Transcriber | video transcription project to make YouTube videos audio | Video Utils library
kandi X-RAY | Transcriber Summary
kandi X-RAY | Transcriber Summary
A video transcription project to make YouTube videos audio content available as text
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Top functions reviewed by kandi - BETA
- Main function .
- Convert audio into string .
- Generates a list of words from candidate transcripts .
- Convert a samplerate to a desired sample rate .
- Write metadata to a JSON file .
- Load a model .
- Write text to a text file .
- Convert metadata to string .
Transcriber Key Features
Transcriber Examples and Code Snippets
Community Discussions
Trending Discussions on Transcriber
QUESTION
Continuing my earlier question. Now I have one dataframe, where I added a new column 'New' with values 1 through 150 for each new file, that can be used as an index if required. I figured it would be easier to make the loop for every file separately. And now I am not sure how to proceed. Let me provide the code, the explanation of the task at hand, and some thoughts how to move on.
New FileName Transcriber Transcription 1 612_000002.wav 100% (80/80) Are we starting off? 612_000002.wav 100% (50/50) shall we starting on 612_000002.wav 100% (2/2) fast mode 612_000002.wav 100% (258/259) Go and start it up 612_000002.wav 100% (20/20) Are we starting off? 612_000002.wav Quartznet there was not inl 612_000002.wav Transducer_M don't start again 612_000002.wav Transducer_L do we start again 2 612_000003.wav 100% (258/259) here we go, hey well woah woah woah 612_000003.wav 100% (23/23) evening gulf air 612_000003.wav 100% (32/32) And as the 1st group reached the bottom of the... 612_000003.wav 100% (80/80) Happy to go off here, woah woah woah 612_000003.wav 100% (10/10) Go boom we'll just 612_000003.wav Transducer_L anything off yeah i'm willing we'll just 612_000003.wav Transducer_S having gone here on will and wolf is ...ANSWER
Answered 2022-Mar-20 at 21:06I assumed you still wanted to know the transcriber of both ground truth and hypothesis. In my approach, I disregarded the new
column, as I don't really see what the purpose is.
First, we join the dataframe with itself, merging only on filename
(thereby creating a cross-join of 113 rows, for this example).
QUESTION
I am struggling to join two dataframes by index (I've made column FileName an index for both tables) which look like this:
Table 1
FileName Transcriber Transcription 612_000002.wav 100% (80/80) Are we starting off? 612_000002.wav 100% (50/50) shall we starting on 612_000002.wav 100% (2/2) fast mode 612_000002.wav 100% (258/259) Go and start it up 612_000002.wav 100% (20/20) Are we starting off? 612_000003.wav 100% (258/259) here we go, hey well woah woah woah 612_000003.wav 100% (23/23) evening gulf air 612_000003.wav 100% (32/32) And as the 1st group reached the bottom of the... 612_000003.wav 100% (80/80) Happy to go off here, woah woah woah 612_000003.wav 100% (10/10) Go boom we'll justTable 2 is similar and looks like this:
FileName Transcriber Transcription 612_000002.wav Quartznet there was not inl 612_000002.wav Transducer_M don't start again 612_000002.wav Transducer_L do we start again 612_000003.wav Transducer_L anything off yeah i'm willing we'll just 612_000003.wav Transducer_S having gone here on will and wolf isSo I've looked into concat, merge, and join. But they don't seem to yield the output I am looking for. What I would like to have is all values from both tables for filename1, all values for filename2 and etc. Basically, adding rows from table2 to table1. Is there any way around it? Thank you <3
...ANSWER
Answered 2022-Mar-19 at 17:46Use:
QUESTION
I have partly faulty transcriptions, which I want to clean-up:
...ANSWER
Answered 2022-Feb-04 at 14:23If I understand this correctly, you want to completely exclude Utterance strings with double brackets? If yes, I think this solves your problem
QUESTION
On the local server this Python app works perfectly, however, when I set up this Python app on my server where my site is hosted, index.html
loads and that means Python is well set up, but when I click the Transcribe button
, after as I select the language
and .wav file
, an error occurs:
ANSWER
Answered 2021-Aug-08 at 13:27I cannot comment in the above discussion due to less points. No it is not due to server response time.
