Qualcomm Incorporated

Eddie L T Choy

Samir Kumar Gupta

Song Wang

Prajakt Kulkarni

Khaled Helmi El-Maleh

Pei Xiang

Nagendra Nagaraja

Dinesh Ramakrishnan

Nidish Ramachandra Kamath

Kuntal Dilipsinh Sampat

Joel Linsky

Suresh Devalapalli

Sharath Manjunath

Stephen A Molloy

Sang-Uk Ryu

Heejong Yoo

Ananthapadmanabhan Arasanipalai Kandhadai

Harinath Garudadri

Asif Mohammad

Johnny Kallacheril John

Anthony Patrick Mauro, Ii

Ananthapadmanabhan Aasanipalai Kandhadai

Amitava Das

Matthew Zivney

Satish Goverdhan

Mingxia Cheng

Yujie Gao

Ravi Satyanarayanan

Prajakt V Kulkarni

Matthew Blaine Zivney

Anthony Patrick Ii Mauro

A method and system for resynchronizing an embedded multimedia system using bytes consumed in an audio decoder. The bytes consumed provides a mechanism to compensate for bit error handling and correction in a system that does not require re-transmission. The audio decoder keeps track of the bytes consumed and periodically reports the bytes consumed. A host microprocessor indexes the actual bytes consumed since bit errors may have been handled or corrected to a predetermined byte count to determine whether resynchronization is necessary.

Method and system for error robust audio playback time stamp reporting

Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

Multiple microphone voice activity detector

A wireless communication device comprises a first wireless interface configured to communicate with a device over a first wireless network, and a second wireless interface configured to communicate with a remote server over a second wireless network, the remote server storing one or more executables. The wireless communication device includes a configured to receive device configuration information from the device over the first wireless network, the device configuration information identifying at least processing hardware resources in the device. The controller is configured to select an executable based on the device configuration information, the executable for configuring the device to process data in a particular format and determine whether the selected executable is stored within the wireless communication device, and if not, to retrieve the selected executable from the remote server by way of the second wireless network, and download the selected executable to the device over the first wireless network.

Dynamically provisioning a device

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.

Power efficient batch-frame audio decoding apparatus, system and method

In accordance with a method for providing a distinct perceptual location for an audio source within an audio mixture, a foreground signal may be processed to provide a foreground perceptual angle for the foreground signal. The foreground signal may also be processed to provide a desired attenuation level for the foreground signal. A background signal may be processed to provide a background perceptual angle for the background signal. The background signal may also be processed to provide a desired attenuation level for the background signal. The foreground signal and the background signal may be combined into an output audio source.

Methods and apparatus for providing a distinct perceptual location for an audio source within an audio mixture

In a digital system with more than one clock source, lack of synchronization between the clock sources may cause overflow or underflow in sample buffers, also called sample slipping. Sample slipping may lead to undesirable artifacts in the processed signal due to discontinuities introduced by the addition or removal of extra samples. To smooth out discontinuities caused by sample slipping, samples are filtered to when a buffer overflow condition occurs, and the samples are interpolated to produce additional samples when a buffer underflow condition occurs. The interpolated samples may also be filtered. The filtering and interpolation operations can be readily implemented without adding significant burden to the computational complexity of a real-time digital system.

Resolving buffer underflow/overflow in a digital system

Encoder-assisted frame loss concealment (FLC) techniques for decoding audio signals are described. A decoder may discard an erroneous frame of an audio signal and may implement the encoder-assisted FLC techniques in order to accurately conceal the discarded frame based on neighboring frames and side-information transmitted from the encoder. The encoder-assisted FLC techniques include estimating magnitudes of frequency-domain data for the frame based on frequency-domain data of neighboring frames, and estimating signs of the frequency-domain data based on a subset of signs transmitted from the encoder as side-information. Frequency-domain data for a frame of an audio signal includes tonal components and noise components. Signs estimated from a random signal may be substantially accurate for the noise components of the frequency-domain data. However, to achieve highly accurate sign estimation for the tonal components, the encoder transmits signs for the tonal components of the frequency-domain data as side-information.

Encoder-assisted frame loss concealment techniques for audio coding

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.

Audio coding selection based on device operating condition

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. Example embodiments may first remove any existing context from a digital audio signal to obtain a context suppressed signal. The context suppressed signal may then be encoded. An audio context may be selected from among a plurality of audio contexts, with the selected audio context inserted into a signal based on the encoded context suppressed signal.

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. Example embodiments may decode two sets of encoded frames from an encoded audio signal. The two frame sets may be encoded using different encoding schemes. For example, the bit rate or coding mode may differ between the two encoded frame sets. Based on information from one of the decoded sets of frames, a context component included in a signal represented by the other frame set may be suppressed. Other embodiments may generate an audio context signal within the mobile user terminal, and mix the generated audio signal with another decoded audio signal.

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. Enhancing the context of a voice communication may first include suppressing an existing context component from the digital audio signal to obtain a context suppressed signal. This signal may then be mixed with a new context signal to create a context enhanced signal, which may then be encoded before transmission. When this new context enhanced signal includes a speech component, it may be encoded and transmitted at a particular bit rate. When the context enhanced signal does not include a speech component, it may also be encoded at a similar bit rate. However, depending on the state of a process control signal, portions of a digital audio signal that lack a speech component may also be transmitted at a lower bit rate.

Systems, methods, and apparatus for context replacement by audio level

An executable is downloaded to an audio output device over a communications link. The executable may configure the audio output device to decode audio encoded in a specified format. The executable may also or alternatively include other audio processing software. The audio may include voice and/or audio playback, e.g., music playback. The ability to download an audio executable allows dynamic provisioning of various decoding and/or audio process capabilities to an audio output device. This may eliminate the need to transcode digitized audio for playback at the audio output device, and may also allow the audio output device to decode multiple audio formats without having multiple audio decoders permanently residing within the audio output device.

Dynamically provisioning a device with audio processing capability

A method for providing an interface to a processing engine that utilizes intelligent audio mixing techniques may include receiving a request to change a perceptual location of an audio source within an audio mixture from a current perceptual location relative to a listener to a new perceptual location relative to the listener. The audio mixture may include at least two audio sources. The method may also include generating one or more control signals that are configured to cause the processing engine to change the perceptual location of the audio source from the current perceptual location to the new perceptual location via separate foreground processing and background processing. The method may also include providing the one or more control signals to the processing engine.

Methods and apparatus for providing an interface to a processing engine that utilizes intelligent audio mixing techniques

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.

Systems and methods for double-talk detection in acoustically harsh environments

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. In one aspect, a method of processing a digital audio signal that includes a first audio context is disclosed. The method comprises based on a first audio signal that is produced by a first microphone, suppressing the first audio context from the digital audio signal to obtain a context-suppressed signal. The method may further comprise selecting a second context based on the first audio context, and mixing the second audio context with a signal that is based on the context-suppressed signal to obtain a context-enhanced signal.

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Carlsbad, CA, United States of America

Eddie L T Choy