[FFmpeg-user] How to generate IRT conform D10 MXFs (IMX50) with FFmpeg?

[Christoph Gerstbauer]

Hello

I am making tests with IMX50/MXF encodings with FFMBC and FFMPEG. (with
24bit audio and 16bit audio)
The tests compare the final files with the IRT MXF Analyzer Pro tool to
be shure to be IRT conform.
(My commandline outputs are at the bottom of this mail)

Both files (generated with ffmbc and ffmpeg) are working very well in
editing tools but there is a little difference:

The IMX50 generated by ffmbc has no errors/warnings at IRT MXF Analyzer
(with 16bit audio). It is really perfect. In fact (with one exception) I 
NEVER had a clean "green" MXF by testing other probes from other 
encoding tools. Only the IRT probes itself are clean (of course). So 
ffmbc MXF is IRT conform when using the

standard target preset "imx50". (it automatically uses 16bit audio)

When I use 24bit audio encoding at ffmbc, I got this error from IRT MXF 
Analyzer:

ERROR: "The Sound Essence Descriptor signals that the number if
quantization bits is 24, which is higher than actually required (16) as
detemined by parsing the essence."

I tried the same with ffmpeg:
When using 16bit audio, I got no audio error. But I get always this warning:
WARNING:
"Fill Item Key with a wrong RP210 (SMPTE Metadata Dictionary) version
number of 0x01. The correct version number is 0x02"

When using 24bit audio I get the same error like in ffmbc with 24bit audio:
ERROR: "The Sound Essence Descriptor signals that the number if
quantization bits is 24, which is higher than actually required (16) as
detemined by parsing the essence."

So, I could always prevent the audio WARNING by producing 16bit audio.
But is there a problem in the MXF when producing 24bit? Sound works in 
editing tools.



So I have 2 questions (because I want to use ffmpeg instead of ffmbc)

1.) Is there an syntax option in ffmpeg/ffmbc for 24bit audio encoding to
prevent this ERROR:
"The Sound Essence Descriptor signals that the number if quantization
bits is 24, which is higher than actually required (16) as detemined by
parsing the essence."

2.) Is there an syntax option in ffmpeg to prevent this warning?
WARNING:
"Fill Item Key with a wrong RP210 (SMPTE Metadata Dictionary) version
number of 0x01. The correct version number is 0x02"


Every hint would be very helpful, thank you!

Best Regards
Christoph




USED SYNTAX:

-

FFmbc (16bit audio):

ffmbc -i I:\IMX_tests\IMX50_CARBON.avi -target imx50
I:\IMX_tests\_FFMBC_PAL_IMX_16bit_2.mxf -loglevel debug
FFmbc version 0.7-rc7
Copyright (c) 2008-2012 Baptiste Coudurier and the FFmpeg developers
Input #0, avi, from 'I:\IMX_tests\IMX50_CARBON.avi':
   Metadata:
 encoder: Lavf55.0.100
 ISMP: 00:00:00:00
   Duration: 00:00:20.00, start: 0.00, bitrate: 113291 kb/s
 Stream #0.0(und): Video: ffvhuff, yuv422p, 720x608p, PAR 152:135
DAR 4:3, 25.00 fps
 Stream #0.1(und): Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s
File 'I:\IMX_tests\_FFMBC_PAL_IMX_16bit_2.mxf' already exists. Overwrite
? [y/N] y
[buffer @ 018E07E0] w:720 h:608 pixfmt:yuv422p tb:1/100
sar:152/135 sws_param:
Incompatible sample format 's32' for codec 'pcm_s16le', auto-selecting
format 's16'
Warning, QMAT_SHIFT is larger than 21, overflows possible
 Last message repeated 1 times
Output #0, mxf_d10, to 'I:\IMX_tests\_FFMBC_PAL_IMX_16bit_2.mxf':
   Metadata:
 encoder: FFmbc 0.7
 Stream #0.0(und), 0, 1/25: Video: mpeg2video, yuv422p, 720x608i tff
[PAR 152:135 DAR 4:3], tbc 1/25, cbr, 5 kb/s, 25.00 fps
 Stream #0.1(und), 0, 1/48000: Audio: pcm_s16le, 48000 Hz, stereo,
s16, 1536 kb/s
Stream mapping:
   Stream #0.0 -> #0.0
   Stream #0.1 -> #0.1
Press [q] to stop, [?] for help
[mxf_d10 @ 0037C9C0] package type:1
[mxf_d10 @ 0037C9C0] package type:2
[mxf_d10 @ 0037C9C0] edit units count 0
frame=  239 fps= 25 q=3.0 size=   72409kB time=00:00:09.52
bitrate=62307.8kbits/s eta=00:00:10.51

-

FFmbc (24bit audio):

ffmbc -i I:\IMX_tests\IMX50_CARBON.avi -target imx50 -acodec pcm_s24le
I:\IMX_tests\_FFMBC_PAL_IMX_24bit_2.mxf -loglevel debug
FFmbc version 0.7-rc7
Copyright (c) 2008-2012 Baptiste Coudurier and the FFmpeg developers
Input #0, avi, from 'I:\IMX_tests\IMX50_CARBON.avi':
   Metadata:
 encoder: Lavf55.0.100
 ISMP: 00:00:00:00
   Duration: 00:00:20.00, start: 0.00, bitrate: 113291 kb/s
 Stream #0.0(und): Video: ffvhuff, yuv422p, 720x608p, PAR 152:135
DAR 4:3, 25.00 fps
 Stream #0.1(und): Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s
[buffer @ 002EFF60] w:720 h:608 pixfmt:yuv422p tb:1/100
sar:152/135 sws_param:
Warning, QMAT_SHIFT is larger than 21, overflows possible
 Last message repeated 1 times
Output #0, mxf_d10, to 'I:\IMX_tests\_FFMBC_PAL_IMX_24bit_2.mxf':
   Metadata:
 encoder: FFmbc 0.7
 Stream #0.0(und), 0, 1/25: Video: mpeg2video, yuv422p, 720x608i tff
[PAR 152:135 DAR 4:3], tbc 1/25, cbr, 5 kb/s, 25.00 fps
 St

Re: [FFmpeg-user] delay problems when using ffserver, ffmpeg for streaming of web cam and vlc for viewing.

[Adrian Schwartz]

I forgot to mention, that over Ethernet there is still a ~2 sec delay?

On Thu, Apr 30, 2015 at 9:00 AM, Adrian Schwartz 
wrote:

> Hi Roger
> When I changed the network interface from wireless to Ethernet, I stopped
> seeing the past duration to large error.
> I do see dup=[number] at the end and drop=0, what does it means?
>
>
> On Wed, Apr 29, 2015 at 9:24 PM, Roger Pack  wrote:
>
>> Do you get past duration too large when just saving to file?
>>
>> On 4/29/15, Adrian Schwartz  wrote:
>> > hello ffmpeg user group,
>> > I'm having delay problem ( when using ffserver and ffmpeg)
>> > I'm getting the error "ffmpeg past duration too large" when using the
>> > ffmpeg for streaming.
>> >  VLC is configured to work in minimum delay and when streaming from //
>> > 192.168.1.1:8090/test1.asf it shows 2 sec delay, when the communication
>> > interface delay is aroung 15 msec.
>> >
>> >
>> > Is there some recommendation on how to lower the delay?
>> > Regards
>> > Adrian
>> >
>> > #
>> > #Streaming command:
>> > #
>> >
>> > ffmpeg -f dshow  -rtbufsize 1M   -i video="Logitech
>> > Webca"":audio="Microphone (2- Webcam C160)" -fflags nobuffer -tune
>> > zerolatency http://192.168.1.1:8090/feed1.ffm
>> >
>> > 
>> > ##ffserver config:
>> > 
>> >
>> >
>> > Port 8090
>> > BindAddress 0.0.0.0
>> > MaxClients 1000
>> > MaxBandwidth 3
>> > CustomLog /var/log/ffserver.log
>> > NoDaemon
>> >
>> > 
>> > File /tmp/feed1.ffm
>> > FileMaxSize 1M
>> > ACL allow 192.168.26.0 192.168.29.255
>> > ACL allow 127.0.0.1
>> > ACL allow 10.0.0.0 10.0.0.255
>> > 
>> >
>> >
>> >
>> > 
>> > Feed feed1.ffm
>> > Format asf
>> > VideoFrameRate 15
>> > VideoSize 352x240
>> > VideoBitRate 256
>> > VideoGopSize 30
>> > StartSendOnKey
>> > NoAudio
>> > 
>> >
>> > 
>> > Format status
>> > ACL allow localhost
>> > ACL allow 192.168.0.0 192.168.255.255
>> >
>> > 
>> > ___
>> > ffmpeg-user mailing list
>> > ffmpeg-user@ffmpeg.org
>> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> >
>> ___
>> ffmpeg-user mailing list
>> ffmpeg-user@ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>
>
>
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] delay problems when using ffserver, ffmpeg for streaming of web cam and vlc for viewing.

