Re: [FFmpeg-user] Generate MPEG Transport stream from audio files only

[Carl Zwanzig]

On 3/8/2018 8:17 AM, Moritz Barsnick wrote:

As Carl Eugen tried to point out: What is missing is that you provide
your actual (non-working) command line and its complete, uncut console
output.


One thing missing is some directed speculation even before the cmd line is 
sent; it's a valuable part of troubleshooting and helps those asking to learn.



To the OP-
You might try providing dummy video to go with the audio (I know some 
software doesn't deal well when there is no video stream).


I think all you need to add something like

  -i color=s=$outFormat:c=black:r=$videoRate

but I haven't tried it.

z!
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Re: [FFmpeg-user] Generate MPEG Transport stream from audio files only

[Pierre Pasquet]

All right, here’s a detailed presentation of what’s happening : 

First of all, everything is on local. I am using the default Mac OS apache 
server

- I launch a process that listens on port  and segments incoming media for 
deployment using HTTP Live Streaming with : 
mediastreamsegmenter -f /Library/WebServer/Documents/live 127.0.0.1:

In theory, when receiving an input file, the segmenter should generate a 
playlist .m3u8 and create subsequent .aac files (because we are dealing only 
with audio)

- Then, I **try** to wrap the audio in a MPEG-2 transport stream and send it 
over UDP to port  with the following command (on a different terminal tab) 
: 
ffmpeg -re -i test.aac -vcodec copy -vbsf h264_mp4toannexb -acodec copy -f 
mpegts udp://127.0.0.1:

At this point, here’s what happens : 

- on the segmenter tab : 

Mar  9 2018 00:59:49.627: audio pid set at 100
Mar  9 2018 00:59:49.928: audio pid change to 100
Mar  9 2018 00:59:50.396: audio pid change to 100
Mar  9 2018 00:59:50.718: audio pid change to 100

And so on … (same prompt)

- on the ffmpeg tab : 

[aac @ 0x7fd242009400] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from 'test.aac':
  Duration: 00:01:08.37, bitrate: 139 kb/s
Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 139 kb/s
[mpegts @ 0x7fd24200a000] frame size not set
Output #0, mpegts, to 'udp://127.0.0.1:':
  Metadata:
encoder : Lavf57.83.100
Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 139 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=  75kB time=00:00:04.06 bitrate= 151.0kbits/s speed=   1x 


At this point, there should be a .m3u8 playlist file under 
/Library/WebServer/Documents/live and appropriate .aac files. But it’s not the 
case. 
Eventually, the ffmpeg tab will return this : 

video:0kB audio:1165kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 12.020084%

And the segmenter tab will return : 

Mar  9 2018 01:02:00.582: Warning: no data received for 0.5 seconds.
Mar  9 2018 01:02:01.083: Warning: no data received for 1.0 seconds.
Mar  9 2018 01:02:01.583: Warning: no data received for 1.5 seconds.
Mar  9 2018 01:02:02.084: Warning: no data received for 2.0 seconds.
Mar  9 2018 01:02:02.589: Warning: no data received for 2.5 seconds.
Mar  9 2018 01:02:03.090: Warning: no data received for 3.0 seconds.
Mar  9 2018 01:02:03.591: Warning: no data received for 3.5 seconds.
Mar  9 2018 01:02:04.095: Warning: no data received for 4.0 seconds.


But still no files were created. 

I know that the ffmpeg command is not write because it applies to video files, 
but I can’t come up with an equivalent command for audio only! 

Thanks for your help, 

Pierre

> On 8 Mar 2018, at 16:17, Moritz Barsnick  wrote:
> 
> On Thu, Mar 08, 2018 at 14:25:54 +, Pierre Pasquet wrote:
>> Not sure to identify what is missing. This command works for a .m4v file and 
>> doesn’t for a .aac file located in the same directory.
> 
> As Carl Eugen tried to point out: What is missing is that you provide
> your actual (non-working) command line and its complete, uncut console
> output. These provide a lot of valuable information, and will help to
> analyze your issue.
> 
> Cheers,
> Moritz
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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Mark Burton]

