30 results about "Wideband speech coding" patented technology

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

The present invention relates to a gain quantization method and device for implementation in a technique for coding a sampled sound signal processed, during coding, by successive frames of L samples, wherein each frame is divided into a number of subframes and each subframe comprises a number N of samples, where N<L. In the gain quantization method and device, an initial pitch gain is calculated based on a number f of subframes, a portion of a gain quantization codebook is selected in relation to the initial pitch gain, and pitch and fixed-codebook gains are jointly quantized. This joint quantization of the pitch and fixed-codebook gains comprises, for the number f of subframes, searching the gain quantization codebook in relation to a search criterion. The codebook search is restricted to the selected portion of the gain quantization codebook and an index of the selected portion of the gain quantization codebook best meeting the search criterion is found.

Provided is a highband coding apparatus and method for a wideband coding system. The coding apparatus and method can reduce a pre-echo phenomenon by encoding the highband based on lowband encoding information and Temporal Noise Shaping technique. A highband encoding apparatus includes: a domain converter for converting the domain of an input highband signal into a frequency domain; a linear prediction order determiner for determining a linear prediction order based on the lowband encoding information; a linear prediction analyzer for analyzing a highband signal of the frequency domain based on the determined linear prediction order to thereby generate a linear prediction coefficient; a linear prediction coefficient quantizer for quantizing the linear prediction coefficient based on the lowband encoding information; and a residual signal quantizer for obtaining a residual signal by dequantizing the quantized linear prediction coefficient and quantizing the residual signal.

Provided are a scalable wide-band speech coding / decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.

A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.

A speech encoder / decoder for wideband speech with a partitioning of wideband into lowband and highband, convenient coding of the lowband, and LP excited by noise plus some periodicity for the highband. The embedded lowband may be extracted for a lower bit rate decoder.

A speech encoder / decoder for wideband speech with a partitioning of wideband into lowband and highband, convenient coding of the lowband, and LP excited by noise plus some periodicity for the highband. The embedded lowband may be extracted for a lower bit rate decoder.

A wideband speech encoder according to one embodiment includes a narrowband encoder and a highband encoder. The narrowband encoder is configured to encode a narrowband portion of a wideband speech signal into a set of filter parameters and a corresponding encoded excitation signal. The highband encoder is configured to encode, according to a highband excitation signal, a highband portion of the wideband speech signal into a set of filter parameters. The highband encoder is configured to generate the highband excitation signal by applying a nonlinear function to a signal based on the encoded narrowband excitation signal to generate a spectrally extended signal.

Speech signal classification and encoding systems and methods are disclosed herein. The signal classification is done in three steps each of them discriminating a specific signal class. First, a voice activity detector (VAD) discriminates between active and inactive speech frames. If an inactive speech frame is detected (background noise signal) then the classification chain ends and the frame is encoded with comfort noise generation (CNG). If an active speech frame is detected, the frame is subjected to a second classifier dedicated to discriminate unvoiced frames. If the classifier classifies the frame as unvoiced speech signal, the classification chain ends, and the frame is encoded using a coding method optimized for unvoiced signals. Otherwise, the speech frame is passed through to the “stable voiced” classification module. If the frame is classified as stable voiced frame, then the frame is encoded using a coding method optimized for stable voiced signals. Otherwise, the frame is likely to contain a non-stationary speech segment such as a voiced onset or rapidly evolving voiced speech signal. In this case a general-purpose speech coder is used at a high bit rate for sustaining good subjective quality.

In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters. When classification of the sound signal determines that the signal-coding parameters should be transmitted using the first communication mode and when a request to transmit the signal-coding parameters from the coder of one station to the decoder of the other station using a second communication mode designed to reduce bit rate during transmission of the signal-coding parameters is received, a portion of the signal-coding parameters from the coder one station is dropped and the remaining signal-coding parameters are transmitting to the decoder of the other station using the second communication mode. The dropped portion of the signal-coding parameters are regenerated before the decoder of the other station decodes the signal-coding parameters.

