Re: [FFmpeg-user] Wave channel split is very slow

2024-02-24 Thread Bouke / Videotoolshed
On 24 Feb 2024, at 12:37, Paul B Mahol  wrote:
> 
> On Sat, Feb 24, 2024 at 11:33 AM Bouke / Videotoolshed <
> bo...@videotoolshed.com> wrote:
> 
>> Trying to split a 64 channel 48Khz 24 bits Wave file.
>> That is VERY slow.
>> 
>> ffmpeg -i /Volumes/Data/magweg/64channel.wav -filter_complex
>> "[0]pan=mono|c0=c0[tr0]" -c:a pcm_s24le -map "[tr0]"
>> /Volumes/Data/magweg/64_test/out_1.wav
>> 
>> Runs at about 15x
>> 
>> But if I do all 64 in one pass, it slows down to about 0.7x
>> 
>> I don’t get it. DiskIO should not be a factor, and there is no math
>> involved. (Granted, the ppm_s24le might involve math, but that should be
>> simple.)
>> 
>> Is this a bug?
>> (Same happens vice versa, going from multiple non-interleaved to one big
>> interleaved Wave.)
>> 
>> Bouke
>> 
> 
> For demuxing case, use -max_size 400 as input option for wav demuxer,

Hi Paul,

Yup, that helps. It now runs 64 channels at about 16x, that is about 20 times 
faster than before!

> order of this option matters.

Where have I heard this before :-) (I won’t go there, no worries.)


> Issue is if single packets have small number of stored samples it hurts
> performance with ffmpeg utility, recently situation got even worse with
> threading work.
I trust you, but I don’t do C
> I raised concerns and reported it to authorities multiple times without
> results, but got responses that is not relevant but very minor problem.

This IS a serious problem, same as stripping audio only from video has become 
dog slow.
FFmbc used to be able to do that (with -c:a copy) at the speed of drive, FFmpeg 
nowadays seems to read the entire file, including video.
> 
> Dunno what command you use for reverse direction so can not help you on
> that one.

My filter compex is HUGE, it splits all inputs to mono, pads / merges / etc. 
(I’m doing auto sequence style.)
But for now, I’ve got something more to toy with.

Thanks,
Bouke

Better command line (way simplified) that indeed runs faster:
/Applications/ffmpeg -max_size 400 -i 
/Volumes/Data/magweg/64_test/64chans.wav -max_size 400 -filter_complex 
"[0]pan=mono|c0=c0[tr0];[0]pan=mono|c0=c1[tr1];[0]pan=mono|c0=c2[tr2];[0]pan=mono|c0=c3[tr3];[0]pan=mono|c0=c4[tr4];[0]pan=mono|c0=c5[tr5];[0]pan=mono|c0=c6[tr6];[0]pan=mono|c0=c7[tr7];[0]pan=mono|c0=c8[tr8];[0]pan=mono|c0=c9[tr9];[0]pan=mono|c0=c10[tr10];[0]pan=mono|c0=c11[tr11];[0]pan=mono|c0=c12[tr12];[0]pan=mono|c0=c13[tr13];[0]pan=mono|c0=c14[tr14];[0]pan=mono|c0=c15[tr15];[0]pan=mono|c0=c16[tr16];[0]pan=mono|c0=c17[tr17];[0]pan=mono|c0=c18[tr18];[0]pan=mono|c0=c19[tr19];[0]pan=mono|c0=c20[tr20];[0]pan=mono|c0=c21[tr21];[0]pan=mono|c0=c22[tr22];[0]pan=mono|c0=c23[tr23];[0]pan=mono|c0=c24[tr24];[0]pan=mono|c0=c25[tr25];[0]pan=mono|c0=c26[tr26];[0]pan=mono|c0=c27[tr27];[0]pan=mono|c0=c28[tr28];[0]pan=mono|c0=c29[tr29];[0]pan=mono|c0=c30[tr30];[0]pan=mono|c0=c31[tr31];[0]pan=mono|c0=c32[tr32];[0]pan=mono|c0=c33[tr33];[0]pan=mono|c0=c34[tr34];[0]pan=mono|c0=c35[tr35];[0]pan=mono|c0=c36[tr36];[0]pan=mono|c0=c37[tr37];[0]pan=mono|c0=c38[tr38];[0]pan=mono|c0=c39[tr39];[0]pan=mono|c0=c40[tr40];[0]pan=mono|c0=c41[tr41];[0]pan=mono|c0=c42[tr42];[0]pan=mono|c0=c43[tr43];[0]pan=mono|c0=c44[tr44];[0]pan=mono|c0=c45[tr45];[0]pan=mono|c0=c46[tr46];[0]pan=mono|c0=c47[tr47];[0]pan=mono|c0=c48[tr48];[0]pan=mono|c0=c49[tr49];[0]pan=mono|c0=c50[tr50];[0]pan=mono|c0=c51[tr51];[0]pan=mono|c0=c52[tr52];[0]pan=mono|c0=c53[tr53];[0]pan=mono|c0=c54[tr54];[0]pan=mono|c0=c55[tr55];[0]pan=mono|c0=c56[tr56];[0]pan=mono|c0=c57[tr57];[0]pan=mono|c0=c58[tr58];[0]pan=mono|c0=c59[tr59];[0]pan=mono|c0=c60[tr60];[0]pan=mono|c0=c61[tr61];[0]pan=mono|c0=c62[tr62];[0]pan=mono|c0=c63[tr63]"
 -map_metadata -1 -c:a pcm_s24le -map "[tr0]" 
/Volumes/Data/magweg/64_test/speed_1.wav -y -map_metadata -1 -c:a pcm_s24le 
-map "[tr1]" /Volumes/Data/magweg/64_test/speed_2.wav -y -map_metadata -1 -c:a 
pcm_s24le -map "[tr2]" /Volumes/Data/magweg/64_test/speed_3.wav -y 
-map_metadata -1 -c:a pcm_s24le -map "[tr3]" 
/Volumes/Data/magweg/64_test/speed_4.wav -y -map_metadata -1 -c:a pcm_s24le 
-map "[tr4]" /Volumes/Data/magweg/64_test/speed_5.wav -y -map_metadata -1 -c:a 
pcm_s24le -map "[tr5]" /Volumes/Data/magweg/64_test/speed_6.wav -y 
-map_metadata -1 -c:a pcm_s24le -map "[tr6]" 
/Volumes/Data/magweg/64_test/speed_7.wav -y -map_metadata -1 -c:a pcm_s24le 
-map "[tr7]" /Volumes/Data/magweg/64_test/speed_8.wav -y -map_metadata -1 -c:a 
pcm_s24le -map "[tr8]" /Volumes/Data/magweg/64_test/speed_9.wav -y 
-map_metadata -1 -c:a pcm_s24le -map "[tr9]" 
/Volumes/Data/magweg/64_test/speed_10.wav -y -map_metadata -1 -c:a pcm_s24le 
-map "[tr10]" /Volumes/Data/magweg/64_test/speed_11.wav -y -map

[FFmpeg-user] Wave channel split is very slow

2024-02-24 Thread Bouke / Videotoolshed
Trying to split a 64 channel 48Khz 24 bits Wave file.
That is VERY slow.

ffmpeg -i /Volumes/Data/magweg/64channel.wav -filter_complex 
"[0]pan=mono|c0=c0[tr0]" -c:a pcm_s24le -map "[tr0]" 
/Volumes/Data/magweg/64_test/out_1.wav

Runs at about 15x

But if I do all 64 in one pass, it slows down to about 0.7x

I don’t get it. DiskIO should not be a factor, and there is no math involved. 
(Granted, the ppm_s24le might involve math, but that should be simple.)

Is this a bug? 
(Same happens vice versa, going from multiple non-interleaved to one big 
interleaved Wave.)

Bouke

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Re: [FFmpeg-user] Packet corrupt on cut, notice on concat

2024-02-19 Thread Bouke / Videotoolshed

> On 19 Feb 2024, at 19:23, Reindl Harald  wrote:
> 
> 
> 
> Am 19.02.24 um 19:20 schrieb Bouke / edit 'B:
>> My most sincere apologies to the list.
>> I have a mental condition that forces me to indulge in these kind of 
>> discussions, while it is clearly pointless.
>> Sorry for the waste of bandwidth / your attention.
>> I’ll make a message rule filtering out the OP.
> 
> given our off-list "conversation" started by you your mental conditions need 
> much more than filter out a single invidual

You are totally right, as you always are.
And you have such a nice way of bringing your point across, as you always make 
an effort to be friendly, and help people.

Now I’m stepping out of this.

Bouke



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Re: [FFmpeg-user] FFmpeg Architecture (was: Codec copy with -ss before -i and after -i is not documented)

2024-02-12 Thread Bouke / Videotoolshed
Even though I used the word "or" in my post, the use of any of the 
architectural techniques can be mixed.

On the subject of you being a troll, you wrote that you posted no more than 6 
posts a year.

That turns out to be ‘a day’

Go away, or at least shut up.


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Re: [FFmpeg-user] Why are these different?

2024-02-11 Thread Bouke / Videotoolshed
So do you, so why ask the question?
Now go away!


> On 11 Feb 2024, at 19:05, Mark Filipak  wrote:
> 
> On 11/02/2024 13.00, Bouke / Videotoolshed wrote:
>> Stop trolling.
>> Read in on the subject. You want to write a FFmpeg manual?
>> Go away, don’t spam the rest of us.
>>> On 11 Feb 2024, at 18:47, Mark Filipak  wrote:
>>> 
>>> Command 1:
>>> ffmpeg -ss 10 -copyts -i this.m2ts -map 0 -c copy -sn -dn -muxdelay 0 
>>> that.m2ts
>>> Command 2:
>>> ffmpeg -copyts -i this.m2ts -map 0 -ss 10 -c copy -sn -dn -muxdelay 0 
>>> that.m2ts
>>> 
>>> Let's ignore all the errors both make. And let's ignore the individual 
>>> errors that they make individually.
>>> 
>>> Should Command 1 and Command 2 make the same outputs? "Yes" or "No"?
> 
> You don't know the answer, do you Bouke?
> 
> 
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Re: [FFmpeg-user] Why are these different?

2024-02-11 Thread Bouke / Videotoolshed
Stop trolling.
Read in on the subject. You want to write a FFmpeg manual?
Go away, don’t spam the rest of us.

> On 11 Feb 2024, at 18:47, Mark Filipak  wrote:
> 
> Command 1:
> ffmpeg -ss 10 -copyts -i this.m2ts -map 0 -c copy -sn -dn -muxdelay 0 
> that.m2ts
> Command 2:
> ffmpeg -copyts -i this.m2ts -map 0 -ss 10 -c copy -sn -dn -muxdelay 0 
> that.m2ts
> 
> Let's ignore all the errors both make. And let's ignore the individual errors 
> that they make individually.
> 
> Should Command 1 and Command 2 make the same outputs? "Yes" or "No"?
> 
> I'm not trying to be rude. I'm simply trying to get a definitive answer to a 
> question that goes to the heart of what ffmpeg is and does.
> 
> Regards,
> Mark.
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Re: [FFmpeg-user] Towards better trims & concatenations

2024-01-08 Thread Bouke / Videotoolshed

> For clarity, I wasn't accusing you of lying and it certainly wasn't my
> intention to imply that; my apologies if it sounded that way!

Don’t feed the trolls


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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

2023-12-20 Thread Bouke / Videotoolshed


> On 19 Dec 2023, at 17:24, Erik Dobberkau  wrote:
> 
>> 
>> @Shuto: I'm sure you've noticed already that you will want to adjust the
>> frame rate (option "-r"). But I'm not sure if that does also require an
>> adjustment of the -bufsize .
>> 
> 
> @Bouke: and it works without explicitly specifying a GOP size of 12 (for
> 50Hz  15 for 60-ish Hz) and a b-frame interval of 2 ?

Hi Eric,
Yup, this is the output of my (commercial) SpotXML app.
https://www.videotoolshed.com/product/spotxml/ 


Broadcasters over here have a Baton, an it eats the files.
(This particular one would be rejected cause having a non 5 sec multiple 
duration, and of course the volume change is very strange, resolution change 
probably not needed, I took the first stupid clip I found…)


Bouke

videotoolshed.com
Van Oldenbarneveltstraat 33
6512 AS Nijmegen, the Netherlands
+31 6 21817248
If you want to send me large files, please use:
https://videotoolshed.wetransfer.com/


> Erik
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Re: [FFmpeg-user] ffmpeg and use of the ildct flag in progressive encoding

2023-12-17 Thread Bouke / Videotoolshed

> On 17 Dec 2023, at 20:44, Nik Armstrong  wrote:
> 
> On 2023-12-16 15:01, Erik Dobberkau wrote:
>> The ildct flag appears only to be making the output file "think" it is
>>> interlaced, when it is, in fact, progressive. It does not seem to be doing
>>> anything to change the footage (quality-wise and it's still progressive).
>>> Since someone other than myself set the flag on this command, I am trying
>>> to figure out what effect, if any, taking it off will do to the output.
>> IIRC, it only has an effect when you are creating an interlaced output
>> using -top 1, (and from here on I‘m guessing) it might instruct the encoder
>> or an automatically inserted filter to apply the DCT in a different fashion
>> compared to a progressive input.
>> But again, that’s a guess which may (also) be outdated, since I last did
>> these kinds of transcodes quite a while ago.
>> Good luck.
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> Thanks Erik. Much appreciated.

Mind you:
Sometimes it’s wanted / needed to have all files ’the same’.
If that is interlaced, it’s interlaced. Codec should say ‘interlaced’, no 
matter the actual content.

For Avid media, that might (depending on the version of MC) be important for a 
‘fast import’, or omitting adapters you don’t want / need.

In broadcast, over here interlaced is mandatory, even if the content is 
progressive.

So no, it’s NOT complicated, and no, it’s NOT outdated.

Bouke

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[FFmpeg-user] Directshow capture and irregular timing of output to StdErr

2023-03-11 Thread Bouke / Videotoolshed
Trying to build a metering tool that displays (almost) RT values.
On Mac, all works fine with AVfoundation, but on Win with Directshow I have an 
issue:

C:\Users\Bouke>ffmpeg.exe -nostats -f dshow -ar 48000 -i audio="Line (Steinberg 
CI2)" -filter_complex "ebur128=peak=true" -f null -

Now, this works fine, BUT, the output to stderr is at very irregular intervals.
Output is as expected lines like:
[Parsed_ebur128_0 @ 01915fe77a40] t: 0.199979   TARGET:-23 LUFSM:-120.7 
S:-120.7 I: -70.0 LUFS   LRA:   0.0 LU  FTPK: -55.5 -71.0 dBFS  TPK: 
-55.2 -69.8 dBFS
[Parsed_ebur128_0 @ 01915fe77a40] t: 0.299979   TARGET:-23 LUFSM:-120.7 
S:-120.7 I: -70.0 LUFS   LRA:   0.0 LU  FTPK: -56.2 -72.7 dBFS  TPK: 
-55.2 -69.8 dBFS

Now, sometimes one line is outputted (as expected) but most of the times it’s 2 
up to 4 lines at the same time.
That does not make for a pretty visual show if I parse it ‘as fast as possible’

I could buffer the output and do my display stuff for each line with my own 
interval / if buffer line length >= 4 or alike, but that would introduce half a 
second delay, and that does not compute well in my brain when I have to ride 
the faders…

Is there anything I can do to have FFmpeg spit out each line ‘as soon as it is 
available’?
(I’m starting FFmpeg from Python, listening to StdOut / StdErr.)

Below the full command, what you cannot see is the irregularity in when the 
lines arrive….

Thx,

Bouke


C:\Users\Bouke>ffmpeg.exe -nostats -f dshow -ar 48000 -i audio="Line (Steinberg 
CI2)" -filter_complex "ebur128=peak=true" -f null -

ffmpeg version 6.0-full_build-www.gyan.dev Copyright (c) 2000-2023 the FFmpeg 
developers
  built with gcc 12.2.0 (Rev10, 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-bzlib --enable-lzma 
--enable-libsnappy --enable-zlib --enable-librist --enable-libsrt 
--enable-libssh --enable-libzmq --enable-avisynth --enable-libbluray 
--enable-libcaca --enable-sdl2 --enable-libaribb24 --enable-libdav1d 
--enable-libdavs2 --enable-libuavs3d --enable-libzvbi --enable-librav1e 
--enable-libsvtav1 --enable-libwebp --enable-libx264 --enable-libx265 
--enable-libxavs2 --enable-libxvid --enable-libaom --enable-libjxl 
--enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass 
--enable-frei0r --enable-libfreetype --enable-libfribidi --enable-liblensfun 
--enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf 
--enable-cuda-llvm --enable-cuvid --enable-ffnvcodec --enable-nvdec 
--enable-nvenc --enable-d3d11va --enable-dxva2 --enable-libvpl 
--enable-libshaderc --enable-vulkan --enable-libplacebo --enable-opencl 
--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  58.  2.100 / 58.  2.100
  libavcodec 60.  3.100 / 60.  3.100
  libavformat60.  3.100 / 60.  3.100
  libavdevice60.  1.100 / 60.  1.100
  libavfilter 9.  3.100 /  9.  3.100
  libswscale  7.  1.100 /  7.  1.100
  libswresample   4. 10.100 /  4. 10.100
  libpostproc57.  1.100 / 57.  1.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, dshow, from 'audio=Line (Steinberg CI2)':
  Duration: N/A, start: 2052080.982000, bitrate: 1536 kb/s
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
[Parsed_ebur128_0 @ 0191617889c0] Summary:

  Integrated loudness:
I: -70.0 LUFS
Threshold:   0.0 LUFS

  Loudness range:
LRA: 0.0 LU
Threshold:   0.0 LUFS
LRA low: 0.0 LUFS
LRA high:0.0 LUFS

  True peak:
Peak:   -inf dBFS
Stream mapping:
  Stream #0:0 (pcm_s16le) -> ebur128:default
  ebur128:out0 -> Stream #0:0 (pcm_s16le)
Press [q] to stop, [?] for help
[Parsed_ebur128_0 @ 01915fe77a40] t: 0.0999792  TARGET:-23 LUFSM:-120.7 
S:-120.7 I: -70.0 LUFS   LRA:   0.0 LU  FTPK: -55.2 -69.8 dBFS  TPK: 
-55.2 -69.8 dBFS
Output #0, null, to 'pipe:':
  Metadata:
encoder : Lavf60.3.100
  Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Metadata:
  encoder : Lavc60.3.100 pcm_s16le
[Parsed_ebur128_0 @ 01915fe77a40] t: 0.199979   TARGET:-23 LUFSM:-120.7 
S:-120.7 I: -70.0 LUFS   LRA:   0.0 LU  FTPK: -55.5 -71.0 dBFS  TPK: 
-55.2 -69.8 dBFS
[Parsed_ebur128_0 @ 01915fe77a40] t: 0.299979   TARGET:-23 LUFSM:-120.7 
S:-120.7 I: -70.0 LUFS   LRA:   0.0 LU  FTPK: -56.2 -72.7 dBFS  TPK: 
-55.2 -69.8 dBFS
[Parsed_ebur128_0 @ 

Re: [FFmpeg-user] ebuR128 display of live source

2023-03-03 Thread Bouke / Videotoolshed
> 
> On 1 Mar 2023, at 11:53, Bouke / Videotoolshed  
> wrote:
> 
> Hi,
> Trying to get a visualisation of my levels during record / stream.
> This works fine for file based material:
> 
> ffplay -f lavfi -i 
> "amovie=/Users/bouke/Downloads/test.wav,ebur128=video=1:meter=18:target=-16 
> [out0][out1]"
> 
> But I would like to have a live source (I’m on Mac, AVfoundation I guess.)


Tried the following:

ffplay -f lavfi -i 
"amovie='\:2':f=avfoundation,ebur128=video=1:meter=18:target=-16 [out0][out1]”

Closer, but still no cigar:
It does not throw an error, but nothing is shown, as if input is not fed to the 
ebur128 filter

 
ffplay version git-2021-03-31-b3e9a84 Copyright (c) 2003-2021 the FFmpeg 
developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.29)
  configuration: --enable-videotoolbox --enable-nonfree --enable-gpl 
--enable-lv2 --enable-openssl --enable-libsvtav1 --enable-libx264 
--enable-libx265 --enable-libvpx --enable-libxvid --enable-libvidstab 
--enable-libaom --enable-libopencore_amrnb --enable-libopencore_amrwb 
--enable-libmp3lame --enable-libopus --enable-libvorbis --enable-libtheora 
--enable-libfdk-aac --enable-libwebp --enable-libsrt --disable-debug 
--disable-doc --disable-shared --enable-pthreads --enable-static --enable-small 
--enable-version3 --enable-libfreetype --enable-libfontconfig --enable-decklink 
--extra-cflags='-I/Users/bouke/ffmpeg-build-script/workspace/include 
-I/Users/bouke/ffmpeg-build-script/include' --extra-ldexeflags= 
--extra-ldflags='-L/Users/bouke/ffmpeg-build-script/workspace/lib 
-L/Users/bouke/ffmpeg-build-script/include' --extra-libs='-ldl -lpthread -lm 
-lz' --pkgconfigdir=/Users/bouke/ffmpeg-build-script/workspace/lib/pkgconfig 
--pkg-config-flags=--static --prefix=/Users/bouke/ffmpeg-build-script/workspace
  libavutil  56. 63.101 / 56. 63.101
  libavcodec 58.117.101 / 58.117.101
  libavformat58. 65.101 / 58. 65.101
  libavdevice58. 11.103 / 58. 11.103
  libavfilter 7. 96.100 /  7. 96.100
  libswscale  5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc55.  8.100 / 55.  8.100
nan:  0.000 fd=   0 aq=0KB vq=0KB sq=0B f=0/0   
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[FFmpeg-user] ebuR128 display of live source

2023-03-01 Thread Bouke / Videotoolshed
Hi,
Trying to get a visualisation of my levels during record / stream.
This works fine for file based material:

ffplay -f lavfi -i 
"amovie=/Users/bouke/Downloads/test.wav,ebur128=video=1:meter=18:target=-16 
[out0][out1]"

But I would like to have a live source (I’m on Mac, AVfoundation I guess.)

How to do that?


