Abstract: A method for simulating spatially extended sound sources comprising: panning a first input signal over a plurality of output channels to generate a first multi-channel directionally encoded signal; panning a second input signal over the plurality of output channels to generate a second multi-channel directionally encoded signal; combining the first and second multi-channel directionally encoded signals to generate a plurality of loudspeaker output channels; and applying a bank of decorrelation filters on the loudspeaker output channels.

Abstract: The present invention provides a frequency-domain spatial audio coding framework based on the perceived spatial audio scene rather than on the channel content. In one embodiment, time-frequency spatial direction vectors are used as cues to describe the input audio scene.

Abstract: A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

Abstract: A frequency domain method for phase-amplitude matrixed surround decoding of 2-channel stereo recordings and soundtracks, based on spatial analysis of 2-D or 3-D directional cues in the recording and re-synthesis of these cues for reproduction on any headphone or loudspeaker playback system.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: An input audio signal is equalized to form an output audio signal on the basis of an intended listening sound pressure level, the output capabilities of a particular playback device, and unique hearing characteristics of a listener. An intended listening level is first determined based on the properties of the audio signal and a mastering sound level. The intended listening level is used to determine an optimal sound pressure level for the particular playback device based on its capabilities and any master volume gain. These two levels are used to determine how much louder to make individual frequencies based on data pertaining to human auditory perception, either standardized or directly measured. The audio is further compensated on the basis of hearing loss data, again either standardized or directly measured, after optionally extending the signal bandwidth. The final, compensated audio signal is sent to the playback device for playback.

Abstract: A method and apparatus processes multi-channel audio by encoding, transmitting or recording “dry” audio tracks or “stems” in synchronous relationship with time-variable metadata controlled by a content producer and representing a desired degree and quality of diffusion. Audio tracks are compressed and transmitted in connection with synchronized metadata representing diffusion and preferably also mix and delay parameters. The separation of audio stems from diffusion metadata facilitates the customization of playback at the receiver, taking into account the characteristics of local playback environment.

Abstract: A method of ambience extraction includes analyzing an input signal to determine the time-dependent and frequency-dependent amount of ambience in the input signal, wherein the amount of ambience is determined based on a signal model and correlation quantities computed from the input signals and wherein the ambience is extracted using a multiplicative time-frequency mask. Another method of ambience extraction includes compensating a bias in the estimation of a short-term cross-correlation coefficient. In addition, systems having various modules for implementing the above methods are disclosed.

Abstract: MIDI-generated audio streams or other input streams of audio events are perceptually associated with specific locations in 3D space with respect to the listener. A conventional pan parameter is redefined so that it no longer specifies the relative balance between the audio being fed to two fixed speaker locations. Instead, the new MIDI pan parameter extension specifies a virtual position of an audio stream in 3D space. Preferably, the relative position of a single audio stream is set along a predefined arc in 3D space.

Abstract: The present invention counterbalances background noise by applying dynamic equalization. A psychoacoustic model representing the perception of masking effects of background noise relative to a desired foreground soundtrack is used to accurately counterbalance background noise. A microphone samples what the listener is hearing and separates the desired soundtrack from the interfering noise. The signal and noise components are analyzed from a psychoacoustic perspective and the soundtrack is equalized such that the frequencies that were originally masked are unmasked. Subsequently, the listener may hear the soundtrack over the noise. Using this process the EQ can continuously adapt to the background noise level without any interaction from the listener and only when required. When the background noise subsides, the EQ adapts back to its original level and the user does not experience unnecessarily high loudness levels.

Abstract: There are provided methods and an apparatus for processing audio signals. According to one aspect of the present invention there is included a method for processing audio signals having the steps of receiving at least one audio signal having at least a center channel signal, a right side channel signal, and a left side channel signal; processing the right and left side channel signals with a first virtualizer processor, thereby creating a right virtualized channel signal and a left virtualized channel signal; processing the center channel signal with a spatial extensor to produce distinct right and left outputs, thereby expanding the center channel with a pseudo-stereo effect; and summing the right and left outputs with the right and left virtualized channel signals to produce at least one modified side channel output.

Abstract: Modifying a panned source in an audio signal comprising a plurality of channel signals is disclosed. Portions associated with the panned source are identified in at least selected ones of the channel signals. The identified portions are modified based at least in part on a user input.

