摘要:
A method that includes; (1) transmitting, at a first transmit time point, a first probe packet over a network connection to a conferencing server immediately before transmitting a data packet, the first probe packet arriving at the conferencing server at a first receive time point; (2) transmitting, at a second transmit time point, a second probe packet over the network connection to the conferencing server immediately after transmitting the data packet, the second probe packet arriving at the conferencing server at a second receive time point, the first and second probe packets being smaller than the data packet; (3) receiving information encoding a first difference between the first and second transmit time points and a second difference between the first and second receive time points; and (4) based on the first and second differences, modifying a transmission parameter associated with data packets to be transmitted thereafter to the conferencing server.
摘要:
According to certain embodiments, training a transcription system includes accessing recorded voice data of a user from one or more sources. The recorded voice data comprises voice samples. A transcript of the recorded voice data is accessed. The transcript comprises text representing one or more words of each voice sample. The transcript and the recorded voice data are provided to a transcription system to generate a voice profile for the user. The voice profile comprises information used to convert a voice sample to corresponding text.
摘要:
In an example embodiment, there is disclosed herein an automatic speech recognition (ASR) system that employs speaker clustering (or speaker type) for transcribing audio. A large corpus of audio with corresponding transcripts is analyzed to determine a plurality of speaker types (e.g., dialects). The ASR system is trained for each speaker type. Upon encountering a new user, the ASR system attempts to map the user to a speaker type. After the new user is mapped to a speaker type, the ASR employs the speaker type for transcribing audio from the new user.
摘要:
In an example embodiment, an example method is provided for echo mitigation in a conference call. In this method, a test audio signal is transmitted to a conference endpoint and as a result, an echo associated with the transmittal of the test audio signal is received. One or more parameters of the echo are then identified and an echo mitigation process is selected from multiple echo mitigation processes based on the identified parameters. The selected echo mitigation process is then applied.
摘要:
A method is provided in one example embodiment that includes monitoring a sound pressure level with an endpoint (e.g., an Internet Protocol (IP) phone), which is configured for communications involving end users; analyzing the sound pressure level to detect a sound anomaly; and communicating the sound anomaly to a sound classification module. The endpoint can be configured to operate in a low-power mode during the monitoring of the sound pressure level. In certain instances, the sound classification module is hosted by the endpoint. In other implementations, the sound classification module is hosted in a cloud network.
摘要:
A method for handling media in a make-before-break handoff between wireless networks is provided that includes communicating with a mobile station to facilitate a session handoff between two call legs of a session. The method also includes identifying a relevant frame of one of the legs, performing a cross correlation operation, generating a delay difference estimate between the two legs, and performing a level difference estimation for the two legs.
摘要:
Methods and apparatus are provided for compressing delay sensitive signals. Frames including multiple samples of the delay sensitive signal are analyzed to determine characteristics associated with the frames, such as the range of quantization levels represented by the samples of the frames. The delay sensitive signal is then compressed by providing an anchor point is sent along with information for determining the variation of each sample from the anchor point.
摘要:
Systems, computer-implemented methods, and non-transitory computer-readable media are provided for performing contact tracing using acoustic communications within or across administrative domains. A computer-implemented method may include obtaining information associated with one or more acoustic tokens from a first user device of a first user where the one or more acoustic tokens were broadcast from an emitter device in an acoustic volume via an audio communication channel, obtaining information associated with one or more acoustic tokens broadcast from the emitter device from a second user device of a second user, determining whether the second user was exposed to a disease associated with the first user in the acoustic volume based on analyzing the information associated with the acoustic tokens from the first user device and the information associated with the acoustic tokens from the second user device, and providing information indicating whether the second user was exposed to the disease.
摘要:
In one embodiment, a signal transmission authentication apparatus includes an input operable to receive a changing signal, a first extractor operable to extract first phoneme data from the received changing signal, a first generator including logic operable to generate first data representative of the extracted first phoneme data, a first output operable to communicate output data corresponding to the received changing signal and the first data from an associated transmitter to an associated receiver, a second extractor associated with the receiver including logic operable to extract second phoneme data from the received output data via the receiver and regenerate the first phoneme data from the received first data, a comparator including logic operable to generate a comparison signal in accordance with a comparison of the first and second phoneme data, and a second output operable to generate a signal corresponding to authenticity of the received changing signal in accordance with an output of the comparator.
摘要:
An echo canceller apparatus comprises a receive side attenuator coupled in a receive side signal path that is configured to couple from a conference call bridge to a caller; a convolution processor coupled to the receive side signal path at a convolution processor pick-off point; a double-talk detector coupled to the receive side signal path and to a sending side signal path that is configured to couple from the caller to the conference call bridge; and logic coupled to the receive side attenuator which when executed is responsive to a double-talk condition detected by the double-talk detector and operable to determine a level of echo canceled by the convolution processor, to determine an additional amount of attenuation to introduce, and to activate the receive side attenuator to introduce the additional attenuation.