Abstract: According to the claimed invention, a DSP is disclosed. The DSP comprises a processing unit for processing the data stream, a first memory coupled to the processing unit for storing a destination volume value, and a second memory coupled to the processing unit for storing a time_determining value wherein the processing unit adjusts the volume of the signal stored in the data stream according to the time_determining value such that the adjustment from a current volume value of the signal to the destination volume value is accomplished within a predetermined time.

Abstract: A computing system includes a digital signal processor, a storage medium for storing parameter tables, a central processing unit coupled to the digital signal processor and the storage medium, and a shared memory coupled to the digital signal processor and the central processing unit. The digital signal processor stores a data request in the shared memory and issues a request command to the central processing unit when access to the parameter tables is desired. Upon receipt of the request command, the central processing unit reads the data request in the shared memory, accesses the storage medium for retrieving requested data that corresponds to the data request, stores the requested data in the shared memory, and issues a response command to the digital signal processor. The digital signal processor retrieves the requested data from the shared memory upon receipt of the response command from the central processing unit.

Abstract: In a method and apparatus for processing a microphone audio signal, the microphone audio signal is converted into an initial digital signal at a first sampling frequency. The initial digital signal is stored in a memory at a second sampling frequency lower than the first sampling frequency. The initial digital signal stored in the memory is processed so as to result in a processed digital signal. The processed digital signal is outputted at an output frequency equal to the first sampling frequency. The initial digital signal and the processed digital signal are combined to result in a composite digital signal. Finally, the composite digital signal is converted into an analog processed audio signal output.

Abstract: A method and apparatus are used for accounting for last and out-of-sequence interprocessor messages in a multiprocessor telephone switching system. Each message is assigned a sequence number by the sending processor. Upon reception of messages, the receiving processor analyzes the sequence numbers to ascertain their proper order and if any messages are missing. The identities of missing messages are entered in a table which is periodically updated.