[Adrian Schwartz]

Hi Roger
When I changed the network interface from wireless to Ethernet, I stopped
seeing the past duration to large error.
I do see dup=[number] at the end and drop=0, what does it means?


On Wed, Apr 29, 2015 at 9:24 PM, Roger Pack  wrote:

> Do you get past duration too large when just saving to file?
>
> On 4/29/15, Adrian Schwartz  wrote:
> > hello ffmpeg user group,
> > I'm having delay problem ( when using ffserver and ffmpeg)
> > I'm getting the error "ffmpeg past duration too large" when using the
> > ffmpeg for streaming.
> >  VLC is configured to work in minimum delay and when streaming from //
> > 192.168.1.1:8090/test1.asf it shows 2 sec delay, when the communication
> > interface delay is aroung 15 msec.
> >
> >
> > Is there some recommendation on how to lower the delay?
> > Regards
> > Adrian
> >
> > #
> > #Streaming command:
> > #
> >
> > ffmpeg -f dshow  -rtbufsize 1M   -i video="Logitech
> > Webca"":audio="Microphone (2- Webcam C160)" -fflags nobuffer -tune
> > zerolatency http://192.168.1.1:8090/feed1.ffm
> >
> > 
> > ##ffserver config:
> > 
> >
> >
> > Port 8090
> > BindAddress 0.0.0.0
> > MaxClients 1000
> > MaxBandwidth 3
> > CustomLog /var/log/ffserver.log
> > NoDaemon
> >
> > 
> > File /tmp/feed1.ffm
> > FileMaxSize 1M
> > ACL allow 192.168.26.0 192.168.29.255
> > ACL allow 127.0.0.1
> > ACL allow 10.0.0.0 10.0.0.255
> > 
> >
> >
> >
> > 
> > Feed feed1.ffm
> > Format asf
> > VideoFrameRate 15
> > VideoSize 352x240
> > VideoBitRate 256
> > VideoGopSize 30
> > StartSendOnKey
> > NoAudio
> > 
> >
> > 
> > Format status
> > ACL allow localhost
> > ACL allow 192.168.0.0 192.168.255.255
> >
> > 
> > ___
> > ffmpeg-user mailing list
> > ffmpeg-user@ffmpeg.org
> > http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> >
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] Creating Streaming Server

[ELAMPOORNAN.ANANTHANESHAN]

Hi,
I would like to stream my web cam to a server url.rtmp://
a.rtmp.myserver.com/live2) and record it on the server. I would like to use
RTMPE. Is that possible with FFMPEG if so can you recommend some code
sample.

I wonder, how would Amazon services help this process.

 *Regards,*
E Ananthaneshan
T.P: +9471 23 07125
E-mail: poor...@gmail.com
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] HLS one liner from /dev/videoX

[Kai Hendry]

Hi guys,

I've received an answer on http://stackoverflow.com/a/29952909/4534
which doesn't work for me. I am simply trying to create a browser
friendly stream in my LAN from my /dev/video1 device.

Here is the report log:
http://s.natalian.org/2015-04-30/ffmpeg-20150430-102645.log
http://ix.io/icO is what I saw
Here is the tarball of the output http://s.natalian.org/2015-04-30/output.tar.gz


IIUC, `curl -s http://s.natalian.org/2015-04-30/output.tar.gz | tar
zxvf -` and then `python -m http.server` gets everything setup on my
desktop for playing back (last few seconds of?) the live recorded
stream. Then accessing playlist.m3u8 on Safari "Mozilla/5.0 (iPhone;
CPU iPhone OS 8_3 like Mac OS X) AppleWebKit/600.1.4 (KHTML, like
Gecko) Version/8.0 Mobile/12F70 Safari/600.1.4" will fail to dark grey
screen.


Before I make a bug report, am I missing something?

I've tried -hls_flags live=1 & -segment_list_flags +live to no avail.

Also I've noticed a discrepancy in the docs, I think!
https://www.ffmpeg.org/ffmpeg-formats.html#Options-2 says the Default
value of playlist entries is 5. Yet as you can see from my tarball
there is about 22 .ts files after running it for just a minute or so.

Sidenote: Would be great to have the MPEG-DASH version of command.
https://www.ffmpeg.org/ffmpeg-formats.html#webm_005fdash_005fmanifest
is not clear to me. Does Chrome playback from the manifest or these
*.chk files? https://www.ffmpeg.org/ffmpeg-formats.html#Example-3
didn't work for me.
http://s.natalian.org/2015-04-30/1430361818_1362x740.png

Many thanks,
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] cannot normalize MP4 from RTMP live recorded stream

[Madovsky]

Hi,

I just recorded some rtmp mp4 live stream from flash media server
and concatenate on the fly the stream with server side function that append
recorded stream to the current one.
once done I can read the stream correctly from flash player
but from other players (like VLC or ffplay) it's absolute crap. size is 
not respected and quality is awful,

more, the concatenate file is not shown.
so I tried to encode with ffmpeg but it says that "concatenated mp4 
files is not supported correctly, patch welcome"
the metadata is not caught by ffmpeg and the quality is still awful, 
even if I use -vcodec copy. sound is OK

the command line I tried is

ffmpeg -i test.mp4 -async 1 -vsync 1 -apad -vpad -acodec libfdk_aac 
-vcodec copy -crf 23 -y output.mp4


I can provide the video sample

thanks

Franck

___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] ffmpeg - avfoundation Audio stutters / sounds broken with screen capture [sample, config, output included]

[kevin]

On Tue, Apr 28, 2015 at 10:08 PM, kevin  wrote:

> On Mon, Apr 27, 2015 at 6:12 AM, kevin  wrote:
>
>> On Mon, Apr 27, 2015 at 5:11 AM, Matthieu Bouron <
>> matthieu.bou...@gmail.com> wrote:
>>
>>> On Mon, Apr 27, 2015 at 1:49 PM, Carl Eugen Hoyos 
>>> wrote:
>>>
>>> > Matthieu Bouron  gmail.com> writes:
>>> >
>>> > > Does reverting the following commit
>>> > > 1e9d035b91100b75edc964914df638863f188a4f helps you ?
>>> >
>>> > I cannot find this revision...
>>> >
>>>
>>>
>>> Sorry this local one was pointing to the revert. The right one is
>>> cf16b459a873b990efc10d93e9912ab01b8a0fba
>>> "libavdevice/avfoundation: use pts/dts provided by the CMSampleBuffer
>>> API"
>>
>>
>> No this does not fix it. Output file, command, git log below:
>> ./ffmpeg -f avfoundation -video_size 640x480 -framerate 30 -i "1:0" -c:v
>> mpeg4 -c:a aac -strict -2 -r 30 ffmpeg-pts-reverted.mp4
>>
>> https://www.dropbox.com/s/72335m3cxajhkbn/ffmpeg-pts-reverted.mp4?dl=1
>>
>
> Any suggestions? Happy to help in anyway to fix this issue.
>

I created a ticket, posting it here for reference

https://trac.ffmpeg.org/ticket/4513
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] Using expressions in "FADE" filter

[Shlomo Morosow]

I need the ability to input mathematical values for setting the start time
(st)  on the "fade" filter, I've attempted to use an expression and this
particular filter does not seem to work with expressions, nor simple math
as "10*1". I've included here the input and output of two commands which
I've run, the first with a number as the "st" value which ran successfully
and the second with an expression as the "st" value which returned errors.
Does anyone have any creative solutions? perhaps using the "blend" filter?

