[FFmpeg-user] Sony FX3 timecode (or the lack of it...)

[Bouke / edit 'B]

This cam writes Mp4 without a QuickTime timecode track, but it does write 
’something’ that ’should’ be (about) correct. (It’s a known issue it’s off by 
one or two frames when slaved.)

ffprobe -v error -print_format json -show_format -show_streams 
/Users/bouke/Downloads/Sony\ FX3/PRIVATE/M4ROOT/CLIP/20240227_C6917.MP4 
Partial return:

{
"index": 2,
"codec_type": "data",
"codec_tag_string": "rtmd",
"codec_tag": "0x646d7472",
"id": "0x3",
"r_frame_rate": "0/0",
"avg_frame_rate": "0/0",
"time_base": "1/3",
"start_pts": 0,
"start_time": "0.00",
"duration_ts": 1261260,
"duration": "42.042000",
"bit_rate": "4664775",
"nb_frames": "1260",
"disposition": {
"default": 1,
"dub": 0,
"original": 0,
"comment": 0,
"lyrics": 0,
"karaoke": 0,
"forced": 0,
"hearing_impaired": 0,
"visual_impaired": 0,
"clean_effects": 0,
"attached_pic": 0,
"timed_thumbnails": 0,
"captions": 0,
"descriptions": 0,
"metadata": 0,
"dependent": 0,
"still_image": 0
},
"tags": {
"creation_time": "2024-02-27T10:22:27.00Z",
"language": "und",
"handler_name": "Timed Metadata Media Handler",
"timecode": "05:37:06:19"
}
}



Now, where is this TC stored? I cannot find it using a HEX editor.
And, more important, FCP, Avid MC and Resolve interpret the “timecode” as Drop 
frame, where there is no indication of that in FFmpeg / FFprobe.
Can anyone shed a light on this?


thx,

Bouke
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Re: [FFmpeg-user] Hyper fast gone in Gyan's binaries

[Bouke / edit 'B]

> On 9 Apr 2024, at 13:45, Gyan Doshi  wrote:
> 
> 
> 
> On 2024-04-09 02:46 pm, Bouke / edit 'B wrote:
>> For my work I try an install of FFmpeg by running it without any params.
>> 
>> It used to be that the return would include ‘Hyper fast’.
>> However, the current binaries from Gyan do not include that, but state 
>> ‘Universal media encoder’.
> 
> That description was changed in 86e8439994d86b33f9b22ad64c4a31a559f05aac
>> 
>> What would be a good, future proof way of testing if FFmpeg is indeed 
>> installed / running properly?
> 
> Try
> 
>  ffmpeg -f lavfi -i nullsrc -t 0.1 -v 0 -progress pipe:1 -f null - | grep end
> 
> If grep returns nothing, then something went wrong.

Hi Gyan, 
Thanks.
But what to youse for FFprobe / FFlay?

Bouke

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[FFmpeg-user] Hyper fast gone in Gyan's binaries

[Bouke / edit 'B]

For my work I try an install of FFmpeg by running it without any params.

It used to be that the return would include ‘Hyper fast’.
However, the current binaries from Gyan do not include that, but state 
‘Universal media encoder’.

What would be a good, future proof way of testing if FFmpeg is indeed installed 
/ running properly?


Bouke



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Re: [FFmpeg-user] ffplay - no delay between playbacks when looping

[Bouke / edit 'B]

 
> On 26 Mar 2024, at 09:57, computermaster360  
> wrote:
> 
> I want to loop a short wav sample, which is an 8-bar melody:
> 
>ffplay -loop 0 sample.wav
> 
> There is a small, but perceptible delay between individual playbacks
> of the sample, which throws the melody off. Is there a way to make the
> playback completely continuous with no delay?

Not tested, but I would try to do the loop in FFmpeg and pipe to FFplay.
This way FFplay will get just samples, bypassing whatever happens on restart 
playback of the file.

Bouke


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

[Bouke / edit 'B]

> 
> On 19 Feb 2024, at 19:14, Mark Filipak  wrote:
> 
> On 19/02/2024 13.11, Bouke / edit 'B wrote:
>>> On 19 Feb 2024, at 19:04, Mark Filipak  wrote:
>>> 
>>> On 19/02/2024 12.59, Bouke / edit 'B wrote:
>>>>> 
>>>>> On 19 Feb 2024, at 18:47, Mark Filipak  wrote:
>>>>> 
>>>>>  I'm not writing a book.
>>>>> That's something someone made up.
>>>> Yes, the ’someone' was you. Search the archives.
>>> 
>>> No, that was not me. You may have misunderstood. I was/am writing 
>>> documentation, not a book.
>> My bad, my apologies if you are indeed correct. Please dive into the 
>> archives.
>>> Kindly stop harassing me.
>> IIRC, you wrote not so long ago that you post not more than 6 entries a year 
>> to this list.
>> What should I think about you?
> 
> I think you might conclude that something has changed.

