Abstract: A method includes determining a first modeled high-band signal based on a low-band excitation signal of an audio signal, where the audio signal includes a high-band portion and a low-band portion. The method also includes determining scaling factors based on energy of sub-frames of the first modeled high-band signal and energy of corresponding sub-frames of the high-band portion of the audio signal. The method includes applying the scaling factors to a modeled high-band excitation signal to determine a scaled high-band excitation signal and determining a second modeled high-band signal based on the scaled high-band excitation signal. The method includes determining gain parameters based on the second modeled high-band signal and the high-band portion of the audio signal.

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: Speech encoders and methods of speech encoding are disclosed that encode inactive frames at different rates. Apparatus and methods for processing an encoded speech signal are disclosed that calculate a decoded frame based on a description of a spectral envelope over a first frequency band and the description of a spectral envelope over a second frequency band, in which the description for the first frequency band is based on information from a corresponding encoded frame and the description for the second frequency band is based on information from at least one preceding encoded frame. Calculation of the decoded frame may also be based on a description of temporal information for the second frequency band that is based on information from at least one preceding encoded frame.

Abstract: A method for redundant frame coding by an electronic device is described. The method includes determining an adaptive codebook energy and a fixed codebook energy based on a frame. The method also includes coding a redundant version of the frame based on the adaptive codebook energy and the fixed codebook energy. The method further includes sending a subsequent frame.

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 of bit allocation is described. The method includes, for each among a plurality of vectors, calculating a corresponding one of a plurality of gain factors. The method also includes, for each among the plurality of vectors, calculating a corresponding bit allocation that is based on the gain factor. The method further includes, for at least one among the plurality of vectors, determining that the corresponding bit allocation is not greater than a minimum allocation value. The method additionally includes, in response to the determining, for each of the at least one vector, changing the corresponding bit allocation.

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 of processing an audio signal is described. The method includes selecting one among a plurality of entries of a codebook based on information from the audio signal. The method also includes determining locations, in a frequency domain, of zero-valued elements of a first signal that is based on the selected codebook entry. The method further includes calculating energy of the audio signal at the determined frequency-domain locations. The method additionally includes calculating a value of a measure of a distribution of the energy of the audio signal among the determined frequency-domain locations. The method also includes calculating a noise injection gain factor based on the calculated energy and the calculated value.

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 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: 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: A method includes receiving a second audio frame at a decoder. The second audio frame follows a first audio frame in an audio signal and includes a first number of bits allocated to primary coding information associated with the second audio frame, a second number of bits allocated to redundant coding information associated with the first audio frame, and an indicator of a frame type of the first audio frame. In response to a frame erasure condition associated with the first audio frame, the second number of bits is determined based on the indicator and used to decode the first audio frame. In clean channel conditions, the first audio frame is received and decoded based on primary coding bits in the first audio frame, and the first number of bits is determined based on the indicator and used to decode the second audio frame.

Abstract: A method includes determining a first modeled high-band signal based on a low-band excitation signal of an audio signal, where the audio signal includes a high-band portion and a low-band portion. The method also includes determining scaling factors based on energy of sub-frames of the first modeled high-band signal and energy of corresponding sub-frames of the high-band portion of the audio signal. The method includes applying the scaling factors to a modeled high-band excitation signal to determine a scaled high-band excitation signal and determining a second modeled high-band signal based on the scaled high-band excitation signal. The method includes determining gain parameters based on the second modeled high-band signal and the high-band portion of the audio signal.