Re: [FFmpeg-user] ffplay frustrations/questions/suggestions

[Toerless Eckert]

Thanks, Marton, - inline.

On Thu, Nov 17, 2016 at 09:56:54PM +0100, Marton Balint wrote:
> > If there is no option to select the output device, i would suggest:
> > - Document that there is no such option, eg: following the documentation for
> >   "-devices".
> > - Print in the output of "-devices" which the actual current output device 
> > is.
> >
> > If there is an option, pls. let me know and document.
> ffplay can't use libavdevice output devices, it uses the SDL drivers.

So "ffplay -devices" lists output devices that itself can not use ?

> That may be caused by the reason that ffplay currently wants to use
> a hardware accelerated renderer with VSYNC support. I guess a
> fallback can be implemented to use any available renderer if that
> fails.

Sounds right.  ffplay works locally, but i had to run it remotely via Xwindows
and that swhere it failed.

Fallback would be lovely. Remote troubleshooting is extremely helpfull
in many cases.

> >Finally: in latest git, ffplay is not even built without SDL2 library
> >discovered on the system. That doesn't seem to be right either.
> 
> Why? FFplay does need SDL2, it can't work without it.

How does someone compiling ffmpeg learn about this dependency ? 
(i was kinda guessing it...)
Eg: i can't remember that i've seen this in a README, and the configure for
ffmpeg does not show this dependency in it's output.

Cheers
Toerless

> Regards,
> Marton
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t...@cs.fau.de
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Re: [FFmpeg-user] ffmpeg slowly get out of sync

[Reuben Martin]

On Wednesday, November 16, 2016 7:59:44 PM CST Luo Xin wrote:
> - capture video from v4l2 device
> - capture audio from alsa device

The drift is going to happen to a certain extent no matter what. You are 
sourceing from two separate pieces of hardware that do not share a common 
hardware clock crystal. 

-Reuben
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Re: [FFmpeg-user] Indeo 4 transparency handling

[Paul Gilbert]

It doesn't look like it; I wasn't able to even find any videos with "indeo"
in the name. It would be inappropriate for me to upload actual game
content, but if you've an interest and a few dollars spare, Starship
Titanic is available on both GOG and Stream for US$6. One example video
from the game is in assets/b16.avi, which contains the animation for a
bedhead coming out of the wall.

Paul.

On Thu, Nov 17, 2016 at 3:17 PM, Carl Eugen Hoyos 
wrote:

> 2016-11-17 20:30 GMT+01:00 Paul Gilbert :
> > Hi there,
> >
> > I'm wondering if anyone has any detailed information about extracting
> > transparency data from an Indeo 4 video.
>
> Is such a sample available on samples.ffmpeg.org?
>
> Carl Eugen
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[FFmpeg-user] Rate control parameters

[Jaka Bac]

Hello

Before I dig through the source code and try to figure out this by myself...
Could someone please explain what exactly do the following options do:

-rc_max_vbv_use
Ratecontrol attempt to use, at maximum,  of what can be used without
an underflow.

-rc_min_vbv_use
Ratecontrol attempt to use, at least,  times the amount needed to
prevent a vbv overflow.

At least / at maximum of what? VBV buffer size?

And also how do these work exactly:
-b_qfactor
qscale factor between IP and B-frames
 * If > 0 then the last P-frame quantizer will be used (q=
lastp_q*factor+offset).
 * If < 0 then normal ratecontrol will be done (q=
-normal_q*factor+offset).

-i_qfactor
qscale factor between P and I-frames
 * If > 0 then the last p frame quantizer will be used (q=
lastp_q*factor+offset).
 * If < 0 then normal ratecontrol will be done (q=
-normal_q*factor+offset).


I know that these set the factor between I and P frames and IP and B frames.
But source says that negative values will use normal rate control and
positive values will just set the factor.

So I am guessing that negative values mean that rate control is done and
then the results are scaled otherwise previous P quantizer value is used
and just scaled.
But what confuses me is that in the -i_qfactor case it also states that
last P quantizer is used.. should not this be last I quantizer?