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.

Bouke


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

[Bouke / edit 'B]

> On 19 Feb 2024, at 19:04, Mark Filipak  wrote:
> 
> On 19/02/2024 12.59, Bouke / edit 'B wrote:
>>> 
>>> On 19 Feb 2024, at 18:47, Mark Filipak  wrote:
>>> 
>>>  I'm not writing a book.
>>> That's something someone made up.
>> Yes, the ’someone' was you. Search the archives.
> 
> No, that was not me. You may have misunderstood. I was/am writing 
> documentation, not a book.

My bad, my apologies if you are indeed correct. Please dive into the archives.

> Kindly stop harassing me.

IIRC, you wrote not so long ago that you post not more than 6 entries a year to 
this list.
What should I think about you?



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

[Bouke / edit 'B]

> 
> On 19 Feb 2024, at 18:47, Mark Filipak  wrote:
> 
>  I'm not writing a book.
> That's something someone made up.

Yes, the ’someone' was you. Search the archives.




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Re: [FFmpeg-user] The Guild

[Bouke / edit 'B]

>  I'd pay $25/month to belong.

I pay 25 bucks for each troll head.

(Or 50 per ball, makes for more fun)

Bouke 

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Re: [FFmpeg-user] '-t' versus '-to'

[Bouke / edit 'B]

> On 14 Jan 2024, at 21:22, Carl Zwanzig  wrote:
> 
> On 1/13/2024 10:26 PM, Mark Filipak wrote:
>> For example:
>> -ss 20.061708 -to 5570.815250
>> would be
>> -frametrim 481..134046
>> meaning frame 481 thru frame 134046.
> 
> Although IME it much more common to specify the frame number after a dot* but 
> ffmpeg doesn't seem to do that. (There may be North American vs Europe vs 
> Asiatic regional conventions on this.)
> 
> *hh:mm:ss.ff, or occasionally hh:mm:ss:ff, but the latter is ambiguous if 
> either hh or ff are omitted
> 
> Perhaps some day I'll look at the command arg time parser to see what 
> assumptions are made and where they're documented.

Don’t feed the trolls.

Bouke


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Re: [FFmpeg-user] sntsc

[Bouke / edit 'B]

> On 26 Dec 2023, at 21:13, Mark Filipak  wrote:

How about stopping trolling if you know it all?

The ‘imdb’ in your mail address makes you an even bigger jerk.

Bouke



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

[Bouke / edit 'B]

> On 19 Dec 2023, at 17:16, Erik Dobberkau  wrote:
> 
>> 
>> Trust me, this command (besides audio parsing, depending on sources) has
>> made thousands of broadcast compliant files over here, where XDCAM is also
>> mandatory.
>> 
> 
> Does that mean the issue mentioned here:
> https://www.mail-archive.com/ffmpeg-user@ffmpeg.org/msg26313.html 
>  (and
> its follow-up messages) is not an issue anymore?

No clue, only thing I know is that the stupid broadcasters over here accept 
files created as I showed
Bouke
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Re: [FFmpeg-user] FFmpeg default processing(invisible processing)

[Bouke / edit 'B]

Top posting, as my mail client does now allow for decent parsing, and I’m fed 
up with that.

My command to create ‘’proper” XDcam HD 50Mb, as required by the broadcasters 
here.
Read between the filenames, it does work.

