Abstract: Methods and apparatus are provided for an echo cancellation system. The echo cancellation system comprises an automatic gain control (AGC), a first scalar, a second scalar, a signal summing stage, and an adaptive filter. The AGC is responsive to a first signal in a reference path and a second signal in a near end path. The signal summing stage is between the first and second scalar in the near end path and the first and second scalars are responsive to the AGC. The adaptive filter is responsive to the first signal and provides a third signal to the signal summing stage that corresponds to an echo signal and is subtracted from a signal in the near end path. The second scalar has a scale rate that is the inverse of the scale rate of the first scalar such that unity scaling occurs on the near end path.

Abstract: An echo canceling system receives and transmits audio signals between a far end and a near end. During single talk, which is when only one end is originating audio, the path back to the originator is impeded by echo cancellation and attenuation. When there is double talk, which is when both ends are originating audio, the attenuation is removed, or at least significantly reduced. This is achieved by using ERLE, which itself is a known signal used for other purposes in an echo cancellation system, to provide information as to when double talking is occurring. This allows for stopping the attenuation for the double talk situation, which is the desired result.

Abstract: A bi-directional hands-free communication device includes a microphone for transmitting a signal along a transmit path and a speaker receiving a signal transmitted along a receive path and outputting a corresponding output signal. An echo canceller, positioned in the transmit path and the receive path, cancels echo signals induced by the microphone from the speaker and outputs a corresponding cancelled signal along the transmit path, and a transparency circuit distributes state-dependent additional loss derived from the noise floor margin to the transmit path and the receive path to reduce residual echo signals output from the echo canceller.

Abstract: A digital satellite communication system including a local terminal coupled to a radio wireline interface through a radio satellite network. The radio wireline interface connects the radio satellite network to a public switched telephone network and a remote communication terminal. The system includes a novel method of establishing an end-to-end communication channel between local and remote terminals wherein the local terminal establishes a direct digital channel between itself and the radio wireline interface and transmits a message describing its signaling capabilities to the radio wireline interface. The radio wireline interface then trains its modem with the modem of the remote terminal such that the signaling capabilities of the local terminal are not violated. By moving the modem training procedure to the radio wireline interface, modem training response delays caused by the radio satellite network do not affect the success of establishing the end-to-end communication channel.

Abstract: An apparatus comprises a first and a second input and a processor coupled to the inputs. The processor estimates peak and minimum levels of each of the first and second input signals. A first signal regulator is coupled to the first input and to a first output for delivering a first output signal from the first signals. A second signal regulator is coupled to the second input and to a second output for delivering a second output signal from the second audio signals. A gain adjustment device is coupled to the processor and the first and second signal regulators. The gain adjustment device provides control signals to the first and second signal regulators to adjust an output signal level of the first output signal in response to minimum and peak levels of the first and second input signals. The output signal level is continuously variable over a range.

Abstract: A new method for Code Excited Linear Predictive (CELP) coding of speech reduces the computational complexity by removing a convolution operation from a recursive loop used to poll the adaptive code book vectors. In a preferred embodiment, an impulse function of a short term perceptually weighted filter is first convolved with perceptual weighted target speech and the result cross-correlated with each vector in the codebook to produce an error function. The vector having the minimum error function is chosen to represent the particular speech frame being examined.

Abstract: A new way of determining correlation coefficients for stochastic codebook vectors for CELP coding of speech takes advantage of the sparsely populated nature of stochastic codebook vectors. N valued input signals (e.g., convolution vectors) to be correlated with N valued codebook vectors are fed to an N by N multiplexer or other selection means and the signal values either passed to an accumulator or not according to the state of N select inputs or other identification means determined from a memory store (e.g., an EPROM) whose entries correspond to the non-zero values of the codebook vectors. The accumulator output is the correlation of the codebook vector with the input signal. A sequencer steps through the entire codebook to provide correlation values for each vectors. The results are used to determine the optimum stochastic codebook vector for replicating the particular speech frame being analyzed.

Abstract: A new way of determining autocorrelation coefficients for adaptive codebook vectors for CELP coding of speech simplifies and improves the accuracy of the autocorrelation coefficient determination for the situation where the codebook vector length being analyzed is less than a speech frame length. This is important in synthesizing short pitch period speech. Copy-up of the shortened codebook vector to equal the frame length is not needed and autocorrelation coefficient errors associated with copy-up are avoided. The improved system relies on calculating autocorrelation coefficients of the first (shortest) vector and then obtaining subsequent autocorrelation coefficients for successive vectors of increasing length by a simple end correction technique until the vector length equals the frame length. The autocorrelation coefficients are scaled by multiplying them by the ratio of the frame length to the vector length.

Abstract: A new way of CELP coding speech simplifies the recursive loop used to poll code adaptive book vectors by reducing the number of autocorrelation operations that must be performed with the K vectors of the codebook each having N entries. Autocorrelation is initially performed for only a small number P<<N autocorrelation coefficients in each codebook vector and the values found are used to scan through all the codebook vectors looking for those S vectors (S<K) which give the best match to the input speech. The autocorrelation function for the S vectors is then recalculated for R entries (P<R.ltoreq.N) in the codebook vectors to determine which of the S codebook vectors and associated gain gives the best match to the input speech.