Abstract: A method of coding for multi-channel audio signals includes estimating comparison values at an encoder indicative of an amount of temporal mismatch between a reference channel and a corresponding target channel. The method includes smoothing the comparison values to generate short-term and first long-term smoothed comparison values. The method includes calculating a cross-correlation value between the comparison values and the short-term smoothed comparison values. The method also includes adjusting the first long-term smoothed comparison values in response to comparing the cross-correlation value with a threshold. The method further includes estimating a tentative shift value and non-causally shifting the target channel by a non-causal shift value to generate an adjusted target channel. The non-causal shift value is based on the tentative shift value. The method further includes generating, based on reference channel and the adjusted target channel, at least one of a mid-band channel or a side-band channel.

Abstract: A method for multi-channel audio or speech signal processing includes receiving a reference channel and a target channel, determining a variation between a first mismatch value and a second mismatch value, and comparing the variation with a first threshold that may have a pre-determined value or may be adjusted based on a frame type or a smoothing factor. The method also includes adjusting a set of target samples of the target channel based on the variation and based on the comparison to generate an adjusted set of target samples. Adjusting the set of target samples includes selecting one among a first interpolation and a second interpolation based on the variation. The method further includes generating at least one encoded channel based on a set of reference samples and the adjusted set of target samples. The method also includes transmitting the at least one encoded channel to a second device.

Abstract: A device includes a receiver configured to receive an encoded bitstream from a second device. The encoded bitstream includes a temporal mismatch value determined based on a reference channel captured at the second device and a target channel captured at the second device. The device also includes a decoder configured to decode the encoded bitstream to generate a first frequency-domain output signal and a second frequency-domain output signal. The decoder is configured to perform inverse transform operations on the frequency-domain output signals to generate a first and second time-domain signals. Based on the temporal mismatch value, the decoder is configured to map the time-domain signals to a decoded target channel and a decoded reference channel. The decoder is also configured to perform a causal time-domain shift operation on the decoded target channel based on the temporal mismatch value to generate an adjusted decoded target channel.

Abstract: An apparatus includes a receiver and a decoder. The receiver is configured to receive a bitstream that includes an encoded mid channel and a quantized value representing a shift between a reference channel associated with an encoder and a target channel associated with the encoder. The quantized value is based on a value of the shift. The value of the shift is associated with the encoder and has a greater precision than the quantized value. The decoder is configured to decode the encoded mid channel to generate a decoded mid channel and to generate a first channel based on the decoded mid channel. The decoder is further configured to generate a second channel based on the decoded mid channel and the quantized value. The first channel corresponds to the reference channel and the second channel corresponds to the target channel.

Abstract: A device for processing audio signals includes an interchannel temporal mismatch analyzer, an interchannel phase difference (IPD) mode selector and an IPD estimator. The interchannel temporal mismatch analyzer is configured to determine an interchannel temporal mismatch value indicative of a temporal misalignment between a first audio signal and a second audio signal. The IPD mode selector is configured to select an IPD mode based on at least the interchannel temporal mismatch value. The IPD estimator is configured to determine IPD values based on the first audio signal and the second audio signal. The IPD values have a resolution corresponding to the selected IPD mode.

Abstract: A residual scaling unit is configured to determine a scaling factor for a residual channel based on an inter-channel mismatch value. The inter-channel mismatch value is indicative of a temporal alignment between a reference channel and a target channel. The residual scaling unit is further configured to scale (e.g., attenuate) the residual channel by the scaling factor to generate a scaled residual channel. A residual channel encoder is configured to encode the scaled residual channel as part of a bitstream.

Abstract: A device includes a processor and a transmitter. The processor is configured to determine a first mismatch value indicative of a first amount of a temporal mismatch between a first audio signal and a second audio signal. The processor is also configured to determine a second mismatch value indicative of a second amount of a temporal mismatch between the first audio signal and the second audio signal. The processor is further configured to determine an effective mismatch value based on the first mismatch value and the second mismatch value. The processor is also configured to generate at least one encoded signal having a bit allocation. The bit allocation is at least partially based on the effective mismatch value. The transmitter configured to transmit the at least one encoded signal to a second device.

