[FFmpeg-user] How to select GPU in a multi GPU system

[DopeLabs]

hello,

how does one select which GPU to use for hardware acceleration when more than 1 
GPU is installed in the system?

some searching seemed to hint at using the -gpu option but it looks like that 
option has been deprecated? the option does not appear in the man pages or on 
the documentation pages on the ffmpeg website. trying to use the option seems 
to fail.

thank you

$ ffmpeg -i test.mov -c:v h264_videotoolbox -gpu 1 test.mp4
ffmpeg version N-107303-g9fec43a92d Copyright (c) 2000-2022 the FFmpeg 
developers
  built with Apple clang version 13.1.6 (clang-1316.0.21.2.5)
  configuration: --enable-static --enable-pthreads --enable-version3 --cc=clang 
--host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl 
--enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame 
--enable-libopus --enable-librubberband --enable-libsnappy 
--enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis 
--enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 
--enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig 
--enable-libfreetype --enable-frei0r --enable-libopencore-amrnb 
--enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex 
--enable-libsoxr --disable-libjack --disable-indev=jack --enable-libcaca 
--extra-libs=-liconv --prefix=/opt/miniconda3 --enable-macos-kperf 
--enable-libkvazaar --disable-shared --pkg-config-flags=--static --enable-neon 
--enable-nonfree --enable-runtime-cpudetect --enable-postproc 
--enable-libfdk_aac
  libavutil  57. 27.100 / 57. 27.100
  libavcodec 59. 36.100 / 59. 36.100
  libavformat59. 26.100 / 59. 26.100
  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
Unrecognized option 'gpu'.
Error splitting the argument list: Option not found
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Re: [FFmpeg-user] Reading or writing audio from audio device on MacOS

[DopeLabs]

> On May 9, 2020, at 7:21 14AM, Michael Glenn Williams 
>  wrote:
> 
> Hi All,
> 
> Is it possible to ask ffmpeg to read or write from/to an audio device on
> MacOS? For example read from the microphone or playback to headphones?

yes, you can use avfoundation. to get a list of input devices use the following 
command

ffmpeg -f avfoundation -list_devices true -i ""

that will provide you with a list of video and audio input devices...

input devices are specified by the ID number separated by a colon provided in 
the previous command.

you specify the devices as -i "videoid:audioid"

here is an example of what mine looks like:

[AVFoundation indev @ 0x7fc3d1703980] AVFoundation video devices:
[AVFoundation indev @ 0x7fc3d1703980] [0] CamTwist
[AVFoundation indev @ 0x7fc3d1703980] [1] FaceTime HD Camera (Built-in)
[AVFoundation indev @ 0x7fc3d1703980] [2] webcamoid
[AVFoundation indev @ 0x7fc3d1703980] [3] CamTwist (2VUY)
[AVFoundation indev @ 0x7fc3d1703980] [4] EpocCam
[AVFoundation indev @ 0x7fc3d1703980] [5] Capture screen 0
[AVFoundation indev @ 0x7fc3d1703980] [6] Capture screen 1
[AVFoundation indev @ 0x7fc3d1703980] [7] Capture screen 2
[AVFoundation indev @ 0x7fc3d1703980] AVFoundation audio devices:
[AVFoundation indev @ 0x7fc3d1703980] [0] Scarlett 18i20 USB
[AVFoundation indev @ 0x7fc3d1703980] [1] Built-in Microphone
[AVFoundation indev @ 0x7fc3d1703980] [2] EpocCam Microphone
[AVFoundation indev @ 0x7fc3d1703980] [3] aux1 


for example if i want to do a screen capture of screen 1 for the video and the 
built in mic for the audio, the command would look like this

ffmpeg -f avfoundation -i "6:1"

if i only want audio

ffmpeg -f avfoundation -i ":1"

im pretty sure that avfoundation is strictly input devices only... that being 
said, you could probably pipe it to ffplay if you wanted to monitor local 
playback, in which it would use the system configured output device.

there are other 3rd party tools you can use to route specific application audio 
output such as loopback or audiohijack.

cheers =]

> 
> Thank you!
> 
> -- 
>  Michael Williams
> 
> CEO | CTO
> 
> Smart Blonde Experiential
> 8054991994
> michael.glenn.willi...@smartblondeco.com
> totalvu.tv
> 13900 Marquesas Way, Marina Del Rey, CA
> 
> [image: facebook]
>  [image:
> twitter]  [image: linkedin]
> 
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Re: [FFmpeg-user] FFmpeg to RTMP streaming issue

[DopeLabs]

> On Jan 31, 2020, at 10:18 01AM, Vladimir Sobolev  wrote:
> 
> I'm trying to stream to online service via FFmpeg. All my test commands works 
> fine with restream.io preview and with local rtmp server viewer. My OBS setup 
> works fine too (same resolution, server and key). Only when I'm about to 
> start ffmpeg stream from my command line it stucks after few seconds.
> 
> My test command:
> 
> ffmpeg -stream_loop -1 \
>   -i noise.mov \
>   -c:v libx264 c:a aac \
>   -f flv 'rtmps://live-upload**'
> 

may i ask what behavior you are expecting when using -stream_loop -1?


> With -debug pict after some progress with input files and rtmp handshaking I 
> can see this:
> 
> Clipping frame in rate conversion by 0.08
> cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if 
> it occurs once at the start per stream)
>Last message repeated 10 times
> [h264 @ 0x7ff12001da00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 0
> cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if 
> it occurs once at the start per stream)
>Last message repeated 5 times
> [h264 @ 0x7ff12001e000] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if 
> it occurs once at the start per stream)
>Last message repeated 10 times
> [h264 @ 0x7ff12001e600] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 0
> cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if 
> it occurs once at the start per stream)
>Last message repeated 5 times
> [h264 @ 0x7ff12001ec00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if 
> it occurs once at the start per stream)
>Last message repeated 8 times
> [h264 @ 0x7ff12001ac00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 0
> cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if 
> it occurs once at the start per stream)
>Last message repeated 5 times
> [h264 @ 0x7ff12001da00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if 
> it occurs once at the start per stream)
>Last message repeated 5 times
> [h264 @ 0x7ff12001e000] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 0
> [libx264 @ 0x7ff120802400] frame=   0 QP=20.00 NAL=3 Slice:I Poc:0   I:3600 
> P:0SKIP:0size=216266 bytes
> [h264 @ 0x7ff12001e600] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> [libx264 @ 0x7ff120802400] frame=   1 QP=31.00 NAL=2 Slice:P Poc:2   I:287  
> P:3267 SKIP:46   size=48597 bytes
> [h264 @ 0x7ff12001ec00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 0
> [libx264 @ 0x7ff120802400] frame=   2 QP=30.00 NAL=2 Slice:P Poc:4   I:710  
> P:2888 SKIP:2size=79226 bytes
> 
> ..and that's it.
> 
> The same command works fine with any other online service except this one. 
> What I can do to debug it? Works fine with OBS, but won't with ffmpeg 
> command. noise.mov — 720×1280px mov file.
> 
> Full log:
> 
> iMac:Desktop 350d$ ffmpeg -stream_loop -1 -re -i noise.mov -c:v libx264 
> -preset ultrafast -c:a aac -f flv -debug pict 'rtmps://live-upload.***' 
> ffmpeg version N-96521-g5e62100152-tessus  https://evermeet.cx/ffmpeg/  
> Copyright (c) 2000-2020 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-libwavpack --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. 38.100 / 56. 38.100
>  libavcodec 58. 67.100 / 58. 67.100
>  libavformat58. 36.100 / 58. 36.100
>  libavdevice58.  9.103 / 58.  9.103
>  libavfilter 7. 71.100 /  7. 71.100
>  libswscale  5.  6.100 /  5.  6.100
>  libswresample   3.  6.100 /  3.  6.100
>  libpostproc55.  6.100 / 55.  6.100
> matched as AVOption 'debug' with argument 'pict'.
> Reading option 'rtmps://live-upload.' ... matched as output url.
> Finished splitting t

Re: [FFmpeg-user] ffmpeg to stream an already h.264 encoded stream

[DopeLabs]

> On Sep 9, 2019, at 5:59 28AM, Verachten Bruno  wrote:
> 
> Hi there,
> 
> this is my first message on this list, so please bear with me.
> I have access to a h.264 encoder that will supply a few rtmp streams
> corresponding to the various HDMI inputs it has. It is supposed to use
> ffmpeg by the way (TBS2605).
> Anyway... I have a Linux machine, and my goal is to:
> - grab one of these streams, add audio coming from a USB sound card
> (FocusRite 2i4 2nd Gen), and stream that to YouTube while recording on
> the disk too.
> - grab two of these streams, compose them on top of a template, add
> audio (like in https://youtu.be/01nM2Shnyz0?t=1638), stream that to
> YouTube while recording on the disk too.
> 
> For the time being, I have managed to stream to YouTube, but the
> YouTube platform is not happy with my stream quality, saying "The
> stream's current bitrate (1653.00 Kbps) is lower than the recommended
> bitrate. We recommend that you use a stream bitrate of 4500 Kbps.".
> The bitrate that is output by the encoder is supposed to be 4000Kbps.
> The ffmpeg command I have gives me around 2200/2400Kbps.
> So I guess I made a mess and decoded with my command the h.264 stream
> before encoding again... as the speed is around 0.962x, and the CPU is
> quite busy. I thought there was no need to decode/encode to add an
> audio stream (but I have no idea what I'm doing, as you had already
> guessed ;-) ).
> 
> I recompiled the 4.2 ffmpeg release sources on this very machine.
> 
> Here is the kind of output I get:
> ffmpeg version 4.2 Copyright (c) 2000-2019 the FFmpeg developers
>  built with gcc 7 (Ubuntu 7.4.0-1ubuntu1~18.04.1)
>  configuration: --enable-gpl --enable-version3 --disable-static
> --enable-shared --enable-small --enable-avisynth --enable-chromaprint
> --enable-frei0r --enable-gmp --enable-gnutls --enable-ladspa
> --enable-libaom --enable-libass --enable-libcaca --enable-libcdio
> --enable-libcodec2 --enable-libfontconfig --enable-libfreetype
> --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack
> --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopencore-amrwb
> --enable-libopenjpeg --enable-libopenmpt --enable-libopus
> --enable-libpulse --enable-librsvg --enable-librubberband
> --enable-librtmp --enable-libshine --enable-libsnappy --enable-libsoxr
> --enable-libspeex --enable-libssh --enable-libtesseract
> --enable-libtheora --enable-libtwolame --enable-libv4l2
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxvid --enable-libxml2 --enable-libzmq --enable-libzvbi
> --enable-lv2 --enable-libmysofa --enable-openal --enable-opencl
> --enable-opengl --enable-libdrm --enable-nonfree --enable-libfdk-aac
> --enable-libbluray --extra-libs='-lnettle -lhogweed -lgmp'
>  libavutil  56. 31.100 / 56. 31.100
>  libavcodec 58. 54.100 / 58. 54.100
>  libavformat58. 29.100 / 58. 29.100
>  libavdevice58.  8.100 / 58.  8.100
>  libavfilter 7. 57.100 /  7. 57.100
>  libswscale  5.  5.100 /  5.  5.100
>  libswresample   3.  5.100 /  3.  5.100
>  libpostproc55.  5.100 / 55.  5.100
> Input #0, flv, from 'rtmp://192.168.1.211:1935/live/stream0':
>  Duration: N/A, start: 15024.271000, bitrate: N/A
>Stream #0:0: Video: h264, yuv420p(progressive), 1920x1080, 30 fps,
> 30 tbr, 1k tbn
>Stream #0:1: Audio: aac, 48000 Hz, stereo, fltp
> Guessed Channel Layout for Input Stream #1.0 : stereo
> Input #1, pulse, from 'default':
>  Duration: N/A, start: 1568033472.683289, bitrate: 1536 kb/s
>Stream #1:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
> Stream mapping:

as you can see below, you are re-encoding... 

>  Stream #0:0 -> #0:0 (h264 (native) -> h264 (libx264))
>  Stream #0:1 -> #0:1 (aac (native) -> mp3 (libmp3lame))
> Press [q] to stop, [?] for help
> [libx264 @ 0x559c4df3e780] using cpu capabilities: MMX2 SSE2Fast SSSE3
> SSE4.2 AVX FMA3 BMI2 AVX2
> [libx264 @ 0x559c4df3e780] profile High, level 4.0
> [libx264 @ 0x559c4df3e780] 264 - core 152 r2854 e9a5903 - H.264/MPEG-4
> AVC codec - Copyleft 2003-2017 - http://www.videolan.org/x264.html -
> options: cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7
> psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1
> 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2
> threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1
> interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2
> b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=60
> keyint_min=6 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf
> mbtree=1 crf=3.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=7000
> vbv_bufsize=512 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40
> aq=1:1.00
> Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/dh5g-r8tf-x5qy-cse8':
>  Metadata:
>encoder : Lavf58.29.100
>Stream #0:

Re: [FFmpeg-user] Book with examples

[DopeLabs]

On Aug 16, 2019, at 7:48 25AM, Michael Koch  wrote:
> 
> Hello all,
> 
> this is not a question. During the last few years I've tested many things 
> with FFmpeg, and now I've written it all in this document:
> http://www.astro-electronic.de/FFmpeg_Book.pdf
> It contains many examples for FFmpeg and a few other programs, also for audio 
> processing, color grading and fulldome video production. Please feel free to 
> use all examples as a starting point for your own projects. If you find 
> errors, please let me know.

its unclear if the builds you are providing at: 
https://ffmpeg.zeranoe.com/builds/ are different from the builds that 
ffmpeg.org provides...

if they are the same, you may wish to consider updating the introduction to 
point directly to ffmpeg.org's download page...


Download site for the Windows and OS-X versions: 
https://ffmpeg.org/download.html


cheers =]


> This is a work in progress, so the chapter numbers will change when I add new 
> chapters in future. For the FFmpeg programming experts, there is a chapter 
> "FFmpeg: Suggestions for improvement".
> 
> Michael
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Re: [FFmpeg-user] audio volume normalize

[DopeLabs]

you can try using the loudnorm filter...

http://ffmpeg.org/ffmpeg-filters.html#loudnorm



> On Aug 10, 2019, at 8:02 49PM, Matthew Reus  wrote:
> 
> Hello , i am recently streaming  5 live stream . the output of video is
> fine but the audio of different live stream are dynamics .. some.are too
> high and some low .
> is there a way to normalize the audio label and volume on same range .. i m
> using ffmpeg on ubuntu server and install the ffmpeg from.compilation guide
> .
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Re: [FFmpeg-user] Help : Preserve meta data

[DopeLabs]

maybe this will help? 

https://libav.org/avconv.html#Metadata

> On Aug 6, 2019, at 2:30 01AM, Rockland  wrote:
> 
> Hello all,
> 
> Can someone tells me which option I can add among my configure options to
> preserve metadata during transcoding? In command line I know that the
> option is map_metadata but I'm using libav
> in my project.
> 
> This is my configure options :
> CONFIGURE_COMMAND
>env LD_PRELOAD=${ALGOLIB_DIR}/libalgolib.so
> /configure
>--prefix=
>--extra-cflags=-I/include
>--extra-ldflags=-L/lib
>--extra-libs=-lalgolib\ -lpthread
>--enable-gpl
>--enable-nonfree
>--enable-libfdk-aac
>--disable-doc
>--enable-pthreads
>--disable-programs
>--enable-shared
>--enable-static
>--enable-runtime-cpudetect
> 
> Regards
> ___
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Re: [FFmpeg-user] ffmpeg build failure on Mac

[DopeLabs]

> On Jul 17, 2019, at 6:12 24PM, Steve Saunders via ffmpeg-user 
>  wrote:
> 
> "
> steve$ ./configure --extra-ldflags=-L"$(pwd)/../libvpx/build" 
> --extra-cflags=-I"$(pwd)/../libvpx/build" --enable-libvpx
> nasm/yasm not found or too old. Use --disable-x86asm for a crippled build.
> 

yasm is available via macports 

$ sudo port install yasm

or via homebrew

$ brew install yasm


> If you think configure made a mistake, make sure you are using the latest
> version from Git.  If the latest version fails, report the problem to the
> ffmpeg-user@ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net.
> Include the log file "ffbuild/config.log" produced by configure as this will 
> help
> solve the problem.
> “
> 
> As part of the following recipe to get a ffmpeg/libvpx install that supports 
> VP9 HDR Profile 2: 
> 
> 
> git clone https://chromium.googlesource.com/webm/libvpx 
> 
> cd libvpx
> mkdir -p build
> cd build
> ../configure --enable-vp9-highbitdepth
> make -j16
> cd ../..
> git clone https://git.ffmpeg.org/ffmpeg.git 
> 
> cd ffmpeg
> ./configure --extra-ldflags=-L"$(pwd)/../libvpx/build" 
> --extra-cflags=-I"$(pwd)/../libvpx/build" --enable-libvpx
> make -j16
> 
> Here is the requested ffbuild log: 
> 
> 
> 
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Re: [FFmpeg-user] -to,-t options not honored...

[DopeLabs]

this is what i use in a script i use to record exactly 1 hour from ip cameras 
at home... 

ffmpeg  -i "rtsp://@IP:PORT/cam/realmonitor?channel=1&subtype=0" -c 
copy -movflags +faststart -t 01:00:00.000 "/Volumes/CCTV/FRONT_EXT_-_$(date 
+%Y-%m-%d_%H-%M-%S).mp4"

and to verify using ffprobe...

 $ ffprobe FRONT_EXT_-_2019-05-01_00-00-00.mp4
ffprobe version N-93632-gdf9ef925f9 Copyright (c) 2007-2019 the FFmpeg 
developers
  built with Apple LLVM version 10.0.1 (clang-1001.0.46.4)
  configuration: --enable-filter=gltransition --extra-libs='-lGLEW -lglfw' 
--enable-shared --enable-gpl --enable-nonfree --enable-avresample 
--enable-avisynth --enable-gnutls --enable-libass --enable-libcaca 
--enable-libfontconfig --enable-libfreetype --enable-libfribidi 
--enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus 
--enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora 
--enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp 
--enable-libx265 --enable-libxvid --enable-libzmq --enable-libx264 
--enable-libfdk-aac --prefix=/opt/local --enable-opengl
  libavutil  56. 26.100 / 56. 26.100
  libavcodec 58. 52.100 / 58. 52.100
  libavformat58. 27.103 / 58. 27.103
  libavdevice58.  7.100 / 58.  7.100
  libavfilter 7. 49.100 /  7. 49.100
  libavresample   4.  0.  0 /  4.  0.  0
  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 'FRONT_EXT_-_2019-05-01_00-00-00.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
title   : Media Server
encoder : Lavf58.27.103
  Duration: 01:00:00.01, start: 0.00, bitrate: 6129 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, 
bt709), 2560x1440 [SAR 1:1 DAR 16:9], 6127 kb/s, 24.98 fps, 25 tbr, 90k tbn, 50 
tbc (default)
Metadata:
  handler_name: VideoHandler


hope this helps =]

DL


> On May 5, 2019, at 9:20 10AM, Calvin Arndt  wrote:
> 
> In a effort to create a camera recording of an exact amount of time.
> 
> FFmpeg/ffmpeg -y -loglevel 16 -i http://10.17.24.6/mjpg/video.mjpg -to 
> 00:01:00 -filter:v fps=fps=7 -b 512k /DVR/Tower2/05-05_11-04.avi
> 
> : ~# ls -l /DVR/Tower2/05-05_11-04.avi
> -rw-r--r-- 1 root root 5314038 May 5 11:07 /DVR/Tower2/05-05_11-04.avi
> 
> Without the -to 00:01:00 the file is perfect. The file plays a exactly the 
> appropriate frame rate.
> the clock is in sync with the timestamps in the video.
> 
> With the -to 00:01:00 option the file plays at wrong frame rate (approx 2.5x)
> And it records for 3-5 minutes!
> 
> using -t 3600 before or after the -i yeilds similar results?
> 
> What am I doing wrong?
> 
> Extra points for explaining why -to effects output frame rate! :)
> 
> -- 
> Calvin Arndt
> (217) 377-2979
> car...@macksrecycling.com
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Re: [FFmpeg-user] ffmpeg - executing 2 or more command in one time

[DopeLabs]

you can probably combine all that into a single filter chain


ffmpeg -f concat -i videos.txt -i music.mp3 -i logo.png -i picture.png 
-filter_complex 
"[0:v][1:v]overlay=main_w-overlay_w-20:20[bg];[bg][2:v]overlay=10:10,drawtext=fontfile=verdanab.ttf:text=HELLO:fontcolor=white:fontsize=36:bordercolor=black:borderw=5:x=170:y=80-text_h/2,subtitles=subtitle.srt"
 -c:a copy OUTPUT.mp4

(untested)


> On Apr 27, 2019, at 1:07 23AM, Ahmet Güler  wrote:
> 
> Hello,
> 
> 
> 
> I have 4 ffmpeg commands, I execute them respectively, they work, but I want
> to execute them in one time and one command. How to do that?
> 
> ffmpeg -f concat -i videos.txt -i music.mp3 -c copy VIDEO1.mp4
> 
> 
> 
> ffmpeg -i VIDEO1.mp4 -i logo.png -i picture.png -filter_complex
> "[0:v][1:v]overlay=main_w-overlay_w-20:20[bg];[bg][2:v]overlay=10:10[v]"
> -map "[v]" -map 0:a -c:a copy VIDEO2.mp4
> 
> 
> 
> ffmpeg -i VIDEO2.mp4 -vf drawtext="fontfile=verdanab.ttf: text='HELLO':
> fontcolor=white: fontsize=36: bordercolor=black: borderw=5: x=170:
> y=80-text_h/2" -codec:a copy VIDEO3.mp4
> 
> 
> 
> ffmpeg -i VIDEO3.mp4 -vf subtitles=subtitle.srt OUTPUT.mp4
> 
> 
> 
> Best Regards,
> 
> 
> 
> ---
> Bu e-posta virüslere karşı Avast antivirüs yazılımı tarafından kontrol 
> edilmiştir.
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Re: [FFmpeg-user] problem with ffmpeg -c copy from pipe

[DopeLabs]

> On Apr 20, 2019, at 10:24 37PM, Ivan Kirillov  wrote:
> 
> im trying to rip my stream with streamlink and then send it to my local rtmp 
> server by using
> 
> |streamlink https://www.twitch.tv/mytwitch best -O | ffmpeg -i pipe: -c copy 
> -f flv rtmp://127.0.0.1/live|
> 

try this

ffmpeg -i "`streamlink --stream-url --default-stream best 
https://www.twitch.tv/mytwitch`"; -c copy -f flv rtmp://

> im geting an error
> 
> |error: Error when writing to output: [Errno 32] Broken pipe, exiting|
> 

output...  again try writing to a local file first, then test your rtmp server 
and make sure that works using local file as input


> what am i doing wrong guys ?
> 
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Re: [FFmpeg-user] problem with ffmpeg -c copy from pipe

[DopeLabs]

> On Apr 20, 2019, at 10:24 37PM, Ivan Kirillov  wrote:
> 
> im trying to rip my stream with streamlink and then send it to my local rtmp 
> server by using
> 
> |streamlink https://www.twitch.tv/mytwitch best -O | ffmpeg -i pipe: -c copy 
> -f flv rtmp://127.0.0.1/live|
> 

im not familiar with streamlink, but i expect there should be an option for 
streamlink to just return the m3u8 url?

with a similar tool called youtube-dl the command would look like this

$ ffmpeg -i "`youtube-dl -f best -g https://twitch.tv/mytwitch`"; -c copy -f flv 
rtmp://127.0.0.1/app/stream

> im geting an error
> 
> |error: Error when writing to output: [Errno 32] Broken pipe, exiting|
> 

the complete console output would make things easier =]

> what am i doing wrong guys ?
> 

can you use the same command but instead save to a local file? 

for example, does this work?

> streamlink https://www.twitch.tv/mytwitch best -O | ffmpeg -i pipe: -c copy 
> outfile.mp4


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Re: [FFmpeg-user] Efficiently doing thousands of edits?

[DopeLabs]

cropping out the problem quarter to a separate file seems like it could be 
faster... less pixels to black detect.

if you know the start and stop times/frames, and also keep in mind that:

OPTIONS FOR FILTERS WITH SEVERAL INPUTS
   Some filters with several inputs support a common set of options.  These 
options can only be set by name, not with the short notation.

   eof_action
   The action to take when EOF is encountered on the secondary input; 
it accepts one of the following values:

   repeat
   Repeat the last frame (the default).

   endall
   End both streams.

   pass
   Pass the main input through.

   shortest
   If set to 1, force the output to terminate when the shortest input 
terminates. Default value is 0.

   repeatlast
   If set to 1, force the filter to extend the last frame of secondary 
streams until the end of the primary stream. A value of 0 disables this 
behavior.  Default value is 1.

setting eof_action=repeat might just do what your looking for... 

though it would be nice to know which filters 'some filters' are...

also the tpad filter looks like it would be a good contender as well..

you can pad either before or after the input stream, and use the clone mode 
which would be clones of the first or last frame respectively.

 
tbh, after seeing the yt video, i personally would probably just overlay the 
bottom left with solid black once it starts to flicker as i, for the most part, 
consider that unwatchable anyways =]

if this is an ongoing problem, maybe its time for a new camera? =]

cheers,

DL

> On Apr 11, 2019, at 3:01 28AM, Bouke  wrote:
> 
> 
> 
>> On 11 Apr 2019, at 04:44, John Hawkinson  wrote:
>> 
>> Michael Shaffer  wrote on Wed, 10 Apr 2019
>> at 19:40:36 -0400 in 
>> :
>> 
>>> I'm pretty sure you could use Python and OpenCV to create a solution..
>> 
>> "Just because you can, doesn't mean you should."
>> 
>> My original post explained how to determine where the runs of black are, 
>> using ffmpeg (libavfilter)'s "blackdetect." The only piece left is applying 
>> the edits in a practical fashion.
>> 
>> Rolling your own with opencv would be a whole lot more work. I'm not 
>> entirely sure that I agree with Carl Eugen that ffmpeg isn't a video editing 
>> tool, but if it's not, Python certainly isn't. Yes, you can make it work. 
>> But you're going to spend a lot more time doing it than if higher level 
>> tools were used.
>> 
>> The goal here is for higher-level tools, not lower-level ones.
>> 
>> Or I might just end up using ffmpeg to split it into 7,000 files and then 
>> concatenating them. That's hardly the worst thing in the world.
> 
> Why split and cat? You are not doing editing, you want to mask (little) 
> pieces, not shorten / extend / reshuffle, right?
> If you have all the in/out point, why not do a (png or alike) overlay on 
> those points?
> Or, a subtitle overlay with a strange custom font (one big black rectangle 
> char).
> 
> Bouke
> 
> 
>> --
>> jh...@alum.mit.edu
>> John Hawkinson
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Re: [FFmpeg-user] Changing start number in segment name - ffmpeg mpeg dash encoder

[DopeLabs]

looks like you can specify if you use webm_dash_manifest.. check the manpage

> On Mar 8, 2019, at 12:05 57AM, 韦林  wrote:
> 
> it seems ffmpeg had hardcode starting from 1.
> you could modify source code file 'libavformat/dashenc.c’ in function 
> dash_init, member ‘segment_index' of OutputStream.
> i’m not sure which unexpected result would be raised up.
> 
>> 在 2019年3月8日，下午2:27，Sai Balaji  写道：
>> 
>> Hi, I'm trying to implement MPEG dash live streaming using FFmpeg dash
>> encoder and shaka player.
>> 
>> Is there any way to set the start number to any value other than 1?
>> 
>> For example, chunk-stream0-9.m4s instead of the default starting
>> segment chunk-stream0-1.m4s
>> 
>> ffmpeg -f avfoundation -video_size 800x600 -framerate 30 -i 0  -vcodec
>> libx264 -x264-params keyint=30:scenecut=0  -acodec aac  -b:v 1M -b:a 192k
>> -f dash  -use_template 0 -use_timeline 0 -single_file 0  -streaming 1
>> -start_number
>> 10 -seg_duration  1 sai.mpd
>> 
>> 
>> This is the command I am using.
>> 
>> 
>> Thanks and regards,
>> 
>> Sai
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Re: [FFmpeg-user] ffplay black screen

[DopeLabs]

i can confirm i also have this same issue since upgrading to mojave.

i have a 5k imac, and additional displays.. a 4k and a 1080p.

things work fine when moving the window between the external displays but goes 
black when i try to move the window to the 5k built in display.

even using ffplays build in showspectrum when playing an audio file shows the 
same behavior..



