Qualcomm Incorporated

Nidish Ramachandra Kamath

Suresh Devalapalli

Prajakt Kulkarni

Samir Kumar Gupta

Eddie L T Choy

Stephen A Molloy

Allister Alemania

Sang-Uk Ryu

Johnny Kallacheril John

Deepti Mani

Can Acar

Robert J Vachon

Alex Jong

Satish Goverdhan

Heejong Yoo

Srilekha Krishnamurthy

Indranil Banerjee

Srivatsan Srinivasan

Ge Wang

Dmitry Beransky

Mesh communications aspects are disclosed in which multiple mobile devices located within a communication area perform direct communication to establish a mesh network. Each device node included in the mesh network provides power profile information to the other nodes on the network. As mesh messages are generated for transmission by the nodes onto the mesh network, an adaptive routing mechanism determines the transmission route based on the power profile of the proposed target node. The selective and adaptive routing determination allows for power to be efficiently conserved within the mesh network.

Power optimized behavior in mesh networks

Embodiments of methods, apparatuses, devices and systems associated with combining or mixing digital media streams are disclosed.

System, method or apparatus for combining multiple streams of media data

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

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.

Unified filter bank for performing signal conversions

This disclosure describes techniques for processing audio files that comply with the musical instrument digital interface (MIDI) format. In particular, this disclosure describes storage of MIDI parameters for efficient access by a processor and a hardware unit. The processor may be a digital signal processor (DSP) and the hardware unit may be specifically designed to process MIDI parameters. In one aspect, this disclosure provides an apparatus comprising a processor that converts a MIDI event into MIDI parameters, a hardware unit that uses MIDI parameters to generate audio samples, and a plurality of storage units that store MIDI parameters which are accessible by both the processor and the hardware unit.

Musical instrument digital interface parameter storage

This disclosure describes techniques that make use of a waveform fetch unit that operates to retrieve waveform samples on behalf of each of a plurality of hardware processing elements that operate simultaneously to service various audio synthesis parameters generated from one or more audio files, such as musical instrument digital interface (MIDI) files. In one example, a method comprises receiving a request for a waveform sample from an audio processing element, and servicing the request by calculating a waveform sample number for the requested waveform sample based on a phase increment contained in the request and an audio synthesis parameter control word associated with the requested waveform sample, retrieving the waveform sample from a local cache using the waveform sample number, and sending the retrieved waveform sample to the requesting audio processing element.

Waveform fetch unit for processing audio files

This disclosure describes techniques that make use of a summing buffer that receives waveform samples from audio processing elements, and sums and stores the waveform sums for a given frame. In one example, a method comprises summing a waveform sample received from an audio processing element to produce a waveform sum associated with a first audio frame, storing the waveform sum in a memory, wherein the memory is logically partitioned into a plurality of memory blocks, and locking memory blocks containing the waveform sum associated with the first audio frame, transferring contents of locked memory blocks to an external processor, unlocking a memory block after contents of the memory block have been transferred to the external processor, and storing a waveform sum associated with a second audio frame within the unlocked memory block concurrently with transferring contents of remaining locked memory blocks associated with the first audio frame.

Shared buffer management for processing audio files

Techniques are described of efficiently identifying sets of audio parameters to be applied during a time frame. For example, a list of indicators may be generated. Each of the indicators in the list may indicate a Musical Instrument Digital Interface (MIDI) voice present in a MIDI frame. Furthermore, in generating the list, the indicators in the list may be restricted to those indicators that indicate the most acoustically significant MIDI voices in the MIDI frame. After the list is generated, a digital waveform may be generated for each of MIDI voices indicated by an indicator in the list. A combination of the waveforms of each MIDI voice may constitute an overall waveform for the MIDI frame.

Efficient identification of sets of audio parameters

This disclosure describes techniques for generating a set of data points that form a triangular wave having a desired gain and a desired frequency. In one example, the method includes the step of (a) determining an increment value based on the desired frequency and the desired gain of the triangular wave. The method further includes the step of (b) adding the increment value to a current data point to generate a next data point, the current data point and the next data point forming a subset of the set of data points. The method further includes the step of iteratively performing (a) and (b) to generate the set of data points that form the triangular wave.

Method and device for generating triangular waves

Techniques are described of generating a digital waveform for a Musical Instrument Digital Interface (MIDI) voice using a set of machine-code instructions that is specialized for the generation of digital waveforms for MIDI voices. For example, a processor may execute a software program that generates a digital waveform for a MIDI voice. The instructions of the software program may be machine code instructions from an instruction set that is specialized for the generation of digital waveforms for MIDI voices. In particular, the execution of one of the instructions may involve a selection of an operation based on a set of parameters that define a MIDI voice and the performance of the selected operation.

Musical instrument digital interface hardware instructions

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.

Audio processing hardware elements

Generating a digital waveform for a Musical Instrument Digital Interface (MIDI) voice using a set of machine-code instructions that is specialized for the generation of digital waveforms for MIDI voices. For example, a processor may execute a software program that generates a digital waveform for a MIDI voice. The instructions of the software program may be machine code instructions from an instruction set that is specialized for the generation of digital waveforms for MIDI voices.

Musical instrument digital interface hardware instruction set

This disclosure describes techniques for processing audio files that comply with the musical instrument digital interface (MIDI) format. In particular, various tasks associated with MIDI file processing are delegated between software operating on a general purpose processor, firmware associated with a digital signal processor (DSP), and dedicated hardware that is specifically designed for MIDI file processing. Alternatively, a multi-threaded DSP may be used instead of a general purpose processor and the DSP. In one aspect, this disclosure provides a method comprising parsing MIDI files and scheduling MIDI events associated with the MIDI files using a first process, processing the MIDI events using a second process to generate MIDI synthesis parameters, and generating audio samples using a hardware unit based on the synthesis parameters.

Growing community of inventors

Placentia, CA, United States of America

Nidish Ramachandra Kamath