Abstract: The present invention relates to improvements of predictive encoding/decoding operations performed on a signal which is transmitted over a packet switched network. The signal is encoded on a block by block basis in such way that a block A-B is predictive encoded independently of any preceding blocks. A start state (715) located somewhere between the end boundaries A and B of the block is encoded using any applicable coding method. Both block parts surrounding the start state is then predictive encoded based on the start state and in opposite directions with respect to each other, thereby resulting in a full encoded representation (745) of the block A-B. At the decoding end, corresponding decoding operations are performed.

Abstract: The disclosure relates to encoding and decoding of digital data, and in particular to lossless arithmetic encoding and decoding of digital data representing audio, image or video data. A probability density function used for lossless arithmetic encoding of digital data is controlled by employing one or more parameters that changes over the set of data to be encoded. A parametric model in the form of an envelope function describes the spread of quantization indices derived from the data in a transform domain. By transmitting the one or more parameters together with the arithmetically encoded data, a receiving decoder may decode the data by exploiting the same parametric model as used by the encoder.

Abstract: The invention relates to a method and a receiver having control logic means for determining a target packet level of a jitter buffer adapted to receive packets with digitized signal samples, which packets are subject to delay jitter, from a packet data network. According to the invention, the jitter buffer is made adaptive to current network conditions, i.e., the nature and magnitude of the jitter observed by the receiver, by collecting statistical measures that describe these conditions. The target buffer level is determined with regard to the effect of packet losses in terms of duration of the discontinued playback of the true signal. This effect is derived from statistical measures of the network conditions as perceived by the receiving side and as reflected by a probability mass function which is continuously updated with packet inter-arrival times. The target buffer level is the result of minimization of a cost function which weights the internal buffer delay and an expected length of buffer underflow.

Abstract: A closed-loop voice-over-Internet-Protocol (VoIP) system has a local and a remote VOIP application. Each VOIP application monitors incoming packet arrival times and durations of audio data in the incoming packets to estimate bandwidth. The bandwidth estimates are forwarded to the other VOIP application. The forwarded bandwidth estimates are compared to a sending bandwidth. When the bandwidth estimate is above the sending bandwidth, compression and audio-frame decimation are reduced to improve voice quality. When the bandwidth estimate falls below the sending bandwidth, audio compression and decimation are increased to improve efficiency. Packet size can also be increased. Congestion estimates can also be sent with the audio data, causing packet transmission to pause until congestion ends. Incoming packet latencies are compared to a moving average to determine the congestion estimate, while bandwidth estimates are made by comparing packet audio duration to time between packet arrivals.

Abstract: The present invention relates to the decoding/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: According to the invention, a method for processing an audio media stream that originates from a packet communication network is disclosed. In one step, packets are received as they arrive from the packet communication network. The packets are part of the audio media stream. A playout buffer in a media playout device is monitored. It is determined that the playout buffer is filled below a threshold. A portion of the audio media stream is retrieved when the playout buffer is filled below the threshold. The portion is stored in the playout buffer of the media playout device.

Abstract: The invention relates to methods and apparatuses for encoding and decoding of a video sequence. In connection with encoding/decoding a video sequence it is desirable to increase the video quality without having to increase the bit-rate for the encoded video too much, thereby still providing a bit-efficient representation of the video. If multiple descriptions of the video sequence is used the invention improves the video quality without any increase of the bit-rate. According to the invention, this is achieved by using two or more coding units for encoding the same video sequence, wherein the encoding units perform their encoding operations displaced in time in relation to each other. Correspondingly, two or more decoding units are used for decoding the same video sequence, wherein the decoding units perform their decoding operations displaced in time in relation to each other.

Abstract: A Voice-over-Internet-Protocol (VoIP) system includes clients that make management connections to an external manager server using web-browsing protocols. A discovery process tests a variety of transport methods by sending packets using different protocols to several ports of the external manager. Some packets may be blocked by firewalls, while other packets are allowed to pass through. Successful packet replies indicate transport capabilities of the client. When a client makes a VoIP call to another client, the external manager compares transport capabilities of the two clients and selects a transport capability that is common to both clients. Transport methods include direct client-to-client connections using TCP or UDP to an arbitrary port or well-known port 80 or 443, and indirect using a voice-proxy server that each client connects to using web standards such as HTTP or SSL. When a client can make only outgoing connections, that client initiates the connection.

Abstract: The present invention relates to a cancellation of echoes in telecommunications systems, more specifically it relates to adaptive alignment of a linear filter (500) used for echo cancellation. According to the invention, it is continuously determined, by means of control logic (520), if a reflection replica delay included in an echo replica signal (110), which delay is provided by a signal buffer (510), should be attempted to be increased or not. Similarly, it is continuously determined if the reflection replica delay should be attempted to be decreased or not. In this way it is possible to provide a delay of the reflection replica which corresponds to the pure delay of a corresponding reflection included in an echo signal (120) received over an echo path. The invention is advantageous since the filter (500) will continuously and quickly adapt to changes in the echo path delay by continuously increasing or decreasing, in an incremental and smooth manner, a present replica delay.

Abstract: The present invention relates to the decoding-/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.