mobility scenarios. A method 200 for controlling end-to-end latency between receiving and processing audio frames includes: receiving 201 a data packet comprising at least one coded audio frame; storing 202 the received data packet in a packet buffer; retrieving 203 the received data packet from the packet buffer and decoding the at least one coded audio frame into audio samples; and processing 204 the audio samples, wherein a scheduling of retrieving 203 the received data packet from the packet buffer and decoding the at least one coded audio frame is based on a target criterion with respect to audio quality of the audio samples and latency between receiving the data packet and processing the audio samples, and wherein the scheduling is dynamically and smoothly shifted in time in order to avoid audio distortions.

    Abstract translation: 本公开涉及用于控制语音质量的方法和设备,特别是通过控制端到端等待时间并且在移动性场景的情况下改善语音质量。 用于控制接收和处理音频帧之间的端到端等待时间的方法200包括:接收201包括至少一个编码音频帧的数据分组; 将所接收的数据分组202存储在分组缓冲器中; 从分组缓冲器检索203所接收的数据分组,并将所述至少一个编码音频帧解码为音频样本; 以及对所述音频样本进行处理204,其中从所述分组缓冲器检索203所接收的数据分组并对所述至少一个编码音频帧进行解码的调度基于关于音频样本的音频质量的目标标准和接收到 数据分组并处理音频样本,并且其中调度在时间上动态平滑地移位以避免音频失真。

    Device and method for processing audio data

    公开(公告)号:US09713192B2

    公开(公告)日:2017-07-18

    申请号:US14670441

    申请日:2015-03-27

    Abstract: A device is described comprising an audio processor configured to perform audio processing during an audio communication connection between the communication device and a second communication device, a memory configured to store at least a first audio processor wakeup schedule in accordance with a first predefined audio communication connection event and a controller configured to set at least one second audio processor wakeup schedule in accordance with a second predefined audio communication connection event being similar to the first audio processor wakeup schedule, and wake up the audio processor in accordance with at least one of the first audio processor wakeup schedule or the second audio processor wakeup schedule in response to the incidence of at least one of the first or second predefined audio communication connection events, wherein the audio processor is configured to enter into a sleep mode after the audio processing is complete and to enter into a processing mode from the sleep mode in response to a wake up signal received from the controller.

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