摘要:
According to the present invention, there is developed a proprietary technology for compressing the window tables of audio coders to ⅛ their original size (or less) without any loss of quality. This technology can be applied to all transform based audio coders, or any audio coder that uses a windowing stage. The novel technique for reducing storage requirements for the window tables of audio coders is based on multiple differentiation. Since the difference between any two adjacent samples in the first difference signal is small, so it is more efficient to store this difference. This technique can be carried out several more times, until the returns get smaller, and the computational requirements to “undo” the compression go up. The optimum number of times to differentiate is dependent on the particular application and the window shape.
摘要:
A method for performing audible fast-forward or reverse of audio content represented in a compressed format, such as, but not limited to, MPEG-1 Layer 3 (MP3) or MPEG-2 Advance Audio Coding (AAC) employs a fast-forward controller which performs fast-forward or reverse by repeatedly skipping forward or reverse in the compressed audio data stream, retrieving a block of data, and then splicing these data blocks back together. A decoder is then used to decode each of these blocks, to detect when a block switch has occurred (a splice in the data stream), and to quickly resynchronize at each transition. Hierarchical or multiplexed data streams may be decoded using a cascade of decoders each employing this technique. The decoder uses a robust sync search for performing resynchronization and error recovery.
摘要:
A data processing system (10) uses a microprocessor host (12) coupled to a decoding system (14). A hardware filter arithmetic unit block (32) retrieves decoded information from the arithmetic unit buffer (30) and dequantizes, transforms and filters the data to generate PCM output data which is loaded into a PCM buffer (34). An address circuit forms several addresses from a single value to accesses multiple sources of data and coefficients simultaneously for use by the hardware filter arithmetic unit.
摘要:
A data processing system (10) is disclosed which comprises a microprocessor host (12) coupled to a decoding system (14). A host interface block (18) receives a bit stream and passes bit stream on to a system decoder block (20). The system decoder block (20) extracts the appropriate data from the bit stream and loads an input buffer (24) or an optional external buffer (26). An audio decoder block (28) retrieves the data from the input buffer (24) and generates scale factor indices, bit per code word values and subband samples which are stored in an arithmetic unit buffer (30). A hardware filter arithmetic unit block (32) retrieves the information from the arithmetic unit buffer (30) and dequantizes, transforms and filters the data to generate PCM output data which is loaded into a PCM buffer (34). The data within the PCM buffer (34) is output by a PCM output block (36) to a digital-to-analog converter (16).
摘要:
A data processing system (10) is disclosed which comprises a microprocessor host (12) coupled to a decoding system (14). A host interface block (18) receives a bit stream and passes bit stream on to a system decoder block (20). The system decoder block (20) extracts the appropriate data from the bit stream and loads an input buffer (24) or an optional external buffer (26). An audio decoder block (28) retrieves the data from the input buffer (24) and generates scale factor indices, bit per code word values and subband samples which are stored in an arithmetic unit buffer (30). A hardware filter arithmetic unit block (32) retrieves the information from the arithmetic unit buffer (30) and dequantizes, transforms and filters the data to generate PCM output data which is loaded into a PCM buffer (34). The data within the PCM buffer (34) is output by a PCM output block (36) to a digital-to-analog converter (16).