Patent classifications
G10L19/09
Systems, methods, apparatus, and computer-readable media for adaptive formant sharpening in linear prediction coding
A method of processing an audio signal includes determining an average signal-to-noise ratio for the audio signal over time. The method includes, based on the determined average signal-to-noise ratio, a formant-sharpening factor is determined. The method also includes applying a filter that is based on the determined formant-sharpening factor to a codebook vector that is based on information from the audio signal.
Encoding method, decoding method, encoder apparatus, decoder apparatus, and recording medium for processing pitch periods corresponding to time series signals
In encoding, pitch periods for time series signals in a predetermined time interval are calculated, and a code corresponding thereto is output. In that encoding, the resolutions for expressing the pitch periods and/or a pitch period encoding mode are switched according to whether an index indicating a periodicity and/or stationarity level of the time series signals satisfies a condition indicating high or low in periodicity and/or stationarity. In that decoding, according to whether an index indicating a periodicity and/or stationarity level, the index being included in or obtained from an input code corresponding to the predetermined time interval, satisfies a condition indicating high periodicity and/or stationarity, a decoding mode for a code, included in the input code, corresponding to pitch periods is switched to decode the code corresponding to the pitch periods to obtain the pitch periods corresponding to the predetermined time interval.
Encoding method, decoding method, encoder apparatus, decoder apparatus, and recording medium for processing pitch periods corresponding to time series signals
In encoding, pitch periods for time series signals in a predetermined time interval are calculated, and a code corresponding thereto is output. In that encoding, the resolutions for expressing the pitch periods and/or a pitch period encoding mode are switched according to whether an index indicating a periodicity and/or stationarity level of the time series signals satisfies a condition indicating high or low in periodicity and/or stationarity. In that decoding, according to whether an index indicating a periodicity and/or stationarity level, the index being included in or obtained from an input code corresponding to the predetermined time interval, satisfies a condition indicating high periodicity and/or stationarity, a decoding mode for a code, included in the input code, corresponding to pitch periods is switched to decode the code corresponding to the pitch periods to obtain the pitch periods corresponding to the predetermined time interval.
Packet loss concealment for speech coding
A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.
Packet loss concealment for speech coding
A speech coding method of reducing error propagation due to voice packet loss, is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class. A pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. A strongly voiced class is decided by checking if the pitch lags are stable and the pitch gains are high enough with the frame; for the strongly voiced frame, the pitch lags and the pitch gains can be encoded more efficiently than other speech classes.
Audio coding device, audio coding method, audio coding program, audio decoding device, audio decoding method, and audio decoding program
An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.
Audio coding device, audio coding method, audio coding program, audio decoding device, audio decoding method, and audio decoding program
An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.
POST FILTER FOR AUDIO SIGNALS
In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.
POST FILTER FOR AUDIO SIGNALS
In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.
AUDIO SIGNAL COMPRESSION METHOD AND APPARATUS USING DEEP NEURAL NETWORK-BASED MULTILAYER STRUCTURE AND TRAINING METHOD THEREOF
A method, executed by a processor for compressing an audio signal in multiple layers, may comprise: (a) restoring, in a highest layer, an input audio signal as a first signal; (b) restoring, in at least one intermediate layer, a signal obtained by subtracting an upsampled signal, which is obtained by upsampling the audio signal restored in the highest layer or an immediately previous intermediate layer, from the input audio signal as a second signal; and (c) restoring, in a lowest layer, a signal obtained by subtracting an upsampled signal, which is obtained by upsampling the audio signal restored in an intermediate layer immediately before the lowest layer, from the input audio signal as a third signal, wherein the first signal, the second signal, and the third signal are combined to output a final restoration audio signal.