Bouke

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Re: [FFmpeg-user] Multiline subtitles without thin dark line

2023-02-15 Thread Bouke / Videotoolshed
Hi Steinar,
>> 
> 
> I know it's due to overlapping boxes. I cannot find anything in the ASS
> specs. or the subtitles filter to alter the line spacing. I would be very
> happy if your confidence could point me to the right setting ;-)

I would if I could, but sorry, that’s beyond my capability. 
For reasons like this, I render my own subtitles as PNG (with metadata), google 
’subbits subtitler’, that’s my work. (And yes, it allows for a box :-)

Then I create a catlist, and overlay, if a burn-in is needed.
Yes, that was quite some work, but I’m a control freak.

> Anyway, overlapping boxes is (to my knowledge) handled by the filter so in
> my opinion it should be addressed there. Cause I don't believe anyone
> needing this style wants the result created by these overlapping
> transparent boxes. Like this dude:
> https://stackoverflow.com/questions/72093116/ffmpeg-subtitles-background-issue-when-multiple-lines-using-srt-format

I hope someone else can help, I can’t…

> Of course suggesting another style is not a solution.

I know, I just gave my opinion on how things “should” look. I come from a 
country where EVERYTHING is subtitled, and I’ve worked (as a pro in movies / 
tv) with subs for over 20 years.

Bouke



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Re: [FFmpeg-user] Multiline subtitles without thin dark line

2023-02-15 Thread Bouke / Videotoolshed

> On 15 Feb 2023, at 13:21, Steinar Apalnes  wrote:
> 
> Hi,
> 
> I want to create subtitles with multiple text lines using BorderStyle=3 in
> order to have the background color adapt to the length of each line. The
> problem is that I get a thin dark line between every text line.

You do not have a stupid black line, you have overlapping boxes so the alpha 
will double.
Increase line spacing and it ’should’ be fine. (I did not try, but I’m pretty 
confident.)
And for my taste, just outline is enough, boxes are ugly and came from stupid 
CC / TeleText.

Bouke


> Is there a
> way to avoid this so that all background areas appear uniform and
> transparent?
> Using BorderStyle=4 is not an option, nor is using a solid black background.
> 
> ffmpeg -f lavfi -i mandelbrot=size=1280x720 -filter_complex
> "subtitles=sub.srt:force_style='Alignment=0,OutlineColour=,BorderStyle=3'"
> -c:v rawvideo -f nut - | ffplay -
> 
> Thanks
> -steinar
> 
> [image: sub.jpg]
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Re: [FFmpeg-user] Any way to estimate moov_size in mov/mp4?

2023-02-10 Thread Bouke / Videotoolshed

> On 10 Feb 2023, at 11:40, Stefan Oltmanns  wrote:
> 
> Hello,
> 
> ffmpeg allows to set a reserved space for the moov at the beginning of
> the file using the "-moov_size" option.
> This will waste some space, but prevent the (in some cases very
> time-consuming) second pass of "-movflags faststart" if you want to have
> the moov atom at the beginning of the file.
> 
> If the reserved space is too small the moov atom gets written at the end
> from my testing (the documentation seems to be wrong here, it states
> muxing will fail, or why could the produced mov file still be played?).
> 
> Is there any way to properly estimate the moov size? In my use case it
> should be rather simple: PCM audio, DNXHD codec (all-I frames, almost
> CBR), known number of frames.
> As the produced files are very big anyway I don't mind wasting a few MB
> by overestimating.

I don’t get it.
Faststart is for internet / network playing only, when the entire file is not 
yet available.
If movflags faststart is needed, it should be darn fast as your files ’should’ 
be fairly small, as they are intended for (relative) low bandwith.
There is NO penalty for having the MOOV atom at the end if a file is local, as 
parsing all atoms will not take longer than a few secs if the file is local.
Having said that:
Checking the size of the moov atom is relative simple, use ‘any’ atom parser, 
or render the same file for 1 frame with a scale as small as you can get, with 
highly compressed sound and use the entire file size for the MOOV atom size 
(plus a bit).

Hth,

Bouke



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Re: [FFmpeg-user] Create a black box over top of the video for a set duration at a set time.

2022-12-31 Thread Bouke / Videotoolshed
On 31 Dec 2022, at 04:53, Gyan Doshi  wrote:
> 
> On 2022-12-31 08:15 am, David Niklas wrote:
>> Sorry for taking so long to reply.
>> It is not working. Terminal output below.
>> 
>> % ffmpeg -i Ducksinarow.mp4 -filter_complex 
>> "[0:0]drawbox=color=black:t=fill=enable='between(t,1.0,2.0)'" -c:a copy -c:v 
>> libvpx ducks.mp4
> 
> There should be a colon after fill, not =
> 
> drawbox=color=black:t=fill:enable='between(t,1.0,2.0)'

And for your next question that might come up (since you want to do multiple 
filters, I do too and just got bitten):
There is a limitation on Windows that won’t accept long arguments.
In that case, save the filter_complex string to a file, then, instead of 
filter_complex you can use filter_complex_script with the filename as argument.

Bouke

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Re: [FFmpeg-user] Create a black box over top of the video for a set duration at a set time.

2022-12-24 Thread Bouke / Videotoolshed
On 23 Dec 2022, at 22:55, David Niklas  wrote:
> 
> Hello,
> What I'd like to achieve is sort of like censoring, but with a black box.
> 
> I can find plenty of examples of how to add a black box to a video, but
> the problem is it is always either the whole video, or only at the
> beginning/end, that the examples add black boxes


Try filter_complex "[0:0]youfilter=enable='between(t,1.0,2.0)’”

Hth
Bouke

> to. I cannot seem to
> figure out how to set the start and end timestamp.
> 
> Thanks,
> David
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Re: [FFmpeg-user] Ubuntu 22.04 audio capture issue

2022-11-30 Thread Bouke / Videotoolshed


> On 30 Nov 2022, at 17:17, Carl Zwanzig  wrote:
> 
> Doubtful, IME.

I enjoyed it, even showed some to my GF. 

Bouke

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Re: [FFmpeg-user] How to reduce interlace artifacts while trancoding to mp4

2022-11-12 Thread Bouke / Videotoolshed

> On 12 Nov 2022, at 01:46, Ulf Zibis  wrote:
> 
> Hi,
> 
> I have transcoded a avi video to mp4 for compressing by factor 8. Now I see 
> interlace artifacts. E.g. the eagle in the back window of the bus or the 
> fence in the background.
> Additionally I would like to reduce the noise for even better compression. 
> Any hint for a good filter?
> 
> Refer:
> http://jugkoeln.de/Videos/2012-01-03_21.46.40_6.29GB_15.avi
> http://jugkoeln.de/Videos/2012-01-03_21.46.40_6.29GB_15.mp4

Hi Ulf,
This is normal DV interlaced video. Open the .avi with VLC and turn OFF 
de-interlace, you’ll see how gorgeous the motion was on an old fashioned 
interlace aware screen.

So, you need to de-interlace the filter.
While you’re at it, you should crop a little, as SD is an over scanned format. 
And, you have non-square pixels that you have to accommodate for, so set aspect 
to 4:3 (in this case), or scale to a 4:3 ratio.

Hth,

Bouke

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Re: [FFmpeg-user] ffserver

2022-11-10 Thread Bouke / Videotoolshed

> On 10 Nov 2022, at 14:22, david stephen  wrote:
> 
> ffserver is not someone that speciality in audio/video. u just need to

Well David,

I have no clue what this is about, but with you write ‘u’ instead of ‘you’, you 
have lost all my respect.
Not signing your post with your name tops it off.

Bouke


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Re: [FFmpeg-user] Wrong TC read with doubled Framenumber read out when input is 50p

2022-11-06 Thread Bouke / Videotoolshed

> 
> On 6 Nov 2022, at 08:42, Björn Büchel via ffmpeg-user 
>  wrote:
> 
> Hi, when I pass a 50p h264 or h265 Video to "ffmpeg -i" without any further 
> code the console shows the "TC In" of the Video. But the Frame coloumn of the 
> File is doubled so the output has the wrong TC. 
> e.g. 
> orginial File TC: 10:00:00:05 
> ffmpeg -i shows: 10:00:00:10
> 
> or e.g. 
> orginial File TC: 10:00:00:45 
> ffmpeg -i shows: 10:00:01:40
> 
> Is this a known issue or does anyone have a workaround?

This I’ve never seen. I know that the TC is written as frames in Mov, Mp4 and 
MXF, so you need the frame rate to convert to HH MM SS FF
This sometimes goes wrong, so different apps calculate different TC values, but 
that is always the entire number doubled / halved.
In your case only the frames are doubled, this makes no sense, a frames to tc 
routine will not use a different FPS for hours / minutes / seconds and frames.

When you say ‘original File TC’, what makes you so sure that is indeed correct? 
What software tells you that? Did you try other apps? (Resolve / Premiere / 
Avid / Quicktime player / whatever you have.)

Bouke


> I would appreciate your help. 
> Björn
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Re: [FFmpeg-user] 16:9 to 2.35:1 and true HD 7.1 to DTS

2022-11-04 Thread Bouke / Videotoolshed

> On 3 Nov 2022, at 22:55, Carl Eugen Hoyos  wrote:
> 
> Am Do., 3. Nov. 2022 um 20:18 Uhr schrieb Bartosz Trzebuchowski
> :
> 
>> 1. How can I convert 16:9 movies to 2.35:1? My projector can’t
>> scale down and 16:9 movies run outside my 2.35:1 canvas at
>> the top and the bottom. I need to black out top and bottom so
>> only 2.35:1 is visible.
> 
> Use the crop filter:
> https://ffmpeg.org/ffmpeg-filters.html#crop
> 
> Carl Eugen
> ___

Definitely not crop, that will remove part of the image.
Instead, pillarbox it, using the pad filter, if needed in combo with scale.

https://ffmpeg.org/ffmpeg-filters.html#pad

Bouke
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Re: [FFmpeg-user] encode to RAW video

2022-10-24 Thread Bouke / Videotoolshed



> On 24 Oct 2022, at 12:52, Naveen.B  wrote:
> 
> Hello Team,
> 
> I have managed to convert raw Images files to .mp4 video,
> I need to encode to a RAW video instead, could you please let me know the
> command for this?
> 
> RAW Image file is,
> resolution - 1600x1300
> fps-30
> bit depth-16bit
> 

Do not specify a video codec, or set -c:v rawhide, set your frame rate / 
resolution / bit depth params and use .raw as extension.

Bouke


> Thanks and Regards,
> Naveen B.
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Re: [FFmpeg-user] Raw video conversion

2022-07-08 Thread Bouke / Videotoolshed
WTF,
 I did NOT, I repeat, NOT write the message below.
And my mail has not been hacked, so what is going on?

Bouke

> On 8 Jul 2022, at 12:17, Bouke / Videotoolshed  
> wrote:
> 
> Please provide full command and console output, and do NOT top post here!
>> 
>> I have a raw video file captured from the monochrome camera, I need to
>> convert the raw video file to visualise it in .avi or .mp4.
>> I tried this command, but this did not worked out to me,
>> ffmpeg -f rawvideo -pixel_format yuv420p -video_size 1600x1300 -framerate
>> 30 -i raw_video.raw raw_video.avi
>> 
>> Could anyone help me on how to do conversion from raw file to .avi or -mp4
>> file?
> 
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Re: [FFmpeg-user] Raw video conversion

2022-07-08 Thread Bouke / Videotoolshed
Please provide full command and console output, and do NOT top post here!
> 
> I have a raw video file captured from the monochrome camera, I need to
> convert the raw video file to visualise it in .avi or .mp4.
> I tried this command, but this did not worked out to me,
> ffmpeg -f rawvideo -pixel_format yuv420p -video_size 1600x1300 -framerate
> 30 -i raw_video.raw raw_video.avi
> 
> Could anyone help me on how to do conversion from raw file to .avi or -mp4
> file?

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Re: [FFmpeg-user] How come after update, it takes so longer, many more process

2022-07-08 Thread Bouke / Videotoolshed


> On 8 Jul 2022, at 04:23, Budi  wrote:
> 
> $ ffmpeg -i "b.mp4" -to 0:2:20  a.mp4
> 
> 
> used to do it nicely done in few seconds

FFmpeg -i "b.mp4" -to 0:2:20  -c: copy a.mp4
Mind the codec copy.

If you omit that, FFmpeg will transcode, resulting in speed and quality loss.

Bouke

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Re: [FFmpeg-user] Make white of photo transparant

2022-06-29 Thread Bouke / Videotoolshed
Pff, make it an option. 
Finding someone who knows this stuff is less work than this shit, including the 
top post flaming.


Boukee use:
https://videotoolshed.wetransfer.com/

> On 29 Jun 2022, at 13:19, Cecil Westerhof via ffmpeg-user 
>  wrote:
> 
> Bouke / Videotoolshed  <mailto:bo...@videotoolshed.com>> writes:
> 
>>> On 29 Jun 2022, at 12:54, Cecil Westerhof via ffmpeg-user 
>>>  wrote:
>>> 
>>> Bouke / Videotoolshed  writes:
>>> 
>>>> No, sorry. But, hire an editor / designer to do this for you,
>>>> results will be better / less pain.
>>>> If you want to have it done for free, be my guest, but don’t ask my
>>>> help, do not expect any half decent results.
>>> 
>>> I am doing it for a school (musical) , so I cannot spend money on it.
>> 
>> What do you want me to say? What is your point?
> 
> You tell me to hire an editor/designer. I explain why that is not an
> option.
> 
> -- 
> Cecil Westerhof
> Senior Software Engineer
> LinkedIn: http://www.linkedin.com/in/cecilwesterhof 
> <http://www.linkedin.com/in/cecilwesterhof>
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Re: [FFmpeg-user] Make white of photo transparant

2022-06-29 Thread Bouke / Videotoolshed


> On 29 Jun 2022, at 12:54, Cecil Westerhof via ffmpeg-user 
>  wrote:
> 
> Bouke / Videotoolshed  writes:
> 
>> No, sorry. But, hire an editor / designer to do this for you,
>> results will be better / less pain.
>> If you want to have it done for free, be my guest, but don’t ask my
>> help, do not expect any half decent results.
> 
> I am doing it for a school (musical) , so I cannot spend money on it.

What do you want me to say? What is your point?


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Re: [FFmpeg-user] Make white of photo transparant

2022-06-29 Thread Bouke / Videotoolshed

>> doing, and IMHO, FFmpeg is not the tool for this.
> 
> A long time ago I tried to work with GUI programs, but I found them to
> bothersome. 

Any half decent video editor / designer works with a GUI, WYSIWYG has been 
invented for a reason.
Doing this without feedback only works in lab circumstances, and I think less 
than 0.01 % of the FFmpeg users have that situation.

>> (But someone recently mentioned a GUI for filters, that might help you big 
>> time on this one.)
> 
> Do you remember the GUI?


No, sorry. But, hire an editor / designer to do this for you, results will be 
better / less pain.
If you want to have it done for free, be my guest, but don’t ask my help, do 
not expect any half decent results.


Bouke

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Re: [FFmpeg-user] Make white of photo transparant

2022-06-29 Thread Bouke / Videotoolshed
On 29 Jun 2022, at 01:48, Cecil Westerhof via ffmpeg-user 
 wrote:
> 
> 
> The background of the photo is white. Would it be possible to make
> that transparent? (So where there is white, I do not see white, but
> the video.)

This is called Luma (or Luminance) key.

Yes, it is possible, yes, it will be ugly unless you know what you are doing, 
and IMHO, FFmpeg is not the tool for this.
(But someone recently mentioned a GUI for filters, that might help you big time 
on this one.)

Bouke



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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

2022-04-20 Thread Bouke / Videotoolshed

> On 20 Apr 2022, at 19:30, Shubham Tiwari  wrote:
> 
> Please accept my apology for putting a large size msg. I appreciate all the
> help received.
> 
> I have another example with similar behavior. The exact commands used are
> below. The detailed output is attached in this email.
> 
> *command 1*: ffmpeg -i call.wav -af
> "volume=enable='between(t,0,1)':volume=0" output.wav
> *behaviour: *executed within a second, input file size 2.1 mb, output file
> size 176 mb
> 
> *command 2*: ffmpeg -i call.wav -af
> "volume=enable='between(t,0,1)':volume=0" output.mp3
> *behaviour: *executed time approx 40 seconds, input file size 2.1 mb,
> output file size 15 mb

You do not specify what kind of output file size you want, so yes, it will be 
‘as it is’.
Having a .wav extension on your input DOES NOT mean your input is uncompressed 
Wave PCM. (Wave Can be compressed, in your case, you have OGG compression with 
a .wav extension.)

What do you expect / why are you doing this? The quality of your file will NOT 
get better, nor will the file size get significant down if you process it.

Concerning your lack of knowledge, the only thing you can accomplish is making 
things worse.

Bouke


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Re: [FFmpeg-user] DNxHD / DNxHR conversions for Avid

2022-04-13 Thread Bouke / Videotoolshed
On 12 Apr 2022, at 21:24, Paul B Mahol  wrote:
> 
>> do cause an error on the Avid side if you drop them directly in an Avid
>> MediaFiles/MXF folder

If you want to use the MXF/xxx folders, you need special wrapped Avid MXF.
The BBC has made a tool for that, https://sourceforge.net/p/bmxlib/home/Home/ 


If you have a recent version of Avid MC, you can use normal Op1 MXF, but those 
have to go to
Avid MediaFiles/UME/xxx

Both can work, I’ve done it before.


Hth,
Bouke

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Re: [FFmpeg-user] Loudnorm or ebu?

2022-04-12 Thread Bouke / Videotoolshed
Learn to live with it, as well as the flaming for top posting, as I’m about to 
receive.
(Or not, if I tell I actually like it to be spanked by nerds.(

Bouke


> On 12 Apr 2022, at 19:02, CMG DiGiTaL  wrote:
> 
> I have noticed, from the harsh and uneducated answers, that the
> distinguished
> Mr. Paul B Mahol, must have serious social relationship problems...
> ...poor man, must be a very lonely person!
> 
> 
> 
> Em ter., 12 de abr. de 2022 às 07:54, Reindl Harald 
> escreveu:
> 
>> 
>> 
>> Am 12.04.22 um 12:29 schrieb Paul B Mahol:
 it's a *users list* if you haven't realized it
>>> 
>>> The tool is not provided by ffmpeg if you have not yet realized it by now
>> 
>> and who gives a shit?
>> it's a users list
>> 
>> just because a few morons on this list pretends wrappers are not
>> supported means nothing on a users-list
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Re: [FFmpeg-user] ffmpeg create wrong time information

2021-12-22 Thread Bouke / Videotoolshed



> On 21 Dec 2021, at 19:37, Cecil Westerhof via ffmpeg-user 
>  wrote:
> 
> Cecil Westerhof via ffmpeg-user  writes:
> 
> After some experiments I solved it by putting -ss and -to behind -i
> and not forgetting to alter the between values.
> But this takes about a third more time.
> Is there be a better solution?
> 
> I am now using:
>ffmpeg -y \
>-i input.mp4  \
>-ss 00:26:05  \
>-to 00:44:04  \
>-vf "
> drawbox=enable='between(t, 1565, 1566.6)':
>color = black :
>w = in_w  :
>h = in_h  :
>thickness = fill
>" \
>-acodec copy  \
>-vcodec libx264   \
>-crf 23   \
>-preset veryfast  \
>output.mkv

I think you are inserting black, not overlaying (blanking) the video. Try -vf 
overlay.
With -ss before input, it will do a faster search, you might have to do a bit 
of math on the enable= range, but that will become very obvious with a small 
test.

Bouke
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[FFmpeg-user] CRF and Preset in H265, am I loosing it?

2020-05-02 Thread Bouke / VideoToolShed
I’ve always thought that setting -crf [x] determines the output quality, while 
(in the same line) -preset [x] affected the output file size.
(Where faster speeds result in bigger output files at the same quality.

But now this:
https://trac.ffmpeg.org/wiki/Encode/H.265 


Under the Constant Rate Factor (CRF):

-Put differently, if you choose ultrafast, the encoding process is going to run 
fast, and the file size will be smaller when compared to medium.

WTF? I would think the output file size will be bigger? (I take it this is just 
a stupid mistake…)

Then, the line continues:
-The visual quality will not be as good. 
 
Now it gets confusing. This sorta kinda suggests that -crf and -preset are not 
to be used together?

What am I missing here?

Thx,

Bouke
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[FFmpeg-user] rw_timeout trouble

2020-04-21 Thread Bouke / VideoToolShed
Hi all,
Strange issue:
Trying to transcode MXF ‘while in the making’ with the rw_timeout option.
It works on my end using Adobe Media Encoder for the MXF, but if Avid MC (on 
another computer that I don’t control) generates the MXF, 
the first video frame gets doubled, and a more than expected by the video frame 
doubled audio offset.

Below is the FFmpeg command (not the ‘real’ command, I use Python to fire 
FFmpeg, and that takes care of the quoting / escaping stuff, the line is a rip 
off from the list I send…),
and the (JSON) FFprobe output I use to generate the command line.