Did you try with
transcript = recognizer.recognize_google(data)
instead
QUESTION
im trying to use my python app to transcribe multiple files in a folder and speed up the process. At present I am able to do it one file at a time -
...ANSWER
Answered 2021-Mar-03 at 16:48have you tried running this script multiple times? you could write a wrapper that launches this script in a subprocess kinda like this:
QUESTION
In my live phone speech recognition project Python's asyncio
and websockets
modules are used basically to enable data exchange between client and server in asynchronous mode. The audio stream which to be recognized comes to the client from inside of a PBX channel (Asterisk PBX works for that) via a local wav
file that cumulates all data from answering call until hangup event. While conversation is going on, an async producer pushes chunks of call record (each of them no larger than 16 kB) to asyncio queue, so that a consumer coroutine can write data to buffer before sending to the recognition engine server (my pick is Vosk
instance with Kaldi
engine designed to connect using websocket interface). Once the buffer exceeds a specific capacity (for example it may be 288 kB), the data should be flushed to recognition by send
function and returned (as a transcript of the speech) by recv
. The real-time recognition does matter here, therefore I need to guarantee that socket operations like recv
will not halt both coroutines throughout websocket session (they should be able to keep queue-based data flow until the hangup event). Let's take a look at whole program, first of all there is a main
where an event loop gets instantiated as well as a couple of tasks:
ANSWER
Answered 2021-Mar-05 at 09:06If I understand the issue correctly, you probably want to replace await self.do_recognition()
with asyncio.create_task(self.do_recognition())
to make do_recognition
execute in the background. If you need to support Python 3.6 and earlier, you can use loop.create_task(...)
or asyncio.ensure_future(...)
, all of which in this case do the same thing.
When doing that you'll also need to extract the value of self._buffer
and pass it to do_recognition
as parameter, so that it can send the buffer contents independently of the new data that arrives.
Two notes unrelated to the question:
The code is accessing internal implementation attributes of queue, which should be avoided in production code because it can stop working at any point, even in a bugfix release of Python. Attributes that begin with
_
like_finished
and_unfinished_tasks
are not covered by backward compatibility guarantees and can be removed, renamed, or change meaning without notice.You can import
CancelledError
from the top-levelasyncio
package which exposes it publicly. You don't need to refer to the internalconcurrent.futures._base
module, which just happens to be where the class is defined by the implementation.
QUESTION
I'm trying to transcribe Telegram audio messages, using Mozillas speech-to-text engine deepspeech.
Using *.wav
in 16bit 16khz works flawless.
I want to add *.ogg
opus support, since Telegram uses this format for it's audio messages.
I have tried pyogg and soundfile so far, with no luck.
Soundfile could outright not read the opus format and pyogg is a pain to install without conda. I had really weird moments where it literally crashed python.
Right now, I'm trying librosa with mixed results.
...ANSWER
Answered 2020-Jun-25 at 10:51librosa returns an array floats in range -1.0 to 1.0. In int16 the maximum value is 32767. So you have to multiply to scale the signal, then convert to int16.
QUESTION
I try to create a speech-to-text transcriber with Python and Google cloud.
Unfortunately, it always gives me the error "ModuleNotFoundError: No module named 'google'".
I installed plenty of packages, google-cloud, google-cloud-storage and many more and nothing seems to work.
I also looked up "How to install Python packages", but after following the respective links, which I though must be the right ones, it still did not work.
The following is part of the code I use:
...ANSWER
Answered 2020-Apr-18 at 18:08Have you looked at this page: https://cloud.google.com/speech-to-text/docs/quickstart-client-libraries#client-libraries-install-python ?
And have you tried pip install --upgrade google-cloud-speech
?
QUESTION
I have a table that has a pkey and a pvalue column. One pkey value is "moderators" and its corresponding pvalue is a JSON string. Currently the string looks like this:
...ANSWER
Answered 2020-Mar-16 at 12:06you can do like a´akina already said, use:
Community Discussions, Code Snippets contain sources that include Stack Exchange Network
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Install Transcriber
[ ] YouTube video transcriber
[x] A video transcriber
[ ] Record your voice and transcriber afterwards
[ ] Transcribe while speaking
clone the repository $ git clone https://github.com/erolrecep/Transcriber.git
Install SoX Swiss army knife for audio processing things $ brew install sox # For Mac Os X $ sudo apt install sox # For Ubuntu
create a new Python virtual environment $ conda create --name transcriber python=3.6 $ conda activate transcriber $ conda install tensorflow==1.13.1 # if you have GPU, then install *conda install tensorflow-gpu==1.13.1* (surprisingly I like this version of tensorflow :) ) $ pip install youtube-dl deepspeech==0.7.4 # if you have GPU, then install *pip install deepspeech-gpu==0.7.4*
Download pre-trained DeepSpeech models from here This repository uses 0.7.4 version of the DeepSpeech, you can try the same setup with newer models. You need to download 0.7.4 pdmm file If you want, you can also download and load scorer provided by Mozilla, scorer
Now, the virtual environment is ready, the next step is running the project. For your convenience, I provided a sample .wav file so you can test your setup if it's working. Also, you can download audio files from here $ python run.py # This will read audio files from the *audio_locations.txt* file. $ python run.py -a audio_files/sample.wav # This will only run inference on this input .wav file
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