[FFmpeg-user] User Question: resize and pix_fmt

[Warprays Alceatraz]

Hello :

i have a question about -s and -pix_fmt

the raw video is 8K yuv420 , and i want get 4K 444

so i use this :

-s 3840x2160 -pix_fmt yuv444p

as we know  420 have half resolution ， 8k have 4k UV channel

what ffmpeg will do when resize ?


keep raw 4k UV channel with downscaled Y channel

or

down UV to 2K them save them in upscale 4K ?




I don't know how to code  , so i cant figure this question through source code 
even its the best way

Thanks for help .


by the way , can yuv4mpegpipe use yuv444p mode ? i failed when try


-s 3840x2160 -pix_fmt yuv444p -f yuv4mpegpipe - | xxx xxx xxx xxx xxx xxx


Regards


Sorry I'm not native english useer . sorry  for  grammar error

And this is my first time use mail list , sorry for disturb ， i used post on 
forums .

And is that the correct way to use a mail list ?

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[FFmpeg-user] Streaming a H264 stream from a named pipe - Windows - to FFServer

[Thadeu Antonio Ferreira de Melo]

Hello.

We are using Nvidia Capture to capture and encode a 3D app. Our solution
writes the stream into a named pipe \\.\pipe\stream -
https://en.wikipedia.org/wiki/Named_pipe

This stream is playable with "ffplay \\.pipe\stream".

The problem now is that we have to channel this stream to our local net
work and internet. Low latency is very important.

We are exploring possible solutions like :

ffmpeg -override_ffserver -f \\.\pipe\stream  http://:8090/feed1.ffm

With mpegts and pure UDP it works for localhost and local network. I also
tried to reencode the data, without success.

So, what is the simplest way to send the stream up to our FFSERVER (or
Gstreamer server)?
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[FFmpeg-user] problems decoding opus from RTP stream

[Dave Horton]

I am trying to decode opus audio from an RTP stream and getting some errors.

I’ve tested two different RTP streams (same conversation, one from caller and 
the other from callee).
I am using HEAD, as of a few days ago.

One decodes fine, the other gives these errors:

size=  15kB time=00:00:02.38 bitrate=  51.4kbits/s speed=0.786x
[opus @ 0x7fcc1b00] LBRR frames is not implemented. Update your FFmpeg 
version to the newest one from Git. If the problem still occurs, it means that 
your file has a feature which has not been implemented.
[opus @ 0x7fcc1b00] Error decoding a SILK frame.
[opus @ 0x7fcc1b00] Error decoding an Opus frame.

The SDP I feed in is:

v=0
o=root 1530041045 1530041045 IN IP4 81.201.82.171
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 40130 RTP/AVP 107 101
a=rtpmap:107 opus/48000/2
a=rtpmap:101 telephone-event/8000
a=fmtp:107 useinbandfec=1
a=fmtp:101 0-16
a=rtcp:40131
a=ptime:20
a=maxptime:60
a=sendrecv
a=silenceSupp:off - - - -

The arguments are:

-loglevel debug -y -protocol_whitelist file,crypto,udp,rtp,pipe -re -i pipe:0 
-ar 48000 foo.flac



The output looks like this

ffmpeg version N-87326-g3ffd3b7 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 7.3.0 (clang-703.0.29)
  configuration: --enable-gpl --enable-nonfree --enable-libopus 
--enable-libspeex --enable-libvpx --enable-libmp3lame --disable-optimizations 
--disable-stripping --extra-cflags='-DDEBUG -O0 -g'
  libavutil  55. 75.100 / 55. 75.100
  libavcodec 57.106.101 / 57.106.101
  libavformat57. 82.100 / 57. 82.100
  libavdevice57.  8.101 / 57.  8.101
  libavfilter 6.105.100 /  6.105.100
  libswscale  4.  7.103 /  4.  7.103
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) 
with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with 
argument '1'.
Reading option '-protocol_whitelist' ... matched as AVOption 
'protocol_whitelist' with argument 'file,crypto,udp,rtp,pipe'.
Reading option '-re' ... matched as option 're' (read input at native frame 
rate) with argument '1'.
Reading option '-i' ... matched as input url with argument 'pipe:0'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in 
Hz)) with argument '48000'.
Reading option 'foo.flac' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.

Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url pipe:0.
Applying option re (read input at native frame rate) with argument 1.
Successfully parsed a group of options.
Opening an input file: pipe:0.
[NULL @ 0x7ffbd1810600] Opening 'pipe:0' for reading
TS score: 0 0
[sdp @ 0x7ffbd1810600] Format sdp probed with size=2048 and score=50
[sdp @ 0x7ffbd1810600] audio codec set to: opus
[sdp @ 0x7ffbd1810600] audio samplerate set to: 48000
[sdp @ 0x7ffbd1810600] audio channels set to: 2
[sdp @ 0x7ffbd1810600] audio codec set to: opus
[sdp @ 0x7ffbd1810600] audio samplerate set to: 8000
[sdp @ 0x7ffbd1810600] audio channels set to: 1

udp @ 0x7ffbd1500ea0] end receive buffer size reported is 65536

 end receive buffer size reported is 65536
[sdp @ 0x7ffbd1810600] After avformat_find_stream_info() pos: 311 bytes 
read:311 seeks:0 frames:1
Input #0, sdp, from 'pipe:0':
  Metadata:
title   : session
  Duration: N/A, start: 0.00, bitrate: N/A
Stream #0:0, 1, 1/8000: Audio: opus, 48000 Hz, mono, fltp
Successfully opened the file.

Parsing a group of options: output url foo.flac.
Applying option ar (set audio sampling rate (in Hz)) with argument 48000.
Successfully parsed a group of options.
Opening an output file: foo.flac.

[file @ 0x7ffbd2900660] Setting default whitelist 'file,crypto'

Successfully opened the file.

Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> flac (native))
cur_dts is invalid (this is harmless if it occurs once at the start per stream)

 [SWR @ 0x7ffbd3008600] Using fltp internally between filters

 detected 8 logical cores

[graph_0_in_0_0 @ 0x7ffbd1500120] Setting 'time_base' to value '1/48000'
[graph_0_in_0_0 @ 0x7ffbd1500120] Setting 'sample_rate' to value '48000'

 [graph_0_in_0_0 @ 0x7ffbd1500120] Setting 'sample_fmt' to value 'fltp'
[graph_0_in_0_0 @ 0x7ffbd1500120] Setting 'channel_layout' to value '0x4'

[graph_0_in_0_0 @ 0x7ffbd1500120] tb:1/48000 samplefmt:fltp samplerate:48000 
chlayout:0x4

[format_out_0_0 @ 0x7ffbd2a003e0] Setting 'sample_fmts' to value 's16|s32'
[format_out_0_0 @ 0x7ffbd2a003e0] Setting 'sample_rates' to value '48000'

[format_out_0_0 @ 0x7ffbd2a003e0] auto-inserting filter 'auto_resampler_0' 
between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
[AVFilterGraph @ 

Re: [FFmpeg-user] importing rtp packet stream and decoding

[Dave Horton]

>> Sorry, my bad — I was able to fix that error — which was a simple typo in my 
>> command that led to a blank output url.

>> With that fixed, I get further:

Now I have figured out why the audio sounded like crap.  The problem was that 
ffmpeg was treating the input stream as pcma, when it in fact was pcmu.
The issue is that the SDP indicated that the audio endpoint was able to receive 
_either_ pcmu or pcma — both were in the SDP, as follows:

v=0
o=root 2116403914 2116403914 IN IP4 81.201.82.161
s=session
c=IN IP4 127.0.0.1
t=0 0
m=audio 40558 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp:40559
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

However, the rtp stream contained only PCMU audio.  The payload type of 0 in 
the RTP header should have clued ffmpeg in that this was PCMU, as per the SDP.  
Yet ffmpeg logged out that it was PCMA and transcoded it as such, incorrectly.

This seems like a bug to me, should I recreate on HEAD and open a ticket 
somewhere?

I’m still wondering about to sync my sending side with ffmpeg being ready to 
receive, and also if lengthier decodes are expected when using rtp vs file


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Re: [FFmpeg-user] importing rtp packet stream and decoding

[Dave Horton]

On Sep 22, 2017, at 2:50 PM, Dave Horton  wrote:

>> So I am trying to feed the captured rtp stream into ffmpeg and get a 
>> transcoded output.  Not quite working and I could use some help

Sorry, my bad — I was able to fix that error — which was a simple typo in my 
command that led to a blank output url.

With that fixed, I get further:

Now I am able to generate a wav file, but I still have several issues and 
questions

1.  It sounds like crap — the audio is recognizable but barely.  Something is 
either wrong in the input feed I am sending, or what ffmpeg thinks it is 
receiving.  I did a wireshark trace during the execution to see exactly what 
was going over the wire, and it was the true and correct RTP packets as best I 
can tell (RTP headers, sequence numbers, payload types, etc all look good).

2.  ffmpeg takes quite a while to decode this file….maybe an additional 10 
seconds after the utility has finished sending….how can I signal to ffmpeg that 
the stream is over?

3.  Is there any good way to synchronize ffmpeg with the sending utility?  
Right now, I start ffmpeg first, but I have no way of knowing when it has 
opened the socket on the local port and is ready to receive.  I’m currently 
just waiting 2 seconds after feeding it the SDP before I start sending RTP, but 
is there a better way?

