Abstract:
The signal (801) of a first code sequence of a predetermined format a part of which is dummy data is sent to a code sequence rewriting means (1802) through a code sequence decomposing means (1801). The signal (806) of a second code sequence with which the part corresponding to the dummy data in the signal (801) is supplemented is sent to the code sequence rewriting means (1802) through a control means (1805), and the dummy data in the first code sequence is rewritten to the second code sequence. For trial viewing/listening, the signal (802) from the code sequence decomposing means (1801) is selected by means of a selector switch (1808). When the second code sequence (806) is acquired by purchase of a content, the signal (803) from the code sequence rewriting means (1802) is selected. Thus trial viewing/listening of a content such as music free of danger of decryption is possible, and a high-quality content can be reproduced by acquiring a relatively small amount of data.
Abstract:
A center (content supply center) (1865) for storing/managing a content is connected to user terminals (1861-1864) used by users through networks (861-867). The center (1865) sends a signal of a content for trial viewing/listening to a user terminal free or at a low price. The user terminal (1861-1864) receives the signal, and the user can select, purchase, and reproduce at high quality only the content that the user likes. Thus trial viewing/listening is possible, and the opportunity to purchase a content is enhanced since it is unnecessary to download a large amount of data with high quality from a content supply center.
Abstract:
A signal encoding process comprises the steps of determining a gain control function of an overlapping portion of the present block on the basis of a gain control function of the previous block at the overlapping portion of the present block and the previous block of an input waveform signal; controlling the gain of the input waveform signal corresponding to the present block by using the gain control function of the overlapping portion; extracting a sample overlapping between adjacent blocks on the time axis to convert the sample to a signal on the frequency axis; encoding the converted signal and the control data for gain control; and outputting the encoded signal. A signal decoding process comprises the steps of decoding the encoded signal to take out the conversion signal and the control data; effecting inverse conversion for the converted signal by the interference of the waveform elements between the adjacent blocks; determining the gain control function of the overlapping portion of the present block on the basis of the control data and the gain control function of the previous block at the overlapping portion between the present block and the previous block; controlling the gain of the inversely converted signal corresponding to the present block by using the gain control function of the overlapping portion; and restoring the original waveform signal. According to this method, the amplification in the encoding and the corresponding correction in the decoding can be performed without any contradiction between the blocks in the case or the interference of the waveform elements of both adjacent blocks at the time of inverse conversion such as MDCT, so that the occurence of the pre-echo can be prevented by a simple construction using spectrum conversion having high encoding efficiency.
Abstract:
In an apparatus for, and a method of, encoding signals according to the present invention, a normalization coefficient for normalizing a first signal is set into a smaller interval then that of a normalization coefficient for normalizing a second signal. In other words, accuracy of the normalization coefficient at the time of normalization of a tone component is set to a higher level than that of the normalization coefficient at the time of normalization of a noise component. Accordingly, the present invention provides an efficient method of encoding. In other words, the present invention improves the accuracy of encoding of the tone component while restricting the number of bits necessary for encoding the noise component, thus improving the encoding efficiency as a whole. In the recording medium of the present invention, since signals efficiently encoded by the signal encoding apparatus or method of the present invention are recorded, a recording capacity can be effectively utilized. Further, because the signal decoder of the present invention decodes the signals encoded by the signal encoding apparatus or method of the present invention, signal error during decoding can be reduced.
Abstract:
An input signal is divided into blocks and converted into spectrum signals. Each of the spectrum signals are further divided into units and normalized. The normalized spectrum signals are transformed into variable-length codes and outputted together with the normalization coefficients and the number of bits of requantization. An upper limit is put on the number of bits of the outputted signals per block. If the numbers of bits of some signals blocks exceed the upper limit, the normalization coefficients of at least one of the units are forcedly changed. The signals whose normalization coefficients have been forcedly changed are requantized, entropy-coded, and outputted. Thus, without influence of the variation of the numbers of bits due to the variable-length encoding, the hardware scale can be smaller that conventional ones, and the encoding/decoding is efficient and not aurally affected.
Abstract:
The signal (801) of a first code sequence of a predetermined format a part of which is dummy data is sent to a code sequence rewriting means (1802) through a code sequence decomposing means (1801). The signal (806) of a second code sequence with which the part corresponding to the dummy data in the signal (801) is supplemented is sent to the code sequence rewriting means (1802) through a control means (1805), and the dummy data in the first code sequence is rewritten to the second code sequence. For trial viewing/listening, the signal (802) from the code sequence decomposing means (1801) is selected by means of a selector switch (1808). When the second code sequence (806) is acquired by purchase of a content, the signal (803) from the code sequence rewriting means (1802) is selected. Thus trial viewing/listening of a content such as music free of danger of decryption is possible, and a high-quality content can be reproduced by acquiring a relatively small amount of data.
Abstract:
Input signals are converted into frequency components, the frequency components are separated into first signals composed of tonal components and second signals composed of other components. The first and second signals are respectively encoded and code string to be transmitted or recorded is generated. For example, only the first signals are encoded, and an information substring generated for, e.g., every quantization accuracy information which has a common value and a parameter related to the encoding of the first signals is included in the code string, by using at least either, for example, the number of spectrum components constituting the tonal components which is the parameter related to the separation or, for example, quantization accuracy information which is the parameter related to the encoding of the first signals as a reference parameter. Thus, the inputted signals can be encoded more efficiently than conventional.
Abstract:
Time-series sampled data of the input signal are grouped (SO1); the data are converted into spectrum data by MDCT (SO2); in order to divide the spectrum data into units, a code table is selected for every unit (SO6); the spectrum data are encoded by quantization using the code tables (SO8); and the encoded spectrum data, the identification signals of the code tables, the normalization coefficients and the number of steps of quantization are outputted (SO9). It is also possible to select a code table for every frame. Thus, the hardware scale of the encoder can be effectively reduced and the encoding efficiency of the encoder can be effectively improved.
Abstract:
This method comprises the steps of converting the inputted acoustic signals into the frequency component; separating the output of conversion means to the tone component and the other component (noise component); coding the tone component; coding the noise component; and generating the code string from the output of the coding step.
Abstract:
In a method and device for signal encoding according to this invention, a first encoder (124) separately encodes noise components of a plurality of channels and a second encoder (125) commonly encodes noise components of a plurality of channels. Then a discriminating means (123) detects the characteristics of the noise components and selects the output from either of the encoders (124 and 125) based on the discriminated results. When the noise components are commonly encoded, therefore, the compression ratio of the noise components of a plurality of channels can be increased. When the components are not commonly encoded, on the other hand, undesired effects of common encoding can be prevented.