Re: [FFmpeg-user] Convert srt Subtitle file into ttml subtitle

[Christian Ebert]

* R J on Tuesday, October 05, 2021 at 16:05:16 +0530:

I want to convert a .srt file into ttml file.

I tried using below command.

ffmpeg -i my_srt.srt -y srt-to-ttml.xml -v verbose


How is ffmpeg supposed to know to which format you intend to 
convert?


Either

ffmpeg -i my_srt.srt -f ttml srt-to-ttml.xml

or

ffmpeg -i my_srt.srt srt-to-ttml.ttml

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Re: [FFmpeg-user] How to check if a youtube URL is valid before starting download?

[Christian Ebert]

* Bo Berglund on Friday, July 09, 2021 at 08:14:28 +0200:

 wrote:

It did. See the line I highlighted above. All you need to do now is
something like filter the output of ffmpeg like:

ffmpeg -i bluearrow.mp4 -vf "freezedetect=n=0.01:d=5" -map 0:v:0 -f null - | 
grep -q 'lavfi.freezedetect.freeze_start: '
nofreeze=$?


I tested the command using two different video snippets as input (with and
without the blue arrow) and the screen output shows the line:

[freezedetect @ 0x555be8939800] lavfi.freezedetect.freeze_start: 0
for *only* the bluearrow.mp4 video so this is a step forward!

However, the grep command does the same (nothing) in both cases!


ffmpeg writes the log output to stderr, so redirect it to stdout:

ffmpeg -i bluearrow.mp4 -vf "freezedetect=n=0.01:d=5" -map  0:v:0 -f null - 
2>&1 | grep -q 'lavfi.freezedetect.freeze_start: '


I tested by replacing the | grep ... stuff with a redirect to file instead
ffmpeg -hide_banner -i bluearrow.mp4 -vf "freezedetect=n=0.01:d=5" -map 0:v:0 -f
null -  > freezeoutput.txt

This showed the same screen output but the freezeoutput.txt file is created but
is empty!
So grep has no chance of ever detecting anything


See above.

But for writing the log into a file look into ffmpeg's -report option, that's 
what it's for.


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Re: [FFmpeg-user] Trying to stream to rtmp from my rsp4 and a GoPro

[Christian Ebert]

* Mark Filipak (ffmpeg) on Monday, February 15, 2021 at 21:02:33 -0500:

On 02/15/2021 06:29 PM, Mike Soultanian wrote:
-bigsnip-

You can - easily - let the audio stream start with 0 but the problem
is that the result will not play in-sync, see the setps documentation.


What is 'setps'? A filter, a protocol, a muxer, what?


A typo for 'setpts'.


I've searched through all the docs and can't find it.


Because the typo is not in the docs ;-)

https://ffmpeg.org/ffmpeg-filters.html#setpts_002c-asetpts

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Re: [FFmpeg-user] Correct conversion of yuvj420p?

[Christian Ebert]

* Peter B. on Thursday, October 01, 2020 at 14:16:02 +0200:

On 29.09.20 09:56, Christian Ebert wrote:

How about doing quick diagnosis with ffprobe before you start,
something like:

ffprobe -v error \
-print_format default=noprint_wrappers=1:nokey=1 \
-select_streams V -show_entries stream=pix_fmt

and make the command depend on the result?


Thanks for the suggestion, but that actually is what I'm doing right
now - yet it means to treat parts of the batches with different
conditions/recipes.


I would be very surprised if you could use the same command on
thousands of videos.


Additionally, in that concrete situation, the source videos (several
thousand...) are stored on an S3-type object based storage (non-Amazon
dialect). So used Mediainfo to avoid having to pull several hundred
Terabyte of videos just to get their tech-MD.


What's wrong with that? But then again I don't understand: You
want to process _all_ the thousands of files, so you need them
anyway?


And Mediainfo cannot tell me what FFmpeg considers yuvj420p, because...


If you can run Mediainfo on them (remotely?), I'd be surprised if
you couldn't run ffprobe on them - and if you haven't
built/installed ffprobe, even parsing `ffprobe -i input` lets you
detect the pix_fmt.


Even if it wasn't for the "S3-extra-fun", I'm running into this issue
every now and then in different video collections - and every "if -
then - else" adds extra points of failure (and work).


I would think if - then - else is the way to _avoid_ points of
failure, but most likely I don't understand you setup and/or
intent, sorry.

Good luck.

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Re: [FFmpeg-user] Correct conversion of yuvj420p?

[Christian Ebert]

Hi Peter,

* Peter B. on Monday, September 28, 2020 at 20:42:56 +0200:

...but I still have to find out which files are interpreted as
"yuvj420p" by ffmpeg - and then fish them out and treat them
with a separate command, since I have batches with and without
color_range set, therefore "-pix_fmt +" throws a (correct)
error when trying to convert to FFV1 :)


How about doing quick diagnosis with ffprobe before you start,
something like:

ffprobe -v error \
-print_format default=noprint_wrappers=1:nokey=1 \
-select_streams V -show_entries stream=pix_fmt

and make the command depend on the result?

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Re: [FFmpeg-user] prores_ks vendor: apl0 or ap10 ?

[Christian Ebert]

* Paul B Mahol on Tuesday, September 01, 2020 at 10:26:15 +0200:

On 9/1/20, Christian Ebert  wrote:

man ffmpeg-codecs recommends apl0 as prores_ks -vendor:
http://ffmpeg.org/ffmpeg-all.html#Private-Options-for-prores_002dks

whereas the WIKI says ap10:
https://trac.ffmpeg.org/wiki/Encode/VFX#Prores

Which one is it?


apl0,


Thank you for the clarification.


and it is not recommendation.


Indeed, it is more a biblical apple temptation, albeit, at least
in my experience, without any consequences ...

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[FFmpeg-user] prores_ks vendor: apl0 or ap10 ?

[Christian Ebert]

Hi,

man ffmpeg-codecs recommends apl0 as prores_ks -vendor:
http://ffmpeg.org/ffmpeg-all.html#Private-Options-for-prores_002dks

whereas the WIKI says ap10:
https://trac.ffmpeg.org/wiki/Encode/VFX#Prores

Which one is it?

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[FFmpeg-user] aresample in_channel_layout has no effect

[Christian Ebert]

Hi,

I'm trying to downsample 4 channel (unknown layout) pcm to
stereo. However, aresample's in_channel_layout option has no
effect:

$ ffmpeg -report -guess_layout_max 0 -i 4ac.wav -filter:a 
aresample=in_channel_layout=4.0:out_channel_layout=stereo -c:a pcm_s16le -y 
out.wav
ffmpeg started on 2020-06-29 at 14:53:17
Report written to "ffmpeg-20200629-145317.log"
Log level: 48
ffmpeg version N-98130-g38737b3d4e Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --enable-gpl --enable-nonfree --enable-shared --enable-openssl 
--enable-pic --enable-libspeex --enable-libopus --enable-libfdk-aac 
--enable-libx265 --enable-libx264 --enable-libmp3lame --enable-libbluray 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libvidstab 
--enable-libfreetype --enable-libzimg --enable-libass 
--extra-cflags=-I/opt/sw/include --extra-libs='-L/opt/sw/lib 
-L/opt/sw/lib/freetype219/lib -L/opt/sw/lib/gcc9/lib' --disable-htmlpages
  libavutil  56. 54.100 / 56. 54.100
  libavcodec 58. 92.100 / 58. 92.100
  libavformat58. 46.101 / 58. 46.101
  libavdevice58. 11.100 / 58. 11.100
  libavfilter 7. 86.100 /  7. 86.100
  libswscale  5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc55.  8.100 / 55.  8.100
Input #0, wav, from '4ac.wav':
  Metadata:
encoder : Lavf58.46.101
  Duration: 00:00:12.00, bitrate: 3072 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 4 
channels, s16, 3072 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'out.wav':
  Metadata:
ISFT: Lavf58.46.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, 
s16, 1536 kb/s
Metadata:
  encoder : Lavc58.92.100 pcm_s16le
size=2250kB time=00:00:12.00 bitrate=1536.1kbits/s speed= 387x
video:0kB audio:2250kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 0.003385%
$ grep -A 2 Matrix ffmpeg-20200629-145317.log
[Parsed_aresample_0 @ 0x7fc03e500a40] [SWR @ 0x1110fe000] Matrix coefficients:
[Parsed_aresample_0 @ 0x7fc03e500a40] [SWR @ 0x1110fe000] FL: FL:0.453082 
FR:0.00 FC:0.320377 BC:0.226541
[Parsed_aresample_0 @ 0x7fc03e500a40] [SWR @ 0x1110fe000] FR: FL:0.00 
FR:0.453082 FC:0.320377 BC:0.226541

$ ffmpeg -report -guess_layout_max 0 -i 4ac.wav -filter:a 
aresample=in_channel_layout=quad:out_channel
_layout=stereo -c:a pcm_s16le -y out.wav
ffmpeg started on 2020-06-29 at 14:59:13
Report written to "ffmpeg-20200629-145913.log"
Log level: 48
ffmpeg version N-98130-g38737b3d4e Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.17)
  configuration: --enable-gpl --enable-nonfree --enable-shared --enable-openssl 
--enable-pic --enable-libspeex --enable-libopus --enable-libfdk-aac 
--enable-libx265 --enable-libx264 --enable-libmp3lame --enable-libbluray 
--enable-libtheora --enable-libvorbis --enable-libvpx --enable-libvidstab 
--enable-libfreetype --enable-libzimg --enable-libass 
--extra-cflags=-I/opt/sw/include --extra-libs='-L/opt/sw/lib 
-L/opt/sw/lib/freetype219/lib -L/opt/sw/lib/gcc9/lib' --disable-htmlpages
  libavutil  56. 54.100 / 56. 54.100
  libavcodec 58. 92.100 / 58. 92.100
  libavformat58. 46.101 / 58. 46.101
  libavdevice58. 11.100 / 58. 11.100
  libavfilter 7. 86.100 /  7. 86.100
  libswscale  5.  8.100 /  5.  8.100
  libswresample   3.  8.100 /  3.  8.100
  libpostproc55.  8.100 / 55.  8.100
Input #0, wav, from '4ac.wav':
  Metadata:
encoder : Lavf58.46.101
  Duration: 00:00:12.00, bitrate: 3072 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 4 
channels, s16, 3072 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'out.wav':
  Metadata:
ISFT: Lavf58.46.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, 
s16, 1536 kb/s
Metadata:
  encoder : Lavc58.92.100 pcm_s16le
size=2250kB time=00:00:12.00 bitrate=1536.1kbits/s speed= 375x
video:0kB audio:2250kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 0.003385%
$ grep -A 2 Matrix ffmpeg-20200629-145913.log
[Parsed_aresample_0 @ 0x7feeca404380] [SWR @ 0x111d48000] Matrix coefficients:
[Parsed_aresample_0 @ 0x7feeca404380] [SWR @ 0x111d48000] FL: FL:0.453082 
FR:0.00 FC:0.320377 BC:0.226541
[Parsed_aresample_0 @ 0x7feeca404380] [SWR @ 0x111d48000] FR: FL:0.00 
FR:0.453082 FC:0.320377 BC:0.226541

Matrix coefficients for icl 4.0 and quad, or for any other 4
channel layout are exactly the same,

What am I overlooking?

Something like aresample=ocl=quad,aresample=ocl=stereo yields a
different matrix, but what is the point of in_channel_layout
then?

TIA

Christian

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Re: [FFmpeg-user] Error message when combining -vf and -af in command line

[Christian Ebert]

* Kieran O'Leary on Wednesday, March 13, 2019 at 15:27:21 +:

It appears that this is ticket 7712: https://trac.ffmpeg.org/ticket/7712


Ah, right, for some reaseon I didn't catch that when searching the bug
tracker.


and it was introduced here: 
https://github.com/FFmpeg/FFmpeg/commit/301cee61fa61c55b1c178ebfbc590872e8b033e6
 as an attempt to fix an existing bug: https://trac.ffmpeg.org/ticket/4184


Indeed, I kind of guessed that this is meant to warn about
duplicate -vf or -af.

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[FFmpeg-user] Error message when combining -vf and -af in command line

[Christian Ebert]

Hi,

Using -vf and -af in 1 command results in this error message:

Only '-vf scale' read, ignoring remaining -vf options: Use ',' to separate 
filters
Only '-af aresample' read, ignoring remaining -af options: Use ',' to separate 
filters

The message does not have any consequences.

Happens also with -filter:v plus -filter:a

Does not happen with -filter_complex

Does this mean that -vf plus -af is deprecated? Or is it a bug?

ffmpeg -i BigBuckBunny_640x360.m4v -vf scale -af aresample out.mpeg
ffmpeg version N-93330-g7ff89574c7 Copyright (c) 2000-2019 the FFmpeg developers
 built with Apple LLVM version 10.0.0 (clang-1000.11.45.5)
 configuration: --enable-gpl --enable-nonfree --enable-shared --enable-openssl 
--enable-pic --enable-libspeex --enable-libopus --enable-libfdk-aac 
--enable-libx265 --enable-libx264 --enable-libmp3lame --enable-libtheora 
--enable-libvorbis --enable-libvpx --enable-libvidstab --enable-libfreetype 
--enable-libzimg --enable-libass --extra-cflags=-I/sw/include 
--extra-libs='-L/sw/lib -L/sw/lib/freetype219/lib -L/sw/lib/gcc7/lib'
 libavutil  56. 26.100 / 56. 26.100
 libavcodec 58. 47.103 / 58. 47.103
 libavformat58. 26.101 / 58. 26.101
 libavdevice58.  6.101 / 58.  6.101
 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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'BigBuckBunny_640x360.m4v':
 Metadata:
   major_brand : M4V 
   minor_version   : 1