***new logs**

args: -loglevel debug -y -protocol_whitelist file,crypto,udp,rtp,pipe -re -i 
pipe:0 foo.wav

ffmpeg stderr says: ffmpeg version N-85641-gdd49eff-tessus Copyright (c) 
2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg 
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl 
--enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm 
--enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopus --enable-libschroedinger 
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora 
--enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzmq --enable-libzvbi --enable-version3 
--disable-ffplay --disable-indev=qtkit

ffmpeg stderr says:   libavutil  55. 61.100 / 55. 61.100
  libavcodec 57. 93.100 / 57. 93.100
  libavformat57. 72.101 / 57. 72.101
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 86.100 /  6. 86.100
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) 
with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with 
argument '1'.
Reading option '-protocol_whitelist' ... matched as AVOption 
'protocol_whitelist' with argument 'file,crypto,udp,rtp,pipe'.
Reading option '-re' ... matched as option 're' (read input at native frame 
rate) with argument '1'.
Reading option '-i' ... matched as input url with argument 'pipe:0'.
Reading option 'foo.wav' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url pipe:0.
Applying option re (read input at native frame rate) with argument 1.
Successfully parsed a group of options.
Opening an input file: pipe:0.
[sdp @ 0x7ff9f400] Format sdp probed with size=2048 and score=50

ffmpeg stderr says: [sdp @ 0x7ff9f400] audio codec set to: pcm_mulaw
[sdp @ 0x7ff9f400] audio samplerate set to: 8000
[sdp @ 0x7ff9f400] audio channels set to: 1
[sdp @ 0x7ff9f400] audio codec set to: pcm_alaw
[sdp @ 0x7ff9f400] audio samplerate set to: 8000
[sdp @ 0x7ff9f400] audio channels set to: 1
[sdp @ 0x7ff9f400] audio codec set to: pcm_alaw
[sdp @ 0x7ff9f400] audio samplerate set to: 8000
[sdp @ 0x7ff9f400] audio channels set to: 1
[udp @ 0x7ff9f3501400] end receive buffer size reported is 65536
[udp @ 0x7ff9f35017c0] end receive buffer size reported is 65536
[sdp @ 0x7ff9f400] setting jitter buffer size to 500
[sdp @ 0x7ff9f400] Before avformat_find_stream_info() pos: 286 bytes 
read:286 seeks:0 nb_streams:1

[Note: I start sending RTP here]

ffmpeg stderr says: [sdp @ 0x7ff9f400] All info found

ffmpeg stderr says: [sdp @ 0x7ff9f400] After avformat_find_stream_info() 
pos: 286 bytes read:286 seeks:0 frames:1
Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, sdp, from 'pipe:0':
  Metadata:
title   : session
  Duration: N/A, start: 0.00, bitrate: 64 kb/s

ffmpeg stderr says: Stream #0:0, 1, 1/8000: Audio: pcm_alaw, 8000 Hz, mono, 
s16, 64 kb/s
Successfully opened the 

Re: [FFmpeg-user] importing rtp packet stream and decoding

[Dave Horton]

So I am trying to feed the captured rtp stream into ffmpeg and get a transcoded 
output.  Not quite working and I could use some help

Basically I have 
1.  An sdp file that describes the rtp, and indicates which port (on localhost) 
the rtp will be arriving on
2.  A process / utility that reads the rtp from a file and then streams it to 
that port.

I have a node.js application managing all of this — the idea is that it will 
spawn ffmpeg, send the SDP in on its stdin, instruct ffmpeg about the output, 
then spawn the utility to start the rtp stream.

I start up ffmpeg with these arguments:

ffmpeg  -loglevel debug -y  -protocol_whitelist file,crypto,udp,rtp,pipe -i 
pipe:0 map 0:0 foo.wav

I feed it the SDP, and at that point all looks good — from the logging (which I 
will show below) it appears like ffmpeg has decoded the SDP and started 
listening on the rtp port specified in the SDP (127.0.0.1:40558 in this case).

However, once it begins receiving rtp it logs an error:  "Unable to find a 
suitable output format for ‘'

Am I doing something wrong with my command above?  My intent is to instruct it 
to take the first rtp stream and encode it a wav file called ‘foo.wav’.

Here is the complete output (note because I am starting ffmpeg as a child 
process the logging is coming through the parent process which is adding the 
'ffmpeg stderr says..’ etc)

 console log
args: -loglevel debug -y  -protocol_whitelist file,crypto,udp,rtp,pipe -i 
pipe:0 map 0:0 foo.wav

ffmpeg stderr says: ffmpeg version N-85641-gdd49eff-tessus Copyright (c) 
2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg 
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl 
--enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm 
--enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopus --enable-libschroedinger 
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora 
--enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzmq --enable-libzvbi --enable-version3 
--disable-ffplay --disable-indev=qtkit

