Abstract:
A method, apparatus and computer-readable medium for transmitting a medium access control (MAC) protocol data unit (PDU) including one or more MAC service data units (SDUs) of a logical channel; and transmitting a first type of MAC subheader including or excluding first logical channel identifier (LCID) information and including or excluding first length information, wherein a first size of the first LCID information is equal to or smaller than a second size of second LCID information of a second type of MAC subheader that is different than the first type of MAC subheader, and a third size of the first length information is equal to or smaller than a fourth size of second length information of the second type of MAC subheader.
Abstract:
Techniques are described for coding audio signals. For example, using a neural network, a residual signal is generated for a sample of an audio signal based on inputs to the neural network. The residual signal is configured to excite a long-term prediction filter and/or a short-term prediction filter. Using the long-term prediction filter and/or the short-term prediction filter, a sample of a reconstructed audio signal is determined. The sample of the reconstructed audio signal is determined based on the residual signal generated using the neural network for the sample of the audio signal.
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:
A device includes a receiver configured to receive an audio frame of an audio stream. The device also includes a decoder configured to generate first decoded speech associated with the audio frame and to determine a count of audio frames classified as being associated with band limited content. The decoder is further configured to output second decoded speech based on the first decoded speech. The second decoded speech may be generated according to an output mode of the decoder. The output mode may be selected based at least in part on the count of audio frames.
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 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:
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 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.