[FFmpeg-user] Fwd: librsvg-2.0 not found using pkg-config

[Charles Lawrence]

Created a debian-based dockerfile a few days ago that successfully
built ffmpeg 3.4.1 from source with librsvg enabled.

However today, without any intervening changes in the build script,
librsvg fails due to "librsvg-2.0 not found using pkg-config".

The 3.4.1 archive hasn't been changed as far as I can tell. If I
disable the librsvg options, ffmpeg builds as expected.

The build instructions are basically verbatim from the ffmpeg
compilation guide for Debian with some slight modifications for the
dockerfile and to add librsvg.

My dockerfile is included below. Hoping someone has some insight.

DOCKERFILE:

FROM debian:stretch


ARG FFMPEG_VERSION


RUN apt-get update -qq && apt-get -y install \

  autoconf \

  automake \

  build-essential \

  cmake \

  git \

  libass-dev \

  libfreetype6-dev \

  librsvg2-dev \

  libtheora-dev \

  libtool \

  libvorbis-dev \

  libx264-dev \

  mercurial \

  pkg-config \

  texinfo \

  curl \

  zlib1g-dev \

  nasm \

  yasm \

  zip


ENV cwd=/usr/local/apps/ffmpeg

WORKDIR ${cwd}


RUN mkdir -p ${cwd} && \

  mkdir -p ${cwd}/ffmpeg_sources \

  mkdir -p ${cwd}/ffmpeg_build \

  mkdir -p ${cwd}/ffmpeg_bin


RUN cd ${cwd}/ffmpeg_sources && \

  curl -O http://ffmpeg.org/releases/ffmpeg-${FFMPEG_VERSION}.tar.gz && \

  tar xvf ffmpeg-${FFMPEG_VERSION}.tar.gz && \

  cd ffmpeg-${FFMPEG_VERSION} && \

  PATH="${cwd}/ffmpeg_bin:$PATH"
PKG_CONFIG_PATH="${cwd}/ffmpeg_build/lib/pkgconfig" ./configure \

--prefix="${cwd}/ffmpeg_build" \

--pkg-config-flags="--static" \

--extra-cflags="-I${cwd}/ffmpeg_build/include" \

--extra-ldflags="-L${cwd}/ffmpeg_build/lib" \

--extra-libs="-lpthread -lm" \

--bindir="${cwd}/ffmpeg_bin" \

--enable-gpl \

--enable-libass \

--enable-libfontconfig \

--enable-libfreetype \

--enable-librsvg \

--enable-libx264 \

--enable-nonfree && \

  PATH="${cwd}/ffmpeg_bin:$PATH" make && \

  make install && \

  hash -r && \

  mv ${cwd}/ffmpeg_bin/* /usr/local/bin
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Re: [FFmpeg-user] Strange results with -preset

[Dave]

On Sunday 28 January 2018 18:38:48 Cecil Westerhof wrote:
> So it looks like I should always use superfast. (But probably a good
> idea to test it with a few other videos.)
> 
> Anyone an idea what could be happening here?

Some good answers already but something I read while researching
similar effects using x265 encoding told me that preset is just a
shorthand for a specific set of encoding options, some affect speed,
some affect quality.  Said combinations are defined by whoever set
up the preset keywords and there is no specific relationship between
size, speed and the key word chosen because the source material
may affect how the pre-selected options work, although in general
the fast presets will, in fact, give a faster encoding time.

If you really want to optimise sped and quality, you need to manually
choose your options to match the source materiel, eg lots of action,
cartoon animation, black and white etc.  I think there's a way (or some
website somewhere) which documents the full command line settings
that are enacted when choosing a preset.

Good luck!
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Re: [FFmpeg-user] Error converting .m4a to .mp3

[Bernhard Döbler]

Am 28.01.2018 um 22:59 schrieb Carl Eugen Hoyos:

2018-01-28 21:53 GMT+01:00 Bernhard Döbler :


I want to convert an mp4 file from iTunes to MP3 using ffmpeg.
For on certain file I get error:

Command line and complete, uncut console output missing.