Thanks very much


Input 1:

ffmpeg.exe -y -ss "0:00:20" -t "0:00:10" -i input.avi -filter_complex
"fade=out:st=9:d=1" -f dvd -target ntsc-dvd -r 29.97 -aspect 4:3 -b:v 4500k
-mbd rd -trellis 1 -flags +mv0 -cmp 0 -subcmp 2 -b:a 128k output.vob


Output 1:

ffmpeg version N-70875-g3c5c687 Copyright (c) 2000-2015 the FFmpeg
developers
  built with gcc 4.9.2 (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-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-lzma --enable-decklink --enable-zlib
  libavutil  54. 20.100 / 54. 20.100
  libavcodec 56. 29.100 / 56. 29.100
  libavformat56. 26.101 / 56. 26.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 13.101 /  5. 13.101
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, avi, from 'input.avi':
  Duration: 00:00:42.54, start: 0.00, bitrate: 30316 kb/s
Stream #0:0: Video: dvvideo (dvsd / 0x64737664), yuv411p, 720x480 [SAR
8:9 DAR 4:3], 28793 kb/s, 29.97 fps, 29.97 tbr, 29.97 tbn, 29.97 tbc
Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2
channels, s16, 1536 kb/s
Output #0, dvd, to 'output.vob':
  Metadata:
encoder : Lavf56.26.101
Stream #0:0: Video: mpeg2video, yuv420p, 720x480 [SAR 8:9 DAR 4:3],
q=2-31, 4500 kb/s, 29.97 fps, 90k tbn, 29.97 tbc
Metadata:
  encoder : Lavc56.29.100 mpeg2video
Stream #0:1: Audio: ac3, 48000 Hz, stereo, fltp, 128 kb/s
Metadata:
  encoder : Lavc56.29.100 ac3
Stream mapping:
  Stream #0:0 (dvvideo) -> fade (graph 0)
  fade (graph 0) -> Stream #0:0 (mpeg2video)
  Stream #0:1 -> #0:1 (pcm_s16le (native) -> ac3 (native))
Press [q] to stop, [?] for help
frame=   88 fps=0.0 q=5.3 size=1736kB time=00:00:03.19
bitrate=4451.6kbits/s
frame=  178 fps=177 q=2.6 size=3332kB time=00:00:06.10
bitrate=4469.8kbits/s
frame=  255 fps=169 q=2.0 size=4378kB time=00:00:09.01
bitrate=3976.7kbits/s
frame=  300 fps=180 q=1.6 Lsize=4536kB time=00:00:10.01
bitrate=3711.9kbits/s
video:4269kB audio:156kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 2.506122%





Input 2:

ffmpeg.exe -y -ss "0:00:20" -t "0:00:10" -i input.avi -filter_complex
"fade=out:st=%{expr\\\: (10-1)}:d=1" -f dvd -target ntsc-dvd -r 29.97
-aspect 4:3 -b:v 4500k -mbd rd -trellis 1 -flags +mv0 -cmp 0 -subcmp 2 -b:a
128k output.vob



Output 2:

ffmpeg version N-70875-g3c5c687 Copyright (c) 2000-2015 the FFmpeg
developers
  built with gcc 4.9.2 (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-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-lzma --enable-decklink --enable-zlib
  libavutil  54. 20.100 / 54. 20.100
  libavcodec 56. 29.100 / 56. 29.100
  libavformat56. 26.101 / 56. 26.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 13.101 /  5. 13.101
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc53.  3.100 / 53.  3.100
Guessed Channel Layout for  Input Stream #0.1 : stereo
Input #0, avi, from 'inpu

Re: [FFmpeg-user] How can I set in a D10 MXF (IMX50) file the flags for output color_range, -space, -transfer and primaries?

[James Heliker]

On 4/29/2015 2:22 PM, Christoph Gerstbauer wrote:
I found out that a IMX50 mxf file encoded with FFmbc and the IMX50 mxf 
file encoded with actual ffmpeg builds are different in these mxf 
metadata flags (by reading out via ffprobe - show_streams)




+1 if anyone can answer this question, it would be very helpful for my 
needs also



FFMBC IMX FILE stream 0:0:
color_range=tv
color_space=smpte170m
color_transfer=bt709
color_primaries=bt470bg


FFMPEG IMX FILE stream 0:0:
color_range=tv
color_space=unknown
color_transfer=unknown
color_primaries=unknown

I want to set these metadata flags to the same values, but with FFmpeg.
How can I produce the same output like ffmbc with these 4 metadata flags?
I didnt found a answer to that in the ffmpeg documentation :/

Best Regards
CHristoph


COMMAND LINE OUTPUT:

FFmbc encoding:

ffmbc -i I:\IMX_tests\IMX50_CARBON.avi -target imx50 -acodec pcm_s24le
I:\IMX_tests\_FFMBC_PAL_IMX_24bit_2.mxf -loglevel debug
FFmbc version 0.7-rc7
Copyright (c) 2008-2012 Baptiste Coudurier and the FFmpeg developers
Input #0, avi, from 'I:\IMX_tests\IMX50_CARBON.avi':
   Metadata:
 encoder: Lavf55.0.100
 ISMP: 00:00:00:00
   Duration: 00:00:20.00, start: 0.00, bitrate: 113291 kb/s
 Stream #0.0(und): Video: ffvhuff, yuv422p, 720x608p, PAR 152:135
DAR 4:3, 25.00 fps
 Stream #0.1(und): Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s
[buffer @ 002EFF60] w:720 h:608 pixfmt:yuv422p tb:1/100
sar:152/135 sws_param:
Warning, QMAT_SHIFT is larger than 21, overflows possible
 Last message repeated 1 times
Output #0, mxf_d10, to 'I:\IMX_tests\_FFMBC_PAL_IMX_24bit_2.mxf':
   Metadata:
 encoder: FFmbc 0.7
 Stream #0.0(und), 0, 1/25: Video: mpeg2video, yuv422p, 720x608i tff
[PAR 152:135 DAR 4:3], tbc 1/25, cbr, 5 kb/s, 25.00 fps
 Stream #0.1(und), 0, 1/48000: Audio: pcm_s24le, 48000 Hz, stereo,
s32, 2304 kb/s
Stream mapping:
   Stream #0.0 -> #0.0
   Stream #0.1 -> #0.1
Press [q] to stop, [?] for help
[mxf_d10 @ 003BF460] package type:1
[mxf_d10 @ 003BF460] package type:2
[mxf_d10 @ 003BF460] edit units count 0
frame=  341 fps= 25 q=3.0 size=  103570kB time=00:00:13.60
bitrate=62385.4kbits/s eta=00:00:06.41

