Re: [FFmpeg-user] Converting Audible AAX

[Robert Chalmers]

They can actually be unencoded if you have the activation key.

ffmpeg -activation_bytes 62689101 -i Volume1.aax -vn -c:a copy -v debug 
Voume1.mp4

This is now my error message.

[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] [aax] file checksum == 
d72f1f04e3c73d0bc68e742db1bc69b58dc3a500
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] [aax] mismatch in checksums!
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] error reading header
[AVIOContext @ 0x7f9683700420] Statistics: 32768 bytes read, 0 seeks
Volume1.aax: Invalid data found when processing input



However, this one works ….. Taken from the github site….

ffmpeg -activation_bytes 62689101 -i Audible_AAX_sample_62689101.aax -vn -c:a 
copy -v debug output_sample.mp4


So either my activation_bytes are wrong …. or something else is amiss.




> On 14 Jan 2016, at 03:58, G A  wrote:
> 
> they are encrypted files, ffmpeg is not going to crack it
> 
> Audible AAX files are encrypted M4B files, and they can be decrypted by
> specifying a 4 byte activation secret.
> 
> 
> 
> --
> View this message in context: 
> http://ffmpeg-users.933282.n4.nabble.com/Converting-Audible-AAX-tp4673787p4674063.html
> Sent from the FFmpeg-users mailing list archive at Nabble.com.
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Robert Chalmers
rob...@chalmers.com.au  Quantum Radio: http://tinyurl.com/lwwddov
Mac mini 6.2 - 2012, Intel Core i7,2.3 GHz, Memory:16 GB. El-Capitan 10.11. 2TB 
Storage made up of - 
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Bay



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[FFmpeg-user] Demux stream with ffmpeg - but decode with HW instead

[Ran Shalit]

Hello,

I would please like to ask if it is possible to demux a stream to raw
h.264 , while do the h.264 decoding in HW instead of ffmpeg.

I think maybe the following can achieve this:

ffmpeg -f mpegts udp://192.168.1.10:48550  -vcodec copy -an -f h264
temp_file.h264 &
temp_file > hw_decoder (hw_decoder is executable which can take
raw h.264 and convert it to yuv)

The only thing I will need to take care of is that this file temp_file
will get larger and larger. Is there a way to deal with it ?

Best Regards,
Ran
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Re: [FFmpeg-user] Demux stream with ffmpeg - but decode with HW instead

[Moritz Barsnick]

On Thu, Jan 14, 2016 at 12:04:34 +0200, Ran Shalit wrote:
> ffmpeg -f mpegts udp://192.168.1.10:48550  -vcodec copy -an -f h264 
> temp_file.h264 &
> temp_file > hw_decoder (hw_decoder is executable which can take
> raw h.264 and convert it to yuv)

$ temp_file(.h264) > hw_decoder
is not valid shell syntax.

$ < temp_file.h264 hw_decoder
or
$ hw_decoder < temp_file.h264
should work to provide the file temp_file to hw_decoder's standard
input.

> The only thing I will need to take care of is that this file temp_file
> will get larger and larger. Is there a way to deal with it ?

Do I understand correctly that you are worried about the intermediate
file, and not so much about having to use "hw_decoder"?

Unix provides you with pipes or "fifos" for this:
$ mkfifo tmp_pipe
$ ffmpeg -f mpegts udp://192.168.1.10:48550 -vcodec copy -an -f h264 tmp_pipe &
$ hw_decoder < tmp_pipe
$ rm tmp_pipe

Moritz
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[FFmpeg-user] Let ffmpeg pick correct fps from stdin

[Jean-Marie Baran]

Hello,

I have ffmpeg reading a stream which is sent to its stdin. ffmpeg then 
record the stream in an mp4 (h.264). The problem is that I don't know 
the fps in advance so I would like that ffmpeg guess it.


So far all my attempts failed with ffmpeg recording systematically at 25 
fps, and the recorded video is then played too fast by players (ffplay, 
VLC, ...). When recording, I see:


fmpeg -y -thread_queue_size 1024 -f s16le -ar 44100 -i 
tcp://localhost:5624?listen=1 -re -f mjpeg -thread_queue_size 1024 -i - 
-c:v libx264 -preset fast -crf 28 -vsync 0 -strict -2 -acodec aac 
-threads 0 -ar 44100 file.mp4

...
frame=8 fps=0.0 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   14 fps= 13 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   21 fps= 13 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   26 fps= 12 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   31 fps= 12 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   37 fps= 11 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   46 fps= 12 q=33.0 size=  33kB time=00:00:00.16 
bitrate=1689.9kbits/s
frame=   52 fps= 12 q=33.0 size=  69kB time=00:00:00.40 
bitrate=1408.2kbits/s
frame=   57 fps= 12 q=33.0 size=  96kB time=00:00:00.60 
bitrate=1299.8kbits/s
frame=   64 fps= 12 q=33.0 size= 136kB time=00:00:00.88 
bitrate=1266.3kbits/s
frame=   71 fps= 12 q=33.0 size= 174kB time=00:00:01.16 
bitrate=1227.3kbits/s

...