C:\Users\bd>"C:\Program 
Files\ffmpeg-20180121-78e884f-win64-shared\bin\ffmpeg.exe" -y -i 
"E:\MP3\iTunes\Music\DJ Tomekk\Kimnotyze (Remastered) [Remixes] 
[feat_\01 Kimnotyze (Remastered) [feat. Lil.m4a" -c:a libmp3lame -q:a 2 
-vn  -compression_level 0  -cutoff 19500 Kimnotyze.mp3
ffmpeg version N-89861-g78e884f3fb Copyright (c) 2000-2018 the FFmpeg 
developers

  built with gcc 7.2.0 (GCC)
  configuration: --disable-static --enable-shared --enable-gpl 
--enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig 
--enable-gnutls --enable-iconv --enable-libass --enable-libbluray 
--enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus 
--enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora 
--enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp 
--enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg 
--enable-lzma --enable-zlib --enable-gmp --enable-libvidstab 
--enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa 
--enable-libspeex --enable-libmfx --enable-amf --enable-cuda 
--enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 
--enable-avisynth

  libavutil  56.  7.100 / 56.  7.100
  libavcodec 58.  9.100 / 58.  9.100
  libavformat    58.  5.100 / 58.  5.100
  libavdevice    58.  0.101 / 58.  0.101
  libavfilter 7. 11.101 /  7. 11.101
  libswscale  5.  0.101 /  5.  0.101
  libswresample   3.  0.101 /  3.  0.101
  libpostproc    55.  0.100 / 55.  0.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 025c9a13e340] stream 0, timescale not set
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'E:\MP3\iTunes\Music\DJ 
Tomekk\Kimnotyze (Remastered) [Remixes] [feat_\01 Kimnotyze (Remastered) 
[feat. Lil.m4a':

  Metadata:
    major_brand : M4A
    minor_version   : 0
    compatible_brands: M4A mp42isom
    creation_time   : 1998-05-11T06:03:56.00Z
    iTunSMPB    :   0840 0314 007C1CAC 
       
    iTunNORM    :  1D6E 1E33 00015D8B 00015527 99D5 
99D5 7B9F 7BA1 00016DB7 00010252
    title   : Kimnotyze (Remastered) [feat. Lil Kim & Trooper 
Da Don]

    artist  : DJ Tomekk
    album_artist    : DJ Tomekk
    composer    : DJ Tomekk
    album   : Kimnotyze (Remastered) [Remixes] [feat. Lil Kim & 
Trooper Da Don] - EP

    genre   : Hip-Hop/Rap
    track   : 1/4
    disc    : 1/1
    compilation : 0
    date    : 2012-10-05T07:00:00Z
    media_type  : 1
    purchase_date   : 2014-05-21 20:24:56
    sort_name   : Kimnotyze (Remastered) [feat. Lil Kim & Trooper 
Da Don]
    sort_album  : Kimnotyze (Remastered) [Remixes] [feat. Lil Kim & 
Trooper Da Don] - EP

    sort_artist : DJ Tomekk
  Duration: 00:03:04.51, start: 0.047889, bitrate: 284 kb/s
    Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, 
stereo, fltp, 259 kb/s (default)

    Metadata:
  creation_time   : 1998-05-11T06:03:56.00Z
    Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 
600x600 [SAR 300:300 DAR 1:1], 90k tbr, 90k tbn, 90k tbc

Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, mp3, to 'Kimnotyze.mp3':
  Metadata:
    major_brand : M4A
    minor_version   : 0
    compatible_brands: M4A mp42isom
    sort_artist : DJ Tomekk
    iTunSMPB    :   0840 0314 007C1CAC 
       
    iTunNORM    :  1D6E 1E33 00015D8B 00015527 99D5 
99D5 7B9F 7BA1 00016DB7 00010252
    TIT2    : Kimnotyze (Remastered) [feat. Lil Kim & Trooper 
Da Don]

    TPE1    : DJ Tomekk
    TPE2    : DJ Tomekk
    TCOM    : DJ Tomekk
    TALB    : Kimnotyze (Remastered) [Remixes] [feat. Lil Kim & 
Trooper Da Don] - EP

    TCON    : Hip-Hop/Rap
    TRCK    : 1/4
    TPOS    : 1/1
    TCMP    : 0
    TDRC    : 2012-10-05T07:00:00Z
    media_type  : 1
    purchase_date   : 2014-05-21 20:24:56
    sort_name   : Kimnotyze (Remastered) [feat. Lil Kim & Trooper 
Da Don]
    sort_album  : Kimnotyze (Remastered) [Remixes] [feat. Lil Kim & 
Trooper Da Don] - EP

    TSSE    : Lavf58.5.100
    Stream #0:0(eng): Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp 
(default)

    Metadata:
  creation_time   : 1998-05-11T06:03:56.00Z
  encoder : Lavc58.9.100 libmp3lame
Assertion failed!

