Re: [FFmpeg-user] Converting text to video for dictation

[John B Morris]

>
> So just to be clear, you’re referring to kinetic typography, right?
> For example stuff like Pulp Fiction: Intonation <
> https://youtu.be/h2m_Z8CWdXI>, derivative graphics work by Jarett Moody
> (slightly NSFW, language)
> Apple has also been playing around with the style since 2016, mostly in
> promo campaigns for non-traditional channels. Don’t Blink <
> https://twitter.com/Apple/status/773604633008680964>
>
(and...)

> This kinda sounds like something you'd with Adobe After Effects.
>
Oh wow. Well, not that advanced
... and I use Shotcut . Might already be able to do
that kind of typography with it somewhat.

I don't yet understand why you want to create a video with transparent
> background.
>
Well, my video editor has an option of creating a transparent-colored
element where you put a bunch of visual filters, one of those being text,
and being able to place that text anywhere. For each block of text at a
different time, I'd have to kind of repeat the process which can get pretty
complex for longer videos, and I can't keyframe text content in it. I'm
looking to simplify that with ffmpeg if possible (and not have to
chroma-key the black background if there is one).

I think you want subtitles with heavy custom styling, maybe as opposed to
> captions like you would do in scenarist, try Aegisub <
> http://www.aegisub.org/> with newer ASS versions.
>
This looks promising actually. Might actually help with most if not
everything I'm looking to do.

Well I actually managed to find a great example... probably should've
provided this initially: https://youtu.be/i3Bno_39qh0
-
Thank you,

John B (JB).


On Tue, Oct 20, 2020 at 12:41 PM Carl Zwanzig  wrote:

>
> On 10/20/2020 6:52 AM, Edward Park wrote:
> > You say it’s not very efficient using your video editor (I assume
> > something along the lines of Avid, PPro, FCPX/Motion), but I don’t think
> > FFmpeg is the right tool for the job, unless you had some really complex
> > ass subtitles already, using a bunch of v4+ features of the script and
> > you just need to burn it into the video.
>
> This kinda sounds like something you'd with Adobe After Effects.
>
> z!
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Re: [FFmpeg-user] afreqshift

[Paul B Mahol]

On Wed, Oct 21, 2020 at 9:29 PM Michael Koch 
wrote:

> Am 21.10.2020 um 21:02 schrieb Michael Koch:
> > Am 21.10.2020 um 20:36 schrieb Paul B Mahol:
> >> On Wed, Oct 21, 2020 at 08:14:50PM +0200, Michael Koch wrote:
> >>> Hi,
> >>>
> >>> I'm just testing the new afreqshift filter.
> >>> In my first test I did use only *.wav audio files. This test was
> >>> successful.
> >>>
> >>> In my next test I'm using MP4 files. First I make a test video with
> >>> a 5kHz
> >>> tone:
> >>> ffmpeg -f lavfi -i sine=f=5000:r=48000:d=5 -f lavfi -i color=black
> >>> -lavfi
> >>> showspectrum=legend=1 -y test.mp4
> >>>
> >>> Now I try to shift the frequency down by 4kHz:
> >>> ffmpeg -i test.mp4 -lavfi afreqshift=-4000,showspectrum=legend=1 -y
> >>> out.mp4
> >>>
> >>> The spectrum display is showing the expected 1kHz, but what I hear
> >>> is 5kHz.
> >>> What's wrong with my command line? Why is the output of the afreqshift
> >>> filter not mapped to the output file?
> >>>
> >> Something like this:
> >>
> >> ffmpeg -i test.mp4 -lavfi
> >> "asplit=2[a][b],[b]afreqshift=-4000,showspectrum=legend=1[b]" -map
> >> "[a]" -map "[b]" -y out.mp4
> >>
> >> You ignored fact that showspectrum is just A->V filter.
> >
> > That did also not work, the audio output is 5kHz. But thanks for
> > pointing me in the right direction. The asplit filter must be after
> > the afreqshift filter. This works:
> >
> > ffmpeg -i test.mp4 -lavfi
> > afreqshift=-4000,asplit[a][b];[b]showspectrum=legend=1 -map "[a]" -y
> > out.mp4
> >
>
> Would it be possible to add an option for deleting all frequencies which
> are smaller than the shift frequency, before shifting down? Because
> otherwise these freqencies are shifted into the negative frequency range
> and can still be heard.
> Example: An audio signal contains normal audio frequencies in the
> [0...12kHz] range and ultrasonic frequencies in the [12kHz...20kHz]
> range. If we want to hear these ultrasonic frequencies, we must shift by
> -12kHz. But then the [0...12kHz] range is shifted to the [-12kHz...0]
> range which means we can still hear it. That could be solved by an
> additional steep highpass filter before the afreqshift filter.
> Maybe it's easier to do that in the frequency domain in the afreqshift
> filter?
>

