Abstract: Techniques are described for performing adaptive noise suppression to improve handling of both speech signals and music signals at least up to super wideband (SWB) bandwidths. The techniques include identifying a context or environment in which audio data is captured, and adaptively changing a level of noise suppression applied to the audio data prior to bandwidth compressing (e.g., encoding) based on the context. For a valid speech context, an audio pre-processor may set a first level of noise suppression that is relatively aggressive in order to suppress noise (including music) in the speech signals. For a valid music context, the audio pre-processor may set a second level of noise suppression that is less aggressive in order to leave the music signals undistorted. In this way, a vocoder at a transmitter side wireless communication device may properly encode both speech and music signals with minimal distortions.

Abstract: A device includes a decoder configured to receive an encoded audio signal at a decoder and to generate a synthesized signal based on the encoded audio signal. The device further includes a classifier configured to classify the synthesized signal based on at least one parameter determined from the encoded audio signal.

Abstract: The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold.

Abstract: A method includes determining an error condition during a bandwidth transition period of an encoded audio signal. The error condition corresponds to a second frame of the encoded audio signal, where the second frame sequentially follows a first frame in the encoded audio signal. The method also includes generating audio data corresponding to a first frequency band of the second frame based on audio data corresponding to the first frequency band of the first frame. The method further includes re-using a signal corresponding to a second frequency band of the first frame to synthesize audio data corresponding to the second frequency band of the second frame.

Abstract: A method includes separating, at a device, an input audio signal into at least a low-band signal and a high-band signal. The low-band signal corresponds to a low-band frequency range and the high-band signal corresponds to a high-band frequency range. The method also includes selecting a non-linear processing function of a plurality of non-linear processing functions. The method further includes generating a first extended signal based on the low-band signal and the non-linear processing function. The method also includes generating at least one adjustment parameter based on the first extended signal, the high-band signal, or both.

Abstract: A method includes selecting, at a device, a first seed generation scheme or a second seed generation scheme based on determining whether audio data satisfies a criterion. The audio data corresponds to a first audio frame of a sequence of frames. The first seed generation scheme includes generating a first seed value based on one or more parameters corresponding to the first audio frame (e.g., the bit-stream indices). The second seed generation scheme includes generating a second seed value based on a seed output value associated with a second audio frame of the sequence of frames. A seed value generated by the selected seed generation scheme is provided to a random noise generator.

Abstract: A device includes a de-jitter buffer configured to receive a packet, the packet including first data and second data. The first data includes a partial copy of first frame data corresponding to a first frame of a sequence of frames. The second data corresponds to a second frame of the sequence of frames. The device also includes an analyzer configured to, in response to receiving the packet, generate a first frame receive timestamp associated with the first data. The analyzer is also configured to, in response to receiving the packet, generate a second frame receive timestamp associated with the second data. The first frame receive timestamp indicates a first time that is earlier than a second time indicated by the second frame receive timestamp.

Abstract: A device includes a first classifier and a second classifier coupled to the first classifier. The first classifier is configured to output first decision data that indicates a classification of an audio frame as a speech frame or a non-speech frame, the first decision data determined based on first probability data associated with a first likelihood of the audio frame being the speech frame and based on second probability data associated with a second likelihood of the audio frame being the non-speech frame. The second classifier is configured to output second decision data based on the first probability data, the second probability data, and the first decision data, the second decision data includes an indication of a selection of a particular encoder of multiple encoders available to encode the audio frame.

Abstract: A method for determining an interpolation factor set by an electronic device is described. The method includes determining a value based on a current frame property and a previous frame property. The method also includes determining whether the value is outside of a range. The method further includes determining an interpolation factor set based on the value and a prediction mode indicator if the value is outside of the range. The method additionally includes synthesizing a speech signal.

