The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains. Because signals in the audio coding format can be mixed with a limited computational effort, the invention may advantageously be applied in a tele-conferencing application.

Layered approach to spatial audio coding

In this paper, a new voice activity detection method is proposed. It is based on the total spectrum energy in the overlapping speech window frames. The noise energy from the higher frequency band is subtracted from the noisy speech spectrum in the lower frequency band. In addition, a moving average filter is used to smooth the speech spectrum energy waveform. The proposed strategy for voice activity detection is robust and works well for a variety of signal to noise ratio levels.

Spectrum energy based voice activity detection

A method for encoding sound field signals includes allocating coding rate by application of a uniform criterion to all subbands of all signals in a joint process. An allocation criterion may be based on a comparison, in a given subband, between a spectral envelope of the signals to be encoded and a coding noise profile, wherein the noise profile may be a sum of a noise shape and a noise offset, which noise offset is computed on the basis of the coding bit budget. The rate allocation process may be combined with an energy-compacting orthogonal transform, for which there is proposed a parameterization susceptible of efficient coding and having adjustable directivity. In a further aspect, the invention provides a corresponding decoding method.

Coding of a sound field signal

In some embodiments, a method for selecting at least one layer of a spatially layered, encoded audio signal. Typical embodiments are teleconferencing methods in which at least one of a set of nodes (endpoints, each of which is a telephone system, and optionally also a server) is configured to perform audio coding in response to soundfield audio data to generate spatially layered encoded audio including any of a number of different subsets of a set of layers, the set of layers including at least one monophonic layer, at least one soundfield layer, and optionally also at least one metadata layer comprising metadata indicative of at least one processing operation to be performed on the encoded audio. Other aspects are systems configured (e.g., programmed) to perform any embodiment of the method, and computer readable media which store code for implementing any embodiment of the method or steps thereof.

Methods and systems for selecting layers of encoded audio signals for teleconferencing

A system and method of signal combining that supports different speakers in a noisy environment is provided. Particularly for deviations in the noise characteristics among the channels, various embodiments ensure a smooth transition of the background noise at speaker changes. A modified noise reduction (NR) may achieve equivalent background noise characteristics for all channels by applying a dynamic, channel specific, and frequency dependent maximum attenuation. The reference characteristics for adjusting the background noise may be specified by the dominant speaker channel. In various embodiments, an automatic gain control (AGC) with a dynamic target level may ensure similar speech signal levels in all channels.

Dynamic microphone signal mixer

Now-a-days, multi-party conferencing is an essential feature in the VoIP calls among desktop PCs, laptops The additional voice processing module required for a conferencing when compared to a normal 2-way call is Mixer Even though some algorithms are available in the literature, they lack in evaluating their performance In this paper, different types of mixing algorithms are investigated and objective methods for evaluating their performance are proposed Also, enhanced mixer algorithm based on an audio stream which has greatest amplitude among all the audio streams to be mixed is proposed Comparison of all the algorithms is done in terms of quality and complexity The proposed mixer algorithm out-performs all the mixer algorithms and the complexity of the proposed method is marginal Also, different types of connection configuration of other voice processing modules with the Mixer are proposed for further quality improvement

Audio mixer for multi-party conferencing in VoIP

This paper deals with different types of configurations and network-architecture models used for multi-party conferencing in VoIP A comparison of multi-party conferencing models is presented based on the bandwidth used by the Real Time Protocol (RTP [5]) streams, quality degradation due to trans-coding if any, etc Also, an efficient and simple architectural design of a media engine for decentralized conferencing with an end-point manager is presented The investigation on the right placement of voice-enhancement modules in the media-engine is presented by analyzing the real time captured data The proposed architectural design is extended for inter-operating with different VoIP-based terminals, which operate at different sampling rates such as 8 kHz, 16 kHz, 24 kHz etc The proposed conferencing model contains ITU-T G722, ITU-T G7221 Wideband codecs, Echo Cancellation (EC), Noise Cancellation (NC), Automatic Gain Controller (AGC) and other VoIP components implemented for IP-Phones running on desktop or laptop computers The proposed system is able to connect up to ten conferees on a computer that has 16GHz speed, 1GB RAM memory and Linux/Window 2000 or later operating system with wideband (16 kHz) voice quality

DSP subsystem for multiparty conferencing in VoIP

An audio gain control system 14 may control the analogue gain 15 and the digital gain 18 applied to a signal from a microphone 12 in a telephone apparatus 10. The system comprises: a power estimator 22 configured to estimate the power represented by digital signal derived from the audio signal, wherein the estimated power is an average power determined from a plurality of frames of the digital signal; a digital gain estimator 23 configured to determine, in dependence on the estimated power, a digital gain 18 which would modify the power represented by the digital signal so as to reach a target power level; and a gain controller 24 configured to adjust an analogue gain applied to the audio signal in dependence on the determined digital gain. The power estimator 22 may receive an input signal via a voice activity detector 21. Analysis of audio in the digital domain allows the targeting of speech alone. Adjustment in both the analogue and digital domains maintains resolution when multiplying digital values and allows a larger dynamic range for the total gain. The system improves the quality of audio received by a far-end listener.

Manikya Prasad.P Bala

Papers

Audio mixer for multi-party conferencing in VoIP

DSP subsystem for multiparty conferencing in VoIP

Dynamic gain controller