afreqshift does not operate in the frequency domain at all.
It uses two groups of allpass sections of biquad IIR filters to do most of
the work.
If you need big latency use afftfilt instead, you have a filtergraph for
that already.
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Re: [FFmpeg-user] afreqshift

[Michael Koch]

Am 21.10.2020 um 21:02 schrieb Michael Koch:

Am 21.10.2020 um 20:36 schrieb Paul B Mahol:

On Wed, Oct 21, 2020 at 08:14:50PM +0200, Michael Koch wrote:

Hi,

I'm just testing the new afreqshift filter.
In my first test I did use only *.wav audio files. This test was 
successful.


In my next test I'm using MP4 files. First I make a test video with 
a 5kHz

tone:
ffmpeg -f lavfi -i sine=f=5000:r=48000:d=5 -f lavfi -i color=black 
-lavfi

showspectrum=legend=1 -y test.mp4

Now I try to shift the frequency down by 4kHz:
ffmpeg -i test.mp4 -lavfi afreqshift=-4000,showspectrum=legend=1 -y 
out.mp4


The spectrum display is showing the expected 1kHz, but what I hear 
is 5kHz.

What's wrong with my command line? Why is the output of the afreqshift
filter not mapped to the output file?


Something like this:

ffmpeg -i test.mp4 -lavfi 
"asplit=2[a][b],[b]afreqshift=-4000,showspectrum=legend=1[b]" -map 
"[a]" -map "[b]" -y out.mp4


You ignored fact that showspectrum is just A->V filter.


That did also not work, the audio output is 5kHz. But thanks for 
pointing me in the right direction. The asplit filter must be after 
the afreqshift filter. This works:


ffmpeg -i test.mp4 -lavfi 
afreqshift=-4000,asplit[a][b];[b]showspectrum=legend=1 -map "[a]" -y 
out.mp4




Would it be possible to add an option for deleting all frequencies which 
are smaller than the shift frequency, before shifting down? Because 
otherwise these freqencies are shifted into the negative frequency range 
and can still be heard.
Example: An audio signal contains normal audio frequencies in the 
[0...12kHz] range and ultrasonic frequencies in the [12kHz...20kHz] 
range. If we want to hear these ultrasonic frequencies, we must shift by 
-12kHz. But then the [0...12kHz] range is shifted to the [-12kHz...0] 
range which means we can still hear it. That could be solved by an 
additional steep highpass filter before the afreqshift filter.
Maybe it's easier to do that in the frequency domain in the afreqshift 
filter?


Michael

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Re: [FFmpeg-user] ffmpeg does nothing but using CPU

[Cecil Westerhof]

Reindl Harald  writes:

> Am 20.10.20 um 20:38 schrieb Cecil Westerhof:
>>  Enter command: |all |-1 [ ]
>>
>>  From this moment on ffmpeg does nothing but using CPU cycles
>
> and you don't realize that it waits for input?

I think over 90% CPU just for waiting for input is a 'little'
overkill.


> just remove the redirection and look how to avoid that situation

Nope, I just added -nostdin.

-- 
Cecil Westerhof
Senior Software Engineer
LinkedIn: http://www.linkedin.com/in/cecilwesterhof
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Re: [FFmpeg-user] afreqshift

[Paul B Mahol]

On Wed, Oct 21, 2020 at 08:14:50PM +0200, Michael Koch wrote:
> Hi,
> 
> I'm just testing the new afreqshift filter.
> In my first test I did use only *.wav audio files. This test was successful.
> 
> In my next test I'm using MP4 files. First I make a test video with a 5kHz
> tone:
> ffmpeg -f lavfi -i sine=f=5000:r=48000:d=5 -f lavfi -i color=black -lavfi
> showspectrum=legend=1 -y test.mp4
> 
> Now I try to shift the frequency down by 4kHz:
> ffmpeg -i test.mp4 -lavfi afreqshift=-4000,showspectrum=legend=1 -y out.mp4
> 
> The spectrum display is showing the expected 1kHz, but what I hear is 5kHz.
> What's wrong with my command line? Why is the output of the afreqshift
> filter not mapped to the output file?
> 

Something like this:

ffmpeg -i test.mp4 -lavfi 
"asplit=2[a][b],[b]afreqshift=-4000,showspectrum=legend=1[b]" -map "[a]" -map 
"[b]" -y out.mp4

You ignored fact that showspectrum is just A->V filter.


