Abstract: A method for managing audio during a conference includes steering, at a spatial steering processor of a mobile device, first decoded audio to be projected from a speaker at a first angle. The method also includes steering, at the spatial steering processor, second decoded audio to be projected from the speaker at a second angle. The first decoded audio corresponds to a decoded version of a first audio stream from a first device associated with a first participant of the conference, and the second decoded audio corresponds to a decoded version of a second audio stream from a second device associated with a second participant of the conference. The first decoded audio is synchronized with the second decoded audio.

Abstract: A method includes extracting a voicing classification parameter of an audio signal and determining a filter coefficient of a low pass filter based on the voicing classification parameter. The method also includes filtering a low-band portion of the audio signal to generate a low-band audio signal and controlling an amplitude of a temporal envelope of the low-band audio signal based on the filter coefficient. The method also includes modulating a white noise signal based on the amplitude of the temporal envelope to generate a modulated white noise signal and scaling the modulated white noise signal based on a noise gain to generate a scaled modulated white noise signal. The method also includes mixing a scaled version of the low-band audio signal with the scaled modulated white noise signal to generate a high-band excitation signal that is used to generate a decoded version of the audio signal.

Abstract: A method for managing audio during a conference includes steering, at a spatial steering processor of a mobile device, first decoded audio to be projected from a speaker at a first angle. The method also includes steering, at the spatial steering processor, second decoded audio to be projected from the speaker at a second angle. The first decoded audio corresponds to a decoded version of a first audio stream from a first device associated with a first participant of the conference, and the second decoded audio corresponds to a decoded version of a second audio stream from a second device associated with a second participant of the conference. The first decoded audio is synchronized with the second decoded audio.

Abstract: A method for managing audio during a conference includes receiving, at a first buffer of a mobile device, a first audio stream from a first device associated with a first participant of the conference. The method also includes receiving, at a second buffer of the mobile device, a second audio stream from a second device associated with a second participant of the conference. The method further includes generating a control signal at a delay controller of the mobile device. The control signal is provided to the first buffer and to the second buffer to synchronize first buffered audio that is output from the first buffer with second buffered audio that is output from the second buffer.

Abstract: A method includes extracting a voicing classification parameter of an audio signal and determining a filter coefficient of a low pass filter based on the voicing classification parameter. The method also includes filtering a low-band portion of the audio signal to generate a low-band audio signal and controlling an amplitude of a temporal envelope of the low-band audio signal based on the filter coefficient. The method also includes modulating a white noise signal based on the amplitude of the temporal envelope to generate a modulated white noise signal and scaling the modulated white noise signal based on a noise gain to generate a scaled modulated white noise signal. The method also includes mixing a scaled version of the low-band audio signal with the scaled modulated white noise signal to generate a high-band excitation signal that is used to generate a decoded version of the audio signal.

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 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 particular method includes determining, at a device, a voicing classification of an input signal. The input signal corresponds to an audio signal. The method also includes controlling an amount of an envelope of a representation of the input signal based on the voicing classification. The method further includes modulating a white noise signal based on the controlled amount of the envelope. The method also includes generating a high band excitation signal based on the modulated white noise signal.

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: Systems and methods of performing blind bandwidth extension are disclosed. In an embodiment, a method includes determining, based on a set of low-band parameters of an audio signal, a first set of high-band parameters and a second set of high-band parameters. The method further includes generating a predicted set of high-band parameters based on a weighted combination of the first set of high-band parameters and the second set of high-band parameters.

Abstract: A method for managing audio during a conference includes receiving, at a first buffer of a mobile device, a first audio stream from a first device associated with a first participant of the conference. The method also includes receiving, at a second buffer of the mobile device, a second audio stream from a second device associated with a second participant of the conference. The method further includes generating a control signal at a delay controller of the mobile device. The control signal is provided to the first buffer and to the second buffer to synchronize first buffered audio that is output from the first buffer with second buffered audio that is output from the second buffer.

Abstract: A device includes a decoder that includes an extractor, a predictor, a selector, and a switch. The extractor is configured to extract a first plurality of parameters from a received input signal. The input signal corresponds to an encoded audio signal. The predictor is configured to perform blind bandwidth extension by generating a second plurality of parameters independent of high band information in the input signal. The second plurality of parameters corresponds to a high band portion of the encoded audio signal. The selector is configured to select a particular mode from multiple high band modes including a first mode using the first plurality of parameters and a second mode using the second plurality of parameters. The switch is configured to output the first plurality of parameters or the second plurality of parameters based on the selected particular mode.

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 particular method includes determining, at a device, a voicing classification of an input signal. The input signal corresponds to an audio signal. The method also includes controlling an amount of an envelope of a representation of the input signal based on the voicing classification. The method further includes modulating a white noise signal based on the controlled amount of the envelope. The method also includes generating a high band excitation signal based on the modulated white noise signal.

Abstract: Systems and methods of performing blind bandwidth extension are disclosed. In an embodiment, a method includes determining, based on a set of low-band parameters of an audio signal, a first set of high-band parameters and a second set of high-band parameters. The method further includes generating a predicted set of high-band parameters based on a weighted combination of the first set of high-band parameters and the second set of high-band parameters.

Abstract: Systems and methods of performing blind bandwidth extension are disclosed. In an embodiment, a method includes receiving, at a decoder of a speech vocoder, a set of low-band parameters as part of a narrowband bitstream. The set of low-band parameters are received from an encoder of the speech vocoder. The method also includes predicting a set of high-band parameters based on the set of low-band parameters.

Abstract: A device includes a decoder that includes an extractor, a predictor, a selector, and a switch. The extractor is configured to extract a first plurality of parameters from a received input signal. The input signal corresponds to an encoded audio signal. The predictor is configured to perform blind bandwidth extension by generating a second plurality of parameters independent of high band information in the input signal. The second plurality of parameters corresponds to a high band portion of the encoded audio signal. The selector is configured to select a particular mode from multiple high band modes including a first mode using the first plurality of parameters and a second mode using the second plurality of parameters. The switch is configured to output the first plurality of parameters or the second plurality of parameters based on the selected particular mode.

Abstract: Compressibility-based reallocation of initial bit allocations for frames of an audio signal is described. Applications to redundancy-based retransmission of critical frames (e.g., for fixed-bit-rate modes of speech codec operation) are also described.

Abstract: A method of audio signal processing is described. The method includes calculating a criticality measure based on information about a first frame of the audio signal. The method also includes calculating a threshold value based on information relating to a state of a transmission channel. The method further includes comparing the calculated criticality measure to the calculated threshold value. The method additionally includes deciding to transmit a redundant copy of the first frame based on a result of the comparison. The method also includes transmitting the redundant copy of the first frame with a second frame. The first frame precedes the second frame in the audio signal.

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.