[MEncoder-users] mencoder, ac3 audio and mts conversion to avi

[Alex Morris]

On Fri, May 23, 2008 at 7:10 PM, Alex Samad <alex at samad.com.au> wrote:
>
> On Fri, May 23, 2008 at 06:40:47PM -0700, Alex Morris wrote:
> > Hello,
> >
>
> [snip]
>
> I am newbie at this as well, but I might be able to shed some light
>
>
> >
> > This creates the avi with no sound.  Video is nice, very smooth and high
> > res. No audio.  I find this in the program output;
> >
> > LAVF_header: av_find_stream_info() failed
> > RAWDV file format detected.
>
> ^^^ this is saying it found rawdv for video input
>
> > VIDEO:  [DVSD]  720x576  24bpp  25.000 fps    0.0 kbps ( 0.0 kbyte/s)
> > [V] filefmt:65536  fourcc:0x30323449  size:1920x1080  fps:25.00
> > ftime:=0.0400
> > ==========================================================================
> > Opening audio decoder: [libdv] Raw DV Audio Decoder
>
> ^^ here is say it found rawdv audio
>
> These don't seem to corrospond to your earlier assumption of x264.
>
> But I think it is better you have the raw DV stream (I believe
> uncompressed images)
>
> > Unknown/missing audio format -> no sound
> > ADecoder init failed :(
> > Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders
> > Cannot find codec 'dvaudio' in libavcodec...
> > ADecoder init failed :(
> > ADecoder init failed :(
> > Cannot find codec for audio format 0x56444152.
> > Read DOCS/HTML/en/codecs.html!
> >
> > I don't understand what's happening.
>
> mencoder can't find a audio codec to match the audio codec presented in
> the original file, thus it can't re encoded it.
>
>
> >
> > Q: If mplayer identifies the audio sample as ac3, should I be using ac3
> > codec in the transcoding process?
> > Q2: When I run mencoder -oac help, it lists lavc as present.  Why does my
> > use of mencoder apparently ignore the lavcopts I chose?
> i beleive you have set the outfile parameters
>
> > Q3: What is mencoder's use of lavc telling me when ffmepg cannot find
> > dvaudio in libavcodec?
>
> you have set outfile codec not infile
>
> >
> > I think this method should work, because I can play the audio-only portion
> > of the sample with mplayer, like this;
> >
> > mplayer -rawaudio format=0x2000:channels=6:bitrate=38400 -demuxer rawaudio
> > 00000.mts.ac3
> >
> > Playing 00000.mts.ac3.
> > rawaudio file format detected.
> > ==========================================================================
> > Opening audio decoder: [liba52] AC3 decoding with liba52
> > Using SSE optimized IMDCT transform
> > Using MMX optimized resampler
> > AUDIO: 48000 Hz, 2 ch, s16le, 384.0 kbit/25.00% (ratio: 48000->192000)
> > Selected audio codec: [a52] afm: liba52 (AC3-liba52)
> > ==========================================================================
> > AO: [oss] 48000Hz 2ch s16le (2 bytes per sample)
> > Video: no video
> > Starting playback...
> > A: 243.8 (04:03.7) of 24.5 (24.4)  0.9%
> >
> > Exiting... (End of file)
>
> This is the way I have extract sound from file, dumping straight to pcm
> and then reworking that back into the final .avi file.
>
> remember avi files are contianers that hold different media streams
> (video, audio) and these streams can be encoded with different codec's
>
>

[snip]

Thanks for the help.  I understand a little better now.

I finally got ffmepg to create an avi "container", here is command that worked;

ffmpeg -r 25 -s 1920x1080 -vcodec copy -i bits0001.yuv -i bits0001.ac3
-acodec ac3 -ab 384k -b 384000 -f avi 00004.avi

The avi file is valid, mplayer can open it and starts to play, but
video freezes after few seconds, and then warns that my system is too
slow.

Time to upgrade to more CPU speed, 64-bit OS, and lots of disk.

Thank you again.