Clues?
Thx,

Bouke

FFmpeg -threads 0 -rw_timeout 200 -i 
/Users/giulio/Desktop/whatchforlder/in/TESTMXF TO MP4_SIMULPRORES.mxf -vsync 
vfr -f lavfi -i anullsrc=channel_layout=mono:sample_rate=48000 -map 0:0 
-filter_complex 
[0:a]channelmap=0-0:channel_layout=0x1[a0];[1:a]channelmap=0-0:channel_layout=0x1[a1];[a0][a1]amerge=inputs=2[out]
 -map [out] -c:a aac -strict -2 -ab 64k -ar 32000 -timecode 01:00:00:00 
-metadata:s:v:0 reel_name=TESTMXF TO MP4_SIMULPRORES -c:v libx264 -pix_fmt 
yuv420p -preset ultrafast -crf 23 
/Users/giulio/Desktop/whatchforlder/out/TESTMXF TO MP4_SIMULPRORES.mp4 -y 

ffmpeg version git-2020-03-15-c467328 Copyright (c) 2000-2020 the FFmpeg 
developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
  configuration: --enable-gpl --enable-version3 --enable-sdl2 
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass 
--enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 
--enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab 
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex 
--enable-libxvid --enable-libaom --enable-appkit --enable-avfoundation 
--enable-coreimage --enable-audiotoolbox
  libavutil  56. 42.100 / 56. 42.100
  libavcodec 58. 75.100 / 58. 75.100
  libavformat58. 41.100 / 58. 41.100
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 77.100 /  7. 77.100
  libswscale  5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
[mxf @ 0x7fcfe680] broken or empty index
Guessed Channel Layout for Input Stream #0.1 : mono
Guessed Channel Layout for Input Stream #0.2 : mono
Input #0, mxf, from '/Users/giulio/Desktop/whatchforlder/in/TESTMXF TO 
MP4_SIMULPRORES.mxf':
  Metadata:
operational_pattern_ul: 060e2b34.04010101.0d010201.01010900
product_uid : d49f75ba-4965-4cfa-9d11-cd87206287e0
uid : 60393316-83b2-11ea-bd8c-4c32759d7b9f
generation_uid  : 60393317-83b2-11ea-acf9-4c32759d7b9f
company_name: Avid Technology, Inc.
product_name: Avid MediaProcessor Plug-In
product_version : 1.0.43.10024.1
application_platform: MXF::SDK (4.6.51) on Mac OS X
modification_date: 2020-04-21T09:27:53.00Z
material_package_umid: 
0x060A2B340101010501010D12131C5C3690ED0702608905A5781F4C32759D7B9F
timecode: 01:00:00:00
  Duration: 00:01:30.04, start: 0.00, bitrate: 7080 kb/s
Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709, progressive), 
1920x1080 [SAR 1:1 DAR 16:9], 5 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
Metadata:
  file_package_umid: 
0x060A2B340101010501010D1213347B3C90ED0702608905A580D84C32759D7B9F
  file_package_name: Source Package
  track_name  : Track 2
Stream #0:1: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D1213347B3C90ED0702608905A580D84C32759D7B9F
  file_package_name: Source Package
  track_name  : Track 3
Stream #0:2: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D1213347B3C90ED0702608905A580D84C32759D7B9F
  file_package_name: Source Package
  track_name  : Track 4
Input #1, lavfi, from 'anullsrc=channel_layout=mono:sample_rate=48000':
  Duration: N/A, start: 0.00, bitrate: 384 kb/s
Stream #1:0: Audio: pcm_u8, 48000 Hz, mono, u8, 384 kb/s
Stream mapping:
  Stream #0:1 (pcm_s24le) -> channelmap (graph 0)
  Stream #1:0 (pcm_u8) -> channelmap (graph 0)
  Stream #0:0 -> #0:0 (mpeg2video (native) -> h264 (libx264))
  amerge (graph 0) -> Stream #0:1 (aac)
Press [q] to stop, [?] for help
[libx264 @ 0x7fcfe6011c00] using SAR=1/1
[libx264 @ 0x7fcfe6011c00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 
AVX FMA3 BMI2 AVX2
[libx264 @ 0x7fcfe6011c00] profile Constrained Baseline, level 4.0, 4:2:0, 8-bit
[libx264 @ 0x7fcfe6011c00] 264 - core 159 - H.264/MPEG-4 AVC codec - Copyleft 
2003-2019 - http://www.videolan.org/x264.html - options: 

Re: [FFmpeg-user] 'image trails'

2020-04-08 Thread Bouke / VideoToolShed


> On 08 Apr 2020, at 14:00, Mark Filipak  
> wrote:
> 
> On 04/08/2020 07:48 AM, Bouke / VideoToolShed wrote:
>>> On 08 Apr 2020, at 13:43, Mark Filipak 
>>>  wrote:
>>> 
>>> Paul, do you really think I have bad intent?
>> you certainly smell like Custom Sift,
>> (a troll on another list.)
>> Bouke
> 
> What I'm attempting to do, Bouke, is perfect the transcode of p24 to 
> judder-free and comb-free 60 FPS for display on 60Hz TVs. I have the 5-5-5-5 
> pull-down telecine working. The result is judder-free and has half the 
> combing of usual telecine, but I endeavor to eliminate the combing, too.
> 
> I'm not as familiar with ffmpeg as the developers of course, so when I run 
> into a problem, I post here. Isn't that what this mailing list is for?
> 
> Paul seems to have a burr under his saddle. He wants me to give up. Why that 
> is, is for someone else to figure out. I plan to forge ahead toward what is a 
> worthy goal.
> ___

Paul can be harsh, but you’ve just proven my point.

Bouke
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Re: [FFmpeg-user] 'image trails'

2020-04-08 Thread Bouke / VideoToolShed


> On 08 Apr 2020, at 13:43, Mark Filipak  
> wrote:
> 
> Paul, do you really think I have bad intent? 

you certainly smell like Custom Sift,
(a troll on another list.)

Bouke


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Re: [FFmpeg-user] Understanding how to use FFmpeg on macOS

2020-03-23 Thread Bouke / VideoToolShed


> On 23 Mar 2020, at 11:29, Ted Park  wrote:
> 
> Hi,

Hi all, since I’m the developer, (btw, @ Ted, this is the project you’ve helped 
me on with audio patchting), let me comment.
> 
> You don’t need binaries to be signed to run on Catalina. If you can run 
> ffmpeg manually other programs probably will be able to as well. If apps have 
> hardened runtime enabled (which is to say all notarized apps) then Gabry is 
> right, usually they can’t use anything other than stuff included in the app 
> and system libraries, but an app that asks for an external library location 
> presumably has the entitlement granting them that exception.

What I do is xatrr -dr com.apple.quarantine (ffpath), and that seems to work 
just fine.
(And yes, my app is notarised.)


> As far as support, the quicktime framework has been deprecated on macOS and 
> is unavailable in Catalina. The replacement is AVFoundtion which dropped 
> support for a lot of features in the format, including some that are still in 
> use (basically any function/codec not available 64bit).

This is what it is about. the OP (Tangier) want to do a destructive change of 
TC on source files.
I’ve already told him that this is a problem now due to the changes in the OS 
(removal of QuickTime)..
Previous version of my work could do that with a plugin, based on the quicktime 
framework.
But that is gone. Now, ‘some’ QT’s have TC written as frame number in the 
beginning of the mdat chunk, but some not.
I can parse ‘some’ TC chunks, that tell me frame duration / frame rate and 
alike, but for the life of me I can’t find where the actual TC data is stored.
(And I can’t say I understand the Apple documentation.)

I’m pretty sure this is in the FFmpeg source code somewhere, but I can’t read 
C….

> How did you convert the mts to a mov? I am pretty sure you have to change the 
> underlying structure of the video to convert from one to the other.

That’s probably FFmpeg work, but then there is no issue, as FFmpeg can easilly 
add a new TC when copying / transcoding to a new file.
The devil is in the destructive TC changing, but that is not something FFmpeg 
is supposed to do.

Bouke

> 
> Regards,
> Ted Park
> 
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Re: [FFmpeg-user] more sound madness, looks SOLVED

2020-03-10 Thread Bouke / VideoToolShed


> On 10 Mar 2020, at 07:28, Ted Park  wrote:
> 
>> ? Start is the same, but then I’m lost...
>> 
>>> channelmap=map=0-0|1-1|2-2:
>> This also confuses me. The order of the second and third mapping looks likes 
>> it comes from the 0 1 1 yadda at the beginning.
>> But where is channel 0 from the second input mapped? Whatever I try, I 
>> always end up with channel 0...
> 
> The channels aren’t on every single bit field, but that’s basically it, I 
> just wanted to try to minimize the variables to see if I could get the same 
> thing to work, because arguments and filter parameters related to audio 
> channels seem to vary (or maybe it’s a discrepancy in the docs)  From 
> libavutil/channel_layout.h:
> #define AV_CH_FRONT_LEFT 0x0001
> #define AV_CH_FRONT_RIGHT0x0002
> #define AV_CH_FRONT_CENTER   0x0004
> #define AV_CH_LOW_FREQUENCY  0x0008
> #define AV_CH_BACK_LEFT  0x0010
> #define AV_CH_BACK_RIGHT 0x0020
> #define AV_CH_FRONT_LEFT_OF_CENTER   0x0040
> #define AV_CH_FRONT_RIGHT_OF_CENTER  0x0080
> #define AV_CH_BACK_CENTER0x0100
> #define AV_CH_SIDE_LEFT  0x0200
> #define AV_CH_SIDE_RIGHT 0x0400
> #define AV_CH_TOP_CENTER 0x0800
> #define AV_CH_TOP_FRONT_LEFT 0x1000
> #define AV_CH_TOP_FRONT_CENTER   0x2000
> #define AV_CH_TOP_FRONT_RIGHT0x4000
> #define AV_CH_TOP_BACK_LEFT  0x8000
> #define AV_CH_TOP_BACK_CENTER0x0001
> #define AV_CH_TOP_BACK_RIGHT 0x0002
> #define AV_CH_STEREO_LEFT0x2000  ///< Stereo downmix.
> #define AV_CH_STEREO_RIGHT   0x4000  ///< See AV_CH_STEREO_LEFT.
> #define AV_CH_WIDE_LEFT  0x8000ULL
> #define AV_CH_WIDE_RIGHT 0x0001ULL
> #define AV_CH_SURROUND_DIRECT_LEFT   0x0002ULL
> #define AV_CH_SURROUND_DIRECT_RIGHT  0x0004ULL
> #define AV_CH_LOW_FREQUENCY_20x0008ULL
> I figured the hex notation was a way to leave little to be 
> interpreted/resolved but apparently that wasn’t the issue. I used the 
> channels labeled stereo mix and top back right thinking they were less likely 
> to overlap, so 0x6002 stands for 0x4000|0x2000|0x0002.

Hi Ted,
> What is the error you get with channel map? Or does it just set the channels 
> to be silent or disable them with no warning again?

Forget it, I think (with your help) I’ve nailed it.

> The multitrack file is actually labeled as the down mix on the first two 
> channels, then the recorded tracks:
>> Input #1, wav, from '/Volumes/Data/test/068.WAV':
>> Metadata:
>>   comment : sSPEED=025.000-ND 
>>   : sTAKE=068 
>>   : sUBITS=$ 
>>   : sSWVER=5.01.8149 
>>   : sSCENE=MixPre 
>>   : sFILENAME=MixPre-068.WAV 
>>   : sTAPE=200130 
>>   : sCIRCLED=FALSE 
>>   : sTRK1=MixL 
>>   : sTRK2=MixR 
>>   : sTRK3=Track 1 
>>   : sTRK4=Track 2 
>>   : sTRK5=Track 3 
>>   : sTRK6=Track 4 
>>   : sTRK7=Track 5 
>>   : sTRK8=Track 6 
>>   : sTRK9
>>   encoded_by  : SoundDev: MixPre-10 II TJ0019225
>>   originator_reference: USSDVTJ001922501520013072A000201
>>   date: 2020-01-30
>>   creation_time   : 07:02:20
>>   time_reference  : 2468718515
>>   coding_history  : A=PCM,F=48000,W=24,M=multi,R=48000,T=10 
>> Ch;Ambisonics=off 

I don’t think this metadata is used. (And it’s incomplete as well, this is just 
a dump of the BEXT chunk that is not large enough to store all the metadata 
there is.
So, the names are completely irrelevant for FFmpeg. (And for everyone, as the 
good stuff is in the iXML.)
I must admit, have not tested it very well, but I would expect that there is a 
-ignore_metadata command as input option if it turns out I’m wrong here.

> Maybe you will get automatic channel detection working again if you don’t 
> include the stereo and use the 8 channels.

Not  an option, as what I’m making is intended to merge cam sound with external 
sound


> Are some of the tracks actually sub-mixes bounced on the device? (Since you 
> get tracks 8 and 9)

Probably, but that’s beside the issue, it’s not up to me what end users want to 
map to where :-)

What DOES work as expected:
(Reverse mapping of BWF sound, with the last channel duplicated, reverse 
mapping of the two video sound channels.)
The channelmap from the BWF is a sum of (zero based) 2 to 12 from you list 
above, to leave room for the 0x3 (first two) for the video sound:

/Applications/ffmpeg -i /Volumes/Data/test/10track.wav -i 
/Volumes/Data/test/input.mp4 -filter_complex 

Re: [FFmpeg-user] more sound madness

2020-03-09 Thread Bouke / VideoToolShed

> On 09 Mar 2020, at 07:16, Ted Park  wrote:
> 
> Hello,
> 
>> Did you try it in Premiere, and got all the channels? Or open the output in 
>> old QT pro, and look at the properties.
>> The last channels are disabled. If I enable them and save the file 
>> (destructive), a new MOOV atom is added and the current one is flagged as 
>> FREE.
>> Then Premiere will accept it as it ‘should’ be.
>> (Warning, Premiere has  a nasty habit to ‘remember’ files. Once it has seen 
>> a specific file, it will use the properties of the first import, even when 
>> you change it and re-import.)
> 
> Premier yes, QuickTime Pro I can’t run anymore. It starts importing all 
> tracks in Premier, and plays all tracks by default in Quicktime Player. For a 
> simple change I opened the file in a hex editor and set/cleared the bits by 
> hand, each track had a flag at their track header “tkhd” box, 0x01 
> enables, 0x02 sets whether it’s part of the main presentation, etc.

Hi Ted,
Ok, that’s what I’m seeing as well! (Now, I did not know that the track header 
contained this info, but indeed, the muted tracks show 0x02, setting them  
to 0x03 fixes the issue!
This is going to be my backup plan!


> (If you set premier to cache metadata to a separate sidecar file alongside 
> media files it can take the guesswork out at least, just delete the xmp to 
> force a refresh)

Good one! Never thought of that.

> 
> I think I see what you are getting though, when I disabled tracks manually 
> they just disappeared from the list in QT, but they were still there, just 
> impossible to see. All ten still imported fine in Premier though.

Not sure what version you are using but it does not on my end...
But, it seems easy enough to fix. (Although I still think this is a FFmpeg bug…)

> I’m wondering what’s causing the different behavior though…
> 
>>> % ffmpeg -i /Volumes/Data/test/C0092.MP4 -i /Volumes/Data/test/068.WAV 
>>> -filter_complex 
>>> “[0:1]pan=1c|c0=c1[o1];[1:0]pan=1c|c0=c9[o2];[1:0]pan=1c|c0=c10[o3]” -map 
>>> “[o1]" -map “[o2]" -map “[o3]"
>>> -c:a pcm_s24le -map 0:0 -c:v copy -shortest 
>>> /Volumes/Data/test/last2muted.mov -y
>> 
>> This gives me exactly the same issue.
> 
> Huh, have you tried switching the order of the input files? I think it’s 
> possible ffmpeg doesn’t bother to check for possibly more channels being used 
> in the filters after guessing stereo from the first input since it’s never 
> actually told otherwise about channels or layouts...
>> Guessed Channel Layout for Input Stream #0.1 : stereo
> 
>> Now, I can live with all the sound channels in one track, only reason I’m 
>> pursuing this is that I can’t remap more than about 8 channels into a single 
>> poly track.
>> (See my post from a few days ago…)
> 
> 
> Actually could using something other than pan sidestep the whole disabling 
> output streams issue?
> 
> % ffmpeg -i /Volumes/Data/test/C0092.MP4 -i /Volumes/Data/test/068.WAV 
> -filter_complex 
> “[0:1]pan=1c|c0=c1[o1];[1:0]pan=1c|c0=c9[o2];[1:0]pan=1c|c0=c10[o3]” -map 
> “[o1]" -map “[o2]" -map “[o3]"
> -c:a pcm_s24le -map 0:0 -c:v copy -shortest /Volumes/Data/test/last2muted.mov 
> -y

Same muted tracks, but now I know how to fix them:

Now for the main course:
> % ffmpeg -i /Volumes/Data/test/068.WAV -i /Volumes/Data/test/C0092.MP4 
> -filter_complex 
> "[0:a]channelmap=map=0-0|1-1:channel_layout=0x6000,[1:a]join=inputs=2:channel_layout=0x6002,channelmap=map=0-0|1-1|2-2:channel_layout=0x6002[out]"
>  -map "[out]" -c:a pcm_s24le -map 0:0 -c:v copy -shortest -y output.mov

This looks promising, but I can’t grasp the filter_complex:
If I break it down:
> "[0:a]
Use sound from input 0

> channelmap=map=0-0|1-1
map channel 0 to output 0, channel 1 to output 1

> :channel_layout=0x6000,
I take it this is means the binary value, in this case 0 1 1 0 0 0 0 0 0 etc 
But this is output? (It looks like ‘max 32 channels out, use 1 and 2, leave 
room to insert the sound from the video at 0? And the mapping auto inserts the 
tracks to the first available slots? (Or how it is called…))

> [1:a]join=inputs=2
Use sound from input 1, start joining that, so total 2 streams input.

> :channel_layout=0x6002,
Now I’m totally lost. This states 0 1 1 0 0 0 0 0 0 0 0 0 0 0 1 0 0 0 0 0 0 0 0 
0 0 0 0 0 0 0 0 0
What am I supposed to see here? Start is the same, but then I’m lost...

> channelmap=map=0-0|1-1|2-2:
This also confuses me. The order of the second and third mapping looks likes it 
comes from the 0 1 1 yadda at the beginning.
But where is channel 0 from the second input mapped? Whatever I try, I always 
end up with channel 0...

> channel_layout=0x6002[out]”


Thanks!

Bouke


> 
> Regards,
> Ted Park
> 
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Re: [FFmpeg-user] more sound madness

2020-03-08 Thread Bouke / VideoToolShed

> On 07 Mar 2020, at 18:05, Ted Park  wrote:
> 
> Hi,
> 
>> Hi Ted,
>> Sorry, I have no clue what you mean here, could you please elaborate?
> I am not sure if this is accessible with ffmpeg, but in the track header, 
> usually only tracks with different alternate_group, or 0, can play at the 
> same time.
> 
>> Nope, moving the  does not make any difference.
>> Did you try this yourself? Does it work for you?
> I mean, yes, but it works either way. But the source I used was a single mxf 
> with multiple tracks, I didn’t try multiple inputs or a bwf.

Hi Ted,
Did you try it in Premiere, and got all the channels? Or open the output in old 
QT pro, and look at the properties.
The last channels are disabled. If I enable them and save the file 
(destructive), a new MOOV atom is added and the current one is flagged as FREE.
Then Premiere will accept it as it ‘should’ be.
(Warning, Premiere has  a nasty habit to ‘remember’ files. Once it has seen a 
specific file, it will use the properties of the first import, even when you 
change it and re-import.)

> 
 Stream mapping:
 Stream #0:0 -> #0:0 (copy)
 Stream #0:1 -> #0:1 (pcm_s16be (native) -> pcm_s24le (native))
 Stream #1:0 -> #0:2 (pcm_s24le (native) -> pcm_s24le (native))
 Stream #1:0 -> #0:3 (pcm_s24le (native) -> pcm_s24le (native))
 Press [q] to stop, [?] for help
 -map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c1.
 [pan @ 0x7fb020e00a00] Pure channel mapping detected: 1
 -map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c8.
 [pan @ 0x7fb020d04740] Pure channel mapping detected: 8
 -map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c9.
 [pan @ 0x7fb020f1d200] Pure channel mapping detected: 9
>>> It shouldn’t matter if audio channels come before others.
>> trust me, it does matter. First input file gets the audio enabled, all other 
>> gets muted in metadata.
> Oh right… But I mean they don’t matter if your tracks aren’t randomly 
> disabled. I still would have thought at least one would get chosen.
> When you say muted in metadata, do you literally mean they have their 
> relative volume set to 0 per track?
> Either way, I’m pretty sure it is because some channels overlap between the 
> two inputs, I think this filter is what is actually being attempted, give it 
> a try.
> % ffmpeg -i /Volumes/Data/test/C0092.MP4 -i /Volumes/Data/test/068.WAV 
> -filter_complex 
> “[0:1]pan=1c|c0=c1[o1];[1:0]pan=1c|c0=c9[o2];[1:0]pan=1c|c0=c10[o3]” -map 
> “[o1]" -map “[o2]" -map “[o3]"
> -c:a pcm_s24le -map 0:0 -c:v copy -shortest /Volumes/Data/test/last2muted.mov 
> -y

This gives me exactly the same issue.

Now, I can live with all the sound channels in one track, only reason I’m 
pursuing this is that I can’t remap more than about 8 channels into a single 
poly track.
(See my post from a few days ago…)

Thanks,
Bouke

>> there is no ‘normal’ in my world, and the Mov format accepts virtually 
>> unlimited channels in virtual unlimited tracks in virtual unlimited 
>> combinations, so that ‘should’ not be an issue.
> The manual panning sets all tracks as mono, at channel 0. The mov format has 
> very little restrictions, but what is supported by software is a different 
> story.
> 
>>> or how the extra channels are mapped wrt gain. Exploding the multitrack 
>>> recording into separate wave files first might make things simpler.
>> 
>> SImpler in what way?
>> Sorry, but merging two files ‘should’ be possible IMHO, and I’m fighting on 
>> doing this for a couple of days now, so ‘simple’ is not in my book anymore.
> Yes I agree, it was just an alternative because what should be working 
> doesn’t seem to be. It’s simpler since you don’t have to worry about the 
> channels being in a known layout for one thing.
> 
> Regards,
> Ted Park
> 
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Re: [FFmpeg-user] more sound madness

2020-03-07 Thread Bouke / VideoToolShed

> On 07 Mar 2020, at 14:38, Ted Park  wrote:
> 
> Hi,
> 
>> command below seems simple enough, and works, except that the last two audio 
>> channels (those coming from the second input) are muted in the metadata.
>> (Meaning, the sound data is there, but QT atom has flagged them as 
>> ‘inactive’.)
> 
>> What is the secret handshake to remap / patch / reroute / matrix / add / 
>> omit the sound channels of multiple input files to one new file?
>> (I can’t imagine I’m the only one doing this kind of stuff. And at this 
>> point I really don’t care if all the channels end up in one track, or in 
>> multiple, just to have them play in QT / Premiere and alike.)
> The reverse of “inactive” is “default” iirc, so they aren’t in alternate 
> groups. The mov muxer seems to do that by default for me though.

Hi Ted,
Sorry, I have no clue what you mean here, could you please elaborate?

