摘要:
The coder generates code information A.sub.1 -A.sub.N, each representing one of Q quantization points of an input signal, where Q is an integer equal to or less than 2.sup.x and X is a positive number, synthesizes the code information A.sub.1 -A.sub.N into a code H through an operation:H=A.sub.1 Q.sup.N-1 +A.sub.2 Q.sup.N-2 + . . . +A.sub.N-1 Q+A.sub.N,and outputs the code H. The decoder inputs the code H, separates the code H into code information A.sub.1 -A.sub.N through operations with decimal fractions of quotients truncated:A.sub.1 =H/Q.sup.N-1A.sub.2 =(H-A.sub.1 Q.sup.N-1)/Q.sup.N-2A.sub.N-1 =(H-A.sub.1 Q.sup.N-1 -A.sub.2 Q.sup.N-2 - . . . -A.sub.N-2 Q.sup.2)/QA.sub.N =H-A.sub.1 Q.sup.N-1 -A.sub.2 Q.sup.N-2 - . . . -A.sub.N-2 Q.sup.2 -A.sub.N-1 Q,and reproduces an output signal based upon the thus-separated code information A.sub.1 -A.sub.N.
摘要:
An apparatus synchronizes a voice coder and a voice decoder which are of the vector-coding type in order to prevent a false synchronization even when a signal having the same period as a string of synchronizing bits is inputted. A noise component adding unit adds a noise component to an input voice signal. Therefore, even if the input voice signal has the same period as that of a string of synchronizing bits and is completely periodic, the periodicity of the input voice signal is lost by the added noise component. Based on the input voice signal which is no longer periodic, a vector-coding unit, a quantizing signal vector generating unit, and a code book index transmitting unit generate code book indexes and transmit the generated code book indexes to a voice decoder. Therefore, the voice decoder is prevented from developing a false synchronization.
摘要:
An audio encoding apparatus for stereo audio encoding an L-channel PCM signal and an R-channel PCM signal efficiently allocates encoded data of the L-channel and the R-channel without varying an existing format and performs MS stereo on/off control and controls of a bit allocation amount or a frame region for the inputted PCM signals while miniaturization of the apparatus can be anticipated. A correlation degree calculation section calculates, based on the PCM signals of the L-channel and the R-channel, a correlation degree between the PCM signals, and decision section decides whether or not a stereo encoding process should be performed based on the calculated correlation degree. An allocation section allocates regions for individually storing a difference signal and a sum signal between the PCM signals based on a result of the decision, and an audio encoding section encodes the difference signal and the sum signal based on the allocated regions.
摘要:
An audio coding device that optimizes quantization parameters for fast convergence of iterations. A quantized bit counter calculates a codeword length representing the number of bits of a Huffman codeword corresponding to quantized values. The quantized bit counter also calculates a codebook number bit count representing how many bits are consumed for optimal Huffman codebook numbers, and a scale factor bit count representing how many bits are consumed for scale factors of each subband. In a first stage of quantization, the quantized bit counter accumulates lengths of Huffman codewords corresponding to quantized values of every nth subband. A bit count estimator calculates a total bit count estimate by adding up n times the accumulated codeword length, the codebook number bit count, and the scale factor bit count. A parameter updater updates quantization parameters if the total bit count estimate exceeds a bit count limit.
摘要:
An encoding device which encodes audio signals comprises a spectrum power calculation unit for calculating the power of each spectrum obtained by analyzing the frequency of an input audio signal, a tonality parameter calculation unit for calculating a tonality parameter indicating the pure tone level of the input audio signal in each sub-band, using the result of the calculation when dividing the frequency range of the spectrum of the input audio signal into a plurality of sub-bands, and a dynamic masking threshold calculation unit for calculating a dynamic masking threshold value of the masking energy of the input audio signal, using the calculated tonality parameter.
摘要:
An audio coding device that optimizes quantization parameters for fast convergence of iterations. A quantized bit counter calculates a codeword length representing the number of bits of a Huffman codeword corresponding to quantized values. The quantized bit counter also calculates a codebook number bit count representing how many bits are consumed for optimal Huffman codebook numbers, and a scale factor bit count representing how many bits are consumed for scale factors of each subband. In a first stage of quantization, the quantized bit counter accumulates lengths of Huffman codewords corresponding to quantized values of every nth subband. A bit count estimator calculates a total bit count estimate by adding up n times the accumulated codeword length, the codebook number bit count, and the scale factor bit count. A parameter updater updates quantization parameters if the total bit count estimate exceeds a bit count limit.