-

FFMPEG encoding:

ffmpegnew -i I:\IMX_tests\IMX50_CARBON.avi -map 0:v -map 0:a -c:v
mpeg2video -r 25 -pix_fmt yuv422p -aspect 4:3 -minrate 5k -maxrate
5k -b:v 5k -intra -flags +ildct+low_del
ay -intra_vlc 1 -non_linear_quant 1 -ps 1 -qmin 1 -qmax 3 -top 1 -dc 10
-bufsize 200 -rc_init_occupancy 200 -rc_max_vbv_use 1
-rc_min_vbv_use 1 -c:a pcm_s24le -ar 48000 -f mxf_d10 I:\IMX_tests
\_FFMPEGNEW_PAL_IMX_24bit_2.mxf
ffmpeg version N-71727-g46778ab Copyright (c) 2000-2015 the FFmpeg
developers
   built with gcc 4.9.2 (GCC)
   configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libblu
ray --enable-libbs2b --enable-libcaca --enable-libdcadec
--enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc
--enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus
--enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-libtwolame
--enable-libvidstab --ena
ble-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena
ble-decklink --enable-zlib
   libavutil  54. 23.101 / 54. 23.101
   libavcodec 56. 35.101 / 56. 35.101
   libavformat56. 31.100 / 56. 31.100
   libavdevice56.  4.100 / 56.  4.100
   libavfilter 5. 14.100 /  5. 14.100
   libswscale  3.  1.101 /  3.  1.101
   libswresample   1.  1.100 /  1.  1.100
   libpostproc53.  3.100 / 53.  3.100
Input #0, avi, from 'I:\IMX_tests\IMX50_CARBON.avi':
   Metadata:
 encoder : Lavf55.0.100
 timecode: 00:00:00:00
   Duration: 00:00:20.00, start: 0.00, bitrate: 113291 kb/s
 Stream #0:0: Video: ffvhuff (FFVH / 0x48564646), yuv422p, 720x608,
96 kb/s, SAR 152:135 DAR 4:3, 25 fps, 25 tbr, 25 tbn, 25 tbc
 Stream #0:1: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo, s32 (24 bit), 2304 kb/s
File 'I:\IMX_tests\_FFMPEGNEW_PAL_IMX_24bit_2.mxf' already exists.
Overwrite ? [y/N] y
Output #0, mxf_d10, to 'I:\IMX_tests\_FFMPEGNEW_PAL_IMX_24bit_2.mxf':
   Metadata:
 timecode: 00:00:00:00
 encoder : Lavf56.31.100
 Stream #0:0: Video: mpeg2video, yuv422p, 720x608 [SAR 152:135 DAR
4:3], q=1-3, 5 kb/s, 25 fps, 25 tbn, 25 tbc
 Metadata:
   encoder : Lavc56.35.101 mpeg2video
 Stream #0:1: Audio: pcm_s24le, 48000 Hz, stereo, s32 (24 bit), 2304
kb/s
 Metadata:
   encoder : Lavc56.35.

[FFmpeg-user] How can I set in a D10 MXF (IMX50) file the flags for output color_range, -space, -transfer and primaries?

[Christoph Gerstbauer]

I found out that a IMX50 mxf file encoded with FFmbc and the IMX50 mxf file 
encoded with actual ffmpeg builds are different in these mxf metadata flags (by 
reading out via ffprobe - show_streams)

FFMBC IMX FILE stream 0:0:
color_range=tv
color_space=smpte170m
color_transfer=bt709
color_primaries=bt470bg


FFMPEG IMX FILE stream 0:0:
color_range=tv
color_space=unknown
color_transfer=unknown
color_primaries=unknown

I want to set these metadata flags to the same values, but with FFmpeg.
How can I produce the same output like ffmbc with these 4 metadata flags?
I didnt found a answer to that in the ffmpeg documentation :/

Best Regards
CHristoph


COMMAND LINE OUTPUT:

FFmbc encoding:

ffmbc -i I:\IMX_tests\IMX50_CARBON.avi -target imx50 -acodec pcm_s24le
I:\IMX_tests\_FFMBC_PAL_IMX_24bit_2.mxf -loglevel debug
FFmbc version 0.7-rc7
Copyright (c) 2008-2012 Baptiste Coudurier and the FFmpeg developers
Input #0, avi, from 'I:\IMX_tests\IMX50_CARBON.avi':
   Metadata:
 encoder: Lavf55.0.100
 ISMP: 00:00:00:00
   Duration: 00:00:20.00, start: 0.00, bitrate: 113291 kb/s
 Stream #0.0(und): Video: ffvhuff, yuv422p, 720x608p, PAR 152:135
DAR 4:3, 25.00 fps
 Stream #0.1(und): Audio: pcm_s24le, 48000 Hz, stereo, s32, 2304 kb/s
[buffer @ 002EFF60] w:720 h:608 pixfmt:yuv422p tb:1/100
sar:152/135 sws_param:
Warning, QMAT_SHIFT is larger than 21, overflows possible
 Last message repeated 1 times
Output #0, mxf_d10, to 'I:\IMX_tests\_FFMBC_PAL_IMX_24bit_2.mxf':
   Metadata:
 encoder: FFmbc 0.7
 Stream #0.0(und), 0, 1/25: Video: mpeg2video, yuv422p, 720x608i tff
[PAR 152:135 DAR 4:3], tbc 1/25, cbr, 5 kb/s, 25.00 fps
 Stream #0.1(und), 0, 1/48000: Audio: pcm_s24le, 48000 Hz, stereo,
s32, 2304 kb/s
Stream mapping:
   Stream #0.0 -> #0.0
   Stream #0.1 -> #0.1
Press [q] to stop, [?] for help
[mxf_d10 @ 003BF460] package type:1
[mxf_d10 @ 003BF460] package type:2
[mxf_d10 @ 003BF460] edit units count 0
frame=  341 fps= 25 q=3.0 size=  103570kB time=00:00:13.60
bitrate=62385.4kbits/s eta=00:00:06.41

-

FFMPEG encoding:

ffmpegnew -i I:\IMX_tests\IMX50_CARBON.avi -map 0:v -map 0:a -c:v
mpeg2video -r 25 -pix_fmt yuv422p -aspect 4:3 -minrate 5k -maxrate
5k -b:v 5k -intra -flags +ildct+low_del
ay -intra_vlc 1 -non_linear_quant 1 -ps 1 -qmin 1 -qmax 3 -top 1 -dc 10
-bufsize 200 -rc_init_occupancy 200 -rc_max_vbv_use 1
-rc_min_vbv_use 1 -c:a pcm_s24le -ar 48000 -f mxf_d10 I:\IMX_tests
\_FFMPEGNEW_PAL_IMX_24bit_2.mxf
ffmpeg version N-71727-g46778ab Copyright (c) 2000-2015 the FFmpeg
developers
   built with gcc 4.9.2 (GCC)
   configuration: --enable-gpl --enable-version3 --disable-w32threads
--enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
--enable-gnutls --enable-iconv --enable-libass --enable-libblu
ray --enable-libbs2b --enable-libcaca --enable-libdcadec
--enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc
--enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enab
le-libopencore-amrwb --enable-libopenjpeg --enable-libopus
--enable-librtmp --enable-libschroedinger --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-libtwolame
--enable-libvidstab --ena
ble-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis
--enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264
--enable-libx265 --enable-libxavs --enable-libxvid --enable-lzma --ena
ble-decklink --enable-zlib
   libavutil  54. 23.101 / 54. 23.101
   libavcodec 56. 35.101 / 56. 35.101
   libavformat56. 31.100 / 56. 31.100
   libavdevice56.  4.100 / 56.  4.100
   libavfilter 5. 14.100 /  5. 14.100
   libswscale  3.  1.101 /  3.  1.101
   libswresample   1.  1.100 /  1.  1.100
   libpostproc53.  3.100 / 53.  3.100
Input #0, avi, from 'I:\IMX_tests\IMX50_CARBON.avi':
   Metadata:
 encoder : Lavf55.0.100
 timecode: 00:00:00:00
   Duration: 00:00:20.00, start: 0.00, bitrate: 113291 kb/s
 Stream #0:0: Video: ffvhuff (FFVH / 0x48564646), yuv422p, 720x608,
96 kb/s, SAR 152:135 DAR 4:3, 25 fps, 25 tbr, 25 tbn, 25 tbc
 Stream #0:1: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz,
stereo, s32 (24 bit), 2304 kb/s
File 'I:\IMX_tests\_FFMPEGNEW_PAL_IMX_24bit_2.mxf' already exists.
Overwrite ? [y/N] y
Output #0, mxf_d10, to 'I:\IMX_tests\_FFMPEGNEW_PAL_IMX_24bit_2.mxf':
   Metadata:
 timecode: 00:00:00:00
 encoder : Lavf56.31.100
 Stream #0:0: Video: mpeg2video, yuv422p, 720x608 [SAR 152:135 DAR
4:3], q=1-3, 5 kb/s, 25 fps, 25 tbn, 25 tbc
 Metadata:
   encoder : Lavc56.35.101 mpeg2video
 Stream #0:1: Audio: pcm_s24le, 48000 Hz, stereo, s32 (24 bit), 2304
kb/s
 Metadata:
   encoder : Lavc56.35.101 pcm_s24le
Stream mapping:
   Stream #0:0 -> #0:0 (ffvhuff (native) -> mpeg2video (native))
   Stream #0:1 -> #0:1 (pcm_s24le (native) -

Re: [FFmpeg-user] Making conditional statements with FFMPEG

[Moritz Barsnick]

Hi Chris,

On Wed, Apr 29, 2015 at 18:50:55 +, Chris Zecco wrote:
> I was wondering if there was a way to create an IF-THEN statement
> using FFMPEG for files in a certain directory.

