avidemux suddenly not allowing me to save audio in 16 bitrate

[leesa]

hi friends

i was wondering if somebody could help me. i am trying to make a voice over for part of my video. what i generally do is i save my original audio file from teh video. i record my voice for part of the video. then i add the voice part of the audio and the end part of my original video audio (because i don't have to instruct the whole time) and i join them together in wavepad sound editor, then i add the joined audio files (the voice over and second half of original audio) back into the original video file. i watched in a youtube video that it's important to save the original audio file in 16 bitrates or lower or else you get artifacts in your audio. up until this point i have had the option to do this avidemux. but for some strange reason, suddenly the minimum number of bitrates i can save is 56. i'm using mp4 files because there's less issues when i join video files together in avidemux. the strange thing is taht i took a video file which i had previously saved a 16 bitrate audio file from and tried to save the original audio file again in 16 bitrate but that option is no longer available. now the lowest audio bitrate i can save in is 56. does anybody know why this is happening with avidemux and how i can fix it? thank you for any replies in advance. leesa

[davexnet]

I'm not sure of your terminology. For example, take a CD quality audio file.
An average to low mp3 made from it may have the following properties:
44.1 KHz (the amount of digital samples per second - same as the CD) 16 bit (the bit depth of each sample - same as the CD)
128 kbps (the bitrate, common for mp3's, far lower than the original CD - this is where the drop in quality comes from).

Are you getting the specifics mixed up ?

The best way to save your existing audio is as it is originally - don't recompress it.
When you edit in the audio editor later, it is temporarily uncompressed. Do all your editing, then when done save as
uncompressed ( a WAV /PCM file) . Then re-add it to your video in avidemux and recompress to a format of your choice.

[Cornucopia]

To assure the best quality, it is best to maintain the equivalent of CD level - 16bits, 44.1kHz, LPCM - (or better) throughout your process until the very end. Then export to your desired compressed format (e.g. mp3, mp4/aac). Also best to have all clips match in their specs, so no unnecessary conversions. Note that the BITRATE for that (stereo) CD quality works out to 1536kbps, well above the common 128 used by mp3, etc.

Scott

[leesa]

hi, thank you so much for your responses. this is where i got the 16 bit thing from on the video on youtube: https://www.youtube.com/watch?v=9rFAKH3V5p8

[Cornucopia]

You misunderstood their talk.

"16bit" = The bitdepth of each individual sample of each channel in audio file (sampled at 48000 or 44100 samples/second). Versus 24bit or 32bit, or even 8 or 20 or 48/64bit (the last is very unlikely).
16bit is equivalent to CD's level of bitdepth. Above 16bit would be BETTER potential for quality (lower noise floor and/or finer volume gradations). Below 16 bit will give lesser and lesser quality. 8bit is considered noticeably worse to just about everyone.

The problem with the higher bit depths is that many programs have not been updated in their coding and are using the old capabilities (which only went up to 16bit), and so they don't know how to correctly handle higher bitdepths. Nothing wrong with the files, just (some of) the apps

Bitrate=speed at which data flows by (or needs to flow by).

Again, CDs calculate their bitrate thus:

2 channels * 16bits / channel / sample (uncompressed/linear) * 44100 samples /second = 2 * 16 * 44100 = 1411200 bits/second = 1378kbps.

DV = 2 * 16 * 48000 = 1500kbps (made a mistake earlier: 1536 was /1000, not /1024).

Common mp3 files compress that 1378 down to ~128kbps. That almost 11:1.

While there is a lot of variability WRT which type of compression (aka "codec") is used, and HOW (settings, relative efficiencies), which can affect the quality. Overall, a lower bitrate is going to be worse quality and a higher bitrate is going to be better quality.

If your benchmark for reasonable casual quality of a compressed file is 128kbps for mp3, it might equate to ~96 or 64kbps for aac/mp4.

