Re: [FFmpeg-user] rtmp Stream timeout

[Andy Furniss]

Marcel Grandemange wrote:

Evening

I sincerely hope someone can assist me with the following case. I
have the command below which is intended to monitor a rtmp stream and
look for silence within the stream. It takes the rtmp stream plays it
for 30 seconds and looks for 1 second of silence within that period,
this works very well however I have the following case where, if for
any reason the rtmp stream is invalid or down, ffmpeg will "hang"
indefinitely trying to play it.


Not the same test, but I've seen with udp/rtp that ffmpeg will wait for
the stream to start. It won't exit with q but will with a signal or
control c. If the stream stops after it's started it takes 2x signals to
quit.

Maybe you need to script something to kill the process after some time,
check it's gone and repeat the kill if needed.


As matter of interest the behavior is not the same with rtsp links.
If the link is invalid I receive a " Output file #0 does not contain
any stream" and ffmpeg quits which I'm quite happy with. How do I
enable similar functionality with rtmp links? A sample of the command
I use at the moment is below:

/usr/local/bin/ffmpeg -hide_banner -loglevel 32 -t 30 -i
rtmp://server/app/stream.stream -af silencedetect=n=-40dB:d=1 -f null
-


You should normally provide full console output as well as command line.
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Re: [FFmpeg-user] move from audio track to another

[juan carlos Rebate]

yes
1 is Spanish 0 is English

2016-05-06 0:50 GMT+02:00 Moritz Barsnick :

> On Fri, May 06, 2016 at 00:44:00 +0200, juan carlos Rebate wrote:
> > I think even did not understand what I want to do, what I need is that
> the
> > audio output is 1 but do not want to be 0 so tnto that is done with -ac
> but
> > does not work
>
> I'm sorry about our language differences, but you will have to express
> yourself more clearly. "audio output is 1" seems unclear to me in any
> language, taking multimedia processing into account.
>
> Are you saying:
> "I am trying to play (not convert) the link, and I want to hear the
> audio track number 1." ?
>
> Moritz
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Re: [FFmpeg-user] move from audio track to another

[Moritz Barsnick]

On Fri, May 06, 2016 at 00:44:00 +0200, juan carlos Rebate wrote:
> I think even did not understand what I want to do, what I need is that the
> audio output is 1 but do not want to be 0 so tnto that is done with -ac but
> does not work

I'm sorry about our language differences, but you will have to express
yourself more clearly. "audio output is 1" seems unclear to me in any
language, taking multimedia processing into account.

Are you saying:
"I am trying to play (not convert) the link, and I want to hear the
audio track number 1." ?

Moritz
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Re: [FFmpeg-user] move from audio track to another

[juan carlos Rebate]

I think even did not understand what I want to do, what I need is that the
audio output is 1 but do not want to be 0 so tnto that is done with -ac but
does not work

2016-05-06 0:31 GMT+02:00 Cley Faye :

> 2016-05-05 23:43 GMT+02:00 juan carlos Rebate :
>
> > https://trac.ffmpeg.org/wiki/AudioChannelManipulation
> >
> >
> > here it is perfectly explained this in version 2.8 and earlier applies to
> > ffmpeg and ffmplay
>
>
> ​The link you provided explicitely say the same thing: -ac allows you to
> set the *number* of output *channels*, and have nothing to do with chosing
> audio *streams* (or tracks as you put it).​
>
> So far, the behavior of -ac explained by others and by your link are the
> same, and match the current behavior of ffplay I'm observing. What exactly
> is your problem?
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Re: [FFmpeg-user] move from audio track to another

[Cley Faye]

2016-05-05 23:43 GMT+02:00 juan carlos Rebate :

> https://trac.ffmpeg.org/wiki/AudioChannelManipulation
>
>
> here it is perfectly explained this in version 2.8 and earlier applies to
> ffmpeg and ffmplay


​The link you provided explicitely say the same thing: -ac allows you to
set the *number* of output *channels*, and have nothing to do with chosing
audio *streams* (or tracks as you put it).​

So far, the behavior of -ac explained by others and by your link are the
same, and match the current behavior of ffplay I'm observing. What exactly
is your problem?
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Re: [FFmpeg-user] move from audio track to another

[juan carlos Rebate]

https://trac.ffmpeg.org/wiki/AudioChannelManipulation
here it is perfectly explained this in version 2.8 and earlier applies to
ffmpeg and ffmplay

2016-05-05 19:11 GMT+02:00 Moritz Barsnick :

> On Tue, May 03, 2016 at 16:23:58 +0200, juan carlos Rebate wrote:
> > >ffplay -i rtp://239.0.0.111:8208 -ac 0
>
> "-ac" selects the number (amount) of audio channels to output, as in
> "stereo to mono". I'm not even sure it applies to ffplay. And "-ac 0"
> doesn't make sense in that context.
>
> If you want to select a different audio *stream* to play (please note
> the semantic difference between channels and streams), you need to use
> the "-ast" option:
>
>   -ast stream_specifier  select desired audio stream
>
> You can also cycle through the available audio streams interactively at
> run time by pressing the 'a' key.
>
> Moritz
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Re: [FFmpeg-user] C compiler test failed. please help me

[Moritz Barsnick]

On Tue, May 03, 2016 at 16:06:27 +0800, 陈晓峥 wrote:
> C compiler test failed, i have try again and again, but useless, the ffpmeg 
> vesion is 3.0.2 and configure.sh and config.log in the
[...]
> /opt/android-ndk-r10b/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi-gcc:
>  1: 
> /opt/android-ndk-r10b/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi-gcc:
>  Syntax error: "(" unexpected

There seems to be something extremely wrong in this binary/script. Are
you sure you installed it correctly?