No, but with Unix scripting, this is a breeze. (For non-native
speakers: This is very simple with Unix scripting. That's what it was
made for.)

> For file sizes in excess of 200MB:
> Encode video bit rate at 1500k
> Encode audio bit rate at 128k
> 
> For file sizes 100-100MB:
> Encode video bit rate at 1100k
> Encode audio bit rate at 128k
> 
> For file sizes less than 100MB:
> Encode video bit rate at 555k
> Encode audio bit rate at 128k

I would inspect the input file. You would grab the size of the file
with something like "stat" first, then compare that against your
numbers. Then you put the command line parameters into a shell
variable, and use that for ffmpeg's command line:

---snip---
#!/bin/sh

for file in *; do
  SIZE=`stat -c %s "$file"`
  if [ $SIZE -gt 209715200 ]; then # > 200 MiB
COMPRESSION_V="-b:v 1500k"
  elif [ $SIZE -ge 104857600 ]; then # >= 100 MiB
COMPRESSION_V="-b:v 1100k"
  else # < 100 MiB
COMPRESSION_V="-b:v 555k"
  fi

  ffmpeg -i "$file" $COMPRESSION_V -b:a 128k "$file".new.extension
done
---snip---

Yeah, there are ways to preserve the file extension. I'm too lazy for
that right now. (It's probably a fixed extension for you anyway.)

You're not taking the run length of the videos into consideration. You
could query the original bitrate and apply e.g. half of that when
converting each file. Just a suggestion. ;-)

Cheers,
Moritz
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] Query_Exporting 5.1 + stereo Audio from prores 422 to mxf using ffmpeg

[Moritz Barsnick]

On Wed, Apr 29, 2015 at 22:04:15 +0530, Hardik Kanakia wrote:
> Hello All,
> 
> I want to export 5.1 channel + Stereo (8 tracks) with prores mov to MXF.
> 
> Since I am very new to this world, and as i read through the post, Audio
> only exports 1 Audio channel from the source (in my case its prores 422 mov
> with 5.1+Stereo)

Who is "Audio"? ("Audio only exports...")

> need your help badly as i am stuck in the middle of the road !!!

What did you try, and what was ffmpeg's output? In other words, please
show us the ffmpeg command line and the complete, uncut console output.

That helps us to see the format of your input file and what you are
trying to achieve.

> Audio output: pcm 24le

Please always be more precise.


Let me guess: You need to look at "-map 0". Your issue - I'm guessing,
because you're unclear! - might be that ffmpeg only takes the first of
each type of stream by default, and "-map 0" instructs it to take all.

Moritz
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] Use force_key_frames to obtain keyframes at exactly the same positions as in the input stream?

[Anatol]

No responses on that one?
It is very important issue.

On Mon, Apr 27, 2015 at 11:47 PM, Haris Zukanovic <
haris.zukanovi...@gmail.com> wrote:

> Hello,
> Can I use force_key_frames in some way to produce keyframes (IDR, not
> I-frames) at exactly the same PTS in output streams as they are found in
> the live input stream? Both input and output are h264 and live streaming.
>
> Something analogous to using 2 pass encoding for VOD and in the second
> pass keyframes are inserted exactly where they are recorded in the first
> pass... Is that something like that even theoretically doable for live
> streaming?
>
>
>
> thanx
>
> --
> --
> Haris Zukanovic
>
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] Making conditional statements with FFMPEG

[Chris Zecco]

I was wondering if there was a way to create an IF-THEN statement using FFMPEG 
for files in a certain directory. For example, I am using FFMPEG to crunch down 
file sizes of videos to 50MB or lower. I would ideally like to do the following:

For file sizes in excess of 200MB:
Encode video bit rate at 1500k
Encode audio bit rate at 128k

For file sizes 100-100MB:
Encode video bit rate at 1100k
Encode audio bit rate at 128k

For file sizes less than 100MB:
Encode video bit rate at 555k
Encode audio bit rate at 128k

Is this possible? And yes, I have run tests on individual file to ensure there 
is no real perceptible quality loss when dropping the bit rates that low.

Any help that you could give would be amazing, and greatly appreciated.

Thanks!

- Chris
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] ffprobe -show_frames values

[Joel Lopez]

Hi,

I'm trying to decipher what the values are in the info presented.  I need
to determine how far apart the keyframes are.  3 seconds is what I hear
they should be for adaptive bitrate switching.

Are these miliseconds and bytes?

pkt_pts=754100

pkt_pts_time=251.37

pkt_dts=754100

pkt_dts_time=251.37

best_effort_timestamp=754100

best_effort_timestamp_time=251.37

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183588

pkt_size=140
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] delay problems when using ffserver, ffmpeg for streaming of web cam and vlc for viewing.

[Roger Pack]

Do you get past duration too large when just saving to file?

On 4/29/15, Adrian Schwartz  wrote:
> hello ffmpeg user group,
> I'm having delay problem ( when using ffserver and ffmpeg)
> I'm getting the error "ffmpeg past duration too large" when using the
> ffmpeg for streaming.
>  VLC is configured to work in minimum delay and when streaming from //
> 192.168.1.1:8090/test1.asf it shows 2 sec delay, when the communication
> interface delay is aroung 15 msec.
>
>
> Is there some recommendation on how to lower the delay?
> Regards
> Adrian
>
> #
> #Streaming command:
> #
>
> ffmpeg -f dshow  -rtbufsize 1M   -i video="Logitech
> Webca"":audio="Microphone (2- Webcam C160)" -fflags nobuffer -tune
> zerolatency http://192.168.1.1:8090/feed1.ffm
>
> 
> ##ffserver config:
> 
>
>
> Port 8090
> BindAddress 0.0.0.0
> MaxClients 1000
> MaxBandwidth 3
> CustomLog /var/log/ffserver.log
> NoDaemon
>
> 
> File /tmp/feed1.ffm
> FileMaxSize 1M
> ACL allow 192.168.26.0 192.168.29.255
> ACL allow 127.0.0.1
> ACL allow 10.0.0.0 10.0.0.255
> 
>
>
>
> 
> Feed feed1.ffm
> Format asf
> VideoFrameRate 15
> VideoSize 352x240
> VideoBitRate 256
> VideoGopSize 30
> StartSendOnKey
> NoAudio
> 
>
> 
> Format status
> ACL allow localhost
> ACL allow 192.168.0.0 192.168.255.255
>
> 
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] Comparing keyframes of videos for proper alignment

[Joel Lopez]

Here are the last 3 keyframes of videos I'm trying to verify.  I'm not 100%
certain on what the values mean.