On 8 Mar 2018, at 22:29, Carl Eugen Hoyos  wrote:
> One possibility is that you run "git diff libavformat/movenc.h" to verify you
> have only changed this one value (this assumes you have originally done
> "git clone" and not "curl..."), then run "git commit libavformat/movenc.h" to
> start the commit line editor, type "i" (I believe this will be necessary to 
> get
> into insert mode) and "lavf/movenc: Change MOV_TIMESCALE
> to 600" followed by two (!) carriage returns and an explanation why this is
> a good idea (because seeking works better in some other software), then
> exit the editor (it's "ESC" ":wq" here) and get back to the shell. There, run
> "git format-patch HEAD^" to get a file 0001-lavf-movenc... that you send
> as attachment to the ffmpeg-devel mailing list and "git reset HEAD^" to
> clean your local checkout.

Fantastic. I’ve just submitted the patch to the developer list. Will do my best 
to answer the questions that come my way!

Thanks again Carl.
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[FFmpeg-user] Generating multiple HLS renditions and a master playlist results in unpredictable behavior.

[Dennis Mungai]

Hello, and good morning.

I have ran into a problem with the HLS muxer, as shown below.

I want to generate three HLS renditions from a single input file, and
a master playlist pointing to the renditions.

Here's the command line used:

ffmpeg -loglevel debug -threads 4 -filter_complex_threads 4 -vsync 1
-i '/home/lin/Desktop/src/sowdtow.webm' -filter_complex \
"[v:0]split=3[s0][s1][s2]; \
 [s0]yadif[v0]; \
 [s1]yadif[v1]; \
 
[s2]scale=w=1280:h=720:force_original_aspect_ratio=decrease:flags=lanczos,yadif[v2]"
\
-map "[v0]" -c:v libx264 -pix_fmt yuv420p -preset medium -profile:v
main -level 4.1 -b:v 5250k -maxrate:v 6375k -bufsize:v 12600k \
-map "[v1]" -c:v libx264 -pix_fmt yuv420p -preset medium -profile:v
main -level 4.1 -b:v 4200k -maxrate:v 4494k -bufsize:v 6300k \
-map "[v2]" -c:v libx264 -pix_fmt yuv420p -preset medium -profile:v
main -level 4.1 -b:v 2100k -maxrate:v 2247k -bufsize:v 3150k \
-map a:0 -map a:0 -map a:0 -c:a aac -ar 48000 -ab 256k -af
"aresample=async=1:min_hard_comp=0.10:first_pts=0" \
-f hls \
-var_stream_map "v:0,a:0 v:1,a:1 v:2,a:2" -hls_wrap 24 -hls_time 4
-master_pl_name master.m3u8 \
"/home/lin/Desktop/dest/vs%v/manifest.m3u8"


File information:

ffprobe -i sowdtow.webm
ffprobe version N-90268-g9fe61b6 Copyright (c) 2007-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609
  configuration: --prefix=/home/lin/ffmpeg_build
--pkg-config-flags=--static
--extra-cflags=-I/home/lin/ffmpeg_build/include
--extra-ldflags=-L/home/lin/ffmpeg_build/lib --bindir=/home/lin/bin
--enable-cuda-sdk --enable-cuvid --enable-libnpp
--extra-cflags=-I../nv_sdk --extra-ldflags=-L../nv_sdk
--extra-cflags=-I/usr/local/cuda/include/
--extra-ldflags=-L/usr/local/cuda/lib64/ --nvccflags='-gencode
arch=compute_61,code=sm_61 -O2' --enable-gpl --enable-libass
--enable-libfdk-aac --enable-libx264 --extra-libs=-lpthread
--enable-libx265 --enable-nvenc --enable-nonfree
  libavutil  56.  8.100 / 56.  8.100
  libavcodec 58. 14.100 / 58. 14.100
  libavformat58. 10.100 / 58. 10.100
  libavdevice58.  2.100 / 58.  2.100
  libavfilter 7. 12.100 /  7. 12.100
  libswscale  5.  0.102 /  5.  0.102
  libswresample   3.  0.101 /  3.  0.101
  libpostproc55.  0.100 / 55.  0.100
Input #0, matroska,webm, from 'sowdtow.webm':
  Metadata:
ENCODER : Lavf58.10.100
  Duration: 00:03:02.18, start: 0.00, bitrate: 16906 kb/s
Stream #0:0(eng): Video: vp9 (Profile 0), yuv420p(tv,
bt709/unknown/unknown), 3840x2160, SAR 1:1 DAR 16:9, 29.97 fps, 29.97
tbr, 1k tbn, 1k tbc (default)
Metadata:
  DURATION: 00:03:02.11500
Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp (default)
Metadata:
  HANDLER_NAME: SoundHandler
  DURATION: 00:03:02.18400