> On Mar 6, 2019, at 6:53 39PM, 韦林  wrote:
> 
> I am trying to use ffplay to play a video on my macOS.
> What I see was a black screen, no matter what file I try.
> When I drag the playing window to another screen (not the main screen), it’s 
> ok.
> When I drag it back, it turns black again.
> I had try the master branch, but the problem is still.
> Before I upgrade my macOS to 10.14.3, it’s ok.
> After I upgrade and build ffplay, it turns black.
> Appreciate any help!
> 
> Here’s the report:
> ffplay started on 2019-03-07 at 10:15:49
> Report written to "ffplay-20190307-101549.log"
> Command line:
> ffplay -report -i /Users/weilin/Downloads/bbb_1080p_h264.mp4
> ffplay version N-93298-g0ce759defd Copyright (c) 2003-2019 the FFmpeg 
> developers
>  built with Apple LLVM version 10.0.0 (clang-1000.10.44.4)
>  configuration: --prefix=/Users/weilin/third-libs/ffmpeg/darwin64-x86_64 
> --enable-gpl --enable-version3 --enable-pic --enable-libx264 --enable-libx265 
> --enable-libfreetype --disable-securetransport --enable-debug=lldb 
> --disable-optimizations --disable-stripping --extra-cflags= --extra-ldflags=
>  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
> Initialized opengl renderer.
> [NULL @ 0x7f85f380] Opening '/Users/weilin/Downloads/bbb_1080p_h264.mp4' 
> for reading
> [file @ 0x7f85ef5d3fc0] Setting default whitelist 'file,crypto'
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f85f380] Format mov,mp4,m4a,3gp,3g2,mj2 
> probed with size=2048 and score=100
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f85f380] ISO: File Type Major Brand: isom
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f85f380] Unknown dref type 0x206c7275 size 
> 12
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f85f380] Processing st: 0, edit list 0 - 
> media time: 0, duration: 737280
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f85f380] Before avformat_find_stream_info() 
> pos: 29904009 bytes read:40216 seeks:1 nb_streams:1
> [h264 @ 0x7f85f400] nal_unit_type: 7(SPS), nal_ref_idc: 1
> [h264 @ 0x7f85f400] nal_unit_type: 8(PPS), nal_ref_idc: 1
> [h264 @ 0x7f85f400] nal_unit_type: 6(SEI), nal_ref_idc: 0
> [h264 @ 0x7f85f400] nal_unit_type: 5(IDR), nal_ref_idc: 1
> [h264 @ 0x7f85f400] Format yuv420p chosen by get_format().
> [h264 @ 0x7f85f400] Reinit context to 1920x1088, pix_fmt: yuv420p
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f85f380] All info found
> [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f85f380] After avformat_find_stream_info() 
> pos: 527 bytes read:72984 seeks:2 frames:1
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 
> '/Users/weilin/Downloads/bbb_1080p_h264.mp4':
>  Metadata:
>major_brand : isom
>minor_version   : 512
>compatible_brands: isomiso2avc1mp41
>encoder : Lavf57.72.101
>  Duration: 00:01:00.00, start: 0.00, bitrate: 3987 kb/s
>Stream #0:0(und), 1, 1/12288: Video: h264 (High) (avc1 / 0x31637661), 
> yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 3986 kb/s, 24 fps, 24 tbr, 12288 tbn, 
> 24576 tbc (default)
>Metadata:
>  handler_name: VideoHandler
> detected 8 logical cores
> [h264 @ 0x7f85f0049a00] nal_unit_type: 7(SPS), nal_ref_idc: 1
> [h264 @ 0x7f85f0049a00] nal_unit_type: 8(PPS), nal_ref_idc: 1
> [h264 @ 0x7f85f0049a00] nal_unit_type: 6(SEI), nal_ref_idc: 0
> [h264 @ 0x7f85f0049a00] nal_unit_type: 5(IDR), nal_ref_idc: 1
> [h264 @ 0x7f85f0049a00] Format yuv420p chosen by get_format().
> [h264 @ 0x7f85f0049a00] Reinit context to 1920x1088, pix_fmt: yuv420p
> [h264 @ 0x7f85f005fc00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> [h264 @ 0x7f85f0079c00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
>nan M-V:nan fd=   0 aq=0KB vq=  498KB sq=0B f=0/0   
> [h264 @ 0x7f85f0853600] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> [h264 @ 0x7f85f0853c00] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> [h264 @ 0x7f85f101b800] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> [h264 @ 0x7f85f009] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> [h264 @ 0x7f85f009d200] nal_unit_type: 1(Coded slice of a non-IDR picture), 
> nal_ref_idc: 1
> Video frame changed from size:0x0 format:none serial:-1 to size:1920x1080 
> format:yuv420p serial:1
> [h264 @ 0x7f85f200] nal_unit_type: 1(Coded slice of a non-IDR picture

Re: [FFmpeg-user] ffmpeg whole directory file conversion question

[DopeLabs]

use the variable for the filename in the output filename

for f in *.mp4; do ffmpeg -i "$f" "$f_%03d.png"; done

cheers =]

DL



> On Feb 27, 2019, at 6:04 52PM, Ashley Smiley  
> wrote:
> 
> Hello,
> 
> I am writing to ask advice about how to augment my ffmpeg command line. I
> am working through my iMac's Terminal.
> 
> The task I am trying to perform is to convert a whole directory of .mp4
> files (all with unique filenames) to .png images whose labels correspond to
> their original .mp4 files.
> 
> The line I've run is:
> 
> *for i in *.mp4; *
> 
> *do ffmpeg -1 "$i" image%03d.png; *
> 
> *done*
> 
> This command line batch converts all .mp4 files to continuous .png files.
> That is, when moving from video 1 to video 2, the labeling does not start
> all over to indicate a break between video files. I would like to insert
> label breaks so that I can track which images correspond to which video and
> know exactly how many frames exist per video by conveniently having them
> labeled starting from 0 or 1 each time.
> 
> However, I don't know how to do this as I've searched online for a few days
> and I think it's time for suggestions.
> 
> An example list of filenames:
> 
> calib_checker_22sep18_154022_top_1x1cm.*mp4*
> calib_checker_22sep18_154022_side_1x1cm.*mp4*
> calib_ball_22sep18_154750_top*.mp4*
> calib_ball_22sep18_154930_side.*mp4*
> 
> and I want each to have a list of corresponding .pngs, i.e.:
> 
> calib_checker_22sep18_154022_top_1x1cm_*001.png*
> calib_checker_22sep18_154022_top_1x1cm_*002.png*
> calib_checker_22sep18_154022_top_1x1cm_*003.png*
> 
> calib_checker_22sep18_154022_side_1x1cm_*001.png*
> calib_checker_22sep18_154022_side_1x1cm_*002.png*
> 
> calib_ball_22sep18_154750_top_*001.png.*etc.
> 
> calib_ball_22sep18_154930_side_*001.png*etc.
> 
> One last note:
> 
> The reason I want to batch process is because there are at least 64 video
> files in the directory total.
> 
> Sincerely,
> 
> Ashley Smiley
> -- 
> Ashley Smiley
> Ph.D. Candidate, Dudley Lab
> Department of Integrative Biology
> University of California, Berkeley
> berkeleyflightlab.org
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Re: [FFmpeg-user] IP Camera stream copy to multiple outputs

[DopeLabs]

assuming h264/aac

ffmpeg -i ipcam -c copy -f flv rtmp://youtube -f mp4 localfile.mp4



> On Dec 18, 2018, at 3:38 01AM, Michael Shaffer  wrote:
> 
> I'm using ffmpeg to stream copy from an IP camera to youtube live. I would
> also like to save a copy of the video to my hard drive. The problem is the
> IP camera only supports one connection at a time. Is there a way to send
> ffmpeg to a program that would "multiply" the video so I could use as many
> copies as I want? I want to write 3 separate python programs to manage the
> different tasks I want to do with the video. One for streaming to youtube.
> One for archiving the footage. A third one for experimenting with OpenCV
> motion detection on the video.
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Re: [FFmpeg-user] filter_complex

[DopeLabs]

fyi...

- is a hyphen

' single quote
" double quote

cheers =]


> On Nov 15, 2018, at 1:20 27AM, Mettavihari D  wrote:
> 
> Greetings all,
> 
> The Problem is solved.
> It was only a problem of the correct hyphens around the object
> 
> '0:a’ should be '0:a'
> 
> ffmpeg -y -threads 16 -i video.mp4 -i logo.png -filter_complex
> "[1:v]scale=100:-1[logo];[0:v]scale=720:576[vid];[vid][logo]overlay=590:40[out]"
> -map '[out]' -map '0:a' -preset slower -profile:v main -level 3.1 -c:v
> libx264 -g 24 -bf 3 -b_strategy 2 -coder 1 -r 25000/1000 -b:v 1650k
> -maxrate:v 1650k -minrate:v 1650k -bufsize:v 330k -flags +ilme+ildct
> -x264opts bff=1 -color_primaries 1 -color_trc 1 -colorspace 1 -c:a mp2
> -b:a 128k -nal-hrd cbr -muxrate 1900k -f mpegts output-10.ts &>
> output-10.log &
> 
> 
> On 11/15/18, Mettavihari D  wrote:
>> Greetings Joel
>> 
>> 
>> -or alternately paste this into the terminal:
>> 
>> ffmpeg -y -threads 16 -i $FILE1.mp4 -i vidasa11.png -filter_complex  \
>> [1:v]scale=100:-1[logo]; "[0:v]scale=720:576[vid];"  \
>> [vid][logo]overlay=590:40[out] -map [out] -map 0:a -preset slower  \
>> -profile:v main -level 3.1 -c:v libx264 -g 24 -bf 3 -b_strategy 2  \
>> -coder 1 -r 25000/1000 -b:v 1650k -maxrate:v 1650k -minrate:v 1650k  \
>> -bufsize:v 330k -flags +ilme+ildct -x264opts bff=1 -color_primaries 1  \
>> -color_trc 1 -colorspace 1 -c:a mp2 -b:a 128k -nal-hrd cbr -muxrate  \
>> 1900k -f mpegts $FILE1.ts &> output.log &
>> 
>> ok?
>> 
>> My script is on one line BUT one never knows :-)
>> Just to make sure I copied your part into my script file and got the same
>> result
>> 
>> Input #1, png_pipe, from 'vidasa11.png':
>> Duration: N/A, bitrate: N/A
>> Stream #1:0: Video: png, rgba(pc), 360x270, 25 tbr, 25 tbn, 25 tbc
>> Filter scale has a unconnected output
>> 
>> It looks like it is at the "filter scale"
>> 
>> But I am not so sure about it.
>> I tried one of the simplest commands
>> 
>> ffmpeg –i video.mp4 -i logo.png -filter_complex "overlay=10:10"
>> top_left.mp4
>> 
>> and got this error.
>> 
>> Input #0, png_pipe, from 'logo.png':
>>  Duration: N/A, bitrate: N/A
>>Stream #0:0: Video: png, rgba(pc), 360x270, 25 tbr, 25 tbn, 25 tbc
>> Cannot find a matching stream for unlabeled input pad 1 on filter
>> Parsed_overlay_0
>> editor@Mettavihari:~/compress$
>> 
>> Thank you for your time.
>> 
>> with metta
>> Mettavihari
>> 
> 
> 
> -- 
> Streaming video from http://learntv.lk
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Re: [FFmpeg-user] Issues Live Streaming Audio Only

[DopeLabs]

have you considered using something other than ffserve (which no longer 
supported if im not mistaken) that has been specifically developed for audio 
transfer over a network?

such as icecast?

ffmpeg can output to an icecast server running on the local machine or running 
on the remote machine


source machine:
ffmpeg -i input -ac 2 -c:a libfdk_aac -ar 48k -b:a 64k -ice_public 0 -ice_genre 
Genre -ice_url http://url.tld -ice_description Description -ice_name Name 
-content_type audio/aacp -f adts icecast://source:p...@remote.host.com/mount

remote machine:
ffmpeg -i http://remote.host/mount


> On Oct 31, 2018, at 2:23 15PM, Brett Garrett 
>  wrote:
> 
> I've been trying to stream an audio feed from a live source using the RTSP
> protocol. The main goal is to eventually utilize jitter buffers or other
> techniques to reduce/remove stutters and skips. Although, at the moment I
> can't seem to get any audio to transfer over the network at all. I do have
> a somewhat unique setup, and therefore don't have many other options to
> work with.
> 
> I'm currently using Debian 9 with no gui for both the server and client(s).
> I would say the biggest hangup or issue is having to use Jack (which uses
> an ALSA device) for my output as it is used in several other applications
> that interact with the audio output. All of which are not running for the
> time being because I'm still just trying to get the stream from one
> computer to another across my local network. Here's what I'm hoping is
> enough information and that I'm not vomiting too much text.
> 
> A little info on our audio card specs...
> 
> root@test-9:~# *aplay -l*
>  List of PLAYBACK Hardware Devices 
> card 0: M44 [M Audio Delta 44], device 0: ICE1712 multi [ICE1712 multi]
>  Subdevices: 1/1
>  Subdevice #0: subdevice #0
> 
> root@test-9:~# *arecord --dump-hw-params -D hw:0,0*
> Recording WAVE 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono
> HW Params of device "hw:0,0":
> 
> ACCESS:  MMAP_INTERLEAVED RW_INTERLEAVED
> FORMAT:  S32_LE
> SUBFORMAT:  STD
> SAMPLE_BITS: 32
> FRAME_BITS: 384
> CHANNELS: 12
> RATE: [8000 96000]
> PERIOD_TIME: (20 341250]
> PERIOD_SIZE: [2 2730]
> PERIOD_BYTES: [96 131040]
> PERIODS: [1 1024]
> BUFFER_TIME: (20 682625]
> BUFFER_SIZE: [2 5461]
> BUFFER_BYTES: [96 262128]
> TICK_TIME: ALL
> 
> arecord: set_params:1299: Sample format non available
> Available formats:
> - S32_LE
> 
> Audio comes in through an analog XLR connection. Verified audio input is
> good by using the following commands...
> 
> root@test-9:~# /usr/bin/dbus-run-session ffmpeg -f jack -i ffmpeg -y
> output.wav
> root@test-9:~# jack_connect system:capture_1 ffmpeg:input_1 && jack_connect
> system:capture_2 ffmpeg:input_2
> 
> Waited a while then killed the tasks. Acquired the output.wav and it played
> with good audio. Although, I need to be able to get it from one computer to
> another through a live feed (not recording to some single file that will
> grow to crazy sizes).
> So here's my server config setup. Without the comments.
> 
> root@test-9:~# *grep ^[^#] /etc/ffserver.conf*
> HTTPPort 8585
> RTSPPort 15151
> HTTPBindAddress 0.0.0.0
> MaxHTTPConnections 2000
> MaxClients 1000
> MaxBandwidth 1000
> CustomLog -
> 
> File /tmp/feed1.ffm
> FileMaxSize 200K
> ACL allow 127.0.0.1
> 
> 
> Format rtp
> Feed feed1.ffm
> NoVideo
> AudioCodec aac
> AudioChannels 2
> AudioBitRate 64
> AudioSampleRate 48000
> AVOptionAudio flags +global_header
> 
> 
> Format status
> ACL allow localhost
> ACL allow 192.168.0.0 192.168.255.255
> 
> 
> URL http://www.ffmpeg.org/
> 
> 
> root@test-9:~# *ffserver -loglevel debug &*
> [1] 11631
> root@test-9:~# ffserver version 3.2.10-1~deb9u1 Copyright (c) 2000-2018 the
> FFmpeg developers
>  built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
>  configuration: --prefix=/usr --extra-version='1~deb9u1'
> --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu
> --incdir=/usr/include/i386-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-libebur128 --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
>  libavutil  55. 34.101 / 55. 34.101
>  libavcodec 57. 64.101 / 57. 64.101
>  libavfor

Re: [FFmpeg-user] How to take perfect screenshots of UHD/HEVC vídeos?

[DopeLabs]

what about using ffmpeg to grab a single frame from the video using -ss to 
specify the time index?

grab 1 frame at 1 minute into the video

ffmpeg -ss 00:01:00.000 -i input.m2ts -vframes 1 -f image2 screenshot.png

> On Oct 20, 2018, at 8:47 56AM, Kleber Duarte  wrote:
> 
> Hi, could someone give a tip to improve the quality of screenshots taken from 
> UHD HEVC files of seedbox running rtorrent/rutorrent installed on Ubuntu 
> Server Armhf 16.04 "Xenial Xerus" (32bits) OS? Many screenshots are shown 
> without any image definition. I don´t have problem with the screenshots of 
> FHD videos. I don´t know if is necessary any adjust in the lines codes of 
> plugins, or if exist a best way to take screenshots through the terminal SSH?
> 
> I verified that the problem occur with .m2ts UHD files. I tested the 
> screenshot plugin with .mkv UHD files and I didn´t have problems.
> 
> I am running the screenshot plugin from rutorrent gui. So, I don´t have any 
> command or log.
> 
> $ ffmpeg -version
> ffmpeg version 2.8.6-1ubuntu2 Copyright (c) 2000-2016 the FFmpeg developers
> built with gcc 5.3.1 (Ubuntu/Linaro 5.3.1-11ubuntu1) 20160311
> configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg 
> --toolchain=hardened --libdir=/usr/lib/arm-linux-gnueabihf 
> --incdir=/usr/include/arm-linux-gnueabihf --cc=cc --cxx=g++ --enable-gpl 
> --enable-shared --disable-stripping --disable-decoder=libopenjpeg 
> --disable-decoder=libschroedinger --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-libmodplug --enable-libmp3lame 
> --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp 
> --enable-libschroedinger --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-libzvbi 
> --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 
> --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 
> --enable-libopencv
> libavutil 54. 31.100 / 54. 31.100
> libavcodec 56. 60.100 / 56. 60.100
> libavformat 56. 40.101 / 56. 40.101
> libavdevice 56. 4.100 / 56. 4.100
> libavfilter 5. 40.101 / 5. 40.101
> libavresample 2. 1. 0 / 2. 1. 0
> libswscale 3. 1.101 / 3. 1.101​
> libswresample 1. 2.101 / 1. 2.101
> libpostproc 53. 3.100 / 53. 3.100
> 
> [cid:447a8209-bbdc-4ea6-af6f-03a8f590da73]
> [cid:46e34f24-9531-4703-9358-66d538380508]
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Re: [FFmpeg-user] FFmpeg getting stuck and prints Finishing stream 0:0 without any data written to it.

[DopeLabs]

do you have single or dual displays? your command is selecting /dev/video1 
which would be a 2nd display.. sources generally start at 0..

can you get a good capture using /dev/video0 ?



> On Oct 19, 2018, at 9:31 57PM, Abhisek Techie  
> wrote:
> 
> Hi,
> Any suggestions,on the below issue.
> 
> Regards
> 
> On Thu, 18 Oct 2018, 15:29 Abhisek Techie, 
> wrote:
> 
>> Hi,
>> 
>> I am using a logitech 310c camera in linux environment to capture video
>> and convert it to a .flv file , by using the below command
>> 
>> ffmpeg -f video4linux2 -s 432*240 -r 5 -pixel_format yuyv422 -i
>> /dev/video1 out3.flv -loglevel debug
>> 
>> But , it gets stuck after printing "*[video4linux2,v4l2 @ 0x5864e400]
>> Setting time per frame to 1/5"* and when I interrupt  with ctrl+c  it
>> prints "*Output file is empty, nothing was encoded". *
>> 
>> Below is the debug log , any suggestions on what's wrong with my approach?
>> 
>> ffmpeg version N-91993-g50d9c85358 Copyright (c) 2000-2018 the FFmpeg
>> developers
>>  built with gcc 6.3.0 (Debian 6.3.0-18+deb9u1) 20170516
>>  configuration: --disable-x86asm
>>  libavutil  56. 19.101 / 56. 19.101
>>  libavcodec 58. 30.100 / 58. 30.100
>>  libavformat58. 18.101 / 58. 18.101
>>  libavdevice58.  4.103 / 58.  4.103
>>  libavfilter 7. 32.100 /  7. 32.100
>>  libswscale  5.  2.100 /  5.  2.100
>>  libswresample   3.  2.100 /  3.  2.100
>> Splitting the commandline.
>> Reading option '-f' ... matched as option 'f' (force format) with argument
>> 'video4linux2'.
>> Reading option '-s' ... matched as option 's' (set frame size (WxH or
>> abbreviation)) with argument '432*240'.
>> Reading option '-r' ... matched as option 'r' (set frame rate (Hz value,
>> fraction or abbreviation)) with argument '5'.
>> Reading option '-pixel_format' ... matched as AVOption 'pixel_format' with
>> argument 'yuyv422'.
>> Reading option '-i' ... matched as input url with argument '/dev/video1'.
>> Reading option 'out3.flv' ... matched as output url.
>> Reading option '-loglevel' ... matched as option 'loglevel' (set logging
>> level) with argument 'debug'.
>> Finished splitting the commandline.
>> Parsing a group of options: global .
>> Applying option loglevel (set logging level) with argument debug.
>> Successfully parsed a group of options.
>> Parsing a group of options: input url /dev/video1.
>> Applying option f (force format) with argument video4linux2.
>> Applying option s (set frame size (WxH or abbreviation)) with argument
>> 432*240.
>> Applying option r (set frame rate (Hz value, fraction or abbreviation))
>> with argument 5.
>> Successfully parsed a group of options.
>> Opening an input file: /dev/video1.
>> [video4linux2,v4l2 @ 0x5864e400] fd:3 capabilities:8421
>> [video4linux2,v4l2 @ 0x5864e400] Current input_channel: 0, input_name:
>> Camera 1, input_std: 0
>> [video4linux2,v4l2 @ 0x5864e400] Setting time per frame to 1/5  * [Place
>> where it is getting stuck ]*
>> [video4linux2,v4l2 @ 0x5864e400] interrupted
>> Input #0, video4linux2,v4l2, from '/dev/video1':
>>  Duration: N/A, bitrate: 8294 kb/s
>>Stream #0:0, 0, 1/100: Video: rawvideo, 1 reference frame (YUY2 /
>> 0x32595559), yuyv422, 432x240 (0x0), 0/1, 8294 kb/s, 5 fps, 5 tbr, 1000k
>> tbn, 1000k tbc
>> Successfully opened the file.
>> Parsing a group of options: output url out3.flv.
>> Successfully parsed a group of options.
>> Opening an output file: out3.flv.
>> [file @ 0x58653a00] Setting default whitelist 'file,crypto'
>> Successfully opened the file.
>> Stream mapping:
>>  Stream #0:0 -> #0:0 (rawvideo (native) -> flv1 (flv))
>> Press [q] to stop, [?] for help
>> Finishing stream 0:0 without any data written to it.
>> detected 1 logical cores
>> [graph 0 input from stream 0:0 @ 0x58654dc0] Setting 'video_size' to value
>> '432x240'
>> [graph 0 input from stream 0:0 @ 0x58654dc0] Setting 'pix_fmt' to value '1'
>> [graph 0 input from stream 0:0 @ 0x58654dc0] Setting 'time_base' to value
>> '1/5'
>> [graph 0 input from stream 0:0 @ 0x58654dc0] Setting 'pixel_aspect' to
>> value '0/1'
>> [graph 0 input from stream 0:0 @ 0x58654dc0] Setting 'sws_param' to value
>> 'flags=2'
>> [graph 0 input from stream 0:0 @ 0x58654dc0] Setting 'frame_rate' to value
>> '5/1'
>> [graph 0 input from stream 0:0 @ 0x58654dc0] w:432 h:240 pixfmt:yuyv422
>> tb:1/5 fr:5/1 sar:0/1 sws_param:flags=2
>> [format @ 0x58655380] Setting 'pix_fmts' to value 'yuv420p'
>> [auto_scaler_0 @ 0x58655cc0] Setting 'flags' to value 'bicubic'
>> [auto_scaler_0 @ 0x58655cc0] w:iw h:ih flags:'bicubic' interl:0
>> [format @ 0x58655380] auto-inserting filter 'auto_scaler_0' between the
>> filter 'Parsed_null_0' and the filter 'format'
>> [AVFilterGraph @ 0x58654a00] query_formats: 4 queried, 2 merged, 1 already
>> done, 0 delayed
>> [auto_scaler_0 @ 0x58655cc0] w:432 h:240 fmt:yuyv422 sar:0/1 -> w:432
>> h:240 fmt:yuv420p sar:0/1 flags:0x4
>> [flv @ 0x58651480] intra_quant_bias = 0 inter_quant_bias = -64
>> Output #0, flv, to 'out3.flv':
>>  

Re: [FFmpeg-user] ffmpeg hang in background

[DopeLabs]

here is an example of one of the commands i have running without any issues...

ffmpeg -d -i http://127.0.0.1:5000 -r 60 -i viz.mp4 -loop 1 -i mask.png 
-filter_complex 
lowpass=600,showwaves=s=1280x720:mode=cline:r=60:colors=random|random[viz];[1:v]scale=1280x720,setsar=1/1[sun];[viz][sun]blend=all_mode=heat:shortest=1[bg];[bg][2:v]blend=all_mode=overlay:shortest=1:repeatlast=0:enable=lt(mod(t\,300)\,5),drawtext=fontfile=d3.ttf:fontsize=38:fontcolor=gray:text=DSFM.TV:x=5:y=(h-text_h)-5,drawtext=fontfile=helvbold.ttf:fix_bounds=true:fontsize=28:textfile=nowplaying.txt:fontcolor=gray:reload=1:y=(h-text_h)-5:x=(w-text_w)/2:enable=lt(mod(t\,60)\,15),drawtext=fontfile=helvbold.ttf:fontsize=24:fontcolor=gray:textfile=tuneid.txt:reload=1:y=5:x=(w-text_w)/2,drawtext=fontfile=helvbold.ttf:fontsize=24:fontcolor=gray:textfile=date.txt:reload=1:y=5:x=(w-text_w)-5:enable=lt(mod(t\,60)\,15),drawtext=fontfile=d3.ttf:fontsize=38:fontcolor=gray:textfile=count.txt:reload=1:x=(w-text_w)-5:y=(h-text_h)+20,hue=H=2*sin(400*PI/600*t)
 -s 1280x720 -c:a libfdk_aac -b:a 256k -c:v libx264 -preset ultrafast 
-profile:v high -level 4.2 -g 120 -bf 4 -x264opts 
keyint=120:min-keyint=120:8x8dct=1 -pix_fmt yuv420p -r 60 -f flv 
rtmp://127.0.0.1/live/viz

in short it grabs my net radio stream and outputs visualizations and adds text 
for the current playing mix, when the air date was, and the current number of 
people watching... i publish the stream at dsfm.tv

nothing that changes pts mid stream though.. that sounds like it wouldnt be 
good though...

> On Oct 16, 2018, at 1:04 42PM, Paul B Mahol  wrote:
> 
> On 10/16/18, DopeLabs  wrote:
>> never happens to me and i have many processes that use -r running in the
>> background 24/7
>> 
>> they all start with
>> 
>> ffmpeg -d -i
>> 
>> and end with
>> 
>> -f flv rtmp://host/app/stream   
> Do you use filtergraphs that change pts at all?
> 
> Try with loop filters and -r and you will get fancy effect.
> 
> 
>> 
>> 
>>> On Oct 16, 2018, at 3:16 16AM, sailor-godkane  wrote:
>>> 
>>> Could using -r with a filtergraph at the same time hang the command at
>>> random?
>>> 
>>> 
>>> 
>>> --
>>> Sent from: http://www.ffmpeg-archive.org/
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>> 
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Re: [FFmpeg-user] ffmpeg hang in background

[DopeLabs]

never happens to me and i have many processes that use -r running in the 
background 24/7

they all start with

ffmpeg -d -i

and end with 

-f flv rtmp://host/app/stream   On Oct 16, 2018, at 3:16 16AM, sailor-godkane  wrote:
> 
> Could using -r with a filtergraph at the same time hang the command at
> random?
> 
> 
> 
> --
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Re: [FFmpeg-user] Drawtext and Metadata

[DopeLabs]

have you tried

 %{metadata=print:key=duration}




> On Oct 16, 2018, at 7:18 04AM, Grant Griffith  
> wrote:
> 
> Hey everyone
> 
> I'm hoping to wanting to create a mosaic tile for video clips. I've got the
> majority of the setup dialed in, but the drawtext metadata is fighting me.
> 
> I'd like to have my mosaic tiles with a box containing filename, duration,
> encoder and optionally file size.
> 
> So far this is where I am...
> 
> ./ffmpeg -i "$1" \
> -vf "select=gt(scene\,0.087),\
> scale=192:-1,\
> tile=10x10,\
> drawtext="fontfile=/System/Library/Fonts/Helvetica.ttc":\
> text=`basename "${1}"` :\
> fontcolor=magenta:\
> shadowx=1.5:\
> shadowy=2:\
> fontsize=60\:\
> x=(w-text_w)/2:\
> y=(h-text_h)/2:\
> box=1:\
> boxcolor=lime@.5:\
> boxborderw=240:" \
> -frames:v 1 \
> -q:v 7 \
> ~/Desktop/test.jpg
> 
> I've tested all kinds of %{metadata:%duration} kind of variants, but I'm
> not having any luck. My last grasp would be to probe, store the data in
> json and then cobble together the key values into the text field, but it
> seems like running to binaries is pointless since ffmpeg already knows
> everything I'm really after.
> 
> Can this be done?
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Re: [FFmpeg-user] How to add 4 streams to one YouTube channel???

[DopeLabs]

an option you may wish to explore if you have the available network and 
encoding resources is youtube live streaming support for multiple cameras for a 
single livestream. each 'camera' has its own rtmp ingestion url and stream key 
and the viewer can choose which of the cameras they want to watch. multiple 
cameras show up as thumbnails above the main video.


https://support.google.com/youtube/answer/2853812?hl=en

cheers,
DL

> On Oct 16, 2018, at 4:27 58AM, Moritz Barsnick  wrote:
> 
> On Mon, Oct 15, 2018 at 23:04:39 -0500, Rad J wrote:
>> However I'm having some issues with the stream quality and frame loss...
>> also YouTube is complaining about my stream type...
> 
> You sometimes need to fiddle against YouTube's expectation. I remember
> that YouTube didn't see my live stream actually starting, without
> giving me detailed errors like yours. When I improved some parameters
> (such as GOP size), it suddenly worked.
> 
>> This is what I'm getting:
>> 
>> *Bad video settings*
>> *The video is encoded with an unsupported codec. Please set the video codec
>> for the stream to a supported codec (H.264).*
>> 
>> *10:54 PM Bad Bad video settings*
>> *The video is encoded with an unsupported codec. Please set the video codec
>> for the stream to a supported codec (H.264).*
>> *Please check the video resolution. The current resolution is (3840x1080),
>> which is not optimal.*
>> *The stream's current bitrate (992.00 Kbps) is lower than the recommended
>> bitrate. We recommend that you use a stream bitrate of 4500 Kbps.*
>> 
>> Do you know how to fix this? How to increase the bitrate?
> 
> Sure we know. But please show your currently used command line(s) which
> provoke(s) these warnings. We can try to fine-tune or reproduce.
> 
> Cheers,
> Moritz
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Re: [FFmpeg-user] Live Streaming in browser just downloads a file no streaming happening

[DopeLabs]

most browsers dont have native flash playback support and usually require the 
flash player plugin.. but.. flash support is basically deprecated in most major 
browsers.

you didnt say what browser your using, but if its chrome you may find this a 
good read...

https://bugs.chromium.org/p/chromium/issues/detail?id=767342

and if you add the site address manually in 'allow' list of 'flash' under 
'content settings' - Chrome>settings>advance>content settings>flash>allow you 
may have some luck.

else you can try a simple embed






 



most of this info was found scrolling through: 
https://productforums.google.com/forum/#!topic/chrome/zVTCfJsssAQ

or you can stream in a format that is natively supported by browsers (html5).

cheers,


> On Oct 15, 2018, at 9:35 42AM, Abhisek Techie  
> wrote:
> 
> Hi,
> 
> I am trying to stream live video from camera in the browser but it
> downloads  file named test.swf and I don't see any video streaming in
> browser
> 
> Below are the steps  that I have performed
> 
> 1.ffserver.conf that I am using
> 
> HTTPPort 8090
> HTTPBindAddress 0.0.0.0
> MaxHTTPConnections 2000
> MaxClients 1000
> MaxBandwidth 15000
> 
> File /tmp/feed1.ffm
> FileMaxSize 200K
> ACL allow 127.0.0.1
> 
> 
> # Flash
> 
> Feed feed1.ffm
> VideoCodec flv
> Format swf
> VideoFrameRate 30
> VideoIntraOnly
> NoAudio
> 
> 
> 
> Format status
> # Only allow local people to get the status
> ACL allow localhost
> ACL allow 192.168.0.0 192.168.255.255
> 
> 
> 
> URL http://www.ffmpeg.org/
> 
> 
> 
> 2.I started the ffserver
> 
> ffserver -d -f /etc/ffserver.conf
> 
> 
> 3.Linked camera with feed
> ffmpeg -r 30 -s 320*240 -hide_banner -f video4linux2 -loglevel debug  -i
> /dev/video0 -f flv http://localhost:8090/feed1.ffm
> 
> 4.Opened browser and typed  http://localhost:8090/test.swf to watch  the
> feed, I found test.swf file getting downloaded and don't see streaming.
> 
> Please suggest if I am missing or doing something wrong
> 
> Regards
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Re: [FFmpeg-user] Lossless cut from a h264 video with audio

[DopeLabs]

try this one...