> 
>> bouke@Boukes-iMac ~ % /Applications/ffmpeg  -i /Volumes/Data/test/C0092.MP4 
>> -i /Volumes/Data/test/068.WAV -map 0:0 -map 0:a -map_channel 0.1.1:0.1  -map 
>> 1:a -map_channel 1.0.8:0.2 -map 1:a -map_channel 1.0.9:0.3  -c:a pcm_s24le 
>> -c:v copy -shortest /Volumes/Data/test/last2muted.mov -y
> 
> I think it would help to do the channel mapping first just to separate the 
> options
> % ffmpeg -i /Volumes/Data/test/C0092.MP4 -i /Volumes/Data/test/068.WAV -map 
> 0:1 -map 1:0 -map 1:0 -map_channel 0.1.1:0.1 -map_channel 1.0.8:0.2 
> -map_channel 1.0.9:0.3 -c:a pcm_s24le -map 0:0 -c:v copy -shortest 
> /Volumes/Data/test/last2muted.mov -y

Nope, moving the  does not make any difference.
Did you try this yourself? Does it work for you?

> 
>> Stream mapping:
>> Stream #0:0 -> #0:0 (copy)
>> Stream #0:1 -> #0:1 (pcm_s16be (native) -> pcm_s24le (native))
>> Stream #1:0 -> #0:2 (pcm_s24le (native) -> pcm_s24le (native))
>> Stream #1:0 -> #0:3 (pcm_s24le (native) -> pcm_s24le (native))
>> Press [q] to stop, [?] for help
>> -map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c1.
>> [pan @ 0x7fb020e00a00] Pure channel mapping detected: 1
>> -map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c8.
>> [pan @ 0x7fb020d04740] Pure channel mapping detected: 8
>> -map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c9.
>> [pan @ 0x7fb020f1d200] Pure channel mapping detected: 9
> 
> It shouldn’t matter if audio channels come before others.

trust me, it does matter. First input file gets the audio enabled, all other 
gets muted in metadata.

> Also I am kind of suspicious of this, I don’t know if using 4 channels is 
> normal in this situation,

there is no ‘normal’ in my world, and the Mov format accepts virtually 
unlimited channels in virtual unlimited tracks in virtual unlimited 
combinations, so that ‘should’ not be an issue.

> or how the extra channels are mapped wrt gain. Exploding the multitrack 
> recording into separate wave files first might make things simpler.

SImpler in what way?
Sorry, but merging two files ‘should’ be possible IMHO, and I’m fighting on 
doing this for a couple of days now, so ‘simple’ is not in my book anymore.

Bouke


> Regards,
> Ted Park



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[FFmpeg-user] more sound madness

2020-03-06 Thread Bouke / VideoToolShed
Hi all,
command below seems simple enough, and works, except that the last two audio 
channels (those coming from the second input) are muted in the metadata.
(Meaning, the sound data is there, but QT atom has flagged them as ‘inactive’.)

From memory, Carl once said when I had similar problems: ‘this is a known 
regression’, but that was ages ago.

What is the secret handshake to remap / patch / reroute / matrix / add / omit 
the sound channels of multiple input files to one new file?
(I can’t imagine I’m the only one doing this kind of stuff. And at this point I 
really don’t care if all the channels end up in one track, or in multiple, just 
to have them play in QT / Premiere and alike.)

Thx,

Bouke

Last login: Fri Mar  6 11:51:28 on ttys001
bouke@Boukes-iMac ~ % /Applications/ffmpeg  -i /Volumes/Data/test/C0092.MP4 -i 
/Volumes/Data/test/068.WAV -map 0:0 -map 0:a -map_channel 0.1.1:0.1  -map 1:a 
-map_channel 1.0.8:0.2 -map 1:a -map_channel 1.0.9:0.3  -c:a pcm_s24le -c:v 
copy -shortest /Volumes/Data/test/last2muted.mov -y

ffmpeg version git-2020-02-03-1c15111 Copyright (c) 2000-2020 the FFmpeg 
developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
  configuration: --enable-gpl --enable-version3 --enable-sdl2 
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass 
--enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 
--enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab 
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex 
--enable-libxvid --enable-libaom --enable-appkit --enable-avfoundation 
--enable-coreimage --enable-audiotoolbox
  libavutil  56. 38.100 / 56. 38.100
  libavcodec 58. 67.100 / 58. 67.100
  libavformat58. 37.100 / 58. 37.100
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 73.100 /  7. 73.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb022001200] st: 0 edit list: 1 Missing key frame 
while searching for timestamp: 1000
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb022001200] st: 0 edit list 1 Cannot find an 
index entry before timestamp: 1000.
Guessed Channel Layout for Input Stream #0.1 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/Data/test/C0092.MP4':
  Metadata:
major_brand : XAVC
minor_version   : 16785407
compatible_brands: XAVCmp42iso2
creation_time   : 2020-01-30T13:31:25.00Z
  Duration: 00:00:02.40, start: 0.00, bitrate: 97881 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, 
bt709), 3840x2160 [SAR 1:1 DAR 16:9], 82959 kb/s, 25 fps, 25 tbr, 25k tbn, 50 
tbc (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z
  handler_name: Video Media Handler
  encoder : AVC Coding
Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, stereo, 
s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z
  handler_name: Sound Media Handler
Stream #0:2(und): Data: none (rtmd / 0x646D7472), 204 kb/s (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z
  handler_name: Timed Metadata Media Handler
  timecode: 01:42:38:10
Input #1, wav, from '/Volumes/Data/test/068.WAV':
  Metadata:
comment : sSPEED=025.000-ND 
: sTAKE=068 
: sUBITS=$ 
: sSWVER=5.01.8149 
: sSCENE=MixPre 
: sFILENAME=MixPre-068.WAV 
: sTAPE=200130 
: sCIRCLED=FALSE 
: sTRK1=MixL 
: sTRK2=MixR 
: sTRK3=Track 1 
: sTRK4=Track 2 
: sTRK5=Track 3 
: sTRK6=Track 4 
: sTRK7=Track 5 
: sTRK8=Track 6 
: sTRK9
encoded_by  : SoundDev: MixPre-10 II TJ0019225
originator_reference: USSDVTJ001922501520013072A000201
date: 2020-01-30
creation_time   : 07:02:20
time_reference  : 2468718515
coding_history  : A=PCM,F=48000,W=24,M=multi,R=48000,T=10 Ch;Ambisonics=off 
: 
  Duration: 00:49:41.90, bitrate: 11520 kb/s
Stream #1:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 10 
channels, s32 (24 bit), 11520 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (pcm_s16be (native) -> pcm_s24le (native))
  Stream #1:0 -> #0:2 (pcm_s24le (native) -> pcm_s24le (native))
  Stream #1:0 -> #0:3 (pcm_s24le 

Re: [FFmpeg-user] Channel mapping misery

2020-03-04 Thread Bouke / VideoToolShed


> On 04 Mar 2020, at 12:06, Carl Eugen Hoyos  wrote:
> 
> Am Mi., 4. März 2020 um 12:02 Uhr schrieb Bouke / VideoToolShed
> :
> 
>> When I add one channel more to the previous command: (the wave file has 10 
>> channels)
>> 
>> -threads 0 -i /Volumes/Data/test/C0092.MP4 -ss 1601.809541692 -i 
>> /Volumes/Data/test/068.WAV -map 0:0 -filter_complex 
>> [0:1][1:0][1:0][1:0][1:0][1:0][1:0]amerge=inputs=7,pan=7c|c0=c0|c1=c2|c2=c3|c3=c4|c4=c5|c5=c6|c6=c7[a]
>>  -map [a] -c:a pcm_s24le -timecode 01:42:38:10 -metadata:s:v:0 
>> reel_name=C0092 -c:v copy /Volumes/Data/test/C0092.mov -y
>> 
>> This does show the pure channel mapping, but fails with:
> 
> Is it possible that you provided the complete, uncut console output
> for the working command line
> but not for the one that fails for you?

Ok, again from the console: 
(This obviously works)

bouke@Boukes-iMac ~ % /Applications/ffmpeg -threads 0 -i 
/Volumes/Data/test/C0092.MP4 -ss 1601.809541692 -i 
/Volumes/Data/test/068.WAV -map 0:0 -filter_complex 
"[0:1][1:0][1:0][1:0][1:0][1:0]amerge=inputs=6,pan=6c|c0=c0|c1=c2|c2=c3|c3=c4|c4=c5|c5=c6[a]"
 -map "[a]" -c:a pcm_s24le -timecode 01:42:38:10 -metadata:s:v:0 
reel_name=C0092 -c:v copy /Volumes/Data/test/C0092.mov -y
ffmpeg version git-2020-02-03-1c15111 Copyright (c) 2000-2020 the FFmpeg 
developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
  configuration: --enable-gpl --enable-version3 --enable-sdl2 
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass 
--enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 
--enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab 
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex 
--enable-libxvid --enable-libaom --enable-appkit --enable-avfoundation 
--enable-coreimage --enable-audiotoolbox
  libavutil  56. 38.100 / 56. 38.100
  libavcodec 58. 67.100 / 58. 67.100
  libavformat58. 37.100 / 58. 37.100
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 73.100 /  7. 73.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fddf6004e00] st: 0 edit list: 1 Missing key frame 
while searching for timestamp: 1000
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fddf6004e00] st: 0 edit list 1 Cannot find an 
index entry before timestamp: 1000.
Guessed Channel Layout for Input Stream #0.1 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/Data/test/C0092.MP4':
  Metadata:
major_brand : XAVC
minor_version   : 16785407
compatible_brands: XAVCmp42iso2
creation_time   : 2020-01-30T13:31:25.00Z
  Duration: 00:00:02.40, start: 0.00, bitrate: 97881 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, 
bt709), 3840x2160 [SAR 1:1 DAR 16:9], 82959 kb/s, 25 fps, 25 tbr, 25k tbn, 50 
tbc (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z
  handler_name: Video Media Handler
  encoder : AVC Coding
Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, stereo, 
s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z
  handler_name: Sound Media Handler
Stream #0:2(und): Data: none (rtmd / 0x646D7472), 204 kb/s (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z
  handler_name: Timed Metadata Media Handler
  timecode: 01:42:38:10
Input #1, wav, from '/Volumes/Data/test/068.WAV':
  Metadata:
comment : sSPEED=025.000-ND 
: sTAKE=068 
: sUBITS=$ 
: sSWVER=5.01.8149 
: sSCENE=MixPre 
: sFILENAME=MixPre-068.WAV 
: sTAPE=200130 
: sCIRCLED=FALSE 
: sTRK1=MixL 
: sTRK2=MixR 
: sTRK3=Track 1 
: sTRK4=Track 2 
: sTRK5=Track 3 
: sTRK6=Track 4 
: sTRK7=Track 5 
: sTRK8=Track 6 
: sTRK9
encoded_by  : SoundDev: MixPre-10 II TJ0019225
originator_reference: USSDVTJ001922501520013072A000201
date: 2020-01-30
creation_time   : 07:02:20
time_reference  : 2468718515
coding_history  : A=PCM,F=48000,W=24,M=multi,R=48000,T=10 Ch;Ambisonics=off 
: 
  Duration: 00:49:41.90, bitrate: 11520 kb/s
Stream #1:0

Re: [FFmpeg-user] Channel mapping misery

2020-03-04 Thread Bouke / VideoToolShed


> On 04 Mar 2020, at 12:06, Carl Eugen Hoyos  wrote:
> 
> Am Mi., 4. März 2020 um 12:02 Uhr schrieb Bouke / VideoToolShed
> :
> 
>> When I add one channel more to the previous command: (the wave file has 10 
>> channels)
>> 
>> -threads 0 -i /Volumes/Data/test/C0092.MP4 -ss 1601.809541692 -i 
>> /Volumes/Data/test/068.WAV -map 0:0 -filter_complex 
>> [0:1][1:0][1:0][1:0][1:0][1:0][1:0]amerge=inputs=7,pan=7c|c0=c0|c1=c2|c2=c3|c3=c4|c4=c5|c5=c6|c6=c7[a]
>>  -map [a] -c:a pcm_s24le -timecode 01:42:38:10 -metadata:s:v:0 
>> reel_name=C0092 -c:v copy /Volumes/Data/test/C0092.mov -y
>> 
>> This does show the pure channel mapping, but fails with:
> 
> Is it possible that you provided the complete, uncut console output
> for the working command line
> but not for the one that fails for you?

Hi Carl,
I did, it is at the bottom.
(Well, it’s not really console, it’s what Python returns. I can’t get it to run 
from the console. I’ll try again though.)

Bouke

> 
> Carl Eugen
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[FFmpeg-user] Channel mapping misery

2020-03-04 Thread Bouke / VideoToolShed
Hi all,
Trying to do create files where I can control the audio channels. (From 
multiple, different sources.)

I’ve got it ‘sorta kinda’ working:

-threads 0 -i /Volumes/Data/test/C0092.MP4 -ss 1601.809541692 -i 
/Volumes/Data/test/068.WAV -map 0:0 -filter_complex 
[0:1][1:0][1:0][1:0][1:0][1:0]amerge=inputs=6,pan=6c|c0=c0|c1=c2|c2=c3|c3=c4|c4=c5|c5=c6[a]
 -map [a] -c:a pcm_s24le -timecode 01:42:38:10 -metadata:s:v:0 reel_name=C0092 
-c:v copy /Volumes/Data/test/C0092.mov 
(Full output at the bottom)
This works, although Wiki says ‘With amerge all inputs must have the same 
sample rate and format.’
But that does not seem to be the case… (In the docs in the -map_channel section 
it’s encouraged to use amerge, since -map_channel cannot do conversions…)
So, I don’t think my problem lies there.

When I add one channel more to the previous command: (the wave file has 10 
channels)

-threads 0 -i /Volumes/Data/test/C0092.MP4 -ss 1601.809541692 -i 
/Volumes/Data/test/068.WAV -map 0:0 -filter_complex 
[0:1][1:0][1:0][1:0][1:0][1:0][1:0]amerge=inputs=7,pan=7c|c0=c0|c1=c2|c2=c3|c3=c4|c4=c5|c5=c6|c6=c7[a]
 -map [a] -c:a pcm_s24le -timecode 01:42:38:10 -metadata:s:v:0 reel_name=C0092 
-c:v copy /Volumes/Data/test/C0092.mov -y 

This does show the pure channel mapping, but fails with:

[Parsed_amerge_0 @ 0x7facfcc28b80] No channel layout for input 1
[Parsed_amerge_0 @ 0x7facfcc28b80] Input channel layouts overlap: output layout 
will be determined by the number of distinct input channels
Assertion outlink->channel_layout == out_layout || !outlink->channel_layout 
failed at src/libavfilter/af_aresample.c:168

Then, adding yet another channel, the error message changes:
-threads 0 -i /Volumes/Data/test/C0092.MP4 -ss 1601.809541692 -i 
/Volumes/Data/test/068.WAV -map 0:0 -filter_complex 
[0:1][1:0][1:0][1:0][1:0][1:0][1:0][1:0]amerge=inputs=8,pan=8c|c0=c0|c1=c2|c2=c3|c3=c4|c4=c5|c5=c6|c6=c7|c7=c8[a]
 -map [a] -c:a pcm_s24le -timecode 01:42:38:10 -metadata:s:v:0 reel_name=C0092 
-c:v copy /Volumes/Data/test/C0092.mov -y 


[Parsed_amerge_0 @ 0x7faa63f0] No channel layout for input 1
[Parsed_amerge_0 @ 0x7faa63f0] Too many channels (max 64)
[Parsed_amerge_0 @ 0x7faa63f0] Query format failed for 'Parsed_amerge_0': 
Invalid argument

Error reinitializing filters!
Failed to inject frame into filter network: Invalid argument
Error while processing the decoded data for stream #1:0
Conversion failed!


What am I missing here / what is the correct approach?

Thanks,
Bouke

Working output:

ffmpeg version git-2020-02-03-1c15111 Copyright (c) 2000-2020 the FFmpeg 
developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
  configuration: --enable-gpl --enable-version3 --enable-sdl2 
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass 
--enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 
--enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab 
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex 
--enable-libxvid --enable-libaom --enable-appkit --enable-avfoundation 
--enable-coreimage --enable-audiotoolbox
  libavutil  56. 38.100 / 56. 38.100
  libavcodec 58. 67.100 / 58. 67.100
  libavformat58. 37.100 / 58. 37.100
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 73.100 /  7. 73.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100

[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb37180] st: 0 edit list: 1 Missing key frame 
while searching for timestamp: 1000
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fb37180] st: 0 edit list 1 Cannot find an 
index entry before timestamp: 1000.

Guessed Channel Layout for Input Stream #0.1 : stereo
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/Data/test/C0092.MP4':
  Metadata:
major_brand : XAVC
minor_version   : 16785407
compatible_brands: XAVCmp42iso2
creation_time   : 2020-01-30T13:31:25.00Z
  Duration: 00:00:02.40, start: 0.00, bitrate: 97881 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, 
bt709), 3840x2160 [SAR 1:1 DAR 16:9], 82959 kb/s, 25 fps, 25 tbr, 25k tbn, 50 
tbc (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z

  handler_name: Video Media Handler
  encoder : AVC Coding
Stream #0:1(und): Audio: pcm_s16be (twos / 0x736F7774), 48000 Hz, stereo, 
s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2020-01-30T13:31:25.00Z
  handler_name: Sound Media Handler
Stream #0:2(und): Data: none (rtmd / 0x646D7472), 204 kb/s (default)
Metadata:
  creation_time   : 

[FFmpeg-user] Strange TC on 50 FPS clips

2020-02-03 Thread Bouke / VideoToolShed
Hi all,
Getting strange TC readings on 50 FPS clips.
FFmpeg reports  timecode: 11:26:32:42
(And that corresponds with the frame counter in the start of the mdat atom.)

Premiere  reports   22:53:05:34
 QT 7 and Avid (older version)  22:53:05:17

Now Avid and QT have the habit of stepping trough the video by two frames if 
fps > 30, so that might be a hint.
(If I take FFmpegs TC to frames, (2059642) multiply that by 2 I’m at the 
‘correct’ frame number, converted back to TC it gives 22:53:05:34)

Now, is this a bug or am I missing something?

(Setting TC with FFmpeg has no issues as far as I’ve seen.)

If interested, the clip can be found here:
Download link 
https://we.tl/t-CP6GlfQIeG 
(Turn your volume down if you want to play it, it has loud LTC on the right 
channel.)

Bouke

bouke@Boukes-iMac ~ % /Applications/ffmpeg -i /Volumes/Samsung\ 2TB\ SSD\ -\ 
Data/Guilliamme\ 50fps/tokyoprod-e8655b/CAMC3472.MOV 
ffmpeg version git-2020-02-03-1c15111 Copyright (c) 2000-2020 the FFmpeg 
developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
  configuration: --enable-gpl --enable-version3 --enable-sdl2 
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass 
--enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 
--enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab 
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex 
--enable-libxvid --enable-libaom --enable-appkit --enable-avfoundation 
--enable-coreimage --enable-audiotoolbox
  libavutil  56. 38.100 / 56. 38.100
  libavcodec 58. 67.100 / 58. 67.100
  libavformat58. 37.100 / 58. 37.100
  libavdevice58.  9.103 / 58.  9.103
  libavfilter 7. 73.100 /  7. 73.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
  libpostproc55.  6.100 / 55.  6.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Volumes/Samsung 2TB SSD - 
Data/clp/tokyoprod-e8655b/CAMC3472.MOV':
  Metadata:
major_brand : qt  
minor_version   : 538247680
compatible_brands: qt  CAEP
com.apple.quicktime.make: Canon
com.apple.quicktime.model: Canon EOS-1D X Mark II
com.apple.quicktime.rating.user: 0.00
com.apple.quicktime.copyright: 
com.apple.quicktime.author: 
creation_time   : 2020-02-01T15:52:40.00Z
  Duration: 00:00:09.68, start: 0.00, bitrate: 843798 kb/s
Stream #0:0(eng): Video: mjpeg (Baseline) (jpeg / 0x6765706A), yuvj422p(pc, 
bt470bg/bt709/bt709), 4096x2160, 841993 kb/s, 50 fps, 50 tbr, 50k tbn, 50k tbc 
(default)
Metadata:
  creation_time   : 2020-02-01T15:52:40.00Z
  encoder : Photo - JPEG
  timecode: 11:26:32:42
Stream #0:1(eng): Audio: pcm_s16le (sowt / 0x74776F73), 48000 Hz, stereo, 
s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2020-02-01T15:52:40.00Z
  timecode: 11:26:32:42
Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 1 kb/s (default)
Metadata:
  creation_time   : 2020-02-01T15:52:40.00Z
  timecode: 11:26:32:42
At least one output file must be specified


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Re: [FFmpeg-user] FFplay sync ext

2019-11-24 Thread Bouke / VideoToolShed

> On 23 Nov 2019, at 17:06, Carl Zwanzig  wrote:
> 
> On 11/23/2019 7:41 AM, Bouke / VideoToolShed wrote:
>> Why do you think frame accuracy is not possible?
> 
> Because you didn't specify the time data available and "clock time" usually 
> means accuracy in seconds.

This makes no sense whatsoever.
I can provide  the time sample accurate. (Where a sample is 1 / 48000 if 
needed.)
I asked how to input ‘sync’ data, but have gotten no answer to that.

And,for other operations FFmpeg takes a float as input, and that sub-second 
accurate.

Bouke

> 
> Later,
> 
> z!
> 
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Re: [FFmpeg-user] FFplay sync ext

2019-11-23 Thread Bouke / VideoToolShed


> On 23 Nov 2019, at 16:31, Carl Zwanzig  wrote:
> 
> On 11/23/2019 3:48 AM, Bouke / VideoToolShed wrote:
>> Seeking a way to slave video playback to an external clock.
>> (Meaning, jump to specific position and restart playing if the difference 
>> between current playback time and external clock is too big.)
> 
> How big is "too big"? A second? Minute? I don't think you're going to get 
> frame-accuracy.
> 


well, subframe accuracy if possible. I have time in samples from the audio,
Why do you think frame accuracy is not possible?


> What do you mean by "playback time" in this context? (The problem may sound 
> simple, but we're left to make a lot of guesses about what you want to do; 
> please fill in some details.)

I meant, if the player shows a wrong playback time, what can I do then?

