[FFmpeg-user] Help with concat and overlay

[Joshua Grauman]

Hello all,

I am working on a command that concats two videos from my camera, and then 
overlays another video on top of the concatonated video. It is mostly 
working, but the timing isn't right. For sake of argument, let's say that 
MVI_0001.MOV is 5 min, and MVI_0002.MOV is 3 min (they are broken up by my 
camera, but I just want them together), and overlay.avi is ~8 minutes. I 
want to delay the overlay video 1.1 seconds to align the videos. But when 
I run this command, unless I'm missing something, the first 5 minutes are 
fine, but then the alignment of the overlay is off for the last three 
minutes (it's like it isn't delayed for the last 3 minutes). Does anyone 
see what is wrong with my command?


ffmpeg -i MVI_0001.MOV -i MVI_0002.MOV -ss 1.1 -i overlay.avi -filter_complex 
"[0:0][0:1][1:0][1:1]concat=n=2:v=1:a=1 [v][a]; 
[v][2:0]overlay=format=rgb[vout]" -strict -2 -shortest -map [vout] -map [a] out.mp4

Josh
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Re: [FFmpeg-user] Help with -video_size

[Tim Hiles]

Hi Moritz, Roger

Hi Moritz

>
> >
> > Can you be a bit more clear? Are you trying to record the full 1366x768
> > and resize it to 1024x768, or are you trying to capture only 1024x768
> > of your screen?
>

I wanted to resize to 1024x768. Not capture part of the screen.

If you only want a segment of your desktop, you can let dshow capture
> the complete 1366x768, and have ffmpeg crop the video to the part you
> want to preserve. Insert (e.g.) "-vf crop=x=0:y=0:w=1024:h=768".


Again, I don't want to crop it, however your suggestion reminded me of the
scaling filter, I see there is -vf scale=1024:-1 The question I would have
though is whether you believe filters create more work for ffmpeg or the
computer as opposed to regular parameters such as -video_size? In other
words, is a filter considered re-encoding where -video_size wouldn't be.
Either way, I plan on testing with the filter.

Hi Roger,

>
> >   By default, it captures the "full screen" of the main desktop monitor
> >   [...] To configure it differently, run the provided "configuration
> >   setup utilities/shortcuts" or adjust registry settings before
> >   starting a run (advanced users only) [...]
>
> I think it might respect video_size as well but I'm not certain, and
> definitely not sure if it i truncates versus scaled or what not,
> anyway you could try it.  Yes, I'm not sure and I wrote it  LOL, guess
> it's been awhile.


I noticed the code hasn't been updated in awhile on the git page. Have you
stopped development? I'm curious, was the initial intention of you writing
this program, a response of ffmpeg not having GDIgrab or disliking the
results at the time?  If you disliked the results, would you say GDIgrab
has improved?  Would you recommend using GDIgrab? Is that something that
ffmpeg devs are consistently developing? And if so, is this why you've
stopped development of screen capture recorder? My concern I suppose is
will screen-capture-recorder be unusable as ffmpeg develops. Let me make it
clear there is no judgement here either way, I just want to know for my own
future development as I like the results of screen-capture-recorder.

Thanks!!

Tim
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[FFmpeg-user] Continue encoding even if one input stream as no more data for a short amount of time ?

[Moff Tigriss]

Hi,

For a project, I need to secure the continuity of a live stream output.
The issue is that ffmpeg pause encoding when any input as no new data.
He restart only when the source is reactivated, but the output stream
is now broken for the final client here before the cut.


So, simple question : is there a solution to force ffmpeg to continue
encoding even if one stream input has no data for a short amount of
time (< 5 seconds) ?
Black screen, alternate stream, last frame repeated, weird bash
pipes... Anything.


I can use a buffer, of course. But the general latency increase each
time, and I want to keep it under 10 seconds.


My only actualy working solution is to have OpenBroadcasterSoftware, a
desktop software, reading the stream and encoding it to the front rtmp
server. Wich is insane.

Gstreamer, VLC, MEncoder... hell, I even tested v4l. All of them also
pause when source is down, even if they do overlay/compositing from
multi sources.