The present invention relates to a gain quantization method and device for implementation in a technique for coding a sampled sound signal processed, during coding, by successive frames of L samples, wherein each frame is divided into a number of subframes and each subframe comprises a number N of samples, where N<L. In the gain quantization method and device, an initial pitch gain is calculated based on a number f of subframes, a portion of a gain quantization codebook is selected in relation to the initial pitch gain, and pitch and fixed-codebook gains are jointly quantized. This joint quantization of the pitch and fixed-codebook gains comprises, for the number f of subframes, searching the gain quantization codebook in relation to a search criterion. The codebook search is restricted to the selected portion of the gain quantization codebook and an index of the selected portion of the gain quantization codebook best meeting the search criterion is found.

A wide-band speech coder and a method thereof and a wide-band speech decoder and a method thereof are provided. The wide-band speech coder includes a speech characteristic classification unit, which stipulates a characteristic of speech corresponding to a current frame statistically using an open-circuit pitch value and a linear prediction coefficient in which a wide-code speech signal to be coded is perceptual weigh filtered, an adaptive codebook retrieving unit, which retrieves a pitch delay value around the open-circuit pitch value, calculates a pitch gain value, generates an adaptive codebook contribution signal corresponding to the retrieved pitch delay value, and outputs a difference between the generated adaptive codebook contribution signal and the perceptual weigh filtered signal as a first fixed codebook target signal, a first fixed codebook retrieving unit, which obtains a first fixed codebook index that can express the first fixed codebook target signal most properly, and a first fixed codebook gain value, generates a first fixed codebook contribution signal corresponding to the retrieved index, and outputs a difference between the first generated fixed codebook contribution signal and the first fixed codebook target signal as a second fixed codebook target signal, a second fixed codebook retrieving unit, which includes at least two second fixed codebooks according to a speech characteristic, selects a second fixed codebook according to the speech characteristic, and retrieves second fixed codebook indices that can express the second fixed codebook target signal most properly, and second fixed codebook gain values, and a parameter multiplexer, which quantizes and multiplexes the speech characteristic information, the pitch delay value, the pitch gain value, the first fixed codebook index, the first fixed codebook gain value, the second fixed codebook indices, and the second fixed codebook gain values, makes them as a bit stream, and transmits the bit stream to an external speech decoding terminal.

The LSF quantizer for a wideband speech coder comprises a subtracter for receiving an input LSF coefficient vector and removing a DC component from it; a memory-based vector quantizer and a memoryless vector quantizer for respectively receiving the DC-component-removed LSF coefficient vector and independently quantizing the same; a switch for receiving quantized vectors respectively quantized by the memory-based vector quantizer and the memoryless vector quantizer, selecting a quantized vector that has less quantized error that is a difference between the received quantized vector and the input LSF coefficent vector from among the received quantized vectors, and outputting the same; and an adder for adding the quantized vector selected by the switch to the DC component of the LSF coefficient vector.

A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.

In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters. When classification of the sound signal determines that the signal-coding parameters should be transmitted using the first communication mode and when a request to transmit the signal-coding parameters from the coder of one station to the decoder of the other station using a second communication mode designed to reduce bit rate during transmission of the signal-coding parameters is received, a portion of the signal-coding parameters from the coder one station is dropped and the remaining signal-coding parameters are transmitting to the decoder of the other station using the second communication mode. The dropped portion of the signal-coding parameters are regenerated before the decoder of the other station decodes the signal-coding parameters.

The invention discloses a method and a device for quantifying the vectors of the induced spectrum frequency (ISF) coefficients, which includes the steps as follows: the ISF coefficients are divided into at least two groups of sub-vectors; the inter-frame strong- and weak-correlation vectors of the induced spectrum are quantified for each group of sub-vectors; the residual signals are calculated after the strong- and weak-correlation vectors are quantified; the quantification errors of the above residual signals are also calculated; the quantification errors of the strong- and weak-correlation residual signals are judged for each group of sub-vectors; and the ISF coefficient vectors of the correlation manner with less quantification error in each group are quantified. The invention is based on the short-time correlation and the long-time correlation of the induced spectrum frequency coefficients, and different forecast coefficients are hereby applied to different dimension of the induced spectrum frequency coefficients, thus increasing forecast gains, narrowing the variation range of the residual signals, enhancing the accuracy for the ISF coefficients quantification, achieving clear quantification, and improving the quality for broadband phonetic coding.