> 
> 
>> Or is FFplay not the tool for this?
> 
> It may not be. mplayer has a "slave mode" which takes commands, including 
> seek-to-time (in seconds). So does CasparCG, but that's probably overkill for 
> a simple player. Remember that any of these will simply jump to that location 
> in the file and may glitch the picture until a key frame comes along.
> 
> Later,
> 
> z!
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[FFmpeg-user] FFplay sync ext

2019-11-23 Thread Bouke / VideoToolShed
Hi all,
Seeking a way to slave video playback to an external clock.
(Meaning, jump to specific position and restart playing if the difference 
between current playback time and external clock is too big.)
Now, FFplay seems pretty simple, but I cannot find anything to control it 
besides the keyboard. Sending keystrokes seems awkward…

There is a -sync ext option, when I use that the video plays, and I notice a 
very small speed change in playback.
So what is ‘ext’ ?
Can I input something while playing? Where, what?

Or is FFplay not the tool for this?
(Using python / QT, any other options?)


Bouke

www.VideoToolShed.com
van Oldenbarneveltstraat 33
6512 AS  Nijmegen
+31 6 21817248
To send files, go here:
https://videotoolshed.wetransfer.com/


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Re: [FFmpeg-user] Adding macrovision style copy protection to a theora video

2019-10-19 Thread Bouke / VideoToolShed
On 19 Oct 2019, at 15:28, Carl Zwanzig  wrote:
> 
> And, of course, nothing will stop a user from aiming their camera at the 
> screen.


This says it all. 
(I don’t like +1)

Bouke
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Re: [FFmpeg-user] Where to cut audio by video frame?

2019-10-10 Thread Bouke / VideoToolShed



> On 10 Oct 2019, at 18:35, Felix Muster via ffmpeg-user 
>  wrote:
> 
> Hello
> 
> 
> 
> I want to cut the audio from a scene of a movie (blu-ray 23.976 pfs).
> 
> I want to create the cut from frame 2928 to 5177.
> 
> Currently I don't know where I have to cut the audio to be accurate.
> 
> Is the given command correct?
> 
Depending on your way of calculating,, the dividing might / might not be needed.
-ss is for the start of what you want, the -t stands for the duration of the 
file you want as output.

Give it a go a couple of times, you’ll see. (It won’t hurt.)

Bouke




> 
> ffmpeg.exe -i audio.flac -ss (2928/(24000/1001)) -t
> (5177/(24000/1001))-(2928/(24000/1001)) -codec flac audio_cut.flac
> 
> ffmpeg.exe -i audio.flac -ss 122.122 -t 93.80204167 -codec flac
> audio_cut.flac
> 
> 
> 
> And from the documentation I should use -ss as an output option to be
> accurate. But what is with -t? As input or output?
> 
> 
> 
> Best
> 
> Felix
> 
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Re: [FFmpeg-user] Licensing ffmpeg

2019-09-23 Thread Bouke / VideoToolShed

> On 23 Sep 2019, at 11:58, Henning Thielemann  
> wrote:
> 
> 
> On Mon, 23 Sep 2019, kushal ghosh wrote:
> 
>> "Use it commercially" means to use the binary as a part of an commercial
>> application. So the takeaway is the compile it without those flags. Thanks
>> for these pointers. I had gone through the Legal page and wanted a second
>> opinion from experts here as I am not professional legally qualified to
>> make best sens of it.
> 
> You may also ship your application with its source code and thus meet the 
> requirements of the GPL.


For the archives:
Do note that you perhaps meet the requirements of the GPL, you still may have 
to deal with other licenses if there is non-free stuff involved.
YMMV (IMHO)

Bouke


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Re: [FFmpeg-user] Windows build starts slow on encoding

2019-04-23 Thread Bouke / VideoToolShed


> On 15 Apr 2019, at 16:36, Moritz Barsnick  wrote:
> 
> On Mon, Apr 15, 2019 at 13:05:08 +0200, Bouke / VideoToolShed wrote:
>> Latest Zeranoe builds (Windows) have a very long delay before starting 
>> transcoding.
>> It seems not to be related on in/output codecs
> 
> Since Zeranoe provides older versions: Can you pin-point with which
> version this changed? And by how much? (E.g. find the version where it
> changed, and post "before/after" logs.)
> 
> I haven't observed this and can't reproduce, so I'd like to a) rule out
> something changed on your side, and b) understand which change may have
> introduced this.

Follow up:
I started testing last sunday, and for the life of me I could not reproduce the 
issue.
(My Windows did do an update that morning.)
Today the problem was back, and I narrowed it down to running FFmpeg.exe from 
(a subfolder) from the desktop.
I am totally clueless how this can happen. (The previous tests  from last 
sunday where the problem did not occur also was from the desktop.)
(Again, it’s a delay introduced before ANY transcoding, but no delay in 
starting FFmpeg itself.)
Same FFmpeg version run from another location is fine. 

Very, very strange.

Bouke



> 
> Thanks,
> Moritz
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Re: [FFmpeg-user] Windows build starts slow on encoding

2019-04-15 Thread Bouke / VideoToolShed

> On 15 Apr 2019, at 13:06, Carl Eugen Hoyos  wrote:
> 
> 2019-04-15 13:05 GMT+02:00, Bouke / VideoToolShed :
>> Latest Zeranoe builds (Windows) have a very long delay
>> before starting transcoding.
> 
> This sentence would indicate that you posted on the wrong
> mailing list.

Perhaps, but I can’t compile myself sadly.
I’ll see if I can get help somewhere else.


> Maybe your issue is that reading mxf files takes longer
> than in the past?

Nope, it happens on all input files.

Bouke

> 
> Carl Eugen
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[FFmpeg-user] Windows build starts slow on encoding

2019-04-15 Thread Bouke / VideoToolShed
Latest Zeranoe builds (Windows) have a very long delay before starting 
transcoding.
It seems not to be related on in/output codecs
-c:v copy is fast as usual, as is just getting metadata (omitting output), but 
anything else has a delay of about 10 seconds (at least) between spitting out 
the initial setup info and the actual start of encoding.

Below is a sample output, but again, it’s not related to the codecs afaik.

Thx,

Bouke

ffmpeg version N-93569-gf9840cd2b5 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 8.3.1 (GCC) 20190414
  configuration: --enable-gpl --enable-version3 --enable-sdl2 
--enable-fontconfig --enable-gnutls --enable-iconv --enable-libass 
--enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack 
--enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 
--enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab 
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex 
--enable-libxvid --enable-libaom --enable-libmfx --enable-amf 
--enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc 
--enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  libavutil  56. 26.100 / 56. 26.100
  libavcodec 58. 51.100 / 58. 51.100
  libavformat58. 27.102 / 58. 27.102
  libavdevice58.  7.100 / 58.  7.100
  libavfilter 7. 48.100 /  7. 48.100
  libswscale  5.  4.100 /  5.  4.100
  libswresample   3.  4.100 /  3.  4.100
  libpostproc55.  4.100 / 55.  4.100
Guessed Channel Layout for Input Stream #0.1 : mono
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Guessed Channel Layout for Input Stream #0.4 : mono
Input #0, mxf, from 'X:\Cam examples\CanonXF705\A003C032H1801230A_CANON.MXF':
  Metadata:
operational_pattern_ul: 060e2b34.04010101.0d010201.01010900
uid : 4ca2d441-8105-4901-802f-ae3689900021
generation_uid  : 4ca2d441-8105-4903-802f-ae3689900021
company_name: CANON
product_name: XF705
product_version : 1.00
product_uid : 060e2b34-0401-010d-0e15-005658460400
modification_date: 2018-01-23T23:45:08.00Z
material_package_umid: 
0x060A2B340101010501010D431300CD7E165541810580853689900021
timecode: 00:25:16;23
  Duration: 00:00:32.03, start: 0.00, bitrate: 156207 kb/s
Stream #0:0: Video: hevc (Rext), yuv422p10le(pc, bt709/bt709/unknown, 
progressive), 3840x2160 [SAR 1:1 DAR 16:9], 29.97 fps, 29.97 tbr, 29.97 tbn, 
29.97 tbc
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300CD7E165441810580853689900021
Stream #0:1: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300CD7E165441810580853689900021
Stream #0:2: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300CD7E165441810580853689900021
Stream #0:3: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300CD7E165441810580853689900021
Stream #0:4: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300CD7E165441810580853689900021
Stream mapping:
  Stream #0:0 -> #0:0 (hevc (native) -> prores (native))
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=0 fps=0.0 q=0.0 size=   0kB time=-577014:32:22.77 bitrate=  
-0.0kbits/s speed=N/A
Output #0, mov, to 'X:\Cam examples\CanonXF705\A003C032H1801230A_CANON.mov':
  Metadata:
operational_pattern_ul: 060e2b34.04010101.0d010201.01010900
uid : 4ca2d441-8105-4901-802f-ae3689900021
generation_uid  : 4ca2d441-8105-4903-802f-ae3689900021
company_name: CANON
product_name: XF705
product_version : 1.00
product_uid : 060e2b34-0401-010d-0e15-005658460400
modification_date: 2018-01-23T23:45:08.00Z
material_package_umid: 
0x060A2B340101010501010D431300CD7E165541810580853689900021
timecode: 00:25:16;23
encoder : Lavf58.27.102
Stream #0:0: Video: prores (Proxy) (apco / 0x6F637061), yuv422p10le, 
320x240 [SAR 4:3 DAR 16:9], q=2-31, 200 kb/s, 0.03 fps, 30k tbn, 29.97 tbc
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300CD7E165441810580853689900021
  encoder : Lavc58.51.100 prores
Stream #0:1: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 
bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300CD7E165441810580853689900021
frame=1 fps=0.0 

Re: [FFmpeg-user] Unable to identify flicker video

2019-04-11 Thread Bouke / VideoToolShed

> On 10 Apr 2019, at 12:40, Carl Eugen Hoyos  wrote:
> 
> 2019-04-10 9:04 GMT+02:00, Bouke / VideoToolShed :
> 
>> The OP may be referring to the Harding test.
> 
> This would be ticket #2104 with a link to a possible future
> video filter.
> 

My gut feeling tells me that a custom scene detect setting will do the trick 
just fine.
but a -scene_detect=harding preset would be nice :-)


Bouke

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Re: [FFmpeg-user] Unable to identify flicker video

2019-04-10 Thread Bouke / VideoToolShed


> On 09 Apr 2019, at 21:06, Ted Park  wrote:
> 
> 
> 
>> On Apr 9, 2019, at 2:34 AM, saurav suman  wrote:
>> 
>> Hi Team,
>> 
>> I have downloaded the latest FFmpeg.exe file from the FFmpeg website. The
>> problem is FFmpeg is unable to identify the flickers present in the video.
>> Flicker present in the video is clearly visible by the naked eye.
>> 
>> If required, I can share the recorded file.
>> 
>> Regards,
>> Saurav
> 
> Can you elaborate on what a flicker is, exactly? Also what you mean by 
> identifying them.

The OP may be referring to the Harding test.
(There was a Simpons episode situated in Japan where everyone but Homer 
suffered from the consequences.)
https://en.wikipedia.org/wiki/Harding_test 


A testfile that fails the test is here:
Download link 
https://videotoolshed.wetransfer.com/downloads/1f40e9977b12f800e4062e939d7178e720190410070204/3ac8160692f3d57b88b7c430987e269720190410070204/077992
 


(Sadly, i have no ide where it fails, it is not per definition the strobe shots 
at the end.)

Bouke


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Re: [FFmpeg-user] Growing file processing

2019-03-10 Thread Bouke / VideoToolShed

> On 02 Mar 2019, at 18:39, Moritz Barsnick  wrote:
> 
> On Sat, Mar 02, 2019 at 11:02:03 +0100, Bouke / VideoToolShed wrote:
>> Next, it is dog slow, sorta kinda defeating the whole idea.
> 
> I thought the whole point was that the "receiving" ffmpeg was too fast,
> in relation to the processing "speed" of the sending ffmpeg. And that
> the "receiving" side needed a chance not to exit on EOF.

Hi Moritz,
Sorry for the late reply, just after testing this my laptop (where I performed 
the tests and have my e-mail) crashed, and my backup mail machine decided not 
to play anymore also,
I reverted to webmail, and missed some stuff here and there. (One of my 
accounts bounces a lot of mails from the FF list for some reason.)

I understood the original idea, and my tests as done were based on an actual 
existing input file, to see what happened with speed.

> The tail "pipe" shouldn't slow anything down, except that there's the
> disk R/W I/O involved.

Thus, sadly, not on my system...

> There could be a minor issue in "tail" processing in text mode,
> buffered line by line. You will find references online to a "bintail",
> but I am otherwise at a loss of tools which can ignore EOF. Named pipes
> as am alternative probably also transfer EOF.

That could have been my issue, I’ve only toyed with ‘tail’ 

> 
>> (Since the data is piped at a speed that FFmpeg will never be able to
>> catch up with the original…)
> 
> Huh? Gregor wrote:
>> 2 - Growing file generation slower than processing
>> ffmpeg.exe -i input.mxf growing.mkv
>> ffmpeg.exe -i growing.mkv -c copy output.mkv
>>> ffmpeg stops when it reaches the current end of the growing file.   
>>> 
>>>   
> 
> So Bouke, what's to catch up? Did you misread the original poster's
> request?

Again, no, I did not, but again, the Tail output seemed slow...

> 
>> bouke$ ffmpeg -i /Users/bouke/Desktop/Judith/Harding.mxf  -an -s 120x40 -t 
>> 15  -y  /Users/bouke/Desktop/Judith/test.mp4
> 
> You may have overlooked that /Users/bouke/Desktop/Judith/Harding.mxf is
> being created at a slower speed, and not an existing finalized file.

Nope, it was finished :-)

> 
>> Stream #0:0 -> #0:0 (mpeg2video (native) -> mpeg4 (native))
>> Press [q] to stop, [?] for help
>> frame=  375 fps=136 q=2.0 Lsize= 365kB time=00:00:15.00 bitrate= 
>> 199.6kbits/s
>> video:363kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB 
>> muxing overhead: 0.674411%
>> 
>> vs
>> bouke$ tail -c +1 -F /Users/bouke/Desktop/Judith/Harding.mxf  | ffmpeg -i - 
>> -an -s 120x40 -t 15  -y  /Users/bouke/Desktop/Judith/test.mp4
>>  Stream #0:0 -> #0:0 (mpeg2video (native) -> mpeg4 (native))
>> frame=  375 fps= 28 q=2.0 Lsize= 365kB time=00:00:15.00 bitrate= 
>> 199.5kbits/s dup=1 drop=0
>> video:363kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB 
>> muxing overhead: 0.674757%
> 
> 28 vs 136 fps is certainly peculiar.


Let’s blame it on not using Bintail, but…

On 02-03-2019 09:57 PM, Marton Balint wrote:
> 
> Another approach is to use -rw_timeout 100 and -follow 1 options. 

is a very elegant solution, so no need to investigate further!

But, thanks, you’re always very helpful!

Bouke


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Re: [FFmpeg-user] How to filter VHS cassette artifacts

2018-12-08 Thread Bouke / VideoToolShed

> On 08 Dec 2018, at 11:34, Carl Eugen Hoyos  wrote:
> 
> 2018-12-07 20:33 GMT+01:00, Ulf Zibis  >:
>> Hi,
>> 
>> I have a VOB-file, originally recorded from a VHS-Kassette.
>> 
>> On the upper edge I see a black line from left to the middle.
>> 
>> Approximately 5 lines from bottom I see 2..3 other horizontally shifted
>> flickering lines.
>> 
>> Filter kerndeint removes the upper one, but not the lowers. But it does
>> not remove comb-like artifacts from vertical edges which bwdif perfectly
>> does.
>> 
>> Do one have an idea for a filter which is restricted to only alter those
>> small lined areas?
> 
> Shouldn't you just crop them?

He should as it is intended for computer display or blowing up to HD, as SD is 
an overscanned format, so about 10% (to tast it seems) should be cropped off.
(Hence you see so many mics dipping in the image, these are NOT mistakes, but 
good practice to get the mic as close as the proper framing allows.)

Bouke (who is feeling quite old now.)

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[FFmpeg-user] amid fails when there are more than 8 input channels (in one file)

2018-05-02 Thread Bouke / VideoToolShed
Hi all,
Strange thing, more than 8 channels give silent output:

ffmpeg -i 9Ch_orMore.WAV -filter_complex "[0:0][0:0] 
amix=inputs=2,pan=mono|c0=c0+c1” -ac 1 mono.wav

gives on errors whatsoever, but output is silent.

Is this a bug or user error?

thx

Bouke




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Re: [FFmpeg-user] Amerge trouble

2017-11-22 Thread Bouke / VideoToolShed


> On 22 Nov 2017, at 06:32, Gyan Doshi <gyando...@gmail.com> wrote:
> 
> 
> 
> On 11/19/2017 6:09 PM, Bouke / VideoToolShed wrote:
> 
>>> ffmpeg -i HD422.mxf -i 8ch.wav -c:v copy -c:a pcm_s24le -filter_complex \
>>> "[1:a]channelsplit=channel_layout=7.1[a0][a1][a2][a3][a4][a5][a6][a7],\
>>> [a1][0:a:1][a0][0:a:0][0:a:2]amerge=inputs=5[a]" \
>>> -map 0:v -map "[a]" -timecode 01:00:00:00 testmerge01.mxf \
>>> -map "[a2]" -map "[a3]" -map "[a4]" -map "[a5]" -map "[a6]" -map "[a7]" -f
>>> null -
> 
>> And after that promote it to 24 bits again.
>> Is there a way to force the channel split to work in 24 bits?
>> (No idea how to squeeze pcm_s24le into the filter_complex part…)
> 
> The default audio codec for the null muxer is pcm_s16le, and your audio codec 
> option is only set for the first output, so add '-c:a pcm_s24le' anywhere 
> after the output mxf filename, and before the pipe character.

Hi Gyan,
AH! Stupid me, now I see only the unused channels (that go down the drain) are 
16 bits :-)
Thanks!

Bouke



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Re: [FFmpeg-user] Amerge trouble

2017-11-19 Thread Bouke / VideoToolShed
On 18 Nov 2017, at 17:24, Erik Dobberkau  wrote:
> 
> or if you want a single audio stream with 5 channels:
> 
> ffmpeg -i HD422.mxf -i 8ch.wav -c:v copy -c:a pcm_s24le -filter_complex \
> "[1:a]channelsplit=channel_layout=7.1[a0][a1][a2][a3][a4][a5][a6][a7],\
> [a1][0:a:1][a0][0:a:0][0:a:2]amerge=inputs=5[a]" \
> -map 0:v -map "[a]" -timecode 01:00:00:00 testmerge01.mxf \
> -map "[a2]" -map "[a3]" -map "[a4]" -map "[a5]" -map "[a6]" -map "[a7]" -f
> null -

Hi Eric,
Great (I would never have come up with the way of disposing unwanted channels.)
One problem though, It seems that FFmpeg pipes the channel split, but does so 
in 16 bits:

  Stream #1:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, 7.1, s32 (24 
bit), 9216 kb/s
Stream mapping:
  Stream #0:1 (pcm_s24le) -> amerge:in3
  Stream #0:2 (pcm_s24le) -> amerge:in1
  Stream #0:3 (pcm_s24le) -> amerge:in4
  Stream #1:0 (pcm_s24le) -> channelsplit
  Stream #0:0 -> #0:0 (copy)
  amerge -> Stream #0:1 (pcm_s24le)
  channelsplit:FC -> Stream #1:0 (pcm_s16le)
  channelsplit:LFE -> Stream #1:1 (pcm_s16le)
  channelsplit:BL -> Stream #1:2 (pcm_s16le)
  channelsplit:BR -> Stream #1:3 (pcm_s16le)
  channelsplit:SL -> Stream #1:4 (pcm_s16le)
  channelsplit:SR -> Stream #1:5 (pcm_s16le)

And after that promote it to 24 bits again.
Is there a way to force the channel split to work in 24 bits?
(No idea how to squeeze pcm_s24le into the filter_complex part…)

Bouke



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[FFmpeg-user] Amerge trouble

2017-11-18 Thread Bouke / VideoToolShed
Hi guys,
Trying to do some audio patching from multiple input files, and output a QT or 
MXF with a poly soundtrack.
Now this turns out to be a hell of a job, I can’t grasp the logic on how the 
patching works.

Y\LTC_Convert\FF\ffmpeg -threads 1 -i V\!patch\test.mxf -ss 0. -i 
V\!patch\snd.wav -map 0:0 -filter_complex 
"[1:a][0:a:2][1:a][0:a:1][0:a:3]amerge=inputs=5,pan=5c|c0=c9|c1=c1|c2=c8|c3=c0|c4=c2[a]"
 -map "[a]" -ca pcm_s24le -cv copy -timecode 01:00:00:00 -y 
V\!patch\active\test.mxf"

FFmpeg returns:
Stream mapping:
  Stream #0:2 (pcm_s24le) -> amerge:in3
  Stream #0:3 (pcm_s24le) -> amerge:in1
  Stream #0:4 (pcm_s24le) -> amerge:in4
  Stream #1:0 (pcm_s24le) -> amerge:in0
  Stream #1:0 (pcm_s24le) -> amerge:in2

This looks all good, so the problem must be in how I define the pan, but for 
the life of me I can’t understand how the numbers work.
It seems (by other tests) that the input track up numbers based on what is 
already set, but it’s still erratic..

