Abstract: An audio coding system in which a plurality of quantization methods are selectable for application to components of a streamed audio signal to achieve a target frame size that is determined by comparing an achieved bit rate against a target bit rate. Based on the target frame size, the system calculates a bit allocation for signal components and compares the bit allocation to the dynamic range of the signal components. Depending on the outcome of the comparison, the system may select to quantize or not quantize a signal component. The system employs lossless coding techniques, but is capable of introducing lossy coding by quantization in order to meet the target bit rate.

Abstract: An encoding apparatus includes a noise detector configured to detect noise included in a certain band in accordance with an audio signal, a gain controller configured to perform gain control on the audio signal so that components in the certain band of the audio signal are attenuated when the noise is detected by the noise detector, a bit allocation calculation unit configured to calculate the numbers of bits to be allocated to frequency spectra of the audio signal which have been subjected to the gain control performed by the gain controller in accordance with the frequency spectra, and a quantization unit configured to quantize the frequency spectra of the audio signal which have been subjected to the gain control in accordance with the numbers of the bits.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: Methods and apparatus for audio and speech processing including generating a plurality of frames, each of the frames comprising a plurality of transform coefficients, and allocating bits to the transform coefficients in each of the frames such that at least two of the transform coefficients in the same frame have different bit allocations and the total number of the bits allocated to the transform coefficients in at least two of the frames is equal.

Abstract: A method system and program for encoding and decoding a speech signal including error correction data. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, analysing the speech signal to determine side information and a residual signal, encoding the residual signal at a first bit rate, and generating an output bitstream based on the residual signal encoded at the first bit rate, and for at least one of the plurality of frames of the speech signal, encoding the residual signal at a second bit rate that is lower than the first bit rate; and generating error correction data based on the residual signal encoded at the second bit rate.

Abstract: An encoder capable of reducing the degradation of the quality of the decoded signal in the case of band expansion in which the high band of the spectrum of an input signal is estimated from the low band. In this encoder, a first layer encoding section (202) encodes an input signal and generates first encoded information, a first layer decoding section (203) decodes the first encoded information and generates a first decoded signal, a characteristic judging section (206) analyzes the intensity of the harmonic structure of the input signal and generates harmonic characteristic information representing the analysis result, and a second layer encoding section (207) changes, on the basis of the harmonic characteristic information, the numbers of bits allocated to parameters included in second encoded information created by encoding the difference between the input signal and the first decoded signal before creating the second information.

Abstract: An apparatus and method of audio encoding and decoding based on a Variable Bit Rate (VBR) is provided. The audio encoding and decoding apparatus and method may determine an optimum bit rate per superframe and per frame, determine an optimum encoding mode by applying an open-loop mode/closed-loop mode based on a characteristic of an audio signal, and perform indexing based on the optimum encoding mode.

Abstract: A system and method are described for controlling the setup of a connection in a communication network that includes a set of nodes, such as Mobile Switching Centers (MSCs) and Media Gateways (MGWs). In one example, a speech connection is established between MGWs subject to the control of the MSCs, which can selectively activate or deactivate codecs along the connection. The codecs are selected from a list of supported codecs, each potentially affecting connection quality by differing amounts. A Total Accumulated Impairment (TAI) element is forwarded between the MSCs and step by step updated by these MSCs that includes individual partially accumulated impairment values corresponding to each of the supported codec candidates. Each individual indicator value provides information representative of the expected accumulated impairment along a candidate connection path leading up to, and including, the corresponding codec.

Abstract: Information is estimated to fill-in even relatively long gaps (e.g., up to 250 ms) that occur in a signal due to physical errors in media or transmission, where the omitted information causes signal distortion. The signal is first divided into a plurality of subbands, since the gaps in each subband are individually easier to interpolate. Coherent demodulation is then employed on each subband signal to reduce the time-varying signals to a collection of pairs of frequency-modulated carriers multiplied by complex-valued envelopes, or modulators. Standard interpolation is then separately applied to the modulators and carriers of these pairs to fill-in the gaps in each of the subbands, and the interpolated pairs are remodulated. The resulting interpolated signals from each of the subbands are recombined to form the final interpolated output signal in which the gaps are filled in with estimated data.

Abstract: An audio encoding method previously estimates better initial iterative values of global-gain and scalefactor for avoiding heavy calculation. The estimating process of the encoding method includes calculating the bit allocation of one frequency sample based on a sampling rate, a bit rate, and the number of audio channels according to an input frame, and the psychoacoustic model, searching one frequency sample having the greatest sample energy in each of a plurality of scalefactor bands, quantizing the frequency sample to comply with the bit allocation and to generate a corresponding scalefactor, searching a maximum scalefactor of all scalefactor bands corresponding to the input frame, and setting initial values of scalefactors and an initial value of global-gain for the quantization iterative loop process according to the corresponding scalefactor and the maximum scalefactor.

Abstract: A method and apparatus for frame classification and rate determination in voice transcoders. The apparatus includes a classifier input parameter preparation module that unpacks the bitstream from the source codec and selects the codec parameters to be used for classification, parameter buffers that store previous input and output parameters of previous frames, and a frame classification and rate decision module that uses the source codec parameters from the current frame and zero or more frames to determine the frame class, rate, and classification feature parameters for the destination codec. The classifier input parameter preparation module separates the bitstream code and unquantizes the sub-codes into the codec parameters. The frame classification and rate decision module comprises M sub-classifiers and a final decision module.

Abstract: Provided are a method and apparatus to encode and decode an audio signal. According to the present invention, by splitting an input signal into a low band signal and a high band signal, converting each of the low band signal and the high band signal from the time domain to the frequency domain, performing quantization and context-dependant bitplane encoding on the converted low band signal, generating and encoding bandwidth extension information that represents a characteristic of the converted high band signal, and outputting the encoded bitplane and the encoded bandwidth extension information as an encoded result of the input signal, high frequency components may be efficiently encoded at a restricted bit rate, thereby improving the quality of an audio signal.