Abstract: The present invention provides for methods and apparatuses for processing audio data. In one embodiment, there is provided a method for achieving bitstream scalability in a multi-channel audio encoder, said method comprising receiving audio input data; organizing said input data by a Code Excited Linear Predictor (CELP) processing module for further encoding by arranging said data according to significance of data, where more significant data is placed ahead of less significant data; and providing a scalable output bitstream; a higher bitrate bitstream is scaled to lower bitrate by discarding less significant data from frame ends. The organized CELP data comprises of a first part and a second part. The first part comprises a frame header, sub frame parameters and innovation vector quantization data from the first frame from all channels. The innovation vector quantization data from the first frames from all channels is arranged according to channel number.

Abstract: A sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

Abstract: Media is selected for video playback through a first device and audio playback through one or more separate devices connected through a wireless network. Different techniques for synchronizing the audio and video can be selected based on one or more factors to improve media playback.

Abstract: A transform-based codec and method with energy smoothing for mitigating vector quantization errors (such as “birdies”) during the encoding process. Embodiments of the codec and method use an encoder to apply in combination an orthogonal transformation and a vector permutation to frequency transform coefficients. In some embodiments the transformation is performed first followed by the permutation and in other embodiments the order is reversed. The order used is reversed at the decoder. A smoothing parameter containing the level of energy smoothing to be applied is passed from the encoder to the decoder and used by both to compute a transform matrix and an inverse transform matrix. In some embodiments the transform matrix is a fraction Hadamard matrix that is invertible, energy preserving, controllable, and stable.

Abstract: A GPEQ filter and tuning methods for asymmetric transaural audio reproduction on mobile consumer devices. The GPEQ filter includes an asymmetric FIR filter bank configured to normalize the magnitude responses of at least two audio speakers to a target response above a low frequency cross-over of the audio speakers as reproduced at the listener position and a symmetric IIR filter bank component configured to high pass filter above the low frequency cross-over and equalize the magnitude responses of the at least two audio speakers. In some embodiments the FIR filter bank is configured to normalize the group delay of the audio channels and in some embodiments the IIR filter bank includes an asymmetric IIR filter bank component configured to correct any residual asymmetry in the magnitude responses of the at least two audio speakers. Phase preservation may be accomplished using phase-balanced or phase-compensated IIR filter sections.

Abstract: A constant-power pairwise panning upmixing system and method for upmixing from a two-channel stereo signal to a multi-channel surround sound (having more than two channels). Each output channel is some combination of the two input channels. Closed-form solutions are used to calculate dematrixing coefficients that are used to weight each input channel. The dematrixing coefficients are computed based on an inter-channel level difference and an inter-channel phase difference between the two input signals. The weighted input channels then are mixed uniquely for each output channel to generate a surround sound output from the stereo input signal. Each dematrixing coefficient has an in-phase component and an out-of-phase component. The phase coefficients for each component vary in time and are based on the phase difference between the input signals. The resultant surround sound output faithfully simulates the audio content as originally mixed.

Abstract: A method and apparatus may be used to perform personalized audio virtualization. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: A method and apparatus for processing object-based audio signals is provided. The apparatus receives a plurality of object-based audio signals. Each object-based audio signal of the object-based audio signals includes audio waveform data and object metadata associated with the audio waveform data. The object metadata includes at least one of a loudness parameter or a power parameter associated with the audio waveform data. The apparatus determines a loudness metric based on the received object-based audio signals and based on the at least one of the loudness parameter or the power parameter for each object-based audio signal of the received object-based audio signals. In one configuration, the apparatus renders the received object-based audio signals to a set of output signals based on the determined loudness metric. In another configuration, the apparatus transmits (e.g., broadcast, file delivery, or streaming) the received object-based audio signals based on the determined loudness metric.

Abstract: A method and system is presented for coordinating the transmission of supplemental digital data to accompany broadcast data, and in particular, analog radio broadcasts, among a plurality of broadcasters. The supplemental digital data may provide information about the particular broadcast data being transmitted or may be supplemental to such data. The supplemental digital data to be presented is sorted based on particular algorithms which may take into account broadcaster-specified criteria such as target audience, time of day, type of broadcast data presented, and the like. The supplemental digital data may be audio data, visual data, or audio-visual data for presentation with the broadcast data. The supplemental digital data may further be advertisement data. The advertisement data may be sold by the broadcasters or the party coordinating the IBOC transmission of the supplemental digital data. The supplemental digital data may play simultaneously with muted broadcast data or at a user-specified time.

Abstract: Accurate modeling of acoustic reverberation can be essential to generating and providing a realistic virtual reality or augmented reality experience for a participant. In an example, a reverberation signal for playback using headphones can be provided. The reverberation signal can correspond to a virtual sound source signal originating at a specified location in a local listener environment. Providing the reverberation signal can include, among other things, using information about a reference impulse response from a reference environment and using characteristic information about reverberation decay in a local environment of the participant. Providing the reverberation signal can further include using information about a relationship between a volume of the reference environment and a volume of the local environment of the participant.