/usr/bin/env ffmpeg -y -ss 0:03:18  -i cook.mp4 -t 0:00:10 -c copy 
cook-start.mp4 &> cook-start.txt



> On Oct 10, 2018, at 8:44 56AM, Samat Galimov  wrote:
> 
> Hello,
> 
> tl;dr: Q: is there a way to cut a 10 second clip from a h264 video 
> (downloaded from youtube) to later use it in web browsers with no quality 
> loss?
> 
> What I have tried so far: http://alpha.rawg.io/question/
> 
> Long version:
> 
> My inputs are http://alpha.rawg.io/question/cook.txt 
> 
> 
> My first attempt is to opt-in for copy codecs using the following command:
> 
>   /usr/bin/env ffmpeg -y -ss 0:03:18 -t 0:00:10 -i cook.mp4 -c copy 
> cook-start.mp4 &> cook-start.txt
> 
> What I've got is:
> - Safari: audio/video sync is OK, beginning is OK, but additional 3 seconds 
> of audio without video in the end
> - Chrome: audio/video is out of sync
> 
> Then I tried to specify -ss as an output option:
> 
>   /usr/bin/env ffmpeg -y -i cook.mp4 -ss 0:03:18 -t 0:00:10 -c copy 
> cook-end.mp4 &> cook-end.txt
> 
> Safari & Chrome: first 3 seconds have no video, but audio only.
> 
> 
> Is there a clever trick to overcome this apparent keyframe location issue 
> while not re-encoding videos? 
> 
> If there is no way to avoid re-encoding, what options should I use to 
> minimise quality loss?
> 
> Here is the webpage to check out my results & source files: 
> https://alpha.rawg.io/question/ 
> 
> Thank you very much in advance.
> 
> Samat.
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Re: [FFmpeg-user] ffmpeg hang in background

[DopeLabs]

its not very well documented but ffmpeg has a daemon mode

ffmpeg -d   On Oct 10, 2018, at 6:56 46AM, Paul B Mahol  wrote:
> 
> On 10/10/18, sailor-godkane  wrote:
>> Hello,
>> 
>> I am using ffmpeg in a big project for video/audio automated processing.
>> Each project for processing requires lots of ffmpeg commands. Those commands
>> are run by a daemon written in ObjC using GnuStep.
>> Since it runs on the background i used -nostdin and -loglevel quiet for
>> production use.
>> To be sure i read everything on stdout and stderr to /dev/null but it should
>> not matter since the command does not output anything.
>> Everything run fine except sometimes a ffmpeg command will hang and i cannot
>> understand why.
>> Sometimes i can see a few % of cpu used by the commands but after hours it
>> was still there.
>> When it happens i have to shutdown my daemon and restart the processing. The
>> same command will then run fine and the process can be finished.
>> 
>> Details of ffmpeg (compiled from source)
>> 
>> ffmpeg version n4.0.2 Copyright (c) 2000-2018 the FFmpeg developers
>>  built with gcc 4.8 (SUSE Linux)
>>  configuration: --disable-programs --enable-ffprobe --enable-ffmpeg
>> --enable-libx264 --enable-gpl --enable-shared --enable-pthreads
>> --enable-libmp3lame --enable-pic
>>  libavutil  56. 14.100 / 56. 14.100
>>  libavcodec 58. 18.100 / 58. 18.100
>>  libavformat58. 12.100 / 58. 12.100
>>  libavdevice58.  3.100 / 58.  3.100
>>  libavfilter 7. 16.100 /  7. 16.100
>>  libswscale  5.  1.100 /  5.  1.100
>>  libswresample   3.  1.100 /  3.  1.100
>>  libpostproc55.  1.100 / 55.  1.100
>> 
>> System is :
>> 
>> openSUSE 13.1 - 3.11.10-29-desktop
>> 
>> The machine is a virtual one running on VMWare.
>> 
>> The command that most often hangs is
>> 
>> ffmpeg -nostdin -loglevel quiet -y -i front_slide_video.mov -an -r 16.00
>> -c:v libx264 -preset slow -b:v 5000k -aspect 1.33 -vf
>> scale=640:480,pad=640:480:0:0:black front_slide_video-rescaled.mov
> 
> What if you do not use -r parameter but fps filter explicitly in filtergraph?
> 
>> 
>> But i had other commands hanging randomly as well so this is just a sample.
>> 
>> Since this is completely random i have no way of testing it. The command
>> that hang works fine if i run it manually and it runs fine most of the time
>> when the daemon runs it.
>> I upgraded from a version 2.X that was running on the previous major version
>> of my software. It's been a few years since i worked on it but i don't
>> recall hangs back in the day with version 2.
>> 
>> I had the same problem with 4.0.1 btw.
>> 
>> Thanks for your help
>> 
>> 
>> 
>> 
>> --
>> Sent from: http://www.ffmpeg-archive.org/
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Re: [FFmpeg-user] Differences between ffmpeg and ffplay

[DopeLabs]

im not sure if windows has the ability to 'pipe' like a *nix system does.. but 
this works to pipe the output of ffmpeg to ffplay instead of a file.. 

hope this helps...

ffmpeg -i test_15kHz.mp3 -filter_complex 
"volume=1,highpass=f=12000,highpass=f=12000,highpass=f=12000,highpass=f=12000[a0];aevalsrc="sin\(12000*2*PI*t\)":c=stereo:s=44100[a1];[a0][a1]amultiply,lowpass=f=1,lowpass=f=1,lowpass=f=1,lowpass=f=1"
 -y -f wav - | ffplay -f wav -

cheers,

DL

> On Oct 3, 2018, at 2:42 27AM, Michael Koch  
> wrote:
> 
> 
>>> oh no, I don't want to learn a new command line syntax only to solve
>>> this simple problem.
>>> I have a long ffmpeg command line that's working fine when writing the
>>> output to a file. Is there really no easy way to send the output to the
>>> speakers instead?
>> It is very easy syntax to learn, you can send output directly to
>> speakers with ffmpeg
>> too but that is extremly limited funcionality.
> 
> How can I send the ffmpeg output to the speakers (on a Windows system)? I 
> would like to test it.
> 
> Michael
> 
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Re: [FFmpeg-user] Why are half the frames "duplicate" ?

[DopeLabs]

input: 90k tbn, 59.94 tbc

output: 30k tbn, 29.97 tbc

try explicitly specifying the output frame rate

-c:v h264 -r 29.97


> On Sep 26, 2018, at 7:27 26PM, sean darcy  wrote:
> 
> I'm transcoding mpeg2 -> x264.
> 
> Stream #0:0[0x64]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002), 
> yuv420p(tv, progressive), 720x480 [SAR 8:9 DAR 4:3], Closed Captions, 29.97 
> fps, 59.94 tbr, 90k tbn, 59.94 tbc
> 
> When I run ffmpeg:
> 
> ffmpeg -i in.mpg -map 0:0 -map 0:2 -c:v libx264 -tune film -preset slow -crf 
> 22 -c:a copy out.mp4
> 
> almost half the frames are dups:
> 
> frame=394550 fps=135 q=-1.0 Lsize= 2218652kB time=03:39:24.71 
> bitrate=1380.6kbits/s dup=175142 drop=0 speed=4.51x
> 
> Does this really mean that there were 175,142 frames with exact copies ? So 
> only ( 394,550 less 175,142 ) unique frames ?
> 
> The output looks fine. The input looks fine. What are these duplicated 
> frames? Is it an interlacing issue ? But the input is progressive.
> 
> Does it have to do with this warning:
> 
> [mp4 @ 0x921c80] track 1: codec frame size is not set
> Output #0, mp4, to 'out.mp4':
>  Metadata:
>encoder : Lavf58.18.103
>Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 720x480 
> [SAR 8:9 DAR 4:3], q=-1--1, 29.97 fps, 30k tbn, 29.97 tbc
>Metadata:
>  encoder : Lavc58.31.100 libx264
> 
> Puzzled.
> 
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Re: [FFmpeg-user] FFMPEG Symbol Lookup Error

[DopeLabs]

try running

$ sudo ldconfig

open a fresh shell session and try ffmpeg again.



> On Sep 18, 2018, at 6:44 56AM, Abhijit Nathwani  
> wrote:
> 
> Hi,
> 
> I installed FFMPEG v3.4.4 via apt-get on Ubuntu 16.04 and on running it,
> I'm facing the symbol lookup error while running it.
> 
> abhijit@AHMCPU2092:~$ ffmpeg
> ffmpeg: symbol lookup error: /usr/lib/x86_64-linux-gnu/libavcodec.so.57:
> undefined symbol: opus_multistream_surround_encoder_create
> 
> I have installed all the opus* and libopus* globs packages available. Still
> I cannot understand the cause and solution to the problem.
> 
> Regards,
> Abhijit Nathwani
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Re: [FFmpeg-user] h.265 streaming to youtube

[DopeLabs]

it would also help if youtube actually supported hevc ingestion...

https://support.google.com/youtube/answer/2853702?hl=en



> On Sep 8, 2018, at 7:09 46AM, Carl Eugen Hoyos  wrote:
> 
> 2018-09-08 12:28 GMT+02:00, Michael Shaffer :
> 
>> c:\ffmpeg\bin\ffmpeg1.exe -re -rtsp_transport tcp -i "rtsp://
>> admin:password@192.168.1.64:554" -f lavfi -f dshow -rtbufsize 2000M
>> -thread_queue_size 5096 -i audio="virtual-audio-capturer" -c:a libmp3lame
>> -ab 128k -ar 44100 -c:v copy -threads 0 -bufsize 512k -f hevc "rtmp://
>> a.rtmp.youtube.com/live2/bmbp-3c10-f2b3-3hgj"
> 
> (Complete, uncut console output missing, "-f hevc" looks wrong,
> "threads 0" seems useless.)
> 
> hevc is not defined for flv, this is therefore not (yet) possible.
> 
> Carl Eugen
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Re: [FFmpeg-user] Overwriting metadata with other existing metadata

[DopeLabs]

players use metadata properly by using artist for artist..

i would not suggest overwriting the artist field because the value from album 
artist and artist may be different and you could end up with incorrectly tagged 
files.

to specify a compilation you would use compilation=1.. so maybe if the 
compilation tag has a value of 1, the player then uses a different field to 
group the album.

but it depends on the logic of the individual player how it handles grouping of 
compilations..

not all players have support for the extra fields that itunes uses to visually 
group tracks together. i use itunes and its nice to have these abilities, but 
you cant expect the same abilities in other players.

i can say with confidence that players will support the basic tags like artist, 
title, and album.

instead what i suggest is to simply copy over any and all metadata from the 
input file with -map_metadata 0

after that, its up to you to use a player that knows how to take advantage of 
all the metadata...


> On Sep 6, 2018, at 9:53 20AM, Zoe Blade  wrote:
> 
> Hi!
> 
> So I've got a lot of FLAC files, meticulously maintained using MusicBrainz's 
> Picard.  And from those, I derive a lot of messier, not at all meticulously 
> maintained MP3 files.  I do this using FFMPEG rather than LAME because it 
> preserves the metadata.  FFMPEG is great for this conversion.  Specifically, 
> I wrote some scripts that at their core use this command:
> 
> ffmpeg -i "$song" -codec:a libmp3lame -q:a 4 -c:v copy "${song%.flac}.mp3" 2> 
> /dev/null
> 
> So far, so good.  The issue is that MP3 players, be they by Apple or Sony or 
> anyone in between, tend to not use the metadata properly.  For example, they 
> pay attention to the "artist" tag when they should instead look at the 
> "album_artist" tag, to group compilations together properly.  Or, better yet, 
> "albumartistsort", which also fixes alphabetisation when it comes to artists 
> whose name starts with "The" and so on.
> 
> I can add to the above line something like this:
> 
> -metadata artist='Test'
> 
> This overwrites the artist tag with custom data, great.  I specifically want 
> to overwrite it with whatever's in the source file's albumartistsort tag.
> 
> Is there a way to do this, so I can overwrite one key with the value of 
> another?
> 
> This inelegant Bash solution seems to work:
> 
> artist=`exiftool "$song" | grep --color=never Albumartistsort | sed 
> 's/^Albumartistsort *: //'`; ffmpeg -i "$song" -metadata artist="$artist" 
> -codec:a libmp3lame -q:a 4 -c:v copy "${song%.flac}.mp3"
> 
> But if there's a simpler way to do it in ffmpeg, I'd love to know!
> 
> Thanks,
> Zoë.
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Re: [FFmpeg-user] How to extract audio and covert audio speech to text using FFFMPEG?

[DopeLabs]

i dont think there is any speech recognition within ffmpeg...

you might want to try

https://cloud.google.com/speech-to-text/



> On Sep 1, 2018, at 9:29 41PM, Dani A  wrote:
> 
> Guys,
> I am working on a project that requires converting audio speed in a video to 
> a text to be used as a script or closed captions.
> 
> Can FFMPEG do that? If not, what do you recommend?
> 
> Dani
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Re: [FFmpeg-user] How To Detect When FFMPEG Stops

[DopeLabs]

if the live stream is truly finished and the stream is no longer available or 
the end is reached, ffmpeg should exit normally and stop running...


> On Aug 27, 2018, at 7:51 28AM, Ray Jender  wrote:
> 
> I am using ffmpeg to capture live streaming segments (.ts) and create an
> MP4.
> 
> 
> 
> How can I tell when the incoming stream is done and ffmpeg is idle?
> 
> 
> 
> Thanks,
> 
> 
> 
> Ray
> 
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Re: [FFmpeg-user] high cpu uses

[DopeLabs]

On Aug 9, 2018, at 4:16 58AM, Remo Rayamajhi  wrote:
> 
> Hello ,
> 
> I m using 8 core cpu and  using ffmpeg to transcode udp input to rtmp out
> to my Wowza server ...

except what i see in the command you provided is rtp input to udp out...

also last time i checked wowza should be able to ingest both udp and rtp 
streams and transcode them

> but only *2 ffmpeg script* is easily run . Is there
> any idea so i can transcode* 5-6 stream *for that 8 core server . Os
> install is centos .
> 
> *ffmpeg -i 'rtp://@239.1.2.10:8002 ' -r "25"
> -keyint_min 25 -refs 3 -level 3.1 -coder 1 -trellis 1 -b:v 800k -vf
> "yadif,hqdn3d=1.5:1.5:6:6,scale=720:576" -aspect 16:9 -c:a aac -ar 44100
> -ac 2 -b:a 128k -c:v libx264 -g 50 -f mpegts udp://@192.168.2.200:1243
>  /dev/null 2>/var/log/ffmpeg9.log &*

have you tried (usually at the expense of higher bitrate): -preset ultrafast


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Re: [FFmpeg-user] Is FFmpeg Smart Enough?

[DopeLabs]

what do you mean by 'stream' ?

are you referring to an audio or a video stream within a single file?

or are you referring to consuming some type of streamed media, such as rtmp or 
hls or dash?


but generally, yes.. ffmpeg can detect when a stream ends, and usually 
(depending on options), exits normally. 


> On Aug 5, 2018, at 7:27 15PM, Ray Jender  wrote:
> 
> Does FFmpeg recognize when a stream has ended?
> 
> 
> 
> Thanks,
> 
> 
> Ray
> 
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Re: [FFmpeg-user] Drawtext and crossfade effect

[DopeLabs]

i copied a short video and inverted the color in second.mp4 by using the negate 
filter so the cross fade is from regular color to inverted color... then i just 
added the drawtext as the next item in the filter_complex 


you can see the output here: http://www.dopelabs.com/video/crosstext.mp4


$ ffmpeg -i first.mp4 -i second.mp4 -filter_complex 
'[0:v]trim=start=0:end=2,setpts=PTS-STARTPTS[0_clip_1];[0:v]trim=start=2:end=3,setpts=PTS-STARTPTS[fadeoutsrc_0];[fadeoutsrc_0]format=pix_fmts=yuva420p,fade=t=out:st=0:d=1:alpha=1[fadeout_0];[fadeout_0]fifo[fadeoutfifo_0];[1:v]trim=start=1,setpts=PTS-STARTPTS[1_clip_2];[1:v]trim=start=0:end=1,setpts=PTS-STARTPTS[fadeinsrc_1];[fadeinsrc_1]format=pix_fmts=yuva420p,fade=t=in:st=0:d=1:alpha=1[fadein_1];[fadein_1]fifo[fadeinfifo_1];[fadeoutfifo_0][fadeinfifo_1]overlay[crossfade_1];[0_clip_1][crossfade_1][1_clip_2]concat=n=3[output];[output]drawtext=text="lets
 be 
friends":fontcolor=white:fontsize=40:box=1:boxcolor=red@0.5:boxborderw=10:x=5:y=5[out]'
 -map "[out]" crosstext.mp4
ffmpeg version N-91398-gd08d4a8c73 Copyright (c) 2000-2018 the FFmpeg developers
  built with Apple LLVM version 9.1.0 (clang-902.0.39.2)
  configuration: --enable-libmp3lame --enable-opencl --enable-audiotoolbox 
--enable-videotoolbox --enable-libx265 --enable-libx264 --enable-libxvid 
--enable-libfdk-aac --enable-lzma --enable-gnutls --enable-fontconfig 
--enable-libfreetype --enable-libfribidi --enable-nonfree --enable-gpl 
--enable-libass --enable-swscale --enable-avfilter --enable-libvorbis 
--enable-libopus --enable-librsvg --enable-libtheora --enable-libopenjpeg 
--enable-libmodplug --enable-libvpx --enable-avresample --enable-libsoxr 
--enable-libspeex --enable-libcaca --enable-opengl --enable-libtesseract
  libavutil  56. 18.102 / 56. 18.102
  libavcodec 58. 20.104 / 58. 20.104
  libavformat58. 17.101 / 58. 17.101
  libavdevice58.  4.101 / 58.  4.101
  libavfilter 7. 25.100 /  7. 25.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale  5.  2.100 /  5.  2.100
  libswresample   3.  2.100 /  3.  2.100
  libpostproc55.  2.100 / 55.  2.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'first.mp4':
  Metadata:
major_brand : mp42
minor_version   : 0
compatible_brands: mp42mp41isom
creation_time   : 2017-07-30T18:31:27.00Z
  Duration: 00:00:29.50, start: 0.00, bitrate: 233 kb/s
Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), 
yuv420p, 400x392, 233 kb/s, 10.03 fps, 10 tbr, 1k tbn, 2k tbc (default)
Metadata:
  creation_time   : 2017-07-30T18:31:27.00Z
  handler_name: Twitter v1.0-757770b7c8e9d79a526cdff77e74666386274fdf
  encoder : AVC Coding
Input #1, mov,mp4,m4a,3gp,3g2,mj2, from 'second.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf58.17.101
  Duration: 00:00:29.60, start: 0.00, bitrate: 335 kb/s
Stream #1:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 400x392, 
334 kb/s, 10 fps, 10 tbr, 10240 tbn, 20 tbc (default)
Metadata:
  handler_name: Twitter v1.0-757770b7c8e9d79a526cdff77e74666386274fdf
[Parsed_drawtext_16 @ 0x7fc17b50e240] Using "/Library/Fonts/Verdana.ttf"
Stream mapping:
  Stream #0:0 (h264) -> trim
  Stream #0:0 (h264) -> trim
  Stream #1:0 (h264) -> trim
  Stream #1:0 (h264) -> trim
  drawtext -> Stream #0:0 (libx264)
Press [q] to stop, [?] for help
[Parsed_drawtext_16 @ 0x7fc17db00a80] Using "/Library/Fonts/Verdana.ttf"
[libx264 @ 0x7fc17d002200] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 
AVX FMA3 BMI2 AVX2
[libx264 @ 0x7fc17d002200] profile High, level 2.1
[libx264 @ 0x7fc17d002200] 264 - core 152 r2854 e9a5903 - H.264/MPEG-4 AVC 
codec - Copyleft 2003-2017 - http://www.videolan.org/x264.html - options: 
cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 
psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 
deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=12 lookahead_threads=2 
sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 
constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 
open_gop=0 weightp=2 keyint=250 keyint_min=10 scenecut=40 intra_refresh=0 
rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 
ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'crosstext.mp4':
  Metadata:
major_brand : mp42
minor_version   : 0
compatible_brands: mp42mp41isom
encoder : Lavf58.17.101
Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv420p, 400x392, 
q=-1--1, 10 fps, 10240 tbn, 10 tbc (default)
Metadata:
  encoder : Lavc58.20.104 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
frame=  316 fps=286 q=-1.0 Lsize=1257kB time=00:00:31.30 bitrate= 
328.9kbits/s speed=28.3x
video:1252kB audio:0kB subtitle:0kB other streams:0kB global headers:0k

Re: [FFmpeg-user] Copying time stamp metadata

[DopeLabs]

-c copy in command a), but not b)?

does this work?

c) fmpeg -i input.mkv -c copy -copyts -movflags +faststart output.mkv 

> On Jul 24, 2018, at 8:06 42PM, Abhinav Kashyap  wrote:
> 
> Replying back to myself.
> 
> I am still not sure if copy is the correct approach or map ? Any
> suggestions ?
> 
> On Mon, Jul 23, 2018 at 10:56 AM, Abhinav Kashyap  wrote:
> 
>> Hello,
>> 
>> I am trying to find out how can I copy timestamp related metadata from a
>> file to another ?
>> 
>> My input file has UNIX timestamp (start time)  which is what i need to
>> copy. Here is what I am doing or have tried:
>> 
>> 1) Read a video file frame by frame in OpenCV
>> 2) Modify frames (image processing such as Gaussian blurring or sharpening)
>> 3) Write back frames in a video using openCV. Written file has a start
>> time of 0.00 ms.
>> 4) Copy metadata from original file to modified file using below command :
>>   a) ffmpeg -i input.mkv -c copy -movflags +faststart output.mkv - This
>> copies bunch of metadata but not the start time which is what I am after.
>>   b) fmpeg -i input.mkv -copyts -movflags +faststart output.mkv - This
>> copies the video content as with all metadata including the start time.
>> 
>> 
>> Any ideas how can I copy just the start time from originial file to
>> modified file ?
>> 
>> Thanks,
>> Abhi
>> 
>> 
>> 
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Re: [FFmpeg-user] How to send sequence of PNG files over network for ffmpeg to render?

[DopeLabs]

heres a 1 liner for you..


scp /full/path/to/images/*.png ubuntu@ip-10-0-0-196:/full/path/to/images/ && 
ssh ubuntu@ip-10-0-0-196 'ffmpeg -pattern_type glob -r 60 -i 
/full/path/to/images/*.png -c:v libvpx-vp9 -pix_fmt yuva420p ffmpeg.webm'



> On Jul 23, 2018, at 11:12 12PM, Andrew Stuart 
>  wrote:
> 
>>> My goal is for ffmpeg to be running and listening on a network port, then 
>>> from another machine to send the PNG sequence over the network directly 
>>> into ffmpeg, and have the resulting video file saved.
> 
>>> So I need to work out how to configure ffmpeg to listen on the network for 
>>> a sequence of PNG files, and I need to work out how to, from another 
>>> machine, connect to ffmpeg, tell it what the output filename should be, and 
>>> then transmit the PNG files.
> 
>>> Is there any way to do this?  Despite many hours research I can’t see if 
>>> this is actually possible or not.
> 
> I’ll answer my own question as I worked it out eventually - maybe it will 
> help someone else in the future.
> 
> The following command sets up ffmpeg to listen on a socket on port 8000, and 
> when PNG images arrive, ffmpeg processes them.
> 
> ffmpeg -loglevel debug -framerate 60 -f image2pipe -i 
> tcp://127.0.0.1:8000?listen=1 -c:v libvpx-vp9 -pix_fmt yuva420p ffmpeg.webm
> 
> You can test it with this Python script which reads all the png files from a 
> directory and sends them to the socket, resulting in ffmpeg running its 
> render and exiting.
> 
> import glob
> path = '/home/ubuntu/tmp/*.png'
> files=glob.glob(path)
> import socket
> 
> TCP_IP = '127.0.0.1'
> TCP_PORT =8000
> BUFFER_SIZE = 1024
> 
> s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
> s.connect((TCP_IP, TCP_PORT))
> print('connection closed')
> for file in files:
>f = open(file, 'rb')
>s.send(f.read())
>f.close()
> s.close()
> 
> 
> 
> With the resulting output:
> 
> ubuntu@ip-10-0-0-196:~$ ffmpeg -y -framerate 60 -f image2pipe -i 
> tcp://127.0.0.1:8000?listen=1 -c:v libvpx-vp9 -pix_fmt yuva420p ffmpeg.webm
> ffmpeg version 2.8.14-0ubuntu0.16.04.1 Copyright (c) 2000-2018 the FFmpeg 
> developers
>  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609
>  configuration: --prefix=/usr --extra-version=0ubuntu0.16.04.1 
> --build-suffix=-ffmpeg --toolchain=hardened 
> --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu 
> --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping 
> --disable-decoder=libopenjpeg --disable-decoder=libschroedinger 
> --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-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus 
> --enable-libpulse --enable-librtmp --enable-libschroedinger --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-libzvbi --enable-openal --enable-opengl --enable-x11grab 
> --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r 
> --enable-libx264 --enable-libopencv
>  libavutil  54. 31.100 / 54. 31.100
>  libavcodec 56. 60.100 / 56. 60.100
>  libavformat56. 40.101 / 56. 40.101
>  libavdevice56.  4.100 / 56.  4.100
>  libavfilter 5. 40.101 /  5. 40.101
>  libavresample   2.  1.  0 /  2.  1.  0
>  libswscale  3.  1.101 /  3.  1.101
>  libswresample   1.  2.101 /  1.  2.101
>  libpostproc53.  3.100 / 53.  3.100
> Input #0, image2pipe, from 'tcp://127.0.0.1:8000?listen=1':
>  Duration: N/A, bitrate: N/A
>Stream #0:0: Video: png, rgba(pc), 800x600, 60 fps, 60 tbr, 60 tbn, 60 tbc
> Incompatible pixel format 'yuva420p' for codec 'libvpx-vp9', auto-selecting 
> format 'yuv420p'
> [libvpx-vp9 @ 0x2428e00] v1.5.0
> Output #0, webm, to 'ffmpeg.webm':
>  Metadata:
>encoder : Lavf56.40.101
>Stream #0:0: Video: vp9 (libvpx-vp9), yuv420p, 800x600, q=-1--1, 200 kb/s, 
> 60 fps, 1k tbn, 60 tbc
>Metadata:
>  encoder : Lavc56.60.100 libvpx-vp9
> Stream mapping:
>  Stream #0:0 -> #0:0 (png (native) -> vp9 (libvpx-vp9))
> Press [q] to stop, [?] for help
> [png @ 0x237e840] Invalid PNG signature 0xBA775E7E29DE8950.bitrate= 
> 247.0kbits/s
> Error while decoding stream #0:0: Invalid data found when processing input
> frame=   98 fps= 18 q=0.0 Lsize=  42kB time=00:00:01.64 bitrate= 
> 208.8kbits/s
> video:41kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing 
> overhead: 2.742833%
> ubuntu@ip-10-0-0-196:~$
> 
> 
> 
> 
> 
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Re: [FFmpeg-user] Using a AMD Radeon RX5xx with ffmpeg

[DopeLabs]

i have found that generally CPU encoding is more efficient and can produce 
better quality results, while GPU may be faster (depending on hw), but produces 
a lower quality result.

for many, it seems people the main goals is to be able to offload the 
encoding/decoding to the GPU, freeing up CPU for the user, or other processes.

you might also want to give -preset ultrafast a shot.



> On Jul 25, 2018, at 12:18 57AM, Lukas Obermann  
> wrote:
> 
> Thanks for that hint! I want to give it a try but where can I get OpenMAX IL 
> from? 
> I can not find a Ubuntu package, nor sources where I could compile it from. 
> Most stuff returned is from xda developers on android.
> 
>> On 25.07.2018, at 06:16, Dennis Mungai  wrote:
>> 
>> From what I can gather, AMD's driver implementation for VAAPI (gallium?
>> through mesa) is a work in progress, and compared to i915 (intel's), is
>> quite behind.
>> 
>> On your system, are you able to build FFmpeg to utilize OMX IL? AMD has
>> support for it via the VCE block. See this for an example on enabling it:
>> https://github.com/legotheboss/YouTube-files/wiki/(RPi)-Compile-FFmpeg-with-the-OpenMAX-H.264-GPU-acceleration
>> 
>> The guide was written for the rPI, but what we're interested in is OpenMAX
>> bellagio and the configuration switches that enable OpenMAX IL encoders.
>> 
>> 
>> 
>> On 24 July 2018 at 12:01, Lukas Obermann  wrote:
>> 
>>> Hello Dennis,
>>> 
>>> thank you for your help! Much appreciate it.
>>> 
>>> Using your command I get a 1.9x speed. So a slight improvement, but not
>>> much.
>>> I pasted the debug output here, maybe you can see something usefull?
>>> https://pastebin.com/W0KKjZbN 
>>> 
>>> ad 1. Yes, there is the onboard intel device and 6 of those RX570 that in
>>> the end I want to have all transcode stuff in parallel.
>>> 
>>> lukas@transcoder:~$ vainfo --display drm --device /dev/dri/card1
>>> libva info: VA-API version 1.1.0
>>> libva info: va_getDriverName() returns 0
>>> libva info: Trying to open /usr/lib/x86_64-linux-gnu/dri/
>>> radeonsi_drv_video.so
>>> libva info: Found init function __vaDriverInit_1_1
>>> libva info: va_openDriver() returns 0
>>> vainfo: VA-API version: 1.1 (libva 2.1.0)
>>> vainfo: Driver version: mesa gallium vaapi
>>> vainfo: Supported profile and entrypoints
>>> VAProfileMPEG2Simple: VAEntrypointVLD
>>> VAProfileMPEG2Main  : VAEntrypointVLD
>>> VAProfileVC1Simple  : VAEntrypointVLD
>>> VAProfileVC1Main: VAEntrypointVLD
>>> VAProfileVC1Advanced: VAEntrypointVLD
>>> VAProfileH264ConstrainedBaseline: VAEntrypointVLD
>>> VAProfileH264ConstrainedBaseline: VAEntrypointEncSlice
>>> VAProfileH264Main   : VAEntrypointVLD
>>> VAProfileH264Main   : VAEntrypointEncSlice
>>> VAProfileH264High   : VAEntrypointVLD
>>> VAProfileH264High   : VAEntrypointEncSlice
>>> VAProfileHEVCMain   : VAEntrypointVLD
>>> VAProfileHEVCMain10 : VAEntrypointVLD
>>> VAProfileJPEGBaseline   : VAEntrypointVLD
>>> VAProfileNone   : VAEntrypointVideoProc
>>> 
>>> lukas@transcoder:~$ ls -la /dev/dri/
>>> total 0
>>> drwxr-xr-x   3 root root   340 Jul 23 15:47 .
>>> drwxr-xr-x  19 root root  5020 Jul 23 15:47 ..
>>> drwxr-xr-x   2 root root   320 Jul 23 15:47 by-path
>>> crw-rw+  1 root video 226,   0 Jul 23 15:47 card0
>>> crw-rw+  1 root video 226,   1 Jul 23 15:47 card1
>>> crw-rw+  1 root video 226,   2 Jul 23 15:47 card2
>>> crw-rw+  1 root video 226,   3 Jul 23 15:47 card3
>>> crw-rw+  1 root video 226,   4 Jul 23 15:47 card4
>>> crw-rw+  1 root video 226,   5 Jul 23 15:47 card5
>>> crw-rw+  1 root video 226,   6 Jul 23 15:47 card6
>>> crw-rw+  1 root video 226, 128 Jul 23 15:47 renderD128
>>> crw-rw+  1 root video 226, 129 Jul 23 15:47 renderD129
>>> crw-rw+  1 root video 226, 130 Jul 23 15:47 renderD130
>>> crw-rw+  1 root video 226, 131 Jul 23 15:47 renderD131
>>> crw-rw+  1 root video 226, 132 Jul 23 15:47 renderD132
>>> crw-rw+  1 root video 226, 133 Jul 23 15:47 renderD133
>>> crw-rw+  1 root video 226, 134 Jul 23 15:47 renderD134
>>> 
>>> 
>>> ad 2. ok, understand. Is there a benefit of doing it that way?
>>> 
>>> ad 3. I have done two tests now with only the decoder running, which are
>>> confusing me now even more.
>>> 
>>> So running following command:
>>> ffmpeg -init_hw_device vaapi=amd:/dev/dri/renderD129 -hwaccel vaapi
>>> -hwaccel_output_format vaapi -hwaccel_device amd -filter_hw_device amd -i
>>> fs_experiental_method.avi  -f null -
>>> 
>>> Results in ~ 12x speed
>>> frame= 5824 fps=340 q=-0.0 Lsize=N/A time=00:03:14.47 bitrate=N/A
>>> speed=11.4x
>>> 
>>> But, using the CPU (a dual core pentium from last year)
>>> ffmpeg -i fs_experiental_method.avi  -f null -
>>> 
>>> Results in ~ 14x speed
>>> frame=10570 fps=408 