ffmpeg stderr says:   libavutil  55. 61.100 / 55. 61.100
  libavcodec 57. 93.100 / 57. 93.100
  libavformat57. 72.101 / 57. 72.101
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 86.100 /  6. 86.100
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) 
with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with 
argument '1'.
Reading option '' ... matched as output url.
Reading option '-protocol_whitelist' ... matched as AVOption 
'protocol_whitelist' with argument 'file,crypto,udp,rtp,pipe'.
Reading option '-i' ... matched as input url with argument 'pipe:0'.
Reading option 'map' ... matched as output url.
Reading option '0:0' ... matched as output url.
Reading option 'foo.wav' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url pipe:0.
Successfully parsed a group of options.
Opening an input file: pipe:0.
[sdp @ 0x7fd483808000] Format sdp probed with size=2048 and score=50

ffmpeg stderr says: [sdp @ 0x7fd483808000] audio codec set to: pcm_mulaw
[sdp @ 0x7fd483808000] audio samplerate set to: 8000
[sdp @ 0x7fd483808000] audio channels set to: 1
[sdp @ 0x7fd483808000] audio codec set to: pcm_alaw
[sdp @ 0x7fd483808000] audio samplerate set to: 8000
[sdp @ 0x7fd483808000] audio channels set to: 1
[sdp @ 0x7fd483808000] audio codec set to: pcm_alaw
[sdp @ 0x7fd483808000] audio samplerate set to: 8000
[sdp @ 0x7fd483808000] audio channels set to: 1
[udp @ 0x7fd4835012e0] end receive buffer size reported is 65536
[udp @ 0x7fd4835014e0] end receive buffer size reported is 65536
[sdp @ 0x7fd483808000] setting jitter buffer size to 500
[sdp @ 0x7fd483808000] Before avformat_find_stream_info() pos: 286 bytes 
read:286 seeks:0 nb_streams:1



start pcap reader..
running 
/Users/dhorton/beachdog-enterprises/beachdog-networks/git/voxbone/node-transcode-pcap/bin/pcapreader
args: ["--port","40558","test/rtpengine.pcap"]
ffmpeg stderr says: [sdp @ 0x7fd483808000] All info found

ffmpeg stderr says: [sdp @ 0x7fd483808000] After avformat_find_stream_info() 
pos: 286 bytes read:286 seeks:0 frames:1

ffmpeg stderr says: Guessed Channel Layout for Input Stream #0.0 : mono
Input #0, sdp, from 'pipe:0':
  Metadata:
title   : 

[FFmpeg-user] RTSP Authentication

[André Silva]

favorite


I've been currently trying to set up an encoding process from RTSP to HLS using 
FFmpeg, with success for most streams.

However, for streams/cameras with digest authentication, FFmpeg seems to fail 
with a classic 401 Unauthorized, as such:

ffmpeg -loglevel debug -i "rtsp://user:password@192.168.0.1/stream" 
/folder/output.m3u8

Giving:

ffmpeg version N-85750-ga75ef15 Copyright (c) 2000-2017 the FFmpeg developers

built with gcc 6.3.0 (GCC)

  configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid 
--enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc 
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r 
--enable-gnutls --enable-iconv --enable-libass --enable-libbluray 
--enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme 
--enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 
--enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libsnappy 
--enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame 
--enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 
--enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-zlib

  libavutil  55. 61.100 / 55. 61.100

  libavcodec 57. 93.100 / 57. 93.100

  libavformat57. 72.101 / 57. 72.101

  libavdevice57.  7.100 / 57.  7.100

  libavfilter 6. 88.100 /  6. 88.100

  libswscale  4.  7.101 /  4.  7.101

  libswresample   2.  8.100 /  2.  8.100

  libpostproc54.  6.100 / 54.  6.100

Splitting the commandline.

Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) 
with argument 'debug'.

Reading option '-i' ... matched as input url with argument 
'rtsp://username:password@192.168.0.1/stream'.

Reading option '/folder/output.m3u8' ... matched as output url.

Finished splitting the commandline.

Parsing a group of options: global .

Applying option loglevel (set logging level) with argument debug.

Successfully parsed a group of options.

Parsing a group of options: input url 
rtsp://username:password@192.168.0.1/stream2.

Successfully parsed a group of options.

Opening an input file: rtsp://username:password@192.168.0.1/stream.

[tcp @ 025c3900] No default whitelist set

[rtsp @ 025c2560] method OPTIONS failed: 401 Unauthorized

[rtsp @ 025c2560] CSeq: 2

WWW-Authenticate: Digest realm="Use 'live' as User Name", 
nonce="2ae726f5557769220b780deb4f562226", algorithm=MD5, qop="auth"

rtsp://user:password@192.168.0.1/stream: Server returned 401 Unauthorized 
(authorization failed)

I've checked that the camera is accessible and the input URI works in VLC. The 
camera I'm trying to obtain the stream from is a Bosch one.

I've searched and searched over the Internet but I didn't find any concrete 
solution to this.

Any ideas?


