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<title>TaSTT.git, branch v0.13.0</title>
<subtitle>Free self-hosted STT for VRChat.</subtitle>
<id>https://git.yummers.dev/TaSTT.git/atom?h=v0.13.0</id>
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<updated>2023-06-30T07:15:50+00:00</updated>
<entry>
<title>Merge pull request #3 from jsopn/fix-gpu-device-index</title>
<updated>2023-06-30T07:15:50+00:00</updated>
<author>
<name>yum-food</name>
<email>114886918+yum-food@users.noreply.github.com</email>
</author>
<published>2023-06-30T07:15:50+00:00</published>
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<id>urn:sha1:ab1d4499e1b3198b387a1e2d7333a93f694cdfae</id>
<content type='text'>
Set GPU device index in whisper model</content>
</entry>
<entry>
<title>fix: set gpu device index in whisper model</title>
<updated>2023-06-30T00:15:30+00:00</updated>
<author>
<name>jsopn</name>
<email>github@jsopn.com</email>
</author>
<published>2023-06-30T00:15:30+00:00</published>
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<id>urn:sha1:95927745b3d3b2ac566f1cfd634a40e760ed29cb</id>
<content type='text'>
</content>
</entry>
<entry>
<title>Fix race condition around audio frames dropping</title>
<updated>2023-06-29T06:50:34+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-29T06:50:34+00:00</published>
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<id>urn:sha1:3b10d3ab3073af2ed716d1607bb92394bb8817fc</id>
<content type='text'>
onAudioFramesAvailable would bail out if audio_state.audio_paused is
set, preventing frames from being dropped. This would cause
transcriptions to get repeated sometimes.

Now that frame dropping code always runs.

Also adjust the code structure of the keyboard/VR input handlers to be
more similar.
</content>
</entry>
<entry>
<title>Bugfix: commit no longer wipes out audio buffer</title>
<updated>2023-06-29T05:18:18+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-29T05:11:46+00:00</published>
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<id>urn:sha1:b1efbf5ce1ebd584796d4a57cf9c7b6517f91fac</id>
<content type='text'>
Audio data is stored in chunks of frames, not in individual frames.
When I commit a transcript, I want to get rid of the portion of the
audio data responsible for that particular transcript. I have code that
does this, but it was dropping a slice of the list assuming that each
sample is stored individually.

Extra fun: Because we have to decimate mic frames, we have to convert
between whisper frames and mic frames to drop the correct amount of
audio data.
</content>
</entry>
<entry>
<title>Add profanity filter</title>
<updated>2023-06-29T04:24:56+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-29T04:24:56+00:00</published>
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<id>urn:sha1:bdaeb1911297d7901a12e3ac51b38c3463789279</id>
<content type='text'>
Add toggle to UI to enable a profanity filter. It replaces vowels in bad
words with asterisks.

Bugfix: filters now apply to OBS
</content>
</entry>
<entry>
<title>Add toggle for debug mode</title>
<updated>2023-06-29T03:35:10+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-29T03:35:10+00:00</published>
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<id>urn:sha1:ff7eb3c212195af71cd0ce4a3cd0c9a081d6ebda</id>
<content type='text'>
Most transcription output is now gone by default. Users can enable a
more verbose output by toggling `Enable debug mode`.

Bugfix: Toggling off transcription would reset audio state, frequently
resulting in the loss of the last few words spoken.
</content>
</entry>
<entry>
<title>Add UI for fuzzy commit threshold</title>
<updated>2023-06-27T23:01:16+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-27T23:01:16+00:00</published>
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<id>urn:sha1:6638993e313773ba6ca8bdb6d7690b798d41f0d4</id>
<content type='text'>
Recap: In the STT there's an algorithm that tries to determine when a
transcript is "stable" enough to commit. If that is too loose, then
accuracy suffers; if too strict, then the audio buffer eventually fills.

To mitigate the problem, I check whether the last N transcripts are
within some edit distance (Levenshtein edit distance) of each other. The
fuzzy matching lets us forgive small instabilities, like differences in
uppercase/lowercase or punctuation, while rejecting large instabilities.

The default value of 8 seems to be in the sweet spot of accuracy &amp;
performance, but it will likely be tuned in the future.
</content>
</entry>
<entry>
<title>Adjust commit logic to use fuzzy string match threshold</title>
<updated>2023-06-27T22:38:42+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-27T22:35:30+00:00</published>
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<id>urn:sha1:241813a5af11093c6b86e70ada729788c1f0dee6</id>
<content type='text'>
... instead of simple equality.

TODO: add UI for threshold.

Bugfix: Frame::onAppStop() joins the OBS app thread.
</content>
</entry>
<entry>
<title>Add ability to preserve transcript while using push to talk</title>
<updated>2023-06-27T22:16:06+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-27T22:16:06+00:00</published>
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<id>urn:sha1:cf75998dab6db1b1d21ca06bde18a56b5e896937</id>
<content type='text'>
This is useful when streaming. Occasionally the STT can get into
a bad state, and manually segmenting clears it up. However doing so
would clear your accumulated transcript, which isn't always desired. Add
ability to preserve the transcript.

A small wrinkle: the new commit logic requires N consecutive identical
windows before committing. To make this feature play nicely with it, I
had to forcibly commit any preview text that hasn't yet been committed.
Failing to do this would usually cause short utterances / the most
recently said stuff to get wiped out.
</content>
</entry>
<entry>
<title>Add grey background to browser src</title>
<updated>2023-06-27T08:50:40+00:00</updated>
<author>
<name>yum</name>
<email>yum.food.vr@gmail.com</email>
</author>
<published>2023-06-27T08:47:53+00:00</published>
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<id>urn:sha1:8ff761153b7de23a9556f2af179ab3bf4b9849a5</id>
<content type='text'>
Should improve legibility.

* Update README
</content>
</entry>
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