If
$ 
/opt/android-ndk-r10b/toolchains/arm-linux-androideabi-4.6/prebuilt/linux-x86_64/bin/arm-linux-androideabi-gcc
 --version
doesn't work, you will have to fix your installation of the compile
environment.

Moritz
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Re: [FFmpeg-user] move from audio track to another

[Moritz Barsnick]

On Tue, May 03, 2016 at 16:23:58 +0200, juan carlos Rebate wrote:
> >ffplay -i rtp://239.0.0.111:8208 -ac 0

"-ac" selects the number (amount) of audio channels to output, as in
"stereo to mono". I'm not even sure it applies to ffplay. And "-ac 0"
doesn't make sense in that context.

If you want to select a different audio *stream* to play (please note
the semantic difference between channels and streams), you need to use
the "-ast" option:

  -ast stream_specifier  select desired audio stream

You can also cycle through the available audio streams interactively at
run time by pressing the 'a' key.

Moritz
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Re: [FFmpeg-user] FFMpeg Streaming via UDP

[Andy Furniss]

Bogdan Mariesan wrote:

Hi Andy,

Sorry for the fuss, my email client added the http part... that was
not intended there.


OK, maybe from an ffmpeg users pov it would be best to test on a lan
without the hole punching code just to see if that part works.


On the client side I am using Java to run the FFMpeg process and in
Android using IJKPlayer to play the stream. I've added pastebin links
since it keeps the code well formatted. Client side -
http://pastebin.com/bkajPR5Q Server used to achieve the UDP hole
punching - http://pastebin.com/vEGdw3QF Android -
http://pastebin.com/gpUUQyjG


Missing / on line 38.

I am far too lazy to check if the ports are reversed properly etc - If I
were doing this I would be looking at tcpdumps to check.



UDP hole punching is the mechanism through which you can establish
P2P connections when your devices are behind routers, secure NATs
etc. where a direct connection between the devices would be
impossible. This can be achieved by having both devices connect to a
public ip server via UDP. When the connection is to the server is
done, on both devices public and private UDP ports are open by the
NAT. Then a direct connection can be established using the public IPs
of the routers of the two devices by specifying the public and
private(local) ports on both sides. Hope i made it clear.

So basically I'm opening a P2P connection between the two devices.

The IPs reported by my UDP hole punching server are below (i've
changed them a bit for security reasons):


OK, maybe it should work "normally" though I don't think it would be
robust with multiple clients behind the same nat as the ffmpeg server is
a different connection from the middle server so the sports could in
theory get changed.

In the UK I think 3g/4g providers do carrier grade nat - I don't know
if/how that would affect things.



My laptop running FFMpeg which sits behind a router 5.2.66.19-56190

My Android phone using 3G/4G 109.55.132.187-52547

And the log generated by FFMpeg is: http://pastebin.com/fyzyk9X1

Hope it makes more sense now.

Kind regards, Bogdan

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Re: [FFmpeg-user] ffmpeg "mpegts" working on linux not on windows ...

[Carl Eugen Hoyos]

debug asm  gmail.com> writes:

> This is a command line that I have used :

If you want support on this mailing list, you have to provide 
both your command line and the complete, uncut console output 
that the command line prints on the console.
And please remove "-v 9 -loglevel 99", the default loglevel 
will answer your question.

(In case anybody wonders: The OP has mixed the console output 
from different versions of FFmpeg.)

Carl Eugen

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Re: [FFmpeg-user] VisualOn vs opencore for amr-wb

[Carl Eugen Hoyos]

sean darcy  gmail.com> writes:

> >> ffmpeg can use the visualon or opencore libraries for amr-wb.
> >>
> >> Any views on which works better ? are the equal ?
> >
> > One is used for decoding (the native decoder does not support
> > all features of the format), the other for encoding.

> OK, visualon only encodes, opencore encodes and decodes, 

> For encoding, does anyone have any preference ?

The preference of anybody does not matter because as said 
above, you have no choice, there is only one (optional) 
amr-wb encoder in FFmpeg.

