Greetings fellow board members. OK, I have been converting an .AVI to DVD format for use in a stand-alone player. The audio is stereo MP3 to begin with, I change it to WAV and then to AC3, which then the audio level is too low when played in the stand-alone DVD player. I have searched the forums for related threads. My question is slightly different from the information I could find.
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Stereo VBR MP3 - Apx. 128 kbps ----> WAV PCM - 16 bit ----> AC3 - 224 kbps
Everything begins and stays at 48000Hz.
I change it from MP3>WAV PCM>AC3.
Step-1 -- I use AviDemux to save the VBR MP3 audio from the AVI as WAV.
Step-2 -- I use Batch ffmpegGUI .NET to convert the WAV to AC3.
The overall audio volume level is just right on the original MP3 and the subsequent WAV, while the overall volume is too low when I take the final step and convert the WAV to AC3.
My questions.
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1. At which format should I increase overall volume?
2. Which tools should I use to increase overall volume?
3. I only want the overall volume to be higher, without changing the dynamic difference in the lows and highs. That alone probably changes things, but I don't know how else to word it. I just want it to be as close to the original as possible, with increased overall volume only. :/
The question I am wondering most about is at what format to increase the sound.
My second question is about frame rate. I use CCE. If I inverse telecine a 29.970 video while encoding with CCE, and then re-add the pull down flags after to make the now 23.976 video playback at 29.97 for DVD compatibility, will I need to do anything to ensure the audio stays in sync? Or is nothing changed? I know the volume needs to be changed to match the FPS when converting PAL>NTSC and vice versa, but I don't think that is needed in this situation.. Also, does converting VBR MP3 to WAV eliminate the audio/video sync issues as the audio would now be CBR(WAV)?
Thanks for reading. I have tried to make this post as clear as possible, so I hope I succeeded in doing so.![]()
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I use VirtualDub Mod for those type of audio conversions. I adjust the volume there, if needed, then save out as a WAV with full processing and encode that to AC3 with ffmpeggui. But you could also use the freeware audio editor Audacity to adjust the WAV audio before encoding to AC3. You could also adjust the dynamic range or add filtering or some other operations there. ffmpeggui may also be able to adjust the volume levels, though I don't know the method.
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Wouldn't that increase the volume of the MP3 before it is saved as WAV? Does the format matter for quality purposes when increasing the volume? Could increasing the volume while the audio is MP3 lower the quality as opposed to doing it in WAV, Or does it not matter what format it is when just increasing the overall volume?
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Adjustment of the volume wouldn't happen till the audio is processed to WAV. I rarely have to increase the volume, so I don't have that much knowledge about where the 'best' place is to increase the volume. I would prefer to use a audio editor for that type of adjustment as you have more control there, and you should be able to monitor it a little better. You could also do some light filtering, if needed, to improve the noise level.
Usually MP3 has sustained quite a bit of damage to the quality in it's initial conversion and compression and about all you can do is try to clean it up.
If this was all analog audio, I wouldn't amplify till the last conversion to keep distortion down. But with digital audio, I don't know. The volume increase may be nothing more than changing a flag setting and not really do anything to the the actual audio file. -
I see.
But as for the frame rate change... Should the audio and video stay in sync when the video is pulled down 23.976>29.97 and vice versa? I have read some guides where it makes mention that the audio can be optionally lengthened/shortened to match the new FPS of the video when converting to or from NTSC/PAL.
I don't understand. Can anyone explain this to me? :\
On a side note, is FFMPEG capable of adjusting audio volume? -
I noticed VirtualdubMOD has an audio filter named "gain."
Does this use a different method compared to using the volume option as you described?
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