摘要:
An apparatus and method for performing speech signal compression, by variable rate coding of frames of digitized speech samples. The level of speech activity for each frame of digitized speech samples is determined and an output data packet rate is selected from a set of rates based upon the determined level of frame speech activity. A lowest rate of the set of rates corresponds to a detected minimum level of speech activity, such as background noise or pauses in speech, while a highest rate corresponds to a detected maximum level of speech activity, such as active vocalization. Each frame is then coded according to a predetermined coding format for the selected rate wherein each rate has a corresponding number of bits representative of the coded frame. A data packet is provided for each coded frame with each output data packet of a bit rate corresponding to the selected rate.
摘要:
An auxiliary data signal is transported with a primary data signal by hiding the auxiliary data signal in the form of colored noise. The colored noise has a spectrum that simulates the spectrum of the primary data signal. By adjusting the gain of individual spread spectrum signal carrier(s) and the power of the colored noise, the auxiliary information stream(s) can be rendered at any desired level below or above an interference threshold in the primary data signal. The power of the colored noise is further compensated to account for a cross-correlation between the primary data signal and the auxiliary data signal to enhance the recovery of the auxiliary data at a decoder.
摘要:
An apparatus and method for performing speech signal compression, by variable rate coding of frames of digitized speech samples. The level of speech activity for each frame of digitized speech samples is determined and an output data packet rate is selected from a set of rates based upon the determined level of frame speech activity. A lowest rate of the set of rates corresponds to a detected minimum level of speech activity, such as background noise or pauses in speech, while a highest rate corresponds to a detected maximum level of speech activity, such as active vocalization. Each frame is then coded according to a predetermined coding format for the selected rate wherein each rate has a corresponding number of bits representative of the coded frame. A data packet is provided for each coded frame with each output data packet of a bit rate corresponding to the selected rate. At the decoder, if a frame is lost due to a channel error, the error is masked by maintaining a fraction of the previous frame's energy and smoothly transitioning to background noise.
摘要:
An apparatus and method for performing speech signal compression, by variable rate coding of frames of digitized speech samples. The level of speech activity for each frame of digitized speech samples is determined and an output data packet rate is selected from a set of rates based upon the determined level of frame speech activity. A lowest rate of the set of rates corresponds to a detected minimum level of speech activity, such as background noise or pauses in speech, while a highest rate corresponds to a detected maximum level of speech activity, such as active vocalization. Each frame is then coded according to a predetermined coding format for the selected rate wherein each rate has a corresponding number of bits representative of the coded frame. A data packet is provided for each coded frame with each output data packet of a bit rate corresponding to the selected rate.
摘要:
A first remote vocoder receives analog voice and produces packetized vocoder data which is transmitted over a wireless link. A first local vocoder receives the packetized vocoder data from the wireless link. The first local vocoder converts the packetized data to a multibit PCM output. The first local vocoder also adds a detection code to one of the least significant bits (LSB) of the PCM output. The first local vocoder passes the PCM signal to the PSTN from the second end user. The first local vocoder also receives PCM input over the PSTN. The first local vocoder constantly monitors the least significant bit of the PCM input for a detection code indicating that a second local vocoder is connected at the receiving end. If the first local vocoder detects the detection code from the second local vocoder, it begins to substitute packetized data and a redundancy check for a second one of the LSB's of the outgoing PCM. The first local vocoder also begins to monitor the second one of the LSB's of the incoming PCM. If the redundancy check indicates that valid packetized data has been received, the first local vocoder stops converting the PCM output into packetized data and simply passes the packetized data on the second one of the LSB's to the first remote vocoder. If at any time the redundancy check fails and the detection code is not detected, the first local vocoder returns to converting the incoming PCM to packetized data. In this way, the tandem vocoding arrangement is avoided.
摘要:
When one vocoding system is coupled to another vocoding system, a tandem arrangement results. The tandem configuration results in voice quality degradation as speech is encoded and decoded, then encoded and decoded again. One reason for the degradation is that postfiltering performed at the output of the speech decoding process introduces distortions in the spectral content of the reconstructed speech as compared to the original speech. The present invention prevents the degradation due to the use of postfilters by modifying the postfiltering within the vocoders where a tandem configuration exists. A detection code is embedded within the data signal to indicate the existence of a tandem configuration. If the detection code is received at a vocoder, modified vocoding is established within the vocoders to prevent the degradation due to the postfiltering.
摘要:
A method and apparatus are described for controlling the data rates for communications to and from a base station and a plurality of remote users. The usage of the communications resource whether the forward link resource (from base station to remote users) or reverse link resource (from remote users to base station) is measured. The measured usage value is compared against at least one predetermined threshold value and the data rates of communications or a subset of communications on said communications resource is modified in accordance with said comparisons.
摘要:
A method and apparatus are described for controlling the data rates for communications to and from a base station and a plurality of remote users. The usage of the communications resource whether the forward link resource (from base station to remote users) or reverse link resource (from remote users to base station) is measured. The measured usage value is compared against at least one predetermined threshold value and the data rates of communications or a subset of communications on said communications resource is modified in accordance with said comparisons.
摘要:
A method and apparatus are described for controlling the data rates for communications to and from a base station and a plurality of remote users. The usage of the communications resource whether the forward link resource (from base station to remote users) or reverse link resource (from remote users to base station) is measured. The measured usage value is compared against at least one predetermined threshold value and the data rates of communications or a subset of communications on said communications resource is modified in accordance with said comparisons.