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[FFmpeg-user] Two frames trimmed of source start

[Bouke / VideoToolShed]

Hi guys,
Trying to do a simple transcode, but the output is missing 2 frames at the 
start.
I can live with that, but the timecode is then off by 2 frames as well.
How can I tell if the trimming will / will not occur so I can compensate the TC 
stamp?
(Or even better, can I avoid the trimming?)
thx,

Bouke

Boukes-portable:~ bouke$ /Users/bouke/Desktop/ffmpeg -i 
/Users/bouke/Desktop/Judith/LV6A3186_0226D1A.mov 
/Users/bouke/Desktop/Judith/test.mp4
ffmpeg version 3.3.1 Copyright (c) 2000-2017 the FFmpeg developers
  built with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads 
--enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx 
--enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 
--enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb 
--enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 
--disable-doc --arch=x86_64 --enable-runtime-cpudetect
  libavutil  55. 58.100 / 55. 58.100
  libavcodec 57. 89.100 / 57. 89.100
  libavformat57. 71.100 / 57. 71.100
  libavdevice57.  6.100 / 57.  6.100
  libavfilter 6. 82.100 /  6. 82.100
  libswscale  4.  6.100 /  4.  6.100
  libswresample   2.  7.100 /  2.  7.100
  libpostproc54.  5.100 / 54.  5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 
'/Users/bouke/Desktop/Judith/LV6A3186_0226D1A.mov':
  Metadata:
major_brand : qt
minor_version   : 0
compatible_brands: qt
creation_time   : 2017-09-22T13:07:25.00Z
  Duration: 00:00:06.17, start: 0.00, bitrate: 14219 kb/s
Stream #0:0(und): Video: prores (apcn / 0x6E637061), yuv422p10le(bt709, 
progressive), 480x270, 12679 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 24k 
tbn, 24k tbc (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  encoder : Apple ProRes 422
  timecode: 08:38:58:09
Stream #0:1(und): Audio: pcm_s16le (lpcm / 0x6D63706C), 48000 Hz, stereo, 
s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
Stream #0:2(und): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  timecode: 08:38:58:09
Stream mapping:
  Stream #0:0 -> #0:0 (prores (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
No pixel format specified, yuv422p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x7fe83b026600] using SAR=1/1
[libx264 @ 0x7fe83b026600] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 
AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 0x7fe83b026600] profile High 4:2:2, level 2.1, 4:2:2 8-bit
[libx264 @ 0x7fe83b026600] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 
2003-2016 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 
deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 
fast_pskip=1 chroma_qp_offset=-2 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=2 
keyint=250 keyint_min=23 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf 
mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to '/Users/bouke/Desktop/Judith/test.mp4':
  Metadata:
major_brand : qt
minor_version   : 0
compatible_brands: qt
encoder : Lavf57.71.100
Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv422p, 
480x270 [SAR 1:1 DAR 16:9], q=-1--1, 23.98 fps, 24k tbn, 23.98 tbc (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  timecode: 08:38:58:09
  encoder : Lavc57.89.100 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1(und): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, 
stereo, fltp, 128 kb/s (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  encoder : Lavc57.89.100 aac
frame=   86 fps=0.0 q=28.0 size=  46kB time=00:00:04.52 bitrate=  
82.6kbits/frame=  148 fps=127 q=-1.0 Lsize= 220kB time=00:00:06.18 bitrate= 
291.9kbits/s speed=5.32x
video:213kB audio:2kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 2.750126%
[libx264 @ 0x7fe83b026600] frame I:1 Avg QP:23.73  size:  2878
[libx264 @ 0x7fe83b026600] frame P:40Avg QP:23.76  size:  3321
[libx264 @ 0x7fe83b026600] frame B:107   Avg QP:26.74  size:   762
[libx264 @ 

Re: [FFmpeg-user] Error converting 4K DCP

[Kieran O Leary]

Hi Carles,

On Fri, Sep 22, 2017 at 3:10 PM, Carles Vila  wrote:
>> Seems like your source has four components, maybe an alpha channel? I'd
>> like to see the pix_fmt, but maybe ffmpeg gives the same error.
>>
>> Not sure if using libopenjpeg as decoder would help?
>> Just add
>> -c:v libopenjpeg
>> Before your
>> -i
>>
>
> Hi! I claimed victory too early. Libopenjpeg starts the conversion, but xyz
> color is not convertred automatically to RGB ...
> Any way to force the conversion?

You can take the advice already given: update ffmpeg!! Older versions
of ffmpeg required an rgb2xyz command, but now the transform is done
for you. You also mentioned that libopenjpeg was slower. Again, you
need to update ffmpeg which will also hopefully update libopenjpeg.
From my experience, libopenjpeg is faster and better than the native
ffmpeg jpeg2000 encoder.

