[FFmpeg-user] FFmpeg website: Start.com certificate trust removal

2017-07-12 Thread Tony Theodore
Hi,

I’m not sure if this is the right list to ask about the certificates 
used on the website. Since October last year, Mozilla and Google have
been distrusting new Start.com certificates[1][2] and Chromium have now
removed trust entirely[3]. Debian has also removed these certs[4] so
tools like wget and curl fail when accessing https links.

Are there any plans to replace these certs (possibly with LetEncrypt)?

Cheers,

Tony

[1] 
https://blog.mozilla.org/security/2016/10/24/distrusting-new-wosign-and-startcom-certificates/
[2] https://security.googleblog.com/2016/10/distrusting-wosign-and-startcom.html
[3] https://groups.google.com/a/chromium.org/forum/#!topic/net-dev/FKXe-76GO8Y
[4] https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=858539
[5] https://letsencrypt.org
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Re: [FFmpeg-user] Converting mkv to mp3 failures

2017-07-12 Thread Tim Hiles
On Wed, Jul 12, 2017 at 7:54 PM, JD  wrote:

> I just wanted to point out that 99% of files I download
> as mkv, and then try to convert them to mp3 with the command:
>
> ffmpeg -i somefile.mkv -vn -c:a libmp3lame -ab 320k -ac 2 -y somefile.mp3
>
> fail to convert, with huge amount of error messages output.
>
> PS: Using ffmpeg version 3.0.2-static
>
> The conversion to mkv is done by youtube-dl automatically.
>
> So, is youtube-dl the culprit by creating bad mkv output?
>
> Would love to hear how I can make this work.
>
>
Please post full uncut command output.
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[FFmpeg-user] Converting mkv to mp3 failures

2017-07-12 Thread JD

I just wanted to point out that 99% of files I download
as mkv, and then try to convert them to mp3 with the command:

ffmpeg -i somefile.mkv -vn -c:a libmp3lame -ab 320k -ac 2 -y somefile.mp3

fail to convert, with huge amount of error messages output.

PS: Using ffmpeg version 3.0.2-static

The conversion to mkv is done by youtube-dl automatically.

So, is youtube-dl the culprit by creating bad mkv output?

Would love to hear how I can make this work.

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[FFmpeg-user] Compile master on i686

2017-07-12 Thread Hans Carlson
I'm unable to compile the latest git master on an old i686 Linux (Fedora) 
system I still use.  See errors below.


The last time I compiled on this system was Mar 29, 2017 and it worked 
fine.  The commit ID was (247d0339ca5ddab692aee49baf43cd1324466028).


Here's truncated output from make.  I'm happy to provide more details, 
just not sure what would be useful.  The git clone, and configure ran 
without any errors.  make showed no errors until the ones below.