Abstract: A method includes generating a windowed time-domain mid channel by applying two first asymmetric windows to a first frame of a time-domain mid channel and applying two second asymmetric windows to a second frame of the time-domain mid channel. The method includes transforming the windowed time-domain mid channel to a transform domain to generate sets of transform-domain mid channel data including first transform-domain mid channel data corresponding to a first mid channel window of the first frame and second transform-domain mid channel data corresponding to a second mid channel window of the first frame. The method includes performing an up-mix operation using the sets of transform-domain mid channel data, stereo parameters from the bit stream, and an interpolated parameter determined using an unevenly weighted interpolation between a first stereo parameter value associated with the first frame and a second stereo parameter value associated with the second frame.

Abstract: A device includes a processor and a transmitter. The processor is configured to determine a first value and a second value indicative of a first amount and a second amount, respectively, of a temporal mismatch between a first audio signal and a second audio signal. The processor is also configured to determine an effective value based on the first value and the second value, to select, based on the effective value, a first coding mode and a second coding mode, and to generate at least one encoded signal having a bit allocation. The at least one encoded signal is based on a first encoded signal and a second encoded signal that are based on the first coding mode and the second coding mode, respectively. The bit allocation is at least partially based on the effective mismatch value. The transmitter is configured to transmit the at least one encoded signal.

Abstract: A device includes an encoder configured to determine, during a first period, that a first audio signal is a leading signal and that a second audio signal is a lagging signal. The encoder is also configured to generate a first frame of at least one encoded signal based on a first modified version of the second audio signal that is generated by adjusting the second audio signal based on a first mismatch value. The encoder is configured to determine, during a second period, that the first audio signal is the leading signal and that the second audio signal is the lagging signal. The encoder is configured to generate a second frame of the at least one encoded signal based on a second modified version of the second audio signal that is generated by adjusting the second audio signal based on the first mismatch value and a second mismatch value.

Abstract: A device includes an encoder and a transmitter. The encoder is configured to generate a first high-band portion of a mid signal based on a left signal and a right signal, to generate a first synthesized signal based on a first gain and linear predictive coefficient (LPC) parameters, and to generate a second synthesized signal based at least in part on a second gain and the LPC parameters. The encoder is also configured to generate a set of first gain parameters based on a comparison of the first synthesized signal and the mid signal, and to generate a set of adjustment gain parameters based on the second synthesized signal and a high-band non-reference signal. The transmitter is configured to transmit information corresponding to the first high-band portion of the mid signal. The transmitter is also configured to transmit the set of adjustment gain parameters corresponding to the high-band non-reference signal.

Abstract: A method includes receiving, at an audio encoder, multiple streams of audio data. The method includes assigning a priority to each stream of the multiple streams and determining, based on the priority of each stream of the multiple streams, a permutation sequence for encoding of the multiple streams. The method also includes encoding at least a portion of each stream of the multiple streams according to the permutation sequence.

Abstract: A method includes decoding a low-band portion of an encoded mid signal to generate a decoded low-band mid signal. The method also includes processing the decoded low-band mid signal to generate a low-band residual prediction signal and generating a low-band left channel and a low-band right channel based partially on the decoded low-band mid signal and the low-band residual prediction signal. The method further includes decoding a high-band portion of the encoded mid signal to generate a time-domain decoded high-band mid signal and processing the time-domain decoded high-band mid signal to generate a time-domain high-band residual prediction signal. The method also includes generating a high-band left channel and a high-band right channel based on the time-domain decoded high-band mid signal and the time-domain high-band residual prediction signal.