/Applications/ffmpeg -hide_banner -threads 0 -i /Users/bouke/yadda.mov -f lavfi 
-i anullsrc=channel_layout=mono:sample_rate=48000 -map 0:0 -map 0:1 -map 0:1 
-map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -map 1:0 -c:a pcm_s24le -ar 48000 
-map_channel 0.1.0:0.1 -c:a pcm_s24le -ar 48000 -map_channel 0.1.1:0.2 -c:a 
pcm_s24le -map_channel 1.0.0:0.3 -c:a pcm_s24le -map_channel 1.0.0:0.4 -c:a 
pcm_s24le -map_channel 1.0.0:0.5 -c:a pcm_s24le -map_channel 1.0.0:0.6 -c:a 
pcm_s24le -map_channel 1.0.0:0.7 -c:a pcm_s24le -map_channel 1.0.0:0.8 -af 
volume=-11.4dB -vf setfield=tff,scale=1920x1080:flags=lanczos -r 25 -flags 
+ilme+ildct -top 1 -c:v mpeg2video -b:v 5k -maxrate 5k -bufsize 3835k 
-minrate 5k -pix_fmt yuv422p -profile:v 0 -level:v 2 -t 26 -timecode 
00:00:00:00 /Volumes/Data/magweg/Mars_Mars-Delight_26_1_19-12-2023_HD.mxf -y 

 Trust me, this command (besides audio parsing, depending on sources) has made 
thousands of broadcast compliant files over here, where XDCAM is also mandatory.

Bouke 

> On 19 Dec 2023, at 06:56, 大石脩斗  wrote:
> 
> I'm trying to use ffmpeg to generate an MXF file that can be played on an
> XDCAM XDS-PD2000.
> Please tell me what default processing is performed in ffmpeg when no
> options are entered.
> 
> [background]
> Sony's check tool outputs an error saying "MXF metadata contains
> non-standard values".
> The data output by the editing machine is OK with the check tool,
> The data generated with ffmpeg will be rejected by the check tool.
> The metadata looks the same for both.
> Please let me know if you are processing anything other than the meta
> information that is being output
> 
> 【Data output by editing machine」
> aftertext.txt
> 
> 【Data output by ffmpeg】
> testffmpeg.txt
> commondlie
> ffmpeg.exe -i input\base.MXF -c:v mpeg2video -r 29.97 -pix_fmt yuv422p
> -aspect 16:9 -flags +ildct+ilme -b:v 5000 -minrate 5000 -maxrate
> 5000 -bufsize 17825792 -bf 2 -metadata creation_time=now -video_format
> component -g 15 -timecode 00:59:45:00 -ac 1 -map 0:0 -map 0:1 -map 0:1 -map
> 0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -map 0:1 -c:a pcm_s24le -ar 48000
> -f mxf -y output/output_test.mxf
> 
> Shuto
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Re: [FFmpeg-user] why reducing the resolution a percentage doesn't reduce the video the same proportion?

[Bouke / edit 'B]

> On 14 Mar 2023, at 14:03, Natalia Molinero Mingorance 
>  wrote:
> 
> what do you mean by " Why is the output size important if you use -preset
> ultrafast?  “

That will mean quick and dirty encoding, at the cost of a higher bitrate.
-preset slow will make the file significant smaller, at the cost of more 
processing time / cpu use.
(And thus energy use, but if you stream too large files a gazillion times, you 
might want to invest in high quality encoding instead of quick ’n dirty.)
It’s a ‘would you like to be bitten by the cat or by the dog’ situation.

Bouke



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Re: [FFmpeg-user] why reducing the resolution a percentage doesn't reduce the video the same proportion?

[Bouke / edit 'B]

On 14 Mar 2023, at 13:07, Phil Rhodes via ffmpeg-user  
wrote:
> 
> Also, it depends what you mean by 50% of the resolution. If your original 
> video is 100x100 pixels and you reduce it to 50x50, then you have half the 
> resolution, but you only have one quarter the pixels.

But, the image will still have a tree, a house and a dog.
>From what I have understood, the resolution in terms of pixels does not 
>compute to output file size (Unless you work uncompressed of course.)

> P
>On Tuesday, 14 March 2023 at 09:37:36 GMT, Natalia Molinero Mingorance 
> mailto:eng.nataliamolin...@gmail.com>> wrote: 
>  
> 
> I'm using this command:
> 
> -y -r -i $inVideoUri -movflags faststart -c:v libx265 -s
> $videoResolution -c:a copy -preset ultrafast  $outPutUri

/// 
> some headers or metadata added during the process but I would like to know
> how to estimate the final video size.

Why is the output size important if you use -preset ultrafast?

Bouke


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Re: [FFmpeg-user] Noticeable transitions between audio chunks

[Bouke / edit 'B]

Oh wait...
> 
>> then I get raw PCM of each chunk and replace some parts of PCM with sine
>> wave data (adding beeps) and then I transform that PCM chunks to mp3 chunks
>> and write those to stream. The final process of concating audio parts is *NOT
>> *performed by ffmpeg, I just write data chunks to the destination stream.