   compatible_brands: M4V M4A mp42isom
   creation_time   : 2008-06-03T05:23:28.00Z
   copyright   : © 2008 Blender Foundation | www.bigbuckbunny.org
   title   : Big Buck Bunny
   artist  : Peach Open Movie Team
   composer: Sacha Goedegebure / Ton Roosendaal
   date: 2008
 Duration: 00:09:56.46, start: 0.00, bitrate: 1626 kb/s
   Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 127 kb/s (default)
   Metadata:
 creation_time   : 2008-06-03T05:23:28.00Z
 handler_name: Apple Sound Media Handler
   Stream #0:1(eng): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), 
yuv420p(tv, smpte170m/smpte170m/bt709), 640x359, 1495 kb/s, 24 fps, 24 tbr, 
2400 tbn, 4800 tbc (default)
   Metadata:
 creation_time   : 2008-06-03T05:23:28.00Z
 handler_name: Apple Video Media Handler
Only '-vf scale' read, ignoring remaining -vf options: Use ',' to separate 
filters
Only '-af aresample' read, ignoring remaining -af options: Use ',' to separate 
filters
Stream mapping:
 Stream #0:1 -> #0:0 (h264 (native) -> mpeg1video (native))
 Stream #0:0 -> #0:1 (aac (native) -> mp2 (native))
Press [q] to stop, [?] for help
[mpeg @ 0x7f84ee004e00] VBV buffer size not set, using default size of 230KB
If you want the mpeg file to be compliant to some specification
Like DVD, VCD or others, make sure you set the correct buffer size
Output #0, mpeg, to 'out.mpeg':
 Metadata:
   major_brand : M4V 
   minor_version   : 1

   compatible_brands: M4V M4A mp42isom
   date: 2008
   copyright   : © 2008 Blender Foundation | www.bigbuckbunny.org
   title   : Big Buck Bunny
   artist  : Peach Open Movie Team
   composer: Sacha Goedegebure / Ton Roosendaal
   encoder : Lavf58.26.101
   Stream #0:0(eng): Video: mpeg1video, yuv420p, 640x359, q=2-31, 200 kb/s, 24 
fps, 90k tbn, 24 tbc (default)
   Metadata:
 creation_time   : 2008-06-03T05:23:28.00Z
 handler_name: Apple Video Media Handler
 encoder : Lavc58.47.103 mpeg1video
   Side data:
 cpb: bitrate max/min/avg: 0/0/20 buffer size: 0 vbv_delay: -1
   Stream #0:1(eng): Audio: mp2, 44100 Hz, stereo, s16, 384 kb/s (default)
   Metadata:
 creation_time   : 2008-06-03T05:23:28.00Z
 handler_name: Apple Sound Media Handler
 encoder : Lavc58.47.103 mp2
frame=14316 fps=1218 q=31.0 Lsize=   47920kB time=00:09:56.46 bitrate= 
658.1kbits/s speed=50.7x
video:19500kB audio:27960kB subtitle:0kB other streams:0kB global headers:0kB 
muxing overhead: 0.968647%


TIA

Christian

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Re: [FFmpeg-user] ProRes422 with 8 audio tracks converted to mp4 -- muffles audio 4

[Christian Ebert]

* Kieran O Leary on Thursday, November 22, 2018 at 19:00:06 +:

On Thu, 22 Nov 2018, 17:28 Harvey Pikelberger 
On Nov 22, 2018, at 8:50 AM, Carl Eugen Hoyos  wrote:
2018-11-22 17:44 GMT+01:00, Harvey Pikelberger :

ffmpeg -i ffmpeg -i SoundMappingTest-I-8Discreet.mov -map 0 -g 48 -c:v
libx264 -profile:v baseline -crf 16 -c:a aac -b:a 256k -vf

scale=1280:720

-pix_fmt yuv420p SoundMappingTest-I-8Discreet.mp4


You (apparently) cannot copy the timecode track from mov to mp4
but this is what you request with "-map 0".

Default behaviour would be to transcode one video and one audio
track.


Questions:
Does the timecode issue explain the 'Conversion Failed'?


Yes I think so. Did you try adding -dn to your command line so that it
ignores data tracks?


To prevent the mov timecode to be processed -dn does not help, 
use -write_tmcd 0. I don't know how this fares with -map 0 
though.


See: https://trac.ffmpeg.org/ticket/5492

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Mark Burton on Wednesday, May 24, 2017 at 13:20:34 +0100
> On 23 May 2017, at 11:20, Christian Ebert <blacktr...@gmx.net> wrote:
>>> So I looked back at your above -af and realised that the 1024 should 
>>> actually be 2112 which is Apple’s chosen fixed encoding delay.
>>> https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
>>>  
>>> <https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html>

Yes, the 'historical solution': implicit encoder delay:
https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html#//apple_ref/doc/uid/TP4939-CH2-SW9

>>> -filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+2112
>> Strange, probably all depends on the demuxing application.
> My take on it is that Quicktime will always assume 2112 padding, so I believe 
> a .mov should be setup with this priming duration.

Yeah, filtering does not add an edit list (apparently the
'modern' solution):
https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFChap2/qtff2.html#//apple_ref/doc/uid/TP4939-CH204-25592
and therefore QuickTime fails over to a hardcoded 'historical'
default. That explains it.

There are some options referring to edit lists, but I haven't
tried whether, nevermind how, they could be used for this
purpose.

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Mark Burton on Monday, May 22, 2017 at 15:22:34 +0100
> On 15 Apr 2017, at 09:22, Christian Ebert <blacktr...@gmx.net> wrote:
>> Somewhat counterintuitive, but you never know:
>> 
>> -filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+1024
>> 
>> combined with the -t incantation.
> 
> It seems this issue is not going to garner much attention which is a little 
> disheartening. I can show how to reproduce it and would love to be able to 
> help.
> 
> So I looked back at your above -af and realised that the 1024 should actually 
> be 2112 which is Apple’s chosen fixed encoding delay.
> https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
>  
> <https://developer.apple.com/library/content/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html>
> 
> -filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+2112

Strange, probably all depends on the demuxing application.

ffprobe -show_entries stream=codec_type,start_time,start_pts,duration result

for me gives 0 for all start values.

> ...brings the native aac encoder almost perfectly into sync when played by a 
> Quicktime based decoder. There is a tiny discrepancy, but its 99.9% better 
> than without the -af line.
> 
> Further to this, using the AudioToolbox AAC encoder (aac_at) available in 
> ffmpeg on macOS only, with the above -af line, this discrepancy is gone and 
> the encoded file is a perfect sync match for the original source file.
> 
> The outstanding issue is the remaining samples for the file which are not 
> being trimmed, so the clip runs past the end of the picture and we get a 
> black frame. Perhaps the remaining samples are not being flagged in a way 
> that the decoder would expect, I’m really not sure.
> 
> I can use the '-t' command with the value for ('total duration of source' - 
> ‘0.041') to trim the end, but its has issues too.
> In my case of 24fps source, 0.041 is the duration of 1 frame. Doing this 
> shortens the overlong audio stream, but it removes the last frame of audio 
> and in some cases does strange things with the last two frames of audio. So 
> its not really usable in a production environment. Without subtracting 0.041, 
> the audio is still overlong.

Did you look at the atrim filter? It offers end, end_sample, and
end_pts parameters. You may have to to some calculations.

Re-reading the article you referenced, it may even a better idea
to use atrim instead of asetpts for the start as well.

Maybe:

-filter:a aresample=async=1:first_pts=0,atrim=start_pts=2112:end={-t value}

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Marton Balint on Saturday, April 15, 2017 at 12:25:58 +0200
> On Sat, 15 Apr 2017, Christian Ebert wrote:
>> * Marton Balint on Saturday, April 15, 2017 at 07:55:22 +0200
>>> Last time I checked (a year ago or so), ffmpeg created a correct .mov
>>> edit list to signal the audio priming.
>>> 
>>> https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
> 
> Hmm, a recent fix changed one of the hunk contexts... Could you try
> this new attached patch?

