1.Responsible for the design of audio processing algorithm such as Biquad PEQ filters, DRC, Mixer, Mux, AGL, Sampling rate converter, level meter, FIR filter, bass boost, echo and Noise reduction.
2.Develop software architecture and design using fixed point C coding, DSP-specific C, assembly programming, Matlab, Python, Simulink, Pearl, or bash.
3.MCU coding to control DSP core data, coefficient and interface back end amp, DAC or ADC.
4.Optimize computation and memory footprint.
5.Develop the prototype of audio processing algorithm on FPGA solution before implement it into DSP in silicon.
6.Performs measurement using audio measurement tools and techniques.
7.Develop the graphical user interface (GUI) for designed audio processing.
8.Generation of detailed relevant documentation.
9.Drive activities to closure, including proper and adequate knowledge sharing in the organization.
10.Work and communicate effectively with talented team within company and with global customers


1.Master’s degree with 5 years of experience or PhD with 2 years of experience in Electronics Engineering, Computer Science, or a related discipline.
2.Outstanding experience in algorithm development, from concept to application in a research or an industry environment.
3.Self-driven and excellent team player.
4.Expert level digital signal processing in theory and practice, with a proven track record in audio applications, and can recognize practical implementation limitations in a fixed point arithmetic.
5.Experience with software development, testing and concepts, making use of existing code and algorithm validation tools and frameworks (SVN, GIT, Jenkins, JIRA).
6.Hands-on experience with various test setups (troubleshoot setup of a test board, use a scope, power supply, soldering, etc.
7.Familiar with subjective and objective audio quality evaluation (standardization, listening tests, objective metrics, etc.)
8.Good understanding of audio DSP and MCU core.
9.Notions of audio amplification techniques and architectures.
10.Good ‘hearing’ skills, i.e. knowing how bass should sound, what can be expected from speech artifacts, buffer underruns, clipping etc will be a plus.


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