Are you trying to concat mp3 chunks? That seems like a bad idea,
Concat the raw PCM, then convert that to mp3

bouke


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Re: [FFmpeg-user] Noticeable transitions between audio chunks

[Bouke / edit 'B]

> On 7 Feb 2023, at 11:05, Alexander Bieliaev via ffmpeg-user 
>  wrote:
> 
> I am processing audio chunks programmatically by using the ffmpeg library
> for C#. First I divide the input audio of .wav format into chunks of 1
> minute each (I can't process the whole audio for specific reasons), then
> prepend it's header to each chunk so it can be recognized and processed,

I don’t get it. If you speak C, and you have uncompressed Wave input, is as 
simple as opening the file, find the DATA riff and copy out X bytes (depending 
on the bit depth / sample rate)


> then I get raw PCM of each chunk and replace some parts of PCM with sine
> wave data (adding beeps) and then I transform that PCM chunks to mp3 chunks
> and write those to stream. The final process of concating audio parts is *NOT
> *performed by ffmpeg, I just write data chunks to the destination stream. I
> am facing the problem that there are noticeable transitions between 1
> minute chunks in result audio(clicks/silence/change of volume/shifting).
> How can I smooth out the start/end of each chunk so when I'm putting them
> together there are no noticeable transitions?

From experience I know that the size of -f segment for Wave does NOT result in 
a sample accurate amount of data, not sure if that is what you are facing. 
(Should not matter if there is a pure concat of the data.)
It ’should’ be seamless.

So, if I can  do this kind of stuff in pure python, why find the cause and not 
do it yourself in C?

Bouke

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[FFmpeg-user] I love the smell of assholes in the morning

[Bouke / edit 'B]

Bonus points if you get the reference.

I also like the smell in the afternoon, and evening, and night.
(Hunting targets.)

IOW:

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

[Bouke / edit 'B]

On 31 Dec 2022, at 10:41, Michael Koch  wrote:
> 
> Am 31.12.2022 um 10:31 schrieb 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.
> 
> Do you have an example? I have used command lines _much_ longer than the 
> above example, and never found a limitation.
> 
> Michael

The limitation is 8091 chars, and afaik only exists on Windows. (Mac is fine, I 
don’t know about Linux, but I suspect it will be no issue there too.)
My latest issue had a line of more than 64000 chars, example from 
filter_complex (that caused the huge length) below.

Bouke

map 0:0 -filter_complex 
[0:0]drawtext=enable='between(t,0.0302083334,75.180166)':timecode='00\:49\:29\:11':fontfile='/Volumes/Data/PycharmProjects/LTC_convert/src/main/resources/base/courbd.ttf':r=25:x=(w-text_w)/2:y=h-th:
 
y=h-(2*lh):fontsize=54.0:fontcolor=white:borderw=2.7:bordercolor=#101010@0.8,drawtext=enable='between(t,75.190167,75.780166)':text='--
 NO LTC 
!':fontfile='/Volumes/Data/PycharmProjects/LTC_convert/src/main/resources/base/courbd.ttf':x=(w-text_w)/2:y=h-th:
 
y=h-(2*lh):fontsize=54.0:fontcolor=white:borderw=2.7:bordercolor=#101010@0.8,drawtext=enable='between(t,75.790166,299.980125)':timecode='00\:47\:41\:21':fontfile='/Volumes/Data/PycharmProjects/LTC_convert/src/main/resources/base/courbd.ttf':r=25:x=(w-text_w)/2:y=h-th:
 
y=h-(2*lh):fontsize=54.0:fontcolor=white:borderw=2.7:bordercolor=#101010@0.8,drawtext=enable='between(t,300.030041667,822.939770833)':timecode='00\:47\:41\:21':fontfile='/Volumes/Data/PycharmProjects/LTC_convert/src/main/resources/base/courbd.ttf':r=25:x=(w-text_w)/2:y=h-th:
 
y=h-(2*lh):fontsize=54.0:fontcolor=white:borderw=2.7:bordercolor=#101010@0.8,drawtext=enable='between(t,822.949770833,1138.661916667)':text='--
 NO LTC 
!':fontfile='/Volumes/Data/PycharmProjects/LTC_convert/src/main/resources/base/courbd.ttf':x=(w-text_w)/2:y=h-th:
 y=h-(2*lh):fontsize=54.0:fontcolor=white:borderw=2.7:bordercolor=#101010@0.8, 
and so on….
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Re: [FFmpeg-user] Changing scan type (progressive<->interlaced) without reencoding.