I'm afraid for my purpose - segmenting - it does not make a
difference. I still get the same overlong first segments;
depending on codec - native aac has much shorter 'delays' than
fdk-aac, they are minute, but they still make the hls muxer set a
1 second higher TARGETDURATION, even when all segments are of the
exact duration up to the sixth decimal - this can be corrected
by hand because the spec tolerates such minimal divergences.

But again, for the purpose of segmenting it has not any effect.

It may well fix Mark's use case. I haven't checked that.

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Marton Balint on Saturday, April 15, 2017 at 07:55:22 +0200
> Last time I checked (a year ago or so), ffmpeg created a correct .mov
> edit list to signal the audio priming.
> 
> https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
> 
> Here is patch which writes Sample Group Description Atom and the
> Sample-To-Group Atom as well, in addition to the already existing edit
> list.
> 
> You can check if it makes a difference or not.

Unfortunately the patch doesn't apply.

> Regards,
> Marton

> From 64d9db76038b7b248bffe459b58b138c6b8fcc03 Mon Sep 17 00:00:00 2001
> From: Marton Balint 
> Date: Sat, 20 Feb 2016 19:32:18 +0100
> Subject: [PATCH] avformat/movenc: add support for writing AAC Audio Priming
> metadata
> 
> https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFAppenG/QTFFAppenG.html
> 
> Signed-off-by: Marton Balint 
> ---
> libavformat/movenc.c| 21 ++---
> tests/ref/fate/adtstoasc_ticket3715 |  4 ++--
> tests/ref/fate/copy-psp |  4 ++--
> tests/ref/fate/movenc   | 12 ++--
> 4 files changed, 24 insertions(+), 17 deletions(-)
> 
> diff --git a/libavformat/movenc.c b/libavformat/movenc.c
> index e6a70bf..5779e3c 100644

I could not find commit e6a70bf either.

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Carl Eugen Hoyos on Saturday, April 15, 2017 at 00:44:43 +0200
> 2017-04-14 23:44 GMT+02:00 Mark Burton :
>> I find it hard having to accept an encode will always play out of
>> sync on certain players.
> 
> Could you elaborate a little?
> So far, for every ticket, it either turned out that out-of-sync was
> not reproducible with MPlayer / vlc (and FFmpeg) or it was off-
> by-one-frame which I consider hard to reproduce...

In theory it is probably correct that it's the task of the
decoding application to make the aac priming work. In many real
world scenarios this does not happen.

I do a lot of segmenting, and can always reproduce it with the
hls muxer:

https://trac.ffmpeg.org/ticket/3859
I closed it because, as per last comment I finally resorted to
the workaround(?) mentioned here.
https://trac.ffmpeg.org/ticket/3859#comment:13
https://sourceforge.net/p/opencore-amr/mailman/message/32912437/

If you see a HLS playlist where the TARGETDURATION is higher than
the duration of any segments than the first you can bet it has
been created by ffmpeg.

And if you don't clip at the end, you can easily get invalid
audio-only segments as last segment, and end of video may not be
detected in the majority of real world scenarios, arguably when
you play HLS in a browser.

So, apart from introducing minimal async, it also has other
repercussions.

Why not a warning, ideally with a hint as to how to best deal
with the situation, similar to the one about pix_fmpt yuv420p:

Use -pix_fmt yuv420p for compatibility with outdated media players.

Maybe there is no correct solution, and the delay hint in the
verbose output was certainly helpful, at least in my case; it
also points to some awareness of a potential problem area. It
just took me a long time to find it.

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Mark Burton on Friday, April 14, 2017 at 22:44:52 +0100
>> On 14 Apr 2017, at 22:22, Christian Ebert <blacktr...@gmx.net> wrote:
>>>> Also, when you run with -v verbose, you'll see a delay (depends
>>>> on audio codec), for you case it's probably 1024. Maybe try:
>>>> 
>>>> -filter:a aresample=first_pts=0,asetpts=PTS-STARTPTS+1024
>>> 
>>> 1024 looks correct. This method successfully changes the start, and the 
>>> encoded audio plays, almost, 100% in sync. Its slightly cut off at the 
>>> head, but only very slightly. However the downside of this method is that 
>>> the audio now overruns the end of the picture further and is not trimmed 
>>> from remaining samples I’m guessing. This results in a blank video frame 
>>> being added to the tail of clip. ffprobe for ffmpeg_v3 attached.
>>> 
>>> I tried using -shortest, but that didn’t help address the above issue.
>> 
>> Inserting
>> 
>> -t `ffprobe -v quiet -of default=nw=1:nk=1 -select_streams V -show_entries 
>> stream=duration SyncTest24p.mov`
>> 
>> into your commandline should do the trick.
> 
> Adding '-t 2.00’ to the command does help a little, but ultimately the 
> final audio packet is still too large - 6ms in this case - and the black 
> frame still occurs.

Works for me with all my sample results I checked.

Somewhat counterintuitive, but you never know:

-filter:a aresample=async=1:first_pts=0,asetpts=PTS-STARTPTS+1024

combined with the -t incantation.

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Mark Burton on Friday, April 14, 2017 at 21:45:07 +0100
>> * Mark Burton on Friday, April 14, 2017 at 16:57:06 +0100
>>> Here is the basic command to reproduce. I have attached the uncut loglevel 
>>> 99 console output for this command:
>>> ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p -movflags faststart 
>>> -c:a aac -b:a 128k ffmpeg.mp4
>> 
>> Can you try:
>> 
>> -filter:a aresample=first_pts=0
> 
> Thanks for your reply and these options to test. This first one does not 
> appear to change the outcome. ffprobe for ffmpeg_v2 attached.
> 
>> Also, when you run with -v verbose, you'll see a delay (depends
>> on audio codec), for you case it's probably 1024. Maybe try:
>> 
>> -filter:a aresample=first_pts=0,asetpts=PTS-STARTPTS+1024
>> 
>> Especially the latter could be exactly the wrong thing for your
>> purpose, but it doesn't hurt trying.
> 
> 1024 looks correct. This method successfully changes the start, and the 
> encoded audio plays, almost, 100% in sync. Its slightly cut off at the head, 
> but only very slightly. However the downside of this method is that the audio 
> now overruns the end of the picture further and is not trimmed from remaining 
> samples I’m guessing. This results in a blank video frame being added to the 
> tail of clip. ffprobe for ffmpeg_v3 attached
> .
> 
> I tried using -shortest, but that didn’t help address the above issue.

Inserting

-t `ffprobe -v quiet -of default=nw=1:nk=1 -select_streams V -show_entries 
stream=duration SyncTest24p.mov`

into your commandline should do the trick.

imho this is a muxing bug in dealing with aac priming in many
situations, but making my case fell on deaf ears.

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Re: [FFmpeg-user] Create an AAC stream matching the Core Media Audio packet format / priming etc?