Re: [FFmpeg-user] RTP streaming and port blocking

[DopeLabs]

ffmpeg does not 'block' ports. it is not listening on port 9000, ffmpeg is 
outputting to the ip on port 9000 so its expecting something to be running on 
that port already, to receive the stream ffmpeg is sending.

i suggest you check your firewall settings, the firewall settings on the 
destination system, and any settings on the switch or router that may be 
between you and the destination system.

also make sure whatever software thats supposed to be listening for the ffmpeg 
stream on port 9000 is configured correctly and running. (log files can be 
informative)



> On Jul 23, 2018, at 7:36 35AM, Lokesh Bhat  wrote:
> 
> I'm using the following command to stream
> 
> ffmpeg -re -f alsa -ac 1 -ar 8000 -i hw:1 -f rtp rtp://192.168.0.25:9000
> 
> when the streaming starts, ffmpeg blocks the port 9000. Hence I can't use
> any other applications to listen on 9000. I have a need where i need to
> write and listen on the same port simultaniously.
> 
> Can someone please tell me how to make ffmpeg not to block the port?
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Re: [FFmpeg-user] Ffmpeg web launcher

[DopeLabs]

to the best of my knowledge, there is no 'ffmpeg web launcher'.

this sounds like something 3rd party or maybe your just looking for a way to 
execute ffmpeg process from a webpage..

either way both are beyond the scope of this mailing list.

also your attachment never made it =]



> On Jul 3, 2018, at 7:56 35AM, Higher Being <2007highfly...@gmail.com> wrote:
> 
> Anyone here know how to install ffmpeg web launcher on to a server ?
> 
> So we can add source and cookie and make a rtmp to send back to our XtremeZ
> panel
> 
> Thanks in advance.
> And picture added so you know what I'm after
> 
> Cheers
> 
> Donald
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Re: [FFmpeg-user] how to use ocr in ffmpeg to produce text message

[DopeLabs]

ffprobe -show_entries frame_tags=lavfi.ocr.text -f lavfi -i 
"movie=ocr_in.png,ocr"  > ocr.txt

[FRAME]
TAG:lavfi.ocr.text=ffmpeg version N-91398-gd08d4a8c73 Copyright (c) 2000-2018 
the FFmpeg developers


[/FRAME]


ffmpeg -f lavfi -i 
"color=black:s=1536x199,drawtext=fontfile=/Users/dopelabs/helvetica_bold.ttf:fontsize=24:fontcolor=white:textfile=ocr.txt"
 -frames 1 ocr_out.png





cheers =]


> On Jun 29, 2018, at 11:50 33AM, Paul B Mahol  wrote:
> 
> On 6/29/18, qw  wrote:
>> Hi,
>> 
>> 
>> ocr is new in ffmpeg 4.0.
>> 
>> 
>> 1)
>> I build ffmpeg 4.0 and its tesseract-3.05.01, and copy english training data
>> to /usr/local/share/tessdata/eng.traineddata.
>> 
>> 
>> I use the following command to run ocr filter in ffmpeg, but fail to find
>> eng.traineddata. But I have set TESSDATA_PREFIX:
>> TESSDATA_PREFIX=/usr/local/share LD_LIBRARY_PATH=/usr/local/lib
>> /usr/local/bin/ffmpeg -f lavfi -i "movie=test_ocr.png,
>> ocr=datapath=tessdata:language=eng, drawgraph=lavfi.ocr.text"
>> test_ocr_out.png
>> 
>> 
>> ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
>>  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-16)
>>  configuration: --enable-version3 --enable-asm --enable-x86asm
>> --enable-avfilter --disable-static --enable-shared --enable-gpl
>> --enable-nonfree --prefix=/usr/local/ --enable-libvidstab --enable-libass
>> --enable-libfreetype --extra-libs=-lfreetype --enable-libtesseract
>> --enable-libfdk_aac --enable-libmp3lame --enable-libx264
>> --enable-libopenjpeg --enable-libwebp --enable-libx265 --enable-libvorbis
>> --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib
>> --enable-stripping --enable-libmfx
>>  libavutil  56. 14.100 / 56. 14.100
>>  libavcodec 58. 18.100 / 58. 18.100
>>  libavformat58. 12.100 / 58. 12.100
>>  libavdevice58.  3.100 / 58.  3.100
>>  libavfilter 7. 16.100 /  7. 16.100
>>  libswscale  5.  1.100 /  5.  1.100
>>  libswresample   3.  1.100 /  3.  1.100
>>  libpostproc55.  1.100 / 55.  1.100
>> Error opening data file /tessdata/eng.traineddata
>> Please make sure the TESSDATA_PREFIX environment variable is set to the
>> parent directory of your "tessdata" directory.
>> Failed loading language 'eng'
>> Tesseract couldn't load any languages!
>> [Parsed_ocr_1 @ 0x1e03d80] failed to init tesseract
>> [lavfi @ 0x1df77c0] Error initializing filter 'ocr' with args
>> 'datapath=tessdata:language=eng'
>> movie=test_ocr.png, ocr=datapath=tessdata:language=eng,
>> drawgraph=lavfi.ocr.text: Invalid argument
>> 
>> 
>> 
>> 
>> If I set TESSDATA_PREFIX to /usr/local/share/tessdata, the same error
>> occurs:
>> 
>> 
>> TESSDATA_PREFIX=/usr/local/share/tessdata LD_LIBRARY_PATH=/usr/local/lib
>> /usr/local/bin/ffmpeg -f lavfi -i "movie=test_ocr.png,
>> ocr=datapath=tessdata:language=eng, drawgraph=lavfi.ocr.text"
>> test_ocr_out.png
>> 
>> 
>> ffmpeg version 4.0 Copyright (c) 2000-2018 the FFmpeg developers
>>  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-16)
>>  configuration: --enable-version3 --enable-asm --enable-x86asm
>> --enable-avfilter --disable-static --enable-shared --enable-gpl
>> --enable-nonfree --prefix=/usr/local/ --enable-libvidstab --enable-libass
>> --enable-libfreetype --extra-libs=-lfreetype --enable-libtesseract
>> --enable-libfdk_aac --enable-libmp3lame --enable-libx264
>> --enable-libopenjpeg --enable-libwebp --enable-libx265 --enable-libvorbis
>> --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib
>> --enable-stripping --enable-libmfx
>>  libavutil  56. 14.100 / 56. 14.100
>>  libavcodec 58. 18.100 / 58. 18.100
>>  libavformat58. 12.100 / 58. 12.100
>>  libavdevice58.  3.100 / 58.  3.100
>>  libavfilter 7. 16.100 /  7. 16.100
>>  libswscale  5.  1.100 /  5.  1.100
>>  libswresample   3.  1.100 /  3.  1.100
>>  libpostproc55.  1.100 / 55.  1.100
>> Error opening data file /tessdata/eng.traineddata
>> Please make sure the TESSDATA_PREFIX environment variable is set to the
>> parent directory of your "tessdata" directory.
>> Failed loading language 'eng'
>> Tesseract couldn't load any languages!
>> [Parsed_ocr_1 @ 0x18a8d80] failed to init tesseract
>> [lavfi @ 0x189c7c0] Error initializing filter 'ocr' with args
>> 'datapath=tessdata:language=eng'
>> movie=test_ocr.png, ocr=datapath=tessda

Re: [FFmpeg-user] ffmpeg support smb protocols

[DopeLabs]

open a new shell or screen and run configure again.


~/github/ffmpeg $ ./configure --enable-shared --enable-gpl --enable-nonfree 
--enable-avresample --enable-avisynth --enable-gnutls --enable-libass 
--enable-libbluray --enable-libbs2b --enable-libcaca --enable-libflite 
--enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme 
--enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg 
--enable-libopus --enable-libpulse --enable-librtmp --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-libzvbi --enable-openal --enable-libdc1394 --enable-libzmq 
--enable-frei0r --enable-libx264 --enable-libopencv --enable-libsmbclient 
--enable-version3
ERROR: libsmbclient not found


~/github/ffmpeg $ sudo apt-get install libsmbclient-dev
Reading package lists... Done
Building dependency tree
Reading state information... Done
The following packages were automatically installed and are no longer required:
  linux-headers-4.4.0-112 linux-headers-4.4.0-112-generic 
linux-headers-4.4.0-116 linux-headers-4.4.0-116-generic linux-headers-4.4.0-119 
linux-headers-4.4.0-119-generic
  linux-headers-4.4.0-121 linux-headers-4.4.0-121-generic 
linux-headers-4.4.0-124 linux-headers-4.4.0-124-generic linux-headers-4.4.0-91 
linux-headers-4.4.0-91-generic
  linux-image-4.4.0-112-generic linux-image-4.4.0-116-generic 
linux-image-4.4.0-119-generic linux-image-4.4.0-121-generic 
linux-image-4.4.0-124-generic
  linux-image-extra-4.4.0-112-generic linux-image-extra-4.4.0-116-generic 
linux-image-extra-4.4.0-119-generic linux-image-extra-4.4.0-121-generic
  linux-image-extra-4.4.0-124-generic
Use 'sudo apt autoremove' to remove them.
The following NEW packages will be installed:
  libsmbclient-dev
0 upgraded, 1 newly installed, 0 to remove and 129 not upgraded.
Need to get 40.8 kB of archives.
After this operation, 298 kB of additional disk space will be used.
Get:1 http://us.archive.ubuntu.com/ubuntu xenial-updates/main amd64 
libsmbclient-dev amd64 2:4.3.11+dfsg-0ubuntu0.16.04.13 [40.8 kB]
Fetched 40.8 kB in 0s (148 kB/s)
Selecting previously unselected package libsmbclient-dev:amd64.
(Reading database ... 528705 files and directories currently installed.)
Preparing to unpack 
.../libsmbclient-dev_2%3a4.3.11+dfsg-0ubuntu0.16.04.13_amd64.deb ...
Unpacking libsmbclient-dev:amd64 (2:4.3.11+dfsg-0ubuntu0.16.04.13) ...
Setting up libsmbclient-dev:amd64 (2:4.3.11+dfsg-0ubuntu0.16.04.13) ...
[5][02:46:04] dopelabs@sjc2:~/github/ffmpeg $


open a new screen

~/github/ffmpeg $ ./configure --enable-shared --enable-gpl --enable-nonfree 
--enable-avresample --enable-avisynth --enable-gnutls --enable-libass 
--enable-libbluray --enable-libbs2b --enable-libcaca --enable-libflite 
--enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme 
--enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg 
--enable-libopus --enable-libpulse --enable-librtmp --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-libzvbi --enable-openal --enable-libdc1394 --enable-libzmq 
--enable-frei0r --enable-libx264 --enable-libopencv --enable-libsmbclient 
--enable-version3

.

Enabled protocols:
asyncfile httpproxy
librtmps md5  rtp  tee
bluray   ftp  https
librtmpt mmsh sctp tls
cachegopher   icecast  
librtmptemmst srtp udp
concat   hls  librtmp  
libsmbclient pipe subfile  
udplite
crypto   http librtmpe 
libssh   prompeg  tcp  unix
data





Enabled indevs:
alsa lavfiopenal   oss  
pulsev4l2 xcbgrab
fbdevlibdc1394

Enabled outdevs:
alsa caca fbdevoss  
pulsev4l2

License: nonfree and unredistributable
libavutil/avconfig.h is unchanged
libavcodec/bsf_list.c is unchanged
[3][02:46:40] dopelabs@sjc2:~/github/ffmpeg $
[3][02:46:40] dopelabs@sjc2:~/github/ffmpeg $ make -j 8


.


> On Jun 26, 2018, at 2:42 28AM, Wang Kiven  wr

Re: [FFmpeg-user] how to use Intel qsv codec for ffmpeg 4.0

[DopeLabs]

have you searched the man page or. this page for "qsv" ?

https://ffmpeg.org/ffmpeg-all.html

> On Jun 19, 2018, at 1:44 46AM, qw  wrote:
> 
> Hi,
> 
> 
> I want to use Intel qsv encoder/decoder for ffmpeg 4.0. I use 
> '--enable-libmfx' to configure ffmpeg, and build it. What's ffmpeg command to 
> use Intel qsv h.264 encoder/decoder to do transcoding?
> 
> 
> Thanks!
> 
> 
> Regards
> 
> 
> Andrew
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Re: [FFmpeg-user] FFMPEG output to append to a text file without overwriting the content

[DopeLabs]

you can always just timestamp each text file so it writes a new file on each 
iteration of ffmpeg running

out.mp4 2> $(date +%Y-%m-%d_%H-%M-%S).txt



> On Jun 4, 2018, at 5:48 57PM, Sana Tafleen  wrote:
> 
> What I am trying to achieve here is to get the number of I,P and B frames
> at the end of streaming. When the disconnection happens, the ffmpeg process
> at the sender stops at the terminal and when I connect it back, it starts
> sending the frames. Doing this, the number of frames at the receiver is
> higher than the number of frames at the sender. I am guessing it is
> because, when the ffmpeg stops, it kills the previous process and starts a
> new process trying to gain connection with the destination. When it gets
> the connection, a new process starts and that output is sent to 'out.txt',
> overwriting the output of the previous ffmpeg process. Which is why the
> frame count at sender is lesser than that at the receiver. I hope this
> makes any sense. Any help here would be appreciated.
> 
> 
> Regards,
> Sana Tafleen
> 
> On Mon, Jun 4, 2018 at 7:36 PM, robertlazarski 
> wrote:
> 
>> On Mon, Jun 4, 2018 at 6:06 PM, Sana Tafleen 
>> wrote:
>> 
>>> Hello,
>>> 
>>> I am sending a UDP stream to the destination and saving the contents of
>> the
>>> FFMPEG output to a text file. I run the ffmpeg command in a loop as
>>> follows,
>>> 
>>> 
>>> while :
>>> do
>>>  echo `ffmpeg -hide_banner -f v4l2 -i /dev/video0 -c:v libx264 -f
>>> mpegts tcp://ip:port -c:v libx264 /path/to/.mp4 2> out.txt -y`
>>> done
>>> 
>>> 
>>> 
>>> When I disconnect the cable connected the destination, the above ffmpeg
>>> process stops and a new one starts oevrwriting the content of the output
>>> that has been saved. And when I reconnect the cable, a new ffmpeg process
>>> starts and its output is what is displayed in the out.txt file.
>>> 
>>> I need the output of each ffmpeg process that runs to an output file. Can
>>> anyone please suggest me a way to do it?
>>> 
>> 
>> This part of the command, '2> out.txt ' says (a) redirect stderr and not
>> stdout, to out.txt. And (b) overwrite the previous contents.
>> 
>> You can append instead of overwrite by using '2>> out.txt ' .
>> 
>> I would not expect ffmpeg to create a new process in that loop but I have
>> no experience with mpegts. I would try -stdin since I have seen unexpected
>> behavior without it in loops.
>> 
>> A long shot would be using nohup, if its a hangup of some sort. I would try
>> that if I was still stuck. Then strace on the command to see why it was
>> creating the extra process.
>> 
>> Kind regards,
>> Robert
>> 
>> 
>> 
>>> ___
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>>> 
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>> 
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> ___
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Re: [FFmpeg-user] How add expression in drawtext's text

[DopeLabs]

i think this is what your looking for... 

this will pad 0's to the left to achieve a 5 digit number

please note the level of escaping you need depends on your environment 


text='%{eif\\:n+1\\:d\\:5}'



> On Jun 3, 2018, at 8:54 46PM, qw  wrote:
> 
> Hi,
> 
> 
> I can use the following command to add frame number on video:
> 
> 
> /usr/local/bin/ffmpeg -t 180 -i input.mp4 -acodec libfdk_aac -ac 2 -b:a 48k 
> -vcodec libx264 -b:v 500k -g 25 \
> -vf "fps=fps=25, scale=w=320:h=240, 
> drawtext='box=1:fontcolor=red:fontsize=20:fontfile=OpenSans-Regular.ttf:text=%{frame_num}'"
>  \
> -f mp4 out.mp4
> 
> 
> But the width of the text box will increase as frame number increases. I want 
> to make the width of the text box unchanged. If the largest frame number is 
> 9, I want to make the width of the text is of 5 numbers. How to do it?
> 
> 
> 
> 
> Another question:
> 
> 
> I want to make the following logic for text:
> 
> 
> if frame number < 10
>text=%{frame_num}
> else if frame number < 100
>text=000%{frame_num}
> else if frame number < 1000
>text=00%{frame_num}
> else if frame number < 1
>text=0%{frame_num}
> else
>text=%{frame_num}
> 
> 
> How to do it? How can I add expression in text? Is there any command example?
> 
> 
> Thanks
> 
> regards
> 
> andrew
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Re: [FFmpeg-user] How add expression in drawtext's text

[DopeLabs]

this should give you a preview...

ffplay -f lavfi 
"color=c=black:s=640x480,drawtext=box=1:fontcolor=red:fontsize=20:text='%{eif\:n+1\:d\:5}':fontfile=OpenSans-Regular.ttf"

> On Jun 3, 2018, at 8:54 46PM, qw  wrote:
> 
> Hi,
> 
> 
> I can use the following command to add frame number on video:
> 
> 
> /usr/local/bin/ffmpeg -t 180 -i input.mp4 -acodec libfdk_aac -ac 2 -b:a 48k 
> -vcodec libx264 -b:v 500k -g 25 \
> -vf "fps=fps=25, scale=w=320:h=240, 
> drawtext='box=1:fontcolor=red:fontsize=20:fontfile=OpenSans-Regular.ttf:text=%{frame_num}'"
>  \
> -f mp4 out.mp4
> 
> 
> But the width of the text box will increase as frame number increases. I want 
> to make the width of the text box unchanged. If the largest frame number is 
> 9, I want to make the width of the text is of 5 numbers. How to do it?
> 
> 
> 
> 
> Another question:
> 
> 
> I want to make the following logic for text:
> 
> 
> if frame number < 10
>text=%{frame_num}
> else if frame number < 100
>text=000%{frame_num}
> else if frame number < 1000
>text=00%{frame_num}
> else if frame number < 1
>text=0%{frame_num}
> else
>text=%{frame_num}
> 
> 
> How to do it? How can I add expression in text? Is there any command example?
> 
> 
> Thanks
> 
> regards
> 
> andrew
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Re: [FFmpeg-user] How to enforce VBR minimum bitrate

[DopeLabs]

you can combine steps 1 and 2 into a single command:

$ ffmpeg- i input -options -f wav - | lame -V 5 -F -b 128 - output.mp3


> On Jun 4, 2018, at 7:52 42AM, Zak  wrote:
> 
> On 2018-06-04 at 3:03 AM EDT, sook...@expressinmusic.com wrote:
> 
>> Hi experts!
>> 
>> I am currently using ffmpeg as a solution to audio file conversion and
>> normalization.
>> 
>> I understand that my setting (-q:a 5) below should produce output with
>> bitrate range in between 120-150kbps, or sometimes even lower than that if
>> the input already in being a lower bitrate than requested.
>> 
>> By using VBR Encoding, is there an option that is able to constrain the
>> bitrate minimum at 128kbps?
>> 
>> ffmpeg -i "input" -af "silenceremove=0:0:0:-1:1:-80dB,
>> dynaudnorm=p=1:m=3:r=1" -vn -acodec libmp3lame -q:a 5 -ar 44100 "output.mp3"
>> 
>> Please kindly advise. Thanks!
>> 
>> Regards,
>> 
>> Sook Sin
> 
> Hello Sook Sin,
> 
> Disclamer: It might depend on your version of FFmpeg and libmp3lame, but I 
> doubt it (assuming versions from 2012 or more recent). The libmp3lame 
> behavior in this respect has been the same since at least 2012, and the 
> FFmpeg behavior has probably been the same in this respect. (If it changed, 
> it would indicate that FFmpeg removed a feature, and I don't think they did 
> this. There is no reason to remove features.)
> 
> Short version: You can set a minimum bitrate in VBR mode with the LAME 
> command line interface, but I think you CANNOT do this with the version of 
> libmp3lame that is linked by FFmpeg during compilation of FFmpeg. I just 
> tried, the minimum bitrate was not respected, every frame of silence was 
> encoded at 32 kbps. A work-around is to use your FFmpeg command to make an 
> uncompressed PCM file, such as a WAV file, and then convert the WAV to MP3 
> using the LAME command line interface (going forward, I will call it the LAME 
> CLI). I give an example of how to do this below.
> 
> Question for the whole list:
> Is it possible to pass arbitrary options from FFmpeg to libmp3lame as if they 
> came from argc and argv? I am guessing no. libx264 seems to take arbitrary 
> options via FFmpeg, but I am guessing libmp3lame does not. Sook Sin wants to 
> pass in the option "-F" as if it came from the command line. (Or rather, if 
> this were possible it would solve the problem at hand.)
> 
> Why:
> 
> In order to tell the LAME CLI to do VBR but also enforce a minimum bitrate 
> for all frames of MP3 audio data, you need to give the LAME CLI a command 
> like this (this is valid Bash syntax for GNU/Linux, MacOS, or Cygwin on 
> Windows):
> 
> bash$ lame -V 5 -F -b 128 input.wav output.mp3
> 
> The option "-V 5" for the LAME CLI is the same as "-q:a 5" in FFmpeg, and in 
> fact FFmpeg passes the number 5 directly to libmp3lame to be used by 
> libmp3lame as it sees fit. The FFmpeg option "-q:a 5" should not be confused 
> with the LAME CLI option "-q 5", which is a different encoder parameter.
> 
> The option -V (LAME) and its sister -q:a (FFmpeg) control the file size and 
> target bitrate.
> 
> The option -q (LAME) and its sister -compression_level:a (FFmpeg) control how 
> fast or slow the encoder is. "-q 0" (LAME CLI) is equal to 
> "-compression_level:a 0" (FFmpeg), and it will be the slowest during 
> compression, equally fast during playback, and give the best-sounding 
> results. "-q 3" is the default in VBR mode for LAME (both CLI and 
> FFmpeg-linked). In CBR mode, -q 3 is often the same as -q 0, but it depends 
> on the sample rate of the input and the output bitrate of the MP3. For CD 
> audio (44.1 kHz stereo) at CBR at 320 kbps, -q 3 is actually a tiny bit 
> faster than -q 0. At lower bitrates they are often identical algorithms, 
> speed, and output. -q 0 is the highest quality and slowest compression, -q 9 
> is the lowest quality and fastest, the output file sizes are generally about 
> the same with no systematic trend larger or smaller.
> 
> The option -F (LAME) is the key option that you want. It means FORCE the 
> encoder to use frames of a certain minimum size, even if it can achieve the 
> target quality (implied by -V 5) using a small frame. You say -F and then you 
> say -b N with the frame size in kbps. So for a minimum frame size of 128 
> kbps, it is the command above with "-F -b 128". This does indeed work in LAME 
> v3.100, I just tested it. Even silence will be 128 kbps. In fact, if I set -V 
> 9 (which should be about 65 kbps and should sound terrible), the whole file 
> is created using 128 kbps frames and the quality is much better, because 128 
> kbps sounds markedly better than ~65 kbps VBR. (These examples and numbers 
> are all for CD audio, in which silence would normally be 32 kbps in -V N for 
> any value of N. Frames smaller than 32 kbps are not valid at CD sample rates 
> in the MP3 standard - VLC will play such files, iTunes will not. LAME CLI 
> will make such a file if you compile in debug mode. FFmpeg will not make such 
> files, as fa

Re: [FFmpeg-user] Tag mp4s incompatible with output codec ?

[DopeLabs]

using -map 0 produced the following error

[mp4 @ 0x7ff51b806e00] Tag mp4s incompatible with output codec id '0' 
([0][0][0][0])
Could not write header for output file #0 (incorrect codec parameters ?): 
Invalid data found when processing input
Error initializing output stream 0:1 --
[libfdk_aac @ 0x7ff51b802e00] 2 frames left in the queue on closing
Conversion failed!

removed -map 0 and ran again.. then got the error

Unrecognized option 'ns'.
Error splitting the argument list: Option not found

removed -ns

then removed the following options as ffmpeg seemed to output the same without 
them

-c:s copy -map_metadata 0 -map_metadata:s:v 0:s:v -map_metadata:s:a 0:s:a

used -c:a copy since the input is already aac and at 96k... if you still really 
want to transcode it to 128k you can, but your still hearing 96k audio.

added -to 00:01:00.000 for the purposes of making a 1 minute preview for you, 
and renamed the input and output file names for cleanliness 


behold:

https://www.dlnetworks.net/ffmpeg/output.mp4

$ ffmpeg -i input.mp4 -c:a copy -c:v hevc -preset faster -vtag hvc1 -crf 25 
-profile:v main -x265-params 
'me=umh:subme=5:ref=4:aq-mode=2:aq-strength=1.0:psy-rd=2.0:psy-rdoq=1.0:rd=6:log-level=1'
 -to 00:01:00.000 -f mp4 output.mp4

ffmpeg version N-90287-g43205df645 Copyright (c) 2000-2018 the FFmpeg developers
  built with Apple LLVM version 9.0.0 (clang-900.0.39.2)
  configuration: --enable-libmp3lame --enable-opencl --enable-audiotoolbox 
--enable-videotoolbox --enable-libx265 --enable-libx264 --enable-libxvid 
--enable-libfdk-aac --enable-lzma --enable-gnutls --enable-fontconfig 
--enable-libfreetype --enable-libfribidi --enable-nonfree --enable-gpl 
--enable-libass --enable-swscale --enable-avfilter --enable-libvorbis 
--enable-libopus --enable-librsvg --enable-libtheora --enable-libopenjpeg 
--enable-libmodplug --enable-libvpx --enable-avresample --enable-libsoxr 
--enable-libspeex --enable-libcaca --enable-opengl
  libavutil  56.  9.100 / 56.  9.100
  libavcodec 58. 14.100 / 58. 14.100
  libavformat58. 10.100 / 58. 10.100
  libavdevice58.  2.100 / 58.  2.100
  libavfilter 7. 12.100 /  7. 12.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale  5.  0.102 /  5.  0.102
  libswresample   3.  0.101 /  3.  0.101
  libpostproc55.  0.100 / 55.  0.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.mp4':
  Metadata:
major_brand : mp42
minor_version   : 0
compatible_brands: mp42isomavc1
creation_time   : 2017-11-09T15:56:49.00Z
compilation : 0
encoder : Sorenson Squeeze
  Duration: 00:26:06.00, start: 0.00, bitrate: 5280 kb/s
Stream #0:0(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 96 kb/s (default)
Metadata:
  creation_time   : 2017-11-09T15:56:49.00Z
  handler_name: soun
Stream #0:1(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p(tv, 
bt709/bt709/unknown), 1920x1080, 5002 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc 
(default)
Metadata:
  creation_time   : 2017-11-09T15:56:49.00Z
  handler_name: vide
  encoder : AVC Coding
Stream #0:2(und): Data: none (mp4s / 0x7334706D) (default)
Metadata:
  creation_time   : 2017-11-09T15:56:58.00Z
  handler_name: sdsm
Stream #0:3(und): Data: none (mp4s / 0x7334706D) (default)
Metadata:
  creation_time   : 2017-11-09T15:56:58.00Z
  handler_name: odsm
Stream #0:4(eng): Data: none (rtp  / 0x20707472), 13 kb/s
Metadata:
  creation_time   : 2017-11-10T15:56:58.00Z
Stream #0:5(eng): Data: none (rtp  / 0x20707472), 162 kb/s
Metadata:
  creation_time   : 2017-11-10T15:57:00.00Z
Stream mapping:
  Stream #0:1 -> #0:0 (h264 (native) -> hevc (libx265))
  Stream #0:0 -> #0:1 (copy)
Press [q] to stop, [?] for help
Output #0, mp4, to 'output.mp4':
  Metadata:
major_brand : mp42
minor_version   : 0
compatible_brands: mp42isomavc1
compilation : 0
encoder : Lavf58.10.100
Stream #0:0(und): Video: hevc (libx265) (hvc1 / 0x31637668), yuv420p, 
1920x1080, q=2-31, 25 fps, 12800 tbn, 25 tbc (default)
Metadata:
  creation_time   : 2017-11-09T15:56:49.00Z
  handler_name: vide
  encoder : Lavc58.14.100 libx265
Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 96 kb/s (default)
Metadata:
  creation_time   : 2017-11-09T15:56:49.00Z
  handler_name: soun
frame= 1500 fps= 10 q=-0.0 Lsize=6264kB time=00:00:59.97 bitrate= 
855.6kbits/s speed=0.408x
video:5509kB audio:704kB subtitle:0kB other streams:0kB global headers:2kB 
muxing overhead: 0.829822%



> On Jun 1, 2018, at 2:19 06AM, Gabriel Zachmann 
>  wrote:
> 
> Thanks a lot for your response.
> 
>>> Could that have any negative side effects when I try to convert some other 
>>> movies?
>> 
>> It depends. If video editing was simple, it wouldn't require complex
>> tools. ;-)

Re: [FFmpeg-user] Infinite loop randomized stream to rtmp server

[DopeLabs]

unfortunately, i still dont think there is way to 'loop' a video... or a 
playlist of videos. it will work with images using -loop 1.

you were pretty close using contcat.

you may wish to try something along the lines of:

> File1.txt
> 
> file 'item1.mp4'

> file 'item2.mp4'
> file 'item3.mp4'
> file 'item4.mp4'
> file 'item5.mp4'
> file 'item6.mp4'
> file 'item7.mp4'
> file 'item8.mp4'
> file 'item9.mp4'
> file 'item10.mp4'


then if youd like to make it random you can

$ cat File1.txt | shuf > RandFiles.txt

also be aware that its generally a good idea to make sure when using concat to 
have all the input files be encoded/formatted using the same codec/rates/frame 
sizes/etc.


i am using wowza and found a nifty little add-on called 'loop until live' which 
when used in combo with a small script i wrote to run ffmpeg in a while loop, 
grabbing a new random file each iteration, allows for uninterrupted playback.

you can msg me directly if you would like further info about this as its beyond 
the scope of this mailing list.