 Technical details

My source is a mpegts stream via udp, from another ffmpeg instance who
convert rtmp to mpegts over udp.
When the rtmp source is down, ffmpeg wait the restart (it work). Here,
I have a script who launch a "standing by image" to the udp stream.
And it switch back to the stream later if RTMP is restarted.
The main ffmpeg restart encoding, and everything is good for a new client.

But every client here before the cut have now a broken playback.


 Command lines
(I removed a lot of fine tuning, so the second one can fail sometime,
just relaunch it.)

ffmpeg \
 -re -i rtmp://server:1935/live/testlive \
 -c copy -bsf:v h264_mp4toannexb -f mpegts udp://127.0.0.1:1

ffmpeg \
 -fflags +genpts -stream_loop -1 \
 -r 30 -i udp://127.0.0.1:1 \
 -vcodec libx264 -preset veryfast -crf 22 \
 -g 60 -c:a aac -b:a 128k -ar 44100 \
 -f flv -rtmp_live live rtmp://192.168.61.108:1935/live/restream


 Buildonf (also tested with the current git, nothing different, so
I stick to the stable version)

ffmpeg -buildconf
ffmpeg version 3.2.2 Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  configuration: --prefix=/usr/local
--extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib
--bindir=/usr/local/bin --disable-doc --disable-static --enable-shared
--enable-ffplay --extra-libs=-ldl --enable-version3
--enable-libfdk_aac --enable-libmp3lame --enable-libopus
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264
--enable-libx265 --enable-libxvid --enable-libxcb --enable-pthreads
--enable-gpl --enable-avresample --enable-postproc --enable-nonfree
--disable-debug --enable-small --enable-openssl
  libavutil  55. 34.100 / 55. 34.100
  libavcodec 57. 64.101 / 57. 64.101
  libavformat57. 56.100 / 57. 56.100
  libavdevice57.  1.100 / 57.  1.100
  libavfilter 6. 65.100 /  6. 65.100
  libavresample   3.  1.  0 /  3.  1.  0
  libswscale  4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc54.  1.100 / 54.  1.100

  configuration:
--prefix=/usr/local
--extra-cflags=-I/usr/local/include
--extra-ldflags=-L/usr/local/lib
--bindir=/usr/local/bin
--disable-doc
--disable-static
--enable-shared
--enable-ffplay
--extra-libs=-ldl
--enable-version3
--enable-libfdk_aac
--enable-libmp3lame
--enable-libopus
--enable-libtheora
--enable-libvorbis
--enable-libvpx
--enable-libx264
--enable-libx265
--enable-libxvid
--enable-libxcb
--enable-pthreads
--enable-gpl
--enable-avresample
--enable-postproc
--enable-nonfree
--disable-debug
--enable-small
--enable-openssl



Thank you if you have an answer, i'm losing my mind to solve that
specific issue, and it's the last element missing for my project.
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Re: [FFmpeg-user] Help with -video_size

[Roger Pack]

On 1/6/17, Moritz Barsnick  wrote:
> On Fri, Jan 06, 2017 at 00:14:31 -0800, Tim Hiles wrote:
>> I've tried recording my screen set at 1366 x 768 and specified -video_size
>> to 1024x768. The command did create a recording however it did not change
>> the size. The output is below (further comments after the output).
>
> Can you be a bit more clear? Are you trying to record the full 1366x768
> and resize it to 1024x768, or are you trying to capture only 1024x768
> of your screen?
>
>> ffmpeg -rtbufsize 1000M -f dshow -i
>> video="screen-capture-recorder":audio="Internal Mic (IDT High Definiti"
>> -flags +global_header -vcodec libx264 -pix_fmt  yuv420p -video_size
>> 1024x768 -preset ultrafast -acodec aac -ac 1 -ar 22050 "Sessiontest2.mp4"
>
> Since -video_size is an input option, it won't work this way. If you
> want to tell the input to use this size, you need to place it *before*
> the respective "-i"; which you did in your second example:
>
>> ...which is switching -video_size parameter before the codec and apply it
>> to my command this is what I get.
>>
>> ffmpeg -rtbufsize 1000M -f dshow -video_size 1024x768 -i
>> video="screen-capture-recorder":audio="Internal Mic (IDT High Definiti"
>> -flags +global_header -vcodec libx264 -pix_fmt yuv420p -video_size
>> 1024x768 -preset ultrafast -acodec aac -ac 1 -ar 22050 "Sessiontest2.mp4"
>
> The second "-video_size" has no effect, as mentioned above.