I would expect to get (output = from input file)
sndTrack 0 = snd.wav 2
sndTrack 1 = test.mxf 2
sndTrack 2 = snd.wav 1
sndTrack 3 = test.mxf 1
sndTrack 4 = test.mxf 3

but, the result has channels mapped as
sndTrack 0 = snd.wav 1
sndTrack 1 = snd.wav 2
sndTrack 2 = test.mxf 3
sndTrack 3 = snd.wav 1
sndTrack 4 = snd.wav 3

Isn’t there a simple way to set the patching? Or, where is my logic off?

thx,
Bouke


ffmpeg version 3.2.2 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-libbs2b --enable-libcaca --enable-libfreetype 
--enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug 
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp 
--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. 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
  libswscale  4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc54.  1.100 / 54.  1.100
Guessed Channel Layout for Input Stream #0.1 : mono
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Guessed Channel Layout for Input Stream #0.4 : mono
Guessed Channel Layout for Input Stream #0.5 : mono
Guessed Channel Layout for Input Stream #0.6 : mono
Guessed Channel Layout for Input Stream #0.7 : mono
Guessed Channel Layout for Input Stream #0.8 : mono
Input #0, mxf, from 'V:\!patch\test.mxf':
  Metadata:
uid : b8b5c18e-2490-6849-ad9f-7f99f053ff76
generation_uid  : 30fcae45-66b3-8245-94b0-ad13bedb033d
company_name: AVID
product_name: TRMG
product_version : 3.01
product_uid : ----
modification_date: 2017-11-16T11:51:43.00Z
material_package_umid: 
0x060A2B340101010501010D13130060373AAE025378409AB925C4319F6FB2
timecode: 01:00:00:00
  Duration: 00:00:16.88, start: 0.00, bitrate: 60691 kb/s
Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709, top first), 
1920x1080 [SAR 1:1 DAR 16:9], 5 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc
Metadata:
  file_package_umid: 
0x060A2B340101010501010D131300CA47B58C572A8D458FF25B9B8DC8F42F
Stream #0:1: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D131300CA47B58C572A8D458FF25B9B8DC8F42F
Stream #0:2: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D131300CA47B58C572A8D458FF25B9B8DC8F42F
Stream #0:3: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D131300CA47B58C572A8D458FF25B9B8DC8F42F
Stream #0:4: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D131300CA47B58C572A8D458FF25B9B8DC8F42F
Stream #0:5: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D131300CA47B58C572A8D458FF25B9B8DC8F42F
Stream #0:6: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D131300CA47B58C572A8D458FF25B9B8DC8F42F

Re: [FFmpeg-user] Qt audio tracks muted

2017-11-12 Thread Bouke / VideoToolShed



> On 12 Nov 2017, at 16:54, Carl Eugen Hoyos <ceffm...@gmail.com> wrote:
> 
> 2017-11-12 16:52 GMT+01:00 Bouke / VideoToolShed <bo...@videotoolshed.com>:
>> When outputting QT with multiple audio tracks, on the first one is flagged 
>> as active.
>> Now I found this documented, and ‘should’ be fixed, but it’s still the case.
> 
> The inconsistency is a known regression.

Ok, thx.
Is there a ticket? Should I send a bug report?
(I can fix it myself but would rather not be dependent on QT anymore)

Bouke

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[FFmpeg-user] Qt audio tracks muted

2017-11-12 Thread Bouke / VideoToolShed
When outputting QT with multiple audio tracks, on the first one is flagged as 
active.
Now I found this documented, and ‘should’ be fixed, but it’s still the case.
(And it’s bad, as Premiere ignores those tracks.)

Anything I can do?

Bouke

ffmpeg -threads 2 -i V\patchTest\352_1157.MXF  -map 0:0 -map 0:4 -map 0:3 -map 
0:2 -map 0:1  -ar 48000 -acodec pcm_s24le /
-map_channel 0.4.0:0.1 -ar 48000 -acodec pcm_s24le -map_channel 0.3.0:0.2 -ar 
48000 -acodec pcm_s24le -map_channel 0.2.0:0.3 /
-ar 48000 -acodec pcm_s24le -map_channel 0.1.0:0.4 -cv copy -timecode 17540620 
-shortest -t 20 -y V\patchTest\352_1157.mov

ffmpeg version N-85653-gb4330a0 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid 
--enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc 
--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-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 
--enable-libopenjpeg --enable-libopus --enable-librtmp --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-zlib
  libavutil  55. 61.100 / 55. 61.100
  libavcodec 57. 93.100 / 57. 93.100
  libavformat57. 72.101 / 57. 72.101
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 87.100 /  6. 87.100
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
[h264 @ 00390fc0] Truncating likely oversized PPS (9469 > 4096)
Last message repeated 8 times
[mxf @ 0038a540] Stream #0: not enough frames to estimate rate; 
consider increasing probesize
Guessed Channel Layout for Input Stream #0.1 : mono
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Guessed Channel Layout for Input Stream #0.4 : mono
Guessed Channel Layout for Input Stream #0.5 : mono
Guessed Channel Layout for Input Stream #0.6 : mono
Guessed Channel Layout for Input Stream #0.7 : mono
Guessed Channel Layout for Input Stream #0.8 : mono
Input #0, mxf, from 'V:\patchTest\352_1157.MXF':
  Metadata:
uid : 6e2255ca-3af5-11e6-b520-080046998630
generation_uid  : 6e2255d4-3af5-11e6-9e7c-080046998630
company_name: Sony
product_name: Mem
product_version : 2.00
product_uid : cede1104-8280-11de-8a39-08004678031c
modification_date: 2016-06-25T16:54:07.00Z
material_package_umid: 
0x060A2B340101010501010D431300E19B49B4647505CD0800460202998630
timecode: 17:54:06:20
  Duration: 00:00:05.76, start: 0.00, bitrate: 124735 kb/s
Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le(pc, bt709, 
progressive), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:1: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:2: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:3: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:4: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:5: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:6: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:7: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:8: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
Stream #0:9: Data: none
Metadata:
  file_package_umid: 
0x060A2B340101010501010D431300E29B49B4647505CD0800460202998630
  data_type   : 

[FFmpeg-user] Two frames trimmed of source start

2017-09-22 Thread Bouke / VideoToolShed
Hi guys,
Trying to do a simple transcode, but the output is missing 2 frames at the 
start.
I can live with that, but the timecode is then off by 2 frames as well.
How can I tell if the trimming will / will not occur so I can compensate the TC 
stamp?
(Or even better, can I avoid the trimming?)
thx,

Bouke

Boukes-portable:~ bouke$ /Users/bouke/Desktop/ffmpeg -i 
/Users/bouke/Desktop/Judith/LV6A3186_0226D1A.mov 
/Users/bouke/Desktop/Judith/test.mp4
ffmpeg version 3.3.1 Copyright (c) 2000-2017 the FFmpeg developers
  built with llvm-gcc 4.2.1 (LLVM build 2336.11.00)
  configuration: --prefix=/Volumes/Ramdisk/sw --enable-gpl --enable-pthreads 
--enable-version3 --enable-libspeex --enable-libvpx --disable-decoder=libvpx 
--enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 
--enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb 
--enable-filters --enable-libgsm --enable-libvidstab --enable-libx265 
--disable-doc --arch=x86_64 --enable-runtime-cpudetect
  libavutil  55. 58.100 / 55. 58.100
  libavcodec 57. 89.100 / 57. 89.100
  libavformat57. 71.100 / 57. 71.100
  libavdevice57.  6.100 / 57.  6.100
  libavfilter 6. 82.100 /  6. 82.100
  libswscale  4.  6.100 /  4.  6.100
  libswresample   2.  7.100 /  2.  7.100
  libpostproc54.  5.100 / 54.  5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 
'/Users/bouke/Desktop/Judith/LV6A3186_0226D1A.mov':
  Metadata:
major_brand : qt
minor_version   : 0
compatible_brands: qt
creation_time   : 2017-09-22T13:07:25.00Z
  Duration: 00:00:06.17, start: 0.00, bitrate: 14219 kb/s
Stream #0:0(und): Video: prores (apcn / 0x6E637061), yuv422p10le(bt709, 
progressive), 480x270, 12679 kb/s, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr, 24k 
tbn, 24k tbc (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  encoder : Apple ProRes 422
  timecode: 08:38:58:09
Stream #0:1(und): Audio: pcm_s16le (lpcm / 0x6D63706C), 48000 Hz, stereo, 
s16, 1536 kb/s (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
Stream #0:2(und): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  timecode: 08:38:58:09
Stream mapping:
  Stream #0:0 -> #0:0 (prores (native) -> h264 (libx264))
  Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
No pixel format specified, yuv422p for H.264 encoding chosen.
Use -pix_fmt yuv420p for compatibility with outdated media players.
[libx264 @ 0x7fe83b026600] using SAR=1/1
[libx264 @ 0x7fe83b026600] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 
AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 0x7fe83b026600] profile High 4:2:2, level 2.1, 4:2:2 8-bit
[libx264 @ 0x7fe83b026600] 264 - core 148 - H.264/MPEG-4 AVC codec - Copyleft 
2003-2016 - 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=-2 threads=6 lookahead_threads=1 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=23 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 '/Users/bouke/Desktop/Judith/test.mp4':
  Metadata:
major_brand : qt
minor_version   : 0
compatible_brands: qt
encoder : Lavf57.71.100
Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv422p, 
480x270 [SAR 1:1 DAR 16:9], q=-1--1, 23.98 fps, 24k tbn, 23.98 tbc (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  timecode: 08:38:58:09
  encoder : Lavc57.89.100 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1(und): Audio: aac (LC) ([64][0][0][0] / 0x0040), 48000 Hz, 
stereo, fltp, 128 kb/s (default)
Metadata:
  creation_time   : 2017-09-22T13:07:25.00Z
  handler_name: Core Media Data Handler
  encoder : Lavc57.89.100 aac
frame=   86 fps=0.0 q=28.0 size=  46kB time=00:00:04.52 bitrate=  
82.6kbits/frame=  148 fps=127 q=-1.0 Lsize= 220kB time=00:00:06.18 bitrate= 
291.9kbits/s speed=5.32x
video:213kB audio:2kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 2.750126%
[libx264 @ 0x7fe83b026600] frame I:1 Avg QP:23.73  size:  2878
[libx264 @ 0x7fe83b026600] frame P:40Avg QP:23.76  size:  3321
[libx264 @ 0x7fe83b026600] frame B:107   Avg QP:26.74  size:   762
[libx264 @ 

[FFmpeg-user] FFmpeg, please talk to me.

2017-08-04 Thread Bouke / VideoToolShed
Guys,
Trying to listen to FFmpeg output, and ‘sometimes’ strange things happen.

On Windows 10, tried running FFmpeg from a batch file while redirecting the 
FFMpeg output to a file.
Then, during it’s work, my work copies the file to a fresh one, open it and 
read the contents.
When FFmpeg tells my work that it’s done, I try to delete the output.txt files.
However, sometimes they are locked. Any clues how that can happen?

Now, I switched to plan B. Shelling to FFmpeg and listen to StdOut.
(Ok, a more logical approach, but there was a reason for Plan A, don’t ask…)
This all seems to work fine, except that FFmpeg every now and then returns:
Exiting normally, received signal 15
(Here I've switched to Win 7 for testing)
But I’m pretty sure I did not send a cmd-Q or alike…
And it’s totally random when it happens. FFmpeg is just rewrapping ProRes QT 
with a bit of audio patching.

It’s not a biggy, as I can catch it and fire the same command again, but 
nevertheless I’d like to know what I’m doing wrong.

thx,

Bouke

ffmpeg version 3.2.2 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-libbs2b --enable-libcaca --enable-libfreetype 
--enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug 
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb 
--enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp 
--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. 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
  libswscale  4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc54.  1.100 / 54.  1.100
Guessed Channel Layout for Input Stream #0.1 : hexadecagonal
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'M:\Petar\input\Capture - Copy 
(9).mov':
  Metadata:
creation_time   : 2017-01-24T03:15:23.00Z
timecode: 01:02:40:28
  Duration: 00:00:25.17, start: 0.00, bitrate: 164864 kb/s
Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le(bt709, 
progressive), 1920x1080, 146776 kb/s, SAR 1:1 DAR 16:9, 30 fps, 30 tbr, 3k tbn, 
3k tbc (default)
Metadata:
  creation_time   : 2017-01-24T03:15:23.00Z
  handler_name: Apple Alias Data Handler
  encoder : Apple ProRes 422
Stream #0:1(eng): Audio: pcm_s24le (lpcm / 0x6D63706C), 48000 Hz, 
hexadecagonal, s32 (24 bit), 18432 kb/s (default)
Metadata:
  creation_time   : 2017-01-24T03:15:23.00Z
  handler_name: Apple Alias Data Handler
Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
Metadata:
  creation_time   : 2017-08-04T12:45:41.00Z
  handler_name: Apple Alias Data Handler
  reel_name   : input
  timecode: 01:02:40:28
-map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c0.
[pan @ 02f95a00] Pure channel mapping detected: 0
-map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c1.
[pan @ 02f78060] Pure channel mapping detected: 1
-map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c2.
[pan @ 02f78360] Pure channel mapping detected: 2
-map_channel is forwarded to lavfi similarly to -af pan=0x4|c0=c3.
[pan @ 02f78660] Pure channel mapping detected: 3
Output #0, mov, to 'M:\Petar\output\Capture - Copy (9).mov':
  Metadata:
timecode: 01:02:40:28
encoder : Lavf57.56.100
Stream #0:0(eng): Video: prores (apcn / 0x6E637061), yuv422p10le(bt709, 
progressive), 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 146776 kb/s, 30 fps, 30 
tbr, 12k tbn, 3k tbc (default)
Metadata:
  creation_time   : 2017-01-24T03:15:23.00Z
  handler_name: Apple Alias Data Handler
  encoder : Apple ProRes 422
  reel_name   : input
Stream #0:1(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 
(24 bit), 1152 kb/s (default)
Metadata:
  creation_time   : 2017-01-24T03:15:23.00Z
  handler_name: Apple Alias Data Handler
  encoder : Lavc57.64.101 pcm_s24le
Stream #0:2(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 
(24 bit), 1152 kb/s (default)
Metadata:
  creation_time   : 2017-01-24T03:15:23.00Z
  handler_name: Apple Alias Data Handler
  encoder : Lavc57.64.101 pcm_s24le
Stream #0:3(eng): Audio: 

Re: [FFmpeg-user] Scan EBU R128 filter only stream#0:1 ?

2017-07-25 Thread Bouke / VideoToolShed

> On 25 Jul 2017, at 16:24, Nopidea  wrote:
> 
> Hello all,
> 
> I try to scan mix stereo from video file with ffmpeg filter EBU R128.
> My video file own 2 mix stereo channels (ch1/2 and ch3/4)
> I don't found the ffmpeg command who scan EBU R128 only on channels 3/4.
> 
> Do you have an idea ?

Audio mapping?
https://trac.ffmpeg.org/wiki/AudioChannelManipulation 


Bouke

> Thanks.
> 
> 
> 
> 
> --
> View this message in context: 
> http://www.ffmpeg-archive.org/Scan-EBU-R128-filter-only-stream-0-1-tp4680715.html
> Sent from the FFmpeg-users mailing list archive at Nabble.com.
> ___
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> 
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Re: [FFmpeg-user] man ffmpeg absent

2017-07-18 Thread Bouke / VideoToolShed

> On 18 Jul 2017, at 17:50, Moritz Barsnick  wrote:
> 
>> 
>> You guys really are a fucking joke, you know that.
> 
> Now, that's an insult. To comedy as well.

Note to self, don’t take a sip and start reading mail.
On the other hand, it was time to clean my screen anyways.

@All:
Lighten up.

Bouke



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Re: [FFmpeg-user] updating com.apple.quicktime.location.name & com.apple.quicktime.location.date

2017-06-27 Thread Bouke / VideoToolShed

> On 27 Jun 2017, at 12:17, Mohamed Ali  wrote:
> 
> hi, Is there a way to update quicktime metadata (
> com.apple.quicktime.location.name & com.apple.quicktime.location.date) that
> specified here
> https://developer.apple.com/library/content/documentation/QuickTime/QTFF/Metadata/Metadata.html
> using ffmpeg

AFAIK, only Reelname, Timecode and album / artist kind of entries can be 
changed, but not destructive, you have to generate a new file.

hth
Bouke

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Re: [FFmpeg-user] Mp3 segmentation fault, why? FIXED

2017-06-26 Thread Bouke / VideoToolShed

> On 24 Jun 2017, at 17:27, Bouke / VideoToolShed <bo...@videotoolshed.com> 
> wrote:
> 
>> A segmentation fault should usually not be your fault. It could be
>> ffmpeg, it could be libmp3lame, or it could be some unfortunate
>> exchange of libraries.
>> 
>> Did you build this ffmpeg yourself? Do you have access to the version

I was feeding 48K to the Mp3 echidna, and ‘sometimes’ it seems to choke on that.
-r 32000 made the problem go away….

Bouke

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Re: [FFmpeg-user] Mp3 segmentation fault, why?

2017-06-24 Thread Bouke / VideoToolShed
Hi Moritz


> On 24 Jun 2017, at 17:18, Moritz Barsnick <barsn...@gmx.net> wrote:
> 
> On Sat, Jun 24, 2017 at 16:41:09 +0200, Bouke / VideoToolShed wrote:
>> Can someone please tell me what I’m doing wrong here?
> 
> A segmentation fault should usually not be your fault. It could be
> ffmpeg, it could be libmp3lame, or it could be some unfortunate
> exchange of libraries.
> 
> Did you build this ffmpeg yourself? Do you have access to the version

Nope, I did not, I can’t :-) (I’m mortal…)

> with debug symbols (ffmpeg_g)? Only then can this issue be debugged.
> How to use gdb (and report the issue) is described here:
> https://ffmpeg.org/bugreports.html
> I don't know whether Mac OSX uses gdb though.
> 
>> [...] -filter_complex "[0:0][0:0] amix=inputs=2,pan=mono|c0=c0+c1" -ac 1
> 
> I wasn't aware you could use an input ([0:0]) twice, and it seems
> you're doing something in a more complicated way than you should - but
> that shouldn't lead to a crash. ;-)

This is a result of my front end that has to do a lot of stuff and be as 
versatile as possible.
It ‘should’ work (and it does, except for this audio codec, change it to 
pcm_s16le it works just fine…)

strange thing is, I did not notice this before, so there is ‘something’ strange 
with the input file.
No clue yet, and it only seems to happen on Mac.
I’ll dive into it tomorrow and try to pinpoint it more exactly.
(That will probably mean get other FFmpeg binaries)
As always, thanks for your help!

Bouke


> 
> Moritz
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[FFmpeg-user] Mp3 segmentation fault, why?

2017-06-24 Thread Bouke / VideoToolShed
Hi guys,
Can someone please tell me what I’m doing wrong here?
thx,
Bouke


Boukes-portable:~ bouke$ ffmpeg -i /Users/bouke/Desktop/Judith/OKC170614T09.WAV 
-filter_complex "[0:0][0:0] amix=inputs=2,pan=mono|c0=c0+c1" -ac 1   -codec:a 
libmp3lame /Users/bouke/Desktop/Judith/OKC170614T09.mp3 
ffmpeg version 3.3-tessus Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg 
--extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl 
--enable-libass --enable-libbluray --enable-libfreetype --enable-libgsm 
--enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopus --enable-libschroedinger 
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora 
--enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx 
--enable-libwavpack --enable-libx264 --enable-libx265 --enable-libxavs 
--enable-libxvid --enable-libzmq --enable-libzvbi --enable-version3 
--disable-ffplay --disable-indev=qtkit
  libavutil  55. 58.100 / 55. 58.100
  libavcodec 57. 89.100 / 57. 89.100
  libavformat57. 71.100 / 57. 71.100
  libavdevice57.  6.100 / 57.  6.100
  libavfilter 6. 82.100 /  6. 82.100
  libswscale  4.  6.100 /  4.  6.100
  libswresample   2.  7.100 /  2.  7.100
  libpostproc54.  5.100 / 54.  5.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from '/Users/bouke/Desktop/Judith/OKC170614T09.WAV':
  Metadata:
comment : sSPEED=023.976-ND 
: sTAKE=09 
: sUBITS=$ 
: sSWVER=4.50.16 
: sSCENE=OKC170614 
: sFILENAME=OKC170614T09.WAV 
: sTAPE=17Y06M14 
: sCIRCLED=FALSE 
: sTRK1=BOOM 
: sTRK2=BOOM -3dB 
: sNOTE=KID INTV - ISMAEL ZAMARANO 
: 
encoded_by  : Sound Dev: Mix633 S#LL0414160003
originator_reference: USSDVLL0414160003170614AW94v  01
date: 2017-06-14
creation_time   : 10:32:18
time_reference  : 1822604785
coding_history  : A=PCM,F=48000,W=24,M=stereo,R=48000,T=2 Ch 
: 
  Duration: 00:06:34.39, bitrate: 2304 kb/s
Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, 
s32 (24 bit), 2304 kb/s
File '/Users/bouke/Desktop/Judith/OKC170614T09.mp3' already exists. Overwrite ? 
[y/N] y
Stream mapping:
  Stream #0:0 (pcm_s24le) -> amix:input0
  Stream #0:0 (pcm_s24le) -> amix:input1
  pan -> Stream #0:0 (libmp3lame)
Press [q] to stop, [?] for help
Output #0, mp3, to '/Users/bouke/Desktop/Judith/OKC170614T09.mp3':
  Metadata:
comment : sSPEED=023.976-ND 
: sTAKE=09 
: sUBITS=$ 
: sSWVER=4.50.16 
: sSCENE=OKC170614 
: sFILENAME=OKC170614T09.WAV 
: sTAPE=17Y06M14 
: sCIRCLED=FALSE 
: sTRK1=BOOM 
: sTRK2=BOOM -3dB 
: sNOTE=KID INTV - ISMAEL ZAMARANO 
: 
TENC: Sound Dev: Mix633 S#LL0414160003
originator_reference: USSDVLL0414160003170614AW94v  01
TDRC: 2017-06-14
coding_history  : A=PCM,F=48000,W=24,M=stereo,R=48000,T=2 Ch 
: 
time_reference  : 1822604785
TSSE: Lavf57.71.100
Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, mono, fltp (default)
Metadata:
  encoder : Lavc57.89.100 libmp3lame
Segmentation fault: 110:06:27.33 bitrate=  64.0kbits/s speed=25.8x


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Re: [FFmpeg-user] Images to video under Windows

2017-05-22 Thread Bouke / VideoToolShed
> On 22 May 2017, at 17:36, Wolfgang Hugemann  wrote:
> 
> Hey,
> 
> I have a bunch of images that I want to convert into a video. These are, 
> however, not numbered consecuitively (not like 001.jpg, 002.jpg, ...).
> 
> Well, I could make a copy of each and rename them in such a pattern, but is 
> there a more elegant way to perform the job? File name globbing is obviously 
> not supported by the Windows version. How about piping?

