many of the video and audio conversion and almost every video and audio editing program can do that, if there is a mp3 encode codec on the machine like "lame". I would use gold wave http://www.goldwave.com/release.php they let you try it for a while, and it can batch that process along with filter batching. It should be able to handle 2-4hour files.
Super the video converter can also do that, it might not need a MP3 encode codec, just turn off the video, drop one or many files in, and push the button. (super comes with junkware that should be avoided on install) Super can also do video convert.
Many video editing programs can "export" the audio seperate, and do not have to have a video nessisarily, Just like Super any video converter that has some parameters could probably do it, like avisynth, and ffmpeg. Mabey even windows movie maker would allow that?
One for all, on one side having a program that has to many features is not easy to use, but if you already have something on the machine for doing "Video with Audio" it might be able to do audio only, fast and easy. it may just need a MP3 codec.
Searching here on this site for "mp3 convert" comes up with many usefull suggestions , because many of the programs can easily take a PCM uncompressed wave input, all they need to do is the MP3 out.
Program to convert flac to mp3?
How to convert .ogg to .mp3?
Convert mp4 to mp3
No. This is an information-theoretic impossibility.
MP3 is what is known as a "lossy" format, meaning that when you encode information into an MP3, some of it is lost (but, usually, not to the point that the listener can detect the loss in quality -- the goal of good lossy encoding is to lose information that the user couldn't detect anyway).
Once that information is "lost" by being encoded from a lossless source format into MP3, that information is permanently gone, unless you still have a copy of the source data. If all you have is the encoded MP3, it is permanently gone. There is absolutely NO way to get it back whatsoever.
You can convert your MP3 to a WAV or FLAC or other lossless format until the cows come home, but the original quality will not come back. If the source data is permanently gone, the only thing you could possibly do is attempt to "remaster" the waveform to increase its quality -- but even then, that is like painting over an existing painting... you are changing the essence of the work. You can't get back to the true, original data. It is gone, unless a copy of the lossless data exists somewhere.
Remastering is a very expensive process that involves a lot of manual labor and very advanced audio processing equipment. Big recording studios sometimes remaster old recordings to make their sound quality as good as (or nearly as good as) the sound quality of songs recorded in the studio today with modern audio equipment.
The basic elements of remastering involve a detailed knowledge of signal processing; as well as the ability to run a number of algorithms that find certain types of anomalies, or synthesize entirely new components to the sound. Remastering works best with music; it does not, typically, work very well with voice or other sound effects. Examples include:
- Expanding the dynamic range of the piece
- Adding new background instruments, or replacing existing ones
- Extrapolating mono (1-channel) audio to stereo (2-channel)
- Removing noise such as clicks, pops, buzzes, etc.
The types of algorithms that can be applied to effectively remaster old analog recordings (from 40+ years ago) do not necessarily apply to restoring very low quality MP3s to a quality similar to the original audio. The reason is that the MP3 codec has somewhat unusual quality reductions that produce irregular and difficult-to-correct patterns in the waveform.
Best Answer
An MP3 stream contains info for generating a set of samples where each sample corresponds to a sample in the original LPCM data (like from the WAV file), but a side-effect of encoding is that there is some extra "junk" added to each end, and a side-effect of decoding is that there is even more junk added to the beginning. The decoder will know how much it adds and will skip those samples, but the encoder-added junk isn't entirely predictable (different encoders add different amounts), so the decoder can't skip those samples unless it is somehow informed of what to skip. Some encoders, like LAME, will add such "gapless playback" or "delay and padding" info (in an encoder-specific format, because there's no standard for it) into the file's VBR Info (VBRI) or Xing header, which is sort-of standard even on CBR files, and which contains other info that is sometimes helpful to the player. This header is actually a frame of silence (usually 1152 samples) with some specially formatted info embedded in between the frame's header and the start of its null audio data. Most decoders/players recognize the frame as special and skip those samples, but some don't, so there's another potential point of failure. so I would look toward making sure you use a compatible encoder/decoder combo to get correct-length, junk-trimmed files as output.
Your second question...did you notice the LAME command-line app's "--decode" option? :) This should solve your problem if you're using LAME as the encoder. The output length should match the input.