I have a couple of previously recorded programs where the volume is just too low. They are the first two I recorded with Myth, before I got all the various places where the volume is set figured out.

When I play them back I have to turn the sound way up, but of course should I change back to regular TV, or another Myth recording, I'm blasted into the back wall.

They both have a high educational value for some volunteer work I do, and I'd like to have them properly burnt to DVDs. I tried hauling them over to my MicroFlacid box, demuxing, boosting the volume, then remuxing them before putting them back on the Myth box. They play ok on that box, but don't survive the transition to DVD after I've played with them. It doesn't surprise me, because there's a file format conversion in this process, and the resulting .m2a file quite a bit smaller than the original. Not to mention the seven or so steps required to perform this magic. I've kept the original files - pre-futzing - just in case it all went to hell.

I'm sure there must be some simpler way to do this - right on my Myth box, that will maintain the file integrity and audio sink of the original.

Any suggestions?

I have a suggestion rather than a solution: Take a look at the 'transcode' command. You may be able to do a volume boost on an in-place transcodek maybe with the gain parameter ('-s').

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Thanks Human. I'll look into that.

I found that I can do it with ffmpeg, with a couple of little snags. I've been meaning to to do some more testing, and maybe try my hand at a script for a more general increase or decrease volume, producing samples from which the user can choose how much of an increase or decrease to apply.

The documentation for ffmpeg says the command should be:

Current versions of ffmpeg seem to have a bug though.

I can easily make a small video sample with the -acodec copy (audio codec) and -vcodec copy (video codec), adding the -ss timecode (start time) and -t timecode (length of clip) parameters.

The -vol parameter's XXX is 256 for the same volume, larger numbers to increase, and smaller numbers to decrease the level. The combination of "-vol XXX -acodec copy" seems to produce a segfault in both Linux and Windows at the moment.

If I specify "-acodec mp2" - ie use the same codec as my Hauppauge _50 card is recording, the output bitrate defaults to 64 kbs. So I have to specify the -ab (audio bitrate) parameter as well - keeping it the same as the original. My gut says that's triggering a re-encoding that may cause some loss in quality, but it may have to do that just to increase the volume anyway. But hey, "some loss of quality" is better than "volume so low I have to crank the TV up to Max just to hear," that is my current situation with those particular recordings.

I assume I could make a script to glean the audio codec and audio bitrate from the original file, and then issue the ffmpeg command with those parameters correctly and automatically specified. I'm a beginner at linux scripting, so I haven't figured out the How, on that part yet.

For those of you following along at home, with glazed over eyes, and a headache from the confusing parameters, the current command that works is:

To make a sample clip of say the first one minute of the recording with and increased or decreased volume, the command would be: