At+t Corp.

Lucent Technologies Inc.

Alcatel-lucent USA Inc.

Donald J Youtkus

Jonathan B Allen

Lee Begeja

Carroll W Creswell

Daniel Selig Furman

John C Songrady

John A McMaster

Michael Joseph Haller

Thomas Wasilewski

Semyon B Mizikovsky

Adam L Berenzweig

Ahmed A Tarraf

Robert J Rance

Doh-Suk Kim

Steven Anthony Falco

Binshi Cao

Winston Edward Pekrul

In a method for echo suppression or cancellation, a reference voice packet is selected from a plurality of reference voice packets based on at least one encoded voice parameter associated with each of the plurality of reference voice packets and the targeted voice packet. Echo in the targeted packet is suppressed or cancelled based on the selected reference voice packet.

Packet based echo cancellation and suppression

A speech processor uses at least two speech processing features to enhance the quality of speech signals received by a user during a telephone call. The speech processing features are applied to the speech signals. However, the user only hears speech signals affected by one speech processing feature until both features have fully converged or ramped-up, and the two features are no longer interfering with each other. At that point, a &ldquo;flash-cut&rdquo; of the second speech processing feature is activated. The flash-cut instantaneously switches to speech signals affected by both features. This quick transition makes the speech processing features more noticeable to the user, and the user is not subjected to the period where the features interfere. Further, an optional audio indicator is generated before implementing the flash-cut, so the user is alerted to the flash-cut, and the speech processing features are even more noticeable.

Flash-cut of speech processing features in a telephone call

Disclosed is an encrypting speech processor architecture that provides enhanced security without the use of external cryptosync and with minimal speech degradation. This is accomplished by incorporating a block encryptor and a keystream generator (instead of a fixed secret mask encryption scheme, such as those implemented in voice ciphers) for encrypting blocks of encoded speech bits at a transmitter side. The block encryptor employs an invertible cryptographic algorithm and internal cryptosync to convert a first block of encoded speech bits into a first ciphertext block. The first ciphertext block is used to generate a keystream, which is then used to encrypt a second ciphertext block.

Encrypting speech coder

A speech processor uses at least two speech processing features to enhance the quality of speech signals received by a user during a telephone call. The speech processing features are applied to the speech signals. However, the user only hears speech signals affected by one speech processing feature until both features have fully converged or ramped-up, and the two features are no longer interfering with each other. At that point, a 'flash-cut' of the second speech processing feature is activated. The flash-cut instantaneously switches to speech signals affected by both features. This quick transition makes the speech processing features more noticeable to the user, and the user is not subjected to the period where the features interfere. Further, an optional audio indicator is generated before implementing the flash-cut, so the user is alerted to the flash-cut, and the speech processing features are even more noticeable.

A speech processor uses a plurality of speech processing features to enhance the quality of speech signals received by a user during a telephone call. The activation of a second speech processing feature is delayed for a time duration measured after the initiation of a particular call or measured after the activation of a first speech processing feature. This delay eliminates any interference between the second speech processing feature and the first speech processing feature that is active when the call is initiated. The delay also increases the time it takes for the speech processing features to become fully effective, thus making the enhancement in the quality of the speech signals more noticeable to the user. Further, an optionally generated audio indicator is provided when the second speech processing feature is applied so that the additional feature is even more noticeable to the user.

Method and apparatus for applying multiple speech processing features to

An automated method for modifying a speech signal communicated over a telephone network by applying a gain factor in a telephone set before any side tone is added thereto. The gain factor is a function of a received signal indicative of the background noise at the location of the telephone set. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands. The individual modified subband signals may then be combined to form the resultant modified speech signal.

Background noise compensation in a telephone set

An automated method for modifying a speech signal in a telephone network by applying a gain factor which is a function of the level of background noise at a given destination, and transmitting the modified speech signal to the destination. The gain applied may be a function of both the background noise level and the original speech signal. Either a linear or a non-linear (e.g., compressed) amplification of the original speech signal may be performed, where a compressed amplification results in the higher level portions of the speech signal being amplified by a smaller gain factor than lower level portions. The speech signal may be separated into a plurality of subbands, each resultant subband signal being individually modified in accordance with the present invention. In this case, each subband speech signal is amplified by a gain factor based on a corresponding subband noise signal, generated by separating the background noise signal into a corresponding plurality of subbands. The individual modified subband signals may then be combined to form the resultant modified speech signal.

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Scotch Plains, NJ, United States of America

Donald J Youtkus