> Michael
> 
> 
> 
> C:\Users\astro\Desktop>c:\ffmpeg\ffmpeg -f lavfi -i sine=f=5000:r=48000:d=5
> -f lavfi -i color=black -lavfi showspectrum=legend=1 -y test.mp4
> ffmpeg version 2020-10-21-git-289e964873-full_build-www.gyan.dev Copyright
> (c) 2000-2020 the FFmpeg developers
>   built with gcc 10.2.0 (Rev3, Built by MSYS2 project)
>   configuration: --enable-gpl --enable-version3 --enable-static
> --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv
> --enable-gnutls --enable-libxml2 --enable-gmp --enable-lzma
> --enable-libsnappy --enable-zlib --enable-libsrt --enable-libssh
> --enable-libzmq --enable-avisynth --enable-libbluray --enable-libcaca
> --enable-sdl2 --enable-libdav1d --enable-libzvbi --enable-librav1e
> --enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx
> --enable-libass --enable-frei0r --enable-libfreetype --enable-libfribidi
> --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf
> --enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec
> --enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libmfx
> --enable-libglslang --enable-vulkan --enable-libcdio --enable-libgme
> --enable-libmodplug --enable-libopenmpt --enable-libopencore-amrwb
> --enable-libmp3lame --enable-libshine --enable-libtheora --enable-libtwolame
> --enable-libvo-amrwbenc --enable-libilbc --enable-libgsm
> --enable-libopencore-amrnb --enable-libopus --enable-libspeex
> --enable-libvorbis --enable-ladspa --enable-libbs2b --enable-libflite
> --enable-libmysofa --enable-librubberband --enable-libsoxr
> --enable-chromaprint
>   libavutil  56. 60.100 / 56. 60.100
>   libavcodec 58.111.101 / 58.111.101
>   libavformat    58. 62.100 / 58. 62.100
>   libavdevice    58. 11.102 / 58. 11.102
>   libavfilter 7. 88.100 /  7. 88.100
>   libswscale  5.  8.100 /  5.  8.100
>   libswresample   3.  8.100 /  3.  8.100
>   libpostproc    55.  8.100 / 55.  8.100
> Input #0, lavfi, from 'sine=f=5000:r=48000:d=5':
>   Duration: N/A, start: 0.00, bitrate: 768 kb/s
>     Stream #0:0: Audio: pcm_s16le, 48000 Hz, mono, s16, 768 kb/s
> Input #1, lavfi, from 'color=black':
>   Duration: N/A, start: 0.00, bitrate: N/A
>     Stream #1:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 320x240 [SAR
> 1:1 DAR 4:3], 25 tbr, 25 tbn, 25 tbc
> Stream mapping:
>   Stream #0:0 (pcm_s16le) -> showspectrum (graph 0)
>   showspectrum (graph 0) -> Stream #0:0 (libx264)
>   Stream #0:0 -> #0:1 (pcm_s16le (native) -> aac (native))
> Press [q] to stop, [?] for help
> [libx264 @ 01a6b13d0fc0] using SAR=1/1
> [libx264 @ 01a6b13d0fc0] using cpu capabilities: MMX2 SSE2Fast SSSE3
> SSE4.2 AVX FMA3 BMI2 AVX2
> [libx264 @ 01a6b13d0fc0] profile High 4:4:4 Predictive, level 3.2,
> 4:4:4, 8-bit
> [libx264 @ 01a6b13d0fc0] 264 - core 161 r3018 db0d417 - H.264/MPEG-4 AVC
> codec - Copyleft 2003-2020 - http://www.videolan.org/x264.html - options:
> cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1
> psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1
> cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12
> lookahead_threads=2 sliced_threads=0 nr=0 decimate=1 interlaced=0
> bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0
> direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40
> intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0
> qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> Output #0, mp4, to 'test.mp4':
>   Metadata:
>     encoder : Lavf58.62.100
>     Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv444p, 924x640
> [SAR 1:1 DAR 231:160], q=-1--1, 46.88 fps, 12k tbn, 46.88 tbc
>     Metadata:
>   encoder : Lavc58.111.101 libx264
>     Side data:
>   cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: N/A
>     Stream #0:1: Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp,
> 69 kb/s
>     Metadata:
>   encoder : 

Re: [FFmpeg-user] Converting mp3 with included cover art into mp4 with cover art as video fails

[Carl Eugen Hoyos]

Am Mi., 21. Okt. 2020 um 12:52 Uhr schrieb Rimvydas :

> Running "ffmpeg -i in.mp3 out.mp4" fails with:

Please test current FFmpeg git head and provide the complete,
uncut console output.

