Adding support for `microphone` streaming within pipeline. (#15046)
* Adding support for `microphone` streaming within pipeline.
- Uses `ffmpeg` to get microphone data.
- Makes sure alignment is made to `size_of_sample`.
- Works by sending `{"raw": ..data.., "stride": (n, left, right),
"partial": bool}`
directly to the pipeline enabling to stream partial results and still
get inference.
- Let's `partial` information flow through the pipeline to enable caller
to get it back and choose to display text or not.
- The striding reconstitution is bound to have errors since CTC does not
keep previous state. Currently most of the errors are we don't know if
there's a space or not between two chunks.
Since we have some left striding info, we could use that during decoding
to choose what to do with those spaces and even extra letters maybe (if
the stride is long enough, it's bound to cover at least a few symbols)
Fixing tests.
Protecting with `require_torch`.
`raw_ctc` support for nicer demo.
Post rebase fixes.
Revamp to split raw_mic_data from it's live chunking.
- Requires a refactor to make everything a bit cleaner.
Automatic resampling.
Small fix.
Small fix.
* Post rebase fix (need to let super handle more logic, reorder args.)
* Update docstrings
* Docstring format.
* Remove print.
* Prevent flow of `input_values`.
* Fixing `stride` too.
* Fixing the PR by removing `raw_ctc`.
* Better docstrings.
* Fixing init.
* Update src/transformers/pipelines/audio_utils.py
Co-authored-by: Anton Lozhkov <aglozhkov@gmail.com>
* Update tests/test_pipelines_automatic_speech_recognition.py
Co-authored-by: Anton Lozhkov <aglozhkov@gmail.com>
* Quality.
Co-authored-by: Anton Lozhkov <aglozhkov@gmail.com>