cheers


> On May 26, 2018, at 11:42 43AM, Manuel Celli <88lloy...@gmail.com> wrote:
> 
> Dear All,
> 
> As the subject implies, i have troubles to make a running ffmpeg command
> that can do the following:
> 
> -Infinite "loop" (must stream undefinetly without breaks between videos)
> -Randomized (must reproduce videos in a random order)
> 
> After much searching in the web, i came to this:
> 
> ffmpeg -y -re -stream_loop -1 -i item1.mp4 -async 1 -vsync 1 -c:a aac
> -filter:a loudnorm=I=-23:LRA=1:dual_mono=true:tp=-1 -f flv -ar 44100 -c:v
> copy -preset fast -keyint_min 25 "rtmp://live.my.stream"
> 
> that correctly streams the video but it breaks on repetition, so it's a no
> go.
> 
> I tried to remove -stream_loop -1 and implement concat that uses File1.txt
> and File2.txt which contain:
> 
> File1.txt
> 
> file 'item1.mp4'
> file 'File2.txt'
> 
> and then File2.txt
> 
> File2.txt
> 
> file 'item2.mp4'
> file 'File1.txt'
> 
> to simulate an infinite loop and use linux symbolic links to make item1 and
> item2 point to different files, with the following command:
> 
> ffmpeg -y -re -f concat -i File1.txt -async 1 -vsync 1 -c:a aac -filter:a
> loudnorm=I=-23:LRA=1:dual_mono=true:tp=-1 -f flv -ar 44100 -c:v copy
> -preset fast -keyint_min 25 "rtmp://live.my.stream"
> 
> it correctly streams the first file but it breaks when it tries to open the
> second file (link to the file File2.txt) with the following output:
> 
> [concat @ 0x55be73c15aa0] DTS 2248 < 8085000 out of order
> [flv @ 0x55be73cf4300] Non-monotonous DTS in output stream 0:0; previous:
> 89833, current: 25; changing to 89833. This may result in incorrect
> timestamps in the output file.
> Error while filtering: Illegal seek
> [flv @ 0x55be73cf4300] Failed to update header with correct duration.
> [flv @ 0x55be73cf4300] Failed to update header with correct filesize.
> 
> So i'm actually out of options to do infinite stream of random mp4 files, i
> really need a hand since i'm new to ffmpeg but i really like it!
> 
> Regards
> ___
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Re: [FFmpeg-user] ffmpeg transcoding multicast to multicast stuck in disk sleep

[DopeLabs]

after a quick look here: 
https://trac.ffmpeg.org/wiki/HWAccelIntro#CUDACUVIDNVDEC

in your provided command, the cuvid decoder is passing the frames through 
system memory.

you can achieve full hardware transcode with CUVID and NVENC:

ffmpeg -hwaccel cuvid -c:v h264_cuvid -i input -c:v h264_nvenc -preset 
ultrafast output.mkv

i also could not find the -gpu option anywhere in the ffmpeg documentation, but 
i did find 

-hwaccel_device 0

i havent had any experience using cuvid or nvenc but this is just what i found 
after a quick look.. maybe it can help =]

cheers


> On May 25, 2018, at 12:20 30PM, Jakub Vojáček  wrote:
> 
> Hello,
> 
> we are receiving multicast and using nvidia, we are transcoding it into 2
> bitrates and sending to another multicast. When we are having just one
> ffmpeg like this, it is working fine. But as we add more channels to
> transcode, the ffmpeg is suddenly having problems. It slows down the
> transcoding speed from 1x to less, usually 0.6-0.999. But as it is
> live content, we cannot have it slower than 1x. The server has plenty of
> available resources (RAM, CPU, GPU) when this problem starts to happen.
> 
> By looking deeper into the problem, we found that the ffmpeg process is in
> the D state (disk sleep) using cat /proc/PID/status. But according to iotop
> the hdd, has still available resources. And why is ffmpeg even using
> hdd when transcoding from multicast to multicast (network to network)?
> There should be no HDD involved, or am I mistaken?
> 
> Sample ffmpeg command:
> 
> ffmpeg -y  -gpu 1 -c:v h264_cuvid  -deint adaptive -drop_second_field 1
> -resize 1920x1080 -i 'udp://@239.192.11.8:5000?fifo_size=1000&bitrate=
> 1200&pkt_size=1316
> '
> -filter_complex "[0:0]drawtext=text='':font=DejaVuSans:fontsize=40:
> fontcolor=#dd:x=(w/2)-(text_w/2):y=h*(0.05):box=1:
> boxborderw=10:boxcolor=#dd[main],[main]split=2[v1][v2],[v1]scale=512:288[v1]"
> \
> -map "[v1]" -flush_packets 0 -gpu 1 -r:v:0 25 -aspect:v:0 16:9 -g:v:0 80
> -vcodec:v:0 h264_nvenc -b:v:0 400K -minrate:v:0 1000k -maxrate:v:0 1000k
> -bufsize:v:0 500k \
> -map "[v2]" -flush_packets 0 -gpu 1 -r:v:1 25 -aspect:v:1 16:9 -g:v:1 80
> -vcodec:v:1 h264_nvenc -b:v:1 4500K -minrate:v:1 4500k -maxrate:v:1 4500k
> -bufsize:v:1 2250k \
> -map i:0xb08 -flush_packets 0 -ab:a:0 128k -ar:a:0 48k -acodec:a:0
> libfdk_aac -ac:a:0 2 -flags:a:0 +global_header \
> -f mpegts "udp://@239.100.100.1:1234"
> 
> ffmpeg configuration (we tried with latest version as well):
> 
> ffmpeg version 3.3.1 Copyright (c) 2000-2017 the FFmpeg developers
>  built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
>  configuration: --prefix=/usr/local --enable-nonfree --enable-gpl
> --enable-version3 --enable-shared --enable-pic --enable-avresample
> --enable-fontconfig --enable-frei0r --enable-ladspa --enable-libass
> --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfdk-aac
> --enable-libflite --enable-libfreetype --enable-libfribidi
> --enable-libmodplug --enable-libmp3lame --enable-libopus --enable-librtmp
> --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtwolame
> --enable-libv4l2 --enable-libwavpack --enable-libwebp --enable-libx264
> --enable-libx265 --enable-libzvbi --enable-decklink --enable-nvenc
> --enable-openssl
> 
> Thank you for your ideas and advice.
> 
> Jakub
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Re: [FFmpeg-user] ffmpeg and two (2) logitech 920 webcams

[DopeLabs]

you could try using the alternate names.. examples are shown here

https://www.ffmpeg.org/ffmpeg-devices.html#dshow


is it always that specific camera?

if you connect the one that isnt working first so that its listed as #1, will 
it work?

have you tried connecting the 2nd camera to a separate USB bus?

you can use the following tool to get usb bus info on your system

https://github.com/Microsoft/Windows-driver-samples/tree/master/usb/usbview



> On May 22, 2018, at 9:35 03AM, Frank Belson  wrote:
> 
> The following is the output from ffmpeg -list_devices true -f dshow -i dummy
> 
> ffmpeg version N-87353-g183fd30 Copyright (c) 2000-2017 the FFmpeg
> developers
>  built with gcc 7.2.0 (GCC)
>  configuration: --enable-gpl --enable-version3 --enable-cuda
> --enable-cuvid --enable-d3d11va --enable-dxva2 --enable-libmfx
> --enable-nvenc --enable-avisynth --enable-bzlib --enable-fontconfig
> --enable-frei0r --enable-gnutls --enable-iconv --enable-libass
> --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype
> --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug
> --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libopenh264 --enable-libopenjpeg --enable-libopus --enable-librtmp
> --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc
> --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp
> --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid
> --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib
>  libavutil  55. 76.100 / 55. 76.100
>  libavcodec 57.106.101 / 57.106.101
>  libavformat57. 82.101 / 57. 82.101
>  libavdevice57.  8.101 / 57.  8.101
>  libavfilter 6.105.100 /  6.105.100
>  libswscale  4.  7.103 /  4.  7.103
>  libswresample   2.  8.100 /  2.  8.100
>  libpostproc54.  6.100 / 54.  6.100
> [dshow @ 002f6c00] DirectShow video devices (some may be both video
> and audio devices)
> [dshow @ 002f6c00]  "Logitech Cam#1"
> [dshow @ 002f6c00] Alternative name
> "@device_pnp_\\?\usb#vid_046d&pid_082d&mi_00#6&10047afe&0&#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{bbefb6c7-2fc4-4139-bb8b-a58bba724083}"
> [dshow @ 002f6c00]  "Logitech Cam#2"
> [dshow @ 002f6c00] Alternative name
> "@device_pnp_\\?\usb#vid_046d&pid_082d&mi_00#6&3a78d907&0&#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
> [dshow @ 002f6c00]  "Kodak S101 Webcam"
> [dshow @ 002f6c00] Alternative name
> "@device_pnp_\\?\usb#vid_0979&pid_0206&mi_00#6&2455fd16&0&#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
> [dshow @ 002f6c00] DirectShow audio devices
> [dshow @ 002f6c00]  "Microphone (4- Kodak S101 Micro"
> [dshow @ 002f6c00] Alternative name
> "@device_cm_{33D9A762-90C8-11D0-BD43-00A0C911CE86}\Microphone (4- Kodak
> S101 Micro"
> [dshow @ 002f6c00]  "Line (WsAudio_Device)"
> [dshow @ 002f6c00] Alternative name
> "@device_cm_{33D9A762-90C8-11D0-BD43-00A0C911CE86}\Line (WsAudio_Device)"
> [dshow @ 002f6c00]  "Microphone (2- HD Pro Webcam C9"
> [dshow @ 002f6c00] Alternative name
> "@device_cm_{33D9A762-90C8-11D0-BD43-00A0C911CE86}\Microphone (2- HD Pro
> Webcam C9"
> [dshow @ 002f6c00]  "Microphone (HD Pro Webcam C920)"
> [dshow @ 002f6c00] Alternative name
> "@device_cm_{33D9A762-90C8-11D0-BD43-00A0C911CE86}\Microphone (HD Pro
> Webcam C920)"
> dummy: Immediate exit requested
> 
> 
> On Tue, May 22, 2018 at 6:59 AM, DopeLabs  wrote:
> 
>> what is the output of
>> 
>> $ ffmpeg -list_devices true -f dshow -i dummy
>> 
>> 
>> if you have multiple entries of a webcam named "Logitech Cam", you can
>> specify which using -video_device_number
>> 
>> the following would select the 2nd identical device (as numbers for
>> devices start at 0)
>> 
>> $ ffmpeg -f dshow -video_device_number 1 -i video="Logitech Cam"
>> 
>> 
>> im also curious.. would this work? video="Logitech Cam#0"
>> 
>> (please include complete uncut console output)
>> 
>> running the latest version of ffmpeg?
>> 
>> 
>>> On May 19, 2018, at 3:34 17PM, Brenda Spinner 
>> wrote:
>>> 
>>> I am using the following line to preview my webcam:
>>> 
>>> ffmpeg -s 1280x720 -framerate 30 -pix_fmt yuv420p -rtbufsize 100MB -f
>> 

Re: [FFmpeg-user] ffmpeg and two (2) logitech 920 webcams

[DopeLabs]

what is the output of

$ ffmpeg -list_devices true -f dshow -i dummy


if you have multiple entries of a webcam named "Logitech Cam", you can specify 
which using -video_device_number

the following would select the 2nd identical device (as numbers for devices 
start at 0)

$ ffmpeg -f dshow -video_device_number 1 -i video="Logitech Cam"


im also curious.. would this work? video="Logitech Cam#0"

(please include complete uncut console output)

running the latest version of ffmpeg?


> On May 19, 2018, at 3:34 17PM, Brenda Spinner  wrote:
> 
> I am using the following line to preview my webcam:
> 
> ffmpeg -s 1280x720 -framerate 30 -pix_fmt yuv420p -rtbufsize 100MB -f dshow
> -i video="Logitech Cam#1" -c:v copy -an -f sdl "WebCAM Preview"
> 
> this line works fine with cam #1, however when try it with cam #2, I get
> the error:
> 
> [dshow @ 002d8920] Could not set video options video=Logitech
> Cam#2: I/O error
> 
> Can anyone tell me why this is happening. I would like to able to preview
> both webcams at the same time?
> 
> Thanks, Frank
> 
> 
> Virus-free.
> www.avast.com
> 
> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>
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Re: [FFmpeg-user] Live streaming with dynamic overlay

[DopeLabs]

there is a method but its a bit hacky..

here is a demonstration file i just created using the below details

http://dopelabs.com/video/dynamic_overlay.mp4


i created 2 image files the same frame size as my output video, one thats 
completely transparent, and one that has the graphic i want to display with a 
transparent background

here are the 2 image files i used

http://www.dopelabs.com/images/alpha001.png
http://www.dopelabs.com/images/ffmpeg.png

i am screen capturing one of my displays...

ffmpeg -loglevel 0 -f avfoundation -r 30 -i "3:" 

i used globbing and the loop option to have ffmpeg read an image sequence (of 1 
image) and loop.

-pattern_type glob -loop 1 -i 'alpha00*.png'

i then overlaid the transparent image on top.

-filter_complex '[0:v][1:v] overlay'

what i end up with is something like this

ffmpeg -loglevel 0 -f avfoundation -r 30 -i "3:" -pattern_type glob -loop 1 -r 
30 -i 'alpha00*.png' -filter_complex '[0:v][1:v] overlay' -c:v h264 -b:v 1k 
-preset ultrafast -pix_fmt yuv420p -movflags +faststart -y -r 30 
dynamic_overlay.mp4

now here comes the tricky bit.. since ffmpeg is reading the image file for 
every frame, we can swap out the image with a new one, but only if done 
atomically.

first i made copies of both files

$ cp alpha001.png alpha1.png
$ cp ffmpeg.png ffmpeg1.png

then i atomically replaced the file ffmpeg is reading with the new copy 
containing the graphic i want to display

$ mv ffmpeg1.png alpha001.png

when i am ready to remove the graphic i atomically replace the with the 
transparent copy

$ mv alpha1.png alpha001.png

if you try to replace the file non atomically such as cp or simply overwriting, 
ffmpeg may (and usually does) fail and shuts down.


for sound effects you might want to look into something like jack or some other 
utility that has the ability to create virtual audio interfaces.

you can then specify said interface as input in ffmpeg and have the sound 
effects player output to the virtual interface, then use something like amix or 
amerge if theres already an audio track

i hope this may help a little.. as i said its a bit hacky, but easily 
scriptable if you wanted to automate.

cheers,

DL 


> On Apr 19, 2018, at 8:38 40AM, Nam Giang  wrote:
> 
> Hi,
> 
> Thanks for your comment. Do you mean we need to tap into the code of ffmpeg
> in order to do that?
> 
> Otherwise what I see is just a bunch of command lines.
> 
> Part of my streaming agenda comes from a Raspberry Pi that does not have a
> gui so obs won’t be an option I’m afraid.
> 
> On Thu, Apr 19, 2018 at 2:41 AM DopeLabs  wrote:
> 
>> it can sort of be done, but not practical or easy to accomplish at all..
>> 
>> youd be better off using something like OBS, which makes the things you
>> want to do very easy.
>> 
>> OBS already uses or can use ffmpeg behind the scenes anyways =]
>> 
>> cheers =]
>> 
>> 
>>> On Apr 18, 2018, at 10:16 55PM, Nam Giang  wrote:
>>> 
>>> Hi there,
>>> 
>>> I'm using ffmpeg to live stream to Youtube. I'm wondering if I can do
>>> dynamic overlay.
>>> 
>>> For example, when my stream is live and the world is watching, at some
>>> points I want to pop up some pictures or add some sound effects to my
>> live
>>> stream.
>>> 
>>> I wonder if there is any command that I can use to add those things into
>> my
>>> live stream?
>>> 
>>> Thanks
>>> ___
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Re: [FFmpeg-user] Live streaming with dynamic overlay

[DopeLabs]

it can sort of be done, but not practical or easy to accomplish at all..

youd be better off using something like OBS, which makes the things you want to 
do very easy.

OBS already uses or can use ffmpeg behind the scenes anyways =]

cheers =]


> On Apr 18, 2018, at 10:16 55PM, Nam Giang  wrote:
> 
> Hi there,
> 
> I'm using ffmpeg to live stream to Youtube. I'm wondering if I can do
> dynamic overlay.
> 
> For example, when my stream is live and the world is watching, at some
> points I want to pop up some pictures or add some sound effects to my live
> stream.
> 
> I wonder if there is any command that I can use to add those things into my
> live stream?
> 
> Thanks
> ___
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> 
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Re: [FFmpeg-user] H265 multi

[DopeLabs]

4.100 / 58. 14.100
  libavformat58. 10.100 / 58. 10.100
  libavdevice58.  2.100 / 58.  2.100
  libavfilter 7. 12.100 /  7. 12.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale  5.  0.102 /  5.  0.102
  libswresample   3.  0.101 /  3.  0.101
  libpostproc55.  0.100 / 55.  0.100
Input #0, mpegts, from '640.ts':
  Duration: 00:00:08.15, start: 1.40, bitrate: 915 kb/s
  Program 1
Metadata:
  service_name: Service01
  service_provider: FFmpeg
Stream #0:0[0x100]: Video: hevc (Main) ([36][0][0][0] / 0x0024), 
yuv420p(tv), 640x360 [SAR 1:1 DAR 16:9], 60 fps, 60 tbr, 90k tbn, 60 tbc
Stream #0:1[0x101]: Audio: aac (HE-AACv2) ([15][0][0][0] / 0x000F), 44100 
Hz, stereo, fltp, 29 kb/s
[22][03:46:51] dopelabs@jeez:/Volumes/CAM/vg_ytrip $



> On Apr 16, 2018, at 12:00 01PM, José María Infanzón  
> wrote:
> 
> Hi all,
>  I'm trying to get multiple outputs (different resuloutions) out
> of the same input. I was able to achieve this with h264 based on the guide:
> https://trac.ffmpeg.org/wiki/Creating%20multiple%20outputs but on the case
> of HEVC, I'm being unable
> 
> I've tried with:
> 
> /root/ffmpeg_sources/ffmpeg/ffmpeg -loglevel error -y -i udp://@
> 224.2.2.3:1012 -s 640X360 -vcodec libx265 -pix_fmt yuv420p -tune
> zerolatency -x265-params "bitrate=700:vbv-bufsize=1400" -acodec libfdk_aac
> -profile:a aac_he_v2 -b:a 32K -f mpegts - -s 768X432 -vcodec libx265
> -pix_fmt yuv420p -tune zerolatency -x265-params
> "bitrate=1100:vbv-bufsize=2200" -acodec libfdk_aac -profile:a aac_he_v2
> -b:a 32K -f mpegts -
> 
> Where I have 2 resolutions (640x360 & 768x432) with the same audio encoding.
> 
> But it doesn't work.
> 
> 
> 
> 
> 
> -- 
> José María Infanzón
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Re: [FFmpeg-user] YouTube streaming of a static image...

[DopeLabs]

there really isnt a easy way to loop audio or video (that i have found).. only 
images.. and im actually surprised they havent added a 'loop 1' option for 
audio or video files yet.. it must be among the top requested things to be 
added...

so instead we can loop (sort of) by using concat.

ffmpeg -f concat -safe 0 -i 'music.txt' -loop 1 -i 'image.jpg' -c:v libx264 
-preset ultrafast -c:a copy -f flv rtmp://rtmp.youtube.com/live/stream

the contents of the text file would look something like this

file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'
file '/path/to/music.mp3'

this would effectively loop the audio 10 times... add more lines if you need to.

ffmpeg by default streams video out at 25fps. you can save a bit of bandwidth 
if you specify a lower frame rate (though youtube might complain if its too 
low).

you should also be able to change the image in real time as ffmpeg is still 
streaming by replacing the image.jpg with a new one, but you must replace it 
atomically, or ffmpeg will most likely crash.

using the mv command instead of cp or overwriting...

mv newimage.jpg image.jpg

its also a good idea to make sure the image you are replacing with has the same 
attributes (size, etc)

cheers

DL

> On Jan 28, 2018, at 1:10 24AM, pigeon33  wrote:
> 
> Hello,
> 
> I'm trying to stream (on YouTube) an image...
> 
> This image is a PNG file on my computer...
> 
> The dimensions of my PNG file are 1280x720...
> 
> This image usually don't change but from times to times it might change and
> of course the stream must refresh/update it...
> 
> The goal is to stream this image on YouTube without any interruption (I
> guess with -stream_loop) even if there is some dropped frames during a
> modification of my image...
> 
> Moreover is it possible to add a background music (with a loop) to the
> stream?
> 
> Thank you in advance for your help,
> 
> Maxime.
> 
> 
> 
> --
> Sent from: http://www.ffmpeg-archive.org/
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Re: [FFmpeg-user] Reencoding video from image sequence doesn't work (sometimes)

[DopeLabs]

have you tried different players such as ffplay, mpv, IINA, mplayer?

have you tried a different encapsulation such as .mp4 or .flv or .mov?

looking at your console output..

keyint=250

maybe try and have those more often?

or maybe try and pad the beginning with a few black frames?

if it works 99% of the time does that mean videos seem to faile 1% of the time 
at random? or can you try again and it will work properly?

 
> On Jan 23, 2018, at 9:54 49AM, serial...@gmail.com wrote:
> 
> DopeLabs wrote
>> once you have your image sequence, encoding them into a movie should be
>> pretty straight forward.
> 
> And that's the rub.  It *is* straightforward, and what I'm trying to do
> works on 99% of my videos, but for some reason, there are a handful of
> videos where the picture DOES NOT START when the rest of the video does. 
> Audio starts, and subtitles might, but the picture DOES NOT.
> 
> Fastforwarding the video to some arbitrary point and then rewinding to NEAR
> the beginning works, but if you rewind ALL THE WAY to the beginning whatever
> WAS on the screen just stays there and never updates.
> 
> One of the things I just realized is playing the video all the way through
> the picture eventually comes in, and I get messages like this:
> 
> ...
>  Stream #1:1 -> #0:1 (aac (native) -> vorbis (libvorbis))
>  Stream #1:2 -> #0:2 (ass (ssa) -> ass (ssa))
>  Stream #1:3 -> #0:3 (copy)
> Press [q] to stop, [?] for help
> [image2 @ 002da3e0] Thread message queue blocking; consider raising
> the thread_queue_size option (current value: 8)
> [matroska @ 032254a0] Starting new cluster due to timestamp= 
> 93.4kbits/s speed=11.2x
> [matroska @ 032254a0] Starting new cluster due to timestamp=
> 963.1kbits/s speed=2.19x
> frame=26853 fps= 40 q=-1.0 Lsize=  179749kB time=00:23:20.00
> bitrate=1051.8kbits/s speed=2.06x
> video:162698kB audio:16217kB subtitle:21kB other streams:0kB global
> headers:6kB muxing overhead: 0.454852%
> [libx264 @ 031c98e0] frame I:638   Avg QP:18.35  size: 41747
> ...
> 
> What's up with the "Starting new cluster due to timestamp= ..." thing?  It
> doesn't record the actual timestamp in the output later, but I recall it did
> that saying it was at about 2 minutes, and at about 2 minutes in the picture
> appears (which doesn't seem to help on my Raspberry Pi at all).
> 
> Also, this problem only seems to happen on *some* players.  Since VLC is the
> only thing that really plays MKV files, it's the only thing I can use to
> test, but while VLC doesn't play these files from the beginning, Chrome
> seems to work just fine.  Again, my this doesn't help on my Pi.
> 
> Recently I've tried to not just copy everything except the picture, but
> reencode everything.  This didn't work either.
> 
> Thanks for any further help.
> 
> p.s. 
> Carl Eugen wrote
>> I assume you tested that this is better than FFmpeg's (old)
>> super2xsai filter. Is it also better than FFmpeg's hqx filter?
>> (You agree that faster alone doesn't really help if you have
>> to reencode twice, no?)
> 
> I checked this out after you recommended it, and it doesn't do nearly as
> good a job as my super-long 2+ hour encoding process makes my crappy
> TV-quality videos into damn-near HD-quality videos.  Thanks for the
> suggestion though. 
> 
> 
> 
> --
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Re: [FFmpeg-user] Reencoding video from image sequence doesn't work (sometimes)

[DopeLabs]

once you have your image sequence, encoding them into a movie should be pretty 
straight forward. i have done lots of work using image sequences as source 
material for audio visual loops, etc...

the following is assuming your image files are named in an ascending sequence 
and there are no missing or skipped numbers, are jpg's, and is 30 fps


$ ffmpeg -pattern_type glob -r 30 -i '*.jpg' -c:v h264 -preset ultrafast 
-pix_fmt yuv420p -movflags +faststart output.mp4




> On Jan 18, 2018, at 1:43 51PM, serial...@gmail.com wrote:
> 
> DopeLabs wrote
>> just with a quick glance.. -c copy means stream copy.. as in not to
>> re-encode.. you should use -c copy OR -vcodec h264(-c:v h264), not
>> both...i bet ffmpeg can do whatever upconvert that utility is doing
>> without the need to go with a sequence first.
> 
> True, but I'm trying to get a specific codec (h264) so that it will decode
> on my Raspberry Pi properly, and I want to keep everything else the same,
> that's why the -c copy.
> 
> The reason I need to do the image sequence is because the image upconverter
> *isn't* in ffmpeg, and can't do what it does.  I'm not just making the video
> bigger in size, I'm increasing the quality, and making it better than it
> was.  There are some good example images on the main projects website here:
> https://github.com/nagadomi/waifu2x It uses a Deep NN to render the images,
> and it takes a bit to render a few hundred images, even using a GPU.
> 
> Either way, video -> image sequence -> video doesn't work, without doing any
> kind of conversion in between.  That is my problem.
> 
> 
> 
> --
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Re: [FFmpeg-user] Reencoding video from image sequence doesn't work (sometimes)

[DopeLabs]

just with a quick glance.. -c copy means stream copy.. as in not to re-encode.. 
you should use -c copy OR -vcodec h264(-c:v h264), not both...i bet ffmpeg can 
do whatever upconvert that utility is doing without the need to go with a 
sequence first.

check out the ffmpeg docs for functions, filters, utilities, scalers, etc ,etc

 

> On Jan 18, 2018, at 12:53 48PM, serial...@gmail.com wrote:
> 
> First of all, I've exhausted all my Google-fu to no avail... so any help is
> greatly appreciated!
> 
> I found a cool utility to upconvert anime images
> (if you're interested https://github.com/lltcggie/waifu2x-caffe) and so I'm
> reencoding my low quality
> anime using this.  First I rip to an image sequence using a command like:
> 
> .\ffmpeg.exe -i full-20s.mkv -f image2 "file\seq-%09d.png"
> 
> I also get some metadata, such as framerate and stuff using ffprobe
> Then I do the upconvert of the image sequence, and then I put it back
> together 
> 
> .\ffmpeg.exe -y -i "full-20s.mkv" -framerate $framerate -f image2 -i
> "file\seq-%09d.png" -map 1:v -c copy -vcodec h264 -pix_fmt yuv420p -map 0
> -map -0:v out.mkv
> 
> This works beautifully for most files, but for some files this breaks in a
> weird way.  The file will encode
> okay, and I can then play it, but I get no video.  If I then seek to a
> random place, pause the video and
> then seek to near the beginning, it works perfectly.
> 
> I've uploaded some example files here:
> https://drive.google.com/open?id=1qog43RZ8hPfQIjc6PbGi1Ii_ugeFNrQO
> (oddly, playing the bad "out.mkv" file from google drive works fine, but vlc
> and other media
> players it doesn't -_- )
> 
> I've already tried making a keyframe at the beginning with adding
> `-force_key_frames 0.0` to my
> command line, and making a video out of the image sequence (which then plays
> fine!!!) and adding it to
> the audio later.  I've done a lot of research, and I'm at a loss.
> 
> "full-20s.mkv" was taken using ./ffmpeg -i full-episode.mkv -t 20 -c copy
> -map 0 full-20s.mkv
> 
> I then made out.mkv using the following 2 commands:
> 
> D:\programmin\Projects\reencode> & .\ffmpeg.exe -i full-20s.mkv -f image2
> ".\inp-full-20s.mkv\seq-%09d.png"
> ffmpeg version N-79690-g78baa45 Copyright (c) 2000-2016 the FFmpeg
> developers
>  built with gcc 5.3.0 (GCC)
>  configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r
> --enable-gnutls --enable-iconv --enable-libass --enable-libblu
> ray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme
> --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx
> --enable-libmp3lame --enable-libopencore-amrnb --enable-
> libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp
> --enable-libschroedinger --enable-libsnappy --enable-libsoxr
> --enable-libspeex --enable-libtheora --enable-libtwolame --enable-
> libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxavs --enable-libxvid --enable-libzimg --enable
> -lzma --enable-decklink --enable-zlib
>  libavutil  55. 23.100 / 55. 23.100
>  libavcodec 57. 38.100 / 57. 38.100
>  libavformat57. 34.103 / 57. 34.103
>  libavdevice57.  0.101 / 57.  0.101
>  libavfilter 6. 44.100 /  6. 44.100
>  libswscale  4.  1.100 /  4.  1.100
>  libswresample   2.  0.101 /  2.  0.101
>  libpostproc54.  0.100 / 54.  0.100
> The tags at index 3 refer to a non-existent track 4.
> The tags at index 7 refer to a non-existent track 4.
> Input #0, matroska,webm, from
> 'D:\programmin\Projects\reencode\full-20s.mkv':
>  Metadata:
>ENCODER : Lavf57.34.103
>  Duration: 00:00:20.12, start: 0.00, bitrate: 640 kb/s
>Stream #0:0: Video: h264 (High), yuv420p, 848x480, SAR 1:1 DAR 53:30,
> 19.18 fps, 19.18 tbr, 1k tbn (default)
>Metadata:
>  DURATION: 00:00:20.12400
>Stream #0:1(jpn): Audio: aac (HE-AAC), 48000 Hz, stereo, fltp (default)
>Metadata:
>  DURATION: 00:00:20.01000
>Stream #0:2(eng): Subtitle: ass (default)
>Metadata:
>  DURATION: 00:00:00.0
>Stream #0:3: Attachment: ttf
>Metadata:
>  filename: RotisWTF.ttf
>  mimetype: application/x-truetype-font
> [image2 @ 004ef9c0] Using AVStream.codec to pass codec parameters to
> muxers is deprecated, use AVStream.codecpar instead.
> Output #0, image2, to '.\inp-full-20s.mkv\seq-%09d.png':
>  Metadata:
>encoder : Lavf57.34.103
>Stream #0:0: Video: png, rgb24, 848x480 [SAR 1:1 DAR 53:30], q=2-31, 200
> kb/s, 19.18 fps, 19.18 tbn (default)
>Metadata:
>  DURATION: 00:00:20.12400
>  encoder : Lavc57.38.100 png
> Stream mapping:
>  Stream #0:0 -> #0:0 (h264 (native) -> png (native))
> Press [q] to stop, [?] for help
> frame=  386 fps=150 q=-0.0 Lsize=N/A time=00:00:20.12 bitr

Re: [FFmpeg-user] concat one video file and multiple audio files

[DopeLabs]

you can chain a few commands into a single '1 liner' if youd like...

for f in 1.mp3 2.mp3 3.mp3 4.mp3; do echo "file $f" >> concat.txt; done && 
ffmpeg -f concat -i 'concat.txt' -i video1.mp4 -c copy videoout.mp4

technically 2 commands... but this is basically scripting, which falls outside 
the scope of this mailing list.