Also you will encounter some colour issues in your output file, mostly
that everything will look too green. A tool like Fraunhofer EasyDCP
PLayer will apply the correct LUT that will fix this but from what I
can see, you can't do this in ffmpeg as of yet. Here's some attempts
of mine to get it working:
https://ffmpeg.org/pipermail/ffmpeg-user/2016-November/034223.html

The issue is described better on page 30/31 of this manual:
https://www.easydcp.com/sync/manuals/easyDCP_Player_User_Manual.pdf

Best,

-Kieran.
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Re: [FFmpeg-user] Error converting 4K DCP

[Carles Vila]

> Seems like your source has four components, maybe an alpha channel? I'd
> like to see the pix_fmt, but maybe ffmpeg gives the same error.
>
> Not sure if using libopenjpeg as decoder would help?
> Just add
> -c:v libopenjpeg
> Before your
> -i
>

Hi! I claimed victory too early. Libopenjpeg starts the conversion, but xyz
color is not convertred automatically to RGB ...
Any way to force the conversion?
Thanks
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Re: [FFmpeg-user] Error converting 4K DCP

[Reto Kromer]

Carles Vila wrote:

>ffmpeg version 2.7.2 Copyright (c) 2000-2015 the FFmpeg
>developers

You might possibly wish to upgrade to the current release 3.3.4.

Best regards, Reto

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Re: [FFmpeg-user] Error converting 4K DCP

[Carles Vila]

>
> Seems like your source has four components, maybe an alpha channel? I'd
> like to see the pix_fmt, but maybe ffmpeg gives the same error.
>
> Not sure if using libopenjpeg as decoder would help?
> Just add
> -c:v libopenjpeg
> Before your
> -i
>
>
Hi thanks for the tip! it did the trick, almost half as slow but it's OK...

for reference this is the info when using libopenjpeg

ffmpeg version 2.7.2 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 5.1 (clang-503.0.40) (based on LLVM 3.4svn)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/2.7.2_1 --enable-shared
--enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables
--enable-avresample --cc=clang --host-cflags= --host-ldflags=
--enable-opencl --enable-libx264 --enable-libmp3lame --enable-libvo-aacenc
--enable-libxvid --enable-libfreetype --enable-libvorbis --enable-libvpx
--enable-libass --enable-ffplay --enable-libfdk-aac --enable-libopus
--enable-libquvi --enable-libx265 --enable-libopenjpeg
--extra-cflags='-I/usr/local/Cellar/openjpeg/1.5.2_1/include/openjpeg-1.5 '
--enable-nonfree --enable-vda
  libavutil  54. 27.100 / 54. 27.100
  libavcodec 56. 41.100 / 56. 41.100
  libavformat56. 36.100 / 56. 36.100
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 16.101 /  5. 16.101
  libavresample   2.  1.  0 /  2.  1.  0
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  2.100 /  1.  2.100
  libpostproc53.  3.100 / 53.  3.100
[mxf @ 0x7fd311816400] "OPAtom" with 2 ECs - assuming OP1a
Input #0, mxf, from 'a297c04a-f068-49d0-8213-2b45d58676be_j2c.mxf':
  Metadata:
uid : a3c74cd5-74f3-4398-ae9a-d6cf689e7907
generation_uid  : 5d574c48-8ad3-459a-904b-a3b0467a1259
company_name: Fraunhofer IIS
product_name: easyDCP Creator
product_version : 2.2.3
product_uid : 7d836e16-37c7-4c22-b2e0-46a717e84f42
modification_date: 2017-09-21 16:21:42
application_platform: i386-apple-darwin9.8.0
material_package_umid:
0x060A2B340101010501010F201300F54E66ED0BFE42B18208713C74126BC0
material_package_name: AS-DCP Material Package
timecode: 00:00:00:00
  Duration: 00:00:07.08, start: 0.00, bitrate: 12475 kb/s
Stream #0:0: Video: jpeg2000, rgb48le(12 bpc), 3996x2160, SAR 1:1 DAR
37:20, 24 tbr, 24 tbn, 24 tbc
Metadata:
  file_package_umid:
0x060A2B340101010501010F201300A297C04AF06849D082132B45D58676BE
  file_package_name: File Package: SMPTE 429-4 frame wrapping of JPEG
2000 codestreams
Output #0, mov, to '/Volumes/VIDEO_E/***.mov':
  Metadata:
uid : a3c74cd5-74f3-4398-ae9a-d6cf689e7907
generation_uid  : 5d574c48-8ad3-459a-904b-a3b0467a1259
company_name: Fraunhofer IIS
product_name: easyDCP Creator
product_version : 2.2.3
product_uid : 7d836e16-37c7-4c22-b2e0-46a717e84f42
modification_date: 2017-09-21 16:21:42
application_platform: i386-apple-darwin9.8.0
material_package_umid:
0x060A2B340101010501010F201300F54E66ED0BFE42B18208713C74126BC0
material_package_name: AS-DCP Material Package
timecode: 00:00:00:00
encoder : Lavf56.36.100
Stream #0:0: Video: prores (apch) (apch / 0x68637061), yuv422p10le,
3996x2160 [SAR 1:1 DAR 37:20], q=2-31, 200 kb/s, 24 fps, 12288 tbn, 24 tbc
Metadata:
  file_package_umid:
0x060A2B340101010501010F201300A297C04AF06849D082132B45D58676BE
  file_package_name: File Package: SMPTE 429-4 frame wrapping of JPEG
2000 codestreams
  encoder : Lavc56.41.100 prores
Stream mapping:
  Stream #0:0 -> #0:0 (jpeg2000 (libopenjpeg) -> prores (native))
Press [q] to stop, [?] for help
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Re: [FFmpeg-user] configuration error when cross compile ffmpeg for arm