So ffmpeg seems to be able to retrieve the fps (13, then 12, then 11, 
...), but it doesn't use it for the recorded video. My best bet was to 
use "-re" but it did not change anything.


Any help would be appreciated,
Thanks

--
*Jean-Marie Baran*

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Re: [FFmpeg-user] Converting Audible AAX

[Moritz Barsnick]

On Thu, Jan 14, 2016 at 08:33:41 +, Robert Chalmers wrote:
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] [aax] file checksum == 
> d72f1f04e3c73d0bc68e742db1bc69b58dc3a500
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] [aax] mismatch in checksums!
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] error reading header
> [AVIOContext @ 0x7f9683700420] Statistics: 32768 bytes read, 0 seeks
> Volume1.aax: Invalid data found when processing input

That's certainly the message when the activation key is incorrect. You
can provoke it on the sample by varying the initially correct key.

> However, this one works ….. Taken from the github site….
> ffmpeg -activation_bytes 62689101 -i Audible_AAX_sample_62689101.aax -vn -c:a 
> copy -v debug output_sample.mp4

Now, is this a copy/paste error, or does your activation key really
happen to be the same as for the github sample? Note your command line:

> ffmpeg -activation_bytes 62689101 -i Volume1.aax -vn -c:a copy -v debug 
> Voume1.mp4

Just wondering,
Moritz
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Re: [FFmpeg-user] Let ffmpeg pick correct fps from stdin

[Moritz Barsnick]

On Thu, Jan 14, 2016 at 11:57:42 +0100, Jean-Marie Baran wrote:
> fmpeg -y -thread_queue_size 1024 -f s16le -ar 44100 -i 
> tcp://localhost:5624?listen=1 -re -f mjpeg -thread_queue_size 1024 -i - 
> -c:v libx264 -preset fast -crf 28 -vsync 0 -strict -2 -acodec aac 
> -threads 0 -ar 44100 file.mp4
> ...

Please show us the full, uncut console output. It contains valuable
information (especially about the detection of the input and the
configuration of the output formats/codecs).

Furthermore, you are using two inputs (tcp://localhost:5624?listen=1
and stdin). Can you reproduce this problem with just your stdin? If so,
please show us that command line and the complete console output.

> So ffmpeg seems to be able to retrieve the fps (13, then 12, then 11, 
> ...), but it doesn't use it for the recorded video. My best bet was to 
> use "-re" but it did not change anything.

That "fps=" is the number of frames per seconds ffmpeg is sending to
its output. In certain situations, and with "-re", that may very well
correspond to the input frame rate.

I think the first task would be to determine whether your stdin
actually can deliver your desired 25 fps, but show us your console
outputs first please.

Moritz
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Re: [FFmpeg-user] Let ffmpeg pick correct fps from stdin

[Jean-Marie Baran]

On 14/01/2016 12:54, Moritz Barsnick wrote:

Please show us the full, uncut console output. It contains valuable
information (especially about the detection of the input and the
configuration of the output formats/codecs).


Here is the full output:

fmpeg -y -thread_queue_size 1024 -f s16le -ar 44100 -i 
tcp://localhost:5624?listen=1 -re -f mjpeg -thread_queue_size 1024 -i - 
-c:v libx264 -preset fast -crf 28 -vsync 0 -strict -2 -acodec aac 
-threads 0 -ar 44100 file.mp4


WARNING: linker: libx264.so has text relocations. This is wasting memory 
and prevents security hardening. Please fix.

ffmpeg version 2.8.3 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.8 (GCC)
  configuration: --arch=arm --target-os=android --enable-cross-compile
  libavutil  54. 31.100 / 54. 31.100
  libavcodec 56. 60.100 / 56. 60.100
  libavformat56. 40.101 / 56. 40.101
  libavfilter 5. 40.101 /  5. 40.101
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.0 : mono
Input #0, s16le, from 'tcp://localhost:39325?listen=1':
  Duration: N/A, bitrate: 705 kb/s
Stream #0:0: Audio: pcm_s16le, 44100 Hz, 1 channels, s16, 705 kb/s
[mjpeg @ 0xb6427c00] Changeing bps to 8
Input #1, mjpeg, from 'pipe:':
  Duration: N/A, bitrate: N/A
Stream #1:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 
720x480 [SAR 1:1 DAR 3:2], 25 tbr, 1200k tbn, 25 tbc