Abstract: A method and apparatus for processing object-based audio signals for reproduction through a playback system is provided. The apparatus receives a plurality of object-based audio signals in at least one audio frame. In addition, the apparatus receives at least one audio object command associated with at least one object-based audio signal of the plurality of object-based audio signals. In addition, the apparatus processes the at least one object-based audio signal based on the received at least one audio object command. Further, the apparatus renders a set of object-based audio signals of the plurality of object-based audio signals to a set of output signals based on the at least one audio object command. The at least one audio frame may be received from one of a set top box, an OD player, or a television. The apparatus may be an AV receiver or a television.

Abstract: A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.

Abstract: An active audio output detection system and method that automatically determines the type of audio output being used on an audio playback device. Embodiments of the system and method include playing back audio content on the audio playback device (such as a mobile phone) and capturing sound during playback using an audio capture device (such as a microphone). The captured sound is analyzed to determine whether the audio playing back on the audio playback device is contained therein. If the captured sound matches the audio playing back on the audio playback device, then the active audio output is a loudspeaker. On the other hand, if the captured sound does not match the audio playing back on the audio playback device, then the active audio output is headphones.

Abstract: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

Abstract: A post-encoding bitrate reduction system and method for generating one more scaled compressed bitstreams from a single encoded plenary file. The plenary file contains multiple audio object files that were encoded separately using a scalable encoding process having fine-grained scalability. Activity in the data frames of the encoded audio object files at a time period are compared with each other to obtain a data frame activity comparison. Bits from an available bitpool are assigned to all of the data frames based on the data frame activity comparison and corresponding hierarchical metadata. The plenary file is scaled down by truncating bits in the data frames to conform to the bit allocation. In some embodiments frame activity is compared to a silence threshold and the data frame contains silence if the frame activity is less than or equal to the threshold and minimal bits are used to represent the silent frame.

Abstract: The methods and apparatus described herein optimally represent full 3D audio mixes (e.g., azimuth, elevation, and depth) as “sound scenes” in which the decoding process facilitates head tracking. Sound scene rendering can be performed for the listener's orientation (e.g., yaw, pitch, roll) and 3D position (e.g., x, y, z), and can be modified for a change in the listener's orientation or 3D position. As described below, the ability to render an audio object in both the near-field and far-field enables the ability to fully render depth of not just objects, but any spatial audio mix decoded with active steering/panning, such as Ambisonics, matrix encoding, etc., thereby enabling full translational head tracking (e.g., user movement) beyond simple rotation in the horizontal plane, or 6-degrees-of-freedom (6-DOF) tracking and rendering.

Abstract: A GPEQ filter and tuning methods for asymmetric transaural audio reproduction on mobile consumer devices. The GPEQ filter includes an asymmetric FIR filter bank configured to normalize the magnitude responses of at least two audio speakers to a target response above a low frequency cross-over of the audio speakers as reproduced at the listener position and a symmetric IIR filter bank component configured to high pass filter above the low frequency cross-over and equalize the magnitude responses of the at least two audio speakers. In some embodiments the FIR filter bank is configured to normalize the group delay of the audio channels and in some embodiments the IIR filter bank includes an asymmetric IIR filter bank component configured to correct any residual asymmetry in the magnitude responses of the at least two audio speakers. Phase preservation may be accomplished using phase-balanced or phase-compensated IIR filter sections.

Abstract: A sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

Abstract: A frequency domain long-term prediction system and method for estimating and applying an optimum long term predictor. Embodiments of the system and method include determining parameters of a single-tap predictor using a frequency-domain analysis having an optimality criteria based on spectral flatness measure. Embodiments of the system and method also include determining parameters of the long-term predictor by accounting for the performance of the vector quantizer in quantizing the various subbands. In some embodiments other encoder metrics (such as signal tonality) are used as well. Other embodiments of the system and method include determining the optimal parameters of the long-term predictor by accounting for some of the decoder operation. Other embodiments of the system and method include extending a 1-tap predictor to a k-th order predictor by convolving the 1-tap predictor with a pre-set filter and selecting from a table of such pre-set filters based on a minimum energy criteria.

Abstract: A transform-based codec and method with energy smoothing for mitigating vector quantization errors (such as “birdies”) during the encoding process. Embodiments of the codec and method use an encoder to apply in combination an orthogonal transformation and a vector permutation to frequency transform coefficients. In some embodiments the transformation is performed first followed by the permutation and in other embodiments the order is reversed. The order used is reversed at the decoder. A smoothing parameter containing the level of energy smoothing to be applied is passed from the encoder to the decoder and used by both to compute a transform matrix and an inverse transform matrix. In some embodiments the transform matrix is a fraction Hadamard matrix that is invertible, energy preserving, controllable, and stable.