[Moritz Barsnick]

On Wed, Sep 20, 2017 at 16:38:55 +0800, 云雀 wrote:
> I planed to cross compiling C++ program from my PC(Ubuntu 64 bits) to Inforce 
> 6540(32 bits Ubuntu 14.10 armV7); this program used openCV library,in order 
> to finish this job, i need cross compiling dependency library of openCV 
> first, ffmpeg is one of them, after cross compiling 
> libjpeg,libpng,yasm,libx264,libxvid, i can not pass cross compiling of 
> ffmpeg, configure program showing error: libx264 not found; config.log show 
> as below showing(enclosure is config log file):
> /lib/libx264.so: file not recognized: File format not recognized
[...]
> # ./configure --prefix= --enable-shared --disable-static --enable-pic 
> --enable-gpl --enable-cross-compile --arch=arm --disable-stripping 
> --target-os=linux --enable-libx264 --enable-libxvid 
> --cc=arm-linux-gnueabihf-gcc --enable-swscale --extra-ldflags=-L/lib 
> --extra-cflags=-I/include

These options in that configure line:
  --extra-ldflags=-L/lib --extra-cflags=-I/include
are probably wrong. They need to point to your cross-compiled libraries
and includes, not to these system directories. ("/include" probably
doesn't even exist.)

I am guessing you called this configure line from a script, and it said
"--extra-ldflags=-L$path_to_something/lib", and $path_to_something was
empty, e.g. due to a typo.

Moritz
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Re: [FFmpeg-user] FFmpeg交叉编译openssl时出错

[Moritz Barsnick]

On Tue, Sep 19, 2017 at 17:54:52 +0800, l...@tybofone.com wrote:
> arm-none-linux-gnueabi-gcc -D_ISOC99_SOURCE -D_FILE_OFFSET_BITS=64 
> -D_LARGEFILE_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -DPIC 
> -march=armv5te -std=c99 -fomit-frame-pointer -fPIC -marm -pthread -E -o 
> /tmp/ffconf.Lhn5rbJM.o /tmp/ffconf.Q2c8r4Tb.c
> /tmp/ffconf.Q2c8r4Tb.c:1:25: fatal error: openssl/ssl.h: No such file or 
> directory

You need to install not only the openssl library, but also the
development part of it (in a packaged system it's openssl-dev or
openssl-devel). And since you're cross compiling, these need to be for
the target platform.

If you built your own openssl for the target platform, this needs to be
found by pointing some configure flags at it:
$ ./configure [...] 
--extra-cflags="-I/my/path/were/i/keep/built/arm/stuff/include" 
--extra-ldflags="-L/my/path/were/i/keep/built/arm/stuff/lib"
approximately as described here:
https://trac.ffmpeg.org/wiki/CompilationGuide/RaspberryPi
(Consider Raspberry Pi as yet another ARM platform.)

Moritz
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Re: [FFmpeg-user] FFMPEG Images & Audio to Video

[Kieran O Leary]

How many source images do you have, and should the output definitely be
25fps?also why not use something like
https://amiaopensource.github.io/ffmprovisr/#images_2_video
To process an image sequence rather than concat? Unless your images are
appended by datetime rather than sequential numbers?
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Re: [FFmpeg-user] configuration error when cross compile ffmpeg for arm

[Yang Zhang]

I am not a expert, but the problem seems is caused by libx264.so,
please try install both 32 and 64 bit libx264. and try again

On 9/20/17, 云雀 <645924...@qq.com> wrote:
> Hi,
>
> Hope this mail found you well!
>
> I planed to cross compiling C++ program from my PC(Ubuntu 64 bits) to
> Inforce 6540(32 bits Ubuntu 14.10 armV7); this program used openCV
> library,in order to finish this job, i need cross compiling dependency
> library of openCV first, ffmpeg is one of them, after cross compiling
> libjpeg,libpng,yasm,libx264,libxvid, i can not pass cross compiling of
> ffmpeg, configure program showing error: libx264 not found; config.log show
> as below showing(enclosure is config log file):
> /lib/libx264.so: file not recognized: File format not recognized
>
>
> I can not find solution from internet network, so i send this email,
> appreciate any advise.
>
>
> Best regards,
>
> Tom
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[FFmpeg-user] FFMPEG Images & Audio to Video

[Vivek Jain]

Hello,

Humble greetings! I have been using ffmpeg to create video from an audio
and set of images (as per command below). Lately I have been noticing
ffmpeg's video output is lengthier than actual length. The produced video
has an extra padding in the end with no audio. Could someone please advise
how can the extra padding in the end be removed?