And what happens if after applying all the factors and offsets, rate
control discovers that the bitrate constraints will not be met? Or even
worse that VBV will owerflow?

Sorry if I am missing something or if I missed some explanation already
exisiting on the interet. If this is explained somewhere already, I kindly
ask if you could point me in the right direction.

Thanks,
Jaka
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[FFmpeg-user] gdigrab on Windows 10 crashing the OS

[Andrew]

Im using gdigrab on Windows 10 (64bit) to capture the screen like this

ffmpeg -f gdigrab -video_size 1620x917 -offset_x 4 -offset_y 90 -i 
desktop -b:v 1000k out.mp4


I can record short clips fine but if I leave it running for more than 
about 30 seconds the entire UI freezes then Windows reboots. The Windows 
Event log doesnt give any clue as to why it crashed.
Could it be running out of RAM (8GB) or has anyone experienced a bug 
with gdigrab that can crash Windows like this?


Thanks
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Re: [FFmpeg-user] ffplay frustrations/questions/suggestions

[Marton Balint]

On Sat, 12 Nov 2016, Toerless Eckert wrote:


It is possible with ffplay to use "-devices" to list output devices, but i could
not find in ffplay(8) or "ffplay --help" or http://ffmpeg.org any indication how
to SELECT a particular output device to use.

 If there is no option to select the output device, i would suggest:
 - Document that there is no such option, eg: following the documentation for
   "-devices".
 - Print in the output of "-devices" which the actual current output device is.

 If there is an option, pls. let me know and document.



ffplay can't use libavdevice output devices, it uses the SDL drivers.


With latest git version (3.2.git), ffplay does not work and gives the following
error: "SDL: could not set video mode - exiting". This is running ffmpeg/ffplay
on linux with environment variable DISPLAY set to remote X display (Mac, 
XQuartz).

ffplay -devices lists:
   sdl,sdl2SDL2 output device
   (and more)

Older versions of ffmpeg (tried <= 2.8x) did work fine with my display
setup and ddplay, but in those versions, ffplay and ffmpeg only include sdl
(not sdl2).

But also: latest ffmpeg works fine with either SDL or SDL2 with the same setup:
   ffmpeg -i  -f [sdl|sdl2] "Window Title"
aka: in same compile, only ffplay does not work with SDL...


That may be caused by the reason that ffplay currently wants to use a 
hardware accelerated renderer with VSYNC support. I guess a fallback can 
be implemented to use any available renderer if that fails.



Finally: in latest git, ffplay is not even built without SDL2 library
discovered on the system. That doesn't seem to be right either.


Why? FFplay does need SDL2, it can't work without it.

Regards,
Marton
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Re: [FFmpeg-user] Indeo 4 transparency handling

[Carl Eugen Hoyos]

2016-11-17 20:30 GMT+01:00 Paul Gilbert :
> Hi there,
>
> I'm wondering if anyone has any detailed information about extracting
> transparency data from an Indeo 4 video.

Is such a sample available on samples.ffmpeg.org?

Carl Eugen
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[FFmpeg-user] Indeo 4 transparency handling

[Paul Gilbert]

Hi there,

I'm wondering if anyone has any detailed information about extracting
transparency data from an Indeo 4 video.

The ffmpeg source sets a flag "has_transp" when transparency data is
present, but doesn't actually use it to decode any data. According to a
format document I read on MultimediaWiki, the transparency is encoded as
"another plane, with Huffman encoded 1 bit per pixel data indicating
whether each pixel is transparent".  But I've tried various experiments
with calling Huffman routines on a sample video, and I'm yet to get
something that looks correct.

For the curious, I'm currently in the middle of reimplementing support for
the game Starship Titanic in ScummVM, and I'm encountering such Indeo
videos being used as object animations.

Thanks for any help.