Abstract: A method for controlling an average encoding rate by an electronic device is described. The method includes obtaining a speech signal. The method also includes determining a first average rate. The method further includes determining a first threshold based on the first average rate. The method additionally includes controlling the average encoding rate by determining at least one other threshold based on the first threshold. The method also includes sending an encoded speech signal.

Abstract: A device includes a receiver, a buffer, and an analyzer. The receiver is configured to receive packets that correspond to at least a subset of a sequence of packets and that include error correction data. The error correction data of a first packet of the packets includes a partial copy of a second packet. The buffer is configured to store the packets. The analyzer is configured to determine whether a first particular packet of the sequence is missing from the buffer, to determine whether a partial copy of the first particular packet is stored in the buffer as error correction data in a second particular packet, to update a value based at least in part on whether the first particular packet is missing from the buffer and the partial copy of the first particular packet is stored in the buffer, and to adjust an error recovery parameter based at least in part on the value.

Abstract: A device includes a receiver, a buffer, a transmitter, and an analyzer. The receiver is configured to receive a plurality of packets that corresponds to at least a subset of a sequence of packets. Error correction data of a first packet of the plurality of packets includes a partial copy of a second packet of the plurality of packets. The analyzer is configured to determine whether a particular packet of the sequence is missing from the buffer, and to determine whether a partial copy of the particular packet is stored in the buffer. The analyzer is also configured to send, via the transmitter, a retransmit message to a second device based at least in part on determining that the buffer does not store the particular packet and that the buffer does not store the partial copy of the particular packet.

Abstract: A method includes determining, at a decoder of a first device, an offset value corresponding to an offset between a first particular packet and a second particular packet. The first device includes a de-jitter buffer. The method also includes transmitting the offset value to an encoder of a second device to enable the second device to send packets to the first device based on the offset value.

Abstract: A method for quantizing phase information on an electronic device is described. The method includes obtaining a speech signal. The method also includes determining a prototype pitch period signal based on the speech signal and transforming the prototype pitch period signal into a first frequency-domain signal. The method additionally includes mapping the first frequency-domain signal into a plurality of subbands. The method also includes determining a global alignment based on the first frequency-domain signal and quantizing the global alignment utilizing scalar quantization to obtain a quantized global alignment. The method additionally includes determining a plurality of band alignments corresponding to the plurality of subbands. The method also includes quantizing the plurality of band alignments utilizing vector quantization to obtain a quantized plurality of band alignments. The method further includes transmitting the quantized global alignment and the quantized plurality of band alignments.

Abstract: The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold.

Abstract: The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold.

Abstract: A method for determining pitch pulse period signal boundaries by an electronic device is described. The method includes obtaining a signal. The method also includes determining a first averaged curve based on the signal. The method further includes determining at least one first averaged curve peak position based on the first averaged curve and a threshold. The method additionally includes determining pitch pulse period signal boundaries based on the at least one first averaged curve peak position. The method also includes synthesizing a speech signal.

Abstract: A method includes separating, at a device, an input audio signal into at least a low-band signal and a high-band signal. The low-band signal corresponds to a low-band frequency range and the high-band signal corresponds to a high-band frequency range. The method also includes selecting a non-linear processing function of a plurality of non-linear processing functions. The method further includes generating a first extended signal based on the low-band signal and the non-linear processing function. The method also includes generating at least one adjustment parameter based on the first extended signal, the high-band signal, or both.

Abstract: A method for mitigating potential frame instability by an electronic device is described. The method includes obtaining a frame subsequent in time to an erased frame. The method also includes determining whether the frame is potentially unstable. The method further includes applying a substitute weighting value to generate a stable frame parameter if the frame is potentially unstable.

Abstract: A method for determining pitch pulse period signal boundaries by an electronic device is described. The method includes obtaining a signal. The method also includes determining a first averaged curve based on the signal. The method further includes determining at least one first averaged curve peak position based on the first averaged curve and a threshold. The method additionally includes determining pitch pulse period signal boundaries based on the at least one first averaged curve peak position. The method also includes synthesizing a speech signal.