>
> Instead, take the 5.1 and _DOWNMIX_ all tracks to a single stereo for
> the phone/tablet by declaring -acodec xxxx -ac 2. No intermediate
> steps should be required. Consider also - Do you need pcm_s32le ?
> pcm_s16le is usual.
I fail to see how that is any different than what I am doing now. I was under
the impression that the flags -acodec and -c:a were the same. Regardless using
-acodec reults in identical clipping and noise generation dts->mp3.
For reference here is the command I used:
ffmpeg -i inter.dts -acodec libmp3lame -ac 2 inter-new.mp3
The resulting mp3 file still has horrendous crackling and noise.
What I thought I stated quite clearly in my OP is the following:
5.1DTS->2.0MP3 results in horrible noise and clipping in the resulting mp3 file
5.1DTS>2.0PCM->2.0MP3 does not generate the same atrocious noise. please
reference the files I've included in the dropbox link in my OP.
inter.dts was the ripped 5.1 audio
inter-test.mp3 was encoded to 2.0mp3 format directly from 5.1dts and when
played back on my laptop(s) (Windows, Linux, Mac; in Windows MP, ffplay,
mplayer, vlc, xine, and more), Phones (s5,s3,iphone,htc one), tablet, ipod and
my Sansa MP3 player all has horrific noise.
inter.mp3 was generated by converting the 5.1DTS to 2.0PCM and then the 2.0PCM
to 2.0MP3, sounds just fine when played back on all of my devices.
I am fully aware that there should be NO NEED to use an intermediary wave
format to downsample to stereo audio from 5.1 for a conversion to mp3. But
that's exactly why I'm writing this problem into the group because it is NOT
working as expected.
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