cheers

> On Jan 12, 2018, at 6:43 33AM, ropiafoldetnezi  
> wrote:
> 
> hi,
> 
> thank you all!
> 
> @dopelabs method worked for me best, and as i will make many videos the
> speed boost is also welcome, thank you!
> 
> as i would like to create videos dynamicly (and generate ffmpeg commands
> also), it would be easier for me to have all input in one command (instead
> of list file), is it possibel?
> 
> thx
> 
> 
> On Fri, Jan 12, 2018 at 11:18 AM, DopeLabs  wrote:
> 
>> you can use concat like this
>> 
>> ffmpeg -f concat -i 'textfile.txt' -i video1.mp4 -c copy videoout.mp4
>> 
>> the text file contains the list of audio files you want to concat.
>> 
>> 
>> file 'f1.mp3'
>> file 'n1.mp3'
>> file 'f2.mp3'
>> file 'm1.mp3'
>> file 'f3.mp3'
>> file 'v1.mp3'
>> 
>> as long as all your audio files are encoded at the same bitrate,
>> samplerate, etc, you shouldnt have any problems.
>> 
>> this also doesnt require a re-encode, you can stream copy which will be
>> much much faster. anything you send through a filter complex usually will
>> need to be re-encoded.
>> 
>> 
>> hope this helps =]
>> 
>> 
>> 
>>> On Jan 11, 2018, at 9:43 50PM, ropiafoldetnezi <
>> ropiafoldetn...@gmail.com> wrote:
>>> 
>>> Thank you!
>>> 
>>> Is it possible to do the concat and the map in one step (one command)?
>>> 
>>> 
>>> On Jan 11, 2018 11:11 PM, "Moritz Barsnick"  wrote:
>>> 
>>> On Thu, Jan 11, 2018 at 22:57:11 +0100, ropiafoldetnezi wrote:
>>>> multiple sources:
>>>> 1 video file (.mp4)
>>>> 6 separate audio files (.mps)
>>> [...]
>>>> the command i tried (on windows 10):
>>>> ffmpeg -i video1.mp4 -i f1.mp3 -i n1.mp3 -i f2.mp3 -i m1.mp3 -i f3.mp3
>> -i
>>>> v1.mp3 -filter_complex
>>>> "[0:v:0][1:a:0][2:a:0][3:a:0][4:a:0][5:a:0][6:a:0]concat=n=
>>> 7:v=1:a=1[outv][outa]"
>>>> -map "[outv]" -map "[outa]" output/v1_n1_m1_v1_t_02.mp4
>>> 
>>> You should be concat'ing the audio separately from the video, and then
>>> mapping video and "[outa]" separately.
>>> 
>>> You can't concat an audio stream to a video.
>>> 
>>> Moritz
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>> 
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> 
> 
> 
> -- 
> 
> üdv
> bal
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Re: [FFmpeg-user] concat one video file and multiple audio files

[DopeLabs]

you can use concat like this

ffmpeg -f concat -i 'textfile.txt' -i video1.mp4 -c copy videoout.mp4

the text file contains the list of audio files you want to concat.


file 'f1.mp3'
file 'n1.mp3'
file 'f2.mp3'
file 'm1.mp3'
file 'f3.mp3'
file 'v1.mp3'

as long as all your audio files are encoded at the same bitrate, samplerate, 
etc, you shouldnt have any problems.

this also doesnt require a re-encode, you can stream copy which will be much 
much faster. anything you send through a filter complex usually will need to be 
re-encoded.


hope this helps =]



> On Jan 11, 2018, at 9:43 50PM, ropiafoldetnezi  
> wrote:
> 
> Thank you!
> 
> Is it possible to do the concat and the map in one step (one command)?
> 
> 
> On Jan 11, 2018 11:11 PM, "Moritz Barsnick"  wrote:
> 
> On Thu, Jan 11, 2018 at 22:57:11 +0100, ropiafoldetnezi wrote:
>> multiple sources:
>> 1 video file (.mp4)
>> 6 separate audio files (.mps)
> [...]
>> the command i tried (on windows 10):
>> ffmpeg -i video1.mp4 -i f1.mp3 -i n1.mp3 -i f2.mp3 -i m1.mp3 -i f3.mp3 -i
>> v1.mp3 -filter_complex
>> "[0:v:0][1:a:0][2:a:0][3:a:0][4:a:0][5:a:0][6:a:0]concat=n=
> 7:v=1:a=1[outv][outa]"
>> -map "[outv]" -map "[outa]" output/v1_n1_m1_v1_t_02.mp4
> 
> You should be concat'ing the audio separately from the video, and then
> mapping video and "[outa]" separately.
> 
> You can't concat an audio stream to a video.
> 
> Moritz
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Re: [FFmpeg-user] AVC3 -> AVC1 conversion

[DopeLabs]

the same thing happens by default when using ffmpeg to encode hevc/x265

ffmpeg creates an hevc format mp4 with a default fourcc of hev1, which osx will 
not play via quicktime, finder/quicklook. preview thumbnails/file icons also do 
not generate in finder.

changing the fourcc to hvc1 will make it work across osx in the above mentioned 
areas...

here is the command i used. you may apply the same methods with x264 with 
-tag:v avc1

$ ffmpeg -i x265.mp4 -c copy -tag:v hvc1 hvc1.mp4
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'x265.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2mp41
encoder : Lavf57.76.100
  Duration: 00:41:50.01, start: 0.00, bitrate: 348 kb/s
Stream #0:0(und): Video: hevc (Main) (hev1 / 0x31766568), yuv420p(tv, 
progressive), 1920x1080, 211 kb/s, 30 fps, 30 tbr, 15360 tbn, 30 tbc (default)
Metadata:
  handler_name: VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 128 kb/s (default)
Metadata:
  handler_name: SoundHandler
Output #0, mp4, to 'hvc1.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2mp41
encoder : Lavf58.2.100
Stream #0:0(und): Video: hevc (Main) (hvc1 / 0x31637668), yuv420p(tv, 
progressive), 1920x1080, q=2-31, 211 kb/s, 30 fps, 30 tbr, 15360 tbn, 15360 tbc 
(default)
Metadata:
  handler_name: VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 128 kb/s (default)
Metadata:
  handler_name: SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=75300 fps=14676 q=-1.0 Lsize=  106784kB time=00:41:49.98 bitrate= 
348.5kbits/s speed= 489x
video:64898kB audio:39220kB subtitle:0kB other streams:0kB global headers:2kB 
muxing overhead: 2.560526%



> On Nov 27, 2017, at 2:26 54PM, Carl Eugen Hoyos  wrote:
> 
> 2017-11-26 23:27 GMT+01:00 antony baxter :
> 
>> I can obviously convert them into H264 avc1 the long way, but
>> wondered if there was a quick way of simply changing the
>> metadata of the avc3 stream into avc1?
> 
> Of course:
> Use the hexeditor of your choice, search for "avc3" (it is either
> on top or at the end of the file) and change it.
> 
> I am not sure if QT (or WMP) will play the resulting file though,
> FFmpeg-based players will.
> 
> As Moritz explained: FFmpeg is not a file editor.
> 
> Carl Eugen
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Re: [FFmpeg-user] Creat mosaic without loose quality

[DopeLabs]

> 
> On Nov 23, 2017, at 4:23 50AM, Carl Eugen Hoyos  wrote:
> 
> 2017-11-23 12:52 GMT+01:00 Marco De Angelis 
> :
> 
>> C:\S7\ffmpeg\bin\ffmpeg -i C:\S7\FTP_video\1.mp4 -i C:\S7\FTP_video\2.mp4
>> -i C:\S7\FTP_video\3.mp4 -i C:\S7\FTP_video\4.mp4
> 
>> -filter_complex "nullsrc=size=1920x1080
> 
> This is your original source, the other videos are overlayed on top of it.
> You can specify a frame-rate for the original source, in your case either
> nullsrc=s=hd1080:r=60 or nullsrc=s=hd1080:r=60001/1001, your
> input videos have different frame rates afaict.
> 
> It may be easier to use the vstack and hstack filters instead of
> overlay.
> 
> Please do not top-post here, Carl Eugen
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you may also want to check out the tile filter


tile=2x2,scale=1920x1080
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Re: [FFmpeg-user] recording @ 120fps on mac fails with 'activeVideoMinFrameDuration not supported'

[DopeLabs]

what you are seeing is from the status line during the encode... this shows you 
the current encoding frame rate. if its below  the specified framerate, the 
'speed' value at the end of that line will be less than 1.

basically your current system hardware, and current ffmpeg encoding options can 
not encode in 'real time'.

if you press q to stop the encode, and then use ffprobe on the file, you will 
notice that it encoded a 120fps file.


 $ ffmpeg -y -f avfoundation -r 120 -pixel_format yuyv422 -i "1:" -s 320x240 
out.avi
ffmpeg version N-86825-g99c5ac2039 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.1.0 (clang-802.0.42)
  configuration: --enable-shared --enable-pthreads --cc=/usr/bin/clang 
--enable-vda --enable-videotoolbox --arch=x86_64 --enable-yasm --enable-gpl 
--enable-postproc --enable-libx264 --enable-libxvid --enable-filter=coreimage 
--enable-filter=coreimagesrc --enable-filter=frei0r --enable-filter=frei0r_src 
--enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame 
--enable-libvorbis --enable-libopus --enable-libtheora --enable-libopenjpeg 
--enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex 
--enable-libass --enable-lzma --enable-gnutls --enable-fontconfig 
--enable-libfreetype --enable-libfribidi --enable-libx265 --enable-version3 
--enable-libwavpack --enable-libssh --enable-librtmp --enable-libfdk-aac 
--enable-nonfree --enable-libzmq --enable-opencl --enable-ffplay 
--enable-libcaca
  libavutil  55. 68.100 / 55. 68.100
  libavcodec 57.102.100 / 57.102.100
  libavformat57. 76.100 / 57. 76.100
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 95.100 /  6. 95.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
[avfoundation @ 0x7fcd75806a00] Stream #0: not enough frames to estimate rate; 
consider increasing probesize
Input #0, avfoundation, from '1:':
  Duration: N/A, start: 33968.23, bitrate: N/A
Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 1920x1080, 1000k 
tbr, 1000k tbn, 1000k tbc
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg4 (native))
Press [q] to stop, [?] for help
Output #0, avi, to 'out.avi':
  Metadata:
ISFT: Lavf57.76.100
Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p, 320x240, q=2-31, 
200 kb/s, 120 fps, 120 tbn, 120 tbc
Metadata:
  encoder : Lavc57.102.100 mpeg4
Side data:
  cpb: bitrate max/min/avg: 0/0/20 buffer size: 0 vbv_delay: -1
frame= 1115 fps= 90 q=31.0 Lsize= 650kB time=00:00:09.29 bitrate= 
573.2kbits/s speed=0.746x
video:618kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 5.142043%



[30][09:44:53] dopelabs@jeez:~ $ ffprobe out.avi
ffprobe version N-86825-g99c5ac2039 Copyright (c) 2007-2017 the FFmpeg 
developers
  built with Apple LLVM version 8.1.0 (clang-802.0.42)
  configuration: --enable-shared --enable-pthreads --cc=/usr/bin/clang 
--enable-vda --enable-videotoolbox --arch=x86_64 --enable-yasm --enable-gpl 
--enable-postproc --enable-libx264 --enable-libxvid --enable-filter=coreimage 
--enable-filter=coreimagesrc --enable-filter=frei0r --enable-filter=frei0r_src 
--enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame 
--enable-libvorbis --enable-libopus --enable-libtheora --enable-libopenjpeg 
--enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex 
--enable-libass --enable-lzma --enable-gnutls --enable-fontconfig 
--enable-libfreetype --enable-libfribidi --enable-libx265 --enable-version3 
--enable-libwavpack --enable-libssh --enable-librtmp --enable-libfdk-aac 
--enable-nonfree --enable-libzmq --enable-opencl --enable-ffplay 
--enable-libcaca
  libavutil  55. 68.100 / 55. 68.100
  libavcodec 57.102.100 / 57.102.100
  libavformat57. 76.100 / 57. 76.100
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 95.100 /  6. 95.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Input #0, avi, from 'out.avi':
  Metadata:
encoder : Lavf57.76.100
  Duration: 00:00:09.29, start: 0.00, bitrate: 573 kb/s
Stream #0:0: Video: mpeg4 (Simple Profile) (FMP4 / 0x34504D46), yuv420p, 
320x240 [SAR 1:1 DAR 4:3], 545 kb/s, 120 fps, 120 tbr, 120 tbn, 120 tbc
 

> On Nov 14, 2017, at 7:58 03PM, Roman Puttkammer  wrote:
> 
> I did run into another issue; though the frame rate 320/240 @ 120fps is set
> successfully, ffmpeg reports only 30+ fps later on. (It starts off at 50,
> then drops down to 30+ shortly thereafter.) This seems less straightforward
> - unless I'm not using the command line options right (or am
> misunderstanding how the frame rate works.)
&g

Re: [FFmpeg-user] OT: How to download a video

[DopeLabs]

first get the download url using youtube-dl


$ youtube-dl -f best -g 
http://www.pbs.org/video/wfyi-local-productions-temple-makers/
https://ga.video.cdn.pbs.org/videos/wfyi-local-productions/2e9d36ba-d5f6-4d45-84df-b5ca5acd551c/250267/hd-mezzanine-16x9/2c382920_l-4055-16x9-mp4-2500k.mp4

then change https to http

$ youtube-dl 
http://ga.video.cdn.pbs.org/videos/wfyi-local-productions/2e9d36ba-d5f6-4d45-84df-b5ca5acd551c/250267/hd-mezzanine-16x9/2c382920_l-4055-16x9-mp4-2500k.mp4
[generic] 2c382920_l-4055-16x9-mp4-2500k: Requesting header
[download] Destination: 
2c382920_l-4055-16x9-mp4-2500k-2c382920_l-4055-16x9-mp4-2500k.mp4
Initializing download: 
http://ga.video.cdn.pbs.org/videos/wfyi-local-productions/2e9d36ba-d5f6-4d45-84df-b5ca5acd551c/250267/hd-mezzanine-16x9/2c382920_l-4055-16x9-mp4-2500k.mp4
File size: 1077791195 bytes
Opening output file 
2c382920_l-4055-16x9-mp4-2500k-2c382920_l-4055-16x9-mp4-2500k.mp4.part
Starting download
Downloaded 1.0 Gigabyte in 17 seconds. (60486.33 KB/s)
[axel] Downloaded 1077791195 bytes
[download] 100% of 1.00GiB


> On Nov 4, 2017, at 11:21 07AM, JD  wrote:
> 
> I am using firefox and Inspect element is not working :(
> Also, I did download the page source, and grep'ed the file for mp4
> and found none :(
> 
> 
> On 11/04/2017 11:51 AM, Moritz Barsnick wrote:
>> On Sat, Nov 04, 2017 at 11:26:42 -0600, JD wrote:
>>> Could someone please tell me a linux tool that will download the video at
>>> http://www.pbs.org/video/wfyi-local-productions-temple-makers/
>>> 
>>> Youtube-dl could not do it.
>> Remember to always use the latest version of youtube-dl, as streaming
>> websites are "moving targets":
>> 
>> $ ./youtube-dl http://www.pbs.org/video/wfyi-local-productions-temple-makers/
>> [pbs] Downloading JSON metadata
>> [pbs] wfyi-local-productions-temple-makers: Downloading webpage
>> [pbs] 2365795021: Downloading widget/partnerplayer page
>> [pbs] 2365795021: Downloading portalplayer page
>> [pbs] 2365795021: Downloading hls-400-2500k-16x9 video url info
>> [pbs] 2365795021: Downloading m3u8 information
>> [pbs] 2365795021: Downloading mp4-2500k-16x9 video url info
>> [pbs] 2365795021: Downloading hls-800k-16x9 video url info
>> [pbs] 2365795021: Downloading m3u8 information
>> [pbs] 2365795021: Downloading mp4-baseline-16x9 video url info
>> [pbs] 2365795021: Checking http-1200k video URL
>> [pbs] 2365795021: Checking http-2500k video URL
>> [pbs] 2365795021: Checking http-800k video URL
>> [pbs] 2365795021: Checking http-400k video URL
>> [download] Destination: The Temple Makers - The Temple Makers-2365795021.mp4
>> [download]   0.8% of 1.00GiB at  1.62MiB/s ETA 10:27^C
>> ERROR: Interrupted by user
>> $
>> 
>> Works just fine!
>> 
>> A little hint: One video URL is directly in the source which my browser
>> gets, I just right-click in the video, select "Inspect element"
>> (Firefox, Chrome(ium), Opera), and see the URL
>> https://ga.video.cdn.pbs.org/videos/wfyi-local-productions/2e9d36ba-d5f6-4d45-84df-b5ca5acd551c/250267/hd-mezzanine-16x9/2c382920_l-4055-16x9-mp4-baseline.mp4
>> which is downloadable (albeit a low quality version, I think).
>> 
>> Cheers,
>> Moritz
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Re: [FFmpeg-user] Latest GIT version doesn't compile with libvpx_vp9 encoder

[DopeLabs]

i didnt think you would need to specify 

> --enable-encoder=libvpx_vp8
> --enable-encoder=libvpx_vp9

it has always worked for me using only 

--enable-libvpx

same goes for
>  --enable-encoder=libopus
> --enable-libopus

you only need

--enable-libopus


but i will also point out you have declared 

> --disable-encoders 

with no other arguments ..



> On Oct 14, 2017, at 12:31 42PM, ZikZak  wrote:
> 
> Hello,
> 
> I freshly recompiled libvpx (for both VP8 and VP9) then FFMPEG and
> strangely the result only allows me libvpx_vp8 as encoder despite a
> configure with enable-encoder=libvpx_vp8 and enable-encoder=libvpx_vp9
> 
> vpxenc's help lists VP9 as a possible encoder
> 
> Configuration for libvpx:
> ./configure --disable-unit-tests --disable-docs --enable-vp8 --enable-vp9
> --enable-postproc --enable-vp9-postproc --enable-vp9-temporal-denoising
> --enable-runtime-cpu-detect
> 
> Configuration for ffmpeg:
> ./configure --enable-nonfree --disable-doc --enable-libvpx
> --prefix=/usr/local --disable-encoders --enable-encoder=libvpx_vp8
> --enable-encoder=libvpx_vp9 --disable-ffserver --enable-gpl
> --enable-encoder=png --enable-encoder=libopus --enable-encoder=webvtt
> --enable-libopus --enable-ffplay --enable-ffprobe
> 
> It is on an ARM platform, if it matters.
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Re: [FFmpeg-user] feature proposal - avoid overwriting the input file

[DopeLabs]

ffmpeg commands are all structured in a specific and standardized way

a few bits from the description on the man page

ffmpeg [global_options] {[input_file_options] -i input_url} ... 
{[output_file_options] output_url} ...

ffmpeg reads from an arbitrary number of input "files" (which can be regular 
files, pipes, network streams, grabbing devices, etc.), specified by the "-i" 
option, and writes to an arbitrary number of output
"files", which are specified by a plain output url. Anything found on the 
command line which cannot be interpreted as an option is considered to be an 
output url.

Each input or output url can, in principle, contain any number of streams of 
different types (video/audio/subtitle/attachment/data). The allowed number 
and/or types of streams may be limited by the container
format. Selecting which streams from which inputs will go into which output is 
either done automatically or with the "-map" option (see the Stream selection 
chapter).

To refer to input files in options, you must use their indices (0-based). E.g.  
the first input file is 0, the second is 1, etc. Similarly, streams within a 
file are referred to by their indices. E.g. "2:3"
refers to the fourth stream in the third input file. Also see the Stream 
specifiers chapter.

As a general rule, options are applied to the next specified file. Therefore, 
order is important, and you can have the same option on the command line 
multiple times. Each occurrence is then applied to the
next input or output file.  Exceptions from this rule are the global options 
(e.g. verbosity level), which should be specified first.

Do not mix input and output files -- first specify all input files, then all 
output files. Also do not mix options which belong to different files. All 
options apply ONLY to the next input or output file and
are reset between files.


in your example 

> $ ffmpeg -i input.mp4 [...] -yo output.mov

ffmpeg -i input.mp4 [..] output.mov

works just fine no need to add -y because .mov is a different filename

> $ ffmpeg -yo output.mov -i input.mp4 [...]

is incorrect structure for ffmpeg...

> The order shouldn't overly matter

it matters... see above



> On Oct 11, 2017, at 8:39 06AM, Douglas Marsh  wrote:
> 
> On 2017-10-10 18:18, Mikhail V wrote:
>> James Girotti wrote:
>> Sure, I am. But anybody can miss the typo, and its not so easy to
>> notice in console or text editor, its just two chars
>> in a pile of chars.
> 
> 
> Perhaps then add a new switch.. something *like*
> 
> $ ffmpeg -i input.mp4 [...] -yo output.mov
> 
> 
> (for "Yes Output/Overwrite" where the parameter that MUST come next is the 
> filename)
> 
> and the following example would NOT work:
> 
> $ ffmpeg -yo -i input/mp4 [...] output.mov
> 
> as -yo has no parameter [the filename to overwrite] and assuming "output.mov" 
> exist so it would not overwrite without prompt. The order shouldn't overly 
> matter as the following COULD work:
> 
> $ ffmpeg -yo output.mov -i input.mp4 [...]
> 
> 
> Just an idea.
> 
> 
> --Doug (dx9s)
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Re: [FFmpeg-user] Audio filter: removesilence doesn't appear to do anything

[DopeLabs]

i created an audio file in ableton live which plays 10 seconds of audio, then 
is silent for 5 seconds and so on for ~5 mins, exported as an aif

ran the ffmpeg silenceremove from your example (except i removed the :1 after 
the -20dB

$ ffmpeg -i silencetest.aif -af silenceremove=0:0:0:-1:0.5:-20dB removed.wav

ffmpeg version N-86825-g99c5ac2039 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.1.0 (clang-802.0.42)
  configuration: --enable-shared --enable-pthreads --cc=/usr/bin/clang 
--enable-vda --enable-videotoolbox --arch=x86_64 --enable-yasm --enable-gpl 
--enable-postproc --enable-libx264 --enable-libxvid --enable-filter=coreimage 
--enable-filter=coreimagesrc --enable-filter=frei0r --enable-filter=frei0r_src 
--enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame 
--enable-libvorbis --enable-libopus --enable-libtheora --enable-libopenjpeg 
--enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex 
--enable-libass --enable-lzma --enable-gnutls --enable-fontconfig 
--enable-libfreetype --enable-libfribidi --enable-libx265 --enable-version3 
--enable-libwavpack --enable-libssh --enable-librtmp --enable-libfdk-aac 
--enable-nonfree --enable-libzmq --enable-opencl --enable-ffplay 
--enable-libcaca
  libavutil  55. 68.100 / 55. 68.100
  libavcodec 57.102.100 / 57.102.100
  libavformat57. 76.100 / 57. 76.100
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 95.100 /  6. 95.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, aiff, from 'silencetest.aif':
  Duration: 00:04:54.19, start: 0.00, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16be, 44100 Hz, stereo, s16, 1411 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16be (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to 'removed.wav':
  Metadata:
ISFT: Lavf57.76.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, 
s16, 1411 kb/s
Metadata:
  encoder : Lavc57.102.100 pcm_s16le
size=   31661kB time=00:03:03.79 bitrate=1411.2kbits/s speed= 418x
video:0kB audio:31661kB subtitle:0kB other streams:0kB global headers:0kB 
muxing overhead: 0.000241%


as you can see the output file is shorter than the input.

here is what the wav forms look like for a visual comparison

http://run.florist.run/images/silencetest.png




> On Sep 25, 2017, at 2:39 22AM, Quinn Wood  wrote:
> 
> I have removed the video and am getting the same results.
> silenceremove is not removing silence.
> 
>  ffmpeg -i ../2017-09-24T123913.mp4.done -vn -af
> silenceremove=0:0:0:-1:0.5:-20dB:1 output.m4a
> 
> https://pastebin.com/raw/xcGQFs30
> 
>  ffprobe -i output.m4a
> 
> https://pastebin.com/raw/183KLwZY
> 
>  ffmpeg -i output.m4a -af silencedetect=n=-20dB:d=0.5 -f null -
> 
> https://pastebin.com/raw/4304SzMh
> 
> Using peak detection makes no difference:
> https://pastebin.com/raw/KwqXudSw https://pastebin.com/raw/462yRwQ7
> https://pastebin.com/raw/s362n0wd
> 
> On Mon, Sep 25, 2017 at 1:45 AM, Quinn Wood  wrote:
>> I thought I'd extracted the video by simply using the m4a extension-
>> that's not the way that works and I should have noticed that. I bet
>> the audio has been truncated but the video remains the full length.
>> Let me strip out the video and try it.
>> 
>> On Mon, Sep 25, 2017 at 12:59 AM, Quinn Wood  wrote:
>>> I have used the following command
>>>ffmpeg -i input.m4a -af silenceremove=0:0:0:-1:0.5:-20dB output.m4a
>>> 
>>> on a file. When I use another tool (Audacity) to truncate periods over
>>> 0.5 seconds of under -20dB volume it removes upwards of an hour from
>>> the input file. When I run this command ffmpeg doesn't remove a single
>>> second. I have tried numerous variations of the command and numerous
>>> input files, and am simply unable to replicate the effect Audacity is
>>> producing.
>>> 
>>> The input files have a period of silence at the beginning that is
>>> predictable, intermittent periods of silence throughout the file of
>>> unknown quantity and length, and a period of silence at the end that
>>> is predictable.
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Re: [FFmpeg-user] Audio filter: removesilence doesn't appear to do anything

[DopeLabs]

in your first output paste you have

 ffmpeg -i input.m4a -af silenceremove=1:5:-20dB output.m4a

looking at the man page the example says

Trim all silence encountered from beginning to end where there is more than 1 
second of silence in audio:

   silenceremove=0:0:0:-1:1:-90dB



so you might try swapping your values in place and running that against the 
audio file...

   silenceremove=0:0:0:-1:0.5:-20dB



> On Sep 24, 2017, at 11:45 35PM, Quinn Wood  wrote:
> 
> I thought I'd extracted the video by simply using the m4a extension-
> that's not the way that works and I should have noticed that. I bet
> the audio has been truncated but the video remains the full length.
> Let me strip out the video and try it.
> 
> On Mon, Sep 25, 2017 at 12:59 AM, Quinn Wood  wrote:
>> I have used the following command
>>ffmpeg -i input.m4a -af silenceremove=0:0:0:-1:0.5:-20dB output.m4a
>> 
>> on a file. When I use another tool (Audacity) to truncate periods over
>> 0.5 seconds of under -20dB volume it removes upwards of an hour from
>> the input file. When I run this command ffmpeg doesn't remove a single
>> second. I have tried numerous variations of the command and numerous
>> input files, and am simply unable to replicate the effect Audacity is
>> producing.
>> 
>> The input files have a period of silence at the beginning that is
>> predictable, intermittent periods of silence throughout the file of
>> unknown quantity and length, and a period of silence at the end that
>> is predictable.
> ___
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> 
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Re: [FFmpeg-user] Audio filter: removesilence doesn't appear to do anything

[DopeLabs]

well the only thing that looks odd to me is you have an h264 video stream in 
your 'mpeg 4 audio' file. (.m4a)

have you tried extracting the audio or converting to uncompressed and running 
the same command?



> On Sep 24, 2017, at 11:23 44PM, Quinn Wood  wrote:
> 
> https://pastebin.com/raw/KGVTRVsR
> 
> On Mon, Sep 25, 2017 at 1:19 AM, DopeLabs  wrote:
>> please provide the complete un-truncated console output
>> 
>> 
>>> On Sep 24, 2017, at 10:59 13PM, Quinn Wood  wrote:
>>> 
>>> I have used the following command
>>>   ffmpeg -i input.m4a -af silenceremove=0:0:0:-1:0.5:-20dB output.m4a
>>> 
>>> on a file. When I use another tool (Audacity) to truncate periods over
>>> 0.5 seconds of under -20dB volume it removes upwards of an hour from
>>> the input file. When I run this command ffmpeg doesn't remove a single
>>> second. I have tried numerous variations of the command and numerous
>>> input files, and am simply unable to replicate the effect Audacity is
>>> producing.
>>> 
>>> The input files have a period of silence at the beginning that is
>>> predictable, intermittent periods of silence throughout the file of
>>> unknown quantity and length, and a period of silence at the end that
>>> is predictable.
>>> ___
>>> ffmpeg-user mailing list
>>> ffmpeg-user@ffmpeg.org
>>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>>> 
>>> To unsubscribe, visit link above, or email
>>> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>> 
>> ___
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>> ffmpeg-user@ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
>> 
>> To unsubscribe, visit link above, or email
>> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
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> 
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Re: [FFmpeg-user] Audio filter: removesilence doesn't appear to do anything

[DopeLabs]

please provide the complete un-truncated console output


> On Sep 24, 2017, at 10:59 13PM, Quinn Wood  wrote:
> 
> I have used the following command
>ffmpeg -i input.m4a -af silenceremove=0:0:0:-1:0.5:-20dB output.m4a
> 
> on a file. When I use another tool (Audacity) to truncate periods over
> 0.5 seconds of under -20dB volume it removes upwards of an hour from
> the input file. When I run this command ffmpeg doesn't remove a single
> second. I have tried numerous variations of the command and numerous
> input files, and am simply unable to replicate the effect Audacity is
> producing.
> 
> The input files have a period of silence at the beginning that is
> predictable, intermittent periods of silence throughout the file of
> unknown quantity and length, and a period of silence at the end that
> is predictable.
> ___
> ffmpeg-user mailing list
> ffmpeg-user@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
> 
> To unsubscribe, visit link above, or email
> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".

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Re: [FFmpeg-user] Create transparent movie with subtitles

[DopeLabs]

i think you were close with the transparent png, you just used png as a video 
codec.

try something like -c:v h264

i dont have a subtitles file so i used drawtext filter in its place and seems 
to work as expected.


