摘要:
The present technology relates to an encoding device and method, a decoding device and method, and a program, which enable improvement of audio quality.A QMF sub-band power calculation unit calculates power of a QMF sub-band signal of a high frequency QMF sub-band among a plurality of the QMF sub-bands constituting an input signal. A high frequency sub-band power calculation unit carries out an operation to weight more a QMF sub-band power having larger power as for a sub-band including a number of the high frequency QMF sub-bands to calculate high frequency sub-band power of the sub-band. The multiplexing circuit multiplexes high frequency encoded data and low frequency encoded data for outputting. The high frequency encoded data is selected based on the high frequency sub-band power and obtained by encoding information used for obtaining a high frequency component of the input signal by estimating, and the low frequency encoded data is obtained by encoding low frequency components of the input signal. The present technology can be applied to encoding devices.
摘要:
The present invention relates to a control device, a control method, and a program capable of improving operability with a simpler configuration.A reproducing apparatus 11 which reproduces sound such as music includes earphones 21 which are worn by a user on the ears and a body 22. A sound pickup unit 31 collects sounds around the reproducing apparatus 11, and a determination unit 34 extracts future amounts from the collected sound and determines whether the sound corresponds to an operation sound generated when the sound pickup unit 31 is directly tapped by the user. A controller 35 executes a process in accordance with a result of the determination performed by the determination unit 34. For example, when a sound pickup unit 31-1 is tapped once within a predetermined period of time, the controller 35 instructs a reproduction controller 39 to stop reproduction of music. In the reproducing apparatus 11, various function control processes may be performed by a simple operation of tapping the sound pickup unit 31 and buttons are not required. The present invention is applicable to a music player.
摘要:
The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis.An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.
摘要:
The present technique relates to a sound processing apparatus, a method, and a program capable of alleviating degradation of the quality of sound in a case where the gain of a sound signal is amplified.When equalizer processing for adjusting the gain of each frequency band of an input signal on the basis of a gain setting value is performed, an input signal is attenuated by an input attenuation amount derived from the gain setting value, and the equalizer processing is performed on the input signal attenuated. The amount of amplification of the gain of the input signal in the equalizer processing is estimated on the basis of the gain setting value and a weight coefficient of each frequency band derived from a generally-available music signal prepared in advance, and a difference of the estimation value and the input attenuation amount is calculated as a gain correction amount. Further, nonlinear amplification processing is performed on the input signal so as to actually amplify the input signal, which has been subjected to the equalizer processing, by a gain correction amount, and an output signal is obtained. The present technique can be applied to a sound processing apparatus.
摘要:
The present technology relates to an encoding device and method, a decoding device and method, and a program, which enable improvement of audio quality.A QMF sub-band power calculation unit calculates power of a QMF sub-band signal of a high frequency QMF sub-band among a plurality of the QMF sub-bands constituting an input signal. A high frequency sub-band power calculation unit carries out an operation to weight more a QMF sub-band power having larger power as for a sub-band including a number of the high frequency QMF sub-bands to calculate high frequency sub-band power of the sub-band. The multiplexing circuit multiplexes high frequency encoded data and low frequency encoded data for outputting. The high frequency encoded data is selected based on the high frequency sub-band power and obtained by encoding information used for obtaining a high frequency component of the input signal by estimating, and the low frequency encoded data is obtained by encoding low frequency components of the input signal. The present technology can be applied to encoding devices.
摘要:
The present technology relates to an encoding device and an encoding method, a decoding device and a decoding method, and a program, configured to obtain a high quality audio with less encoding amount. A number-of-sections determining feature amount calculating circuit calculates a number-of-sections determining feature amount for determining the number of divisions to divide a process target section into continuous frame sections each including a frame for which the same estimation coefficient is selected, based on sub-band signals of a plurality of sub-bands constituting an input signal. A quasi-high frequency sub-band power difference calculating circuit determines the number of continuous frame sections in the process target section based on the number-of-sections determining feature amount, selects an estimation coefficient for obtaining a high frequency component of the input signal by estimation for each continuous frame section, and generates data including a coefficient index for obtaining the estimation coefficient. A high frequency encoding circuit encodes the obtained data, and generates high frequency encoded data. The present technology can be applied to an encoding device.
摘要:
The present technology relates to a signal processing device, method, and program that may obtain audio at a higher audio quality when decoding an audio signal.An envelope information generating unit 24 generates envelope information representing an envelope form of high frequency components of an audio signal to be encoded. A sine wave information generating unit 26 extracts a sine wave signal from the high frequency components of the audio signal, and generates a sine wave information representing an emergence start position of the sine wave signal. An encoding stream generating unit 27 multiplexes the envelope information, the sine wave information, and low frequency components of the audio signal that have been encoded, and outputs an encoding stream obtained as the result. As a result, the high frequency components included in the sine wave signal may be predicted at a higher accuracy from the envelope information and the sine wave information at the receiving side of the encoding stream. The present invention may be applied to a signal processing device.
摘要:
An audio signal processing apparatus which includes an input analysis unit which analyses the characteristics of an input signal and generates an input sound feature value; an environment analysis unit which analyses the characteristics of the environmental sound and generates an environmental sound feature value; a mapping control information generation unit which generates mapping control information as control information of amplitude conversion processing to the input signal by application of the input sound feature value and the environmental sound feature value; and a mapping process unit which performs amplitude conversion on the input signal based on a linear or non-linear mapping function determined according to the mapping control information and generates an output signal.
摘要:
Digital data divided into blocks each having a predetermined number of samples is transformed into data on a frequency axis for each block to generate coefficient data for each frequency. Coefficient data of a predetermined number of blocks are stored in a buffer. From the buffer, coefficient data are inputted to a floating-point transforming circuit for each one block. The coefficient data are divided into a plurality of sub-bands, each sub-band including one or a plurality of coefficient data. The coefficient data are floating-point transformed for each sub-band and transformed into one sub-band common characteristic data which is common to the coefficient data included in each sub-band and mantissa data of the number equal to the number of coefficient data included in each sub-band. The sub-band common characteristic data and the mantissa data are stored in a memory. Sub-band common characteristic data of a predetermined number of blocks are read from the memory for each sub-band. One block-common characteristic data which is common to the read sub-band common characteristic data is generated. Coefficient data of a predetermined number of blocks are read from the memory for each sub-band. The read coefficient data is divided by an exponent expressed by the block-common characteristic data to generate new mantissa data.