Abstract: A method and an audio conference device for carrying out an audio conference are disclosed, whereby classification information associated with a respective audio date flow is recorded for supplied audio data flows. According to a result of an evaluation of the classification information, the audio data flows are associated with at least three groups which are homogeneous with regard to the results. The individual audio data flows are processed uniformly in each group in terms of the signals thereof, and said audio data flows processed in this way are superimposed in order to form audio conference data flows to be transmitted to the communication terminals.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: Methods, systems, and computer programs for monitoring quality of audio delivered over a communications channel are presented. One method includes an operation for defining timestamps. The timestamps are associated with a measure of time while delivering audio to a client computer, where each timestamp includes a plurality of timestamp bits. Further, the method includes an operation for modulating an audio signal with pseudo noise (PN) codes when a timestamp bit has a first logical value, and modulating the audio signal with a negative of the PN codes when the timestamp bit has a second logical value. After transmitting the modulated audio signal to the client computer, the timestamp bits are extracted from a received modulated audio signal to obtain received timestamps. The quality of the audio is assessed based on the received timestamps, and the quality of the audio is stored in computer memory.

Abstract: An audio data receiving apparatus includes a receiving unit configured to receive audio data sampled in accordance with a first clock signal; a synchronization unit configured to generate a second clock signal that is synchronized with the first clock signal by extracting clock components contained in the audio data; a demodulator configured to demodulate the audio data in accordance with the second clock signal; an oversampling unit configured to oversample the audio data demodulated by the demodulator by using a frequency higher than a frequency of the second clock signal; a clock generator configured to generate a third clock signal having a frequency nearly equal to the first clock signal; and a data output unit configured to output the audio data oversampled by the oversampling unit in accordance with the third clock signal generated by the clock generator.

Abstract: Disclosed is an audio signal decoding device and a method of balance adjustment that reduces a fluctuation of a decoded signal orientation and maintains a stereo perception. An interchannel correlation computation unit (224) computes a correlation between a left channel decoded stereo signal and a right channel decoded stereo signal, and if the interchannel correlation is low, a peak detection unit (225) uses a peak component of a decoded monaural signal of the current frame and a peak component of either a left or a right channel of the preceding frame to detect a peak component with a high temporal correlation. The peak detection unit (225) combines and outputs, from among the frequencies of the detected peak components, a peak frequency of a frame n?1 and a peak frequency of a frame n. A peak balance coefficient computation unit (226) computes, from the peak frequency of the frame n?1, a balance parameter that is used in converting a peak frequency component of the monaural signal to stereo.

Abstract: An audio channel of a time-based media presentation provides a basis for synchronizing to the presentation across a variety of platforms independent of when and where the presentation is being viewed. By pre-processing the media into a series of non-unique hashes, and similarly processing an audio stream of the media captured at a client device, a comparison can be made that yields an accurate time offset within the presentation. The comparison may usefully be performed over a data network using a server that hosts data from the pre-processed media, and a variety of applications may be deployed on the client device based on the resulting synchronization.

Abstract: A sound output control device is disclosed. The sound output control device includes an application information acquisition section and a control section. The application information acquisition section is configured to acquire application information from an external device having multiple applications for sound output. The control section is configured to process audio data streamed from the external device through causing the external device to start one of the multiple applications. The control section has a predetermined selection criterion to select a start-candidate application from the plurality of applications. The control section is further configured to select the start-candidate application according to the application information acquired from the external device and the predetermined selection criterion, and cause the external device to start the selected start-candidate application.