The command above is supposed to achieve the following goals:

1. Split the primary video stream into three, such that separate
outputs governed by the selected encoder (libx264) and separate
filters can be applied, including a re-scale for the third stream.

2. Encode the audio stream to three different renditions, and

3. Using the HLS muxer's -var_stream_map , generate appropriate
groupings for the renditions above and a master playlist pointing to
the renditions.

Here's where all goes to hell:

Only the first variable stream map pair (output under
~/Desktop/dest/vs1) has the correct HLS time set in the segments, as
shown below:


less vs0/manifest.m3u8

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:4
#EXT-X-MEDIA-SEQUENCE:41
#EXTINF:3.937267,
manifest17.ts
#EXTINF:4.004000,
manifest18.ts
#EXTINF:4.070733,
manifest19.ts
#EXTINF:3.903900,
manifest20.ts
#EXTINF:2.202200,
manifest21.ts
#EXT-X-ENDLIST


And now, on to the other broken manifests rendered unplayable:

less vs1/manifest.m3u8

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:0
#EXT-X-MEDIA-SEQUENCE:41
#EXTINF:0.00,
manifest17.ts
#EXTINF:0.00,
manifest18.ts
#EXTINF:0.00,
manifest19.ts
#EXTINF:0.00,
manifest20.ts
#EXTINF:0.00,
manifest21.ts
#EXT-X-ENDLIST


And vs2/manifest.m3u8:

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:0
#EXT-X-MEDIA-SEQUENCE:41
#EXTINF:0.00,
manifest17.ts
#EXTINF:0.00,
manifest18.ts
#EXTINF:0.00,
manifest19.ts
#EXTINF:0.00,
manifest20.ts
#EXTINF:0.00,
manifest21.ts
#EXT-X-ENDLIST


Note the anomaly in the #EXTINF: period field on vs1 and 2's manifest,
zeroed out, and compare it to the functioning vs0's manifest.

The second anomaly arises in the main manifest:

less master.m3u8

#EXTM3U
#EXT-X-VERSION:3
#EXT-X-STREAM-INF:BANDWIDTH=2591600,RESOLUTION=3840x2160,CODECS="avc1.4d4029,mp4a.40.2"
vs0/manifest.m3u8

#EXT-X-STREAM-INF:BANDWIDTH=2591600,RESOLUTION=3840x2160,CODECS="avc1.4d4029,mp4a.40.2"
vs1/manifest.m3u8

#EXT-X-STREAM-INF:BANDWIDTH=2591600,RESOLUTION=1280x720,CODECS="avc1.4d4029,mp4a.40.2"
vs2/manifest.m3u8


Note that according to the master manifest, all three renditions have
the same BANDWIDTH value, despite having 

Re: [FFmpeg-user] How to extract DVD subtitle from this VOB file?

[Carl Eugen Hoyos]

2018-03-08 21:49 GMT+01:00, Nicolas George :
> Farhad Mohammadi Majd (2018-03-08):

>> I need to those dvd_subtitle(s). How to extract them?
>
> FFmpeg can read them, but it cannot put them into
> anything useful at this time.

That surprises me: I would have expected dvdsub conversion
to work fine. I believe some issues were fixed.

Carl Eugen
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Re: [FFmpeg-user] ffmpeg compile with fdk-aac error

[Carl Eugen Hoyos]

2018-03-08 8:50 GMT+01:00, 黄龙飞 :

> i use ffmpeg3.4.1 compile with fdk-aac 0.1.15 occur error.