Yeah I suppose in the perfect world FFmpeg would at least *complain*
about unused parameters, but it is as Maritz described.

>> [dshow @ 00558fe0] Could not set video options
>> video=screen-capture-recorder:audio=Internal Mic (IDT High Definiti: I/O
>> error
>
> As the documentation says: "If the device does not support the
> requested options, it will fail to open."
>
> I would guess that the "-video_size" option is not supported by
> screen-capture-recorder. Reading about it does confirm this:
>
> https://github.com/rdp/screen-capture-recorder-to-video-windows-free
> See the section "Configuration":
>
>   By default, it captures the "full screen" of the main desktop monitor
>   [...] To configure it differently, run the provided "configuration
>   setup utilities/shortcuts" or adjust registry settings before
>   starting a run (advanced users only) [...]

I think it might respect video_size as well but I'm not certain, and
definitely not sure if it i truncates versus scaled or what not,
anyway you could try it.  Yes, I'm not sure and I wrote it  LOL, guess
it's been awhile.
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[FFmpeg-user] Change Volume of a Track Without Affecting Other Tracks

[Brandon A. Olivares]

I'm trying to basically take a main audio track (podcast), and add an intro and 
outro to it. However, the music is very loud, so I need to turn down the volume.

However, when I use the volume filter on the intro/outro tracks, I notice that 
the main audio track is also lower than in the original audio.

Here's the command I'm using right now:

ffmpeg -i ccyl011-audio.mp3 -i ambient_travel_stinger.wav -i 
ambient_travel_stinger.wav -filter_complex 
"[0:a]adelay=4600,apad,aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo[audio];[1:a]volume=0.15:precision=fixed,apad,aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo[intro];[2:a]volume=0.15:precision=fixed,adelay=2354600,aformat=sample_fmts=fltp:sample_rates=44100:channel_layouts=stereo[outro];
 [audio][intro][outro]amerge=inputs=3,pan=stereo|c0

Re: [FFmpeg-user] Help with -video_size

[Moritz Barsnick]

On Fri, Jan 06, 2017 at 00:14:31 -0800, Tim Hiles wrote:
> I've tried recording my screen set at 1366 x 768 and specified -video_size
> to 1024x768. The command did create a recording however it did not change
> the size. The output is below (further comments after the output).

Can you be a bit more clear? Are you trying to record the full 1366x768
and resize it to 1024x768, or are you trying to capture only 1024x768
of your screen?

> ffmpeg -rtbufsize 1000M -f dshow -i 
> video="screen-capture-recorder":audio="Internal Mic (IDT High Definiti" 
> -flags +global_header -vcodec libx264 -pix_fmt  yuv420p -video_size 1024x768 
> -preset ultrafast -acodec aac -ac 1 -ar 22050 "Sessiontest2.mp4"

Since -video_size is an input option, it won't work this way. If you
want to tell the input to use this size, you need to place it *before*
the respective "-i"; which you did in your second example:

> ...which is switching -video_size parameter before the codec and apply it
> to my command this is what I get.
> 
> ffmpeg -rtbufsize 1000M -f dshow -video_size 1024x768 -i 
> video="screen-capture-recorder":audio="Internal Mic (IDT High Definiti" 
> -flags +global_header -vcodec libx264 -pix_fmt yuv420p -video_size 1024x768 
> -preset ultrafast -acodec aac -ac 1 -ar 22050 "Sessiontest2.mp4"

The second "-video_size" has no effect, as mentioned above.

> [dshow @ 00558fe0] Could not set video options 
> video=screen-capture-recorder:audio=Internal Mic (IDT High Definiti: I/O error

As the documentation says: "If the device does not support the
requested options, it will fail to open."

I would guess that the "-video_size" option is not supported by
screen-capture-recorder. Reading about it does confirm this:

https://github.com/rdp/screen-capture-recorder-to-video-windows-free
See the section "Configuration":

  By default, it captures the "full screen" of the main desktop monitor
  [...] To configure it differently, run the provided "configuration
  setup utilities/shortcuts" or adjust registry settings before
  starting a run (advanced users only) [...]

