Abstract: A double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

Abstract: An echo detector includes means (34) for forming a set of distance measures between pitch estimates of a first signal and pitch estimates of a second signal at predetermined delays with respect to the first signal. A selector (36) selects a distance measure from the set corresponding to the highest similarity between the first and second signals. A classifier (32) classifies the second signal as including an echo if the selected distance measure has a predetermined relation to a similarity threshold (TH).

Abstract: An apparatus for reducing a digital filter delay includes means for determining the magnitude response of a desired filter. Means form the real cepstrum of this magnitude response. Means transform the real cepstrum into a complex cepstrum of a corresponding minimum-phase filter having the same magnitude response as the desired filter. A filter applies a smoothly decaying shaping window to the complex cepstrum. Means transform the shaped complex cepstrum into an estimated minimum-phase filter.

Abstract: A method and arrangement in a receiving communication device for compensating for the difference between the clock-frequency controlled sample rate of the receiving device and the sample rate of a sending communication device. The sending device transmits packets comprising M audio samples to be stored in a buffer in the receiving device accommodating at least 2·M samples before play-out. An estimation of the clock skew is continuously updated from a calculated accumulated difference between an expected and an actual point of time of reception of the M audio samples. Before play-out, an adjusted number N of audio samples to be read from the buffer before play-out is calculated using the estimated clock skew. Thereafter, the N audio samples are resampled by interpolation to M audio samples to play-out.

Abstract: A double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

Abstract: A method and arrangement in a receiving communication device for compensating for the difference between the clock-frequency controlled sample rate of the receiving device and the sample rate of a sending communication device. The sending device transmits packets comprising M audio samples to be stored in a buffer in the receiving device accommodating at least 2·M samples before play-out. An estimation of the clock skew is continuously updated from a calculated accumulated difference between an expected and an actual point of time of reception of the M audio samples. Before play-out, an adjusted number N of audio samples to be read from the buffer before play-out is calculated using the estimated clock skew. Thereafter, the N audio samples are resampled by interpolation to M audio samples to play-out.

Abstract: An echo detector includes means (34) for forming a set of distance measures between pitch estimates of a first signal and pitch estimates of a second signal at predetermined delays with respect to the first signal. A selector (36) selects a distance measure from the set corresponding to the highest similarity between the first and second signals.

Abstract: An apparatus for reducing a digital filter delay includes means for determining the magnitude response of a desired filter. Means form the real cepstrum of this magnitude response. Means transform the real cepstrum into a complex cepstrum of a corresponding minimum-phase filter having the same magnitude response as the desired filter. A filter applies a smoothly decaying shaping window to the complex cepstrum. Means transform the shaped complex cepstrum into an estimated minimum-phase filter.

Abstract: A network noise suppressor includes a decoder for partially decoding a CELP coded bit-stream. A noise suppressing filter H(z) is determined from the decoded parameters. The filter is used to determine modified LP and gain parameters. Corresponding parameters in the coded bit-stream are overwritten with the modified parameters.

Abstract: An apparatus for enhancing audibility of a far-end speech signal from a far-end user to a near-end user in a telephony system includes a near-end background noise signal level estimator (26) at the near-end user. A near-end speech signal level estimator (24) is also provided at the near-end user. A gain control logic (22) determines a gain (G) for amplification of the far-end speech signal based on both estimated speech and background noise signal levels.

Abstract: An apparatus for enhancing audibility of a far-end speech signal from a far-end user to a near-end user in a telephony system includes a near-end background noise signal level estimator (26) at the near-end user. A near-end speech signal level estimator (24) is also provided at the near-end user. A gain control logic (22) determines a gain (G) for amplification of the far-end speech signal based on both estimated speech and background noise signal levels.

Abstract: An inter-network line level adjustment system includes a switch (SW1) and a processing unit (24) for identifying calls between a remote telephone network and a local telephone network. Echo cancellers (20) and/or speech coders (22) collect line level data associated with the identified calls. The processing unit (24) determines statistical properties of the collected line level data. Finally, the processing unit (24) instructs the echo cancellers (20) and/or speech coders (22) to adjust, based on the determined statistical properties, line levels of future calls between the remote telephone network and the local telephone network to a desired line level.

Abstract: A pure echo delay estimate of an echo path impulse response estimate represented by echo canceller filter coefficients is calculated. This is done by incrementing (S9) an assumed pure delay m, where m is a positive integer representing an offset of m filter coefficients into the echo path impulse response estimate, until (S8′) a predetermined filter coefficient dependent decision function exceeds a predetermined threshold function, and by representing (S10) the pure delay by the smallest integer mSTART that makes the decision function exceed the threshold function.

Abstract: A network noise suppressor includes means (113) for partially decoding a CELP coded bit-stream. Means (116) determine a noise suppressing filter H(z) from the decoded parameters. Means (118, 120) use this filter to determine modified LP and gain parameters. Means (122) overwrite corresponding parameters in the coded bit-stream with the modified parameters.

Abstract: A novel method and apparatus are disclosed for cancelling echoes originating from a digital mobile handset using an echo canceller located in the network. The method and apparatus thus solve a practical echo problem that many mobile telephone operators have long been faced with. The method includes an algorithm that differs significantly from a conventional network echo canceller algorithm, because of the following basic considerations that influenced a solution to the echo problem: (1) the algorithm should not worsen speech quality for digital mobiles that do not generate echoes; and (2) the echo path is non-linear due to two speech coder/decoder pairs in the echo path and a low echo level.

Abstract: A single two-ended echo canceller is provided that uses an adaptive FIR filter to model the impulse response of the echo path and cancel the echo originating from the PSTN side of the connection, and a non-linear processor controlled by input signal levels to cancel the echo originating from the mobile phone side of the connection. Since the linear processing function is not used for cancelling echoes of the mobile phone side, the algorithms directed to estimating and cancelling echoes on the two sides of the connection can be executed in a single DSP, with all of the attendant advantages.

Abstract: A double talk and echo path change detector is provided for use in an echo canceller that makes a determination about whether a residual signal is dominated by echo or by a "near end" signal. In order to make this determination, a first measure of linear dependency is computed between the residual signal and an echo estimate, and a second measure of linear dependency is computed between the residual signal and a desired signal. The two results are compared with each other, and if they are of about the same order, no further action is needed. However, if the comparison determines that the dependence between the residual signal and the desired signal is much stronger than the dependence between the residual signal and the echo estimate, the present detector assumes that double talk has been detected and a signal denoting that result is output.

Abstract: A digital signal processor that is used for echo cancellation computations is disclosed, which includes parallel multiplier/adders, an input signal sample memory, and a plurality of accumulators for temporarily storing the results of the computations performed. Consequently, not all of the computational results have to be saved in an external data memory, which minimizes data transfer bottlenecks and significantly increases the computational efficiency of the digital signal processor. The adaptive filter coefficient updates can thereby be computed much faster, and the adaptive filter coefficients can be copied much faster from one filter to the other. The computational complexity of the digital signal processor is significantly less than that of prior digital signal processors.

Abstract: A method and apparatus for determining when an echo canceller should be disabled in a telephoning system is disclosed. First, a first power estimate is determined from an input signal. In addition, the input signal is divided into in-phase and quadrature components. The in-phase and quadrature components are then subsampled and used to determine a second power estimate. The first and second power estimates are compared to determine whether a predetermined tone is present. When the predetermined tone is detected, the present invention determines whether a phase reversal in the predetermined tone is present by using the in-phase and quadrature components. When the phase reversal is detected, the echo cancellers are disabled.