The present disclosure relates to encoding apparatuses, encoding methods, and programs, and particularly relates to an encoding apparatus, an encoding method, and a program which are capable of accurately encoding an audio signal including noise in a certain band.
In general, examples of a method for encoding an audio signal include a method for performing normalization and quantization on frequency spectra obtained by performing time-frequency transform on an audio signal (refer to Japanese Unexamined Patent Application Publication No. 2006-11170, for example).
FIG. 1 is a block diagram illustrating a configuration of an audio encoding apparatus which performs encoding in such an encoding method.
An audio encoding apparatus 10 shown in FIG. 1 includes a time-frequency transform unit 11, a normalization unit 12, a bit allocation calculation unit 13, a quantization unit 14, and a code-string encoder 15. The audio encoding apparatus 10 encodes an audio signal input as a time-series signal and outputs a code string.
Specifically, the time-frequency transform unit 11 included in the audio encoding apparatus 10 performs time-frequency transform on an audio signal input as a time-series signal and outputs frequency spectra mdspec. For example, the time-frequency transform unit 11 performs time-frequency transform on a time-series signal of 2N samples using orthogonal transform such as MDCT (Modified Discrete Cosine Transform) and outputs N MDCT coefficients obtained as a result of the time-frequency transform as the frequency spectra mdspec.
The normalization unit 12 performs normalization on the frequency spectra mdspec supplied from the time-frequency transform unit 11 for each predetermined processing unit using normalization coefficients obtained in accordance with amplitudes of the frequency spectra mdspec. The normalization unit 12 outputs normalization information idsf which is information on integer numbers corresponding to the normalization coefficients and normalization frequency spectra nspec obtained by normalizing the frequency spectra mdspec.
The bit allocation calculation unit 13 performs bit allocation calculation such that the numbers of bits to be allocated to the normalization frequency spectra nspec are calculated for each predetermined processing unit in accordance with the normalization information idsf supplied from the normalization unit 12 so as to output quantization information idwl representing the numbers of bits. Furthermore, the bit allocation calculation unit 13 outputs the normalization information idsf supplied from the normalization unit 12.
The quantization unit 14 quantizes the normalization frequency spectra nspec supplied from the normalization unit 12 in accordance with the quantization information idwl supplied from the bit allocation calculation unit 13. Specifically, the quantization unit 14 quantizes the normalization frequency spectra nspec for each predetermined processing unit using quantization coefficients corresponding to the quantization information idwl. The quantization unit 14 outputs a quantization frequency spectra qspec as a result of the quantization.
The code-string encoder 15 encodes the normalization information idsf and the quantization information idwl which are supplied from the bit allocation calculation unit 13 and the frequency spectra qspec supplied from the quantization unit 14 and outputs a code string obtained as a result of the encoding. The output code string may be transmitted to another apparatus or may be recorded in a certain recording medium.
Furthermore, in recent years, an audio signal processed by audio encoding apparatuses is expanded from a PCM (Pulse Code Modulation) signal of a frequency of 44.1 kHz and a PCM word length of 16 bits and a PCM signal of a frequency of 48 kHz and a PCM word length of 16 bits to a PCM signal having high-quality multi bits such as a PCM signal of a frequency of 96 kHz and a PCM word length of 24 bits and a PCM signal of a frequency of 192 kHz and a PCM word length of 24 bits.
Such a high-quality multi-bit PCM signal is not generated as a multi-bit PCM signal from the beginning but is generated using a PDM (Pulse Density Modulation) signal such as a DSD (Direct Stream Digital) signal as a source in many cases.
This is because, in a field of an A/D converter used to convert an analog audio signal into a digital audio signal, a replacement of a successive-approximation A/D converter by a delta-sigma A/D converter has been rapidly progressed.
More specifically, a general successive-approximation A/D converter may directly generate a multi-bit PCM signal but conversion accuracy is considerably restricted by element accuracy. Therefore, when a PCM word length is equal to or larger than 24 bits, it is difficult to ensure linearity of the A/D conversion. On the other hand, in a delta-sigma A/D converter, A/D conversion is easily performed with high accuracy using a single threshold value. In view of such a background, as an A/D converter, the delta-sigma A/D converter has been widely used instead of the general successive-approximation A/D converter.
FIG. 2 is a diagram illustrating an input signal and an output signal of an 1-bit delta-sigma A/D converter. As shown in FIG. 2, in the 1-bit delta-sigma A/D converter, an analog audio signal serving as an input signal is converted into a 1-bit PDM signal which has amplitude represented by time density of +1 and which serves as an output signal.
FIG. 3 is a diagram illustrating quantization noise in the delta-sigma A/D converter. As shown in FIG. 3, first, in the delta-sigma A/D converter, the quantization noise included in an audio band (0 to fs/2 in the example shown in FIG. 3) is dispersed in a wide band (0 to nfs/2 in the example shown in FIG. 3) by performing oversampling. Next, the quantization noise is shifted out of the audio band by performing noise shaping. Accordingly, the delta-sigma A/D converter may realize a high S/N (signal/noise) ratio in the audio band.
As described above, when a source of a high-quality multi-bit PCM signal is a PDM signal obtained by the delta-sigma A/D converter, the multi-bit PCM signal is generated by performing a LPF (Low Pass Filter) process on the PDM signal.
The multi-bit PCM signal obtained as described above is represented as a delta-sigma type A as shown in FIG. 4. This quantization noise is undesired noise for the multi-bit PCM signal.
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