| Patent application number | Description | Published |
|---|
| 20080229919 | AUDIO PROCESSING HARDWARE ELEMENTS - This disclosure describes techniques that make use of a plurality of hardware elements that operate simultaneously to service synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises storing audio synthesis parameters generated for one or more audio files of an audio frame, processing a first audio synthesis parameter using a first audio processing element of a hardware unit to generate first audio information, processing a second audio synthesis parameter using a second audio processing element of the hardware unit to generate second audio information, and generating audio samples for the audio frame based at least in part on a combination of the first and second audio information. | 09-25-2008 |
| 20080269926 | AUTOMATIC VOLUME AND DYNAMIC RANGE ADJUSTMENT FOR MOBILE AUDIO DEVICES - A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal. | 10-30-2008 |
| 20090024397 | UNIFIED FILTER BANK FOR PERFORMING SIGNAL CONVERSIONS - A unified filter bank for performing signal conversions may include an interface that receives signal conversion commands in relation to multiple types of compressed audio bitstreams. The unified filter bank may also include a reconfigurable transform component that performs a transform as part of signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include complementary modules that perform complementary processing as part of the signal conversion for the multiple types of compressed audio bitstreams. The unified filter bank may also include an interface command controller that controls the configuration of the reconfigurable transform component and the complementary modules. | 01-22-2009 |
| 20090070119 | POWER EFFICIENT BATCH-FRAME AUDIO DECODING APPARATUS, SYSTEM AND METHOD - Power savings in a mobile device is accomplished by generating audio samples by decoding a bitstream with a decoding system within the mobile device. The generated audio samples are transferred into at least one memory bank in a set of memory banks in a power saver block within the mobile device. Parts of the decoding system not involved in the storing of the generated audio samples are switched off after batch decoding a bitstream associated with multiple audio frames. The bitstream includes bits less than that found in one audio file. At least one of the memory banks in the set of memory banks is power collapsible. The fetching of the decoded by the decoding system can be synchronized with a paging channel of a modem in the mobile device. The transferred audio samples is a lossless compression and may occur after a re-encoding. | 03-12-2009 |
| 20090190769 | SOUND QUALITY BY INTELLIGENTLY SELECTING BETWEEN SIGNALS FROM A PLURALITY OF MICROPHONES - Sound signal reception is improved by utilizing a plurality of microphones to capture sound signals which are then weighed to dynamically adjust signal quality. A first sound signal and a second sound signal are obtained from first and second microphones, respectively, where the first and second sound signals originate from one or more sound sources. A first signal characteristic (e.g., signal power, signal signal-to-noise ratio, etc.) is obtained for the first sound signal and a second signal characteristic is obtained for the second sound signal. The first and second sound signals are weighed or scaled based on their respective first and second signal characteristics. The weighed first and second sound signals are then combined to obtain an output sound signal. | 07-30-2009 |
| 20090190774 | ENHANCED BLIND SOURCE SEPARATION ALGORITHM FOR HIGHLY CORRELATED MIXTURES - An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages. | 07-30-2009 |
| 20090190780 | SYSTEMS, METHODS, AND APPARATUS FOR CONTEXT PROCESSING USING MULTIPLE MICROPHONES - Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. | 07-30-2009 |
| 20090192790 | SYSTEMS, METHODS, AND APPARATUS FOR CONTEXT SUPPRESSION USING RECEIVERS - Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. | 07-30-2009 |
| 20090192791 | SYSTEMS, METHODS AND APPARATUS FOR CONTEXT DESCRIPTOR TRANSMISSION - Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. | 07-30-2009 |
| 20090192803 | SYSTEMS, METHODS, AND APPARATUS FOR CONTEXT REPLACEMENT BY AUDIO LEVEL - Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. | 07-30-2009 |
| 20090196429 | SIGNALING MICROPHONE COVERING TO THE USER - A mechanism is provided that monitors secondary microphone signals, in a multi-microphone mobile device, to warn the user if one or more secondary microphones are covered while the mobile device is in use. In one example, smoothly averaged power estimates of the secondary microphones may be computed and compared against the noise floor estimate of a primary microphone. Microphone covering detection may be made by comparing the secondary microphone smooth power estimates to the noise floor estimate for the primary microphone. In another example, the noise floor estimates for the primary and secondary microphone signals may be compared to the difference in the sensitivity of the first and second microphones to determine if the secondary microphone is covered. Once detection is made, a warning signal may be generated and issued to the user. | 08-06-2009 |
| 20100135483 | SYSTEMS AND METHODS FOR DOUBLE-TALK DETECTION IN ACOUSTICALLY HARSH ENVIRONMENTS - A communications device that is configured to detect double talk is described. An echo canceller is configured to cancel an echo from an input signal using an adaptive filter. A double-talk detector provides a double-talk statistic. The double-talk statistic is proportional to the ratio of the remaining echo energy in the cancellation error signal and the total cancellation error energy. | 06-03-2010 |
| 20100191536 | AUDIO CODING SELECTION BASED ON DEVICE OPERATING CONDITION - A sensor is configured to determine at least one operating condition of a device and a selector is configured to select an audio coding process for the device, based on the operating condition. The operating condition may include remaining battery life of the device and/or ambient noise level. The selected audio coding process may consume less power than another possible audio coding process during audio processing. The audio may include voice and/or audio playback, e.g., music playback. | 07-29-2010 |
