Actually, I'm using VLC to play back what it just created! I just tried QT, and it behaved 'the same'; that is, no audio, but if I move the position slider around, I hear bursts of off-key audio for a split second.
So -
I can use VLC to play the original movie
I can use VLC to transcode to wmv no problem (result plays in VLC and Windows Media Player)
I can use mediaCoder to transcode to h.264/mp4a and mp4v/mp4a no problem (result plays in VLC and QT)
I can use toast to transcode to h.264/mp4a no problem (result plays in VLC and QT)
When using VLC to transcode to mp4v/mp4a, choosing different samplerate values has the results described.
Is there something I can do in VLC with the original media to change ONLY it's sample rate, to make it more compatible, THEN try the transcode? Or perhaps convert the audio from its native format to something more compatible? As mentioned above,
When I play the original video in VLC, it has the following audio characteristics ('codec info'):
codec: twos; ... channels: 1; ... sample rate: 16000; bits per sample: 16; bitrate: 256 kb/s
Thanks!
Edit/Update ...
This REALLY does not make sense to me ... please tell me what is going on ... ) !
above, I' was initially using the following command line:
Code: Select all
"C:\Program Files\VideoLAN\VLC\vlc" P1000098.MOV --extraintf=logger --verbose=3 :sout=#transcode{vcodec=mp4v,vb=1024,scale=0.25,acodec=mp4a,ab=128,channels=2}:duplicate{dst=std{access=file,mux=mp4,dst="p1000098-MP4-test.mp4"}}
then, at your suggestion, I added various 'samplerate' values and got weird (but slightly better) results. Finally, for no particular reason, I completely removed any 'qualifiers' from the audio spec - just left acodec=mp4a. That is, I removed the ab=128, channels=2, samplerate= ... . The darned thing transcoded (and played in VLC) just fine !!!
Looking at the resulting output file, in VLC, 'codec info' shows:
Codec: mp4a; ... Channels:2; sample rate 32000; bits per sample 16; Bitrate 256kb/s, AND, 'AAC extension: SBR'
So right off the bat, I can get a good transcode as long as I don't try to specify the bitrate, samplerate, etc.
So I tried using those settings in the audio section, to see if it would accept them. I tried
Code: Select all
acodec=mp4a,ab=256,channels=2,samplerate=32000
But I got the error shown way up above -
Code: Select all
[00000477] avcodec encoder warning: libfaac doesn't support this output format!
(libfaac@0181CA40)
[00000477] avcodec encoder error: cannot open encoder
[00000403] stream_out_transcode stream out error: cannot find audio encoder (mod
ule:any fourcc:mp4a)
You'd think, if the OUTPUT produced with no qualifiers had a certain audio characteristic, then explicitly setting those characteristics would be essentially the same thing ... but of course it is not!
Does this make sense to anyone? Thanks!