Hi so got a lil question here, can anybody advise me on a program that will scan folders of mp3 and auto remove the ones that will not play, Iv got way to many to do it my hand 1 by 1. Thanks for any info aswell. peace.
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The livestream doesn't end (for now, and if ended, I think will be erased of YouTube, that's my reason to download), actually, the video is still in transmission, you know, like the streams of the NASA or streams of channels news. The detail is that the transmission lasts about 10-11 hours, and the transmission has lasted about 3 days. So it was a matter of time before the first concerts were no longer available to watch on the broadcast.
This is the video: https://www.youtube.com/watch?v=rE6QI0ywr0c
I want to download some concerts, but the things that I wanted, are disappearing with the passing of time. Right now, I'm only interested in the Disclosure concert. His concert starts at approximately -3:38:12. I mention it in case someone wants to help me.
I was trying this command, but only appear a text that i don't understand (I'll post it in the comments, all the images with his info). The command is this → yt-dlp.exe -f (bestvideo+bestaudio/best) "link" --postprocessor-args "ffmpeg:-ss 00:00:00 -to 00:00:00" -o "%(title)s_method1.%(ext)s"
The idea of that command emerged on this ideas
https://www.reddit.com/r/youtubedl/wiki/howdoidownloadpartsofavideo/
https://github.com/yt-dlp/yt-dlp/issues/686
Also, I was trying to do this How do you use youtube-dl to download live streams (that are live)?, but I can't get the HLS m3u8 URL in Chrome and Chrome Dev (yes, I go to F12 (Chrome Developer Tools) - Network and I write m3u8, I didn't find anything.
I should mention that I don't have extensive knowledge on codes and yt-dlp. I only learned the necessary to download videos, you know, yt-dlp.exe -F (link) and then yt-dlp.exe -f (numbers of resolution and audio) (link).
So if you recommend any programs or commands, please let me know as precisely as possible.
Any new info I'm gonna update in the comments.
PS: sorry for my english
I have more than 600 MP3 files, I just want to listen a part of each MP3 file. I know some MP3 cutter tools can do this job manually. Repeating the similar job more than 600+ times via MP3 cutter tool, that must be a very very boring and hard thing. So I need to write a program to finish this job automatically.
The specified part in each MP3 file have the almost same start/end part (5~8 seconds), if I can write a program to detect the start point and the end point, my question is nearly to be resolved.
PS:
1) All the MP3 files are English learning podcast.
2) C or JAVA solution is prefered
Thanks
I find the specified part time was already listed in MP3 metadata, so my problem is resolved:
1) Read MP3 metadata, get the start time and end time, thanks to https://stackoverflow.com/a/21746702/3367812
2) cut MP3 file, thanks to https://stackoverflow.com/a/10056024/3367812
This has been asked in several places.
You can follow this answer, which shows how to cut a wave file, to come up with your own program.
cutting a wave file
I'm interested in creating a drm type program. I want to put code into an mp3 file that checks online to see if it is licensed out. The online aspect is not something I'm concerned with at the moment, but I've looked online for some resource on doing this, but I've found nothing really that useful. There are people talking about viruses, but I want this to be intentional and not malicious. Simply when you play the the mp3 it quickly checks online to see if the license is actually allowed to play. If it is it plays if it isn't it gives an error and says either you need to log on or buy the song.
I'm pretty sure this is impossible. mp3 files can't contain executable code. They could contain byte data that could represent code, but the mp3 player would need to interpret it as such and execute it instead of trying to play it as music, which no player does. You would have to program your own player to do this, but since there's no way to constrain an mp3 file to only one player (again, because they can't contain code), you'd have to create your own competing standard. Further, there would be nothing preventing someone from converting this mp3 to a normal non-constrained mp3 and sharing it.
From a less technical standpoint, as a user (who despises DRM) it would irk me to no end if an mp3 was checking up on whether I was "allowed" to have it, and even more so if I was required to be online in order to listen to it, to the point that I would just delete it and never buy/download/steal anything in that format again.
(Does this count as an answer? It was too long for a comment.)
Its surely possible but very difficult. Im not very experienced in coding but using discord i was able to turn an mp3 file into a txt file, But the problem is that it becomes strange characters
So all I need is a simple function that sets it up (eating incoming PCM RATE (for example: rate near to 44100) It's channels (for example: 2) and -bits (for example: 16) and desirable 128 kb\s rate) and another one that takes PCM data and encodes it into pure MP3 frames.
I know it looks like a silly homework task but I assure you - it is not.
I hope it will be of help to all C++ developers starting with MP3s.
So can anybody please help me with that?
See the example I gave in your other question for the basic usage of Lame. It should contain everything you need.
It's a long time since i messed with this, but Lame lib contains all you need to do it, check out lame.h , there is some test code you could look into.
Personally, I'd be looking at ffmpeg's libavcodec. There's an example file containing an audio_encode_example which ought to be more or less what you're looking for.
See also this question.
Note that not all ffmpeg packagings include mp3 codec support (patent issues) by default, although there's usually some simple way of enabling it.
GStreamer should definitely be able to handle this.
If you are on Windows you can use the GStreamer Winbuilds to get started.
Basic Info:
Programming Language - C++
Platform - Windows
Audio Formats - wav and mid
I recently finished a game and was fooling around with figuring out the best way to upload it to a file hosting site. I eventually decided on using 7zip's self-extracting feature. However, I think the mistake I made was that instead of just copying what I needed to another folder and zipping that up for the distribution (i.e., not copying source files, etc.) I rearranged the actual folder that held all of my source files etc. and split it into 2 sub folders for the C++ files, and then everything else (that folder being the one that got zipped up.)
I tested downloading it and playing it and it worked fine. However, I went back because I decided to change the background music and that's when the problem started happening.
To sum the problem up, Mix_PlayMusic() is being called and is working correctly. However, for some reason no sound is playing (and neither are any of the sound effects called from Mix_PlayChannel()). The odd thing is that you can hear the music when Mix_FadeOutMusic() is called. I also have a sound toggling feature, but after thorough testing I've come to the conclusion that it isn't the problem.
I finally decided to create a completely new project and just bring all of the files I needed into that project in the same "organization" that they were in originally. However, the problem is still there.
I have no idea what's wrong. The files are being loaded in fine, it's just that when the music is supposed to be playing (and according to testing it is), it's not playing. This also applies to sound effects.
Edit: I actually wrote a test for each game loop for whether the music is playing and apparently the music is playing. It's just that for some reason it isn't being heard.
This could be a number of things. It could be an issue with the SDL_Mixer library you have, so you could try getting it again to rule that out. Your volume may have somehow got set to zero somewhere, so I would check the volume as a test. And the final thought would be that the source sound file you are playing is incompatible in some way (not likely if you can play it in another sound player, but possible). Besides those suggestions I don't believe I can help you any further with the data you have provided.