GStreamer
Índex
- General
- Instal·lació / Installation
- Ús / Usage
- Desenvolupament / Development
-
- Documentació / Documentation
- Llenguatges / Languages
- gst
- Resum / Summary
- Application
Development
Manual
- About GStreamer
- Building an application
- Advanced
GStreamer
concepts
- Position tracking and seeking
- Metadata
- Interfaces
- Clocks and synchronization in GStreamer
- Buffering
- Dynamic controllable parameters
- Threads
- Autoplugging
- Pipeline manipulation
- Higher-level interfaces for GStreamer applications
- Appendices
- Tutorials
- Encoding
- Encoding target
- Encoding profile
- Encoding preset
- gst-editing-services (GES)
- gst-transcoding
- Projectes / Projects
- Web overlay
- Live mixers
General
- GStreamer (.com) (wp)
- Releases
- Latest stable release: 1.24.2
- Latest development release: git master
- Plug-ins
- GStreamer SDK documentation
- Development
-
- Developer information
-
- modules
- stable version
- modules
- cgit.freedesktop.org/gstreamer/
- Releases
- Gstreamer cheat sheet
- Introducing GStreamer (CInLUG)
Instal·lació / Installation
- From packages
-
- Mageia
-
urpmi ... gstreamer1.0-plugins-bad gstreamer1.0-plugins-ugly gstreamer1.0-x264
- Mono repository: compilació des de codi
font / compilation
from source
- (28th September 2021) replaces gst-build and multiple modules
- GStreamer mono repository FAQ
- Dependències / Dependencies
GitLab dependencies
Mageia
Alma 8
Ubuntu (22.04)
packages subproject package subproject package subproject general
dnf install git ninja
sudo pip install --upgrade meson
pipx install meson
- orc
dnf -y install epel-release
dnf config-manager --set-enabled powertools
# for libx264-devel, fdk-aac-devel:
dnf -y install --nogpgcheck https://mirrors.rpmfusion.org/free/el/rpmfusion-free-release-8.noarch.rpm https://mirrors.rpmfusion.org/nonfree/el/rpmfusion-nonfree-release-8.noarch.rpm
dnf install git python3 python3-pip ninja-build
pip3 install meson
apt install git python3 python3-pip ninja-build
gstreamer gstreamer dnf install flex bisonlib64girepository-devel
dnf install gcc-c++ glib2-devel cmake flex bison gtk3-devel libunwind-devel gmp-devel gsl-devel gobject-introspection-devel bash-completion libcap-devel elfutils-devel
1.24.8: gstreamer| Dependency glib-2.0 found: NO. Found 2.56.4 but need: '>= 2.64.0'
1.22.12: gstreamer| Dependency glib-2.0 found: NO. Found 2.56.4 but need: '>= 2.62.0'
apt install g++ libglib2.0-dev-bin cmake flex bison libgtk-3-dev libunwind-14-dev libgmp-dev libgsl-dev gobject-introspection bash-completion libcap-dev libdw-dev
gst-plugins-base gst-plugins-base dnf install lib64ogg-devel lib64opus-devel lib64vorbis-devel- gl-headers
- graphene
dnf install libogg-devel opus-devel libvorbis-devel libtheora-devel SDL2-devel iso-codes-devel libgudev-devel mesa-libgbm-devel alsa-lib-devel cdparanoia-devel libjpeg-turbo-devel
# do not activate raven, as it messes up with x264-devel and installs ffmpeg-libraries
# dnf install https://pkgs.dyn.su/el8/base/x86_64/raven-release-1.0-3.el8.noarch.rpm
# dnf install graphene-devel- gl-headers
- graphene