If you only want a segment of your desktop, you can let dshow capture
the complete 1366x768, and have ffmpeg crop the video to the part you
want to preserve. Insert (e.g.) "-vf crop=x=0:y=0:w=1024:h=768".

Cheers,
hope this helps,
Moritz
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Re: [FFmpeg-user] Samsung HT-D330K codec issue

[Moritz Barsnick]

On Fri, Jan 06, 2017 at 03:44:11 +0100, James Darnley wrote:
> > * -b:a 4M   # restrict the video bitrate to 4 Mbit/s
> 
> I would suggest -vb (or -b:v in the modern parlance) to set the video
> bitrate.

Sorry, I did mean -b:v of course.
Thanks,
Moritz
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Re: [FFmpeg-user] Why do these blank spots appear in spectrograms of converted youtube videos?

[Paul B Mahol]

On 1/6/17, Ethan Lewis  wrote:
> Hello
>
> I was converting a bunch of youtube videos into 16kHz wav files using ffmpeg
>
> I used the following commands
>
> youtube-dl -f best /path/to/youtube/video/ -f 'bestvideo,bestaudio' -o
> '%(title)s.f%(format_id)s.%(ext)s'
>
> ffmpeg -i /path/to/download/m4a/audio -acodec pcm_s16le -ar 16000 -ac
> test1.wav
>
> When I use these youtube videos
>
> https://www.youtube.com/watch?v=E1mGPeLqg2g
> https://www.youtube.com/watch?v=XBFOP_pF7PY
>
> I see many blank spots in their spectrograms  (see the ones uploaded here
> https://drive.google.com/file/d/0BzNdoGke8tyRSGgwUUtSTkgxdEE/view?usp=sharing
> )
>
> Natural speech should not have such artefacts.
>
> I am confused why this is happening. Is it because of ffmpeg conversion of
> m4a audio OR was it somehow fault of the original recording?

Have you compared with original audio? prior to resampling with ffmpeg?
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[FFmpeg-user] Help with -video_size

[Tim Hiles]

Hi all,

I've tried recording my screen set at 1366 x 768 and specified -video_size
to 1024x768. The command did create a recording however it did not change
the size. The output is below (further comments after the output).

