Right now, our transcribers manually do the typing. If we have a ASR based system, which produces even a 80% accurate transcript of the audio, we can improve our efficiency manifold.
We have a lot of data, audio files and their corresponding transcript. That data can be used to train a DNN based system. A software solution or framework which can do that would be ideal for us so that we can keep feeding in the data and continuously improve our model.
See https://github.com/pannous/caffe-speech-recognition (still under development)
At one time I invested in a medical transcription business that experimented with voice-to-text technologies but couldn't generate any additional efficiency in transcription by using technology. What worked in improving efficiency was the transcriber's knowledge of medical terminology and medical field, matching the same transcriber with the same speaker as transcriber became acclimated with the nuances and tones of the speaker, and letting transcriber handle only certain type of transcription.
Transcription is a manual operation and Toyota Way is a good way to maximize efficiency.
Our goal with ASR is not replace the typists, but to assist them. On an average typing takes 4X the duration of actual time. An 80% accurate ASR may reduce it to 2X, which in turn should translate to faster turnaround and improved throughput.