Carl Eugen
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[FFmpeg-user] RTSP-over-Websocket

[Thomas Schmiedl]

Hello,

I try to display this stream https://herberts.meinekameras.de:10169
(RTSP-over-Websocket) via xupnpd2-mediaserver on the TV-device. Is it
possible in ffmpeg to receive RTSP-over-Websocket? Then I could produce
a local HLS-stream for xupnpd2.

Thanks and regards,
Thomas
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[FFmpeg-user] Converting mp3 with included cover art into mp4 with cover art as video fails

[Rimvydas]

Hi,

I have an mp3 file with two streams:
Stream #0:0: Audio: mp3
Stream #0:1: Video: mjpeg

I'm trying to produce an mp4 file with two streams as well:
Stream #0:0: Audio: aac
Stream #0:1: Video: h264

Running "ffmpeg -i in.mp3 out.mp4" fails with:
[mp4 @ 0x55c1527c66c0] Could not find tag for codec h264 in stream #0,
codec not currently supported in container
Could not write header for output file #0 (incorrect codec parameters ?):
Invalid argument
Error initializing output stream 0:1 --

However, if I extract the image in a separate step and then mux it back, it
works as expected. This works:
ffmpeg -i in.mp3 cover.jpg
ffmpeg -y -i in.mp3 -i cover.jpg -map 0:a:0 -map 1:0 out.mp4

Also, converting to a webm container works as expected too:
ffmpeg -i in.mp3 out.webm

How come when creating an mp4 a "Video: mjpeg" stream works from ".jpg"
input, but not from ".mp3" input? How come this issue appears only when
converting to "mp4", but not to "webm". This looks like a bug, but I hope
there's a simple explanation and I'm missing something.

Versions:
ffmpeg version 4.3.1-4ubuntu1
x264 version: core 160 r3011 cde9a93

Input's ffprobe stream info:
  Duration: 00:21:32.59, start: 0.011995, bitrate: 130 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 128 kb/s
Metadata:
  encoder : Lavf
Stream #0:1: Video: mjpeg (Baseline), yuvj420p(pc,
bt470bg/unknown/unknown), 2228x2228 [SAR 300:300 DAR 1:1], 90k tbr, 90k
tbn, 90k tbc (attached pic)
Metadata:
  title   : Cover.jpg
  comment : Cover (front)

Extracted cover's ffprobe stream info:
  Duration: 00:00:00.04, start: 0.00, bitrate: 20836 kb/s
Stream #0:0: Video: mjpeg (Baseline), yuvj420p(pc,
bt470bg/unknown/unknown), 2228x2228 [SAR 300:300 DAR 1:1], 25 tbr, 25 tbn,
25 tbc

--
Rimvydas
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Re: [FFmpeg-user] ffprobe/ffplay ERROR :: Could not find codec parameters for stream 0 unspecified pixel format,

[Edward Park]

Hi,

> I have a set of mp4 files, which can`t be handled via ffplay/ffprobe.

Try a recent version, you can easily download static builds 
 if you don’t want to compile 
on your own, package manager stable version is usually years old, I think 
because of the standard of vulnerability testing required. Or you could get the 
testing/unstable distro version, but if you don’t have the sources added it’s 
extra steps and even the latest release is usually a few months old. And can 
you confirm that the files work with different programs?

> I did attach the complete log files


Just an fyi, usually helps if you just copy paste into the body of the email 
instead of attaching log files if you can.

Regards,
Ted Park

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Re: [FFmpeg-user] Overlay Error reoccurring

[Edward Park]

Hi,

> I am not sure if I have the overlay code in the correct position or should
> I be adding it to the  "-filter_complex "[0:a][1:a]amerge=inputs=2[a]"" as
> an additional Item.