[Christian Ebert]

* Mark Burton on Friday, April 14, 2017 at 16:57:06 +0100
> I appreciate this is a tricky area and there appear to be different ways that 
> some encoders create AAC streams with regards to the padding and remaining 
> samples etc. I won’t pretend to fully understand all the factors, but I would 
> like to ask a genuine question which purely comes from the point of view of 
> wanting to create a file for my working environment - an environment 
> dominated by Quicktime 7 and X playback / decoding tools. I’m no great fan of 
> Quicktime and appreciate its not well loved here also. In my industry it is 
> still very much the defecto playback engine though, so if I’m able to tailor 
> a file for this decoder, it would be an enormous help.
> 
> Let me also say, I am not accusing ffmpeg of having an issue, I have read a 
> number of ‘bug’ reports surrounding ffmpeg and AAC priming and there being a 
> sync discrepancy in the resultant encode when played back in certain 
> decoders. I happen to see the exact same issue, but from some of the 
> developers replies, I accept their position is that they feel ffmpeg is doing 
> it the right way and its the decoders that are at fault.
> 
> With that said, I’d like to approach this question purely from the point of 
> view of finding out whether there is a way to tweak a command in order to 
> change this way the aac stream is created to produce an mp4 or mov file using 
> the native aac encoder which decodes in Quicktime 7 or X, in sync. Currently 
> an encoded file plays back 1 frame out of sync (audio is early by approx. 1 
> frame). In VLC its about 1/2 a frame out of sync.
> 
> The source file is a .mov, DNx115 24p (true 24p, not 23.976), PCM 24bit 48khz 
> audio, which is in sync. This is film material where sync is crucial and 
> always expected.
> 
> Here is the basic command to reproduce. I have attached the uncut loglevel 99 
> console output for this command:
> ffmpeg -i SyncTest24p.mov -c:v libx264 -pix_fmt yuv420p -movflags faststart 
> -c:a aac -b:a 128k ffmpeg.mp4

Can you try:

-filter:a aresample=first_pts=0

Also, when you run with -v verbose, you'll see a delay (depends
on audio codec), for you case it's probably 1024. Maybe try:

-filter:a aresample=first_pts=0,asetpts=PTS-STARTPTS+1024

Especially the latter could be exactly the wrong thing for your
purpose, but it doesn't hurt trying.

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Re: [FFmpeg-user] HLS stream playing issue - Mobile Firefox Android

[Christian Ebert]

* Walter Ebert on Tuesday, November 08, 2016 at 09:14:51 +0100
>> On Mobile Firefox - via http://dailymotion.github.io/hls.js/demo/ - i tr
>> to load local m3u8 file but i receive the following error in the demo's
>> info area:
>> 
>> Buffer Add Codec Error for video/mp4; codecs=avc1.f4001f :Operation is not
>> supported.
> Mobile Firefox does not suppot HLS.
>> On desktop browser no error but playing fine.
> Also Firefox on desktop does not support HLS out of the box. It can play
> HLS if an extension is installed. This can be provided by VLC or
> Gstreamer for example. Most people won't have this installed.
> 
> In HTML5 you can provide MP4 and WebM as a fallback:
> 
>|   type="application/x-mpegurl" />  />  |
> 
> 
> If you want a cross-browser streaming solution, you could look into
> MPEG-DASH (http://dashif.org/).

You'd still need HLS for iOS where the MediaSource extension is
not supported. Try hls.js:
https://github.com/dailymotion/hls.js

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Re: [FFmpeg-user] Removing Timecode Track

[Christian Ebert]

* William Caulfield on Tuesday, July 19, 2016 at 10:24:12 -0700
> On Tue, Jul 19, 2016 at 6:53 AM, Christian Ebert <blacktr...@gmx.net> wrote:
>> * Kieran O Leary on Sunday, July 17, 2016 at 18:14:59 +0100
>>> On 17 Jul 2016 7:05 p.m., "Kelly Haydon" <kellyhay...@bavc.org> wrote:
>>>> Hello - I'd like to use ffmpeg to remove the QT timecode  which exists
>>>> in Stream #0:2 of the file expressed below. Basic re-encoding with only the
>>>> first two streams mapped was recommended, but has not worked. I have not
>>>> found a flag like -an that will do the trick. Would anyone have insight on
>>>> how to do what (I thought) is a relatively simple task?
>>> 
>>> -dn should do the trick!
>> 
>> It should, but does not:
>> https://trac.ffmpeg.org/ticket/5492
> 
> Maybe   "-map -0:d"  ?  It works for me when using -map 0 .

Nope. Neither positive, nor negative mapping helps.

It obviously depends on the target format; if you don't encode to
mp4, mov, or friends you don't need any gymnastics. But otherwise
the timecode is always copied.

Or use an ffmpeg version before the regression.

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Re: [FFmpeg-user] Removing Timecode Track

[Christian Ebert]

* Kieran O Leary on Sunday, July 17, 2016 at 18:14:59 +0100
> On 17 Jul 2016 7:05 p.m., "Kelly Haydon"  wrote:
>> Hello - I'd like to use ffmpeg to remove the QT timecode  which exists
>> in Stream #0:2 of the file expressed below. Basic re-encoding with only the
>> first two streams mapped was recommended, but has not worked. I have not
>> found a flag like -an that will do the trick. Would anyone have insight on
>> how to do what (I thought) is a relatively simple task?
> 
> -dn should do the trick!

It should, but does not:
https://trac.ffmpeg.org/ticket/5492

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Re: [FFmpeg-user] Cutting on video frame

[Christian Ebert]

* Louis Letourneau on Thursday, July 07, 2016 at 14:05:37 -0400
> I just tried, converting audio for cutting works fine:
> ffmpeg -y -ss 0.042000 -i 2708-1.mp4 -codec:a pcm_s32le -codec:v libx264
> -crf 23 -preset fast -pix_fmt yuv420p -flags +global_header
> -force_key_frames "expr:gte(t,n_forced*5)" -bf 0 a.mp4

Sure. pcm is not a problem, the aac padding is.

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Re: [FFmpeg-user] Cutting on video frame

[Christian Ebert]

* Louis Letourneau on Thursday, July 07, 2016 at 12:38:26 -0400
> When cutting a video, even when transcoding, I often have a small delay
> between the video and audio.
> 
> When looking at the video with either vlc or ffplay, I see the first frame
> shown twice.(The first is probably a dup until it the real pts is hit).
> 
> I was wondering if there is a way to force cutting on video frame
> boundaries to avoid this.
> 
> Also when using -ss to cut exactly to the video frame closest to where I
> want to cut. I still often get audio packets and frames with negative pts.

You may be interested in this:
https://trac.ffmpeg.org/ticket/3859

Basically there's always some aac padding. I usually resort to
the atrim filter to butcher that off at the beginning, and
sometimes other nasty stuff.

> If It then convert this video from mp4 -> hls, the negative frames are now
> positive and my video frame is push forward again.

Yeah, the behavior of the hls or other segment muxers doesn't
make things easier.

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Re: [FFmpeg-user] translate rtp to http

[Christian Ebert]

* juan carlos Rebate on Wednesday, June 01, 2016 at 04:38:12 +0200
> the problem esque presupposes that I want to play in ffplay, but I do not
> want to play in ffplay I want reproduvcirlo in a html5 player for
> android

Most Android devices cannot play opus audio, use vorbis instead.

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Re: [FFmpeg-user] How can I extract 8 linear thumbnails from videos with differnt length?

[Christian Ebert]

* Christoph Gerstbauer on Thursday, April 07, 2016 at 10:12:14 +0200
>> Depending on the calculation it may be easier/more intuitive to
>> use framerate:
>> 
>> ffmpeg -i input -filter:v framerate=1/10 out-%d.jpg
>> 
>> will create one image every ten secs.
> 
> I dont need every x seconds one image ;)

That's why I wrote 'Depending on the calculation' - in many cases
I would find it easier to grab the duration with ffprobe first,
and calculate the required interval in secs before I run ffmpeg.