Paul.
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[FFmpeg-user] Android FFmpeg high memory usage sigkill

[Jonathan Girven]

I am a developer for an Android app that uses FFmpeg to merge,
compress and apply effects to video files. Previously we used a
compiled FFmpeg executable that was run using a shell from our Android
application (with good success). Recently I discovered a patch
included in the FFmpeg master branch that takes advantage of Android's
MediaCodec hardware acceleration
(http://ffmpeg.org/pipermail/ffmpeg-devel/2016-July/196434.html). To
utilise it, I understand FFmpeg needs to be compiled using the JNI, so
I roughly followed the method described in:

https://github.com/appunite/AndroidFFmpeg

I modified the code to use the current git master branch. I have used
the --enable-jni and --enable-mediacodec options in my configure
script. I have written some code to take a command string and run the
ffmpeg.c main function. I believe I now have the system working, but I
am seeing much higher memory usage than with the old executable file,
for the same command. In many cases, Android kills FFmpeg / the whole
app because of it. I have monitored
`Debug.getNativeHeapAllocatedSize()` and see the heap growing to over
1GB on my Nexus 5X.

You can see my FFmpeg command and output here:

http://pastebin.com/gicyqRry

I have added some of my own logs monitoring the progress file and
memory usage in case that is useful. Let me know if more verbose logs
are useful. I am not sure if my problem comes from the FFmpeg command,
build, usage or something else. Any ideas on debugging this would be
gratefully received.
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Re: [FFmpeg-user] how to concatenate several flv files into one flv file

[Jim Worrall]

On Thu, Nov 17, 2016 at 6:21 AM, qw  wrote:

> >> There are several flv file, such as file1.flv, file2.flv, and
> >> file3.flv, which have different encoding settings. How to concatenate
> >> several flv files into one single flv file?
>

Since your files have different encoding settings, you will probably
have to use the concat filter method, transcoding as you go.
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Re: [FFmpeg-user] how to concatenate several flv files into one flv file

[Cley Faye]

2016-11-17 14:21 GMT+01:00 qw :

> >All methods and each of their restrictions are listed here:
> >https://trac.ffmpeg.org/wiki/Concatenate
>
>
> I use the following command, which doesn't work since vlc can play the 1st
> file but not 2nd file in the resulting flv file.
>
>
> ffmpeg -i "concat:file1.flv|file2.flv" -c copy -f flv file_mixed.flv
>
>
> does ffmpeg support the function of merging two flv files correctly?
>

The Concatenate page explicitely tells about this command. It works at the
file level, and only certain files can be concatenated this way. Use any of
the other method.
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[FFmpeg-user] how to send three flv files to rtmp server continuously

[qw]

Hi,


I have three flv files, i.e. file1.flv, file2.flv, file3.flv. I know how to 
send one flv file to rtmp server, as shown below:


ffmpeg.exe -re -i file1.flv -c:v copy -c:a copy -f flv 
rtmp://192.168.183.76:1935/live1/abc


But how to send file1.flv, file2.flv, and file3,flv to rtmp server continuously?


Thanks!


Regards


Andrew

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Re: [FFmpeg-user] how to concatenate several flv files into one flv file

[qw]

>> There are several flv file, such as file1.flv, file2.flv, and
>> file3.flv, which have different encoding settings. How to concatenate
>> several flv files into one single flv file?
>
>All methods and each of their restrictions are listed here:
>https://trac.ffmpeg.org/wiki/Concatenate


I use the following command, which doesn't work since vlc can play the 1st file 
but not 2nd file in the resulting flv file.


ffmpeg -i "concat:file1.flv|file2.flv" -c copy -f flv file_mixed.flv


does ffmpeg support the function of merging two flv files correctly?


Thanks!


Regards


andrew


At 2016-11-17 20:35:23, "Moritz Barsnick"  wrote:
>On Thu, Nov 17, 2016 at 20:12:19 +0800, qw wrote:
>> There are several flv file, such as file1.flv, file2.flv, and
>> file3.flv, which have different encoding settings. How to concatenate
>> several flv files into one single flv file?
>
>All methods and each of their restrictions are listed here:
>https://trac.ffmpeg.org/wiki/Concatenate
>
>Cheers,
>Moritz
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Re: [FFmpeg-user] how to concatenate several flv files into one flv file

[Moritz Barsnick]

On Thu, Nov 17, 2016 at 20:12:19 +0800, qw wrote:
> There are several flv file, such as file1.flv, file2.flv, and
> file3.flv, which have different encoding settings. How to concatenate
> several flv files into one single flv file?