> /Users/huanglongfei/Library/Android/sdk/ndk-bundle/toolchains/arm-linux-androideabi-4.9/prebuilt/darwin-x86_64/bin/../lib/gcc/arm-linux-androideabi/4.9.x/../../../../arm-linux-androideabi/bin/ld:
>  error: fdk-aac: no archive symbol table (run ranlib)

Try running ranlib ;-)

Carl Eugen
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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Carl Eugen Hoyos]

2018-03-08 22:20 GMT+01:00, Mark Burton :
> On 8 Mar 2018, at 17:21, Carl Eugen Hoyos  wrote:
>> You need a compiler, make and git. Iirc, all three are installed if you
>> type something like "gcc -v" on the command line (but I assume they
>> are now installed by brew).
>
> Thank you very much! That filled in a few blanks for me and I was able to
> get it all working. So I built from the current git and tested the ffmpeg
> build. The timescale issue was still the same, with inaccurate seeking.
>
> Then in movenc.h I changed MOV_TIMESCALE from 1000 to 24000 and built ffmpeg
> again. This time the output .mov worked, with no seeking issues!
>
> Finally in movenc.h I changed MOV_TIMESCALE to 600 and built ffmpeg again.
> Again, the output .mov worked, with no seeking issues!
>
> I imagine some exhaustive testing is needed for other workflows, and I’m
> obviously not aware yet of why 1000 was chosen for ffmpeg (??), but based on
> the Quicktime specification documents and what the default timescale is set
> to there, it would seem that a switch from 1000 to 600 could be a very
> positive change.

> Please could you advise what I would need to do from here to move things
> forward.

One possibility is that you run "git diff libavformat/movenc.h" to verify you
have only changed this one value (this assumes you have originally done
"git clone" and not "curl..."), then run "git commit libavformat/movenc.h" to
start the commit line editor, type "i" (I believe this will be necessary to get
into insert mode) and "lavf/movenc: Change MOV_TIMESCALE
to 600" followed by two (!) carriage returns and an explanation why this is
a good idea (because seeking works better in some other software), then
exit the editor (it's "ESC" ":wq" here) and get back to the shell. There, run
"git format-patch HEAD^" to get a file 0001-lavf-movenc... that you send
as attachment to the ffmpeg-devel mailing list and "git reset HEAD^" to
clean your local checkout.

Carl Eugen
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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Mark Burton]

On 8 Mar 2018, at 17:21, Carl Eugen Hoyos  wrote:
> You need a compiler, make and git. Iirc, all three are installed if you
> type something like "gcc -v" on the command line (but I assume they
> are now installed by brew).

Thank you very much! That filled in a few blanks for me and I was able to get 
it all working. So I built from the current git and tested the ffmpeg build. 
The timescale issue was still the same, with inaccurate seeking.

Then in movenc.h I changed MOV_TIMESCALE from 1000 to 24000 and built ffmpeg 
again. This time the output .mov worked, with no seeking issues!

Finally in movenc.h I changed MOV_TIMESCALE to 600 and built ffmpeg again. 
Again, the output .mov worked, with no seeking issues!

I imagine some exhaustive testing is needed for other workflows, and I’m 
obviously not aware yet of why 1000 was chosen for ffmpeg (??), but based on 
the Quicktime specification documents and what the default timescale is set to 
there, it would seem that a switch from 1000 to 600 could be a very positive 
change.

Please could you advise what I would need to do from here to move things 
forward.

Many thanks again for your time and help getting ffmpeg built.

Regards
Mark
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Re: [FFmpeg-user] How to extract DVD subtitle from this VOB file?

[Nicolas George]

Farhad Mohammadi Majd (2018-03-08):
> $ ffmpeg -i VTS_01_1.VOB

Do not do that. Never access the VOB files in a DVD-video structure
directly, they do not contain only the title video data but also menus
and garbage. You can use tools/dvd2concat to extract only the parts of
the VOB file that matters. Or MPlayer to dump them into a file.

> I need to those dvd_subtitle(s). How to extract them?

