DTS, Inc. Patent applications |
Patent application number | Title | Published |
20160100268 | DIGITAL AUDIO FILTERS FOR VARIABLE SAMPLE RATES - Various exemplary embodiments relate to a method and apparatus for processing audio signals to influence the reproduction of the audio signals. 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 a virtualization profile including a digital audio filter with a design sample rate, resample the virtualization profile to a different sample rate, filter the audio signal with the resampled virtualization profile, and reproduce the filtered audio signal as sound. | 04-07-2016 |
20150332663 | SPATIAL AUDIO ENCODING AND REPRODUCTION OF DIFFUSE SOUND - 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. | 11-19-2015 |
20150269951 | RESIDUAL ENCODING IN AN OBJECT-BASED AUDIO SYSTEM - Lossy compression and transmission of a downmixed composite signal having multiple tracks and objects, including a downmixed signal, is accomplished in a manner that reduces the bit-rate requirement as compared to redundant transmission or lossless compression, while reducing upmix artifacts. A compressed residual signal is generated and transmitted along with a compressed total mix and at least one compressed audio objects. In the reception and upmix aspect the invention decompresses a downmixed signal and other compressed objects, calculates an approximate upmix signal, and corrects specific base signals derived from the upmix, by subtracting a decompressed residual signal. The invention thus allows lossy compression to be used in combination with downmixed audio signals for transmission through a communication channel (or for storage). Upon later reception and upmix, additional base signals are recoverable in capable systems providing multi-object capability (while legacy systems can easily decode a total mix without upmix). | 09-24-2015 |
20150255076 | POST-ENCODING BITRATE REDUCTION OF MULTIPLE OBJECT AUDIO - 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. | 09-10-2015 |
20150230041 | ROOM CHARACTERIZATION AND CORRECTION FOR MULTI-CHANNEL AUDIO - Devices and methods are adapted to characterize a multi-channel loudspeaker configuration, to correct loudspeaker/room delay, gain and frequency response or to configure sub-band domain correction filters. | 08-13-2015 |
20140350944 | ENCODING AND REPRODUCTION OF THREE DIMENSIONAL AUDIO SOUNDTRACKS - The present invention provides a novel end-to-end solution for creating, encoding, transmitting, decoding and reproducing spatial audio soundtracks. The provided soundtrack encoding format is compatible with legacy surround-sound encoding formats, so that soundtracks encoded in the new format may be decoded and reproduced on legacy playback equipment with no loss of quality compared to legacy formats. | 11-27-2014 |
20140303984 | LAYERED AUDIO CODING AND TRANSMISSION - Embodiments of systems and methods are described for generating layered audio such that computing devices can request a variable amount of data based on criteria such as their available bandwidth, device capability, or user selection. A base layer and one or more enhancement layers that incrementally enhance the previous layers may be generated. A computing device may retrieve the base layer and/or one or more enhancement layers, adjusting, in real-time or near real-time, which layers are retrieved based on fluctuations in the available bandwidth among other possible criteria. | 10-09-2014 |
20140303762 | LAYERED AUDIO RECONSTRUCTION SYSTEM - A computing device may receive or otherwise access a base audio layer and one or more enhancement audio layers. The computing device can reconstruct the retrieved base layer and/or enhancement layers into a single data stream or audio file. The local computing device may process audio frames in a highest enhancement layer retrieved in which the data can be validated (or a lower layer if the data in audio frames in the enhancement layer(s) cannot be validated) and build a stream or audio file based on the audio frames in that layer. | 10-09-2014 |
20140301556 | DIRECTIONAL BASED AUDIO RESPONSE TO AN EXTERNAL ENVIRONMENT EMERGENCY SIGNAL - An audio signal attenuation system and method for detecting an audio emergency warning signal (or alarm) in a vehicle in which an audio signal is being played. Embodiments of the system and method make it easier for a police, fire, or other emergency alarm or siren to be heard in a loud or noisy listening environment when audio signal is being reproduced. This is achieved using selective frequency attenuation, which identifies a frequency of the alarm and then selectively attenuates the alarm frequency in the audio signal. Moreover, direction data that includes information about from which direction the alarm is coming can be used to selectively attenuate the alarm frequency in certain channels (or speakers) of the audio signal. In some embodiments, audio cues are used to alert the listener to the alarm signal and are adjusted based on alarm distance from the vehicle, speed, and the type of alarm. | 10-09-2014 |