I think that would be the cleanest way, separated by a semicolon. And you could 
get rid of the output pad label and the explicit stream mappings. I’m not sure 
if you can mix simple and complex filters, maybe putting the -vf after the 
gdigrab input is valid too?

> The overlay runs correctly on its own using windows cmd.


Is that the actual command you type what does \k and * do in windows command 
line??

Regards,
Ted Park

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Re: [FFmpeg-user] ffmpeg audio encoding

[Edward Park]

Hi,

> i have a question what is the recommended way for transcoding only audio
> and keep video as is.
> 
> I tried this with command:
> ffmpeg -n -i something.mkv something_else.mkv -map 0 -vcodec copy -scodec
> copy -acodec ac3 -b:a 640k
> 
> The thing is that using this command with a 10+GB file the process is very
> very slow and it would take more than 1 day.
> 
> I tried manual to remove audio from mkv, encode it in ac3 and repack it
> back which was done in an h.
> 
> Do you have any idea what can be improved?


You need the output options before the output for them to have an effect. The 
command that takes 1 day is transcoding everything using default settings for 
mkv (including audio) and ignoring the trailing options.

Regards,
Ted Park

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[FFmpeg-user] ffmpeg audio encoding

[Simon Medved]

Hi,

i have a question what is the recommended way for transcoding only audio
and keep video as is.

I tried this with command:
ffmpeg -n -i something.mkv something_else.mkv -map 0 -vcodec copy -scodec
copy -acodec ac3 -b:a 640k

The thing is that using this command with a 10+GB file the process is very
very slow and it would take more than 1 day.

I tried manual to remove audio from mkv, encode it in ac3 and repack it
back which was done in an h.

Do you have any idea what can be improved?

File i was using:




Thank you.

*Lep pozdrav // Best regards,*
Simon Medved
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[FFmpeg-user] Overlay Error reoccurring

[John Henderson via ffmpeg-user]

Hi There,

I have this code, which works correctly without the overlay code

/k ffmpeg.exe -y -rtbufsize 1500M

*-vf drawtext=fontfile=C:\Windows\ARLRDBD.TTF:text='Snatch Media
Player':fontcolor=white:fontsize=24:
box=1:shadowcolor=darkblue:shadowx=1:shadowy=1:boxcolor=blue@0.6:boxborderw=5:x=50:y=H-th-50*


-f dshow -i audio="Microphone (High Definition Audio Device)" -f dshow -i
audio=virtual-audio-capturer -f gdigrab -framerate 50 -i desktop -codec:v
h264_nvenc -qp 0 -filter_complex "[0:a][1:a]amerge=inputs=2[a]" -map 2 -map
"[a]"
 
C:\Users\User\source\repos\Mini-Media-Streamer\Mini-Media-Streamer\bin\Debug\Recordings\ScreenRecorder20201021064941.mkv

The error I keep getting is...

Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, dshow, from 'audio=Microphone (High Definition Audio Device)':
  Duration: N/A, start: 1171381.31, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Guessed Channel Layout for Input Stream #1.0 : stereo
Input #1, dshow, from 'audio=virtual-audio-capturer':
  Duration: N/A, start: 1171382.071000, bitrate: 1536 kb/s
Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
[gdigrab @ 01c801d08740] Capturing whole desktop as 1366x768x32 at (0,0)
[gdigrab @ 01c801d08740] Stream #0: not enough frames to estimate rate;
consider increasing probesize
Input #2, gdigrab, from 'desktop':
  Duration: N/A, start: 1603259834.426866, bitrate: 1678562 kb/s
Stream #2:0: Video: bmp, bgra, 1366x768, 1678562 kb/s, 50 fps, 1000k
tbr, 1000k tbn, 1000k tbc
[NULL @ 01c801d10040]


*Unable to find a suitable output format for
'box=1:shadowcolor=darkblue:shadowx=1:shadowy=1:boxcolor=blue@0.6:boxborderw=5:x=50:y=H-th-50'box=1:shadowcolor=darkblue:shadowx=1:shadowy=1:boxcolor=blue@0.6:boxborderw=5:x=50:y=H-th-50:
Invalid argument*
Received stop event after 7 passes

I am not sure if I have the overlay code in the correct position or should
I be adding it to the  "-filter_complex "[0:a][1:a]amerge=inputs=2[a]"" as
an additional Item.

The overlay runs correctly on its own using windows cmd.

Any help or suggestions would be awesome.

John
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