Hi Wolfgang,
Piping is a bad idea, it craps out after some 40 images in my experience. (Pipe 
to pipe to pipe to….)
What about creating a Cat.txt file first, and use that as input? That works for 
me just fine.
(Although the cat.txt is done outside FFmpeg…)

hth,

Bouke


> Wolfgang Hugemann
> ___
> 

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Re: [FFmpeg-user] Avid opAtom trouble

2017-05-05 Thread Bouke (VideoToolShed)


- Original Message - 
From: "Jaka Bac" <jaka...@gmail.com>

To: "FFmpeg user questions" <ffmpeg-user@ffmpeg.org>
Sent: Friday, May 05, 2017 12:31 PM
Subject: Re: [FFmpeg-user] Avid opAtom trouble



Hello

This probably happens because the ffmpeg MXF muxer does not write some
metadata that Avid expects.

Namely the following entries in the CDCI descriptor will be missing:
SampledWidth = 1920
SampledHeight = 1080

and this will be set (where 1920x1088 comes from):
StoredWidth = 1920
StoredHeight = 544

But the actual problem I believe is the missing metadata.

There is a patch floating around the mailing lists which fixes this (and
also same problem which happens where OP1a XDCAM HD files generated by
ffmpeg are imported into Avid editors).
https://ffmpeg.org/pipermail/ffmpeg-devel/2016-June/194830.html

Is there any reason why this patch is not accepted/applied?


Hi Jaka,
Is this a question for me? I can't answer it, just took the latest Zeranoe 
build...

thx,
Bouke


Jaka

On 2 May 2017 at 17:37, Bouke (VideoToolShed) <bo...@videotoolshed.com>
wrote:

- Original Message - From: "Carl Eugen Hoyos" 
<ceffm...@gmail.com>

To: "FFmpeg user questions" <ffmpeg-user@ffmpeg.org>
Sent: Tuesday, May 02, 2017 4:31 PM
Subject: Re: [FFmpeg-user] Avid opAtom trouble


2017-05-02 14:46 GMT+02:00 Bouke (VideoToolShed) <bo...@videotoolshed.com

>:

Toying around with Avid opAtom mxf, generating test clips.
When I look at the clip in Avid, the bin info shows a raster dimension 
of

1920x1088, pixel aspect ratio of 1.007 (expected 1920x1080 / 1.000)
And of course, when editing the clip into a seqence, frameflex kicks 
in.


same line but output to Mov and AMA the clip in does give the
expected result.



What happens if you use FFmpeg to remux the working mov file
into mxf?



-vcodec copy NEEDS the FPS, or FFmpeg bitches. When I add -r 25, it does
the re-wrap, but output gives the same trouble in Avid.

What happens if you use FFmpeg to remux the broken mxf file

into mov?



FFmpeg reports the frame size has changed from 1088 to 1080, and all is
well.
(Except the UUID or something is copied over, so doing the dance twice to
two filenames makes Avid refuse to import both, as it 'knows' it 
already...)


Nice details, with -vcodec copy also timecode comes across (From MXF to
MOV and vice-versa.)
Also the interlace flag is set without issue.

Can you provide a working mxf file similar to the one FFmpeg

produces?



Yes, see here:
Download link
https://wetransfer.com/downloads/d35f135d397837227c965ee7b95
ea7bf20170502152440/018e64f2872c5c6e4de8bca6e08ca3f620170502152440/fd3691


Note, this is an AVID generated mixdown of the (good) AMA linked MOV. Tc
has changed to seqence TC (00:00:00:00)

(I believe I remember what the issue is but would like to

verify.)



Let me know if you need more, and thanks!

Bouke

Carl Eugen

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Re: [FFmpeg-user] Avid opAtom trouble

2017-05-02 Thread Bouke (VideoToolShed)
- Original Message - 
From: "Carl Eugen Hoyos" <ceffm...@gmail.com>

To: "FFmpeg user questions" <ffmpeg-user@ffmpeg.org>
Sent: Tuesday, May 02, 2017 4:31 PM
Subject: Re: [FFmpeg-user] Avid opAtom trouble


2017-05-02 14:46 GMT+02:00 Bouke (VideoToolShed) 
<bo...@videotoolshed.com>:



Toying around with Avid opAtom mxf, generating test clips.
When I look at the clip in Avid, the bin info shows a raster dimension of
1920x1088, pixel aspect ratio of 1.007 (expected 1920x1080 / 1.000)
And of course, when editing the clip into a seqence, frameflex kicks in.

same line but output to Mov and AMA the clip in does give the
expected result.


What happens if you use FFmpeg to remux the working mov file
into mxf?


-vcodec copy NEEDS the FPS, or FFmpeg bitches. When I add -r 25, it does the 
re-wrap, but output gives the same trouble in Avid.



What happens if you use FFmpeg to remux the broken mxf file
into mov?


FFmpeg reports the frame size has changed from 1088 to 1080, and all is 
well.
(Except the UUID or something is copied over, so doing the dance twice to 
two filenames makes Avid refuse to import both, as it 'knows' it already...)


Nice details, with -vcodec copy also timecode comes across (From MXF to MOV 
and vice-versa.)

Also the interlace flag is set without issue.


Can you provide a working mxf file similar to the one FFmpeg
produces?


Yes, see here:
Download link
https://wetransfer.com/downloads/d35f135d397837227c965ee7b95ea7bf20170502152440/018e64f2872c5c6e4de8bca6e08ca3f620170502152440/fd3691


Note, this is an AVID generated mixdown of the (good) AMA linked MOV. Tc has 
changed to seqence TC (00:00:00:00)



(I believe I remember what the issue is but would like to
verify.)


Let me know if you need more, and thanks!

Bouke


Carl Eugen
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[FFmpeg-user] Avid opAtom trouble

2017-05-02 Thread Bouke (VideoToolShed)

Hi guys,
Toying around with Avid opAtom mxf, generating test clips.
When I look at the clip in Avid, the bin info shows a raster dimension of 
1920x1088, pixel aspect ratio of 1.007 (expected 1920x1080 / 1.000)

And of course, when editing the clip into a seqence, frameflex kicks in.

same line but output to Mov and AMA the clip in does give the expected 
result.


Am I missing something or is this a bug?

Next, what kind of metadata besides the clip name / timecode is supported?

thx,
Bouke

ffmpeg -f lavfi -i "testsrc=size=1920x1080:rate=25" -frames:v 10 -f 
mxf_opatom -vcodec dnxhd -pix_fmt yuv422p -vb 120M -timecode 
01:02:03:04 -metadata material_package_name="rastertest" -flags +ildct -top 
1 -y "M:\Avid Mediafiles\MXF\2\test.mxf"


ffmpeg version N-85653-gb4330a0 Copyright (c) 2000-2017 the FFmpeg 
developers

 built with gcc 6.3.0 (GCC)
 configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid 
--enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --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-libmp3lame 
--enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264  
--enable-libopenjpeg --enable-libopus --enable-librtmp --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-zlib

 libavutil  55. 61.100 / 55. 61.100
 libavcodec 57. 93.100 / 57. 93.100
 libavformat57. 72.101 / 57. 72.101
 libavdevice57.  7.100 / 57.  7.100
 libavfilter 6. 87.100 /  6. 87.100
 libswscale  4.  7.101 /  4.  7.101
 libswresample   2.  8.100 /  2.  8.100
 libpostproc54.  6.100 / 54.  6.100
Input #0, lavfi, from 'testsrc=size=1920x1080:rate=25':
 Duration: N/A, start: 0.00, bitrate: N/A
   Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 1920x1080 
[SAR 1:1 DAR 16:9], 25 tbr, 25 tbn, 25 tbc

Stream mapping:
 Stream #0:0 -> #0:0 (rawvideo (native) -> dnxhd (native))
Press [q] to stop, [?] for help
Output #0, mxf_opatom, to 'M:\Avid MediaFiles\MXF\2\tst1.mxf':
 Metadata:
   timecode: 01:02:03:04
   material_package_name: raster test
   encoder : Lavf57.72.101
   Stream #0:0: Video: dnxhd (DNXHD), yuv422p, 1920x1080 [SAR 1:1 DAR 
16:9], q=2-1024, 12 kb/s, 25 fps, 25 tbn, 25 tbc

   Metadata:
 encoder : Lavc57.93.100 dnxhd
frame=   10 fps=0.0 q=1.0 Lsize=5927kB time=00:00:00.40 
bitrate=121375.9kbits/s speed=2.38x
video:5920kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB 
muxing overhead: 0.110738%



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[FFmpeg-user] OGG metadata

2017-04-06 Thread Bouke / VideoToolShed
Guys,
Got an OGG file that is supposed to contain timecode. (It’s generated for 
transcription.)
Output is below. Can someone shed some light on what I’m seeing? Is there 
indeed some time info there? If so, how to interpret it?
(The file indeed seems catted from multiple shots.)

thx,
Bouke


ffmpeg -threads 0 -i /Users/bouke/Desktop/GF-JAX-GFT02.ogg 
ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
  configuration: --enable-ffplay
  libavutil  54. 20.100 / 54. 20.100
  libavcodec 56. 26.100 / 56. 26.100
  libavformat56. 25.101 / 56. 25.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 11.102 /  5. 11.102
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
Input #0, ogg, from '/Users/bouke/Desktop/GF-JAX-GFT02.ogg':
  Duration: 00:07:01.02, start: 0.00, bitrate: 222 kb/s
Stream #0:0: Audio: vorbis, 48000 Hz, 5.1, fltp, 276 kb/s
Metadata:
  COM.TSCRIPTS.01 : 
1E5C95610BC686788DD615F25970B3281061A95035D2CFFF19147F8A5DAE25E0
  COM.TSCRIPTS.02 : 
5ABA2D1453DCF207AA27222DE6CF3FD3C4E8E2E3A8F2AB00E9A57699D8BCF4F5
  COM.TSCRIPTS.03 : 
A7070CB03DF58595D9F01D5D678340BDC9139D2F8AF390AC4F18485490ABF654
  COM.TSCRIPTS.04 : 
CC4B800B763738712BD70DA466BCEA5C29054E676FC6BDD59B7C9F8592D696ED
  COM.TSCRIPTS.05 : 
52E0A3C76FD61576727C383C9AE1090BF9CB6B2A683AF0D0C7DEC3ECD270B23F
  COM.TSCRIPTS.06 : 
5F1903E01BD8CD5D912A678ADB589A0E23C9EB4F30E0EC02CC164C89FA9C67B7
  COM.TSCRIPTS.07 : 
728E56975DB9494FBBE4C6A5478DD60A75D7FD781FCE45E0FAFCBD219EC033F0
  COM.TSCRIPTS.08 : 
F5E807F49793E4080CE56D8346012D7BB48F6B91AC73B48885AD339639E225ED
  COM.TSCRIPTS.09 : 
79AFC1640E54AAEA2F850199D83913F5AB5FF5A7828982DB211958F631515BE3
  COM.TSCRIPTS.10 : 
40B8B44B241A107815D903E19B69FCD0F02E34CBCAD5BC6CF80CFD7988DA61FE
  COM.TSCRIPTS.11 : 
3C8DA70E04F5269C06B55B09A494D461D706BA8FD5EE536B472C032FAB0FD4CF
  COM.TSCRIPTS.12 : 
7F4018F3CB2FAFFED0A07233B40C7F501D0AA4A3F660EB780890F9E0D126F06C
  COM.TSCRIPTS.13 : 
ECCCDF905A74CA49C2FEA3B4A27EFE9D7794FA711D495A180F3EF65FF66BABF6
  COM.TSCRIPTS.14 : 
6E225333688EE8A1AD889C9F5E9FFA2112871A50290BC55E2EED8FB2A891B406
  COM.TSCRIPTS.15 : 
01C6C8CDD02100D44BF9B1108F8DECFB473D59C162D984C0780F52EAC985627E
  COM.TSCRIPTS.16 : 
7DA487B3C1048ABFC3208880A91B898F307A052B555D70833ADEE7A7474DA556
  COM.TSCRIPTS.17 : 
E0B60EB257CE69CEEA8178C5CF7215595BFAA550E26891042F13B8E2EF362BC1
  COM.TSCRIPTS.18 : 
074CEFF545DEBF9FC8E1306D6347580CB5DFC7B20B927E139ECB537F91166F89
  COM.TSCRIPTS.19 : 
CD922DADE6A957FB435BB636AEEF8A82F5E99452E8E8DA16F83AE1893D6D9819
  COM.TSCRIPTS.20 : 
7BB7D4434AD4F11645D04EDB0E2815D3C444B461FD9E11D5C76916AD182C980A
  COM.TSCRIPTS.21 : 
879517EFE2247916012F770BF9C7182FC385776AF8D63D2125CC8A248CA59A23
  COM.TSCRIPTS.22 : 
23117D1A621EA6856F0A5FA9668877DAED17126EC1B527540E54864DD68C0575
  COM.TSCRIPTS.23 : 
3D13CB746B5146BDACB9EDA3FB994671F111062619DC347E782CB136981873C4
  COM.TSCRIPTS.24 : 
68BA976F67E7D5316E5DEDB6BF14D8442D18FE99574AA7C6D1A14372D82755CE
  COM.TSCRIPTS.25 : 
EDC0864C7D7835824BAD241BC3935AD31A95EE839ED335ED56A7266197805896
  COM.TSCRIPTS.26 : 
53ADCC66C1FC5FE75419D888648F19AC5A896E423C581293791623C5C1EE9938
  COM.TSCRIPTS.27 : 
E3972DFEF7457BB496D99AB66EA05C464AB8FC2AE3F1AE59457ED11A3EE4C971
  COM.TSCRIPTS.28 : 0BC7F22890C169C8451BB1B8B22E21F813699F8C223AD23D8AFF99
At least one output file must be specified



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Re: [FFmpeg-user] lut 3d trouble

2017-03-13 Thread Bouke / VideoToolShed

Bouke

VideoToolShed.com
van Oldenbarneveltstraat 33
6512 AS  Nijmegen
+31 6 21817248
To send files, go here:
https://videotoolshed.wetransfer.com/


> On 13 Mar 2017, at 12:46, Paul B Mahol <one...@gmail.com> wrote:
> 
> On 3/13/17, Bouke / VideoToolShed <bo...@videotoolshed.com 
> <mailto:bo...@videotoolshed.com>> wrote:
>>> On 12 Mar 2017, at 19:26, Paul B Mahol <one...@gmail.com> wrote:
>>> 
>>> On 3/12/17, Bouke / VideoToolShed <bo...@videotoolshed.com> wrote:
>>>> 
>>>>> On 12 Mar 2017, at 15:49, Paul B Mahol <one...@gmail.com> wrote:
>>>>> 
>>>>>> 
>>>>>> Boukes-portable:3d bouke$ ffmpeg -threads 0 -i
>>>>>> /Users/bouke/Desktop/input.mov  -vf
>>>>>> lut3d=/Users/bouke/Desktop/LegalLuts/3d/Hard_Clip_Legal_3D.cube -y -an
>>>>> 
>>>>> Where can this Hard_Clip_Legal_3D.cube can be found?
>>>> 
>>>> Hi Paul,
>>>> Thanks for looking into this.
>>>> You can get the LUT here: (WeTransfer link.)
>>>> Download link
>>>> https://we.tl/AzKVwaB0sF <https://we.tl/AzKVwaB0sF>
>>> 
>>> As workaround just remove 2 DOMAIN_ line entries?
>> 
>> Eeh, sorry, I have no clue what you are telling me here...
> 
> Open lut file in editor. And remove 2nd and 3rd line.

Aha! Jackpot. (I never had seen a LUT from the inside before…)
No clue if the output makes sense, but at least it’s doing ‘something’, and I 
can test!
Thanks for your help!

Bouke



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Re: [FFmpeg-user] lut 3d trouble

2017-03-13 Thread Bouke / VideoToolShed
> On 12 Mar 2017, at 19:26, Paul B Mahol <one...@gmail.com> wrote:
> 
> On 3/12/17, Bouke / VideoToolShed <bo...@videotoolshed.com> wrote:
>> 
>>> On 12 Mar 2017, at 15:49, Paul B Mahol <one...@gmail.com> wrote:
>>> 
>>>> 
>>>> Boukes-portable:3d bouke$ ffmpeg -threads 0 -i
>>>> /Users/bouke/Desktop/input.mov  -vf
>>>> lut3d=/Users/bouke/Desktop/LegalLuts/3d/Hard_Clip_Legal_3D.cube -y -an
>>> 
>>> Where can this Hard_Clip_Legal_3D.cube can be found?
>> 
>> Hi Paul,
>> Thanks for looking into this.
>> You can get the LUT here: (WeTransfer link.)
>> Download link
>> https://we.tl/AzKVwaB0sF <https://we.tl/AzKVwaB0sF>
> 
> As workaround just remove 2 DOMAIN_ line entries?

Eeh, sorry, I have no clue what you are telling me here…

Bouke


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Re: [FFmpeg-user] Convert Audio Track To Numbers and Back

2017-03-13 Thread Bouke / VideoToolShed
On 07 Mar 2017, at 03:43, Michele Monteferrante  wrote:
> 
> Sorry,
> I was not clear. What I want is a function of time, a graph.
> x: time in seconds
> Y: amplitude in whatever format
> This file must be a simple text file which,for example, I can plot with
> Gnuplot and edit whit vim.
> Ciao
> Michele

Not sure if FFmpeg can do this, but if you hire a coder he’ll be done in no 
time.
If the sound is uncompressed, it’s nothing more of finding the start of the 
actual sound data, and read the numbers.
Creating a sound file from numbers (uncompressed) is pretty easy too. (Although 
it’s your task to make it sound like ‘something’ .:-)

Now, since you want to do math on it, and you have already a nice format to 
store a couple of numbers (being X the position in the file, Y the value of the 
bits), why not stick to that as a file format as input? (And to plot, any half 
decent audio editor can plot a sound file…)

hth
Bouke


>> On 6 Mar 2017, at 10:05, Carl Eugen Hoyos  wrote:
>> 
>> 2017-03-06 4:22 GMT+01:00 Michele Monteferrante :
>>> I have a very simple task.
>> 
>>> Given an audio track (aiff or mp3) I want to convert it in series of real 
>>> numbers.
>> 
>> Are you searching for the pcm_s32be format? Or pcm_u8?
>> 
>> Carl Eugen
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Re: [FFmpeg-user] lut 3d trouble

2017-03-12 Thread Bouke / VideoToolShed

> On 12 Mar 2017, at 15:49, Paul B Mahol  wrote:
> 
>> 
>> Boukes-portable:3d bouke$ ffmpeg -threads 0 -i
>> /Users/bouke/Desktop/input.mov  -vf
>> lut3d=/Users/bouke/Desktop/LegalLuts/3d/Hard_Clip_Legal_3D.cube -y -an
> 
> Where can this Hard_Clip_Legal_3D.cube can be found?

Hi Paul,
Thanks for looking into this.
You can get the LUT here: (WeTransfer link.)
Download link 
https://we.tl/AzKVwaB0sF 

Bouke
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[FFmpeg-user] lut 3d trouble

2017-03-12 Thread Bouke / VideoToolShed
Hi guys,
Toying around on holiday, trying to apply a lut on OSX, having no luck.
Is it my FFmpeg version? (I’m on very limited bandwidth at the moment, no 
option to download a recent version for the next days…)
Any other stupid simple thing I’m missing?
thx,
Bouke

Boukes-portable:3d bouke$ ffmpeg -threads 0 -i /Users/bouke/Desktop/input.mov  
-vf lut3d=/Users/bouke/Desktop/LegalLuts/3d/Hard_Clip_Legal_3D.cube -y -an 
-vcodec prores /Users/bouke/Desktop/HARD.mov 
ffmpeg version 2.6.1 Copyright (c) 2000-2015 the FFmpeg developers
  built with Apple LLVM version 6.1.0 (clang-602.0.49) (based on LLVM 3.6.0svn)
  configuration: --enable-ffplay
  libavutil  54. 20.100 / 54. 20.100
  libavcodec 56. 26.100 / 56. 26.100
  libavformat56. 25.101 / 56. 25.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 11.102 /  5. 11.102
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '/Users/bouke/Desktop/input.mov':
  Metadata:
major_brand : qt  
minor_version   : 512
compatible_brands: qt  
encoder : Lavf57.56.100
  Duration: 00:00:10.00, start: 0.00, bitrate: 21173 kb/s
Stream #0:0(eng): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, 
bt709), 1920x1080 [SAR 1:1 DAR 16:9], 21172 kb/s, 25 fps, 25 tbr, 12800 tbn, 
25600 tbc (default)
Metadata:
  handler_name: DataHandler
[AVFilterGraph @ 0x7fdf30d03580] Error initializing filter 'lut3d' with args 
'/Users/bouke/Desktop/LegalLuts/3d/Hard_Clip_Legal_3D.cube'
Error opening filters!
Boukes-portable:3d bouke$ 


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Re: [FFmpeg-user] ogg encoder always adds 128-255 extra samples

2017-03-02 Thread Bouke / VideoToolShed

> On 02 Mar 2017, at 17:19, Basin Ilya  wrote:
> 
> You can't concatenate .wav files

Huh? Uncompressed PCM seems easy enough to cat, and I have done so a zillion 
times with FFmpeg…
Bouke
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Re: [FFmpeg-user] Video and Audio is off-sync on Decklink SDI output

2017-03-02 Thread Bouke / VideoToolShed

> On 22 Feb 2017, at 23:32, Tung Tran  wrote:
> 
> Dear all
> 
> I am testing to setup a decoder with ffmpeg, take live stream in
> (RTMP/HLS,.) and output to Decklink SDI.
> Its working but I have the issue of audio and video is off-sync on decklink
> SDI output, I tried to play around with vf setpts and af asetpts but
> couldnt make video/audio to sync.

Perhaps unrelated, but on my system, depending on the state of my machine (Eg 
if I have run Avid MC before), the output of my decklink is out of sync on 
known good sync input.
(No matter the player, it seems a Decklink driver issue.)
So you might want to test output with a known good local input file. (While 
looking / listening to Decklink output.)
hth,
Bouke

> Any input will be very appreciated.
> Here is my command so far
> 
> * ffmpeg -i rtmp://something  -vf "setpts=PTS-STARTPTS" -af
> "asetpts=PTS-STARTPTS" -f decklink -pix_fmt uyvy422  "DeckLink SDI" *
> 
> Video and audio of input rtmp is synced on VLC or other software.
> ---
> Best regards,
> 
> *Tung Tran*
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Re: [FFmpeg-user] Need help generating field based Program Stream

2017-01-24 Thread Bouke (VideoToolShed)
- Original Message - 
From: "kumar vikram" 




Hello All,
I am having a Program stream which has progressive sequence. I want to
convert this stream to Field based PS files.