No pixel format specified, yuvj420p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0xb6428800] using SAR=1/1
[libx264 @ 0xb6428800] using cpu capabilities: ARMv6 NEON
[libx264 @ 0xb6428800] profile High, level 3.0
[libx264 @ 0xb6428800] 264 - core 148 - H.264/MPEG-4 AVC codec - 
Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: 
cabac=1 ref=2 deblock=1:-1:-1 analyse=0x3:0x113 me=hex subme=6 psy=1 
psy_rd=1.00:0.15 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 
cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-3 threads=6 
lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 
bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 
b_bias=0 direct=1 weightb=1 open_gop=0 weightp=1 keyint=250 
keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=crf 
mbtree=1 crf=28.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 
aq=1:1.00

Output #0, mp4, to '/storage/emulated/0/DCIM/AMA/NP/record//test.mp4':
  Metadata:
encoder : Lavf56.40.101
Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), 
yuvj420p(pc), 720x480 [SAR 1:1 DAR 3:2], q=-1--1, 25 fps, 12800 tbn, 25 tbc

Metadata:
  encoder : Lavc56.60.100 libx264
Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, mono, 
fltp, 128 kb/s

Metadata:
  encoder : Lavc56.60.100 aac
Stream mapping:
  Stream #1:0 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Stream #0:0 -> #0:1 (pcm_s16le (native) -> aac (native))
frame=8 fps=0.0 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   14 fps= 13 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   21 fps= 13 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   26 fps= 12 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   31 fps= 12 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   37 fps= 11 q=0.0 size=   0kB time=00:00:00.00 bitrate=N/A
frame=   46 fps= 12 q=33.0 size=  33kB time=00:00:00.16 
bitrate=1689.9kbits/s
frame=   52 fps= 12 q=33.0 size=  69kB time=00:00:00.40 
bitrate=1408.2kbits/s
frame=   57 fps= 12 q=33.0 size=  96kB time=00:00:00.60 
bitrate=1299.8kbits/s
frame=   64 fps= 12 q=33.0 size= 136kB time=00:00:00.88 
bitrate=1266.3kbits/s
frame=   71 fps= 12 q=33.0 size= 174kB time=00:00:01.16 
bitrate=1227.3kbits/s

...


Furthermore, you are using two inputs (tcp://localhost:5624?listen=1
and stdin). Can you reproduce this problem with just your stdin? If so,
please show us that command line and the complete console output.
The TCP server is the audio stream. I will try to reproduce it without 
the audio, see what happens.



I think the first task would be to determine whether your stdin
actually can deliver your desired 25 fps, but show us your console
outputs first please.

Just to precise: 25 fps is not the required fps, it is the one put in 
the recorded video file by ffmpeg, which is not correct.


Thank you.

--
*Jean-Marie Baran*
AMA developer

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Re: [FFmpeg-user] Converting Audible AAX

[Robert Chalmers]

Thanks. I solved the problem. There is a script for grabbing the activation 
code from the registered audible account. That is, you have to have an Audible 
login account, and books in your library.

The original has a small error, so I forked it from Github, and applied the 
patch. Well, the patch is already there, but not synched…. and I can see no way 
of contacting the author?
So I forked my own copy.

https://github.com/ShanghaiTimes/audible-activator

This is not my code - and it’s simple Python code, and easy to run from the 
command line. 

and then1.  
ffmpeg -activation_bytes  -i Volume1.aax -vn -c:a copy -v debug 
Voume1.mp4 
will work.