ffmpeg -rtbufsize 1000M -f dshow -i
video="screen-capture-recorder":audio="Interna
l Mic (IDT High Definiti" -flags +global_header -vcodec libx264 -pix_fmt
yuv420p -video_size 1024x768 -preset ultrafast
-acodec aac -ac 1 -ar 22050 "Sessiontest2.mp4"
ffmpeg version N-82966-g6993bb4 Copyright (c) 2000-2016 the FFmpeg
developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --enable-dxva2
--enable-libmfx --enable-nvenc --enable-avisynth --enable
-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv
--enable-libass --enable-libbluray --enable-li
bbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm
--enable-libilbc --enable-libmodplug --enabl
e-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libopenh264 --enable-libopenjpeg --enable-li
bopus --enable-librtmp --enable-libsnappy --enable-libsoxr
--enable-libspeex --enable-libtheora --enable-libtwolame --en
able-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-zl
ib
  libavutil  55. 43.100 / 55. 43.100
  libavcodec 57. 70.100 / 57. 70.100
  libavformat57. 61.100 / 57. 61.100
  libavdevice57.  2.100 / 57.  2.100
  libavfilter 6. 68.100 /  6. 68.100
  libswscale  4.  3.101 /  4.  3.101
  libswresample   2.  4.100 /  2.  4.100
  libpostproc54.  2.100 / 54.  2.100
leaving aero onGuessed Channel Layout for Input Stream #0.1 : stereo
Input #0, dshow, from 'video=screen-capture-recorder:audio=Internal Mic
(IDT High Definiti':
  Duration: N/A, start: 40602.935000, bitrate: N/A
Stream #0:0: Video: rawvideo, bgr0, 1366x768, 30 fps, 30 tbr, 1k
tbn, 1k tbc
Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
[libx264 @ 005c6400] using cpu capabilities: MMX2 SSE2Fast SSSE3
SSE4.2 AVX
[libx264 @ 005c6400] profile Constrained Baseline, level 3.2
[libx264 @ 005c6400] 264 - core 148 r2744 b97ae06 - H.264/MPEG-4
AVC codec - Copyleft 2003-2016 - http://www.vid
eolan.org/x264.html - options: cabac=0 ref=1 deblock=0:0:0 analyse=0:0
me=dia subme=0 psy=1 psy_rd=1.00:0.00 mixed_ref=0
 me_range=16 chroma_me=1 trellis=0 8x8dct=0 cqm=0 deadzone=21,11
fast_pskip=1 chroma_qp_offset=0 threads=12 lookahead_th
reads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
constrained_intra=0 bframes=0 weightp=0 keyint=250
 keyint_min=25 scenecut=0 intra_refresh=0 rc=crf mbtree=0 crf=23.0
qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq
=0
Output #0, mp4, to 'Sessiontest2.mp4':
  Metadata:
encoder : Lavf57.61.100
Stream #0:0: Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p,
1366x768, q=-1--1, 30 fps, 1k tbn, 30 tbc
Metadata:
  encoder : Lavc57.70.100 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1: Audio: aac (LC) ([64][0][0][0] / 0x0040), 22050 Hz, mono,
fltp, 69 kb/s
Metadata:
  encoder : Lavc57.70.100 aac
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
[swscaler @ 005e0fe0] Warning: data is not aligned! This can lead
to a speedloss
Past duration 0.937492 too large4348kB time=00:00:51.53 bitrate=
691.1kbits/s speed=0.993x
frame=175139 fps= 30 q=-1.0 Lsize=  216992kB time=01:37:17.92 bitrate=
304.5kbits/s dup=15 drop=13 speed=   1x
video:165641kB audio:47853kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 1.638695%
[libx264 @ 005c6400] frame I:701   Avg QP:10.05  size:209412
[libx264 @ 005c6400] frame P:174438 Avg QP:13.22  size:   131
[libx264 @ 005c6400] mb I  I16..4: 100.0%  0.0%  0.0%
[libx264 @ 005c6400] mb P  I16..4:  0.1%  0.0%  0.0%  P16..4:  0.1%
 0.0%  0.0%  0.0%  0.0%skip:99.9%
[libx264 @ 005c6400] coded y,uvDC,uvAC intra: 34.1% 30.5% 29.8%
inter: 0.0% 0.1% 0.0%
[libx264 @ 005c6400] i16 v,h,dc,p: 45% 48%  4%  4%
[libx264 @ 005c6400] i8c dc,h,v,p: 49% 36% 12%  3%
[libx264 @ 005c6400] kb/s:232.43
[aac @ 005c8340] Qavg: 48946.633

Inspection of the output file is below:

ffmpeg -i Sessiontest2.mp4
ffmpeg version N-82889-g54931fd Copyright (c) 2000-2016 the FFmpeg
developers
  built with gcc 5.4.0 (GCC)
  configuration: --enable-gpl --enable-version3 --enable-dxva2
--enable-libmfx --enable-nvenc --enable-avisynth --enable
-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv
--enable-libass --enable-libbluray --enable-li
bbs2b 

[FFmpeg-user] Why do these blank spots appear in spectrograms of converted youtube videos?

[Ethan Lewis]

Hello

I was converting a bunch of youtube videos into 16kHz wav files using ffmpeg

I used the following commands

youtube-dl -f best /path/to/youtube/video/ -f 'bestvideo,bestaudio' -o
'%(title)s.f%(format_id)s.%(ext)s'

ffmpeg -i /path/to/download/m4a/audio -acodec pcm_s16le -ar 16000 -ac
test1.wav

When I use these youtube videos

https://www.youtube.com/watch?v=E1mGPeLqg2g
https://www.youtube.com/watch?v=XBFOP_pF7PY

I see many blank spots in their spectrograms  (see the ones uploaded here
https://drive.google.com/file/d/0BzNdoGke8tyRSGgwUUtSTkgxdEE/view?usp=sharing
)

Natural speech should not have such artefacts.

I am confused why this is happening. Is it because of ffmpeg conversion of
m4a audio OR was it somehow fault of the original recording?

Thanks.
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