LSmash weird Audio [JFYI]

DUFF THREAD, WILL BE DELETED, Please Ignore. There is something real weird with new LWLibavAudioSource(ff_options="drc=0,...") stuff on WAV audio, so far as I know, the eg "NICAC3Source(...,DRC=0)" is supposed to remove EXISTING DRC from AC3 files, but if used on WAV, LWLibavAudioSource(ff_options = "drc=0,...") then seems to affect WAV, even when no drc is applied, think should only apply any drc stuff to AC3 [or other DRC applied audio], and not apply to WAV ie linear non-compressed [without drc applied]. MeGUI Produces real nasty audio when "Normalize Peaks to:- 98" used in AAC audio settings with new LWLibavAudioSource(ff_options = "drc=0,...") stuff on linear WAV source file. I'm going back to older LSmash before recent DRC thingy (before ff_options ="drc=0,..."), until I at least figure out what the new plug is doing wrong/different, I got many encodes with nasty audio. I've long converted all source audio to 16 bit stereo WAV using FFMPEG script, with DRC removed, prior to encode script. (to reduce excessive [eg 20->30 min] MEGUI normalize times with eg 24/96 bit, 7.1 audio source, if converted to 16 bit stereo within script) EDIT: Same nasty result whether DRC box ticked or not in MEGUI audio encode AAC settings. EDIT: It is totally possible, [even probable] that I totally misunderstand the drc thingy, but, what I always used to do with good output audio, has now totally horrible result when ff_options="drc=0,..." rather than drc=0 as in earlier version LSMash audio source filter with WAV source.

I cannot reproduce the issue. I used the following script to test: Code:

I don't use MeGUI. I installed but I didn't find NeroAAC for audio encoding (only FFmpeg AAC..). I did one more test: 1. Converted ac3 5.1 to 16-bit wav 2.0 (ffmpeg -i in.ac3 -c:a pcm_s16le -ac 2 out.wav) 2. Used LWLibavAudioSource("out.wav", ff_options="drc_scale=0") with LSMASHSource_x64_1167.0.0.0 (ffmpeg -i script.avs -c:a -pcm_s32le 1.wav) 3. Used LWLibavAudioSource("out.wav") with LSMASHSource_x64_1167.0.0.0 (ffmpeg -i script.avs -c:a -pcm_s32le 2.wav) 4. Used LWLibavAudioSource("out.wav", drc_scale=0) with LSMASHSource_x64_1147.0.0.0 (ffmpeg -i script.avs -c:a -pcm_s32le 3.wav) All three files (1.wav, 2.wav, 3.wav) have same SHA256 hash. Edit: I downloaded . I tested: 1. ffmpeg -i script.avs -f wav -

Sorry StvG, this thread is totally bogus, is self inflicted problem, not problem of LSMash. I made some changes somewhere in big (35KB,700 line) script, still not sure where the problem resides and as I updated on all machines, I dont positively know what I did wrong. I have however managed to kludge the problem until I find it. Anyway, part of fix was to convert all input WAVs to float, I'm using this if anybody finds it of use. ConvStereoAudio.Bat Code:

If somebody else is happen to run into the same condition it maybe helpful to leave the thread up...

OK boss, up it stays.