> On 14 Jan 2016, at 09:22, Moritz Barsnick  wrote:
> 
> On Thu, Jan 14, 2016 at 08:33:41 +, Robert Chalmers wrote:
>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] [aax] file checksum == 
>> d72f1f04e3c73d0bc68e742db1bc69b58dc3a500
>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] [aax] mismatch in checksums!
>> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f968400] error reading header
>> [AVIOContext @ 0x7f9683700420] Statistics: 32768 bytes read, 0 seeks
>> Volume1.aax: Invalid data found when processing input
> 
> That's certainly the message when the activation key is incorrect. You
> can provoke it on the sample by varying the initially correct key.
> 
>> However, this one works ….. Taken from the github site….
>> ffmpeg -activation_bytes 62689101 -i Audible_AAX_sample_62689101.aax -vn 
>> -c:a copy -v debug output_sample.mp4
> 
> Now, is this a copy/paste error, or does your activation key really
> happen to be the same as for the github sample? Note your command line:
> 
>> ffmpeg -activation_bytes 62689101 -i Volume1.aax -vn -c:a copy -v debug 
>> Voume1.mp4
> 
> Just wondering,
> Moritz
> ___
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Robert Chalmers
rob...@chalmers.com .au  Quantum Radio: 
http://tinyurl.com/lwwddov
Mac mini 6.2 - 2012, Intel Core i7,2.3 GHz, Memory:16 GB. El-Capitan 10.11. 2TB 
Storage made up of - 
Drive 0:HGST HTS721010A9E630. Upper bay. Drive 1:ST1000LM024 HN-M101MBB. Lower 
Bay



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Re: [FFmpeg-user] Compiling ffmpeg with libx264

[Rayden]

Hi
I tried contacting the user community in the mail address provided. Didn't
receive any response, that's why I came here. Thought there would be people
who would have done this and willing to help. My apologies if this is the
wrong place to put this.



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Sent from the FFmpeg-users mailing list archive at Nabble.com.
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[FFmpeg-user] Compiling ffmpeg with libx264

[Krishnan Unni]

Hi

I am trying to compile ffmpeg with libx264 in windows 10 x64. Based on the 
guideline in ffmpeg wiki I am able to compile ffmpeg alone successfully. But 
when I try to compile x264 lib based on the guide provided in ffmpeg wiki it's 
failing saying "could not identify os" and give two links to update the 
configure.* (2 files). Once I updated the file from the provided link and try 
to compile again MSYS is throwing error "could not identify OS" . The same MSYS 
compiled the ffmpeg fine.   please help me to compile libx264 . I know how to 
enable it for ffmpeg once its cookies using the enable-libx264 attribute.
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[FFmpeg-user] help: resampling audio settings

[G A]

hello,

i’m resampling some audio with soxr with flac.  do these seem reasonable?
thanks

-codec:a flac -prediction_order_method 4 -lpc_passes 12 -exact_rice_parameters 
1 -lpc_type 3 -ch_mode indep -sample_fmt s16 -ar 44100 -b:a 253588 -filter:a 
"volume=.7dB" -filter:a aresample=resampler=soxr=cheby=1:precision=28, 
dynaudnorm=f=24=g=64=p=.96=m=20=r=1=n=1=c=1=b=0 -filter:a shibata -ac 2



Sent from somewhere in … The Twilight Zone

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Re: [FFmpeg-user] Compiling ffmpeg with libx264

[Lou]

On Thu, 14 Jan 2016 07:14:26 -0500
Krishnan Unni  wrote:

> I am able to compile ffmpeg alone successfully. But when I try to
> compile x264 lib based on the guide provided in ffmpeg wiki it's
> failing saying "could not identify os"

If you're having problems compiling x264 you should get help from x264.
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Re: [FFmpeg-user] help: resampling audio settings

[Moritz Barsnick]

On Wed, Jan 13, 2016 at 20:14:29 -0800, G A wrote:
> i’m resampling some audio with soxr with flac.  do these seem reasonable?

I know nothing about the optimal resampling and FLAC encoding options.
You don't say what you are trying to achieve.

That said, this I can comment:

> -codec:a flac -prediction_order_method 4 -lpc_passes 12 
> -exact_rice_parameters 1 -lpc_type 3 -ch_mode indep -sample_fmt s16 -ar 44100 
> -b:a 253588 -filter:a "volume=.7dB" -filter:a 
> aresample=resampler=soxr=cheby=1:precision=28, 
> dynaudnorm=f=24=g=64=p=.96=m=20=r=1=n=1=c=1=b=0 -filter:a shibata -ac 2

- You cannot specify multiple filters with "-vf" or "-filter:v". You
  need to read about how to chain filters.

- The syntax for handing options to your filters is incorrect. It is
  not
"filter=opt1=val1=opt1=val2=..."
  but rather
"filter=opt1=val1:opt2=val2:opt3=val2:..."

- The whitespace in your (first) filter chain (", dynaudionorm") needs
  to be protected from the shell.

- "shibata" is not a filter. It is an argument to the resampler's
  "-dither_method" option.