Video 1

[FRAME]

key_frame=0

pkt_pts=754000

pkt_pts_time=251.33

pkt_dts=754000

pkt_dts_time=251.33

best_effort_timestamp=754000

best_effort_timestamp_time=251.33

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183172

pkt_size=138

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754100

pkt_pts_time=251.37

pkt_dts=754100

pkt_dts_time=251.37

best_effort_timestamp=754100

best_effort_timestamp_time=251.37

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183588

pkt_size=140

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754200

pkt_pts_time=251.40

pkt_dts=N/A

pkt_dts_time=N/A

best_effort_timestamp=754200

best_effort_timestamp_time=251.40

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183450

pkt_size=138

[/FRAME]




Video 2


[FRAME]

key_frame=0

pkt_pts=754000

pkt_pts_time=251.33

pkt_dts=754000

pkt_dts_time=251.33

best_effort_timestamp=754000

best_effort_timestamp_time=251.33

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183172

pkt_size=138

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754100

pkt_pts_time=251.37

pkt_dts=754100

pkt_dts_time=251.37

best_effort_timestamp=754100

best_effort_timestamp_time=251.37

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183588

pkt_size=140

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754200

pkt_pts_time=251.40

pkt_dts=N/A

pkt_dts_time=N/A

best_effort_timestamp=754200

best_effort_timestamp_time=251.40

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183450

pkt_size=138

[/FRAME]




Video 3


[FRAME]

key_frame=0

pkt_pts=754000

pkt_pts_time=251.33

pkt_dts=754000

pkt_dts_time=251.33

best_effort_timestamp=754000

best_effort_timestamp_time=251.33

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183172

pkt_size=138

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754100

pkt_pts_time=251.37

pkt_dts=754100

pkt_dts_time=251.37

best_effort_timestamp=754100

best_effort_timestamp_time=251.37

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183588

pkt_size=140

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754200

pkt_pts_time=251.40

pkt_dts=N/A

pkt_dts_time=N/A

best_effort_timestamp=754200

best_effort_timestamp_time=251.40

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183450

pkt_size=138

[/FRAME]


Video 4


[FRAME]

key_frame=0

pkt_pts=754000

pkt_pts_time=251.33

pkt_dts=754000

pkt_dts_time=251.33

best_effort_timestamp=754000

best_effort_timestamp_time=251.33

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183172

pkt_size=138

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754100

pkt_pts_time=251.37

pkt_dts=754100

pkt_dts_time=251.37

best_effort_timestamp=754100

best_effort_timestamp_time=251.37

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183588

pkt_size=140

[/FRAME]

[FRAME]

key_frame=0

pkt_pts=754200

pkt_pts_time=251.40

pkt_dts=N/A

pkt_dts_time=N/A

best_effort_timestamp=754200

best_effort_timestamp_time=251.40

pkt_duration=100

pkt_duration_time=0.03

pkt_pos=69183450

pkt_size=138

[/FRAME]

On Tue, Apr 28, 2015 at 4:09 PM, Joel Lopez  wrote:

> Hi,
>
> I'm trying to verify that the keyframes of videos we are using are
> properly aligned.  I've found ffprobe to extract the some info but I'm not
> sure what data I should be comparing.
>
> This is so that we can do some adaptive bitrate switching using an SMIL
> file for RTMP, HLS and eventually MPEG-DASH.  I've been reading that they
> keyframes should be a constant 3 seconds apart.  Hopefully someone has some
> experience in this.
>
> [FRAME]
>
> media_type=video
>
> key_frame=1
>
> pkt_pts=0
>
> pkt_pts_time=0.00
>
> pkt_dts=0
>
> pkt_dts_time=0.00
>
> best_effort_timestamp=0
>
> best_effort_timestamp_time=0.00
>
> pkt_duration=100
>
> pkt_duration_time=0.03
>
> pkt_pos=48
>
> pkt_size=2950
>
> width=1280
>
> height=720
>
> pix_fmt=yuv420p
>
> sample_aspect_ratio=N/A
>
> pict_type=I
>
> coded_picture_number=0
>
> display_picture_number=0
>
> interlaced_frame=0
>
> top_field_first=0
>
> repeat_pict=0
>
> [/FRAME]
>
> [FRAME]
>
> media_type=video
>
> key_frame=0
>
> pkt_pts=100
>
> pkt_pts_time=0.03
>
> pkt_dts=100
>
> pkt_dts_time=0.03
>
> best_effort_timestamp=100
>
> best_effort_timestamp_time=0.03
>
> pkt_duration=100
>
> pkt_duration_time=0.03
>
> pkt_pos=3136
>
> pkt_size=140
>
> width=1280
>
> height=720
>
> pix_fmt=yuv420p
>
> sample_aspect_ratio=N/A
>
> pict_type=B
>
> coded_picture_number=2
>
> display_picture_number=0
>
> interlaced_frame=0
>
> top_field_first=0
>
> repeat_pict=0
>
> [/FRAME]
>
> [FRAME]
>
> media_type=video
>
> key_frame=0
>
> pkt_pts=200
>
> pkt_pts_time=0.07
>
> pkt_dts=200
>
> pkt_dts_time=0.07
>
> best_effort_timestamp=200
>
> best_effort_timestamp_time=0.07
>
> pkt_duration=100
>
> 

[FFmpeg-user] Query_Exporting 5.1 + stereo Audio from prores 422 to mxf using ffmpeg

[Hardik Kanakia]

Hello All,

I want to export 5.1 channel + Stereo (8 tracks) with prores mov to MXF.

Since I am very new to this world, and as i read through the post, Audio
only exports 1 Audio channel from the source (in my case its prores 422 mov
with 5.1+Stereo)

need your help badly as i am stuck in the middle of the road !!!

please help me with the command for getting an output to mxf

Audio output: pcm 24le

Thanks

Hardik
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] ffserver crashes from time to time

[En Figureo Canal]

As stated on a previous post, I'm feeding the server from my home pc like
this:

ffmpeg –rtbufsize 1500M –f show audio=”input” –vn –acodec libmp3lame –ab
> 128k –abufsize 256k http://ffserverip:8090/feedname.ffm




>
> File /root/radio56.ffm
>
> FileMaxSize 200K
>
> 
>
>
>> 
>
> Feed radio56.ffm
>
> Format rtp
>
>
>> AudioBitRate 128
>
> AudioChannels 2
>
> AudioSampleRate 44100
>
> AudioCodec libmp3lame
>
>
>> NoVideo
>
> 
>
>
>
I'm listening to the stream on iPhone running MDC Live Media Player.
Sometimes if I close the app while receiving the stream causes the server
to crash. This time I had to do it several time but, it crashed it.

It looks like something just kills the process some how and it just locks
me out. I'm not feeding a file is just the audio from my pc.




On Wed, Apr 29, 2015 at 4:37 AM, Moritz Barsnick  wrote:

> On Tue, Apr 28, 2015 at 20:50:16 -0400, En Figureo Canal wrote:
> > No clue on this problem?
>
> A crash in ffserver is a serious problem, assuming it isn't caused by
> other code (external library). It usually "shouldn't happen".
>
> But without you recompiling ffmpeg and running ffserver_g in gdb, there
> is next to nothing anyone can do to identify the problem.
>
> You could give us the exact file you're feeding to ffserver, the exact
> config, and tell us (once more) exactly what you did with the client to
> cause the crash. We _could_ try to reproduce, but it's probably not
> feasible. Debugging on your end would be much more efficient.
>
> Moritz
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] delay problems when using ffserver, ffmpeg for streaming of web cam and vlc for viewing.

[Adrian Schwartz]

hello ffmpeg user group,
I'm having delay problem ( when using ffserver and ffmpeg)
I'm getting the error "ffmpeg past duration too large" when using the
ffmpeg for streaming.
 VLC is configured to work in minimum delay and when streaming from //
192.168.1.1:8090/test1.asf it shows 2 sec delay, when the communication
interface delay is aroung 15 msec.