$ ffmpeg -loop 1 -i alpha1080.png -vf drawtext=text="some 
text":x=10:y=10:fontcolor="white":fontsize=72 -t 00:00:10.000 -c:v h264 -f mp4 
test.mp4


ffmpeg version N-86825-g99c5ac2039 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.1.0 (clang-802.0.42)
  configuration: --enable-shared --enable-pthreads --cc=/usr/bin/clang 
--enable-vda --enable-videotoolbox --arch=x86_64 --enable-yasm --enable-gpl 
--enable-postproc --enable-libx264 --enable-libxvid --enable-filter=coreimage 
--enable-filter=coreimagesrc --enable-filter=frei0r --enable-filter=frei0r_src 
--enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame 
--enable-libvorbis --enable-libopus --enable-libtheora --enable-libopenjpeg 
--enable-libmodplug --enable-libvpx --enable-libsoxr --enable-libspeex 
--enable-libass --enable-lzma --enable-gnutls --enable-fontconfig 
--enable-libfreetype --enable-libfribidi --enable-libx265 --enable-version3 
--enable-libwavpack --enable-libssh --enable-librtmp --enable-libfdk-aac 
--enable-nonfree --enable-libzmq --enable-opencl --enable-ffplay 
--enable-libcaca
  libavutil  55. 68.100 / 55. 68.100
  libavcodec 57.102.100 / 57.102.100
  libavformat57. 76.100 / 57. 76.100
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 95.100 /  6. 95.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Input #0, png_pipe, from 'alpha1080.png':
  Duration: N/A, bitrate: N/A
Stream #0:0: Video: png, rgba(pc), 1920x1080 [SAR 2835:2835 DAR 16:9], 25 
fps, 25 tbr, 25 tbn, 25 tbc
File 'test.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (png (native) -> h264 (libx264))
Press [q] to stop, [?] for help
[Parsed_drawtext_0 @ 0x7fa67e1001a0] Using "/Library/Fonts/Verdana.ttf"
[libx264 @ 0x7fa67c005e00] using SAR=1/1
[libx264 @ 0x7fa67c005e00] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 
AVX FMA3 AVX2 LZCNT BMI2
[libx264 @ 0x7fa67c005e00] profile High 4:4:4 Predictive, level 4.0, 4:4:4 8-bit
[libx264 @ 0x7fa67c005e00] 264 - core 148 r2748 97eaef2 - H.264/MPEG-4 AVC 
codec - Copyleft 2003-2016 - http://www.videolan.org/x264.html - options: 
cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 
psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 
deadzone=21,11 fast_pskip=1 chroma_qp_offset=4 threads=12 lookahead_threads=2 
sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 
constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 
open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 
rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 
ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'test.mp4':
  Metadata:
encoder : Lavf57.76.100
Stream #0:0: Video: h264 (libx264) (avc1 / 0x31637661), yuv444p, 1920x1080 
[SAR 1:1 DAR 16:9], q=-1--1, 25 fps, 12800 tbn, 25 tbc
Metadata:
  encoder : Lavc57.102.100 libx264
Side data:
  cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
frame=  250 fps= 90 q=-1.0 Lsize=  26kB time=00:00:09.88 bitrate=  
21.8kbits/s speed=3.57x
video:23kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: 16.684706%
[libx264 @ 0x7fa67c005e00] frame I:1 Avg QP: 9.87  size:  3260
[libx264 @ 0x7fa67c005e00] frame P:63Avg QP: 9.16  size:80
[libx264 @ 0x7fa67c005e00] frame B:186   Avg QP:12.78  size:76
[libx264 @ 0x7fa67c005e00] consecutive B-frames:  0.8%  0.0%  0.0% 99.2%
[libx264 @ 0x7fa67c005e00] mb I  I16..4:  0.1% 99.0%  0.8%
[libx264 @ 0x7fa67c005e00] mb P  I16..4:  0.0%  0.0%  0.0%  P16..4:  0.0%  0.0% 
 0.0%  0.0%  0.0%skip:100.0%
[libx264 @ 0x7fa67c005e00] mb B  I16..4:  0.0%  0.0%  0.0%  B16..8:  0.1%  0.0% 
 0.0%  direct: 0.0%  skip:99.9%  L0:64.3% L1:35.7% BI: 0.0%
[libx264 @ 0x7fa67c005e00] 8x8 transform intra:98.7% inter:0.0%
[libx264 @ 0x7fa67c005e00] coded y,u,v intra: 0.6% 0.0% 0.0% inter: 0.0% 0.0% 
0.0%
[libx264 @ 0x7fa67c005e00] i16 v,h,dc,p: 35% 62%  3%  0%
[libx264 @ 0x7fa67c005e00] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 93%  5%  2%  0%  0%  
0%  0%  0%  0%
[libx264 @ 0x7fa67c005e00] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 40% 16% 19%  5%  5%  
4%  5%  2%  5%
[libx264 @ 0x7fa67c005e00] Weighted P-Frames: Y:0.0% UV:0.0%
[libx264 @ 0x7fa67c005e00] ref P L0: 96.7%  0.0%  3.3%
[libx264 @ 0x7fa67c005e00] ref B L0: 51.6% 48.4%
[libx264 @ 0x7fa67c005e00] ref B L1: 99.6%  0.4%
[libx264 @ 0x7fa67c005e00] kb/s:17.93


[24][12:37:07] dopelabs@jeez:~/Pictures $ ffprobe test.mp4


ffprobe version N-86825-g99c5ac2039 Copy

Re: [FFmpeg-user] ffmpeg as proxy

[DopeLabs]

ffmpeg -i rtmp://hostname.com/app/inputstream -c copy -f flv 
rtmp://hostname.com/app/outputstream


> On Sep 5, 2017, at 2:00 05AM, Simone Savi  
> wrote:
> 
> Hi all,
> I have the need to use ffmpeg as proxy for rtmp stream.
> 
> Particularly I need to know if is it possible to use a live rtmp streaming as 
> input and, if the answer is yes, what are the instructions to do this.
> 
> Best regards
> ___
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Re: [FFmpeg-user] How to livestream from youtube video link

[DopeLabs]

$ youtube-dl -F https://www.youtube.com/watch?v=mosPaiR-nrs
[youtube] mosPaiR-nrs: Downloading webpage
[youtube] mosPaiR-nrs: Downloading video info webpage
[youtube] mosPaiR-nrs: Extracting video information
WARNING: unable to extract uploader nickname
[youtube] mosPaiR-nrs: Downloading MPD manifest
[info] Available formats for mosPaiR-nrs:
format code  extension  resolution note
139  m4aaudio only DASH audio   51k , m4a_dash container, 
mp4a.40.5@ 48k (22050Hz), 20.02MiB
249  webm   audio only DASH audio   64k , opus @ 50k, 21.96MiB
250  webm   audio only DASH audio   83k , opus @ 70k, 28.86MiB
140  m4aaudio only DASH audio  131k , m4a_dash container, 
mp4a.40.2@128k (44100Hz), 53.46MiB
171  webm   audio only DASH audio  141k , vorbis@128k, 52.00MiB
251  webm   audio only DASH audio  160k , opus @160k, 56.51MiB
160  mp4256x144DASH video  113k , avc1.4d400c, 25fps, video 
only, 34.41MiB
278  webm   256x144144p  170k , webm container, vp9, 25fps, 
video only, 41.41MiB
242  webm   426x240240p  350k , vp9, 25fps, video only, 88.67MiB
133  mp4426x240DASH video  374k , avc1.4d4015, 25fps, video 
only, 70.73MiB
243  webm   640x360360p  655k , vp9, 25fps, video only, 
164.62MiB
134  mp4640x360DASH video  779k , avc1.4d401e, 25fps, video 
only, 149.69MiB
244  webm   854x480480p 1193k , vp9, 25fps, video only, 
289.00MiB
135  mp4854x480DASH video 1696k , avc1.4d401e, 25fps, video 
only, 309.33MiB
247  webm   1280x720   720p 2299k , vp9, 25fps, video only, 
579.09MiB
136  mp41280x720   DASH video 2426k , avc1.4d401f, 25fps, video 
only, 542.95MiB
137  mp41920x1080  DASH video 3961k , avc1.640028, 25fps, video 
only, 854.86MiB
248  webm   1920x1080  1080p 4034k , vp9, 25fps, video only, 1.04GiB
17   3gp176x144small , mp4v.20.3,  mp4a.40.2@ 24k
36   3gp320x180small , mp4v.20.3,  mp4a.40.2
43   webm   640x360medium , vp8.0,  vorbis@128k
18   mp4640x360medium , avc1.42001E,  mp4a.40.2@ 96k
22   mp41280x720   hd720 , avc1.64001F,  mp4a.40.2@192k (best)


$ youtube-dl -f 'bestvideo[ext=mp4]+bestaudio[ext=m4a]' 
https://www.youtube.com/watch?v=mosPaiR-nrs
[youtube] mosPaiR-nrs: Downloading webpage
[youtube] mosPaiR-nrs: Downloading video info webpage
[youtube] mosPaiR-nrs: Extracting video information
WARNING: unable to extract uploader nickname
[youtube] mosPaiR-nrs: Downloading MPD manifest
[download] Destination: Маша та Ведмідь - На кухні з Машею (всі серії підряд) 
Masha and the Bear-mosPaiR-nrs.f137.mp4
[download] 100% of 854.86MiB in 00:26
[download] Destination: Маша та Ведмідь - На кухні з Машею (всі серії підряд) 
Masha and the Bear-mosPaiR-nrs.f140.m4a
[download] 100% of 53.46MiB in 00:02
[ffmpeg] Merging formats into "Маша та Ведмідь - На кухні з Машею (всі серії 
підряд) Masha and the Bear-mosPaiR-nrs.mp4"
Deleting original file Маша та Ведмідь - На кухні з Машею (всі серії підряд) 
Masha and the Bear-mosPaiR-nrs.f137.mp4 (pass -k to keep)
Deleting original file Маша та Ведмідь - На кухні з Машею (всі серії підряд) 
Masha and the Bear-mosPaiR-nrs.f140.m4a (pass -k to keep)

 
$ ffprobe -hide banner Маша\ та\ Ведмідь\ -\ На\ кухні\ з\ Машею\ \(всі\ серії\ 
підряд\)\ Masha\ and\\ thрії\ підряд\)\ Masha\ and\ the.mp4
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'Маша та Ведмідь - На кухні з Машею 
(всі серії підряд) Masha and the Bear-mosPaiR-nrs.mp4':
  Metadata:
major_brand : isom
minor_version   : 512
compatible_brands: isomiso2avc1mp41
encoder : Lavf56.40.101
  Duration: 00:58:49.50, start: 0.00, bitrate: 2161 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p(tv, 
bt709), 1920x1080 [SAR 1:1 DAR 16:9], 2029 kb/s, 25 fps, 25 tbr, 90k tbn, 50 
tbc (default)
Metadata:
  handler_name: VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, 
fltp, 125 kb/s (default)
Metadata:
  handler_name: SoundHandler

the only problem is youtube-dl downloads each file sequentially, and for live 
streaming, thats no good. so we use a specific output template.

from the man pages

 You can merge the video  and  audio  of  two  formats  into  a  single  file  
using  -f +  (requires ffmpeg or avconv installed), for example 
-f bestvideo+bestaudio will
   download the best video-only format, the best audio-only format and mux  
them  together  with  ffm-
   peg/avconv.

but the most important bit is...

Note that if you use youtube-dl to stream to stdout (and most likely to pipe it
   to your media player then), i.e.  you explicitly specify output template 
as -o -, youtube-dl  still
   uses -f best format selection in order to start cont

Re: [FFmpeg-user] Can ffpmeg output to stdout?

[DopeLabs]

ok.. you also my want to reply with the full command output that you are trying 
in the email  for whoever can help you next =]

cheers


> On Sep 3, 2017, at 3:38 49AM, Mike Brown  wrote:
> 
> On Sun, Sep 03, 2017 at 12:22:36AM -0700, DopeLabs wrote:
>> any specific reason your not using ffmpeg to output a dts file?
> 
> According to users on Doom9, ffdcaenc is a better DTS encoder.
> 
>> and have you tried outputting to /dev/stdout?
> 
> Doesn't exist under Win XP.  I'm running a cygwin terminal, using the Z-shell.
> Sorry for not listing my environment in the first post.
> 
> MB
> 
>>> On Sep 2, 2017, at 11:58 41PM, Mike Brown  wrote:
>>> 
>>> I've tried using - for the output as well as nothing.  I set the audio codec
>>> with -acodec, so that the correct codec will be used, but nada.
>>> 
>>> I'd like to be able to pipe the output into another program that I'll be 
>>> using
>>> to create a DTS file.
>>> 
>>> I'm beginning to think that it can't be done.
>>> 
>>> MB
>>> -- 
>>> e-mail: vid...@vidiot.com | vid...@vidiot.net/~\ The ASCII
>>>   6082066...@email.uscc.net (140 char limit)   \ / Ribbon Campaign
>>> Visit - URL: http://vidiot.com/   X  Against
>>>http://vidiot.net/  / \ HTML Email
>>> "You're Sherlock Holmes, wear the damn hat!" - Watson to Sherlock
>>> Sherlock - The Abominable Bride - 1/01/16
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> e-mail: vid...@vidiot.com | vid...@vidiot.net/~\ The ASCII
>6082066...@email.uscc.net (140 char limit)   \ / Ribbon Campaign
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> http://vidiot.net/  / \ HTML Email
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Re: [FFmpeg-user] Can ffpmeg output to stdout?

[DopeLabs]

any specific reason your not using ffmpeg to output a dts file?

and have you tried outputting to /dev/stdout?


> On Sep 2, 2017, at 11:58 41PM, Mike Brown  wrote:
> 
> I've tried using - for the output as well as nothing.  I set the audio codec
> with -acodec, so that the correct codec will be used, but nada.
> 
> I'd like to be able to pipe the output into another program that I'll be using
> to create a DTS file.
> 
> I'm beginning to think that it can't be done.
> 
> MB
> -- 
> e-mail: vid...@vidiot.com | vid...@vidiot.net/~\ The ASCII
>6082066...@email.uscc.net (140 char limit)   \ / Ribbon Campaign
> Visit - URL: http://vidiot.com/   X  Against
> http://vidiot.net/  / \ HTML Email
> "You're Sherlock Holmes, wear the damn hat!" - Watson to Sherlock
> Sherlock - The Abominable Bride - 1/01/16
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Re: [FFmpeg-user] Set a background color to a mosaic

[DopeLabs]

instead of  -filter_complex "nullsrc=size=1920x1080
try  -filter_complex "color=s=1920x1080:c=black


> On Aug 28, 2017, at 3:16 57AM, Grégory S  wrote:
> 
> Howard can i do that ?
> 
> Le 28 août 2017 12:14 PM, "Carl Eugen Hoyos"  a écrit :
> 
>> 2017-08-28 12:07 GMT+02:00 Grégory S :
>>> Hi,
>>> 
>>> i use FFMpeg to create a mosaic.
>>> It works but the background color of the mosaic of my output video is
>> green
>>> by default.
>>> How can i modify my command line in order to have a black color as
>>> background for my mosaic ?
>>> 
>>> "ffmpeg.exe -i C:/source/video1.mp4 -i C:/source/video2.mp4
>> -filter_complex
>>> "nullsrc=size=1920x1080 [base]
>> 
>> Did you try using the color filter instead of nullsrc for the background?
>> 
>> Carl Eugen
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Re: [FFmpeg-user] How to livestream from youtube video link

[DopeLabs]

this would be a question for youtube-dl support.


> On Aug 23, 2017, at 9:16 57PM, johnsmithbin  wrote:
> 
> Hello.
> I want livestream from youtube video link. I tried using /youtube-dl
> https://www.youtube.com/watch?v=mosPaiR-nrs -g/ but the link returned is not
> a direct link. How can you help me?
> 
> 
> 
> --
> View this message in context: 
> http://www.ffmpeg-archive.org/How-to-livestream-from-youtube-video-link-tp4680974.html
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Re: [FFmpeg-user] Speeding up a script.

[DopeLabs]

you can search the ffmpeg-all man pages or online documentation for 'threads'

-filter_complex_threads nb_threads (global)
Defines how many threads are used to process a filter_complex graph. Similar to 
filter_threads but used for -filter_complex graphs only. The default is the 
number of available CPUs.

-filter_threads nb_threads (global)
Defines how many threads are used to process a filter pipeline. Each pipeline 
will produce a thread pool with this many threads available for parallel 
processing. The default is the number of available CPUs.

threads integer (decoding/encoding,video)
Set the number of threads to be used, in case the selected codec implementation 
supports multi-threading.

you can also set -preset ultrafast.

then there are the hevc threading options (man x265).

Threading, performance:
   --threads 
  Number of threads for thread pool (0: detect CPU core count, 
default)

   -F/--frame-threads 
  Number of concurrently encoded frames. 0: auto-determined by core 
count

   --[no-]wpp
  Enable Wavefront Parallel Processing. Default enabled

   --[no-]pmode
  Parallel mode analysis. Default disabled

   --[no-]pme
  Parallel motion estimation. Default disabled

   --[no-]asm 
  Override CPU detection. Default: auto

 
> On Aug 6, 2017, at 6:16 01AM, Evert Vorster  wrote:
> 
> Hi there.
> I am using a quite convoluted filter in ffmpeg.
> -
> #!/bin/bash
> #This will split, defish, blend and re-assemble Samsung Gear 360 video
> map_dir="/data/Projects/RemapFilter"
> ffmpeg -y -i "$1" \
> -i $map_dir/lx.pgm -i $map_dir/ly.pgm -loop 1 \
> -i $map_dir/Alpha-Map.png \
> -i $map_dir/rx.pgm -i $map_dir/ry.pgm \
> -c:v hevc_nvenc -rc constqp -qp 26 -cq 26 \
> -filter_complex \
> "[0:v]eq=contrast=0.8:brightness=-0.01:gamma=0.7:saturation=0.8[bright]; \
> [bright]split=2[in1][in2]; \
> [in1]crop=in_w/2:in_h:0:in_h[l_crop];\
> [in2]crop=in_w/2:in_h:in_w/2:in_h[r_crop]; \
> [3]alphaextract[alf]; \
> [l_crop]vignette=angle=PI/4.6:mode=backward[l_vignette]; \
> [l_vignette][1][2]remap[l_remap]; \
> [r_crop]vignette=angle=PI/4.8:mode=backward[r_vignette]; \
> [r_vignette][4][5]remap[r_remap]; \
> [l_remap]crop=in_w:1920:0:(in_h-1920)/2[l_rm_crop]; \
> [r_remap]crop=in_w:1920:0:(in_h-1920)/2[r_rm_crop]; \
> [l_rm_crop][alf]alphamerge[l_rm_crop_a]; \
> [l_rm_crop_a]split=2[l_rm_crop1][l_rm_crop2]; \
> [l_rm_crop1]crop=in_w/2:in_h:0:0[l_rm_crop_l]; \
> [l_rm_crop2]crop=in_w/2:in_h:in_w/2:0[l_rm_crop_r]; \
> [0:v][r_rm_crop]overlay=(1920-(2028/2)):0[ov1]; \
> [ov1][l_rm_crop_l]overlay=((1920+2028/2)-(2028-1920)):0[ov2]; \
> [ov2][l_rm_crop_r]overlay=0:0[out]" \
> -map [out] -map 0:a "$1_Remapped.mp4"
> -
> 
> When this runs, only one of my CPU is showing any activity.
> Is there a way of telling ffmpeg to process these steps in the filter in
> parallel?
> 
> Kind regards,
> Evert Vorster
> 
> Isometrix Acquistion Superchief
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Re: [FFmpeg-user] displaying real time text on output

[DopeLabs]

yea right now i have a cron fired script that will query the audio streaming 
server for the title, and if its different from the title thats in the txt 
file, atomically updates it.

it works just fine, but if i can get it all natively within ffmpeg that would 
be preferred . =]

oh.. and the code example i provided should place the text at the bottom 
center, not the top. sorry for the typo...


> On Aug 6, 2017, at 2:50 19PM, tasos  wrote:
> 
> Hello.
> Thank you very much i will try give it a try!
> Well you could use an external program that would read the metadata and write 
> to your txt file.
> But i suppose there's another solution by using ffmpeg and not an external 
> program.
> I hope someone will tell you how to do it.
> Thanks again!
> 
> 
> On 8/7/2017 12:43 AM, DopeLabs wrote:
>> you can use the drawtext filter.
>> 
>> i have several drawtext filters i run, displaying audio track titles, site 
>> name, original air date (or 'LIVE' if live).
>> 
>> the trick to having the text update mid stream or mid encode is to specify 
>> textfile="/path/file.txt":reload=1
>> 
>> reload=1 will cause ffmpeg to check the file every frame.
>> 
>> you need to update the file atomically or ffmpeg may partially read the 
>> file, or even fail.
>> 
>> you can achieve this by not editing the file with an editor or even using 
>> echo "text" > file.txt...
>> 
>> but instead save a new temp file, then mv tempfile.txt file.txt
>> 
>> here is an example of one that i use:
>> 
>> __
>> 
>> "drawtext=fontfile=/path/font.ttf:fontsize=28:fix_bounds=true:fontcolor=gray:alpha=.5:textfile=/path/file.txt:reload=1:y=(h-text_h)-5:x=(w-text_w)/2"
>> __
>> 
>> 
>> this creates semi transparent gray text at the top center of the frame, will 
>> auto scale if the amount of text exceeds the width of the frame so text wont 
>> get truncated.
>> 
>> if you are using this in a livestream scenario, you can see the text updates 
>> in real time.
>> 
>> 
>> 
>> i have not tried this when the input or output are not live streams...
>> 
>> i would suspect it would be kind of hard to get the timing right since the 
>> input/output can be read/written faster than real time.
>> 
>> 
>> on a side note:
>> im currently trying to figure out if its possible to use text expansion to 
>> pull metadata values from within the stream, and use in drawtext output.
>> 
>> an example would be an mp3 audio live stream such as icecast or shoutcast, 
>> when song titles change in the stream, to display that value in drawtext 
>> output which updates mid-stream.
>> 
>> it looks like it could be done but im just struggling a bit trying to figure 
>> out the correct syntax, metadata keys, etc.
>> 
>> so if anyone has any insight on that. im all ears.
>> 
>> -DL
>> 
>>> On Aug 6, 2017, at 11:39 43AM, tasos  wrote:
>>> 
>>> Hello.
>>> I know that this may be a stupid question but is it possible while encoding 
>>> to apply a filter that displays text for example?
>>> I mean ffmpeg is already running and produces output.
>>> I don't want to take that output and apply the text filter there.I want to 
>>> apply it on the already
>>> running ffmpeg process.
>>> Thanks!
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> 
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Re: [FFmpeg-user] displaying real time text on output

[DopeLabs]

you can use the drawtext filter.

i have several drawtext filters i run, displaying audio track titles, site 
name, original air date (or 'LIVE' if live).

the trick to having the text update mid stream or mid encode is to specify 
textfile="/path/file.txt":reload=1

reload=1 will cause ffmpeg to check the file every frame. 

you need to update the file atomically or ffmpeg may partially read the file, 
or even fail.

you can achieve this by not editing the file with an editor or even using echo 
"text" > file.txt... 

but instead save a new temp file, then mv tempfile.txt file.txt

here is an example of one that i use:

__

"drawtext=fontfile=/path/font.ttf:fontsize=28:fix_bounds=true:fontcolor=gray:alpha=.5:textfile=/path/file.txt:reload=1:y=(h-text_h)-5:x=(w-text_w)/2"
__


this creates semi transparent gray text at the top center of the frame, will 
auto scale if the amount of text exceeds the width of the frame so text wont 
get truncated.

if you are using this in a livestream scenario, you can see the text updates in 
real time.



i have not tried this when the input or output are not live streams... 

i would suspect it would be kind of hard to get the timing right since the 
input/output can be read/written faster than real time.  


on a side note:
im currently trying to figure out if its possible to use text expansion to pull 
metadata values from within the stream, and use in drawtext output. 

an example would be an mp3 audio live stream such as icecast or shoutcast, when 
song titles change in the stream, to display that value in drawtext output 
which updates mid-stream.

it looks like it could be done but im just struggling a bit trying to figure 
out the correct syntax, metadata keys, etc. 

so if anyone has any insight on that. im all ears.

-DL

> On Aug 6, 2017, at 11:39 43AM, tasos  wrote:
> 
> Hello.
> I know that this may be a stupid question but is it possible while encoding 
> to apply a filter that displays text for example?
> I mean ffmpeg is already running and produces output.
> I don't want to take that output and apply the text filter there.I want to 
> apply it on the already
> running ffmpeg process.
> Thanks!
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Re: [FFmpeg-user] Streaming HLS from image in watch folder

[DopeLabs]

use -loop 1 -i image.jpg

and remember to update the file atomically or ffmpeg may partially read the 
file, or even fail.

you can achieve this by not overwriting the new image in the normal sense, but 
instead save the new image to a temp file, then mv temp.jpg image.jpg

you may also wish to specify the input frame rate before the image as well =]


cheers,

DL

> On Aug 2, 2017, at 12:56 42PM, Michael Davis  wrote:
> 
> Hello,
> 
> I want to create a live HLS stream that displays a single image. However, the 
> image is updated by an external program on average about every 3 minutes.  
> The filename remains the same, but it's contents are always different.  I am 
> able to create a live HLS stream using the image as input.  But ffmpeg does 
> not check the input file for changes and so outputs continuously the first 
> rendition of the image that it loads.  Also, I am using a live audio stream 
> coming from icecast as the audio portion of this HLS stream.
> 
> Can ffmpeg be instructed to check file modification and load new, or to load 
> new based on time interval?
> 
> 
> Thanks,
> Mike
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Re: [FFmpeg-user] Compile FFmpeg with openssl in mac without Homebrew?

[DopeLabs]

You can check out the wiki which has instructions for installation via 
homebrew, or manually compiling.

Though something as simple as 

$ brew install ffmpeg --with-gnutls

Or

$ brew install ffmpeg --with-openssl

Should do the trick for you =]

Heres the wiki

https://trac.ffmpeg.org/wiki/CompilationGuide/MacOSX#CompilingFFmpegyourself


> On Jul 20, 2017, at 5:30 31AM, willin  wrote:
> 
> Hi 
> 
> I know that some people posted this question but I didn't quite get how to
> do from the answers..
> 
> I want to record some live streaming videos which contain https.
> I installed and set up FFmpeg using this link:
> http://ericholsinger.com/install-ffmpeg-on-a-mac. I found that initial
> setting of FFmpeg does not include https in its available protocols..
> 
> I'm pretty much beginner for doing this and I want to know more about FFmpeg
> so I am not going to use homebrew or any other dependencies so far.
> 
> I got this error message when I tried to record streaming videos with https:
> "https protocol not found, recompile FFmpeg with openssl, gnutls or
> securetransport enabled."
> 
> It would be very very much appreciated if someone could help me with it..
> 
> Thanks in advance
> 
> 
> 
> 
> --
> View this message in context: 
> http://www.ffmpeg-archive.org/Compile-FFmpeg-with-openssl-in-mac-without-Homebrew-tp4680639.html
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Re: [FFmpeg-user] Using FFMPEG to encode multiple outputs in 'parallel' (doesn't work?)

[DopeLabs]

Option 2 in your question looks fine to me... 

I have used ffmpeg to transcode audio and video streams before to achieve 
adaptive streaming.

Here is an example of an audio transcode... input is 256k mp3, output is 128k, 
64k, and 24k

$ ffmpeg -i http://stream.dubstep.fm -c:a mp3 -b:a 128k -f null /dev/null -c:a 
mp3 -b:a 64k -f null /dev/null -c:a mp3 -b:a 24k -ar 22050 -f null /dev/null


ffmpeg version N-86413-g50be8f2142 Copyright (c) 2000-2017 the FFmpeg developers
  built with Apple LLVM version 8.1.0 (clang-802.0.42)
  configuration: --enable-libvidstab --enable-shared --enable-pthreads 
--cc=/usr/bin/clang --enable-vda --enable-videotoolbox --arch=x86_64 
--enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid 
--enable-filter=coreimage --enable-filter=coreimagesrc --enable-filter=frei0r 
--enable-filter=frei0r_src --enable-swscale --enable-avfilter 
--enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus 
--enable-libtheora --enable-libopenjpeg --enable-libmodplug --enable-libvpx 
--enable-libsoxr --enable-libspeex --enable-libass --enable-libbluray 
--enable-lzma --enable-gnutls --enable-fontconfig --enable-libfreetype 
--enable-libfribidi --enable-libx265 --enable-version3 --enable-libwavpack 
--enable-libssh --enable-librtmp --enable-libfdk-aac --enable-nonfree 
--enable-libzmq --enable-opencl --enable-ffplay --enable-libcaca 
--enable-libvidstab
  libavutil  55. 63.100 / 55. 63.100
  libavcodec 57. 96.101 / 57. 96.101
  libavformat57. 72.101 / 57. 72.101
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 91.100 /  6. 91.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
[mp3 @ 0x7fa3b200] Skipping 88 bytes of junk at 0.
Input #0, mp3, from 'http://stream.dubstep.fm':
  Metadata:
icy-genre   : Dubstep, Dub, Electronic, Garage
icy-name: Dubstep.fm
icy-notice1 : This stream requires http://www.winamp.com";>Winamp
icy-notice2 : SHOUTcast DNAS/posix(linux x64) v2.5.1.724
icy-pub : 1
icy-url : https://www.dubstep.fm
StreamTitle : ARCHIVE - 2011-01-09 - The Dented McStank With Phaded 
From Chicago
StreamUrl   : https://www.dubstep.fm
  Duration: N/A, start: 0.00, bitrate: 256 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 256 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (native) -> mp3 (libmp3lame))
  Stream #0:0 -> #1:0 (mp3 (native) -> mp3 (libmp3lame))
  Stream #0:0 -> #2:0 (mp3 (native) -> mp3 (libmp3lame))
Press [q] to stop, [?] for help
Output #0, null, to '/dev/null':
  Metadata:
icy-genre   : Dubstep, Dub, Electronic, Garage
icy-name: Dubstep.fm
icy-notice1 : This stream requires http://www.winamp.com";>Winamp
icy-notice2 : SHOUTcast DNAS/posix(linux x64) v2.5.1.724
icy-pub : 1
icy-url : https://www.dubstep.fm
StreamTitle : ARCHIVE - 2011-01-09 - The Dented McStank With Phaded 
From Chicago
StreamUrl   : https://www.dubstep.fm
encoder : Lavf57.72.101
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 128 kb/s
Metadata:
  encoder : Lavc57.96.101 libmp3lame
Output #1, null, to '/dev/null':
  Metadata:
icy-genre   : Dubstep, Dub, Electronic, Garage
icy-name: Dubstep.fm
icy-notice1 : This stream requires http://www.winamp.com";>Winamp
icy-notice2 : SHOUTcast DNAS/posix(linux x64) v2.5.1.724
icy-pub : 1
icy-url : https://www.dubstep.fm
StreamTitle : ARCHIVE - 2011-01-09 - The Dented McStank With Phaded 
From Chicago
StreamUrl   : https://www.dubstep.fm
encoder : Lavf57.72.101
Stream #1:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, 64 kb/s
Metadata:
  encoder : Lavc57.96.101 libmp3lame
Output #2, null, to '/dev/null':
  Metadata:
icy-genre   : Dubstep, Dub, Electronic, Garage
icy-name: Dubstep.fm
icy-notice1 : This stream requires http://www.winamp.com";>Winamp
icy-notice2 : SHOUTcast DNAS/posix(linux x64) v2.5.1.724
icy-pub : 1
icy-url : https://www.dubstep.fm
StreamTitle : ARCHIVE - 2011-01-09 - The Dented McStank With Phaded 
From Chicago
StreamUrl   : https://www.dubstep.fm
encoder : Lavf57.72.101
Stream #2:0: Audio: mp3 (libmp3lame), 22050 Hz, stereo, s16p, 24 kb/s
Metadata:
  encoder : Lavc57.96.101 libmp3lame
size=N/A time=00:00:13.14 bitrate=N/A speed=2.69x
video:0kB audio:347kB subtitle:0kB other streams:0kB global headers:0kB muxing 
overhead: unknown





> On Jul 5, 2017, at 2:23 39PM, sh


> alin  wrote:
> 
> Reuben Martin wrote
>> On Thursday, June 29, 2017 6:56:51 PM CDT shalin wrote:
>>> Cley Faye wrote
>>> 
 More to the point, as far as I know, ffmpeg will not do p

Re: [FFmpeg-user] FFplay - Audio bars on video

[DopeLabs]

I have only been able to use local files with just ffplay and using amovie= or 
movie=

Anyone know how to use http or rtmp ?