Abstract: A method includes decoding a low-band portion of an encoded mid channel to generate a decoded low-band mid channel. The method also includes filtering the decoded low-band mid channel according to one or more filter coefficients to generate a low-band filtered mid channel. The method also includes generating an inter-channel predicted signal based on the low-band filtered mid channel and the inter-channel prediction gain. The method further includes generating a low-band left channel and a low-band right channel based on an up-mix factor, the decoded low-band mid channel, and the inter-channel predicted signal.

Abstract: An apparatus includes a receiver configured to receive at least one encoded signal that includes inter-channel bandwidth extension (BWE) parameters. The device also includes a decoder configured to generate a mid channel time-domain high-band signal by performing bandwidth extension based on the at least one encoded signal. The decoder is also configured to generate, based on the mid channel time-domain high-band signal and the inter-channel BWE parameters, a first channel time-domain high-band signal and a second channel time-domain high-band signal. The decoder is further configured to generate a target channel signal by combining the first channel time-domain high-band signal and a first channel low-band signal, and to generate a reference channel signal by combining the second channel time-domain high-band signal and a second channel low-band signal. The decoder is also configured to generate a modified target channel signal by modifying the target channel signal based on a temporal mismatch value.

Abstract: A device includes an encoder. The encoder is configured to receive two audio channels. The encoder is also configured to determine a mismatch value indicative of an amount of a temporal mismatch between the two audio channels. The encoder is further configured to determine, based on the mismatch value, at least one of a target channel or a reference channel. The target channel corresponds to a lagging audio channel of the two audio channels and the reference channel corresponds to a leading audio channel of the two audio channels. The encoder is also configured to generate a modified target channel by adjusting the target channel based on the offset value. The encoder is further configured to generate at least one encoded channel based on the reference channel and the modified target channel.

Abstract: The disclosure generally relates to various methods to increase network coverage with respect to Voice-over-Internet Protocol (VoIP) sessions and/or other voice-based multimedia services. In particular, when establishing a voice session, two or more terminals may perform a codec negotiation to exchange information related to supported multimedia codecs and/or codec modes, jitter buffer management (JBM), and packet loss concealment (PLC) capabilities. Based on the exchanged information, a message may be sent to a base station to indicate the maximum packet loss rate (PLR) for each terminal. Additional techniques may ensure that the terminals use the most robust codecs or codec modes that are available when nearing the edge of coverage to help avoid unnecessary and/or excessive handovers to different radio access technologies.

Abstract: An apparatus includes a first calculator configured to determine a long-term noise estimate of the audio signal. The apparatus also includes a second calculator configured to determine a formant-sharpening factor based on the determined long-term noise estimate. The apparatus includes a filter configured to filter a codebook vector to generate a filtered codebook vector. The filter is based on the determined formant-sharpening factor, and the codebook vector is based on information from the audio signal. The apparatus further includes an audio coder configured to generate a formant-sharpened low-band excitation signal based on the filtered codebook vector.

Abstract: A device includes a processor and a transmitter. The processor is configured to determine a first value and a second value indicative of a first amount and a second amount, respectively, of a temporal mismatch between a first audio signal and a second audio signal. The processor is also configured to determine an effective value based on the first value and the second value, to select, based on the effective value, a first coding mode and a second coding mode, and to generate at least one encoded signal having a bit allocation. The at least one encoded signal is based on a first encoded signal and a second encoded signal that are based on the first coding mode and the second coding mode, respectively. The bit allocation is at least partially based on the effective mismatch value. The transmitter is configured to transmit the at least one encoded signal.

Abstract: An apparatus includes a receiver and a decoder. The receiver is configured to receive a bitstream that includes an encoded mid channel and a quantized value representing a shift between a reference channel associated with an encoder and a target channel associated with the encoder. The quantized value is based on a value of the shift. The value of the shift is associated with the encoder and has a greater precision than the quantized value. The decoder is configured to decode the encoded mid channel to generate a decoded mid channel and to generate a first channel based on the decoded mid channel. The decoder is further configured to generate a second channel based on the decoded mid channel and the quantized value. The first channel corresponds to the reference channel and the second channel corresponds to the target channel.