The invention discloses a vector quantization method of an immittance spectrum frequency (ISF) parameter and a vector quantization device of the ISF. The invention also closes a vector quantization coding end method of the ISF parameter, a vector quantization decoding end method of the ISF parameter, a vector quantization device of the ISF parameter, a vector quantization coding end device of the ISF parameter and a vector quantization decoding end device of the ISF parameter. The methods or devices utilize prediction coefficients of unequal coefficient interframe predictor in each dimension to compute the prediction vector of the ISF parameter. Applying the invention can equalize the quantization performances when losing the frame or not in the process of bandwidth voice coding and decoding, improving the quality of voice synthesis.

The invention relates to a method for realizing the inter-system voice communication through a TFO means, a system and a TFO conversion device. The method comprises steps of receiving TFO negotiation information, performing conversion according to a receiver's requirement of the format of the TFO negotiation information, if the information comprises a speech encoding capability, converting the TFO negotiation information format to an encoding type supported by the receiver under a condition that the speech coding capability supports the broadband speech encoding type, if the information doesn't comprise the speech encoding capability, forwarding the TFO negotiation information which is going through the format conversion to the receiver, receiving the TFO frame, performing code conversion on voice data in the TFO frame and performing conversion on the control information and synchronization information in the frame, forming the TFO frame format required by the receiving through utilizing the converted voice data, control information and synchronization and forwarding the control information and the synchronization information to the receiver. The invention can openly realize the inter-system voice communication.

A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.

A method, apparatus, and system for encoding or decoding a broadband voice signal are provided. The method includes extracting a linear prediction coefficient (LPC) from the broadband voice signal; outputting a linear prediction (LP) residual signal; pitch-searching a spectrum of the LP residual signal; extracting spectral magnitudes and phases of the LP residual signal, which correspond to a damping factor; obtaining, from among the extracted spectral magnitudes and phases, a first spectral magnitude and a first phase at which a power value of the LP residual signal is minimized; quantizing the first spectral magnitude and the first phase; and decoding the broadband voice signal. The apparatus includes a linear prediction coefficient (LPC) analyzer; an LPC inverse filter; a pitch searching unit; a sinusoidal analyzer; and a phase and spectral magnitude quantizer. The system includes a broadband voice encoding apparatus and a broadband voice decoding apparatus.

The invention discloses a method and device for realizing TFO and TrFO intercommunication conversion between different systems. The method comprises the following steps: starting calling by a calling party, and reporting speech coding support ability of a calling terminal; sending a service request signaling from a first system to a GMSCe of a second system; negotiating speech coding through an out-of-band signaling by the GMSCe of the second system and an MSCe of the second system; negotiating whether broadband speech coding is used, and if the broadband speech coding is used, the GMSCe of the second system requesting the GMSCe of the second system to allocate Etc, finishing signaling continuing and carrying out user-plane TFO negotiation with the first system; and negotiating whether the broadband speech coding is used, and if so, entering broadband speech calling. The method and device can prevent the broadband speech coding from being converted into G.711 for transmission, and broadband speech effect is achieved.

A wideband speech coder is provided, wherein a frequency division module extracts high and low frequency components from wideband digital speech signals and gets high and low sub-band signals to output the signals thereof; a low band coder codes a low sub-band signal into a low band code stream data and outputs the data; a low band decoder decodes the low band code stream data into low sub-band speech signals and outputs the signals; a difference operation module leads the original low sub-band signal to subtract the low sub-band speech signal to obtain low sub-band differential signals and outputs the signals; a low band enhanced coder codes low sub-band differential signals into low band enhanced code stream data and outputs the data; a high band enhanced coder codes high sub-band signals into high band enhanced code stream data and outputs the data; a code stream multiplexing module processes the low band code stream data, the low band enhanced code stream data and the high band enhanced code stream data into the wideband speech code stream data by multiplexing and packaging and outputs the data. The wideband speech coder has the characteristics of applicability to the wideband speech by expanding the bandwidth of voice coders, low delay and low complexity.