apt install zlib1g-dev libopus-dev libogg-dev libvorbis-dev libtheora-dev libsdl2-dev iso-codes libgudev-1.0-dev libgbm-dev libasound2-dev libcdparanoia-dev libjpeg-turbo8-dev... gst-plugins-good gst-plugins-good dnf install lib64mp3lame-devel lib64dv-devel lib64jpeg-devel lib64qt5platformsupport-devel
- libnice
dnf install libsoup-devel nasm libnice-devel libvpx-devel pulseaudio-libs-devel lame-devel libdv-devel libcaca-devel libv4l-devel flac-devel jack-audio-connection-kit-devel libshout-devel
optional:dnf installqt5-devel- libnice
apt install libsoup2.4-dev nasm libnice-dev libvpx-dev libpulse-dev libmp3lame-dev libdv-dev libcaca-dev
apt install libv4l-dev libflac-dev libshout-dev... gst-plugins-bad gst-plugins-bad dnf install lib64openjpeg2-devel lib64microdns-devel lib64fdk-aac-devel nasm- avtp
- dssim
- openh264
dnf -y install --nogpgcheck https://mirrors.rpmfusion.org/free/el/rpmfusion-free-release-8.noarch.rpm https://mirrors.rpmfusion.org/nonfree/el/rpmfusion-nonfree-release-8.noarch.rpm
dnf install fdk-aac-devel libmicrodns-devel openjpeg2-devel libva-devel libdrm-devel libass-devel opencv-devel- avtp
- openh264
- dssim
apt install libdrm-dev libass-dev libopencv-dev libva-dev libfdk-aac-dev libopenjp2-7-dev... gst-plugins-ugly gst-plugins-ugly dnf install lib64x264-devel
dnf install libmpeg2-devel x264-devel
to activate x264 and mpeg2dec, meson must be called with -Dgpl=enabled
apt install libmpeg2-4-dev
gst-libav gst-libav only when not using ffmpeg compiled from source:
dnf install lib64ffmpeg-devel
if you must compile ffmpeg, do it before compiling gstreamer, so as it does not install ffmpeg as a subproject; but you must call meson with:
PKG_CONFIG_PATH=/usr/local/lib/pkgconfig/ meson ...
gst-rtsp-server
gst-devtools
dnf install lib64json-glib-devel
dnf install json-glib-devel
apt install libjson-glib-dev
gst-integration-testsuites
gst-editing-services gst-editing-services dnf install python3-gobject-devel
dnf install pygobject3-devel python3-cairo-devel
apt install python-gi-dev python3-cairo-dev
gstreamer-vaapi
(disabled)
gst-omx
(disabled)
gstreamer-sharp
(disabled)
gst-python gst-python
gst-examples
dnf install lib64soup-devel
gst-plugins-rs
- Passos / Steps
sudo sh -c 'echo "/usr/local/lib64" > /etc/ld.so.conf.d/local64.conf'
sudo ldconfig
mkdir -p ~/src && cd ~/srcgit clone https://gitlab.freedesktop.org/gstreamer/gstreamer.gitcd gstreamer- if you compiled ffmpeg:
- if you installed ffmpeg-devel:
sudo ninja -C build install- sudo ldconfig
- to uninstall
sudo ninja -C build uninstall
- Problems
- ...
- check
jq '' ./build/meson-info/intro-installed.json
- check
- ges-timeline.h:31: syntax error, unexpected ';' in '# 31
- ...
- ...
- Compilació des de codi font / Compilation from source
-
- Mòduls i dependències / Modules and dependencies
-
-
module desc
GitLab dependencies
Mageia ... CentOS 7 Ubuntu
/data/doc/gstreamer/head
urpmi ...
yum install ...
apt-get install ...
gstreamer
automake autoconf gettext-devel libtool bison flex gtk-doc yasm
autoconf gettext-devel libtool bison flex gtk-doc yasm glib2-devel gcc-c++ freetype freetype-develautoconf bison flex...