Is there some recommendation on how to lower the delay?
Regards
Adrian

#
#Streaming command:
#

ffmpeg -f dshow  -rtbufsize 1M   -i video="Logitech
Webca"":audio="Microphone (2- Webcam C160)" -fflags nobuffer -tune
zerolatency http://192.168.1.1:8090/feed1.ffm


##ffserver config:



Port 8090
BindAddress 0.0.0.0
MaxClients 1000
MaxBandwidth 3
CustomLog /var/log/ffserver.log
NoDaemon


File /tmp/feed1.ffm
FileMaxSize 1M
ACL allow 192.168.26.0 192.168.29.255
ACL allow 127.0.0.1
ACL allow 10.0.0.0 10.0.0.255





Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoGopSize 30
StartSendOnKey
NoAudio



Format status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255


___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] Fwd: Getting green Images when compressing with ffmpeg for Android 5.0

[Manikanta Vasu]

Hi, I am getting the image,but it is green colour image,

here is the logs


*WARNING: linker: /data/local/tmp/ffmpeg has text relocations. This is
wasting memory and prevents security hardening. Please fix.*

*ffmpeg version n2.4.2 Copyright (c) 2000-2014 the FFmpeg developers*

*  built on Oct  7 2014 15:08:46 with gcc 4.8 (GCC)*

*configuration: --target-os=linux
--cross-prefix=/home/sb/Source-Code/ffmpeg-android/toolchain-android/bin/arm-linux-androideabi-
--arch=arm --cpu=cortex-a8 --enable-runtime-cpudetect
--sysroot=/home/sb/Source-Code/ffmpeg-android/toolchain-android/sysroot
--enable-pic --enable-libx264 --enable-libass --enable-libfreetype
--enable-libfribidi --enable-fontconfig --enable-pthreads --disable-debug
--disable-ffserver --enable-version3 --enable-hardcoded-tables
--disable-ffplay --disable-ffprobe --enable-gpl --enable-yasm --disable-doc
--disable-shared --enable-static
--pkg-config=/home/sb/Source-Code/ffmpeg-android/ffmpeg-pkg-config
--prefix=/home/sb/Source-Code/ffmpeg-android/build/armeabi-v7a-neon
--extra-cflags='-I/home/sb/Source-Code/ffmpeg-android/toolchain-android/include
-U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -fno-strict-overflow
-fstack-protector-all -mfpu=neon'
--extra-ldflags='-L/home/sb/Source-Code/ffmpeg-android/toolchain-android/lib
-Wl,-z,relro -Wl,-z,now -pie' --extra-libs='-lpng -lexpat -lm'
--extra-cxxflags=*

*  libavutil  54.  7.100 / 54.  7.100*

*  libavcodec 56.  1.100 / 56.  1.100*

*  libavformat56.  4.101 / 56.  4.101*

*  libavdevice56.  0.100 / 56.  0.100*

*  libavfilter 5.  1.100 /  5.  1.100*

*  libswscale  3.  0.100 /  3.  0.100*

*  libswresample   1.  1.100 /  1.  1.100*

*  libpostproc53.  0.100 / 53.  0.100*

*Input #0, rawvideo, from 'pipe:':*

*  Duration: N/A, start: 0.00, bitrate: 294912 kb/s*

*Stream #0:0: Video: rawvideo (RGB[0] / 0x424752), rgb0, 720x1280,
294912 kb/s, 10 tbr, 10 tbn, 10 tbc*

*[Parsed_scale_0 @ 0xb7b11d70] w:281.25 h:500.0 flags:'0x4' interl:0*

*[graph 0 input from stream 0:0 @ 0xb7b125d0] w:720 h:1280 pixfmt:rgb0
tb:1/10 fr:10/1 sar:0/1 sws_param:flags=2*

*[graph 0 input from stream 0:0 @ 0xb7b125d0] TB:0.10
FRAME_RATE:10.00 SAMPLE_RATE:nan*

*[Parsed_scale_0 @ 0xb7b11d70] w:720 h:1280 fmt:rgb0 sar:0/1 -> w:281 h:500
fmt:rgb24 sar:0/1 flags:0x4*

*Output #0, image2pipe, to '/sdcard/test1.png':*

*  Metadata:*

*encoder : Lavf56.4.101*

*Stream #0:0: Video: png, rgb24, 281x500, q=2-31, 200 kb/s, 10 fps, 10
tbn, 10 tbc*

*Metadata:*

*  encoder : Lavc56.1.100 png*

*   Stream mapping:*

*  Stream #0:0 -> #0:0 (rawvideo (native) -> png (native))*

*  No more output streams to write to, finishing.*

*  frame=1 fps=0.0 q=0.0 Lsize=  64kB time=00:00:00.10
bitrate=5250.2kbits/s*


*   video:64kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 0.00%*

*  Input file #0 (pipe:):*

*  Input stream #0:0 (video): 1 packets read (3686400 bytes); 1 frames
decoded; *

*  Total: 1 packets (3686400 bytes) demuxed*

*  Output file #0 (/sdcard/test1.png):*
*  Output stream #0:0 (video): 1 frames encoded; 1 packets muxed (65628
bytes); *
*  Total: 1 packets (65628 bytes) muxed*




On Wed, Apr 29, 2015 at 2:17 PM, Moritz Barsnick  wrote:

> On Wed, Apr 29, 2015 at 13:45:51 +0530, Manikanta Vasu wrote:
> > ffmpeg -loglevel panic -f rawvideo -pix_fmt bgr0 -r 10 -i - -vf scale=1
> -f
> > image2 -vcodec png -vframes 1 /sdcard/screen.png
>
> You gave "-loglevel panic", but don't show us the log. But that's what
> the log is good for. Please attach!
>
> Moritz
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] streaming only audio from YouTube

[DopeLabs]

installing youtube-dl is a snap on most *nix

$ sudo apt-get install youtube-dl

or

$ pip install youtube-dl

once installed if you can use  back ticks in the ffmpeg command as such:

$ ffmpeg -re -i "`youtube-dl -f bestaudio -g 
https://www.youtube.com/watch?v=holhZM1OpTc`"; -c:a mp3 -b:a 256k audio_only.mp3

you will need ffmpeg built with openssl enabled to access https://



> On Apr 29, 2015, at 1:45 AM, Moritz Barsnick  wrote:
> 
> Hi Kath,
> 
> On Wed, Apr 29, 2015 at 05:02:03 +0100, kath g wrote:
>> Can we stream and save audio from YouTube using ffmpeg. The audio is about 
>> 10 hrs long. I use manjaro with ffmpeg. 
> 
> Is it really a stream or a static video?
> 
> Static videos - and often even audio only - can be downloaded with the
> tool youtube-dl: http://yt-dl.org/
> With its "-F" option, you can figure out the formats (some are
> audio-only). With "-f " or "-f bestaudio", you can tell
> youtube-dl to grab you that audio. Or you can query the URL by adding
> "-g", and pass the given URL to ffmpeg.
> 
> ffmpeg can also be built with libquvi support, where ffmpeg would
> directly understand a YouTube URL. I don't know whether that works
> anymore though.
> (If you don't build ffmpeg youself, you can figure out libquvi support
> by running:
> $ ffmpeg -formats | grep quvi
> and checking for this line:
> D  libquvi libquvi demuxer
> )
> 
> Moritz
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user

___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] Fwd: Getting green Images when compressing with ffmpeg for Android 5.0

[Moritz Barsnick]

On Wed, Apr 29, 2015 at 13:45:51 +0530, Manikanta Vasu wrote:
> ffmpeg -loglevel panic -f rawvideo -pix_fmt bgr0 -r 10 -i - -vf scale=1 -f
> image2 -vcodec png -vframes 1 /sdcard/screen.png

You gave "-loglevel panic", but don't show us the log. But that's what
the log is good for. Please attach!

Moritz
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] streaming only audio from YouTube

[Moritz Barsnick]

Hi Kath,

On Wed, Apr 29, 2015 at 05:02:03 +0100, kath g wrote:
> Can we stream and save audio from YouTube using ffmpeg. The audio is about 10 
> hrs long. I use manjaro with ffmpeg. 