*Please note that audio's length and combined images duration length is
exactly the same.*

*Command:*

ffmpeg -safe 0 -f concat -i /tmp/90340433636652_config.txt -i
/tmp/90340433636652_audio.mp3 -y -vb 8M -vcodec libx264 -pix_fmt
yuv420p -vf 'scale=trunc(iw/2)*2:trunc(ih/2)*2'
/tmp/90340433636652_mashupVideo.mp4

In the example attached, below are the media lengths:

Audio length = 00:10:54.39

Combined Images length = 00:10:54.36

Output Video length = 00:14:18.12 *(which is 3 minutes 24 seconds extra)*

*Output Attached!*
Appreciate much for your help!
-- 
Thanks,
Vivek

FFMPEG Command Used to create video


ubuntu@ffmpeg-workers:~$ ffmpeg -safe 0 -f concat -i 
/tmp/90340433636652_config.txt -i /tmp/90340433636652_audio.mp3 -y -vb 
8M -vcodec libx264 -pix_fmt yuv420p -vf 'scale=trunc(iw/2)*2:trunc(ih/2)*2' 
/tmp/90340433636652_mashupVideo.mp4
ffmpeg version git-2017-01-22-f1214ad Copyright (c) 2000-2017 the FFmpeg 
developers
  built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.3)
  configuration: --extra-libs=-ldl --prefix=/opt/ffmpeg --mandir=/usr/share/man 
--enable-avresample --disable-debug --enable-nonfree --enable-gpl 
--enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb 
--disable-decoder=amrnb --disable-decoder=amrwb --enable-libpulse 
--enable-libfreetype --enable-gnutls --enable-libx264 --enable-libx265 
--enable-libfdk-aac --enable-libvorbis --enable-libmp3lame --enable-libopus 
--enable-libvpx --enable-libspeex --enable-libass --enable-avisynth 
--enable-libsoxr --enable-libxvid --enable-libvidstab --enable-libwavpack 
--enable-nvenc
  libavutil  55. 44.100 / 55. 44.100
  libavcodec 57. 75.100 / 57. 75.100
  libavformat57. 63.100 / 57. 63.100
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 69.100 /  6. 69.100
  libavresample   3.  2.  0 /  3.  2.  0
  libswscale  4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100
Input #0, concat, from '/tmp/90340433636652_config.txt':
  Duration: 00:10:54.36, start: 0.00, bitrate: 0 kb/s
Stream #0:0: Video: png, rgb24(pc), 720x540 [SAR 72:72 DAR 4:3], 25 tbr, 25 
tbn, 25 tbc
Input #1, mp3, from '/tmp/90340433636652_audio.mp3':
  Metadata:
creationdate: Thu Sep 15 11:24:10
description : Recorded using WebcamRecording example.
encoder : Lavf57.25.100
  Duration: 00:10:54.39, start: 0.025057, bitrate: 128 kb/s
Stream #1:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
  encoder : Lavc57.24
[libx264 @ 0x24a77a0] using SAR=1/1
[libx264 @ 0x24a77a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX 
FMA3 AVX2 LZCNT BMI2
[libx264 @ 0x24a77a0] profile High, level 3.1
[libx264 @ 0x24a77a0] 264 - core 148 r2643 5c65704 - H.264/MPEG-4 AVC codec - 
Copyleft 2003-2015 - http://www.videolan.org/x264.html - options: cabac=1 ref=3 
deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 
mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 
fast_pskip=1 chroma_qp_offset=-2 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=2 
keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=abr 
mbtree=1 bitrate=8000 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 
ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to '/tmp/90340433636652_mashupVideo.mp4':
  Metadata:
encoder : Lavf57.63.100
Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 
720x540 [SAR 1:1 DAR 4:3], q=-1--1, 8000 kb/s, 25 fps, 12800 tbn, 25 tbc
Metadata:
  encoder : Lavc57.75.100 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/800 buffer size: 0 vbv_delay: -1
Stream #0:1: Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, stereo, 
fltp, 128 kb/s
Metadata:
  encoder : Lavc57.75.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (png (native) -> h264 (libx264))
  Stream #1:0 -> #0:1 (mp3 (native) -> aac (native))
Press [q] to stop, [?] for help
More than 1000 frames duplicated 58kB time=00:00:04.08 bitrate= 
115.9kbits/s dup=153 drop=0 speed=8.14x
More than 1 frames duplicated  4799kB time=00:03:52.44 bitrate= 
169.1kbits/s dup=5861 drop=0 speed=10.5x
frame=21453 fps=276 q=-1.0 Lsize=   18179kB time=00:14:18.00 bitrate= 
173.6kbits/s dup=26695 drop=0 speed=11.1x
video:7435kB audio:10369kB subtitle:0kB other