Otherwise the select filter is indeed your friend.

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Re: [FFmpeg-user] How can I extract 8 linear thumbnails from videos with differnt length?

[Christian Ebert]

* Moritz Barsnick on Wednesday, April 06, 2016 at 23:13:25 +0200
> On Wed, Apr 06, 2016 at 21:02:17 +0200, Christoph Gerstbauer wrote:
>> I want to extract 8 thumbnails from every video I have which represents 
>> a linear "timeline" over the complete length (100percent) of the video.
> [...]
>> Is there a filter which can be used for that in combination with the 
>> -vframe command?
> 
> I don't think you'd need "-vframes" (that's what you meant?).
> 
>> I want to use ONE command line, where the video length is used as 100 
>> percent.
> 
> The select filter would be appropriate, *except* that the total length
> of the video is not available for its expression evaluation. So you
> would actually need two commands - one to determine the length, the
> other to extract the frames.

Depending on the calculation it may be easier/more intuitive to
use framerate:

ffmpeg -i input -filter:v framerate=1/10 out-%d.jpg

will create one image every ten secs.

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Re: [FFmpeg-user] Live WebM Streaming

[Christian Ebert]

* MKNwebsolutions . on Thursday, November 12, 2015 at 13:05:56 -0500
> I'm converting a live stream (-i rtmp) into libvpx-vp9 output to
> -webm_chunk.  This looks like it's working perfectly (minus a few must have
> future features i.e. auto delete chunks / segments).  Next we generate the
> webm dash manifest (webm_dash_manifest), where we pull the header generated
> in the prior thread.  This here seems to constantly break and is not able
> to generate the manifest.  Output responses yield Operation not permitted
> and unable to detect codec / pixels.
> 
> I've found maybe 2 threads on this subject with no answers, I'm going to
> assume no one's determined a solution out of the box yet?

I did not try chunking, but had some success with VOD and the
-dash 1 option - I guess this would not applyi to webm_chunk.

Which audio codec do you use? For me the muxer failed with libopus,
had to use libvorbis.

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Re: [FFmpeg-user] Setting up a watch folder to encode adaptive bit rate videos

[Christian Ebert]

* James Darnley on Tuesday, November 03, 2015 at 01:29:43 +0100
> libx264 definitely supports changing resolution between first and later
> passes.

Also when then second pass involves a change in profile and/or
level, meaning change of frame types?

> It also determines frame types in the first pass.

So, only when the same profile and level and frame related
settings are the same, I assume.

otoh without slow-firstpass for instance the reduction to ref=1
might affect already frame types?

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Re: [FFmpeg-user] ffmpeg and multiple outputs

[Christian Ebert]

* Ricardo Kleemann on Monday, August 03, 2015 at 10:57:01 -0700
 Thanks everyone for the follow-ups. Does anyone know if SDL also works on
 OSX,

yes

 or what would be the equivalent?

For this specific purpose (ffmpeg output device) there is no
equivalent as far as I know. But I haven't looked hard.

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Re: [FFmpeg-user] ffmpeg and multiple outputs

[Christian Ebert]

* Moritz Barsnick on Monday, August 03, 2015 at 09:48:20 +0200
 On Sun, Aug 02, 2015 at 17:51:39 -0700, Ricardo Kleemann wrote:
 Good point, how would I display ffmpeg on OS X? I’m not quite sure
 what the output device would be?
 
 The output device sdl is the first that comes to mind.
 
 $ ffmpeg -i $infile -f sdl SDL output
 and you can add another output, i.e. to file, after those arguments.
 
 Moritz
 
 P.S.: You may have to add -pix_fmt yuv420p to get it to work,
  depending on the input format. I believe my sdl didn't like
  another format, and ffmpeg didn't convert automatically.

Almost, it needs yuyv422:

$ ffmpeg -i $infile -pix_fmt yuyv422 -f sdl SDL Output

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Re: [FFmpeg-user] ffprobe -show_frames values

[Christian Ebert]

* Nicolas George on Tuesday, May 05, 2015 at 17:28:33 +0200
 As an additional note, the second solution if by far preferable, because
 forcing the frame type too frequently ruins x264's bit allocation
 algorithms.

As per this thread elsewhere there are different opinions on
that. Others say that no-scenecut is not adviseable because it
obviously removes scene detection.

Which one is it then:

-force_key_frames
or
-x264opts keyint=xxx:no-scenecut

?

Personally I got ok looking results both ways with various
material.

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Re: [FFmpeg-user] ffprobe -show_frames values

[Christian Ebert]

* Werner Robitza on Tuesday, May 12, 2015 at 15:10:07 +0200
 On Tue, May 12, 2015 at 2:45 PM, Christian Ebert blacktr...@gmx.net wrote:
 * Nicolas George on Tuesday, May 05, 2015 at 17:28:33 +0200
 As an additional note, the second solution if by far preferable, because
 forcing the frame type too frequently ruins x264's bit allocation
 algorithms.
 
 As per this thread elsewhere there are different opinions on
 that. Others say that no-scenecut is not adviseable because it
 obviously removes scene detection.
 
 Which one is it then:
 
 -force_key_frames
 or
 -x264opts keyint=xxx:no-scenecut
 
 ?
 
 Personally I got ok looking results both ways with various
 material.
 
 I'm currently involved in ITU-T standardization of a quality model for
 adaptive streaming, and in our video sequences that we use for
 (subjective user) tests, we've disabled scene cut detection
 throughout, with good-looking results. Nothing unexpected.

Thanks for confirming. IIRC I had trouble with (some) older x264
versions and no-scenecut and had therefore worked with
-force_key_frames for a while. But I cannot reproduce the problem
with current x264, so I switched to x264opts with no-scenecut.

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Re: [FFmpeg-user] Faster vp9

[Christian Ebert]

* tim nicholson on Wednesday, April 08, 2015 at 08:03:52 +0100
 On 07/04/15 15:18, Matt Zagrabelny wrote:
 On Mon, Apr 6, 2015 at 11:10 PM, Andrew Sinclair ajsincl...@gmail.com 
 wrote:
 Does anyone have any suggestions for faster vp9 encoding?
 
 I don't have the ffmpeg technical chops to analyze your command-line,
 but I just came across a Google VP9 encoding guide:
 
 https://sites.google.com/a/webmproject.org/wiki/ffmpeg/vp9-encoding-guide
 
 Maybe that will be helpful.
 
 I see it advises using 2 pass encoding with crf, which seems a bit odd
 to me given that This (crf) provides maximum compression efficiency
 with a single pass or is that only true of x264? I would have thought
 that it should be true for all coders that implement such a mode.

The wiki has it for vp8 as well:
https://trac.ffmpeg.org/wiki/Encode/VP8#VariableBitrate

Not sure what to make of it.

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Re: [FFmpeg-user] Reduce mux overhead of HLS stream creation

[Christian Ebert]

* Claudiu Rad-Lohanel on Monday, February 16, 2015 at 20:05:50 +0200
 Can't this actually be related to the player (maybe not implementing
 HLS standard 100% precisely)?

