John L wrote:
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.
IIRC this has come up before. The issue seems to be that sometimes -ac 2
normalises and sometimes it doesn't (depending on what codec is used).
You can see whether or not the matrix is normalised with -loglevel debug.
FWIW dts may contain meta data specifying a matrix typically (and by
default if there is none) this will be partially normalised.
To use this you put -request_channels 2 before -i infile.dts.
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