Use Pre-Recorded Tracks as input/change active track

[edrihan]

It's really great how you can record in the command line, however there are some issues with that. Firstly, the transcription happens per-file which completely breaks the flow. Why can't it batch them once you're done recording? But that is neither here nor there.. as recording in the cli is just not what we are used to in the audio field. There are free (to use) programs like Reaper and Audacity, and paid DAWs which are enjoyed by people doing recordings.

So people have their own workflow for recording audio... and it's not that. You want control over what happens in your audio.

Description of proposed feature

I want to be able to swap audio files easily. I want to be able to pass an audio file, relative to a folder in manim/media/, for example:

• manim/media/voiceovers
• manim/media/recordings
• manim/media/audio

I think the best way for this to work would probably be to have a small json file which has the section name corresponding with the file path. That way, a check of that file can be performed to determine what needs to be done for that section. Then if I update a section later, I want to be able to change manim links to the new file, and have the manim command automatically create transcription and metadata for it.

If a speech file doesn't exist, I want to be prompted whether I want to record it, or whether I want to supply a path. If I choose a path outside the manim directory, I want the option to copy the file into the right place. Maybe it adds the hash to the end of the filename too?

Now it can rebuild the data in manim/media/voiceovers/cache.json where it needs to

How can the new feature be used?

Ideally you just have the section text, as the key, and the filepath as the value.

Additional comments

The other feature I want is related to this, but I don't think it deserves its own issue as for the moment, using your own audio seems difficult, but if this feature request goes through, I would love to have the audio not be hardcoded to mp3. Normally in audio you would use WAV, FLAC, or equivalent lossless codecs. We can let the internet destroy the audio later, but on the master copy you don't use mp3s. I know that it made sense to use mp3s as that is probably what the ai models put out. But for our own audio we do not want lossy compression.

This will be more in line with normal audio workflows and will make this so much easier to use. Then we'll have more of peoples' voices and less of AI, which is a good thing.

Issue #88 seems related