Abstract: MIDI-generated audio streams or other input streams of audio events are perceptually associated with specific locations in 3D space with respect to the listener. A conventional pan parameter is redefined so that it no longer specifies the relative balance between the audio being fed to two fixed speaker locations. Instead, the new MIDI pan parameter extension specifies a virtual position of an audio stream in 3D space. Preferably, the relative position of a single audio stream is set along a predefined arc in 3D space.

Abstract: There are provided methods and an apparatus for conditioning an audio signal. According to one aspect of the present invention there is included a method for conditioning an audio signal having the steps of: receiving at least one audio signal, each audio signal having at least one channel, each channel being segmented into a plurality of frames over a series of time; calculating at least one measure of dynamic excursion of the audio signal for a plurality of successive segments of time; filtering the audio signal into a plurality of subbands, each frame being represented by at least one subband; deriving a dynamic gain factor from the successive segments of time; analyzing at least one subband of the frame to determine if a transient exists in the frame; and applying the dynamic gain factor to each frame having a transient.

Abstract: The overall loudness of an audio track is calculated by combining a number of weighted loudness measures for segments of the audio track, where the weight applied to each individual loudness measure is a function of the loudness measure. By comparing the original overall loudness measure to a desired overall loudness measure, a gain can be determined that will adjust the loudness level to the desired value. Also disclosed is a dynamic compression method that analyzes the dynamic characteristics of an audio track and determines appropriate compressor parameters. Additionally, the loudness of a post-compressor audio track can be estimated for any given compressor parameters, thus permitting post-compression loudness matching to be done even if the compression is performed in real-time.

Abstract: There are provided methods and an apparatus for processing audio signals. According to one aspect of the present invention there is included a method for processing audio signals having the steps of receiving at least one audio signal having at least a center channel signal, a right side channel signal, and a left side channel signal; processing the right and left side channel signals with a first virtualizer processor, thereby creating a right virtualized channel signal and a left virtualized channel signal; processing the center channel signal with a spatial extensor to produce distinct right and left outputs, thereby expanding the center channel with a pseudo-stereo effect; and summing the right and left outputs with the right and left virtualized channel signals to produce at least one modified side channel output.

Abstract: A frequency-domain method for format conversion or reproduction of 2-channel or multi-channel audio signals such as recordings is described. The reproduction is based on spatial analysis of directional cues in the input audio signal and conversion of these cues into audio output signal cues for two or more channels in the frequency domain.

Abstract: A method and apparatus for processing sound sources to simulate environmental effects includes source channel blocks for each source and single reverberation block. The source channel blocks include direct, early reflection, and late reverberation blocks for conditioning the source feeds to include delays, spectral changes, and attenuations depending on the position, orientation and directivity of the sound sources, the position and orientation of the listener, and the position and sound transmission and reflection properties of obstacles and walls in a modeled environment. The outputs of the source channel blocks are combined and provided to single reverberation block generating both the early reflections and the late reverberation for all sound sources.

Abstract: This invention describes a method for decentralized decoding of a multichannel audio signal by broadcasting the original encoded data and distributing the decoding process between a plurality of receiving units. This allows for the design and manufacture of scalable multichannel audio reproduction systems having an arbitrary number of output channels, composed of a plurality of generic decoder and loudspeaker units each generating fewer output channels. With distributed decoding, a manufacturer can use “off-the-shelf” stereo or mono signal processors, digital-to-analog converters and amplifier components in each generic decoding module, thus reducing manufacturing costs and complexity requirements for each module while offering unlimited scalability in the total number of output channels.

Abstract: A two-channel phase-amplitude stereo encoding and decoding scheme enabling flexible and spatially accurate interactive 3-D audio reproduction via standard audio-only two-channel transmission. The encoding scheme allows associating a 2-D or 3-D positional localization to each of a plurality of sound sources by use of frequency independent inter-channel phase and amplitude differences. The decoder is based on frequency-domain spatial analysis of 2-D or 3-D directional cues in a two-channel stereo signal and re-synthesis of these cues using any preferred spatialization technique, thereby allowing faithful reproduction of positional audio cues and reverberation or ambient cues over arbitrary multi-channel loudspeaker reproduction formats or over headphones, while preserving source separation despite the intermediate encoding over only two audio channels.