Moritz
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Re: [FFmpeg-user] Let ffmpeg pick correct fps from stdin

[Carl Eugen Hoyos]

Jean-Marie Baran  ama.bzh> writes:

> Input #1, mjpeg, from 'pipe:':
>Duration: N/A, bitrate: N/A
>  Stream #1:0: Video: mjpeg, yuvj420p(pc, bt470bg/unknown/unknown), 
> 720x480 [SAR 1:1 DAR 3:2], 25 tbr, 1200k tbn, 25 tbc

mjpeg has no timestamps and therefore no framerate, 
25fps is the default.
The mjpeg demuxer understands the "framerate" option, 
you can use it to specify a framerate.

Carl Eugen

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Re: [FFmpeg-user] OSX:deprecated

[Moritz Barsnick]

On Thu, Jan 14, 2016 at 00:39:34 -0800, G A wrote:
> i get this message from ffplay.  is there a fix for this?

Please (always) show us the full command line and the complete, uncut
console output. All of its information is valuable.

Moritz
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Re: [FFmpeg-user] help: resampling audio settings

[Moritz Barsnick]

On Thu, Jan 14, 2016 at 21:18:20 +0100, Moritz Barsnick wrote:
> - You cannot specify multiple filters with "-vf" or "-filter:v". You
>   need to read about how to chain filters.

I meant "-af" or "-filter:a" of course. Sorry.
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[FFmpeg-user] Arvato proxy file

[Bouke (VideoToolShed)]

Guys,
Have a testfile that won't play nice.
This file comes from an Arvato asset management system.
AFAIK, it's a 'growing' file, that 'should' be playable while it's still 
being created.


FFmpeg (Latest Zerano build, ffmpeg-20160114-git-62dfe1d-win64-static) does 
not like it, and it tells me to upload a sample.
That i've done (Called Arvato_Proxy.zip), if there is no solution here, can 
someone please post this on the Develop list?


Thx,
Bouke

-i Q:\Philip\VRT\test.mxf -codec copy -bsf:a aac_adtstoasc 
Q:\Philip\VRT\test.mov

(Just trying to rewrap to QT)

ffmpeg version N-77836-g62dfe1d Copyright (c) 2000-2016 the FFmpeg 
developers

 built with gcc 5.2.0 (GCC)
 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth 
--enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv 
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec 
--enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug 
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr 
--enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab 
--enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib

 libavutil  55. 13.100 / 55. 13.100
 libavcodec 57. 22.100 / 57. 22.100
 libavformat57. 21.101 / 57. 21.101
 libavdevice57.  0.100 / 57.  0.100
 libavfilter 6. 23.100 /  6. 23.100
 libswscale  4.  0.100 /  4.  0.100
 libswresample   2.  0.101 /  2.  0.101
 libpostproc54.  0.100 / 54.  0.100
Input #0, mxf, from 'Q:\Philip\VRT\39014.mxf':
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   modification_date: 2016-01-12 22:08:24
   uid : 6267ef5a-7b76-733d-2767-e74b011fd76c
   generation_uid  : b30ba62e-db12-3c15-877e-0c393e0f9900
   company_name: arvato Systems S4M GmbH
   product_name: LoRes-Transcoder
   product_version : 1.0.0.0.0
   product_uid : d0675d57-0f31-2003-e726-5007c5758762
   application_platform: S4M MediaFramework
   material_package_umid: 
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   timecode: 00:00:00:00
 Duration: N/A, start: 0.00, bitrate: N/A
   Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 640x360 [SAR 
1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc

   Metadata:
 file_package_umid: 
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   Stream #0:1: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
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   Stream #0:2: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
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   Stream #0:3: Audio: aac (LC), 48000 Hz, stereo, fltp
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   Stream #0:4: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
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   Stream #0:6: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
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   Stream #0:7: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
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   Stream #0:8: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
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[mov @ 032dfd00] Codec for stream 0 does not use global headers but 
container format requires global headers
[mov @ 032dfd00] Codec for stream 1 does not use global headers but 
container format requires global headers

Output #0, mov, to 'Q:\Philip\VRT\test.mov':
 Metadata:
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   uid : 6267ef5a-7b76-733d-2767-e74b011fd76c
   generation_uid  : b30ba62e-db12-3c15-877e-0c393e0f9900
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   product_name: LoRes-Transcoder
   product_version : 1.0.0.0.0
   product_uid : d0675d57-0f31-2003-e726-5007c5758762
   application_platform: S4M MediaFramework
   material_package_umid: 
0x060A2B340101010501010D11130098863E05997305A5F2B500110A680FC0

   timecode: 00:00:00:00
   encoder : Lavf57.21.101
   Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 
DAR 16:9], q=2-31, 25 fps, 25 tbr, 12800 tbn, 25 tbc

   Metadata:
 file_package_umid