Nokia Corporation

Nokia Technologies Oy

Jari Selin

Pasi Sakari Ojala

Jussi Virolainen

Ari Lakaniemi

Pekka Pessi

Jukka Saunamaki

Jukka Saunamä Ki

Teleconferencing systems may adapt to changes associated with participating terminals by monitoring characteristics of the terminals and modifying audio streams in accordance with detected changes. A terminal moving from a location shared with one or more other participating terminals may be detected in a variety of ways. In response to the location change, a teleconference server may modify the audio stream transmitted to the terminal so that audio originating from the one or more other participating terminals is included in the stream. Additionally, audio streams transmitted to the one or more other terminals may also be modified to include audio signals originating from the moved terminal. Other changes to a terminal that may also affect audio processing include activating a muting function. Muting a terminal while the terminal is among a group of participating terminals may cause the audio signals from the other terminals to also be muted.

Teleconferencing configuration based on proximity information

An improved application programming interface (API) as described can control media resources in numerous Internet multimedia applications. The API may be independent of the application itself and the media resources underneath. The API may be referred to as a multimedia subsystem (MSS) interface.

Multimedia application interface

It is disclosed a method comprising registering a respective target network system state from each one of a plurality of programs, each of the plurality of programs comprising a first program state, detecting a current network system state, selecting one or more of the plurality of programs based on a result matching the detected current network system state against the registered target network system states, and commanding transition of the selected one or more programs from the first program state to a second program state different from the first program state.

Method, apparatus and computer program for registering a respective target network system state from each one of a plurality of programs

Circuitry, embodied in a media subsystem (A), reproduces a speech or other type of audio signal, and is operable when playing back audio data for reducing synchronization delay. A method operates by, when a frame containing audio data is sent to a decoder (), measuring the synchronization delay; determining by how much the synchronization delay should be adjusted; and adjusting the synchronization delay in a content-aware manner by adding or removing one or more audio samples in a selected current frame or in a selected subsequent frame so as not to significantly degrade the quality of the played back audio data. When the synchronization delay is adjusted by more than one audio sample, the adjustment can be made by all of the determined audio samples in one adjustment, or the adjustment is made by less than all of the determined audio samples by a plurality of adjustments. The step of adjusting selects, if possible, an unvoiced frame and discriminates against a transient frame. The step of determining includes measuring an average amount of time that a frame resides in a jitter buffer (), and adjusting the synchronization delay so that the average duration approaches a desired jitter buffer residency duration.

Method and apparatus for reducing synchronization delay in packet-based voice terminals by resynchronizing during talk spurts

Circuitry, embodied in a media subsystem, reproduces a speech or other type of audio signal, and is operable during a time that comfort noise is being generated for storing data from an arriving data packet that contains data representing a beginning of an interval of speech. The circuitry detects that data representing speech has been stored and synchronizes the circuitry to the beginning of the interval of speech by terminating the generation of comfort noise that precedes the interval of speech, and reproduces the speech by decoding the stored data from the data packet and converting the decoded data to an audio speech signal. The arriving data packet contains at least one frame of speech data having a duration of X milliseconds, and the circuitry includes an audio device, such as a DAC or a PCM connection, that requests decoded speech data at a rate given by X/Y milliseconds, where Y is greater than one. By example, the frame of speech data has a duration in the range of about 20 milliseconds to about 60 milliseconds, and an audio device requests decoded speech data at a rate of less than every 10 milliseconds (e.g., every 5 milliseconds). The circuitry may be contained within a wireless communications device, such as a cellular telephone or a personal communicator.

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Helsinki, Finland

Jari Selin