All methods and each of their restrictions are listed here:
https://trac.ffmpeg.org/wiki/Concatenate

Cheers,
Moritz
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[FFmpeg-user] how to send rtmp stream with different encoding setting to rtmp server

[qw]

Hi,


I have one question about ffmpeg usage. I have one long-time flv file, i.e. 
file.flv, and use the following commands to do av transcoding and send it to 
rtmp server.


ffmpeg.exe -re -i file.flv -c:v copy -c:a copy -f flv 
rtmp://192.168.183.76:1935/live1/abc
ffmpeg.exe -re -i file.flv -c:v libx264 -v:b 2000k -c:a libfdk_aac -v:a 32k -f 
flv rtmp://192.168.183.76:1935/live1/abc


How to change the encoding settings in the middle of av transcoding? Then, I 
can send rtmp stream with different encoding settings.


Thanks!


Regards


andrew

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[FFmpeg-user] how to concatenate several flv files into one flv file

[qw]

Hi,


I have one question about ffmpeg usage.


There are several flv file, such as file1.flv, file2.flv, and file3.flv, which 
have different encoding settings. How to concatenate several flv files into one 
single flv file?


Thanks!


Regards


Andrew
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[FFmpeg-user] 答复: how to use swr_convert_frame convert audio form s16p to s16

[He Lei]

FFMpeg 3.2





发件人: He Lei
发送时间: 2016年11月17日 17:44
收件人: ffmpeg-user@ffmpeg.org
主题: [FFmpeg-user] how to use swr_convert_frame convert audio form s16p to s16



Hi,
My sources like this:
  AVFrame *pOutFrame = av_frame_alloc();
  pOutFrame->channel_layout = AV_CH_LAYOUT_STEREO;
  pOutFrame->format = AVSampleFormat::AV_SAMPLE_FMT_S16;
  pOutFrame->sample_rate = 44100;
  static SwrContext* m_pSwrCtx = swr_alloc_set_opts(NULL, 
AV_CH_LAYOUT_STEREO,AV_SAMPLE_FMT_S16, 44100,  
pFrame->channel_layout,(AVSampleFormat)pFrame->format, pFrame->sample_rate, 0, 
NULL);
  swr_init(m_pSwrCtx);

  swr_convert_frame(m_pSwrCtx, pOutFrame, pFrame));//return 0
  av_frame_free(&pOutFrame);

pFrame { nb_channel =2;fotmat=s16p;nb_sample=1024,rate=44100}
After called swr_convert_frame;
I found the linesize[0] of pOutFrame is incorrect,;
according to common sense:
pOutFrame linesize[0] should be nb_channel * nb_sample * sample_bytes,
That is 2*1024*2 = 4096 (format is S16), but it’s 4224;
What's wrong with me……



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[FFmpeg-user] how to use swr_convert_frame convert audio form s16p to s16

[He Lei]

Hi,
My sources like this:
  AVFrame *pOutFrame = av_frame_alloc();
  pOutFrame->channel_layout = AV_CH_LAYOUT_STEREO;
  pOutFrame->format = AVSampleFormat::AV_SAMPLE_FMT_S16;
  pOutFrame->sample_rate = 44100;
  static SwrContext* m_pSwrCtx = swr_alloc_set_opts(NULL, 
AV_CH_LAYOUT_STEREO,AV_SAMPLE_FMT_S16, 44100,  
pFrame->channel_layout,(AVSampleFormat)pFrame->format, pFrame->sample_rate, 0, 
NULL);
  swr_init(m_pSwrCtx);

  swr_convert_frame(m_pSwrCtx, pOutFrame, pFrame));//return 0
  av_frame_free(&pOutFrame);

pFrame { nb_channel =2;fotmat=s16p;nb_sample=1024,rate=44100}
After called swr_convert_frame;
I found the linesize[0] of pOutFrame is incorrect,;
according to common sense:
pOutFrame linesize[0] should be nb_channel * nb_sample * sample_bytes,
That is 2*1024*2 = 4096 (format is S16), but it’s 4224;
What's wrong with me……



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