摘要:
A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve.
摘要:
A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve.
摘要:
A hybrid audio encoding technique incorporates both ABR, or CBR, and VBR encoding modes. For each audio coding block, after a VBR quantization loop meets the NMR target, a second quantization loop might be called to adaptively control the final bitrate. That is, if the NMR-based quantization loop results in a bitrate that is not within a specified range, then a bitrate-based CBR or ABR quantization loop determines a final bitrate that is within the range and is adaptively determined based on the encoding difficulty of the audio data. Excessive bitrates from use of conventional VBR mode are eliminated, while still providing much more constant perceptual sound quality than use of conventional CBR mode can achieve.
摘要:
Methods, systems, and apparatus are presented for decoding an audio signal that includes bandwidth extension data. An audio signal that includes core audio data and bandwidth extension data can be received in a decoder. The core audio data can be associated with a core portion of an audio signal, such as the frequency range below a cutoff frequency, and the bandwidth extension data can be associated with an extended portion of the audio signal, such as a frequency range above the cutoff frequency. The core audio data can be decoded to generate a decoded core audio signal in a time domain representation. Further, an extended portion of the audio signal can be reconstructed in accordance with extension data and decoded core audio signal. Additionally, the decoded core audio signal can be lowpass filtered and the extended portion can be highpass filtered before being combined to generate a decoded output signal.
摘要:
A system and method of retrieving a watermark in a watermarked signal are disclosed. The watermarked signal comprises odd and even overlapped blocks where the watermark is contained in the even blocks. The method comprises, for each k-th even block, subtracting the two adjacent odd numbered blocks from the k-th even block of the watermarked signal to retrieve s *k(n), transforming s *k(n) into the frequency domain to generate S k(f), calculating a phase of S k(f) as φ (f) and a phase of Sk(f) as φ(f), calculating the difference Ψ (f) between φ (f) and φ(f), unwrapping Ψ (f) to obtain the phase modulation {tilde over (Φ)} k(f), and using a Viterbi search to retrieve the watermark embedded in {tilde over (Φ)} k(f).
摘要:
A system, method and computer-readable medium are disclosed for using filters signal processing. The system includes a module that calculates a filter for each of a plurality of frequency bands, a module that groups the filters into a plurality of groups, a module that determines a representative filter for each group of the plurality of groups and a module that uses the representative filter of each group for frequency bands of the each group. The filters are temporal noise shaping filters (TNS) filters.
摘要:
A system and method of generating a watermarked signal are disclosed. The system segments the signal into overlapping blocks using a window function and processes the overlapping blocks according to whether each block is odd- or even-numbered. The system windows the odd-numbered blocks, modulates the phase of each block in the frequency domain, transforms each modulated block in the time domain, windows each block transformed into the time domain and overlap-adds each odd-numbered block with each even-numbered block to generate the watermarked signal.
摘要:
The present invention provides a method and apparatus for enhancing and recognizing connected and degraded text. The enhancement process comprises filtering a scanned image to determine whether a binary image value of an image pixel should be complemented, determining whether complementing the value of the pixel reduces the sharpness of wedge-like figures in the image, and complementing the binary value of the pixel when doing so does not reduce sharpness. The recognition process may comprise determining primitive strokes in a scanned image, segmenting the scanned image into sub-character segments based on the primitive strokes, identifying features which characterize the sub-character segments, and comparing identified features to stochastic models of known characters and determining an optimum sequence of known characters based on the comparisons through the use of Viterbi scoring and level building procedures.
摘要:
According to one embodiment, an improved audio coding technique encodes audio having a low frequency transient signal, using a long block, but with a set of adapted masking thresholds. Upon identifying an audio window that contains a low frequency transient signal, masking thresholds for the long block may be calculated as usual. A set of masking thresholds calculated for the 8 short blocks corresponding to the long block are calculated. The masking thresholds for low frequency critical bands are adapted based on the thresholds calculated for the short blocks, and the resulting adapted masking thresholds are used to encode the long block of audio data. The result is encoded audio with rich harmonic content and negligible coder noise resulting from the low frequency transient signal.
摘要:
System, method and computer-readable medium are disclosed for using filters signal processing. The system includes a module that receives information regarding a first filter, a module that receives information regarding a second filter, and a module that receives date to indicate switching between the first filter and the second filter across the spectrum of the received audio signal, and a module that processes the received audio signal according to the received data and switching between the first filter and the second filter, wherein at least one of the first filter and the second filter represent a merger of two initial filters.