  $ uname -srvmpi
  Linux 2.6.35.14-106.fc14.i686 #1 SMP Wed Nov 23 13:57:33 UTC 2011 i686 i686 
i386
  $ git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg-20170712
  ...
  $ cd ffmpeg-20170712
  $ configure
  ...
  $ make
  ...
  X86ASMlibavfilter/x86/af_afir.o
  libavutil/x86/x86inc.asm:45: error: symbol `HAVE_ALIGNED_STACK' not defined 
before use
  libavutil/x86/x86inc.asm:58: error: symbol `ARCH_X86_64' not defined before 
use
  libavutil/x86/x86inc.asm:193: error: (DECLARE_REG_SIZE:11) symbol 
`ARCH_X86_64' not defined before use
  libavutil/x86/x86inc.asm:194: error: (DECLARE_REG_SIZE:11) symbol 
`ARCH_X86_64' not defined before use
  libavutil/x86/x86inc.asm:195: error: (DECLARE_REG_SIZE:11) symbol 
`ARCH_X86_64' not defined before use
  libavutil/x86/x86inc.asm:196: error: (DECLARE_REG_SIZE:11) symbol 
`ARCH_X86_64' not defined before use
  libavutil/x86/x86inc.asm:197: error: (DECLARE_REG_SIZE:11) symbol 
`ARCH_X86_64' not defined before use
  libavutil/x86/x86inc.asm:198: error: (DECLARE_REG_SIZE:11) symbol 
`ARCH_X86_64' not defined before use
  libavutil/x86/x86inc.asm:199: error: (DECLARE_REG_SIZE:11) symbol 
`ARCH_X86_64' not defined before use
  libavutil/x86/x86inc.asm:225: error: symbol `ARCH_X86_64' not defined before 
use
  libavutil/x86/x86inc.asm:938: error: (INIT_XMM:5) symbol `ARCH_X86_64' not 
defined before use
  libavutil/x86/x86inc.asm:938: error: (INIT_CPUFLAGS:32) symbol `ARCH_X86_64' 
not defined before use
  libavutil/x86/x86inc.asm:938: error: (CPUNOP:1) symbol `HAVE_CPUNOP' not 
defined before use
  libavutil/x86/x86inc.asm:938: error: unknown 'cpu' type
  libavfilter/x86/af_afir.asm:30: error: (INIT_XMM:5) symbol `ARCH_X86_64' not 
defined before use
  libavfilter/x86/af_afir.asm:30: error: (INIT_CPUFLAGS:32) symbol 
`ARCH_X86_64' not defined before use
  libavfilter/x86/af_afir.asm:30: error: (CPUNOP:1) symbol `HAVE_CPUNOP' not 
defined before use
  libavfilter/x86/af_afir.asm:30: error: unknown 'cpu' type
  libavfilter/x86/af_afir.asm:31: error: (INIT_XMM:5) symbol `ARCH_X86_64' not 
defined before use
  libavfilter/x86/af_afir.asm:31: error: (INIT_CPUFLAGS:32) symbol 
`ARCH_X86_64' not defined before use
  libavfilter/x86/af_afir.asm:31: error: (CPUNOP:1) symbol `HAVE_CPUNOP' not 
defined before use
  libavfilter/x86/af_afir.asm:31: error: unknown 'cpu' type
  make: *** [libavfilter/x86/af_afir.o] Error 1

Based on the error messages, it looks like it might not be getting the 
correct values for HAVE_ALIGNED_STACK and ARCH_X86_64?


  $ egrep 'HAVE_ALIGNED_STACK|ARCH_X86_64' config* ffbuild/*
  config.asm:%define ARCH_X86_64 0
  config.asm:%define HAVE_ALIGNED_STACK 1
  config.h:#define ARCH_X86_64 0
  config.h:#define HAVE_ALIGNED_STACK 1
  ffbuild/config.mak:!ARCH_X86_64=yes
  ffbuild/config.mak:HAVE_ALIGNED_STACK=yes

It appears the "ffbuild" subdir that contains all the config files was 
added at some point between Mar 29th and now... Maybe that has something 
to do with it?


FYI.  If I do the following, everything works fine:

  $ git clone git://source.ffmpeg.org/ffmpeg.git ffmpeg-20170712-247d0339ca
  $ cd ffmpeg-20170712-247d0339ca
  $ git checkout 247d0339ca
  $ configure
  $ make
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Re: [FFmpeg-user] MTS to MP4 lossless conversion issues

2017-07-12 Thread Nick Gilmour
Thanks for the reponses!



*What is the output of$ ffmpeg -i 0.ffmpeg1.mp4??*

ffmpeg version 3.3.2-1~16.04.york1 Copyright (c) 2000-2017 the FFmpeg
developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  configuration: --prefix=/usr --extra-version='1~16.04.york1'
--toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu
--incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping
--enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libflite --enable-libfontconfig
--enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm
--enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt
--enable-libopus --enable-libpulse --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh
--enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx
--enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid
--enable-libzmq --enable-libzvbi --enable-omx --enable-openal
--enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883
--enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264
--enable-shared
  WARNING: library configuration mismatch
  avcodec configuration: --prefix=/usr --extra-version='1~16.04.york1'
--toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu
--incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping
--enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa
--enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca
--enable-libcdio --enable-libflite --enable-libfontconfig
--enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm
--enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt
--enable-libopus --enable-libpulse --enable-librubberband --enable-libshine
--enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh
--enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx
--enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxvid
--enable-libzmq --enable-libzvbi --enable-omx --enable-openal
--enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libiec61883
--enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264
--enable-shared --enable-version3 --disable-doc --disable-programs
--enable-libopencore_amrnb --enable-libopencore_amrwb --enable-libtesseract
--enable-libvo_amrwbenc --enable-netcdf
  libavutil  55. 58.100 / 55. 58.100
  libavcodec 57. 89.100 / 57. 89.100
  libavformat57. 71.100 / 57. 71.100
  libavdevice57.  6.100 / 57.  6.100
  libavfilter 6. 82.100 /  6. 82.100
  libavresample   3.  5.  0 /  3.  5.  0
  libswscale  4.  6.100 /  4.  6.100
  libswresample   2.  7.100 /  2.  7.100
  libpostproc54.  5.100 / 54.  5.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '0.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf57.71.100
  Duration: 00:00:15.10, start: 0.00, bitrate: 12936 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv),
1280x720 [SAR 1:1 DAR 16:9], 12760 kb/s, 25 fps, 25 tbr, 90k tbn, 100 tbc
(default)
Metadata:
  handler_name: VideoHandler
Stream #0:1(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, stereo,
fltp, 192 kb/s (default)
Metadata:
  handler_name: SoundHandler
Side data:
  audio service type: main
At least one output file must be specified