Nope:

$ ffprobe -v warning -show_entries format=duration 
http://media.blacktrash.org/ccc_trailer1.m3u8
[FORMAT]
duration=254.28
[/FORMAT]
$ ffprobe -v warning -show_entries format=duration 
http://media.blacktrash.org/fp/enc/ccc_trailer1-as.m3u8
[FORMAT]
duration=252.274700
[/FORMAT]

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Re: [FFmpeg-user] Reduce mux overhead of HLS stream creation

[Christian Ebert]

* Claudiu Rad-Lohanel on Monday, February 16, 2015 at 12:31:58 +0200
 On 2/16/2015 10:46 AM, Moritz Barsnick wrote:
 For one, we might identify that your problem has been fixed already.
 Here's a (fixed) ticket concerning a similar matter:
 
 https://trac.ffmpeg.org/ticket/2857
 
 No, that didn't fix what Wesley reports.
 
 I am watching the subject for a quite long time, seen also when that
 ticket was closed. I compiled, tested afterwards, the HLS segmenting
 overhead seems the same. I've compiled and tested many times since,
 the problem is the same.
 Maybe actually what we report is a different problem than #2857 with
 similar effect so I didn't comment on it without proof.
 I wanted to report this but never had the time to gather all the
 samples and create a nice report of it. Maybe now I'll be more
 motivated if Wesley doesn't do it before me.
 
 Bottom line: Apple's segmenter introduces low overhead of only a few
 precent at low bitrates like 150-200kbit, while ffmpeg may reach
 15-20% no matter what mpegts path you choose. And this is a problem
 not only because of such high waste but also as it yields Apple
 validation warnings for the HLS stream generated with ffmpeg.
 Also the mpegts output is the same for ffmpeg no matter if you
 generate .ts directly or use ssegment or whatever, which would make
 sense as from what I figured from the source, the muxer used is the
 same? Or maybe here I am wrong..
 
 And to also clarify something: ffmpeg's overhead is ALWAYS bigger than
 Apple's mediafilesegmenter, no matter the bitrate. But I assume it
 tends to be more constant (or it grows slower with bitrate), thus, you
 don't see such high percentages with higher bitrates. But if you
 segment a 1mbit 1GB video with ffmpeg vs. Apple's segmenter, you'll
 see tens of MB of difference between them.
 It's clearly a general efficiency problem in ffmpeg in HLS case, I
 assume that's because Apple squeezed everything it could from mpegts
 while ffmpeg uses the generic mpegts muxer which must create valid and
 compatible streams for all kinds of devices.

Don't have blind faith in Apple's segmenter though. It seems to
overoptimize and may get the duration wrong (and actually quite
massively):

http://flowplayer.blacktrash.org/fp5/hls-segmenter.html

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Re: [FFmpeg-user] Output PTS is different from the source

[Christian Ebert]

* Wesley Wen on Thursday, December 18, 2014 at 15:43:22 +
 I'm transcoding one MPEG2-TS file to MP4, but I noticed the start PTSs of
 video and audio of the generated MP4 file are different from the source.
 The first video frame starts at 0, while the first audio PTS is negative. I
 would expect the PTS of source and output should be identical.

Me too, but: https://trac.ffmpeg.org/ticket/3859

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Re: [FFmpeg-user] Use ffmpeg to convert stereo audio to mono

[Christian Ebert]

* Moritz Barsnick on Friday, December 05, 2014 at 10:37:14 +0100
 Could we know how ffmpeg determine audio channel number from MP4 file? MP4
 container indicates it's channel count is 2, but ffprobe shows mono as
 expected.
 
 ffprobe probably looks at the actual AAC stream?
 
 Here's a hint:
 http://sourceforge.net/p/opencore-amr/mailman/message/31876733/
 
 The cause for this is that libavformat always writes 2 as number of 
 channels in the m4a header regardless of the actual stream.
 (This was about HE-AAC, but may be valid here as well.)
 
 Is this causing any issues except your trust in the nature of the file?
 
 I must say I'm not sure whether that is actually true, whether it's
 okay for ffmpeg to do so, and whether this is still valid with recent
 ffmpeg. I can confirm the behavior though.
 
 Moritz
 
 P.S.: I like adding other tools to the confusion, without knowing
 what they look at:
 
 $ mediainfo audio_48k_mono.mp4 | grep -Fi chann
 Channel(s)   : 2 channels
 Channel(s)_Original  : 1 channel
 Channel positions: Front: C
 $ mplayer -vo null -ao null audio_48k_mono.mp4 2/dev/null | grep -F AUDIO
 AUDIO: 44100 Hz, 1 ch, s16le, 48.3 kbit/6.85% (ratio: 6040-88200)
 $ # vlc says: Codec - Stream 0 - Channels: Stereo

There's more confusion:
https://trac.ffmpeg.org/ticket/4106

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Re: [FFmpeg-user] Mjpeg to h264 audio sync problem

[Christian Ebert]

* Petr Tresnak on Friday, December 05, 2014 at 10:34:50 +
 the sound start is cut and audio is ahead even more with your command line.
 
 ffmpeg -i mjpeg.avi -qscale 2 -strict -2 -vcodec mpeg4 -acodec aac out.mp4
 
 encoder : Lavf56.14.100
 Duration: 00:00:14.39, start: 0.092880, bitrate: 873 kb/s
 Stream #0:0(und): Video: mpeg4 (Simple Profile) (mp4v / 0x7634706D), yuv420p, 
 640x480 [SAR 1:1 DAR 4:3], 818 kb/s, 30 fps, 30 tbr, 65521 tbn, 65521 tbc 
 (default)
 Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 11025 Hz, mono, fltp, 
 57 kb/s (default)
 
 Here you have full console output:
 