Carl Eugen

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Re: [FFmpeg-user] FFMpeg Streaming via UDP

[Bogdan Mariesan]

Hi Andy,

Sorry for the fuss, my email client added the http part... that was not
intended there.
On the client side I am using Java to run the FFMpeg process and in Android
using IJKPlayer to play the stream. I've added pastebin links since it
keeps the code well formatted.
Client side - http://pastebin.com/bkajPR5Q
Server used to achieve the UDP hole punching - http://pastebin.com/vEGdw3QF
Android - http://pastebin.com/gpUUQyjG

UDP hole punching is the mechanism through which you can establish P2P
connections when your devices are behind routers, secure NATs etc. where a
direct connection between the devices would be impossible. This can be
achieved by having both devices connect to a public ip server via UDP. When
the connection is to the server is done, on both devices public and private
UDP ports are open by the NAT. Then a direct connection can be established
using the public IPs of the routers of the two devices by specifying the
public and private(local) ports on both sides. Hope i made it clear.

So basically I'm opening a P2P connection between the two devices.

The IPs reported by my UDP hole punching server are below (i've changed
them a bit for security reasons):

My laptop running FFMpeg which sits behind a router
5.2.66.19-56190

My Android phone using 3G/4G
109.55.132.187-52547

And the log generated by FFMpeg is:
http://pastebin.com/fyzyk9X1

Hope it makes more sense now.

Kind regards,
Bogdan




*Bogdan Emil Mariesan*
Senior Software Engineer

*BuddyGuard UG* (haftungsbeschränkt)
Dircksenstr. 40
10178 Berlin

m: +40 743 901 331
w.   www.buddyguard.io

On Thu, May 5, 2016 at 2:33 AM, Andy Furniss  wrote:

> Bogdan Mariesan wrote:
>
>> Hi everyone,
>>
>> I am trying to achieve a *P2P *connection via *UDP *and for that I'm
>> following the wiki guide.
>>
>> On my server I am using:
>>
>> *ffmpeg -f dshow -video_size 640x360 -rtbufsize 702000k -framerate 30 -i
>> video="Integrated Camera":audio="Microphone (5- Logitech USB Headset
>> H340)"
>> -r 30 -threads 4 -vcodec libx264 -pix_fmt yuv420p -tune zerolatency
>> -preset
>> ultrafast -f mpegts udp:/10.166.141.198:23202 <
>> http://10.166.141.198:23202>*
>>
>
> Well I'm lost already as 10/8 is a local address, plus what's the http bit?
>
>
>> Where *10.166.141.198 *is the IP of an Android device and *23202 *is the
>> UDP port opened by the NAT after doing UDP hole punching.
>>
>
> What is this udp hole punching (and why do you need it with a local
> address)?
>
>
>> Based on the example in the guide (
>> https://trac.ffmpeg.org/wiki/StreamingGuide) I assume on the client I
>> have
>> to connect to the client in a similar manner. I might be wrong here since
>> the guide uses the same IP address for both client and server.
>>
>> So on the client I am connection the URL *udp://
>> 5.2.55.19:51810?localport=50341
>> *.
>> Where *5.2.55.19 *is the IP of the serverm *51810 *is the public port and
>> *50341* is the private port obtained after doing UDP hole punching.
>>
>> On the client side the stream seems to work but on my mobile device the
>> app
>> i have built using IJKPlayer (https://github.com/Bilibili/ijkplayer)
>> doesn't display my stream. So is it due to the way I am listening for the
>> UDP stream?
>>
>>
>> Client side log:
>>
>> ffmpeg version N-79630-g9ac154d Copyright (c) 2000-2016 the FFmpeg
>> developers
>>built with gcc 5.3.0 (GCC)
>>configuration: --enable-gpl --enable-version3 --disable-w32threads
>> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
>> --enable-gnutls --enable-iconv --enable-libass --enable-libbluray
>> --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme
>> --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx
>> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
>> --enable-libopenjpeg --enable-libopus --enable-librtmp
>> --enable-libschroedinger --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-decklink --enable-zlib
>>libavutil  55. 22.101 / 55. 22.101
>>libavcodec 57. 38.100 / 57. 38.100
>>libavformat57. 34.103 / 57. 34.103
>>libavdevice57.  0.101 / 57.  0.101
>>libavfilter 6. 44.100 /  6. 44.100
>>libswscale  4.  1.100 /  4.  1.100
>>libswresample   2.  0.101 /  2.  0.101
>>libpostproc54.  0.100 / 54.  0.100
>> Guessed Channel Layout for  Input Stream #0.1 : stereo
>> Input #0, dshow, from 'video=Integrated Camera:audio=Microphone (5-
>> Logitech USB Headset H340)':
>>Duration: N/A, start: 1386931.56, bitrate: N/A
>>  Stream #0:0: Video: rawvideo, bgr24, 640x360, 30 fps, 31 tbr, 1k
>> tbn
>>  

[FFmpeg-user] Multi rtmp input overlayed into 1 not sync

[mulianto]

Hi,

I have multi rtmp input, which have same object to capture, so this for multi 
camera application (picture in picture)

My goal was to make 3 input overlayed and ffmpeg output 1 rtmp stream .

This was done, the challange was each video not sync. Last input will be 
delayed 5-8 second. Second input delayed 4 second, and first input is the 
leading stream inside the overlayed output video. Other words ffmpeg generating 
picture in picture video.

Anyone know how to sync this live stream? I try using a mp4 file, all sync 
inside the pip, but not sync in live stream.

Thanks for help,


Regards

Mulianto
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