I have used the following command

ffmpeg.exe -y -i INPUT -c:v mpeg2video -flags +ildct+ilme -vf yadif=1 -top
1 -b:a 256k -aspect 16:9 OUTPUT

But this is creating Frame based interlaced stream.

Looks like I have missed some options. Can you please help me fix this
problem?


Hi Vikram,
What do you expect? If the source is progressive, an interlaced output will 
have two fields taken at the same time, thus still look progressive.
If the framerate does not change, besides doing this for being compatible 
with streams that must be interlaced, there is no reason to do so.
If you do want to change the framerate (eg from 50p to 25i, or add 
pulldown), there is logic. Have a look at the Tinterlace filter.

hth,
Bouke



Regards,
Vikram



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Re: [FFmpeg-user] ffmpeg and handler_name in a mov

2016-12-10 Thread Bouke / VideoToolShed

> On Dec 7, 2016, at 15:42, David Cecco  wrote:
> 
> Is there any way to change the handler_name metadata field in a mov file?  I 
> am trying to avoid having a Final Cut Pro message pop up regarding media 
> optimisation.
> 
> Here is what I am currently trying:
> 
> ffmpeg -y -i test.mxf -vf setfield=tff -flags +ilme+ildct -map 0:1 -map 0:0 
> -metadata:s handler_name="Apple Alias Data Handler" -metadata:s 
> encoder="\"Apple ProRes 422 LT\"" -acodec pcm_s16le -vcodec prores_ks 
> -profile:v 1 -copytb -1 test.mov


Hi David,

Did you try adding a Reelname & Timecode? (And, a gotcha, don’t forget to quote 
drop frame timecode…)
And / or in combination with faststart.
I can’t remember having issues when this was all done. (But it’s some time ago 
for me…)

Bouke

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Re: [FFmpeg-user] Sony FS7 AVC long group rewrap trouble

2016-11-15 Thread Bouke / VideoToolShed



> On Nov 13, 2016, at 18:52, Carl Eugen Hoyos <ceffm...@gmail.com> wrote:
> 
> 2016-11-13 11:49 GMT+01:00 Bouke (VideoToolShed) <bo...@videotoolshed.com>:
> 
>> ffmpeg.exe -fflags +genpts -i K:\mxftest\test4\clip001.MXF -t 10 -codec
>> copy -map 0:0 -map 0:1 -map 0:2 -map 0:3 -map 0:4 K:\mxftest\test4.mov
>> 
>> ffmpeg version 3.1.4 Copyright (c) 2000-2016 the FFmpeg developers
> 
> Please test current FFmpeg git head and please provide an input sample.
> 
> Carl Eugen
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Did try latest build, and makes no difference.
I can send an input sample, but the only one I have is 2 gig, Is that a problem?

Bouke

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[FFmpeg-user] Sony FS7 AVC long group rewrap trouble

2016-11-13 Thread Bouke (VideoToolShed)

Hi guys,
Got trouble rewrapping Sony MXF clips to something else. (Main goal is to
change TC / alter sound a bit)
If I go to Mov, the result is a green image in both Avid and Premiere.
(Tried -vtag avc1 / aivx, but that seems to do nothing...)

If I go to MXF, the video is heavilly distorted / Premiere complains about a
gazillion errors.
VLC however likes the output just fine, but that's not good enough for rock
'n roll.
(Transcoding is not an option.)

Am I missing something, or is it just not implemented?

Bouke

ffmpeg.exe -fflags +genpts -i K:\mxftest\test4\clip001.MXF -t 10 -codec
copy -map 0:0 -map 0:1 -map 0:2 -map 0:3 -map 0:4 K:\mxftest\test4.mov

ffmpeg version 3.1.4 Copyright (c) 2000-2016 the FFmpeg developers
 built with gcc 5.4.0 (GCC)
 configuration: --enable-gpl --enable-version3 --disable-w32threads 
--enable-dxva2
--enable-libmfx --enable-nvenc --enable-avisynth --enable-bzlib 
--enable-libebur128
--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-libmp3lame 
--enable-libopencore-amrnb
--enable-libopencore-amrwb --enable-libopenh264 --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. 28.100 / 55. 28.100
 libavcodec 57. 48.101 / 57. 48.101
 libavformat57. 41.100 / 57. 41.100
 libavdevice57.  0.101 / 57.  0.101
 libavfilter 6. 47.100 /  6. 47.100
 libswscale  4.  1.100 /  4.  1.100
 libswresample   2.  1.100 /  2.  1.100
 libpostproc54.  0.100 / 54.  0.100
Guessed Channel Layout for Input Stream #0.1 : mono
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Guessed Channel Layout for Input Stream #0.4 : mono
Input #0, mxf, from 'K:\mxftest\Clip0001.MXF':
 Metadata:
   uid : b097b50a-e434-122d-a48b-
   generation_uid  : b097b514-e434-122d-b7fe-
   company_name: Sony
   product_name: Mem
   product_version : 2.00
   product_uid : ----
   modification_date: 2016-10-30 07:50:42
   material_package_umid:
0x060A2B340101010501010D4313007A14AE53917605C400AEFAFFFE729B3D
   timecode: 18:14:04:12
 Duration: 00:05:43.34, start: 0.00, bitrate: 51933 kb/s
   Stream #0:0: Video: h264 (High 4:2:2), yuv422p10le(tv, bt709), 1920x1080
[SAR 1:1 DAR 16:9], 23.98 fps, 23.98 tbr, 23.98 tbn, 47.95 tbc
   Metadata:
 file_package_umid:
0x060A2B340101010501010D4313007B14AE53917605C400AEFAFFFE729B3D
   Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152
kb/s
   Metadata:
 file_package_umid:
0x060A2B340101010501010D4313007B14AE53917605C400AEFAFFFE729B3D
   Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152
kb/s
   Metadata:
 file_package_umid:
0x060A2B340101010501010D4313007B14AE53917605C400AEFAFFFE729B3D
   Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152
kb/s
   Metadata:
 file_package_umid:
0x060A2B340101010501010D4313007B14AE53917605C400AEFAFFFE729B3D
   Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152
kb/s
   Metadata:
 file_package_umid:
0x060A2B340101010501010D4313007B14AE53917605C400AEFAFFFE729B3D
   Stream #0:5: Data: none
   Metadata:
 file_package_umid:
0x060A2B340101010501010D4313007B14AE53917605C400AEFAFFFE729B3D
 data_type   : vbi_vanc_smpte_436M
File 'K:\mxftest\test4.mov' already exists. Overwrite ? [y/N] [mov @
02f63020] Using AVStream.codec to pass codec parameters to muxers is
deprecated, use AVStream.codecpar instead.
   Last message repeated 4 times
Output #0, mov, to 'K:\mxftest\test4.mov':
 Metadata:
   uid : b097b50a-e434-122d-a48b-
   generation_uid  : b097b514-e434-122d-b7fe-
   company_name: Sony
   product_name: Mem
   product_version : 2.00
   product_uid : ----
   modification_date: 2016-10-30 07:50:42
   material_package_umid:
0x060A2B340101010501010D4313007A14AE53917605C400AEFAFFFE729B3D
   timecode: 18:14:04:12
   encoder : Lavf57.41.100
   Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv422p10le(tv, bt709),
1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 0.04 fps, 23.98 tbr, 24k tbn, 23.98
tbc
   Metadata:
 file_package_umid:
0x060A2B340101010501010D4313007B14AE53917605C400AEFAFFFE729B3D
   Stream #0:1: Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono (24
bit), 1152 kb/s
   Metadata:
 

Re: [FFmpeg-user] Green screen

2016-05-28 Thread Bouke (VideoToolShed)
- 
From: "Cecil Westerhof" <ce...@decebal.nl>

To: <ffmpeg-user@ffmpeg.org>
Sent: Saturday, May 28, 2016 12:05 AM
Subject: [FFmpeg-user] Green screen



I want to shoot video's with a green screen and then replace the green
part with a picture (not another video). How would I do that?


Hi Cecil,
You don't want to do this with FFmpeg.
Lighting a green (or blue) screen set is difficult already, and you want to 
have a zillion tools at your fingertips to get a decent end result.

Get a proper posthouse to do this for you. (I happen to run one :-)

Bouke

VideoToolShed
van Oldenbarneveltstraat 33
6512 AS  NIJMEGEN, the Netherlands
+31 24 3553311 



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Re: [FFmpeg-user] stabilisation

2016-05-10 Thread Bouke (VideoToolShed)
- Original Message - 
From: "Carl Eugen Hoyos" 

To: 
Sent: Tuesday, May 10, 2016 1:13 PM
Subject: Re: [FFmpeg-user] stabilisation



edouard chalaron  xtra.co.nz> writes:


when using -vf "deshake" a black frame is coming separating the
original frame from the "interpolated" pixels along the boders.


Please provide the command line you tested together with the 
complete, uncut console output.


Just a hunch, to stabilize you move the frame within is boundaries.
Crop filter after the deshake may be the answer...

Bouke


Carl Eugen

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[FFmpeg-user] Segment sound to multiple Waveform images

2016-04-09 Thread Bouke (VideoToolShed)

Hi guys,
Trying to split a soundfile to generate multiple audio waveform images.

ffmpeg.exe -i K:\test\test.wav -f segment -segment_time 8  -filter_complex 
"aformat=channel_layouts=mono,compand,showwavespic=s=640x120" -f image2 
K:\test\test_%03d.png


However, this gives me just one file, showing the entire duration of the 
sound clip. What am I missing here?


To make it slightly more complex, if this can be done, how can I make the 
last file to have the same timescale as the other ones?
(say the file is 12 seconds, the first image would show second 0 to 8, and I 
want the second one to be as wide, but have the audio data show only in the 
first half of the image.)


thx,

Bouke

-- "ffmpeg version N-79253-g8005b6d 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. 20.100 / 55. 20.100
 libavcodec 57. 34.100 / 57. 34.100
 libavformat57. 30.100 / 57. 30.100
 libavdevice57.  0.101 / 57.  0.101
 libavfilter 6. 40.102 /  6. 40.102
 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 #0.0  stereo
Input #0, wav, from 'K\test\test.wav'
 Metadata
   encoder  Lavf57.30.100
 Duration 000104.08, bitrate 1536 kb/s
   Stream #00 Audio pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 
channels, s16, 1536 kb/s

Output #0, image2, to 'K\test\test_%03d.png'
 Metadata
   encoder  Lavf57.30.100
   Stream #00 Video png, rgba, 640x120 [SAR 11 DAR 163], q=2-31, 200 kb/s, 
75 fps, 75 tbn, 75 tbc

   Metadata
 encoder  Lavc57.34.100 png
Stream mapping
 Stream #00 (pcm_s16le) -> aformat
 showwavespic -> Stream #00 (png)
Press [q] to stop, [?] for help
frame=1 fps=0.0 q=-0.0 Lsize=N/A time=00.01 bitrate=N/A 
speed=0.0452x
video12kB audio0kB subtitle0kB other streams0kB global headers0kB muxing 
overhead unknown




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Re: [FFmpeg-user] Arvato proxy file

2016-01-15 Thread Bouke (VideoToolShed)
- Original Message - 
From: "Carl Eugen Hoyos" <ceho...@ag.or.at>

To: <ffmpeg-user@ffmpeg.org>
Sent: Friday, January 15, 2016 11:06 AM
Subject: Re: [FFmpeg-user] Arvato proxy file



Bouke (VideoToolShed  videotoolshed.com> writes:


This file comes from an Arvato asset management system.


Does the Arvato asset management system allow
playback of the file 39014.mxf you uploaded?


Hi Carl,
I'm not the one behind that system. I've asked the guy who is if he can 
confirm this is indeed 'good'.
(It was created for me to toy with, so I take it this is indeed 'as it is', 
but you never know...)
I'll let you know when I've got a definite answer. Here, direct or on the 
Develop list?


Thanks for looking into it!
Bouke



Carl Eugen

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[FFmpeg-user] Arvato proxy file

2016-01-14 Thread Bouke (VideoToolShed)

Guys,
Have a testfile that won't play nice.
This file comes from an Arvato asset management system.
AFAIK, it's a 'growing' file, that 'should' be playable while it's still 
being created.


FFmpeg (Latest Zerano build, ffmpeg-20160114-git-62dfe1d-win64-static) does 
not like it, and it tells me to upload a sample.
That i've done (Called Arvato_Proxy.zip), if there is no solution here, can 
someone please post this on the Develop list?


Thx,
Bouke

-i Q:\Philip\VRT\test.mxf -codec copy -bsf:a aac_adtstoasc 
Q:\Philip\VRT\test.mov

(Just trying to rewrap to QT)

ffmpeg version N-77836-g62dfe1d Copyright (c) 2000-2016 the FFmpeg 
developers

 built with gcc 5.2.0 (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-libdcadec 
--enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug 
--enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg 
--enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr 
--enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab 
--enable-libvo-aacenc --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. 13.100 / 55. 13.100
 libavcodec 57. 22.100 / 57. 22.100
 libavformat57. 21.101 / 57. 21.101
 libavdevice57.  0.100 / 57.  0.100
 libavfilter 6. 23.100 /  6. 23.100
 libswscale  4.  0.100 /  4.  0.100
 libswresample   2.  0.101 /  2.  0.101
 libpostproc54.  0.100 / 54.  0.100
Input #0, mxf, from 'Q:\Philip\VRT\39014.mxf':
 Metadata:
   modification_date: 2016-01-12 22:08:24
   uid : 6267ef5a-7b76-733d-2767-e74b011fd76c
   generation_uid  : b30ba62e-db12-3c15-877e-0c393e0f9900
   company_name: arvato Systems S4M GmbH
   product_name: LoRes-Transcoder
   product_version : 1.0.0.0.0
   product_uid : d0675d57-0f31-2003-e726-5007c5758762
   application_platform: S4M MediaFramework
   material_package_umid: 
0x060A2B340101010501010D11130098863E05997305A5F2B500110A680FC0

   timecode: 00:00:00:00
 Duration: N/A, start: 0.00, bitrate: N/A
   Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 640x360 [SAR 
1:1 DAR 16:9], 25 fps, 25 tbr, 25 tbn, 50 tbc

   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:1: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:2: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:3: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:4: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:5: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:6: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:7: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0

   Stream #0:8: Audio: aac (LC), 48000 Hz, stereo, fltp
   Metadata:
 file_package_umid: 
0x060A2B340101010501010D11130058863E05997305A5B50900110A680FC0
[mov @ 032dfd00] Codec for stream 0 does not use global headers but 
container format requires global headers
[mov @ 032dfd00] Codec for stream 1 does not use global headers but 
container format requires global headers

Output #0, mov, to 'Q:\Philip\VRT\test.mov':
 Metadata:
   modification_date: 2016-01-12 22:08:24
   uid : 6267ef5a-7b76-733d-2767-e74b011fd76c
   generation_uid  : b30ba62e-db12-3c15-877e-0c393e0f9900
   company_name: arvato Systems S4M GmbH
   product_name: LoRes-Transcoder
   product_version : 1.0.0.0.0
   product_uid : d0675d57-0f31-2003-e726-5007c5758762
   application_platform: S4M MediaFramework
   material_package_umid: 
0x060A2B340101010501010D11130098863E05997305A5F2B500110A680FC0

   timecode: 00:00:00:00
   encoder : Lavf57.21.101
   Stream #0:0: Video: h264 (avc1 / 0x31637661), yuv420p, 640x360 [SAR 1:1 
DAR 16:9], q=2-31, 25 fps, 25 tbr, 12800 tbn, 25 tbc

   Metadata:
 file_package_umid: 

Re: [FFmpeg-user] Why is it possible to set a timecode with 24:00:00:00 and beyond?

2016-01-13 Thread Bouke / VideoToolShed


> On Jan 13, 2016, at 14:17, Robert Krüger  wrote:
> 
> On Wed, Jan 13, 2016 at 12:53 PM, Christoph Gerstbauer <
> christophgerstba...@gmail.com> wrote:
> 
>> FFmpeg allows me to set a timecode of 24:00:00:00 and beyond.
>> 
>> e.g "-timecode 24:00:00:00"
>> 
>> When I make an MXF D10 for example, the value of 24:00:00:00 is still in
>> the file.
>> Most but not all programs ignore this and display 00:00:00:00 correctly.
>> 
>> Here are some sample extractions of the timecode offset value in different
>> mxf d10 files generated by ffmpeg:
>> 
>> PAL 25fps:
>> 
>> -timecode 12:34:56:11  -> in file: 11477b hex -> 1132411 dec
>> 
>> -timecode 10:00:00:00 -> in file: 0dbba0 hex -> 90 dec
>> 
>> -timecode 20:00:00:00 -> in file: 1b7740 hex -> 180 dec
>> 
>> -timecode 23:59:59:24 -> in file: 20f57f hex -> 215 dec
>> 
>> -timecode 24:00:00:00 -> in file: 20f580 hex -> 216 dec -> 24h
>> 
>> -timecode 24:00:00:01 -> in file: 20f581 hex -> 2160001 dec
>> 
>> -timecode 25:00:00:00 -> in file: 225510 hex -> 225 dec -> 25h
>> 
>> 
>> NTSC 30fps
>> 
>> -timecode 23:59:59;29 -> in file: 2782df hex -> 2589407 dec
>> 
>> -timecode 24:00:00;00 -> in file: 2782e0 hex -> 2589408 dec
>> 
>> 
>> Why is this possible to set a timecode after 23:59:59:XX?
>> Is there a usecase?
>> 
> 
> I wasn't able to find any official spec that says what the limit for the
> hour part is but since it in the general case does not define a time of day
> but an offset, why limit it to 25 hours and not use the full 100? On a
> side-note, I have not come across a real-world use case for a timecode of
> that magnitude either.
> 
Some forms of TC just lack the space to store higher numbers IIRC (LTC / VITC, 
perhaps also in MpegII frame headers)
But there is logic in not going higher, as it is ‘time’.
For a real world use case, in the old days of tape, a reelname most of the time 
corresponded with the hour.
So a production with more than 24 tapes needed better administration in reel 
names that had to be written on the tape and box.

Bouke
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Re: [FFmpeg-user] Grab stills on scene change math question

2015-08-25 Thread Bouke (VideoToolShed)

Hi Moritz,
Thanks!


Hi Bouke,

On Tue, Aug 25, 2015 at 13:00:40 +0200, Bouke (VideoToolShed) wrote:
assuming that gt(scene\,0.4) will return a number that I could do math 
on,

but this fails.


ffmpeg's expressions give the select filter a true/false (take or don't
take this frame) decision. In this case: scene detection shows a change
of 0.4 or more, principally.


Ah, that makes sense!


And to make it more complex, can I also grab a shot out of the middle
of a scene? (that would require to get the current and next
framenumber, subtract and divide by two to an integer and add to the
first...)


I wish I knew. I don't think such a filter exists. Part of the issue is
that ffmpeg would need to look ahead, and buffer a lot of data (which
it does for other filters as well).


Well, in theory it should only have to remember the last frame number...
But this is acedemic, as your solution is way easier.


What you could do is to parse the video for scene changes first,


Ok, this I can do.


calculate the images you want, and present them to the select filter in
the second run.


But, I'm clueless on how to do that?.
(Except restarting FFmpeg for each frame with a -ss)
Btw, is there a -ss  option that i can input frames instead of time?

Thanks,
Bouke


Moritz
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Re: [FFmpeg-user] ProRes - MPEG2 bug: color shifting?

2015-08-13 Thread Bouke (VideoToolShed)
 Original Message - 
From: linxs lin...@gmail.com

To: FFmpeg user questions ffmpeg-user@ffmpeg.org
Sent: Thursday, August 13, 2015 7:21 AM
Subject: [FFmpeg-user] ProRes - MPEG2 bug: color shifting?



Hi,

I need helps for color shifting. Is it bug? or I am wrong with the usage?

I get a .mov file and its video format is ProRes 422, 24 fps, 1920x1080,
BT.709.
I put the MOV file in this mail as attachment proress_hq.mov, and only
one frame in this MOV.

Using ffmpeg to convert it to MPEG-2:
ffmpeg -i proress_hq.mov -c:v mpeg2video -b:v 30M -pix_fmt yuv420p my.m2v

I compared them by previewing proress_hq.mov and my.m2v in Edius and I
found there is a little color difference:
proress_hq.mov seems to be a small little brighter than my.m2v and the
color in my.m2v seems to be a little more saturate.


AFAIK, Edius uses Quicktime to decode.
There is a horrible bug in Quicktime that causes gamma shifts.
What you could try is to convert both your outputs to testfiles, both using 
the same codec/format.

Then compare those two. If they are alike, it's Edius that messes up things.

hth,
Bouke


How can i convert this mov to mpeg2 without color shifting?



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Re: [FFmpeg-user] ffmpeg blue ray iso

2015-06-18 Thread Bouke (VideoToolShed)
- Original Message - 
From: Moritz Barsnick barsn...@gmx.net

To: FFmpeg user discussions ffmpeg-user@ffmpeg.org
Sent: Wednesday, June 17, 2015 11:53 PM
Subject: Re: [FFmpeg-user] ffmpeg blue ray iso



On Wed, Jun 17, 2015 at 17:31:56 +0200, Bouke (VideoToolShed) wrote:

But you can use FFmpeg to generate a compliant video file
(Hint, if you need fast, you can use HDV at the cost of a bit of
resolution.)


Sure! There are few, but nice web pages about that. It seems this has
been a top hit on Google for the last few years:

https://irishjesus.wordpress.com/2010/10/17/blu-ray-movie-authoring-in-linux/

The ffmpeg/x264 parameters are probably still valid, as are the
caveats.

May I add: I have read that ImgBurn works fine using Wine (if you
happen to be on x86 and on a platform supporting Wine), and that it
supports UDF 2.6, i.e. proper Blu Ray mastering. I know ImgBurn well
from Windows, and it's an excellent ISO and burning tool, but I have
neither used it under Wine/Linux no for Blu Ray yet (but plan to -
both).

https://appdb.winehq.org/objectManager.php?sClass=applicationiId=4625


WARNING!
My last download of IMGburn came with an installer that installed A LOT of 
adware / browser malware.

IMGburn itself was good, but it took me a wile to get a proper system again.

Bouke




Cheers,
Moritz
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