Abstract: A method for forming a thin film by a deposition process, including: a substrate is placed in a deposition chamber; a precursor is introduced into the deposition chamber to form an adsorption layer on a surface of the substrate; a reactant is introduced into the deposition chamber and reacts with the adsorption layer to form a thin film layer on the surface of the substrate and generate reaction byproducts; a vacuuming operation is performed on the deposition chamber to decrease a chamber pressure therein to reduce desorption energy of the reaction byproducts formed at the surface of the thin film layer; plasma is introduced into the deposition chamber to increase energy of the surface of the formed thin film layer; a cleaning gas is introduced into the deposition chamber to discharge the reaction byproducts and the residual precursor and reactant in the deposition chamber.

Abstract: A method for forming a film layer includes: a substrate is provided; a pretreatment step is performed, in which the pretreatment step includes providing a reaction source gas, which forms attachment points on the substrate; and a deposition step is performed, in which the reaction source gas is formed into a plasma, which is deposited on the substrate based on the attachment points to form a first film layer.

Abstract: Systems and methods for device compatibility testing and reporting are disclosed. An exemplary method includes outputting, by a testing computer station, instructions for executing a plurality of certification tests on a device under test. The method includes receiving result indications and corresponding certification test identifiers corresponding to each of the plurality of certification tests, and generating, by the testing computer station, compressed test results for each of the plurality of certification tests. The method includes combining at least a portion of the compressed test results into a reduced data report. The method includes sending, by a computer network in communication with the testing computer station and a remote server, the reduced data report to the remote server.

Abstract: Disclosed is a method, a system and a computer readable medium for encoding a video stream. The method includes determining a first time period based on a first estimated processing time algorithm, determining a second time period based on a second estimated processing time algorithm, comparing an encoding time period of the video frame to the first time period and the second period, and changing an encoding scheme for encoding a subsequent video frame if two times the encoding time of the video frame is one of less than the first time period and greater than the second time period.

Abstract: Systems and methods for device compatibility testing and reporting are disclosed. An exemplary method includes outputting, by a testing computer station, instructions for executing a plurality of certification tests on a device under test. The method includes receiving result indications and corresponding certification test identifiers corresponding to each of the plurality of certification tests, and generating, by the testing computer station, compressed test results for each of the plurality of certification tests. The method includes combining at least a portion of the compressed test results into a reduced data report. The method includes sending, by a computer network in communication with the testing computer station and a remote server, the reduced data report to the remote server.

Abstract: Provided are methods and systems for processing audio using a combination of interrupts and timer events. Buffer queues and timer events are configured and utilized between audio capture and encoding processes, and also between audio decoding and playout processes to deliver a fixed-rate audio packet and higher quality audio output. Thread execution priorities may be assigned to the capture and/or playout timer events, and such priorities may be automatically adjusted to conserve processing resources. By configuring timer events to drive audio encoding and decoding processes, and utilizing buffer queues to absorb variances in the timing of audio capture and playout interrupts, the methods and systems provided are able to deliver improved audio quality while conserving processing resources.

Abstract: Methods and systems are provided for determining audio latency in both audio capture and audio playout processes in a real-time audio processing system by calculating time differences with respect to interrupts. The audio capture and playout latencies may be utilized for improved audio processing, such as AEC. The methods provided address two major problems with existing techniques. First, on platforms where the system does not provide API to obtain audio latency, the methods allow the application to obtain more accurate audio latency. Second, the application may utilize a unified and simplified technique to obtain audio latency instead of using platform-specific APIs on different platforms.

Abstract: Methods and systems are provided for negotiating codecs between different platforms and devices such that an audio application selects a codec to use based on the capabilities of the platforms and devices. Processing or resource requirements for various combinations of encoders and decoders may be compared against resource thresholds defined for each client. A combination of an encoder and a decoder may be selected for clients wishing to participate in a communication session such that the selected combination is within the resource threshold applicable to each client.

Abstract: Provided are methods and systems for processing audio using a combination of interrupts and timer events. Buffer queues and timer events are configured and utilized between audio capture and encoding processes, and also between audio decoding and playout processes to deliver a fixed-rate audio packet and higher quality audio output. Thread execution priorities may be assigned to the capture and/or playout timer events, and such priorities may be automatically adjusted to conserve processing resources. By configuring timer events to drive audio encoding and decoding processes, and utilizing buffer queues to absorb variances in the timing of audio capture and playout interrupts, the methods and systems provided are able to deliver improved audio quality while conserving processing resources.

Abstract: Methods and systems are provided for determining audio latency in both audio capture and audio playout processes in a real-time audio processing system by calculating time differences with respect to interrupts. The audio capture and playout latencies may be utilized for improved audio processing, such as AEC. The methods provided address two major problems with existing techniques. First, on platforms where the system does not provide API to obtain audio latency, the methods allow the application to obtain more accurate audio latency. Second, the application may utilize a unified and simplified technique to obtain audio latency instead of using platform-specific APIs on different platforms.