摘要:
A speech encoding or decoding arrangement (711, 721, 811, 821) comprises a speech signal input and a multiple mode speech encoder (402) or decoder (411) for encoding or decoding speech signals coupled to the speech signal input selectabily with a first encoding or decoding mode associated with a first bandwidth or a second encoding or decoding mode associated with a second bandwidth. It comprises a soft bandwidth switching block (401, 412, 500) with an input (IN) and an output (OUT). In an encoding arrangement the input (IN) is coupled to the speech signal input and the output (OUT) is coupled to the multiple mode speech encoder (402). In a decoding arrangement the input (IN) is coupled to the multiple mode speech decoder (411) and the output (OUT) is the output of the decoding arrangement. The soft bandwidth switching block (401, 412, 500) is arranged to gradually change the bandwidth of a speech signal coupled to the multiple mode speech encoder or decoder as a response to an instruction for changing speech signal bandwidth (421).
摘要:
A speech decoder comprises a decoder (103) for converting a linear prediction encoded speech signal into a first sample stream having a first sampling rate and representing a first frequency band. Additionally it comprises a vocoder (105) for converting an input signal into a second sample stream having a second sampling rate and representing a second frequency band, and combination means (107) for combining the first and second sample streams in processed form. It comprises also means (301) for generating a second linear prediction filter, to be used by the vocoder (105) on the second frequency band, on the basis of a first linear prediction filter used by the decoder (103) on the first frequency band. Extrapolation through an infinite impulse response filter is the preferable method of generating the second linear prediction filter.
摘要:
A method for use by a speech decoder in handling bad frames received over a communications channel a method in which the effects of bad frames are concealed by replacing the values of the spectral parameters of the bad frames (a bad frame being either a corrupted frame or a lost frame) with values based on an at least partly adaptive mean of recently received good frames, but in case of a corrupted frame (as opposed to a lost frame), using the bad frame itself if the bad frame meets a predetermined criterion. The aim of concealment is to find the most suitable parameters for the bad frame so that subjective quality of the synthesized speech is as high as possible.
摘要:
A method and corresponding apparatus for encoding a sequence of bits for transmission as symbols, some of the bit positions of the symbols having a higher bit error rate than other bit positions. A plurality of sequences of bits is provided using a convolutional encoder, in response to a sequence of input bits, each sequence of bits being defined by a predetermined generator polynomial having a predetermined level of sensitivity to puncturing. Then the bits of each sequence of bits are mapped to symbol positions based on the level of sensitivity of the generator polynomial defining the sequence of bits. With interleaving, the mapping of bits of each sequence of bits to symbol positions can precede a symbol interleaving step, or it can follow a bit interleaving step.
摘要:
A method for use by a speech decoder in handling bad frames received over a communications channel a method in which the effects of bad frames are concealed by replacing the values of the spectral parameters of the bad frames (a bad frame being either a corrupted frame or a lost frame) with values based on an at least partly adaptive mean of recently received good frames, but in case of a corrupted frame (as opposed to a lost frame), using the bad frame itself if the bad frame meets a predetermined criterion. The aim of concealment is to find the most suitable parameters for the bad frame so that subjective quality of the synthesized speech is as high as possible.
摘要:
A method and system for concealing errors in one or more bad frames in a speech sequence as part of an encoded bit stream received in a decoder. When the speech sequence is voiced, the LTP-parameters in the bad frames are replaced by the corresponding parameters in the last frame. When the speech sequence is unvoiced, the LTP-parameters in the bad frames are replaced by values calculated based on the LTP history along with an adaptively-limited random term.
摘要:
The invention relates to a method for transmitting background noise information including a silence descriptor identifier and background noise parameters in a communication system in which the information to be transmitted is formed into data frames. The data frames are subjected to channel coding to form channel-coded frames. The channel-coded frames are interleaved to be transmitted in two or more data transmission frames, and information of two channel-coded frames is transmitted in each data transmission frame. A first silence descriptor frame is formed provided with the silence descriptor identifier. The first silence descriptor frame is subjected to channel coding to form a channel-coded silence descriptor frame. The channel-coded silence descriptor frame is transmitted in two or more data transmission frames, and at least one data transmission frame transmitting part of the channel-coded silence descriptor frame is also used to transmit at least the background noise parameters.
摘要:
A speech coding method and device for encoding and decoding an input signal and providing synthesized speech, wherein the higher frequency components of the synthesized speech are achieved by high-pass filtering and coloring an artificial signal to provide a processed artificial signal. The processed artificial signal is scaled by a first scaling factor during the active speech periods of the input signal and a second scaling factor during the non-active speech periods, wherein the first scaling factor is characteristic of the higher frequency band of the input signal and the second scaling factor is characteristic of the lower frequency band of the input signal. In particular, the second scaling factor is estimated based on the lower frequency components of the synthesized speech and the coloring of the artificial signal is based on the linear predictive coding coefficients characteristic of the lower frequency of the input signal.
摘要:
A wireless telecommunications system comprises a mobile station (MS) and a network. The mobile station has a multi-rate speech encoder which produces an encoded speech signal which is transmitted to the network. The network has a multi-rate speech decoder which decodes the encoded speech signal to produce a decoded speech signal. The network also comprises a signal analyser which measures speech characteristics of the decoded speech signal to produce speech characteristics information and an up-link mode control unit which receives the speech characteristics information and produces a mode command. The mode command is transmitted by the network to the mobile station where it is used to control the speech encoding bit rate of the multi-rate speech encoder.
摘要:
A method for use by a speech decoder in handling bad frames received over a communications channel a method in which the effects of bad frames are concealed by replacing the values of the spectral parameters of the bad frames (a bad frame being either a corrupted frame or a lost frame) with values based on an at least partly adaptive mean of recently received good frames, but in case of a corrupted frame (as opposed to a lost frame), using the bad frame itself if the bad frame meets a predetermined criterion. The aim of concealment is to find the most suitable parameters for the bad frame so that subjective quality of the synthesized speech is as high as possible.