FFmpeg can read them, but it cannot put them into anything useful at
this time. MEncoder can extract them.

Regards,

-- 
  Nicolas George


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Re: [FFmpeg-user] probleme lors de la compilation de ffmpeg

[Lou Logan]

On Wed, Mar 7, 2018, at 1:44 PM, Cley Faye wrote:
>
> The direct cause of this is that you ask configure to enable librtmp with
> --enable-librtmp but the configure script can't find the development files
> for librtmp. If you really want to support this, on a debian system
> installing "librtmp-dev" should be enough if you're not doing anything
> specific like cross-compiling or using custom prefix.

Alternatively, remove "--enable-librtmp". FFmpeg has native rtmp support, so 
enabling librtmp may not be needed (but I'm unsure what additional features or 
whatever librtmp provides over the native implementation).
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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Carl Eugen Hoyos]

2018-03-08 18:21 GMT+01:00, Carl Eugen Hoyos :
> 2018-03-08 17:58 GMT+01:00, Mark Burton :
>> Thanks. I have managed to compile a usable ffmpeg using brew (I work
>> exclusively on macOS), but I have not figured out how I would be able to
>> edit the libavformat/movenc.h file before compiling using this method. It
>> seems I would need to go down the route, so any help to get that far
>> would
>> be so welcome.
>
> You need a compiler, make and git. Iirc, all three are installed if you
> type something like "gcc -v" on the command line (but I assume they
> are now installed by brew).
>
> $ git clone https://git.ffmpeg.org/ffmpeg.git ffmpeg

An faster equivalent that does not involve git is:
$ curl -O http://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2
$ tar xf ffmpeg-snapshot-git.tar.bz2

> $ cd ffmpeg
> $ ./configure && make ffmpeg
> This could for example fail if you don't have a new enough yasm or nasm
> binary, in that case please try "./configure --disable-yasm", feel free to
> add "--enable-gpl". If you know hat your cpu has - for example - four
> cores, you can use "make -j4 ffmpeg" to speed up compilation.
>
> This creates a current ffmpeg binary, please test (from the build
> directory) if it allows to reproduce your issue.
> Then edit libavformat/movenc.h and run "make ffmpeg" again to
> get a binary that allows testing if you found the actual issue.
>
> Carl Eugen
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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Carl Eugen Hoyos]

2018-03-08 17:58 GMT+01:00, Mark Burton :
> Thanks. I have managed to compile a usable ffmpeg using brew (I work
> exclusively on macOS), but I have not figured out how I would be able to
> edit the libavformat/movenc.h file before compiling using this method. It
> seems I would need to go down the route, so any help to get that far would
> be so welcome.

You need a compiler, make and git. Iirc, all three are installed if you
type something like "gcc -v" on the command line (but I assume they
are now installed by brew).

$ git clone https://git.ffmpeg.org/ffmpeg.git ffmpeg
$ cd ffmpeg
$ ./configure && make ffmpeg
This could for example fail if you don't have a new enough yasm or nasm
binary, in that case please try "./configure --disable-yasm", feel free to
add "--enable-gpl". If you know hat your cpu has - for example - four
cores, you can use "make -j4 ffmpeg" to speed up compilation.

This creates a current ffmpeg binary, please test (from the build
directory) if it allows to reproduce your issue.
Then edit libavformat/movenc.h and run "make ffmpeg" again to
get a binary that allows testing if you found the actual issue.

Carl Eugen
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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Mark Burton]

On 8 Mar 2018, at 16:38, Carl Eugen Hoyos  wrote:
> 2018-03-08 15:57 GMT+01:00, Mark Burton :
>> 
>> Understood. A better option would no doubt be the ability to specify this
>> value in the same way we can specify the video_track_timescale value.
> If it would not fix your issue why would it be a good option?