I have rerun the command and now I'm getting a WARNING: library
configuration mismatch. I thinks it's because the Software Center forced me
to remove a package...


*Subtitles? Interesting, what do they contain? :)*

Actually there are no subtitles. This is weird...

To me this AVCHD format is uncomfortable. How am I supposed to separate the
files? I want to have everything in a single file. Anyway, I have decided
to convert the MTS files to mp4 and keep the AVCHD structure as a backup. I
see no quality loss with the generated mp4.


On Tue, Jul 11, 2017 at 1:34 PM, Moritz Barsnick  wrote:

> On Mon, Jul 10, 2017 at 15:03:49 +0200, Nick Gilmour wrote:
> > *track 1: codec frame size is not set*
> > 1. I googled also about this but I still don't know what does that mean
> and
> > how I should deal with it. Is it an error or should I change something?
>
> It about the audio track and I think it shouldn't matter. (I don't know
> absolutely for sure though.)
>
> > 2. I also see with VLC > Tools > Codec Information that the output file
> has
> > 25 frames per sec, whereas the input file has 50 frames per sec. This
> > although the output of the previous command shows 50 fps for the output
> > file. So what is going on? Is VLC wrong? How can I get the 50 fps in the
> > output file? Do 25 fps less mean reduced quality?
>
> If you copy the stream ("-c[:x] copy"), no quality 

Re: [FFmpeg-user] Issue when encoding a MP4 video

2017-07-12 Thread Cley Faye
2017-07-12 16:21 GMT+02:00 Léonce ANDRIANJAFY :

> Here is the complete output of the command:
>
> ffmpeg version 0.8.20-6:0.8.20-0+deb7u1, Copyright (c) 2000-2014 the Libav
> developers
>   built on Jan 19 2017 11:13:36 with gcc 4.7.2
> The ffmpeg program is only provided for script compatibility and will be
> removed
> in a future release. It has been deprecated in the Libav project to allow
> for
> incompatible command line syntax improvements in its replacement called
> avconv
>

​You are not using ffmpeg, you're using another product that masquerade
itself as an ffmpeg replacement using less than honest claims of
deprecation (to say the least).
​
You can get ffmpeg from https://www.ffmpeg.org/download.html , the exact
mean to do so depend on your system.
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Re: [FFmpeg-user] Issue when encoding a MP4 video

2017-07-12 Thread Léonce ANDRIANJAFY
Here is the complete output of the command:

ffmpeg version 0.8.20-6:0.8.20-0+deb7u1, Copyright (c) 2000-2014 the Libav
developers
  built on Jan 19 2017 11:13:36 with gcc 4.7.2
The ffmpeg program is only provided for script compatibility and will be
removed
in a future release. It has been deprecated in the Libav project to allow
for
incompatible command line syntax improvements in its replacement called
avconv
(see Changelog for details). Please use avconv instead.
[libspeex @ 0x646260] Missing Speex header, assuming defaults.