[Bouke (Edit 'B)]

It’s no problem at all.
Btw, it’s very common to store progressive as interlaced, as broadcast still 
requires interlaced material, no matter how it is shot.

From memory, just do a -c copy and you’ll end up with the source in a 
progressive container.

Bouke


> On 05 Dec 2022, at 11:56, Steinar Apalnes  wrote:
> 
> Hi,
> 
> As the title says, is this at all possible? My concern is mostly with mpeg2
> and h264 that have content of one type but encoded/marked as another. Most
> notably content that has progressive video but stored interlaced. So I'm
> probably talking about a bit stream filter here because just setting the
> "field_order", "top" etc. flags when rewrapping does nothing.But I have not
> found anything in the official docs that suggest this is currently possible
> so it would be nice if some of you developers could shed some light on this.
> 
> Thanks,
> Steinar Apalnes
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Re: [FFmpeg-user] Wrong TC read with doubled Framenumber read out when input is 50p

[Bouke (Edit 'B)]

> On 06 Nov 2022, at 21:35, Björn Büchel via ffmpeg-user 
>  wrote:
> 
> Hi Bouke, 
> Thanks for the reply. You are right, that makes no sense. But this is what I 
> found out. 
> I checked the TC with Avid Media Composer as well as with Resolve and 
> Premiere. 
> In all three Apps the TC is identical. 
> Björn 

Hi Björn,
Did you try QTchange as well? (If not, please do, the free demo will do just 
fine.)
https://www.videotoolshed.com/handcrafted-timecode-tools/qtchange/

It’s partly based on FFmpeg, so it ‘should’ give the same results. 
There are a couple of thousend end users, including some major broadcasters / 
studios, and I have not have any complaint like you describe…

So, this is strange.

Bouke



> 
> —-
> 
> Björn Büchel
> Am Hüpplingsgraben 5
> 42799 Leichlingen
> 0152 / 051 051 00
> bjoernbuec...@me.com
> 
> 
>> Am 06.11.2022 um 11:10 schrieb 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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>> 
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Re: [FFmpeg-user] encode to RAW video

[Bouke / edit 'B]

> On 24 Oct 2022, at 14:29, Naveen.B  wrote:
> 
>> 
>> 
>> 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.
>> 
>> I tried as you suggested, I get the below error,
> 
> C:\Naveen\projects\DMS\software\ffmpeg_full\ffmpeg\bin>ffmpeg -pixel_format
> gray10le -s 1600x1300 -r 30 -i CapturedImage-%03d.raw -c:v rawhide -crf 1
> raw_video.raw

Rawhide?
WTF?
Rawvideo does not carry a pix format afaik.


> ffmpeg version 2022-06-20-git-56419428a8-full_build-www.gyan.dev Copyright
> (c) 2000-2022 the FFmpeg developers
>  built with gcc 11.3.0 (Rev1, 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-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-libmfx --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  57. 27.100 / 57. 27.100
>  libavcodec 59. 33.100 / 59. 33.100
>  libavformat59. 25.100 / 59. 25.100
>  libavdevice59.  6.100 / 59.  6.100
>  libavfilter 8. 41.100 /  8. 41.100
>  libswscale  6.  6.100 /  6.  6.100
>  libswresample   4.  6.100 /  4.  6.100
>  libpostproc56.  5.100 / 56.  5.100
> Input #0, image2, from 'CapturedImage-%03d.raw':
>  Duration: 00:00:01.20, start: 0.00, bitrate: N/A
>  Stream #0:0: Video: rawvideo (Y1[0][10] / 0xA003159), gray10le,
> 1600x1300, 25 fps, 25 tbr, 25 tbn
> [NULL @ 023c862e6740] Unable to find a suitable output format for
> 'raw_video.raw'
> raw_video.raw: Invalid argument
> 
> C:\Naveen\projects\DMS\software\ffmpeg_full\ffmpeg\bin>
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Re: [FFmpeg-user] Make white of photo transparant

[Bouke / edit 'B]

Beledig me niet, lijk met niet in lijn met je princiepes / de Heer, iig siert 
het niemand.