Fair enough. The current ffmpeg default is 1000. This is not divisible by 24, 
30 or 60fps. The default in the Quicktime specification is 600. This is cleanly 
divisible by 24, 25, 30, 50, 60, so is a very good value for the majority of 
professional fixed frame rates. I accept these do not represent all fixed frame 
rates, but it is evidently a better value than 1000.

https://developer.apple.com/library/content/documentation/QuickTime/RM/Fundamentals/QTOverview/QTOverview_Document/QuickTimeOverview.html
 

"Media Time Scale
A QuickTime movie always has a time scale, expressed in units per second. You 
can specify a time scale when you create a movie, but the time scale cannot be 
changed once a movie exists. When you perform operations on a QuickTime movie, 
you frequently need to specify a point in the movie timeline at which to begin 
the operation; this is specified using a time value, expressed in movie time 
scale units. You may also need to specify a duration; this is also expressed in 
movie time scale units. The default movie time scale is 600…"

>> Since I don’t have the know how to make changes to the FFmpeg code
>> (changing MOV_TIMESCALE to a higher value in libavformat/movenc.h),
> Sorry, this is not an acceptable answer:
> movenc.h is a textfile that can be changed with any text editor.
I understand that, but I do not yet know how to bring the files locally, edit 
the values and then compile everything into a binary. I have absolutely tried 
this a number of ways, following the Compilation Guide 
https://trac.ffmpeg.org/wiki/CompilationGuide 
, but I did not manage to do it 
successfully. I would love to get there though and so any help is hugely 
appreciated. I’m very determined and will do my utmost to be able to contribute 
a complete answer to this issue.

>> or even better, build a binary for me to test with.
> Could you elaborate why you cannot compile yourself?
> FFmpeg is supposed to build out-of-the-box on every common
> operating system and an a few very uncommon OS's.
> This is the right mailing list for compilation help.

Thanks. I have managed to compile a usable ffmpeg using brew (I work 
exclusively on macOS), but I have not figured out how I would be able to edit 
the libavformat/movenc.h file before compiling using this method. It seems I 
would need to go down the route, so any help to get that far would be so 
welcome.

Thanks
Mark
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[FFmpeg-user] RTMP Timeout

[Marcel Grandemange]

Good Day

Does anyone have a working example of a ffmpeg command that includes a
timeout for rtmp streams as an input?
In certain circumstances ffmpeg simply hangs indefinitely.

As an example:

ffmpeg -stimeout 1 -t 10 -i
rtmp://46.4.25.213/radiotoday/radiotoday.stream
This hangs indefinitely as the -stimeout is meant for rtsp only

ffmpeg -stimeout 1 -t 10 -i
rtsp://46.4.25.213/radiotoday/radiotoday.stream
Gives me following error, which is expected given the short value of
-stimeout
"rtsp://46.4.25.213/radiotoday/radiotoday.stream: Operation timed out"

However Even without the parameter it dies eventually, which is also the
experience with HLS/HTTP

It would seem that rtmp has a "bug" that causes indefinite hang and no way
to work around it?



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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Carl Eugen Hoyos]

2018-03-08 15:57 GMT+01:00, Mark Burton :
> On 8 Mar 2018, at 11:03, Carl Eugen Hoyos  wrote:
>> It is zero as long as nobody confirms that this would fix the issue you
>> see.
>
> Understood. A better option would no doubt be the ability to specify this
> value in the same way we can specify the video_track_timescale value.

If it would not fix your issue why would it be a good option?

[...]

> Since I don’t have the know how to make changes to the FFmpeg code
> (changing MOV_TIMESCALE to a higher value in libavformat/movenc.h),

Sorry, this is not an acceptable answer:
movenc.h is a textfile that can be changed with any text editor.

> I’ll need to find someone who is willing to either test,

This sounds difficult with the given information.

> or even better, build a binary for me to test with.

Could you elaborate why you cannot compile yourself?
FFmpeg is supposed to build out-of-the-box on every common
operating system and an a few very uncommon OS's.
This is the right mailing list for compilation help.

Carl Eugen
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Re: [FFmpeg-user] Generate MPEG Transport stream from audio files only

[Moritz Barsnick]

On Thu, Mar 08, 2018 at 14:25:54 +, Pierre Pasquet wrote:
> Not sure to identify what is missing. This command works for a .m4v file and 
> doesn’t for a .aac file located in the same directory.