Program: C:\Program 

Re: [FFmpeg-user] ADPCM block size for XAudio

[Martin (Imagine Earth)]

>
> Is the issue also reproducible with WMP or only with "XAudio"?


No, only with XAudio. WMP plays all of them properly.
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Re: [FFmpeg-user] Avid opAtom trouble

[algia]

Hi all,
I have the same trouble.
I used the latest version of FFmpeg downloaded 2018-01-26 from Zeranoe
The trouble is only if I import the file.
but if I use AMA relink all is ok.
my workaround is FFmbc but i loking for ffmpeg solution
Thk
Livio



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[FFmpeg-user] Need help Axis encoder

[Cushman, Aaron]

I have Axis q7406 axis video encoders that send h.264.  I am trying to get them 
to work with Blonder Tongue and they will not as axis does not send H.264 over 
RTP (RFC 3984).  I am trying to see if there is a way to use FFMPEG to take the 
stream from the axis and resend UDP/RTP do you know if that would be possible.
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Re: [FFmpeg-user] Strange results with -preset

[Carl Eugen Hoyos]

2018-01-29 16:15 GMT+01:00 Nicolas George :
> Carl Eugen Hoyos (2018-01-29):
>> And therefore, no specific output file size can be expected
>> afaiu.
>
> Indeed, but you can expect the same quality, and therefore a
> decreasing size-vs-time mapping.

I didn't know that.

In this case, there likely is a bug in x264 (or FFmpeg).

Carl Eugen
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[FFmpeg-user] Replace part of video with series of images

[Joshua Grauman]

Hi all,

Would someone mind helping me to get a commandline correct?

I have a video, screencast.avi, of utvideo with transparency.

I extract a couple seconds of images from screencast.avi like:

ffmpeg -ss 31 -t 2 -i screencast.avi filename%03d.png

I would like to edit the images and then "put them back" into the video, 
replacing the appropriate frames.


I know how to use the overlay filter, but don't know how to make it use 
the whole series of images at the correct time. Plus, I can't simply 
overlay the video as it has transparency. I would need to replace the 
frames, or blank the appropriate frames before overlaying.


Thanks for help getting the command correct!

Josh

Here are my attempts for anyone interested:

This command uses all the images, but doesn't start and end them at the 
right place:


ffmpeg -i screencast.avi -pattern_type glob -i "filename*.png" -filter_complex 
overlay output.avi

This command puts the overlay in the correct frames, but only uses a 
single .png file, not all of them:


ffmpeg -i screencast.avi -pattern_type glob -i "filename*.png" -filter_complex 
"overlay=0:0:enable='between(t,2,3)'" output.avi

Plus, both commands incorrectly overlay the transparent images, rather 
than replacing the frames.


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Re: [FFmpeg-user] Strange results with -preset

[Nicolas George]

Carl Eugen Hoyos (2018-01-29):
> And therefore, no specific output file size can be expected
> afaiu.

Indeed, but you can expect the same quality, and therefore a decreasing
size-vs-time mapping.

Regards,

-- 
  Nicolas George


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[FFmpeg-user] HLS: multiple retries on HTTP 404 errors

[den....@yandex.ru]

Hello

I copy live HLS stream with the command like this: 
ffmpeg -probesize 1000 -y -i 
"https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/index_0720_av-p.m3u8?sd=10=on;
 \
-t 7200 -map 0:0 -map 0:1 -c copy o.mkv

It works most of the time, but sometimes some requests for new HLS segments 
result in "HTTP error 404 Not Found" errors:
...
[hls,applehttp @ 0x55e993a7ef80] Opening 
'https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/segment151719160_0540_av-p.ts?sd=10=on'
 for reading
[hls,applehttp @ 0x55e993a7ef80] Opening 
'https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/segment151719161_0540_av-p.ts?sd=10=on'
 for reading
[https @ 0x55e9942e2420] HTTP error 404 Not Found
[hls,applehttp @ 0x55e993a7ef80] Failed to open segment of playlist 0
[hls,applehttp @ 0x55e993a7ef80] Opening 
'https://rfe-lh.akamaihd.net/i/rfe_tvmc5@383630/segment151719162_0540_av-p.ts?sd=10=on'
 for reading
frame=101178 fps= 25 q=-1.0 size= 1048779kB time=01:07:39.98 
bitrate=2116.2kbits/s speed=1.01x   
...