 ffmpeg version git-2014-11-20-e39f8fa Copyright (c) 2000-2014 the FFmpeg 
 developers
  built on Nov 20 2014 12:58:03 with gcc 4.9.2 (Rev2, Built by MSYS2 project)
  configuration: --arch=x86 --disable-debug --disable-shared --disable-doc 
 --enable-gpl --enable-version3 --enable-runtime-cpudetect --enable-avfilter 
 --enable-
 bzlib --enable-zlib --enable-librtmp --enable-gnutls --enable-avisynth 
 --enable-frei0r --enable-filter=frei0r --enable-libbluray --enable-libcaca 
 --enable-libop
 enjpeg --enable-fontconfig --enable-libfreetype --enable-libass 
 --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame 
 --enable-libopencore-am
 rnb --enable-libopencore-amrwb --enable-libvo-amrwbenc 
 --enable-libschroedinger --enable-libsoxr --enable-libtwolame 
 --enable-libspeex --enable-libtheora --enab
 le-libutvideo --enable-libvorbis --enable-libvo-aacenc --enable-openal 
 --enable-libopus --enable-libvidstab --enable-libvpx --enable-libwavpack 
 --enable-libxavs
 --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzvbi 
 --enable-nonfree --enable-libfaac --enable-libfdk-aac
  libavutil  54. 14.100 / 54. 14.100
  libavcodec 56. 12.101 / 56. 12.101
  libavformat56. 14.100 / 56. 14.100
  libavdevice56.  3.100 / 56.  3.100
  libavfilter 5.  2.103 /  5.  2.103
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  1.100 /  1.  1.100
  libpostproc53.  3.100 / 53.  3.100
 Guessed Channel Layout for  Input Stream #0.1 : mono
 Input #0, avi, from 'input.avi':
  Metadata:
creation_time   : 2005-12-30 14:13:20
encoder : CanonMVI02
  Duration: 00:00:14.30, start: 0.00, bitrate: 14883 kb/s
Stream #0:0: Video: mjpeg (MJPG / 0x47504A4D), yuvj422p(pc, 
 bt470bg/unknown/unknown), 640x480, 14823 kb/s, 30 fps, 30 tbr, 30 tbn, 30 tbc
Stream #0:1: Audio: pcm_u8 ([1][0][0][0] / 0x0001), 11024 Hz, 1 channels, 
 u8, 88 kb/s
 [swscaler @ 051d8080] deprecated pixel format used, make sure you did set 
 range correctly
 Output #0, mp4, to 'output.mp4':
  Metadata:
encoder : Lavf56.14.100
Stream #0:0: Video: mpeg4 ( [0][0][0] / 0x0020), yuv420p, 640x480, q=2-31, 
 200 kb/s, 30 fps, 65521 tbn, 30 tbc
Metadata:
  encoder : Lavc56.12.101 mpeg4
Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 11025 Hz, mono, fltp, 64 
 kb/s
Metadata:
  encoder : Lavc56.12.101 aac
 Stream mapping:
  Stream #0:0 - #0:0 (mjpeg (native) - mpeg4 (native))
  Stream #0:1 - #0:1 (pcm_u8 (native) - aac (native))
 Press [q] to stop, [?] for help
 frame=  429 fps=266 q=31.0 Lsize=1535kB time=00:00:14.30 bitrate= 
 879.0kbits/s
 video:1428kB audio:101kB subtitle:0kB other streams:0kB global headers:0kB 
 muxing overhead: 0.374799%
 Press any key to continue . . .

Seems related to
https://trac.ffmpeg.org/ticket/3859

aac duration is rarely the same as the video, in your case it
looks like the audio is truncated.

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Re: [FFmpeg-user] Audio advanced when encoding from ProRes to H.264

[Christian Ebert]

* Michael Connolly on Monday, August 18, 2014 at 00:40:53 -0700
 I'm new-ish to encoding with FFMPEG and am hoping you guys can
 assist on a confusing issue.
 
 I'm transcoding from ProRes with PCM audio to H.264 with AAC
 audio.  After transcoding, my audio track is advanced (appears
 earlier in the file) by about 1.5 frames.  This doesn't occur
 if I trancode from H.264 to H.264 at a different bitrate.  In
 that case the audio is preserved.I did try converting from
 XDCAM - h.264 as well and encountered the same issue -- the
 audio is advanced.

I _always_ experience a video delay of about half a frame with
x264, even when encoding x264 - x264:

https://trac.ffmpeg.org/ticket/3859

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Re: [FFmpeg-user] HLS for Apple TV?

[Christian Ebert]

* Mark Bogdanoff on Tuesday, July 29, 2014 at 06:36:36 -0700
 Have you run Apple's mediastreamvalidator on your streams?
 https://developer.apple.com/library/ios/technotes/tn2235/_index.html

Except that these 'HTTP Live Streaming Tools' are not available
anymore.

(I'd be happy to be proven wrong)

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Re: [FFmpeg-user] only audio ffmpeg

[Christian Ebert]

* Lou on Sunday, July 27, 2014 at 13:51:25 -0800
 On Sun, 27 Jul 2014 19:23:45 +0100
 Christian Ebert blacktr...@gmx.net wrote:
 * Carl Eugen Hoyos on Monday, July 21, 2014 at 15:01:08 +
 
 FFmpeg contains a native opus decoder, libopus is not 
 needed.
 
 Then the fine new website is not up to speed:
 https://www.ffmpeg.org/ffmpeg-codecs.html#libopus-1
 
 Fixed. It may take a day for the new docs to generate.

Thanks.

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Re: [FFmpeg-user] Still happening - Using AVStream.codec.time_base as a timebase hint to the muxer is deprecated. Set AVStream.time_base instead.

[Christian Ebert]

* Carl Eugen Hoyos on Saturday, July 26, 2014 at 21:14:14 +
 Luke Davis l1 at newanswertech.com writes:
 I believe somebody (Carl) said this was fixed in the 
 latest version, but in a compilation of a version 
 obtained an hour ago from latest, it is still happening 
 when segmenting...
 
 This is ticket #3741.

Yes, I re-opened it:
https://trac.ffmpeg.org/ticket/3741#comment:10

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Re: [FFmpeg-user] only audio ffmpeg

[Christian Ebert]

* Carl Eugen Hoyos on Monday, July 21, 2014 at 15:01:08 +
 Reindl Harald h.reindl at thelounge.net writes:
 at least for opus you need to enable the external 
 library --enable-libopus is your friend
 
 FFmpeg contains a native opus decoder, libopus is not 
 needed.

Then the fine new website is not up to speed:
https://www.ffmpeg.org/ffmpeg-codecs.html#libopus-1

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Re: [FFmpeg-user] Packet header is not contained in global metadata when segmenting and encoding x264

[Christian Ebert]

* Werner Robitza on Wednesday, July 23, 2014 at 13:53:51 +0200
 On Wed, Jul 23, 2014 at 12:42 PM, Carl Eugen Hoyos ceho...@ag.or.at wrote:
 Why are you using a bitstream filter when re-encoding?
 The normal usecase for a bitstream filter is remuxing.
 
 If I don't use it, e.g.
 
 ffmpeg -y -i tmp/tmpPass2.mp4 -c:v libx264 -acodec copy -map 0 -f
 segment -segment_time 5 -segment_format mpegts
 ./tsSegmentsDisplay/SRC04_Q1_%05d.ts
 
 I get:
 
 [mpegts @ 0x41437e0] H.264 bitstream malformed, no startcode found,
 use the h264_mp4toannexb bitstream filter (-bsf h264_mp4toannexb)

I guess you just want to copy (-c copy both video and audio), not
re-encode (-c:v libx264):

ffmpeg -y -i tmp/tmpPass2.mp4 -c copy -bsf:v h264_mp4toannexb
-map 0 -f segment -segment_time 5 -segment_format mpegts
./tsSegmentsDisplay/SRC04_Q1_%05d.ts

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Re: [FFmpeg-user] Alternative options for comma-separated filters?

[Christian Ebert]

* Sam Marrocco on Thursday, July 10, 2014 at 12:31:02 -0400
 I have an application that uses ffmpeg to perform file conversions including 
 applying the filter -vf colormatrix=bt601:bt709. I would like to also apply a 
 gamma changing filter such as -vf mp=eq2=1:1:0:1:2:1:1:1.
 
 The problem is that the application using ffmpeg (which I can *not* change) 
 cannot have commas included in the arguments provided. No combination of 
 escapes, quoting, etc will work if there is a comma anywhere in the arguments.
 
 This seems to preclude the possibility of having two filters since as near as 
 I can tell, the only way to do so would be
 
 -vf colormatrix=bt601:bt709, mp=eq2=1:1:0:1:2:1:1:1

I guess the space (argument separator in the shell) is the
problem here, not the comma. Try:

-vf colormatrix=bt601:bt709,mp=eq2=1:1:0:1:2:1:1:1

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