摘要:
An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization means having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.
摘要:
A semiconductor memory card for storing audio information with corresponding text information and type information, the type information indicating a type of the text information. The type is classified into at least (a), (b), and (c) in which the text information respectively includes a 1-byte character code sequence, a 2-byte character code sequence, and a 1-byte character code sequence and a 2-byte character code sequence.
摘要:
The present invention is a signal processing device which performs parallel processes A and B efficiently. There is a deviation in the throughputs of the process A and B in processing an audio signal. First to Nth sub signal processing sections have capabilities to complete the process A within a period (N×T). A main signal processing sections has a capability to complete the process B within a period T. Efficient signal processing can be achieved by processing an input digital signal by means of distinct sub signal processing devices one after another and then processing the signal by the main signal processing section.
摘要:
An encoder of the present invention includes: a number G of storage sections (G is an integer equal to or greater than 1) for storing a number G of groups of data; a Huffman codebook selection section for selecting one of a number H of Huffman codebooks (H is an integer equal to or greater than 1) for each of the groups of data stored in the respective storage sections, each of the Huffman codebooks having a codebook number; a number G of Huffman encoding sections, each of the Huffman encoding sections Huffman-encoding a corresponding one of the G groups of data by using one of the Huffman codebooks which is selected by the Huffman codebook selection section for the one group of data; and a codebook number encoding section for encoding the codebook number of each Huffman codebook selected by the Huffman codebook selection section. The Huffman codebook selection section includes a code length calculation section for calculating a code length which would result from a Huffman encoding operation of each of the G groups of data using each Huffman codebook, and a control section for selecting one of the Huffman codebooks which is suitable for the group of data based on the code length calculated by the code length calculation section. When the Huffman codebook selected is an unsigned codebook, a number of bits required for sign information has previously been added to the code length calculated by the code length calculation section.
摘要:
An audio stream is divided into a plurality of audio object (AOB) files that are recorded having each been encrypted using a different encryption key. At least one piece of track management information (TKI) is provided corresponding to each track. Playlist information (PLI) assigns a playback position in a playback order to each track when a plurality of tracks are to be played back one after the other.
摘要:
An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization means having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing the residual signal using the weighting coefficients. Therefore, a low frequency band, which is auditively important, can be analyzed with a higher frequency resolution as compared with a high frequency band, whereby efficient signal compression utilizing human auditory characteristics is realized.