CMU Sphinx 4 to Asterisk or Freeswitch

$750-1500 USD

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超过 12 年前

$750-1500 USD

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We need someone with experience with the CMU Sphinx 4 toolkit, and Asterisk or Freeswitch. We seek professional real-time dictation of SIP channels to text, with the ability to continue to tune or improve quality of transcription. Requirements: 1) The system shall be hosted on Amazon EC2 2) The system language shall be English 3) The system shall allow the users to improve the basic accuracy in some way, such as asking the user to read from a known script of training words. 4) The system shall translate multiple simultaneous SIP channels. Please do not submit quotes if you do not have the prerequisite experience with CMU Sphinx, or have not done similar work in acoustic recognition.

Algorithm

Face Recognition

Machine Learning (ML)

Pattern Matching

Scientific Research

项目 ID: 1320649

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@akshuja

My research is in machine learning, and some of it is in the area of speech recognition. I do have experience with CMU Sphinx 4 (in particular with HMM's which I believe is what you are going to need), but not so much with Asterisk. I have also been doing system-level programming in linux for about 11 years so I should be able to figure out what we need for Asterisk and how it hooks into the voice recognition engine. To be realistic, for this project to be truly effective, it would need to go beyond statistical learning and do some linguistic processing, but this would go well beyond your limits (perhaps in the future). Let me know if you are interested.

$1,500 USD 在60天之内