Bouke / edit 'B

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/

> On 29 Jun 2022, at 13:45, Cecil Westerhof via ffmpeg-user 
>  wrote:
> 
> Bouke / Videotoolshed  writes:
> 
>> Pff, make it an option.
> 
> You probably would tell a homeless person that is hungry and cold:
> just put on clothes and eat something.
> (Without providing him anything.)
> 
> -- 
> Cecil Westerhof
> Senior Software Engineer
> LinkedIn: http://www.linkedin.com/in/cecilwesterhof
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Re: [FFmpeg-user] Get the file size in advance when encoding from one another format

[Bouke (Edit 'B)]

> 
> Please help me do this; How do I get the file size in advance if I encode
> any media format to another format so that I can set the Header
> Content-Length in the header. What is the formula for this? I'm new to
> FFMpeg.

Didn’t you already get flamed on this very subject not very long ago?

Bouke

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

[Bouke / edit 'B]

> On 20 Apr 2022, at 19:50, Shubham Tiwari  wrote:
> 
> I think you are not reading the message properly.
 
I think I do.

> If you notice, the size
> of the file i am passing is 2 mb, the return file size is 176 mb. Does 88
> times increase seem reasonable to you?

Yes, as far as it goes for your command line, this is to be expected, and 
totally comprehensive.

> I want the file to be in approximately similar quality and file size should
> go at max 2 to 3 times. In this case upto 6 mb file size is fine.

State what you want, do you want to trim (cut out parts), just to forward it to 
other people?
So, what do you want, what are you after?
> 
> I am doing this because we have a redaction logic to be executed on
> approximately 1 million audio files per day.  And all these calls are
> stored in aws s3. so storing multiple files of size 176 mb doesn't seem to
> be a good idea.

Again, what are you after? If you need ’some’ kind of computer logic / 
analysis, uncompressed Wave / PCM CAN be faster / more efficient than 
compressing it.
In that case, file size does not matter if it’s local, it’s just math on what 
is cheaper / more efficient.

BUT, since you don’t give any clue what / who has to process the output of your 
work, trust me, hire a specialist to drag this out of you.

I’m trying to help you, but for that, you need to provide more input.

Bouke / edit 'B

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/


> 
> Concerning my lack of knowledge, I read the documentation page multiple
> times but couldn't find something that gives me the right information.  In
> the end I reached out here. Concerning my lack of knowledge, probably some
> help would be better instead of passing remarks.
> 
> On Wed, Apr 20, 2022 at 11:07 PM Bouke / Videotoolshed <
> bo...@videotoolshed.com> wrote:


Bouke / edit 'B

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

[Bouke / edit 'B]

> On 20 Apr 2022, at 19:50, Shubham Tiwari  wrote:
> 
> 1 million audio files per day.

Hire a specialist. Now. Immediately. Pay him / her well. Do NOT hesitate. Do 
NOT think, just do it. Right now.
It WILL pay off.
Did I mention you DO NOT HAVE time to waste?


Bouke / edit 'B

videotoolshed.com
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+31 6 21817248
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Re: [FFmpeg-user] Large Sized output files recieved while encoding the audio

[Bouke / edit 'B]

Hi Shubham,
When I copy your mail below (not done on this list, mind you!), it’s exactly as 
long as it was before. What do you expect?
Bouke