The present invention relates to a gain quantization method and device for implementation in a technique for coding a sampled sound signal processed, during coding, by successive frames of L samples,wherein each frame is divided into a number of subframes and each subframe comprises a number N of samples, where N<L. In the gain quantization method and device, an initial pitch gain is calculated based on a number f of subframes, a portion of a gain quantization codebook is selected in relation to the initial pitch gain, and pitch and fixed-codebook gains are jointly quantized. This joint quantization of the pitch and fixed-codebook gains comprises, for the number f of subframes, searching the gain quantization codebook in relation to a search criterion. The codebook search is restricted tothe selected portion of the gain quantization codebook and an index of the selected portion of the gain quantization codebook best meeting the search criterion is found.

The invention discloses a method and a device for selecting an adaptive codebook excitation signal. The method calculates high frequency excitation correlation according to the high frequency part of the searched adaptive codebook and the residual signals of the input speech signals. If the calculated high frequency excitation correlation is higher than the preset correlation threshold value, the searched adaptive codebook signal then is selected as the current adaptive codebook excitation signal; otherwise, the low frequency part of adaptive codebook searched and low pass filter disposed is selected as the current adaptive codebook excitation signal. The invention can improve wideband speech coding performance. Tests shows that the technical scheme improves coding signal-to-noise ratio (SNR) compared with prior art. Besides, the complexity of selection calculation is greatly decreased. Statistical analyses suggests that the complexity of the calculation method is only 60 percent of that in prior art.

The disclosure relates to a method, a system and a TFO conversion device for realizing voice communication between different systems in a TFO manner. The method includes receiving a TFO negotiation message; converting the TFO negotiation message according to the format requirements of the receiving party; if the message contains voice coding capability, converting it to the receiving party in the case that the voice coding capability supports wideband voice coding type For the supported encoding type, forward the TFO negotiation message for format and encoding type conversion to the receiver; if it does not include voice encoding capability, forward the TFO negotiation message for format conversion to the receiver; Voice data is coded and converted, and the control information and synchronization information in the TFO frame are converted, and the converted voice data, control information and synchronization information are used to form the TFO frame format required by the receiver and forwarded to the receiver. The present disclosure can realize voice intercommunication between different systems.

The invention discloses a method and a device for interworking and converting TFO and TrFO between different systems. The calling party initiates a call and reports the voice coding support capability of the calling terminal; the service request signaling is sent from the first system to the GMSCe of the second system; the GMSCe of the second system and the MSCe of the second system negotiate voice coding through out-of-band signaling; Negotiate whether to use wideband voice coding, if so, the GMSCe of the second system requires the GMGW of the second system to allocate eTC, complete the signaling connection, and conduct user plane TFO negotiation with the first system; negotiate whether to use wideband voice coding, if yes, enter Wideband voice calls. The present invention avoids wideband voice code conversion into G.711 transmission, and obtains wideband voice effect.

The invention discloses a call method and device, an electronic terminal and a medium. In the invention, When a first call instruction is received in a GSM network environment, A first call request message is sent through a target cell, when a first call failure response generated for a first call instruction is received, communication parameters of the target cell are recorded, and when a secondcall instruction is received, whether a currently registered cell is the target cell or not is judged based on the communication parameters. And if the target cell is in the target cell, second call request information which does not carry the identifier supporting the broadband voice coding function is sent through the target cell. By applying the technical scheme provided by the invention, the call request information supporting the broadband voice coding function is not sent to the cell when the registered cell having the call failure is connected. Therefore, the defect that when the electronic terminal is located in a cell which does not support broadband voice coding in a GSM network environment, a call cannot be initiated to an opposite-end user and a call function can not be carriedout is avoided, so that the user experience is improved.

The embodiment of the invention discloses voice signal coding and decoding methods, devices and systems. The coding method comprises: coding an input voice signal to obtain a broadband code stream, wherein the broadband code stream comprises a core layer bit stream and an expanded and reinforced layer bit stream; compressing the core layer bit stream to obtain a compressed code stream; and packing the compressed code stream and the expanded and reinforced layer bit stream to obtain a packed code stream. In the embodiment of the invention, by pressing the core layer bit stream and packing the compressed code stream and the expanded and reinforced layer bit stream, the transmission bandwidth occupied by the input voice signal is reduced; as the input voice signal is subjected to broadband voice coding, the transmission of broadband voice code stream by the transmission bandwidth of a narrow band is realized, and the performance-to-price ratio of the voice signal transmission is improved.