A signal decoding method and device, where the method includes decoding a bit stream of a voice signal or an audio signal to acquire a decoded signal, predicting an excitation signal of an extension band according to the decoded signal, where the extension band is adjacent to a band of the decoded signal, and the band of the decoded signal is lower than the extension band; selecting a first band and a second band from the decoded signal, and predicting a spectral envelope of the extension band according to a spectral coefficient of the first band and a spectral coefficient of the second band; and determining a frequency-domain signal of the extension band according to the spectral envelope of the extension band and the excitation signal of the extension band.

In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.

An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

An apparatus for determining an estimated pitch lag is provided. The apparatus includes an input interface for receiving a plurality of original pitch lag values, and a pitch lag estimator for estimating the estimated pitch lag. The pitch lag estimator is configured to estimate the estimated pitch lag depending on a plurality of original pitch lag values and depending on a plurality of information values, wherein for each original pitch lag value of the plurality of original pitch lag values, an information value of the plurality of information values is assigned to the original pitch lag value.

Provided are a terminal device and method of performing a call function transmitting ambient audio with high sensitivity.

A terminal device performing a call function with at least one external device via a network may include a receiver configured to receive at least one of an audio transmission signal and a video transmission signal to be transmitted to the external device; a processor configured to analyze at least one of the audio transmission signal and the video transmission signal, select one of a speech mode and an audio mode, based on a result of the analysis, and compress the audio transmission signal, based on the selected mode; a communicator configured to transmit the compressed audio transmission signal to the external device, and receive an audio reception signal from the external device; and an output unit configured to output the audio reception signal.

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.

A frame loss compensation processing method and apparatus is presented, where the method includes, when a i.sup.th frame is a lost frame, estimating a spectrum frequency parameter, a pitch period, and a gain of the i.sup.th frame according to at least one of an inter-frame relationship between first N frames of the i.sup.th frame or an intra-frame relationship between first N frames of the i.sup.th frame. A parameter of the i.sup.th frame is determined using the signal correlation between the first N frames, the signal energy stability between the first N frames, intra-frame signal correlation of each frame, and intra-frame signal energy stability of each frame.

A frame loss compensation processing method and apparatus is presented, where the method includes, when a i.sup.th frame is a lost frame, estimating a spectrum frequency parameter, a pitch period, and a gain of the i.sup.th frame according to at least one of an inter-frame relationship between first N frames of the i.sup.th frame or an intra-frame relationship between first N frames of the i.sup.th frame. A parameter of the i.sup.th frame is determined using the signal correlation between the first N frames, the signal energy stability between the first N frames, intra-frame signal correlation of each frame, and intra-frame signal energy stability of each frame.

Provided are a vector quantization device, a voice coding device, a vector quantization method, and a voice coding method which enable a reduction in the calculation amount of voice codec without deterioration of voice quality. In the vector quantization device, a first reference vector calculation unit (201) calculates a first reference vector by multiplying a target vector (x) by an auditory weighting LPC synthesis filter (H), and a second reference vector calculation unit (202) calculates a second reference vector by multiplying an element of the first reference vector by a filter having a high pass characteristic. A polarity preliminary selection unit (205) generates a polar vector by disposing a unit pulse having a positive or negative polarity, which is selected on the basis of the polarity of an element of the second reference vector, in the position of said element.