As Carl Eugen tried to point out: What is missing is that you provide
your actual (non-working) command line and its complete, uncut console
output. These provide a lot of valuable information, and will help to
analyze your issue.

Cheers,
Moritz
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Re: [FFmpeg-user] How to extract DVD subtitle from this VOB file?

[Farhad Mohammadi Majd]

> I guess your DVD-Film is concatenated ...

The full film is consists of three files, this is the first file.

> in this case you have to extract the whole concatenated VOB with
> mplayer and dump it into a file before you can  apply ffmpeg

I don't have mplayer on the system, I have mpv instead, is it possible
to do that with MPV or ffmpeg itself? although installing mplayer is
not so hard but I prefer to do that with mpv or ffmpeg.

THANKS
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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Mark Burton]

On 8 Mar 2018, at 11:03, Carl Eugen Hoyos  wrote:
> It is zero as long as nobody confirms that this would fix the issue you see.

Understood. A better option would no doubt be the ability to specify this value 
in the same way we can specify the video_track_timescale value.

For the record, I have used a 3rd party tool to alter the timescale to 24000 
and it completely fixed the issue with the ffmpeg output files.

Since I don’t have the know how to make changes to the FFmpeg code (changing 
MOV_TIMESCALE to a higher value in libavformat/movenc.h), I’ll need to find 
someone who is willing to either test, or even better, build a binary for me to 
test with.

Is there anyone out there willing to do this?
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Re: [FFmpeg-user] Generate MPEG Transport stream from audio files only

[Pierre Pasquet]

Not sure to identify what is missing. This command works for a .m4v file and 
doesn’t for a .aac file located in the same directory.
How can it be? 

Thanks for your help, 

> On 8 Mar 2018, at 11:02, Carl Eugen Hoyos  wrote:
> 
> 2018-03-07 19:16 GMT+01:00, Pierre Pasquet :
> 
>> However, It doesn't work with a AAC file.
> 
> Command line and complete, uncut console output missing.
> 
> Carl Eugen
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Re: [FFmpeg-user] use question

[Jim Shupert]

I see that you are a windows usr.
1st - i would suggest linux ( i could expand on the benefits )
2nd-
try something like this in a file.bat

@echo off
cd 
for /f %%a IN ('dir /b *.mp4') do  (
ffmpeg.exe -i %%a -y < your ffmpeg cmd 
instructions of choice>   %%~na_frame%d.png


)
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Re: [FFmpeg-user] use question

[Adi Marvillo]

Am 2018-03-08 um 14:33 schrieb Vuerstaek, Maarten (EC):
> Hi everyone,
>
> I'm trying to do something simple that I can't find the documentation for.  I 
> want to get all the video-files from a folder and cut them all into frames, 
> all together in 1 folder.
> I am opening ffmpeg in powershell and using this command
> .\ffmpeg.exe -i 
> C:\Users\VuerstaekM\Desktop\ffmpeg-20180307-5ab0ecf-win64-static\03072018\video.mp4
>  
> C:\Users\VuerstaekM\Desktop\ffmpeg-20180307-5ab0ecf-win64-static\pics2\frame%d.png
>
> Which works fine for a single video, but I have about a 100 a day, so I need 
> a better solution.
>
> Thanks in advance guys!
> Maarten Vuerstaek
> IT Support
> [cid:image002.png@01D2F4AB.E232CA10]
> Avnet Logistics
> Limesweg 4
> 3700 Tongeren
> BELGIUM
> +32 12 242 888
> it-iso-tonge...@avnet.eu
>
>
>
>
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I do not exactly understand what you want to do, but echo-ing or
performing a operation with all files in a directory runs like this in
linux shell == powershell:

>>> find . -name "*.mp4" -exec echo {} \;        # "." after "find "
means the actual directory
>>> find . -name "*.mp4" -exec ffmpeg  -i{} blabla \;

Cheers
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[FFmpeg-user] use question

[Vuerstaek, Maarten (EC)]

Hi everyone,

I'm trying to do something simple that I can't find the documentation for.  I 
want to get all the video-files from a folder and cut them all into frames, all 
together in 1 folder.
I am opening ffmpeg in powershell and using this command
.\ffmpeg.exe -i 
C:\Users\VuerstaekM\Desktop\ffmpeg-20180307-5ab0ecf-win64-static\03072018\video.mp4
 
C:\Users\VuerstaekM\Desktop\ffmpeg-20180307-5ab0ecf-win64-static\pics2\frame%d.png

Which works fine for a single video, but I have about a 100 a day, so I need a 
better solution.

