46 results about "Advanced Audio Coding" patented technology

An audio coding method and apparatus capable of improving efficiency of a MPEG-4 AAC (Moving Picture Expert Group-4 Advanced Audio Coding) process are disclosed. The audio coding method and apparatus reduce the number of calculations of an audio coding algorithm to improve efficiency of an audio coding process. Specifically, the audio coding method and apparatus reduce the number of calculations required for a Psychoacoustic model process of the MPEG-4 AAC algorithm capable of coding an audio signal.

A system provides lossless split and merge processes of integer discrete cosine transform (DCT) transformed data such that the discrete cosine transform of one data block may be split into two half length DCT odd and even blocks for merging, with split and merge processes being lossless and are generated in the discrete cosine transformed domain. After splitting, the redundancy existing between the two integer discrete cosine transformed half data blocks allows one to approximately reconstruct the original data block in case one of the discrete cosine transformed half data block is lost during transmission. The system may be used with existing JPEG and MPEG compressors and decompressors because both use the discrete cosine transform for image and video compression and decompression, may be used as a resolution conversion device for transcribing from digital high-definition TV to analog low-definition TV, and may be used for lossless splitting and merging type-IV discrete cosine transformed data for audio compression and decompression in the international standard MPEG-4 Advanced Audio Coding (AAC), such as AC-3 or MP3.

In the MPEG2 Advanced Audio Coder (AAC) standard, Temporal Noise Shaping (TNS) is currently implemented by defining one filter for a given frequency band, and then switching to another filter for the adjacent frequency band when the signal structure in the adjacent band is different than the one in the previous band. The AAC standard limits the number of filters used to either one filter for a “short” block or three filters for a “long” block. In cases where the need for additional filters is present but the limit of permissible filters has been reached, the remaining frequency spectra are simply not covered by TNS. This current practice is not an effective way of deploying TNS filters for most audio signals. We propose two solutions to deploy TNS filters in order to get the entire spectrum of the signal into TNS. The first method involves a filter bridging technique and complies with the current AAC standard. The second method involves a filter clustering technique. Although the second method is both more efficient and accurate in capturing the temporal structure of the time signal, it is not AAC standard compliant. Thus, a new syntax for packing filter information derived using the second method for transmission to a receiver is also outlined.

The present invention relates to the computing method and Huffman computing circuits for improving the correctness and efficiency of the nonlinear inverse quantization when the MPEG-2 AAC (Advanced Audio Coding) or MPEG-4 AAC algorithm which is used as an audio compression algorithm in multi-channel high-quality audio systems is implemented on programmable processors. In accordance with the present invention, while the architecture of the existing digital signal processor is reused, the performance can be improved by means of the addition of Huffman decoder and bit processing architecture. Accordingly, to design and change the programmable processor can be facilitated.

A system provides lossless split and merge processes of integer discrete cosine transform (DCT) transformed data such that the discrete cosine transform of one data block may be split into two half length DCT odd and even blocks for merging, with split and merge processes being lossless and are generated in the discrete cosine transformed domain. After splitting, the redundancy existing between the two integer discrete cosine transformed half data blocks allows one to approximately reconstruct the original data block in case one of the discrete cosine transformed half data block is lost during transmission. The system may be used with existing JPEG and MPEG compressors and decompressors because both use the discrete cosine transform for image and video compression and decompression, may be used as a resolution conversion device for transcribing from digital high-definition TV to analog low-definition TV, and may be used for lossless splitting and merging type-IV discrete cosine transformed data for audio compression and decompression in the international standard MPEG-4 Advanced Audio Coding (AAC), such as AC-3 or MP3.

An audio coding method and apparatus capable of improving efficiency of a MPEG-4 AAC (Moving Picture Expert Group-4 Advanced Audio Coding) process are disclosed. The audio coding method and apparatus reduce the number of calculations of an audio coding algorithm to improve efficiency of an audio coding process. Specifically, the audio coding method and apparatus reduce the number of calculations required for a Psychoacoustic model process of the MPEG-4 AAC algorithm capable of coding an audio signal.

The invention discloses a coding parameter statistical feature-based AAC (advanced audio coding) sound recording document source identification method. According to the design thought of the invention, the use features and statistical features of a plurality of coding parameters of AAC sound recording documents of smart phones are analyzed; the tendencies and characteristics of using the coding parameters when the mobile phones generate the AAC sound recording documents are found out; and features for distinguishing phone models are constructed; and therefore, accurate identification of the sources of the AAC sound recording documents can be realized. The AAC sound recording document source identification method has the advantages of high recognition accuracy, convenience in operation and the like.

The techniques described are utilized for detection of noise and noise-like segments in audio coding. The techniques can include performing a prediction gain calculation, an energy compaction calculation, and a mean and variation energy calculation. Signal adaptive noise decisions can be made both in time and frequency dimensions. The techniques can be embodied as part of an AAC (advanced audio coding) encoder to detect noise and noise-like spectral bands. This detected information is transmitted in a bitstream using a signaling method defined for a perceptual noise substitution (PNS) encoding tool of the AAC encoder.

The present invention discloses an AAC (Advanced Audio Coding) audio double compression detection method based on a QMDCT (Quantized Modified Discrete Cosine Transform) coefficient. The method comprises the steps of: obtaining single compression AAC audio and a dual compression AAC audio with different bit rate; removing sampling points to obtain single compression removal sampling point AAC audios and dual compression removal sampling point AAC audios; obtaining a corresponding feature vector according to QMDCT coefficient distribution histograms of the single compression AAC audio and the corresponding single compression removal sampling point AAC audios; and obtaining a corresponding feature vector according to QMDCT coefficient distribution histograms of the dual compression AAC audioand the corresponding dual compression removal sampling point AAC audios; training a LIBSVM classifier according to the feature vectors of the single compression AAC audios and the dual compression AAC audios; and when test is performing, inputting the bit rate of the AAC audio to be subjected to dual compression detection to a trained corresponding LIBSVM classifier to obtain a detection result.The AAC audio double compression detection method based on the QMDCT coefficient achieves effective detection of the AAC audios of the dual compression from low code rate to high code rate and dual compression with the same code rate, and is high in detection accuracy, low in computing complexity and high in robustness.

The invention discloses an audio processing method in mobile digital television recording. The method comprises the following steps of: packaging each advanced audio coding + (AAC+) code stream naked data frame which is received from the air into a frame in an audio data transport stream (ADTS) format; analyzing an ADTS frame head to find the original AAC+ code stream naked data of the frame; decoding the AAC+ code stream naked data, removing spectral band replication (SBR) information from the AAC+ code stream naked data in an audio decoding, recording and buffering area, and converting the AAC+ code stream naked data in the audio decoding, recording and buffering area into AAC naked data; and performing a subsequent decoding process by using an AAC+ decoder, and transmitting the AAC naked data to a recording module to record video and audio files. By the method, the mobile digital television recording also can be performed even if the signal is poor, and a recorded video file can benormally played on a computer by mainstream video play software, and severe abnormal phenomena such as harsh noise, silence, player breakdown, system crash and the like are avoided.

The invention brings forward an audio coding and decoding system. An advanced audio coding (AAC) and decoding algorithm is adopted by the audio coding and decoding system at the same audio coding and decoding unit (DSP). Meanwhile, high-fidelity audio coding and decoding are realized so that resource consumption is substantially reduced, and product power consumption, volume and weight are well controlled.