Scott

[leesa]

hi scott
thank you so much for that detailed explanation. i've read the responses several times but honestly, i'm struggling to figure out what you mean exactly. you guys are talking about so many concepts that i am totally unfamiliar with. where do you learn all about concepts like samples and channels and bit depths and CDs? i'm SO new to video editing.

what i was doing before seemed to be working ... saving the original audio file in the mp3 LAME configuration, with the bitrate mode CBR and then choosing the bitrate of 16 from teh drop down list. then when i used the wavepad editor to do my explanation bit of the video, i didn't even try to make any modifications with these types of configurations in the wavepad editor. i just saved my voice in mp3 format.

so can i please ask you .... when you open an mp4 video file in avidemux, and you save the original audio file from the video using the mp3 LAME configuration, with the bitrate mode CBR .... then which bitrate would you choose from the list (it goes from 56 to 224) (with default quality being "2"), given that i will then make a voice audio file in wavepad sound editor, to be joined with part of the original audio file later on and put it back into the original video using avidemux ....... in order not to get any artifacts that the guy was talking about in that youtube link i posted?

thank you so much for your patience in advance.
leesa

[hello_hello]

When audio is sampled (lots of times per second), each sample needs to be assigned a value. The greater the number of values which can be assigned, the more accurate it can be. That's the bitdepth. When you save a wave file, 16 bit (or more) is generally used. Whoever told you it should be 16 bit or less was wrong. The bitdepth only applies to lossless formats such as wave. You can't set a bitdepth for MP3 (explained later).
But 8 bit, 16 bit, 24 bit etc..... that's the bitdepth.

The "lots of samples per second" part is the bitrate. As a very general rule, the higher the bitrate the better the quality for lossy audio (ie MP3). For LAME MP3 a constant bitrate of 128kbps is often used for "good enough" conversions, especially for soundtrack audio. It'd be plenty for just speech. For a higher quality you'd select a higher bitrate. Well..... the LAME constant bitrate encoding methods also have a quality setting. I think the default for the LAME encoder is q3. The quality setting doesn't effect the bitrate (they're independent). Rather, the quality setting effects how much time the encoder spends encoding. In theory, q2 at 128kbps should be capable of better quality than q3 at 128kbps. And q1 and q0 might be better again. They're very slow though. I think most people stick with q2 or q3.
You can't set a bitrate for wave files (it's fixed).
But 128kbps (kilobits per second), 192kbp/s, 256kbp/s etc. That's the bitrate.

Back to bitdepth...... lossless formats such as wave files have a bitdepth, and 16 bit is "CD quality". Lossy formats such as MP3 don't have a fixed bitdepth, just a bitrate. So when you convert a wave file to MP3 you can select the bitrate, but not the bitdepth, because there isn't one (it's a long explanation as to why which I don't fully understand myself).
When you convert an MP3 to a wave file you're converting the MP3 audio (with no fixed bitdepth) to a wave file with a fixed bitdepth. Once again, the higher the bitdepth the more accurate the conversion from MP3 to wave can be, but 16 bit is fine.

The upshot of it all is pretty simple though.... when you convert to MP3 you pick the bitrate (and optionally a quality/speed setting). When you convert to wave, pretty much all audio programs default to outputting a 16 bit wave file and the bitrate is fixed, so for wave files you generally don't need to think about it. Just "convert to wave".

I haven't used wavepad and I'm not sure what you're doing exactly, but keep in mind MP3 is "lossy". It throws information away, so every time you convert to MP3 (or AAC or AC3 etc) you potentially lose quality. Wave files are lossless, so aside from the bitdepth choice you can convert to wave lots of times and not lose quality. Going from wave to MP3 to wave to MP3 again is two lossy conversions. You should always try to keep that to a minimum. Ideally, you'd work with lossless audio while editing etc and only convert to MP3 once for the final output.

when you open an mp4 video file in avidemux, and you save the original audio file from the video using the mp3 LAME configuration, with the bitrate mode CBR .... then which bitrate would you choose from the list (it goes from 56 to 224) (with default quality being "2")

I'd probably stick with a minimum of 128kbps, and either Q2 or Q3, but what format is the original audio (AAC or AC3 etc)? If it's already in a "lossy" format, which it probably is, converting it to a different "lossy" format throws more information away, so if file size isn't an issue, using a higher bitrate wouldn't hurt.

[Cornucopia]

If quality MATTERS to you, you should do this:

MP4 original file -> Editing -> Save as 16bit WAV (lossless) -> (whatever other stuff you are going to do to the file) -> Re-open and re-save as 128kbps MP3 or similar.

MP3 can go up to 320 for regular stereo files. I would suggest you do a test and, creating alternates stepwise, raise your 128kbps up to a level you find acceptable (transparent?) and then use that for your final output bitrate (and no higher).

Scott

[leesa]

thank you so much for the effort you guys put in to helping me with this. i really appreciate it.