> On 20 Apr 2022, at 15:01, Shubham Tiwari  wrote:
> 
> Hi,
> 
> When I run ffmpeg command on a wav file, the received output file is of
> large size as compared to the original file. The command and the output are
> below,
> 
> *FFMPEG command*
> 
> % ~/Downloads/audio/ffmpeg -i call-redacted.wav output.wav
> 
> ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
> 2000-2021 the FFmpeg developers
> 
>  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
> 
>  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
> --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
> --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
> --enable-libfreetype --enable-libgsm --enable-libmodplug
> --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> --enable-libopus --enable-librubberband --enable-libshine
> --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-libtwolame --enable-libvidstab --enable-libvmaf
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
> --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
> --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
> --pkg-config-flags=--static --disable-ffplay
> 
>  libavutil  56. 70.100 / 56. 70.100
> 
>  libavcodec 58.134.100 / 58.134.100
> 
>  libavformat58. 76.100 / 58. 76.100
> 
>  libavdevice58. 13.100 / 58. 13.100
> 
>  libavfilter 7.110.100 /  7.110.100
> 
>  libswscale  5.  9.100 /  5.  9.100
> 
>  libswresample   3.  9.100 /  3.  9.100
> 
>  libpostproc55.  9.100 / 55.  9.100
> 
> Input #0, mp3, from 'call-redacted.wav':
> 
>  Metadata:
> 
>encoder : Lavf58.45.100
> 
>  Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s
> 
>  Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s
> 
> Stream mapping:
> 
>  Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
> 
> Press [q] to stop, [?] for help
> 
> Output #0, wav, to 'output.wav':
> 
>  Metadata:
> 
>ISFT: Lavf58.76.100
> 
>  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, stereo,
> s16, 256 kb/s
> 
>Metadata:
> 
>  encoder : Lavc58.134.100 pcm_s16le
> 
> size=   24569kB time=00:13:06.17 bitrate= 256.0kbits/s speed=2.97e+03x
> 
> video:0kB audio:24569kB subtitle:0kB other streams:0kB global headers:0kB
> muxing overhead: 0.000310
> 
> 
> *Input and output file sizes*
> 
> du -sh call-redacted.wav
> 
> 6.5M call-redacted.wav
> 
> du -sh output.wav
> 
> 24M output.wav
> 
> 
> Alternatively, when I run the same command but change the output
> file's extension to mp3, the correct sized file is returned. But the time
> taken (6 seconds) in this case is quite higher than the previous case(less
> than 1 second). The command and output is below,
> 
> 
> 
> ~/Downloads/audio/ffmpeg -i call-redacted.wav output.mp3
> 
> ffmpeg version 4.4.1-tessus  https://evermeet.cx/ffmpeg/  Copyright (c)
> 2000-2021 the FFmpeg developers
> 
>  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
> 
>  configuration: --cc=/usr/bin/clang --prefix=/opt/ffmpeg
> --extra-version=tessus --enable-avisynth --enable-fontconfig --enable-gpl
> --enable-libaom --enable-libass --enable-libbluray --enable-libdav1d
> --enable-libfreetype --enable-libgsm --enable-libmodplug
> --enable-libmp3lame --enable-libmysofa --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg
> --enable-libopus --enable-librubberband --enable-libshine
> --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-libtwolame --enable-libvidstab --enable-libvmaf
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwebp
> --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
> --enable-libzimg --enable-libzmq --enable-libzvbi --enable-version3
> --pkg-config-flags=--static --disable-ffplay
> 
>  libavutil  56. 70.100 / 56. 70.100
> 
>  libavcodec 58.134.100 / 58.134.100
> 
>  libavformat58. 76.100 / 58. 76.100
> 
>  libavdevice58. 13.100 / 58. 13.100
> 
>  libavfilter 7.110.100 /  7.110.100
> 
>  libswscale  5.  9.100 /  5.  9.100
> 
>  libswresample   3.  9.100 /  3.  9.100
> 
>  libpostproc55.  9.100 / 55.  9.100
> 
> Input #0, mp3, from 'call-redacted.wav':
> 
>  Metadata:
> 
>encoder : Lavf58.45.100
> 
>  Duration: 00:13:06.38, start: 0.138125, bitrate: 64 kb/s
> 
>  Stream #0:0: Audio: mp3, 8000 Hz, stereo, fltp, 64 kb/s
> 
> Stream mapping:
> 
>  Stream #0:0 -> #0:0 (mp3 (mp3float) -> mp3 (libmp3lame))
> 
> Press [q] to stop, [?] for help
> 
> Output #0, mp3, to 'output.mp3':
> 
>  Metadata:
> 
>TSSE: Lavf58.76.100
> 
>  Stream #0:0: 

Re: [FFmpeg-user] Fluorescent White Balance Video Filters

[Bouke / edit 'B]

On 19 Apr 2022, at 20:44, Jim DeLaHunt  wrote:
> 
> Not that the FFmpeg tool is stupid overall — it is quite capable and useful 
> in many ways. But there are aspects of the tool, and its documentation, and 
> its contributors' behaviour, and the project culture, which are indeed quite 
> stupid.

Hear hear!

Now, IMHO, audio mapping in FFmpeg is plain stupid. FFMBC’s approach what so 
much easier / better.
(On top of that, ripping just sound from an AV file with FFMBC was WAY faster, 
as FFMBC was able to ignore the video data, where FFmpeg seems to feel a need 
to plough trough it.)