From this log we can see that the request for the segment with name 
"segment151719161_0540_av-p.ts" results in HTTP 404 error. It looks like a bug 
of the streaming service, but I also think that this segment could be available 
if we make another request(maybe after some timeout). 

Is it possible to configure ffmpeg to do multiple retries on HTTP errors for 
buggy/missed/not-yet-created HLS segments?
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Re: [FFmpeg-user] FFmpeg dumps audio raw data of more size than expected

[Cley Faye]

2018-01-29 14:16 GMT+01:00 m.kamalasubha m.kamalasubha <
m.kamalasu...@gmail.com>:

> Hello all,
> I need to generate audio raw buffer for some 30 seconds from a clip whose
> sample rate is 44100 Hz. I used the following FFmpeg command to dump the
> raw data.
>   ffmpeg -i  -ss 00:00:00  -t 00:00:30 -f s16le -acodec pcm_s16le
> -ac 1 -ar 44100 audi.raw
>
> Expected audio raw file size : 44100 * 2 * 30 = 2646000
> Actual audio raw file size : 2646176
>
> I am wondering from where that extra 176 bytes arrive for 30 secs samples
> with sample rate 44100 Hz and sample size 2 bytes respectively.
>
>
>
​You really should post the complete output from ffmpeg here, it's the
*only* way to know what might be happening when the unexpected happens :)

(for the record, I tried your command and the output was as expected).​
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[FFmpeg-user] FFmpeg dumps audio raw data of more size than expected

[m.kamalasubha m.kamalasubha]

Hello all,
I need to generate audio raw buffer for some 30 seconds from a clip whose
sample rate is 44100 Hz. I used the following FFmpeg command to dump the
raw data.
  ffmpeg -i  -ss 00:00:00  -t 00:00:30 -f s16le -acodec pcm_s16le
-ac 1 -ar 44100 audi.raw

Expected audio raw file size : 44100 * 2 * 30 = 2646000
Actual audio raw file size : 2646176

I am wondering from where that extra 176 bytes arrive for 30 secs samples
with sample rate 44100 Hz and sample size 2 bytes respectively.

Kindly help on those lines.

Thanks,
Shalini
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Re: [FFmpeg-user] FFmpeg doing slow encoding

[Reindl Harald]

Am 29.01.2018 um 10:42 schrieb Sai Krishna Kothapalli:

My set up includes a input live RTMP stream, and apply a custom video
filter and output  RTMP stream which is converted to HLS and served to
clients. Now my issue is that FFmpeg is taking a lot of time to do this
process (like it is encoding at 0.3 - 0.4X speed). If I give a mp4 file as
input and output a mp4 file for testing purposes it is taking  nearly 1:30
sec for a 25 sec 4K video clip. The problem is caused when by the time the
client finished watching that 25 sec .ts file and by that time the next .ts
file is produced the video player just buffers. The video player buffers
for 1:05 sec to play the next chunk and this continues. How should I
overcome this problem? i am using 8 core aws server for now. Should I use a
GPU instance and compile FFmpeg with GPU compatibility or can I solve this
issue by using more number of CPUs. I want it to encode at 1x times so that
it doesn't buffer on the client video player.


so what answer do you expect with no techincal information at all?

* sample stream
* command line you used
* output of the command
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[FFmpeg-user] FFmpeg doing slow encoding

[Sai Krishna Kothapalli]

Hello all,

My set up includes a input live RTMP stream, and apply a custom video
filter and output  RTMP stream which is converted to HLS and served to
clients. Now my issue is that FFmpeg is taking a lot of time to do this
process (like it is encoding at 0.3 - 0.4X speed). If I give a mp4 file as
input and output a mp4 file for testing purposes it is taking  nearly 1:30
sec for a 25 sec 4K video clip. The problem is caused when by the time the
client finished watching that 25 sec .ts file and by that time the next .ts
file is produced the video player just buffers. The video player buffers
for 1:05 sec to play the next chunk and this continues. How should I
overcome this problem? i am using 8 core aws server for now. Should I use a
GPU instance and compile FFmpeg with GPU compatibility or can I solve this
issue by using more number of CPUs. I want it to encode at 1x times so that
it doesn't buffer on the client video player.

Thank you.
-- 
K. Mohan Sai Krishna
Roll No: 130101039
Computer Science and Engineering
IIT Guwahati
India
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