Seems stream 0 codec frame rate differs from container frame rate: 40.00
(40/1) -> 1000.00 (1000/1)
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '1281904405.mp4':
  Metadata:
major_brand : f4v
minor_version   : 0
compatible_brands: isommp42m4v
creation_time   : 2016-11-24 09:18:54
  Duration: 00:01:35.92, start: 0.00, bitrate: 186 kb/s
Stream #0.0(eng): Video: h264 (Main), yuv420p, 640x480 [PAR 1:1 DAR
4:3], 165 kb/s, 19.60 fps, 1k tbr, 90k tbn, 40 tbc
Metadata:
  creation_time   : 2016-11-24 09:18:54
Stream #0.1(eng): Audio: libspeex, 16000 Hz, 1 channels, s16, 18 kb/s
Metadata:
  creation_time   : 2016-11-24 09:18:54
[libspeex @ 0x646260] Missing Speex header, assuming defaults.
Output #0, mp4, to '1281904405_aac.mp4':
  Metadata:
major_brand : f4v
minor_version   : 0
compatible_brands: isommp42m4v
creation_time   : 2016-11-24 09:18:54
encoder : Lavf53.21.1
Stream #0.0(eng): Video: libx264, yuv420p, 640x480 [PAR 1:1 DAR 4:3],
q=2-31, 165 kb/s, 20 tbn, 20 tbc
Metadata:
  creation_time   : 2016-11-24 09:18:54
Stream #0.1(eng): Audio: libvo_aacenc, 16000 Hz, 1 channels, s16, 32
kb/s
Metadata:
  creation_time   : 2016-11-24 09:18:54
Stream mapping:
  Stream #0.0 -> #0.0
  Stream #0.1 -> #0.1
Press ctrl-c to stop encoding
[mp4 @ 0x646be0] Application provided invalid, non monotonically increasing
dts to muxer in stream 0: 16 >= 16
av_interleaved_write_frame(): Invalid argument

Regards,

2017-07-12 16:16 GMT+03:00 Cley Faye :

> 2017-07-12 14:39 GMT+02:00 Léonce ANDRIANJAFY :
>
> > Hello,
> > I try to convert a MP4 video with a libspeex audio codec to an another
> one
> > but with an aac audio codec. In other words, I would like to only get the
> > audio stream re-encoded
> >
> > I launch the command below:
> > *ffmpeg -i 1281904405.mp4 -vcodec copy -acodec libvo_aacenc -ab 32k
> > 1281904405_aac.mp4*
> >
> > But still I got this error:
> > *[mp4 @ 0x646be0] Application provided invalid, non monotonically
> > increasing dts to muxer in stream 0: 16 >= 16*
> > *av_interleaved_write_frame(): Invalid argument*
> >
> > Could you help me please?
> >
>
> ​You should post the complete output of ffmpeg.
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-- 

*Léonce ANDRIANJAFY*
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[FFmpeg-user] Issue when encoding a MP4 video

2017-07-12 Thread Léonce ANDRIANJAFY
Hello,
I try to convert a MP4 video with a libspeex audio codec to an another one
but with an aac audio codec. In other words, I would like to only get the
audio stream re-encoded

I launch the command below:
*ffmpeg -i 1281904405.mp4 -vcodec copy -acodec libvo_aacenc -ab 32k
1281904405_aac.mp4*

But still I got this error:
*[mp4 @ 0x646be0] Application provided invalid, non monotonically
increasing dts to muxer in stream 0: 16 >= 16*
*av_interleaved_write_frame(): Invalid argument*

Could you help me please?

Regards

-- 

*Léonce ANDRIANJAFY*
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Re: [FFmpeg-user] m4a won't play on Alexa

2017-07-12 Thread Evert Vorster
Recode to mp3.

Nothing beats it for compatibility.


On 11 July 2017 at 00:12, sean darcy  wrote:

> I've got a bunch of sonatas I've uploaded to Amazon Music. They were
> originally opus files, but Amazon requires mp3 or m4a . ( Why ? Got me.)
>
> So I re-encoded the opus files:
>
> ffmpeg -i sonata.opus -vn -c:a aac -b:a 128k sonata.m4a
>
> The m4a files play just fine on my laptop, even in the Amazon Music
> windows app, but not in Alexa.
>
> "Sorry, I am having trouble playing the music."
>
> In the Alexa forum I found the following helpful advice:
>
> Make sure you are not sending a content type header of video/mpg4 which
> would be wrong if this is an audio-only stream. Also, it could be that the
> encoding is incorrect, for example the moov element is not present in the
> stream, etc.
>
> Any suggestions on how to encode that's Alexa proof ?
>
> sean
>
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-- 
Evert Vorster
Isometrix Acquistion Superchief
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