stream output & file processing

[kqvanity]

Looking for a feature adjustment

• [ ] [feature] Can continuous stream be a single chunk of text that gets updated with new transcriptions, instead of printing on new lines

• [x] [Bug] File processing doesn't seem to work. I pass an 8M flac audio file, so gstt takes it time, but output nothing at the end

[giulianopz]

Hi @kqvanity,

No problem for the feature request: I've already thought to implement it and I'm going to do it in the next few days.

As to the problem you mentioned: could you please provide me with some more info to reproduce the issue? I would need at least the exact command you typed and the FLAC file itself (upload it somewhere, e.g. Google Drive). But bear in mind that if you don't pass the right sample rate of the audio file, the Google service won't be able to properly transcribe the input audio.

[kqvanity]

No problem for the feature request: I've already thought to implement it and I'm going to do it in the next few days.

Thanks for your time

bear in mind that if you don't pass the right sample rate of the audio file

I've tried to grab the sample rate using mediainfo then explicitly pass, but getting the following error

  gstt --sample-rate --file Using\ Wget\ As\ A\ Download\ Manager.flac
flag provided but not defined: -sample-rate
Usage:
    gstt [OPTION]... --interim --continuous [--file FILE]

Options:
        --verbose
        --file, path of audio file to trascript
        --key, api key built into chromium
        --language, language of the recording transcription, use the standard webcodes for your language, i.e. 'en-US' for English-US, 'ru' for Russian, etc. please, see https://en.wikipedia.org/wiki/IETF_language_tag
        --continuous, to keep the stream open and transcoding as long as there is no silence
        --interim, to send back results before its finished, so you get a live stream of possible transcriptions as it processes the audio
        --max-alts, how many possible transcriptions do you want
        --pfilter, profanity filter ('0'=off, '1'=medium, '2'=strict)
        --user-agent, user-agent for spoofing
        --sample-rate, audio sampling rate

Audio file

[giulianopz]

Hi @kqvanity,

there was a typo with the sample rate that is now fixed, sorry.

Anyway, I looked at your file and I found out that it is composed of two channels. You can convert it to 1 channel with: ffmpeg -i afogr3.flac -ac 1 mono.flac. I added a sidenote about it in the readme linking the ffmepg docs that explains this command: https://trac.ffmpeg.org/wiki/AudioChannelManipulation.

Then, use the new flag to write output on the same line as follows: gstt --interim --continuous --subtitle-mode --file mono.flac

[kqvanity]

Great. it does work now.

Not aware of the complexity involved, but was wondering if it could be processed on the fly. Be it audio/video stream (dual-channel live streams which is only processed IRT). performance-wise, re-encoding would take place either way.

-subtitle-mode

Not available yet. Only continuous stream.

[giulianopz]

Make sure you are using the latest version (tag v0.1.0).

[kqvanity]

my bad. The binary was go-gsttt It does eliminate the extra noise, however, i faced two issues

• the current raw text basically keeps overwriting the current terminal header.
• reading currently running stream from rec, no longer outputs anything (i've tried it with a youtube video, flac file running using mpv)