Abstract: A multi-party control unit (MCU) generates, based on audio data streams that represent sounds associated terminal devices, a mixed audio data stream. In addition, the MCU modifies the mixed mono audio data to steganographically embed sub-streams that include representations of the mono audio data streams. A terminal device receives the modified mixed audio data stream. When the terminal device is configured for stereo playback, the terminal device performs an inverse steganographic process to extract, from the mixed audio data stream, the sub-streams. The terminal device generates and outputs multi-channel audio data based on the extracted sub-streams and the mixed audio data stream. When the terminal device is not configured for stereo playback, the terminal device outputs sound based on the mixed audio data stream without extracting the embedded sub-streams.

Abstract: A method for applying error resiliency based on speed is disclosed. A speed of the wireless communication device is detected. Source-based error resiliency is dynamically applied to a signal based on the speed. The signal is sent on an uplink. In addition to the speed, the source-based error resiliency may be dynamically applied to the signal based on a characteristic of the signal.

Abstract: A method for quantizing prediction memory during fast recovery coding is disclosed. A best shape vector that describes prediction memory for a current frame is quantized. It is determined whether to send the quantized best shape vector. The quantized best shape vector is sent based on the determination. An encoded current frame is sent.

Abstract: Methods of audio coding are described in which an excitation signal for a first frequency band of the audio signal is used to calculate an excitation signal for a second frequency band of the audio signal that is separated from the first frequency band.

Abstract: A method for determining an upperband speech signal from a narrowband speech signal is disclosed. A list of narrowband line spectral frequencies (LSFs) is determined from the narrowband speech signal. A first pair of adjacent narrowband LSFs that have a lower difference between them than every other pair of adjacent narrowband LSFs in the list is determined. A first feature that is a mean of the first pair of adjacent narrowband LSFs is determined. Upperband LSFs are determined based on at least the first feature using codebook mapping.

Abstract: A method for reconstructing an erased speech frame is described. A second speech frame is received from a buffer. The index position of the second speech frame is greater than the index position of the erased speech frame. The type of packet loss concealment (PLC) method to use is determined based on one or both of the second speech frame and a third speech frame. The index position of the third speech frame is less than the index position of the erased speech frame. The erased speech frame is reconstructed from one or both of the second speech frame and the third speech frame.

Abstract: A method for preventing the loss of information within a speech frame is described. A first speech frame to be encoded is selected. A determination is made as to whether or not a second speech frame is a critical speech frame based on the information within the second speech frame and one or more adjacent speech frames. At least a part of an encoded version of the second speech frame is created according to a selected forward error correction (FEC) mode if the second speech frame is a critical speech frame. The first speech frame and the at least a part of the encoded version of the second speech frame are transmitted.

Abstract: A method for decoding a signal on an electronic device is described. The method includes receiving a signal. The method also includes extracting a bitstream from the signal. The method further includes performing watermark error checking on the bitstream for multiple frames. The method additionally includes determining whether watermark data is detected based on the watermark error checking. The method also includes decoding the bitstream to obtain a decoded second signal if the watermark data is not detected.

Abstract: An electronic device configured for adaptively encoding a watermarked signal is described. The electronic device includes modeler circuitry that determines watermark data based on a first signal. The electronic device also includes coder circuitry coupled to the modeler circuitry. The coder circuitry determines a low priority portion of a second signal and embeds the watermark data into the low priority portion of the second signal to produce a watermarked second signal.

Abstract: An electronic device configured for encoding a watermarked signal is described. The electronic device includes modeler circuitry. The modeler circuitry determines parameters based on a first signal and a first-pass coded signal. The electronic device also includes coder circuitry coupled to the modeler circuitry. The coder circuitry performs a first-pass coding on a second signal to obtain the first-pass coded signal and performs a second-pass coding based on the parameters to obtain a watermarked signal.

Abstract: A method for quantizing prediction memory during fast recovery coding is disclosed. A best shape vector that describes prediction memory for a current frame is quantized. It is determined whether to send the quantized best shape vector. The quantized best shape vector is sent based on the determination. An encoded current frame is sent.

Abstract: Methods of audio coding are described in which an excitation signal for a first frequency band of the audio signal is used to calculate an excitation signal for a second frequency band of the audio signal that is separated from the first frequency band.

Abstract: A method for determining an upperband speech signal from a narrowband speech signal is disclosed. A list of narrowband line spectral frequencies (LSFs) is determined from the narrowband speech signal. A first pair of adjacent narrowband LSFs that have a lower difference between them than every other pair of adjacent narrowband LSFs in the list is determined. A first feature that is a mean of the first pair of adjacent narrowband LSFs is determined. Upperband LSFs are determined based on at least the first feature using codebook mapping.

Abstract: A method for preventing the loss of information within a speech frame is described. A first speech frame to be encoded is selected. A determination is made as to whether or not a second speech frame is a critical speech frame based on the information within the second speech frame and one or more adjacent speech frames. At least a part of an encoded version of the second speech frame is created according to a selected forward error correction (FEC) mode if the second speech frame is a critical speech frame. The first speech frame and the at least a part of the encoded version of the second speech frame are transmitted.

Abstract: A method for reconstructing an erased speech frame is described. A second speech frame is received from a buffer. The index position of the second speech frame is greater than the index position of the erased speech frame. The type of packet loss concealment (PLC) method to use is determined based on one or both of the second speech frame and a third speech frame. The index position of the third speech frame is less than the index position of the erased speech frame. The erased speech frame is reconstructed from one or both of the second speech frame and the third speech frame.

Abstract: A speech converter in a speech processing system modifies various aspects of input speech. The speech converter receives a formants signal representing an input speech signal. The speech converter may also receive a formant scaling command or a user selection of one of multiple control signals, each specifying a manner of modifying one or more of the received signals (i.e., formants, voicing, pitch, gain). The speech converter modifies at least one of the formants, voicing, pitch, and/or gain signals as specified by the selected voice font.

Abstract: Techniques for training a call center operator trainee are described. A database contains queries including caller speech providing information to be used in routing a call. Each query is associated with an operator response as well as numerical values produced by analyzing the query and the operator response. A training module presents a query to a trainee and receives the trainee response. The training module compares the trainee response to one or more of the numerical values in the database to determine if the trainee response is consistent with the correct destination for the call. If the trainee response is not consistent with the correct destination, the training module provides guidance to assist the trainee in associating the query with the correct destination.

Abstract: Techniques for training a call center operator trainee are described. A database contains queries including caller speech providing information to be used in routing a call. Each query is associated with an operator response as well as numerical values produced by analyzing the query and the operator response. A training module presents a query to a trainee and receives the trainee response. The training module compares the trainee response to one or more of the numerical values in the database to determine if the trainee response is consistent with the correct destination for the call. If the trainee response is not consistent with the correct destination, the training module provides guidance to assist the trainee in associating the query with the correct destination.