Bouke

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Re: [FFmpeg-user] Fluorescent White Balance Video Filters

[Bouke / edit 'B]

> On 19 Apr 2022, at 21:00, Carl Zwanzig  wrote:
> 
> On 4/19/2022 11:52 AM, Bouke / edit 'B wrote:
>>> How would you do that when there are only parameters for R and B?
> 
>> AFAIK, YUV = R-Y B-Y, so it’s just math.
>> Doing corrections in RGB is simpler for a mortal,
> 
> and in analog :)
> 
> What I'm getting at is there are no green terms on that filter's controls so 
> I don't see a way to directly adjust green (my point). This makes sense for a 
> filter that's trying to correct what we call white balance- like a 
> three-legged stool, you only ever have to adjust two legs to make the top 
> level.
> 
Could be I’m missing the point, but cutting / extending two legs of a three 
legged chair will close to NEVER make the desired height.
But yes, it will level it if needed, to stick to the reference.

I'm missing out on the ‘analog’ reference. AFAIK, YUV is still in use in 
(digital) codecs.
And it’s hard to grasp if you do color correction, and are used to fancy tools 
like Resolve. (That used to be a 100.000+ tool not so long ago.)

Bouke



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Re: [FFmpeg-user] Fluorescent White Balance Video Filters

[Bouke / edit 'B]

> On 19 Apr 2022, at 20:41, Carl Zwanzig  wrote:
> 
> On 4/19/2022 11:37 AM, Paul B Mahol wrote:
>> colorcorrect filter operates in YUV colorspace and not RGB colorspace
>> So green tint can be adjusted with it.
> 
> How would you do that when there are only parameters for R and B?
> 
> z!

AFAIK, YUV = R-Y B-Y, so it’s just math.
Doing corrections in RGB is simpler for a mortal, but should / could result 
exactly the same. Now, let someone write something to convert the matrix to 
support RGB.

Bouke

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Re: [FFmpeg-user] linear loudnorm

[Bouke / edit 'B]

>> 
>> I'm trying to nomalize an audio file with FFmpeg. I'm using the loudnorm
>> filter. The source loudness is -23 LUFS and I want to make it -17 LUFS.
>> As far as I know, loudnorm has 2 modes of normalizing audio: linear and


>> 2. Encode the audio with:
>> 
>> ffmpeg -i input.wav -filter:a
>> loudnorm=I=-17:TP=-1:LRA=9:measured_I=-22.72:measured_TP=-2.67:measured_LRA=6.10:measured_thresh=-33.31:offset=-0.05:linear=true:print_format=summary

Why not just add 6 dB to the input?

That ’should’ do the trick just fine imho. (As one LU should correspond with 1 
dB)


Bouke


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Re: [FFmpeg-user] Video Notation, A Video Lingua Franca

[Bouke / edit 'B]

> On 9 Feb 2022, at 18:32, Mark Filipak  wrote:
> 
> Hello, All,

Bye Mark.go away!
Now. No time to waste. Unsub / close your internet account / kill yourself / 
hire someone to do that for you, I don’t care, just, stop.

Bouke

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Re: [FFmpeg-user] Please advise

[Bouke / edit 'B]

> On 30 Dec 2021, at 15:16, Reindl Harald  wrote:
> 
> 
> 
> Am 30.12.21 um 15:05 schrieb Bouke / edit 'B:
>>> On 30 Dec 2021, at 14:58, Reindl Harald  wrote:
>>> 
>>> it's amazing that you
>> Hey Harald,
>> Are you allright? You almost sound polite!
> 
> no, not if you read the whole sentence

I did, you wrote:
"it's amazing that you managed to subscribe but not to understand a ML”
I fail to see a good insult in that line.

> i forgot to mention that "New post" is a idiotic subject for any email no 
> matter in what context

I agree, but since these are your words, I could not care less.

> 
>> Is everything well? Should someone call a doctor / inform the authorities?
> 
> mind your own business

I AM. My business is (sometimes) to make this world a better place.
You are trying to do the same thing, in the most irritating offensive annoying 
way I’ve ever encountered.
(Not that I disagree about the stupidity in this thread, but that’s beside the 
point.)

Bouke

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Re: [FFmpeg-user] Please advise

[Bouke / edit 'B]

> On 30 Dec 2021, at 14:58, Reindl Harald  wrote:
> 
> it's amazing that you 

Hey Harald,
Are you allright? You almost sound polite!
Is everything well? Should someone call a doctor / inform the authorities?

Bouke


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Re: [FFmpeg-user] Concatenating video cuts - audio gets out of sync...

[Bouke (Edit 'B)]

> On 17 Dec 2021, at 11:08, Bo Berglund  wrote:
> 
> Can I keep the copy for video but re-encode for audio?
> Some ffmpeg command suggestion?

-c:v copy -c:a ‘something’, but no clue if it will help you…

Bouke
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