> On Jun 24, 2017, at 10:25 27AM, Lou  wrote:
> 
> On Sat, Jun 17, 2017, at 10:39 PM, Manmathan wrote:
>> Please guide how to do this with* ffplay*
> 
> ffplay -f lavfi
> "amovie=input.mp4,asplit[a][out1];[a]showvolume[fg];movie=input.mp4[bg];[bg][fg]overlay[out0]"
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Re: [FFmpeg-user] FFplay - Audio bars on video

[DopeLabs]

It will work if you use ffmpeg and pipe to ffplay... here is one example

ffmpeg -loglevel 0 -i rtmp://video/live/stream -filter_complex 
"[0:a]pan=1c|c0=c0,showvolume=r=60:b=0:t=0:v=0:w=720:h=80:f=0.5:o=1[left];[0:a]pan=1c|c0=c1,showvolume=r=60:b=0:t=0:v=0:w=720:h=80:f=0.5:o=1[right];[left][0:v]hstack[tmp1];[tmp1][right]hstack"
 -c:v h264 -preset ultrafast -c:a copy -f flv - | ffplay -f flv -




> On Jun 17, 2017, at 11:39 45PM, Manmathan  wrote:
> 
> Hello Everyone.
> We need to monitor audio (vertical bar-graph) over video of* live *stream. 
> Audio is able to monitor separately with ffplay :
> (http://www.ffmpeg-archive.org/Error-configuring-filters-AudioBars-td4680275.html#a4680277)
> ffplay -f lavfi "amovie=in.mp4, asplit [a][out1]; [a] 
> showfreqs=mode=line:fscale=log [out0]"
> 
> Audio over video is possible with ffmpeg on recorded
> file.(http://www.ffmpeg-archive.org/how-to-create-audio-bar-in-video-td4680323.html)
> ffmpeg -i inputfile_or_stream -filter_complex "[0:a]asplit[a][aout];
> [a]showvolume=o=v[bars]; [0:v][bars]overlay=x=main_w-overlay_w:y=0[vout]"
> -map "[vout]" -map "[aout]" output.mp4 
> 
> Please guide how to do this with* ffplay* or live viewing on ffmpeg or pipe
> out of the above to vlc.
> 
> 
> 
> 
> 
> --
> View this message in context: 
> http://www.ffmpeg-archive.org/FFplay-Audio-bars-on-video-tp4680331.html
> Sent from the FFmpeg-users mailing list archive at Nabble.com.
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Re: [FFmpeg-user] Automatically set codec

[DopeLabs]

Generally this would be done first by analyzing the file before running the 
ffmpeg command. Usually in the form of a small script. 

Though scripting is not on topic for this mailing list, here is one anyways.

dependencies:

mediainfo
ffmpeg 

--

#!/bin/bash

cid="$(mediainfo --Inform="Video;%CodecID%" "$1")";

if [ "$cid" = "avc1" ]
  then
echo "CodecID: $cid - Copy";
ffmpeg -i "$1" -c:v copy ;
  else
echo "CodecID: $cid - Encode";
ffmpeg -i "$1" -c:v libx264 ;
fi

--

Execute the script using the input filename as the first argument.

$ ./script.sh /path/to/input/video.mp4


If you have any questions please feel free to reply direct =]

Cheers


> On Jun 22, 2017, at 10:57 23PM, tony_rui  wrote:
> 
> I want to automatically set a codec.
> For example, if the codec of a movie is h264, set -c:v copy. Otherwise, set 
> -c:v libx264.
> Please tell me the way to do.
> 
> Thank you for reading this mail.
> 
> tony_rui
> 
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Re: [FFmpeg-user] Is it possible to use ffmpeg as a rtmp downloader like rtmpdump?

[DopeLabs]

Your command should work just fine. Is the file not playable when the stream 
completes or after you stop ffmpeg?


$ ffmpeg -i rtmp://stream.dsfm.tv/live/dubstepfm_720p -c copy -f flv output.flv
ffmpeg version N-86169-g55b56a8 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  configuration: --enable-shared --disable-static --enable-version3 
--enable-openal --enable-libxvid --enable-libx264 --enable-libwavpack 
--enable-libvpx --enable-libvorbis --enable-libtheora --enable-libssh 
--enable-librtmp --enable-libopenjpeg --enable-libmp3lame --enable-libfdk-aac 
--enable-libass --enable-avresample --enable-nonfree --enable-gpl 
--enable-libfreetype --enable-openssl --enable-fontconfig --enable-libzmq 
--enable-libfribidi --enable-libsoxr --enable-libcaca --enable-chromaprint 
--enable-avisynth --disable-txtpages --disable-podpages --disable-htmlpages 
--enable-libx265
  libavutil  55. 63.100 / 55. 63.100
  libavcodec 57. 96.101 / 57. 96.101
  libavformat57. 72.101 / 57. 72.101
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 90.100 /  6. 90.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Metadata:
  duration  0.00
  width 1280.00
  height720.00
  videodatarate 19531.25
  framerate 60.00
  videocodecid  7.00
  audiodatarate 250.00
  audiosamplerate   44100.00
  audiosamplesize   16.00
  stereoTRUE
  audiocodecid  10.00
  major_brand   dash
  minor_version 0
  compatible_brands iso6avc1mp41
  encoder   Lavf57.28.100
  filesize  0.00
Input #0, flv, from 'rtmp://stream.dsfm.tv/live/dubstepfm_720p':
  Metadata:
major_brand : dash
minor_version   : 0
compatible_brands: iso6avc1mp41
encoder : Lavf57.28.100
  Duration: 00:00:00.00, start: 0.00, bitrate: N/A
Stream #0:0: Video: h264 (High), yuv420p(progressive), 1280x720 [SAR 1:1 
DAR 16:9], 2 kb/s, 60 fps, 60 tbr, 1k tbn, 120 tbc
Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 256 kb/s
Output #0, flv, to 'output.flv':
  Metadata:
major_brand : dash
minor_version   : 0
compatible_brands: iso6avc1mp41
encoder : Lavf57.72.101
Stream #0:0: Video: h264 (High) ([7][0][0][0] / 0x0007), 
yuv420p(progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 2 kb/s, 60 fps, 
60 tbr, 1k tbn, 1k tbc
Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 44100 Hz, stereo, 
fltp, 256 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame=  328 fps= 61 q=-1.0 Lsize=   16525kB time=00:00:05.80 
bitrate=23331.7kbits/s speed=1.08x
video:16332kB audio:181kB subtitle:0kB other streams:0kB global headers:0kB 
muxing overhead: 0.066842%



$ ffprobe output.flv
ffprobe version N-86169-g55b56a8 Copyright (c) 2007-2017 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.4) 20160609
  configuration: --enable-shared --disable-static --enable-version3 
--enable-openal --enable-libxvid --enable-libx264 --enable-libwavpack 
--enable-libvpx --enable-libvorbis --enable-libtheora --enable-libssh 
--enable-librtmp --enable-libopenjpeg --enable-libmp3lame --enable-libfdk-aac 
--enable-libass --enable-avresample --enable-nonfree --enable-gpl 
--enable-libfreetype --enable-openssl --enable-fontconfig --enable-libzmq 
--enable-libfribidi --enable-libsoxr --enable-libcaca --enable-chromaprint 
--enable-avisynth --disable-txtpages --disable-podpages --disable-htmlpages 
--enable-libx265
  libavutil  55. 63.100 / 55. 63.100
  libavcodec 57. 96.101 / 57. 96.101
  libavformat57. 72.101 / 57. 72.101
  libavdevice57.  7.100 / 57.  7.100
  libavfilter 6. 90.100 /  6. 90.100
  libavresample   3.  6.  0 /  3.  6.  0
  libswscale  4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc54.  6.100 / 54.  6.100
Input #0, flv, from 'output.flv':
  Metadata:
major_brand : dash
minor_version   : 0
compatible_brands: iso6avc1mp41
encoder : Lavf57.72.101
  Duration: 00:00:05.85, start: 0.00, bitrate: 23140 kb/s
Stream #0:0: Video: h264 (High), yuv420p(progressive), 1280x720 [SAR 1:1 
DAR 16:9], 2 kb/s, 60 fps, 60 tbr, 1k tbn, 120 tbc
Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 256 kb/s






> On May 19, 2017, at 3:20 36AM, Xu Pu Ti  wrote:
> 
> A simple 'ffmpeg -i rtmp://url -c copy -f flv -' is not expected, since ffmpeg
> cares too much about the actual content of the stream and will complain about
> bad flv tags and ignore tags it doesn't understand.
> 
> Is it possible to make ffmpeg just download from rtmp and save it as a flv 
> file
> and do not inspect what it contains?
> 

Re: [FFmpeg-user] Problem with ffmpeg command when recording live streams

[DopeLabs]

Have you tried a stream copy or only encoding the audio to mp3/aac since the 
video looks like its already in h264?

Stream copy video and re-encode audio

ffmpeg -I rtsp:/ip -c:v copy -c:a mp3 -b:a 128k -movflags +faststart -f mp4 
outfile.mp4



> On Apr 28, 2017, at 6:12 24AM, Yanet Giuse 
>  wrote:
> 
> Hello!I am trying to record a live rtsp stream and save videos locally using 
> ffmpeg, but i cant.This stream is a camera, and Ive tested the video stream 
> in VLC. It works.
> First, the ffplay command (ffplay rtsp://ip ) returns
> [rtsp @ 0x7fa574009280] UDP timeout, retrying with TCP   0B f=0/0   
> [rtsp @ 0x7fa574009280] method PAUSE failed: 455 Method Not Valid in This 
> State
> [rtsp @ 0x7fa574009280] Could not find codec parameters for stream 0 (Video: 
> h264, none): unspecified size
> Consider increasing the value for the 'analyzeduration' and 'probesize' 
> options
> Input #0, rtsp, from 'rtsp://ip':
>   Metadata:
> title   : Media Server
>   Duration: N/A, bitrate: 128 kb/s
> Stream #0:0: Video: h264, none, 90k tbr, 90k tbn, 180k tbc
> Stream #0:1: Audio: pcm_alaw, 16000 Hz, 1 channels, s16, 128 kb/s
> [rtsp @ 0x7fa574009280] UDP timeout, retrying with TCP   0B f=0/0   
> [rtsp @ 0x7fa574009280] method PAUSE failed: 455 Method Not Valid in This 
> State
> So, i installed h264 and h265 codecs and re-installed ffmpeg. But the same 
> error happened.Alsp, i tried the ffmpeg command
> ffmpeg -i rtsp://ip -r 15 outvideo.mp4
> the result is:
> [rtsp @ 0x11e30c0] UDP timeout, retrying with TCP
> [rtsp @ 0x11e30c0] method PAUSE failed: 455 Method Not Valid in This State
> [rtsp @ 0x11e30c0] Could not find codec parameters for stream 0 (Video: h264, 
> none): unspecified size
> Consider increasing the value for the 'analyzeduration' and 'probesize' 
> options
> Guessed Channel Layout for Input Stream #0.1 : mono
> Input #0, rtsp, from 'rtsp://ip':
>   Metadata:
> title   : Media Server
>   Duration: N/A, bitrate: 128 kb/s
> Stream #0:0: Video: h264, none, 90k tbr, 90k tbn, 180k tbc
> Stream #0:1: Audio: pcm_alaw, 16000 Hz, 1 channels, s16, 128 kb/s
> [mp4 @ 0x1237260] Using AVStream.codec to pass codec parameters to muxers is 
> deprecated, use AVStream.codecpar instead.
> Output #0, mp4, to 'outvideo.mp4':
>   Metadata:
> title   : Media Server
> encoder : Lavf57.50.100
> Stream #0:0: Audio: aac (LC) ([64][0][0][0] / 0x0040), 16000 Hz, mono, 
> fltp, 69 kb/s
> Metadata:
>   encoder : Lavc57.57.101 aac
> Stream mapping:
>   Stream #0:1 -> #0:0 (pcm_alaw (native) -> aac (native))
> Press [q] to stop, [?] for help
> [rtsp @ 0x11e30c0] UDP timeout, retrying with TCP
> [rtsp @ 0x11e30c0] method PAUSE failed: 455 Method Not Valid in This State
> rtsp://ip: Operation not permitted
> size=   0kB time=00:00:00.00 bitrate=N/A speed=   0x
> video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing 
> overhead: unknown
> [aac @ 0x1238a60] Qavg: -nan
> 
> I tried with several parameters I found on the web. No chances to work.
> 
> I think it is a codec problem, or maybe the reason is that my camera is in 
> 192.168.32. and my computer works in 192.168.31.I am searching for suggests, 
> reasons, commands .. Everything that helps..
> 
> 
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Re: [FFmpeg-user] change playing options in ffplay

[DopeLabs]

mpv is pretty nice. You can use a local config file in ~/.config/mpv/config 
where you can set the keybinding and seek values.

IINA if your on a Mac is also looking pretty good...






> On Apr 22, 2017, at 5:13 43AM, Moritz Barsnick  wrote:
> 
> On Sat, Apr 22, 2017 at 08:38:23 +, Bernd Butscheidt wrote:
> 
>> am I right that there is no way to adjust these playing options of
>> ffplay from the command line/user interface:
> 
> You are right.
> 
> Other players such as mplayer can be configured, perhaps you want to
> look for such an alternative.
> 
> Moritz
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Re: [FFmpeg-user] Can't register to trac

[DopeLabs]

I was just able to register successfully... it also said that it found my ip in 
a black list.

I checked mxtoolbox and dnsbl databases for my ip and as I figured, it wasn't 
listed

but I suspect it will find any ip in there


cheers,

DL

> On Feb 27, 2017, at 1:49 31AM, Martin Ždila  wrote:
> 
> Hello
> 
> I am unable to register to trac.ffmpeg.org because it thinks I am a
> spammer. I've tried two networks(178.253.172.70  and 85.237.234.248) ant
> twice correctly retyped recaptcha, but still it gets me errors about
> spammer. I've never seen such an overprotection in any of the bug tracking
> systems I've used so far. Is there a way to get in for me without trying
> other networks?
> 
> Best regards
> -- 
> Martin Ždila
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Re: [FFmpeg-user] -movflags faststart revisited

[DopeLabs]

I utilize ffplay a lot for previewing encoding settings.. make you might find 
it handy as well =]


> On Feb 26, 2017, at 4:49 39PM, Reuben Martin  wrote:
> 
> On Sunday, February 26, 2017 5:44:56 PM CST JD wrote:
>> On Sun, Feb 26, 2017 at 4:07 PM, Cley Faye  wrote:
>>> 2017-02-26 23:32 GMT+01:00 JD :
 Are there any flags that will let me playback an unfinished transcode?
 Reason I am asking is that if my various options in the transcode
 command
 did not yield what I like, then I would like to abort the transcode.
>>> 
>>> ​mp4 files will never be playable without "finishing" them. But you can
>>> interrupt the process at anytime with 'q'; that will properly close the
>>> file so you can play it.​
>> 
>> ​Thanks - that is somewhat a panacea, especially if one
>> is not 50% or more into the transcoding - so that issuing
>> a 'q' will not lose too much is only say the transcoding is
>> into 10 minutes of the video.
>> 
>> Now how about ffplay? I mean is/are there options
>> to ffplay that would allow the playing of the unfinsihed
>> mp4 file (without quitting the trancoding)?
>> ​
> 
> Set the output format to flv. You can play flv before it’s finished encoding. 
> If you decide you want to keep it, you can dump the video and audio streams 
> from the flv into an mp4 file. (generally only takes a few seconds).
> 
> -Reuben
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Re: [FFmpeg-user] Using the flags -movflags +faststart

[DopeLabs]

Does the file work for you if you let the encode process complete and exit 
cleanly?

How did you stop the ffmpeg process once it was 'about 3 minutes' into the 
encoding process?

I have found if you stop ffmpeg by control-c, it may fail to properly finish 
writing the file and seems broken. ffprobe will complain about a missing MOOV 
ATOM.

Though if you press 'q' to stop instead, it should exit cleanly and the file 
should be watchable.

cheers

DL

> On Feb 24, 2017, at 4:29 24PM, JD  wrote:
> 
> I played the output file when only about 3 minutes of it had been
> transcoded. The file did not play at all.
> mplayer was belching out:
> 
> Starting playback...
> [dvvideo @ 0x7f5461b94540]could not find dv frame profile
> Error while decoding frame!
> [dvvideo @ 0x7f5461b94540]could not find dv frame profile
> Error while decoding frame!
> [dvvideo @ 0x7f5461b94540]could not find dv frame profile
> Error while decoding frame!
> [dvvideo @ 0x7f5461b94540]could not find dv frame profile
> Error while decoding frame!
> [dvvideo @ 0x7f5461b94540]could not find dv frame profile
> Error while decoding frame!
> [dvvideo @ 0x7f5461b94540]could not find dv frame profile
> Error while decoding frame!
> 
> and the video window was all twinkling square pixels.
> 
> $ ~/bin/ffmpeg.d/ffmpeg -i video_Z11.mp4 -movflags +faststart -vb 8000k -ab
> 384k -s 1920x1080 -y video_Z11-1920x1080.mp4
> ffmpeg version 3.2.4-static http://johnvansickle.com/ffmpeg/  Copyright (c)
> 2000-2017 the FFmpeg developers
>  built with gcc 5.4.1 (Debian 5.4.1-5) 20170205
>  configuration: --enable-gpl --enable-version3 --enable-static
> --disable-debug --disable-ffplay --disable-indev=sndio
> --disable-outdev=sndio --cc=gcc-5 --enable-fontconfig --enable-frei0r
> --enable-gnutls --enable-gray --enable-libass --enable-libfreetype
> --enable-libfribidi --enable-libmp3lame --enable-libopencore-amrnb
> --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus
> --enable-librtmp --enable-libsoxr --enable-libspeex --enable-libtheora
> --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis
> --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxvid --enable-libzimg
>  libavutil  55. 34.101 / 55. 34.101
>  libavcodec 57. 64.101 / 57. 64.101
>  libavformat57. 56.101 / 57. 56.101
>  libavdevice57.  1.100 / 57.  1.100
>  libavfilter 6. 65.100 /  6. 65.100
>  libswscale  4.  2.100 /  4.  2.100
>  libswresample   2.  3.100 /  2.  3.100
>  libpostproc54.  1.100 / 54.  1.100
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video_Z11.mp4':
>  Metadata:
>major_brand : isom
>minor_version   : 512
>compatible_brands: isomiso2avc1mp41
>creation_time   : 2016-11-29T13:42:34.00Z
>encoder : Lavf56.1.0
>  Duration: 02:22:55.48, start: 0.00, bitrate: 379 kb/s
>Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p,
> 480x360 [SAR 1:1 DAR 4:3], 246 kb/s, 29.95 fps, 29.97 tbr, 90k tbn, 59.91
> tbc (default)
>Metadata:
>  creation_time   : 2016-11-29T13:42:34.00Z
>  handler_name: VideoHandler
>Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz,
> stereo, fltp, 125 kb/s (default)
>Metadata:
>  creation_time   : 2016-11-29T13:42:34.00Z
>  handler_name: SoundHandler
> [libx264 @ 0x575c240] using SAR=3/4
> [libx264 @ 0x575c240] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
> [libx264 @ 0x575c240] profile High, level 4.0
> [libx264 @ 0x575c240] 264 - core 148 r333 90a61ec - H.264/MPEG-4 AVC codec
> - Copyleft 2003-2017 - http://www.videolan.org/x264.html - options: cabac=1
> ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00
> mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11
> fast_pskip=1 chroma_qp_offset=-2 threads=3 lookahead_threads=1
> sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0
> constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1
> weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40
> intra_refresh=0 rc_lookahead=40 rc=abr mbtree=1 bitrate=8000 ratetol=1.0
> qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
> Output #0, mp4, to 'video_Z11-1920x1080.mp4':
>  Metadata:
>major_brand : isom
>minor_version   : 512
>compatible_brands: isomiso2avc1mp41
>encoder : Lavf57.56.101
>Stream #0:0(und): Video: h264 (libx264) ([33][0][0][0] / 0x0021),
> yuv420p, 1920x1080 [SAR 3:4 DAR 4:3], q=-1--1, 8000 kb/s, 29.97 fps, 30k
> tbn, 29.97 tbc (default)
>Metadata:
>  creation_time   : 2016-11-29T13:42:34.00Z
>  handler_name: VideoHandler
>  encoder : Lavc57.64.101 libx264
>Side data:
>  cpb: bitrate max/min/avg: 0/0/800 buffer size: 0 vbv_delay: -1
>Stream #0:1(und): Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz,
> stereo, fltp, 384 kb/s (default)
>Metadata:
>  creation_time   : 2016-11-29T13:42:34.00

Re: [FFmpeg-user] -c:v or -vcodec copy or -codec:v

[DopeLabs]

c:v is an abbreviated version of codec:v

vcodec is an alias of codec:v

So all 3 function the same, but are not limited to 'copy' only. It is used to 
specify the codec, or copy.

For example

ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT

encodes all video streams with libx264 and copies all audio streams.

This could also be written as

ffmpeg -i INPUT -map 0 -codec:v libx264 -acodec copy OUTPUT

or

ffmpeg -i INPUT -map 0 -vcodec libx264 -codec:a copy OUTPUT




From the man pages:

-vcodec codec (output)

Set the video codec. This is an alias for "-codec:v".


-c[:stream_specifier] codec (input/output,per-stream)
-codec[:stream_specifier] codec (input/output,per-stream)

Select an encoder (when used before an output file) or a decoder (when used 
before an input file) for one or more streams. codec is the name of a 
decoder/encoder or a special value "copy" (output only to indicate that the 
stream is not to be re-encoded.


For example

ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT

encodes all video streams with libx264 and copies all audio streams.






> On Feb 21, 2017, at 11:39 51PM, Tim Hiles  wrote:
> 
> Hi all,
> 
> What is the deal with all the different ways to copy?  Do they all do the
> same thing? Cause if not, it is NOT clear what they are and if soseems
> like a lot of redundancy.
> 
> Thanks!
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Re: [FFmpeg-user] Question about mp3 bit rate mode

[DopeLabs]

In your earlier console output for both commands...

>> size=2344kB time=00:05:00.01 bitrate=  64.0kbits/s speed= 128x 

>> size=2344kB time=00:05:00.01 bitrate=  64.0kbits/s speed= 134x

What method are you using to 'verify' the length?

What is the ffprobe output for each of those files?



> On Jan 30, 2017, at 8:23 13AM, 桃源老師  wrote:
> 
> In Addition...
> 
>> 2017/01/30 11:54 A.M. 桃源老師 dream.jp> wrote:
>> 
>> Hello ffmpeg experts...
>> 
>> I have question about mp3 bit rate mode.  
>> 
>> When I run the following command, the resulting duration.mp3 is VBR.  I have 
>> confirmed it's bit rate mode with mediainfo, mpg123, and SinkuSuperLite (on 
>> Windows).
>> 
>> $ ffmpeg -f lavfi -i sine=d=300 duration.mp3
>> ffmpeg version N-83152-gf7e9275 Copyright (c) 2000-2017 the FFmpeg developers
>> built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
>> configuration: --prefix=/Volumes/ffmpeg_compile --pkg-config-flags=--static 
>> --disable-ffserver --enable-gpl --enable-version3 --enable-nonfree 
>> --enable-libmp3lame --enable-libfdk-aac --enable-libvo-amrwbenc 
>> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvorbis 
>> --enable-libspeex --enable-libopus --enable-libgsm --enable-libtwolame 
>> --enable-libsoxr --enable-libwavpack --enable-libmodplug 
>> --enable-libopenjpeg --enable-libwebp --enable-libtheora --enable-libx264 
>> --enable-libopenh264 --enable-libx265 --enable-libxvid --enable-libvpx 
>> --enable-libxavs --enable-libfreetype --enable-fontconfig 
>> --enable-libfribidi --enable-libass --enable-libbluray --enable-libvidstab 
>> --enable-libschroedinger --enable-libsnappy --enable-libzmq
>> libavutil  55. 43.100 / 55. 43.100
>> libavcodec 57. 74.100 / 57. 74.100
>> libavformat57. 62.100 / 57. 62.100
>> libavdevice57.  2.100 / 57.  2.100
>> libavfilter 6. 69.100 /  6. 69.100
>> libswscale  4.  3.101 /  4.  3.101
>> libswresample   2.  4.100 /  2.  4.100
>> libpostproc54.  2.100 / 54.  2.100
>> Input #0, lavfi, from 'sine=d=300':
>> Duration: N/A, start: 0.00, bitrate: 705 kb/s
>>   Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
>> Output #0, mp3, to 'duration.mp3':
>> Metadata:
>>   TSSE: Lavf57.62.100
>>   Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, mono, s16p
>>   Metadata:
>> encoder : Lavc57.74.100 libmp3lame
>> Stream mapping:
>> Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
>> Press [q] to stop, [?] for help
>> size=2344kB time=00:05:00.01 bitrate=  64.0kbits/s speed= 128x
>> video:0kB audio:2344kB subtitle:0kB other streams:0kB global headers:0kB 
>> muxing overhead: 0.009457%
>> 
>> And when I run the following command, the resulting duration.mp3 is CBR.  
>> Confirmed with same program as above.
>> 
>> $ ffmpeg -f lavfi -i sine=d=300 -write_xing 0 duration.mp3
>> ffmpeg version N-83152-gf7e9275 Copyright (c) 2000-2017 the FFmpeg developers
>> built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
>> configuration: --prefix=/Volumes/ffmpeg_compile --pkg-config-flags=--static 
>> --disable-ffserver --enable-gpl --enable-version3 --enable-nonfree 
>> --enable-libmp3lame --enable-libfdk-aac --enable-libvo-amrwbenc 
>> --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvorbis 
>> --enable-libspeex --enable-libopus --enable-libgsm --enable-libtwolame 
>> --enable-libsoxr --enable-libwavpack --enable-libmodplug 
>> --enable-libopenjpeg --enable-libwebp --enable-libtheora --enable-libx264 
>> --enable-libopenh264 --enable-libx265 --enable-libxvid --enable-libvpx 
>> --enable-libxavs --enable-libfreetype --enable-fontconfig 
>> --enable-libfribidi --enable-libass --enable-libbluray --enable-libvidstab 
>> --enable-libschroedinger --enable-libsnappy --enable-libzmq
>> libavutil  55. 43.100 / 55. 43.100
>> libavcodec 57. 74.100 / 57. 74.100
>> libavformat57. 62.100 / 57. 62.100
>> libavdevice57.  2.100 / 57.  2.100
>> libavfilter 6. 69.100 /  6. 69.100
>> libswscale  4.  3.101 /  4.  3.101
>> libswresample   2.  4.100 /  2.  4.100
>> libpostproc54.  2.100 / 54.  2.100
>> Input #0, lavfi, from 'sine=d=300':
>> Duration: N/A, start: 0.00, bitrate: 705 kb/s
>>   Stream #0:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s
>> File 'duration.mp3' already exists. Overwrite ? [y/N] y
>> Output #0, mp3, to 'duration.mp3':
>> Metadata:
>>   TSSE: Lavf57.62.100
>>   Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, mono, s16p
>>   Metadata:
>> encoder : Lavc57.74.100 libmp3lame
>> Stream mapping:
>> Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
>> Press [q] to stop, [?] for help
>> size=2344kB time=00:05:00.01 bitrate=  64.0kbits/s speed= 134x
>> video:0kB audio:2344kB subtitle:0kB other streams:0kB global headers:0kB 
>> muxing overhead: 0.001875%
> 
> The command which produces VBR mp3 file, "$ ffmpeg -f lavfi -i sine=d=300 
> duration.mp3"'s duration is 5:43 on Mac and is 5:01 on Windo

[FFmpeg-user] anequalizer example error on doc site and manpage

[DopeLabs]

i noticed a small error in the example provided for the anequalizer filter…

ffmpeg . org/ffmpeg-filters.html#Examples-7
ffmpeg . org/ffmpeg-all.html#Examples-42
$ man ffmpeg-all


the params specify:

cf
Set central frequency for band. If input doesn’t have that frequency the entry 
is ignored.

…

f
Set filter type for band, optional, can be:

‘0’
Butterworth, this is default.

‘1’
Chebyshev type 1.

‘2’
Chebyshev type 2.



yet the example shows:

anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1


which should actually be:

anequalizer=c0 cf=200 w=100 g=-10 f=1|c1 cf=200 w=100 g=-10 f=1


cheers,

DL
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Re: [FFmpeg-user] Problem with a YT video

[DopeLabs]

be sure you are running the latest YouTube-DL as well. -U to update. I've had 
similar errors which worked after the update.

Sent from my iPhone

> On Apr 27, 2016, at 6:30 PM, Xen  wrote:
> 
> jd1008 schreef op 27-04-2016 23:35:
>> Hi Moritz,
>> How do I use the command to save the output file instead of viewing it?
>> Thanx!!!
> 
> I haven't followed this as I just arrived on this list. But.
> 
> Normally if you were downloading from a video site you'd use youtube-dl, 
> right?
> 
> Maybe I should be reading this thread online, but.
> 
> Normally "youtube-dl -F " would show all the available formats.
> 
> Then "youtube-dl -f  " would download a format specified by 
> number as you got it from the previous command.
> 
> The URL specified in below message gives me a 403 error though (forbidden).
> 
> When you do -F the first list gives audio-only formats, the middle part gives 
> video-only formats, and the remaining part gives both. Usually there will be 
> something like "webm" and "mp4" in it.
> 
> Just some common knowledge here, like I said I haven't been able to read the 
> rest yet.
> 
> Regards.
> 
> 
> 
>>> On 04/24/2016 11:02 AM, Moritz Barsnick wrote:
 On Sun, Apr 24, 2016 at 10:46:50 -0600, jd1008 wrote:
 OK, that plays one screen. Well and good !!!
 I had to download it to get one viewing screen.
>>> How did your download manage to cut it in half? BTW, you can feed the
>>> stream URL directly to ffplay (your URL may vary):
>>> $ ffplay -i 
>>> 'https://r6---sn-5hnedn7s.googlevideo.com/videoplayback?mime=video%2Fmp4&dur=9590.119&upn=jYWGghj_TK4&itag=22&ip=80.144.60.117&ratebypass=yes&mn=sn-5hnedn7s&mm=31&source=youtube&expire=1461538813&pl=14&mv=m&mt=1461517049&ms=au&id=o-AImla_PoEMqA6TrhRrLn4QozlmIZAZYKbo8o_jfqycn9&initcwndbps=247500&key=yt6&sparams=dur%2Cid%2Cinitcwndbps%2Cip%2Cipbits%2Citag%2Clmt%2Cmime%2Cmm%2Cmn%2Cms%2Cmv%2Cnh%2Cpl%2Cratebypass%2Crequiressl%2Csource%2Cupn%2Cexpire&lmt=1460246095785485&requiressl=yes&ipbits=0&nh=IgpwcjAxLmFtczE2KgkxMjcuMC4wLjE&fexp=9405960%2C9416126%2C9416891%2C9420095%2C9421503%2C9422596%2C9424134%2C9426926%2C9428398%2C9431012%2C9431708%2C9431814%2C9432367%2C9432683%2C9432751%2C9433096%2C9433180%2C9433947%2C9434126%2C9434204&signature=0559118CDA43302FF412EB7DA728B559D0265AE1.B86A5764633D44AE3165B71ECAFE036563DA7786&sver=3'
>>>  -vf "crop=w=iw/2:h=ih:x=0:y=0,scale=w=2*iw:h=ih,setdar=2*dar"
 BUT - No Audio :( :(
>>> Mine has. Perhaps your download tool downloaded the video-only stream.
>>> (Have you checked "youtube-dl -F"?)
 PS: Even if you play it in the browser , it is 2 virtual screens.
>>> Not here. YouTube recognizes it as 3D and does some magic when playing
>>> (in Firefox). (Note, I think I am using YouTube without Flash. Not that
>>> it should matter.)
 on my laptop using any one of the linux media players - but without the
 dual viewing windows.
>>> What are those Linux YouTube players? smtube possibly?
>>> You're still not correctly writing what you're trying to achieve and
>>> what you're actually doing. Like really doing.
>>> Moritz
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