Texas Instruments Corporation

John Thomas Dowdal

Dunling Li

Sachin Adlakha

Zoran Mladenovic

Gokhan Sisli

Gaurav Agarwal

Qin Su

A device manages data for a digital signal processor. The device includes an external random access memory (RAM), configured to store channel specific data for plural different channels; and a microprocessor, in communication with the external RAM. When receiving the packet, in a transport layer processing the packet, the microprocessor determines a channel of the plural different channels corresponding to an indication in a transport layer header of the received packet. The microprocessor fetches channel specific data specific to the channel into an internal memory internal to the microprocessor from the external RAM, by the transport layer, before the packet is passed to an application layer, thereby avoiding a wait for reading the packet at the application layer.

External memory data management with data regrouping and channel look ahead

A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. A voice playout unit in the receiver shares its nominal delay buffer with a history buffer of a packet loss concealment algorithm up to a maximum limit described in a standard. This reduces or eliminates need to allocate memory for the history buffer. A history buffer can also be extended to retain an original portion of voice signal packets received prior to a packet loss as well as generated voice signals as they are generated. A scratch buffer is used as a working buffer and replaces the function of a pitch buffer.

Memory optimization packet loss concealment in a voice over packet network

An overflow problem of LSF quantization in G.729 Annex B speech encoding which may lead to non-assignment of a codebook index. Preferred embodiments fix the problem with default or limited random variable assignments or flagging the overflow and adjusting the frame encoding such as by limiting spectral components or changing quantization targets.

Speech coder and method

In order to reduce distortion in playout of audio received in a packet over a packet network, the playout unit needs to determine the relative delay of adjacent packets, jitter, present in the network. The jitter is used to determine buffering for smoothing playout of audio in a packet network. The jitter in packets received from a packet network is calculated based upon the arrival time of a packet, the length of the packet and the arrival time of a subsequent packet. The receiving gateway notes the arrival time of a packet. The length of the audio payload is determined from the size of the payload and the codec used to encode the payload. The length of the payload is added to the arrival time to determine the anticipated arrival of the next subsequent packet. The actual arrival time is noted and the difference between anticipated arrival and actual arrival is used to determine jitter. The present invention is protocol independent because jitter is determined without reference to any time stamp information which may be or may not be contained in a header for certain protocols. The present invention can be implemented in a wider variety of environments because jitter can be calculated without reference to the network packet time stamp.

Adaptive playout of digital packet audio with packet format independent jitter removal

A system and method to measure the clock skew between transmitting and receiving devices operating with independent clock sources over a packet network is described. To provide adaptive playout in an IP telephony device without a sequencing scheme in the packets, the clock skew is measured and recorded. Using a PCM resampler that is implemented with an interpolation filter bank of FIR subfilters, the change in depth of the playout buffer during transmission is analyzed, and this change infers the clock rate associated with the transmission.

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Gaithersburg, MD, United States of America

John Thomas Dowdal