Abstract: A packet-based or cell-based data communication system is capable of determining whether a new call is a modem call or a voice call before a modem connection is established between the two end devices. The network switches are configured to monitor for distinctive facsimile modem and data modem calling and answer signals. In response to the detection of such calling or answer signals, the data communication system selects a speech coding scheme and/or a data transmission technique for use in connection with the current call, while a conventional 64 kbps PCM transmission scheme is maintained for communications between the network and the end devices. A speech coding protocol, such as ADPCM, may be enabled for voice calls, thus conserving network bandwidth. The 64 kbps PCM scheme may be employed to facilitate high speed modem transmissions.

Abstract: An apparatus and method for issue grouping of instructions in a VLIW processor is disclosed. There can be one, two, or three issue groups (but no greater than three issue groups) in each VLIW packet. In one embodiment, a template in the VLIW packet comprises two issue group end markers where each issue group end marker comprises three bits. The three bits in the first issue group end marker identifies the instruction which is the last instruction in the first issue group. Likewise, the three bits in the second issue group end marker identifies the instruction which is the last instruction in the second issue group. Any instructions in the VLIW packet falling outside the two expressly defined first and second issue groups are placed in a third issue group. As such, three issue groups can be identified by use of the two issue group end markers. In one embodiment, the template of the VLIW packet includes a chaining bit.

Abstract: A speech-coding device includes a fixed codebook, an adaptive codebook, a short-term enhancement circuit, and a summing circuit. The short-term enhancement circuit connects an output of the fixed codebook to a summing circuit. The summing circuit adds an adaptive codebook contribution to a fixed codebook contribution. The short-term enhancement circuit can also be connected to a synthesis filter to emphasize the spectral formants in an encoder and a decoder.

Abstract: A circuit for monitoring and controlling input to a VCO, comprises: (A) a monitoring sub-circuit for monitoring input; (B) a stepping sub-circuit for stepping the VCO's frequency range; (C) an input reduction sub-circuit for reducing input into the monitoring sub-circuit; and (D) a VCO.

Abstract: An exemplary decoder comprises a receiver that receives parameters of a speech signal on a frame-by-frame basis, a control logic for decoding parameters and for resynthesizing the speech signal, the control logic including a minimum spacing indicative of a minimum difference required between LSFs of consecutive frames, a frame recovery logic that, when a lost frame detector detects a lost frame, sets the minimum spacing for the lost frame to a first value which is greater than the minimum spacing for the previously received frame, and/or uses pitch lag parameters of a plurality of previously received frames to extrapolate a pitch lag parameter for the lost frame, and/or sets gain parameter of a subframe of the lost frame in a first manner if the lost gain parameter is an adaptive codebook gain parameter and in a second manner if the lost gain parameter is a fixed codebook gain parameter.

Abstract: An extended signal coding system that accommodates substantially music-like signals within a signal while maintaining a high perceptual quality in a reproduced signal during discontinued transmission (DTX) operation. The extended signal coding system contains internal circuitry that performs detection and classification of the speech signal, depending on numerous characteristics of the signal, to ensure the high perceptual quality in the reproduced signal. In certain embodiments of the invention, the signal is a speech signal, and the speech signal has a substantially music-like signal contained therein, and the extended signal coding system overrides any voice activity detection (VAD) decision that is used to determine which among a plurality of source coding modes are to be employed using a voice activity detection (VAD) correction/supervision circuitry. This is particularly relevant for discontinued transmission (DTX) operation.

Abstract: In a cellular telephone system where a digital cellular telephone is connected to a regular telephone through the public switched telephone network (PSTN), a speech encoder/decoder is used with an A/&mgr;-Law encoder/decoder causing annoying audible noise at very low levels because of the quantization characteristics of the A/&mgr;-Law encoder/decoder. This noise is eliminated by adding a digital constant to the output of the speech coder, shifting the low level signal away from zero. The resulting DC level added to the speech signal is inaudible to the PSTN telephone user and does not degrade speech quality. Alternatively, the constant added to the output of the speech coder is confined to a small value added to the speech coder output to move the entire speech coder output during the silence period, between speech periods, above zero or below zero.

Abstract: In a speech communications network, continuous play of audio packets is achieved using a jitter buffer in a receiver. Audio packets are stored in the jitter buffer before decoding the audio packets into an audible output. When the level of stored audio packets approaches the full capacity of the jitter buffer, the rate at which the audio packets are played out of the jitter buffer is increased signaling a compression operation in the decoder. When the level of stored audio packets approaches an empty level of the jitter buffer, the rate which the audio packets are played out of the jitter buffer is reduced signaling an expansion operation in the decoder. Audio packets are not modified when the level of stored audio packets is within a predetermined range. A speed controller is provided to instruct the decoder to decode the audio packets according to either a compressed, expanded or normal audio packet status.