Is it really a stream or a static video?

Static videos - and often even audio only - can be downloaded with the
tool youtube-dl: http://yt-dl.org/
With its "-F" option, you can figure out the formats (some are
audio-only). With "-f " or "-f bestaudio", you can tell
youtube-dl to grab you that audio. Or you can query the URL by adding
"-g", and pass the given URL to ffmpeg.

ffmpeg can also be built with libquvi support, where ffmpeg would
directly understand a YouTube URL. I don't know whether that works
anymore though.
(If you don't build ffmpeg youself, you can figure out libquvi support
by running:
$ ffmpeg -formats | grep quvi
and checking for this line:
 D  libquvi libquvi demuxer
)

Moritz
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] ffmpeg not working on server

[Reindl Harald]

Am 29.04.2015 um 10:35 schrieb Waleed Afridi:

*Message By Hosting Suport Team.*

We've uploaded the file '20051210-w50s.flv' to your images/videos folder,
and run the following command from shell directly:

/usr/bin/ffmpeg -i 20051210-w50s.flv -vcodec libx264 -ar 22050 -crf 28
20051210-w50s.mp4 -r 1 -y -vcodec mjpeg -f mjpeg 20051210-w50s.jpg

The following FFMPEG output results:

=
FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)


ah come on you can't seriously expect support for such and outdated 
build - recent is 2.6.2


[harry@rh:~]$ ffmpeg
ffmpeg version 2.6.2 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 4.9.2 (GCC) 20150212 (Red Hat 4.9.2-6)



signature.asc
Description: OpenPGP digital signature
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] ffserver crashes from time to time

[Moritz Barsnick]

On Tue, Apr 28, 2015 at 20:50:16 -0400, En Figureo Canal wrote:
> No clue on this problem?

A crash in ffserver is a serious problem, assuming it isn't caused by
other code (external library). It usually "shouldn't happen".

But without you recompiling ffmpeg and running ffserver_g in gdb, there
is next to nothing anyone can do to identify the problem.

You could give us the exact file you're feeding to ffserver, the exact
config, and tell us (once more) exactly what you did with the client to
cause the crash. We _could_ try to reproduce, but it's probably not
feasible. Debugging on your end would be much more efficient.

Moritz
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] ffmpeg not working on server

[Waleed Afridi]

*Message By Hosting Suport Team.*

We've uploaded the file '20051210-w50s.flv' to your images/videos folder,
and run the following command from shell directly:

/usr/bin/ffmpeg -i 20051210-w50s.flv -vcodec libx264 -ar 22050 -crf 28
20051210-w50s.mp4 -r 1 -y -vcodec mjpeg -f mjpeg 20051210-w50s.jpg

The following FFMPEG output results:

=
FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers
built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6)
configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64
--mandir=/usr/share/man --incdir=/usr/include --disable-avisynth
--extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions
-fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC'
--enable-avfilter --enable-avfilter-lavf --enable-libdc1394
--enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin
--enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger
--enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl
--enable-nonfree --enable-postproc --enable-pthreads --enable-shared
--enable-swscale --enable-vdpau --enable-version3 --enable-x11grab
libavutil 50.15. 1 / 50.15. 1
libavcodec 52.72. 2 / 52.72. 2
libavformat 52.64. 2 / 52.64. 2
libavdevice 52. 2. 0 / 52. 2. 0
libavfilter 1.19. 0 / 1.19. 0
libswscale 0.11. 0 / 0.11. 0
libpostproc 51. 2. 0 / 51. 2. 0
[flv @ 0x18b6670]Estimating duration from bitrate, this may be inaccurate

Seems stream 0 codec frame rate differs from container frame rate: 1000.00
(1000/1) -> 25.00 (25/1)
Input #0, flv, from '20051210-w50s.flv':
Metadata:
duration : 17
width : 360
height : 288
videodatarate : 260
framerate : 25
videocodecid : 4
audiodatarate : 40
audiodelay : 0
audiocodecid : 2
canSeekToEnd : 1
creationdate : Fri Feb 03 11:52:46 2006

Duration: 00:00:16.92, start: 0.00, bitrate: 306 kb/s
Stream #0.0: Video: vp6f, yuv420p, 360x288, 266 kb/s, 25 tbr, 1k tbn, 1k
tbc
Stream #0.1: Audio: mp3, 22050 Hz, 2 channels, s16, 40 kb/s
File '20051210-w50s.mp4' already exists. Overwrite ? [y/N] y
[libx264 @ 0x18ba170]broken ffmpeg default settings detected
[libx264 @ 0x18ba170]use an encoding preset (e.g. -vpre medium)
[libx264 @ 0x18ba170]preset usage: -vpre  -vpre 
[libx264 @ 0x18ba170]speed presets are listed in x264 --help
[libx264 @ 0x18ba170]profile is optional; x264 defaults to high
Output #0, mp4, to '20051210-w50s.mp4':
Stream #0.0: Video: libx264, yuv420p, 360x288, q=2-31, 200 kb/s, 90k tbn,
25 tbc
Stream #0.1: Audio: libfaac, 22050 Hz, 2 channels, s16, 64 kb/s
Output #1, mjpeg, to '20051210-w50s.jpg':
Stream #1.0: Video: mjpeg, yuvj420p, 360x288, q=2-31, 200 kb/s, 90k tbn, 1
tbc
Stream mapping:
Stream #0.0 -> #0.0
Stream #0.1 -> #0.1
Stream #0.0 -> #1.0
Error while opening encoder for output stream #0.0 - maybe incorrect
parameters such as bit_rate, rate, width or height
===

On Wed, Apr 29, 2015 at 12:36 PM, Jan Ehrhardt  wrote:

> Carl Eugen Hoyos in gmane.comp.video.ffmpeg.user (Tue, 28 Apr 2015
> 19:00:10 + (UTC)):
> >Waleed Afridi  gmail.com> writes:
> >
> >> ffmpeg-php version 0.6.0-svn
> >
> >This is an unrelated project for which no support
> >was ever offered here.
>
> He does not use it in his PHP-script. He is calling ffmpeg there with
> exec(). @Waleed: try running this command on your server from a console
> (not using PHP):
>
> /usr/bin/ffmpeg -i input_file -vcodec libx264 -ar 22050 -crf 28 \
> output.mp4 -r 1 -y -vcodec mjpeg -f mjpeg output.jpg
>
> Jan
>
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>



-- 

*Waleed Afridi*
*Web Engineer*

*Alliance IT Solutions*
*www.waleedafridi.com *
*Mob: 0333-9010689*
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

[FFmpeg-user] Fwd: Getting green Images when compressing with ffmpeg for Android 5.0

[Manikanta Vasu]

I am trying to fetch screenshots of android devices using adb screencap command
along with the ffmpeg command. The command works well for devices other
than Android 5.0

ffmpeg -loglevel panic -f rawvideo -pix_fmt bgr0 -r 10 -i - -vf scale=1 -f
image2 -vcodec png -vframes 1 /sdcard/screen.png

The issue is for Android 5.0 devices after every rendering of image ,a
green image is coming.

Regards
Manikanta Vasu
___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user

Re: [FFmpeg-user] ffmpeg not working on server

[Jan Ehrhardt]

Carl Eugen Hoyos in gmane.comp.video.ffmpeg.user (Tue, 28 Apr 2015
19:00:10 + (UTC)):
>Waleed Afridi  gmail.com> writes:
>
>> ffmpeg-php version 0.6.0-svn
>
>This is an unrelated project for which no support 
>was ever offered here.

He does not use it in his PHP-script. He is calling ffmpeg there with
exec(). @Waleed: try running this command on your server from a console
(not using PHP):

/usr/bin/ffmpeg -i input_file -vcodec libx264 -ar 22050 -crf 28 \
output.mp4 -r 1 -y -vcodec mjpeg -f mjpeg output.jpg

Jan

___
ffmpeg-user mailing list
ffmpeg-user@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-user