20140270263 | AUTOMATIC MULTI-CHANNEL MUSIC MIX FROM MULTIPLE AUDIO STEMS - There are disclosed automatic mixers and methods for creating a surround audio mix. A set of rules may be stored in a rule base. A rule engine may select a subset of the set of rules based, at least in part, on metadata associated with a plurality of stems. A mixing matrix may mix the plurality of stems in accordance with the selected subset of rules to provide three or more output channels. | 09-18-2014 |
20140270184 | AUDIO DEPTH DYNAMIC RANGE ENHANCEMENT - An audio depth dynamic range enhancement system and method for enhancing the dynamic range of depth in audio sound systems as perceived by a human listener. Embodiments of the system and method process an input audio signal by applying a gain function to at least one of a plurality of sub-signals of the audio signal having different values of a spatial depth parameter. The sub-signals are combined to produce a reconstructed audio signal carrying modified audio information. The reconstructed audio signal is output from the system and method for reproduction by the audio sound system. The gain function alters the gain of the at least one of the plurality of sub-signals such that the reconstructed audio signal, when reproduced by the audio sound system, results in modified depth dynamic range of the audio sound system with respect to the spatial depth parameter. | 09-18-2014 |
20140185811 | SYSTEM AND METHOD FOR VARIABLE DECORRELATION OF AUDIO SIGNALS - Various embodiments relate to a system and method for decorrelating an audio signal with a hybrid filter. The hybrid filter is generated by first generating a decorrelation filter. A frequency-dependent warping is applied to the decorrelation filter. The warped decorrelation filter is then mixed with a carrier filter to generate the hybrid filter. The carrier filter may include filters for spatial processing of an audio signal, filters for upmixing an audio signal, and/or filters for downmixing an audio signal. | 07-03-2014 |
20140153727 | METHOD AND APPARATUS FOR PERSONALIZED AUDIO VIRTUALIZATION - 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. | 06-05-2014 |
20140142959 | RECONSTRUCTION OF A HIGH-FREQUENCY RANGE IN LOW-BITRATE AUDIO CODING USING PREDICTIVE PATTERN ANALYSIS - A predictive pattern high-frequency reconstruction system and method that finds patterns in high-frequency components of an audio signal, encodes the audio signal into an encoded bitstream along with pattern information, and then uses the patterns to reconstruct the high-frequency components during decoding. The high-frequency components can be reconstructed using the pattern information alone. Embodiments of the system and method map normalized subband signals of the audio signal to a scaled representation of a time-frequency grid containing multiple tiles and perform statistical analysis on each tile to estimate subband parameters and determine whether a pattern exists. If a pattern does exist, it can be encoded in the encoded bitstream, transmitted, and used to reconstruct the high-frequency components at the decoder. A direct search technique and a fast Fourier transform (FFT) technique may be used to perform the statistical analysis. | 05-22-2014 |
20140074460 | SCALABLE CODE EXCITED LINEAR PREDICTION BITSTREAM - The present invention provides for methods and apparatuses for processing audio data. In one embodiment, there is a 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. 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. | 03-13-2014 |
20140016791 | LOUDNESS CONTROL WITH NOISE DETECTION AND LOUDNESS DROP DETECTION - Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels. | 01-16-2014 |
20140010515 | PLAYBACK SYNCHRONIZATION - Various exemplary embodiments relate to method and media devices for synchronizing media playback between a receiving media device and sending media device, including: receiving, at the receiving media device, a plurality of messages from the sending media device, wherein the plurality of messages include a plurality of sender timestamps; generating a plurality of clock offset values based on the plurality of sender timestamps and a clock of the receiving media device; identifying a minimum clock offset value from the plurality of clock offset values; locating first media data for playback and a first presentation time associated with the first media data; and causing the first media data to be rendered at a first time that matches the first presentation time based on the minimum clock offset. | 01-09-2014 |
20100303246 | VIRTUAL AUDIO PROCESSING FOR LOUDSPEAKER OR HEADPHONE PLAYBACK - 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. | 12-02-2010 |