Thanks in advance guys!
Maarten Vuerstaek
IT Support
[cid:image002.png@01D2F4AB.E232CA10]
Avnet Logistics
Limesweg 4
3700 Tongeren
BELGIUM
+32 12 242 888
it-iso-tonge...@avnet.eu

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Re: [FFmpeg-user] Quicktime - Specify a Movie Header timescale of 24000

[Carl Eugen Hoyos]

2018-03-07 17:58 GMT+01:00, Mark Burton :
> On 7 Mar 2018, at 00:45, Mark Burton  wrote:
>> Does it help if you edit MOV_TIMESCALE in libavformat/movenc.h?
>
> What are the chances that this value could be changed to a much
> higher and more useful value?

It is zero as long as nobody confirms that this would fix the issue you see.

Carl Eugen
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Re: [FFmpeg-user] Generate MPEG Transport stream from audio files only

[Carl Eugen Hoyos]

2018-03-07 19:16 GMT+01:00, Pierre Pasquet :

> However, It doesn't work with a AAC file.

Command line and complete, uncut console output missing.

Carl Eugen
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[FFmpeg-user] ffplay RTSP listen mode with UDP

[Roee Kashi]

Hi,

i'm trying to use RTSP to push video from client to server.
it works on TCP, but fails on UDP.

server side:
> ffplay.exe -rtsp_flags listen rtsp://0.0.0.0:14000/test

client side (TCP mode):
> ffmpeg.exe -re -i  -rtsp_flags prefer_tcp -f rtsp
rtsp://SERVER_IP:14000/test

cliend side (UDP mode):
> ffmpeg.exe -re -i  -f rtsp rtsp://SERVER_IP:14000/test


when I open the ffmpeg in the client side in TCP mode, everything works,
and i see the video stream in the server.
But when i open the ffmpeg in the client side in UDP mode, the server
writes an error:
"bind failed error number -10048" (already in use)

to try and investigate this problem a little bit, i set RTSP min port in
the server to 6001.
the result was the same, yet i could see (using netstat) ffplay listening
on 6001 in UDP (and still writes the same binding error).

seems like something in the RTSP reverse mode is not fully compatible with
UDP mode, and it tries to bind the same port twice.

is there a workaround to this matter?

tested with Zeranoe build 20180307-5ab0ecf

Roee.
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Re: [FFmpeg-user] make error af_afir.o

[helloCgy]

i fixed it, wowwowow,. i know it, just download an yasm-win32, because vsyasm
could'nt konw -M



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[FFmpeg-user] How to extract DVD subtitle from this VOB file?

[Farhad Mohammadi Majd]

Hello, I have ffmpeg v3.2.9 on Debian v9.3

I have searched WWW for suitable command but none of them worked for
me, and ffmpeg is complex.

== My file ==
$ ffmpeg -i VTS_01_1.VOB
.
.
.
.
Input #0, mpeg, from 'VTS_01_1.VOB':
  Duration: 00:24:40.83, start: 0.040033, .
Stream #0:0[0x1bf]: Data: dvd_nav_packet
Stream #0:1[0x1e0]: Video: mpeg2video (Main), .
Stream #0:2[0x80]: Audio: .
Stream #0:3[0x21]: Subtitle: dvd_subtitle
Stream #0:4[0x20]: Subtitle: dvd_subtitle

=

I need to those dvd_subtitle(s). How to extract them?

THANKS
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Re: [FFmpeg-user] make error af_afir.o

[helloCgy]

I missed the question too, i can't understand "Please test with a general
usage yasm binary instead of the special Visual Studio binary. ".
 how should i do to fixed the erro:'LINK : fatal error LNK1181: cannot open
input file 'libavfilter/x86/af_afir.o' '?



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