摘要:
A video server or other processing device obtains availability information for a wireless network and modifies a manner in which video segments of a video service are delivered to a user device over the wireless network based on the obtained availability information. The availability information may comprise at least one of network congestion measurement information and transmission pricing information. In an illustrative embodiment, the processing device comprises a video server configured to utilize the availability information to generate a list of video segments available for transmission for at least a portion of the video service, to provide the list of video segments to the user device responsive to a request received from the user device, to receive an additional request from the user device for at least one particular video segment identified in the list, and to provide the requested at least one video segment to the user device.
摘要:
A video server or other processing device obtains availability information for a wireless network and modifies a manner in which video segments of a video service are delivered to a user device over the wireless network based on the obtained availability information. The availability information may comprise at least one of network congestion measurement information and transmission pricing information. In an illustrative embodiment, the processing device comprises a video server configured to utilize the availability information to generate a list of video segments available for transmission for at least a portion of the video service, to provide the list of video segments to the user device responsive to a request received from the user device, to receive an additional request from the user device for at least one particular video segment identified in the list, and to provide the requested at least one video segment to the user device.
摘要:
Embodiments relate to an apparatus and method for encoding/decoding data for motion detection in a communication system. The method for encoding data includes receiving, by an encoder, video data including a plurality of frames. Each frame is represented by a pixel vector including a number of pixel values. The method further includes generating, by the encoder, sets of measurements representing the plurality of frames. Each set of measurements represents a different frame of the plurality of frames. The generating step generates the sets of measurements by applying sensing matrices to the pixel vectors, and a same sensing matrix is used for at least two sets of measurements.
摘要:
A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The decoder buffer level limits are specified in terms of a maximum number of encoded frames (or duration). The transmitter can predict the number of encoded frames, Fpred, in the decoder buffer and transmit the value, Fpred, to the receiver with the audio data. If the transmitter determines that the decoder buffer level is becoming too high, the frames being generated by the encoder are too small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming too low, the frames being generated by the encoder are too big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver. The receiver fills the decoder buffer until Fpred frames are received before commencing decoding frames when the decoder first starts up or possibly when a new audio program is selected. The transmitter and receiver clocks may be synchronized by adjusting the clock at the receiver by using a feedback loop that compares the actual level of the decoder buffer to the predicted value, Fpred, received from the transmitter (a higher number of encoded frames in the buffer indicates that the clock of the receiver is too slow and should be increased, and a lower number of encoded frames in the buffer indicates that the clock of the receiver is too fast and needs to be slowed down).
摘要:
A side tone generator which adjusts gain to compensate for echo effects is described. The side tone generator includes a gain unit which receives a voice signal and applies a gain to the voice signal to produce a side tone, and a summing unit which adds the side tone to a speech decoder signal. The side tone generator also includes a side tone gain adapter to adjust the gain added by the gain unit. The side tone gain adapter computes a default gain which would be applied in the absence of echo and multiplies the default gain by an echo correction factor based on the prevailing level of echo in order to determine the gain to be applied. In another embodiment, an echo canceller employs a filter which receives a filter input based on a speech decoder signal and produces a filter output reflecting an estimate of echo present in the speech decoder signal. The echo canceller also includes an output summing unit which subtracts the filter output from a voice signal to produce an echo canceller output. The echo canceller also includes a feedback loop which employs feedback from the filter output to refine the filter input. The echo canceller also includes a filter adapter which produces updated coefficients for the filter based on the echo canceller output, the voice signal and the filter input.
摘要:
A speech recognizer system for use with a telecommunication network wherein an input signal generated onto the network from a first terminal is directed to a speech recognizer for estimating the verbal content of the input signal. The speech recognizer or associated equipment then directs an estimate of the verbal content as an output signal back to the first terminal, the estimate including one or more approximations of the verbal content of the input signal. At the first terminal the user then confirms a correct estimate, or selects from a plurality of approximations, the verbal content of the input signal.
摘要:
A controller in a video headend or other transmission element of a signal distribution system is operative to detect a condition in which unicast transmissions of a given content stream to a plurality of terminals meet a specified threshold. The controller starts a multicast transmission of the given content stream in response to the detected condition, and transitions at least one of the terminals to the multicast transmission. In one embodiment, the controller identifies at least one of the terminals as a terminal that will receive the multicast transmission of the given content stream in place of its unicast transmission prior to one or more of the other terminals receiving the multicast transmission. The controller stops the unicast transmission to the identified terminal if that unicast transmission has already been started, starts the multicast transmission, switches the identified terminal to the multicast transmission, and subsequently transitions one or more of the other terminals to the multicast transmission. The identified terminal may be a leading terminal or a trailing terminal.
摘要:
A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The transmitter predicts the number of encoded frames, Fpred, in the buffer having a limited level and transmits the value, Fpred, to the receiver with the frame. If the transmitter determines that the decoder buffer level is high, the frames being generated by the encoder are small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming low, the frames being generated by the encoder are big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver clock using feedback depending on the compared level of the decoder to the actual level to Fpred.
摘要:
A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The decoder buffer level limits are specified in terms of a maximum number of encoded frames (or duration). The transmitter can predict the number of encoded frames, Fpred, in the decoder buffer and transmit the value, Fpred, to the receiver with the audio data. If the transmitter determines that the decoder buffer level is becoming too high, the frames being generated by the encoder are too small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming too low, the frames being generated by the encoder are too big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver. The receiver fills the decoder buffer until Fpred frames are received before commencing decoding frames when the decoder first starts up or possibly when a new audio program is selected. The transmitter and receiver clocks may be synchronized by adjusting the clock at the receiver by using a feedback loop that compares the actual level of the decoder buffer to the predicted value, Fpred, received from the transmitter (a higher number of encoded frames in the buffer indicates that the clock of the receiver is too slow and should be increased, and a lower number of encoded frames in the buffer indicates that the clock of the receiver is too fast and needs to be slowed down).
摘要:
A wireless telephone with record and playback capability is disclosed. The telephone has an operation module, which transmits near-end signals and receives far-end signals, and a record module which writes transmission packets formed from the near-end signals in a first location of a memory and reception packets formed from the far-end signals in a second location of the memory. A playback module reads the transmission packets and the reception packets from the memory, and decodes the transmission packets into transmission speech samples and the reception packets into reception speech samples using transmission and reception decoders, respectively. Further, the playback module has a mixer to mix the transmission speech samples with the reception speech samples to form mixed speech signals for playback on a speaker. The playback module also includes a voice activity detector which outputs a skip signal to the transmission and reception decoders in response to detection of either voice in the transmission speech samples or silence in the reception speech samples. In response to the skip signal, the transmission and reception decoders discard a current transmission packet and a current receive packet and read a next transmission packet and a next reception packet from the memory.