摘要:
An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.
摘要:
A person is provided with the ability to auditorily determine the spatial geometry of his current physical environment. A spatial map of the current physical environment of the person is generated. The spatial map is then used to generate a spatialized audio representation of the environment. The spatialized audio representation is then output to a stereo listening device which is being worn by the person.
摘要:
A person is provided with the ability to auditorily determine the spatial geometry of his current physical environment. A spatial map of the current physical environment of the person is generated. The spatial map is then used to generate a spatialized audio representation of the environment. The spatialized audio representation is then output to a stereo listening device which is being worn by the person.
摘要:
A “speech onset detector” provides a variable length frame buffer in combination with either variable transmission rate or temporal speech compression for buffered signal frames. The variable length buffer buffers frames that are not clearly identified as either speech or non-speech frames during an initial analysis. Buffering of signal frames continues until a current frame is identified as either speech or non-speech. If the current frame is identified as non-speech, buffered frames are encoded as non-speech frames. However, if the current frame is identified as a speech frame, buffered frames are searched for the actual onset point of the speech. Once that onset point is identified, the signal is either transmitted in a burst, or a time-scale modification of the buffered signal is applied for compressing buffered frames beginning with the frame in which onset point is detected. The compressed frames are then encoded as one or more speech frames.
摘要:
An energy based technique to estimate the positions of people speaking from an ad hoc network of microphones. The present technique does not require accurate synchronization of the microphones. In addition, a technique to normalize the gains of the microphones based on people's speech is presented, which allows aggregation of various audio channels from the ad hoc microphone network into a single stream for audio conferencing. The technique is invariant of the speaker's volumes thus making the system easy to deploy in practice.
摘要:
A media recommendation and sharing technique that employs agents on media players/devices to expand the scope of media sharing scenarios. The technique assists a user in discovering media items, such as, for example, music, recordings, play lists, pictures, video games, on nearby media players or devices (devices which are capable of receiving, storing and playing media) which are interesting to the user. The collaborative media recommendation and sharing technique contemporaneously determines a user's media preferences based on media stored on a pair of media devices and recommends media for potential sharing based on these determined user preferences.
摘要:
An energy based technique to estimate the positions of people speaking from an ad hoc network of microphones. The present technique does not require accurate synchronization of the microphones. In addition, a technique to normalize the gains of the microphones based on people's speech is presented, which allows aggregation of various audio channels from the ad hoc microphone network into a single stream for audio conferencing. The technique is invariant of the speaker's volumes thus making the system easy to deploy in practice.
摘要:
A template and/or knowledge associated with a synchronous meeting are obtained by a computing device. The computing device then adaptively manages the synchronous meeting based at least in part on the template and/or knowledge.
摘要:
Described herein is a method that includes receiving multiple requests for access to an exposed media object, wherein the exposed media object represents a live media stream that is being generated by a media source. The method also includes receiving data associated with each entity that provided a request, and determining, for each entity, whether the entities that provided the request are authorized to access the media stream based at least in part upon the received data and splitting the media stream into multiple media streams, wherein a number of media streams corresponds to a number of authorized entities. The method also includes automatically applying at least one policy to at least one of the split media streams based at least in part upon the received data.
摘要:
The described implementations relate to distributed network management and more particularly to enhancing distributed network utility. One technique selects multiple trees to distribute content to multiple receivers in a session where individual receivers can receive the distributed content at one of a plurality of rates. The technique further adjustably allocates content distribution across the multiple trees to increase a sum of utilities of the multiple receivers.