gst-plugins-base
lib64opus-devel libvorbis-devel libogg-devel libtheora-devel libxv-devel
opus-devel libvorbis-devel libogg-devel libtheora-devel libxv-devel pango-devel wayland-devellibopus-dev libvorbis-dev libogg-dev libtheora-dev libxv-dev libpango1.0-dev
gst-plugins-good
libvpx-devellibsoup-devel
libvpx-devel pulseaudio-libs-devel libsoup-devellibvpx-devgst-plugins-bad
gst-plugins-bad librtmp-devel(from source: srt)
librtmp-devel(from source: srt)librtmp-dev(from source: srt)gst-plugins-ugly
libx264-devel
libx264-devellibx264-devgst-python
python-gobject3-devel
python-devel pygobject3-devel
gst-libav
gstreamer-editing-services
gst-editing-services libxml2-devel
libxml2-devel
- Mageia
-
urpmiautoconf gettext-devel libtool bison flex gtk-doc yasm- For plugins-base:
-
urpmi lib64opus-devel lib64vorbis-devel lib64ogg-devel lib64theora-devel lib64xv-devel libsoup-devel
- Raspbian
-
- ...
- CentOS
-
- CentOS 8
- see gst-build
- CentOS 7
- gstreamer
- automake >=1.14 (CentOS 7 provides version 1.13)
yum install -y autoconf gettext-devel libtool bison flex gtk-doc yasm glib2-devel gcc-c++ freetype freetype-devel
- gst-plugins-base
-
yum install opus-devel libvorbis-devel libogg-devel libtheora-devel libxv-devel pango-devel
- gst-plugins-good
-
yum install libvpx-devel
- gst-plugins-bad
-
yum -y install http://li.nux.ro/download/nux/dextop/el7/x86_64/nux-dextop-release-0-5.el7.nux.noarch.rpmyum -y install librtmp-devel
- gst-plugins-ugly
-
yum install libx264-devel
- gstreamer
- CentOS 8
-
- Modules
-
module
/usr/lib64/
/usr/local/lib/
/usr/lib64/gstreamer-1.0/
/usr/local/lib/gstreamer-1.0//usr/lib64/girepository-1.0/
/usr/local/lib/girepository-1.0/
/usr/share/gir-1.0/
/usr/local/share/gir-1.0/
/usr/lib64/pkgconfig/
/usr/local/lib/pkgconfig/gstreamer
packages
urpmi lib64gstreamer1.0_0 urpmi lib64gst-gir1.0
urpmi lib64gstreamer1.0-devel files
- libgstreamer-1.0.so
- libgstbase-1.0.so
- libgstcheck-1.0.so
- libgstcontroller-1.0.so
- libgstnet-1.0.so
- libgstcoreelements.so
- libgstcoretracers.so
- Gst-1.0.typelib
- GstBase-1.0.typelib
- GstController-1.0.typelib
- GstNet-1.0.typelib
- GstCheck-1.0.typelib
- Gst-1.0.gir
- GstBase-1.0.gir
- GstController-1.0.gir
- GstNet-1.0.gir
- GstCheck-1.0.gir
- gstreamer-1.0.pc
- gstreamer-base-1.0.pc
- gstreamer-check-1.0.pc
- gstreamer-controller-1.0.pc
- gstreamer-net-1.0.pc
gst-plugins-base packages
urpmi lib64gstreamer-plugins-base1.0_0
urpmi lib64gstreamer-plugins-base-gir1.0
files
gst-plugins-good packages urpmi gstreamer1.0-plugins-good
files
gst-plugins-bad packages urpmi gstreamer1.0-plugins-bad urpmi lib64gstreamer-plugins-bad-gir1.0 lib64gstreamer-plugins-bad1.0-devel
files
- GstGL-1.0.typelib
- GstInsertBin-1.0.typelib
- GstMpegts-1.0.typelib
- GSTGL-1.0.gir
- GstInsertBin-1.0.gir
- GstMpegts-10.0.gir
- GstPlayer-1.0.gir
- gstreamer-bad-audio-1.0.pc
- gstreamer-bad-base-1.0.pc
- gstreamer-bad-video-1.0.pc
- gstreamer-codeparsers-1.0.pc
- gstreamer-gl-1.0.pc
- gstreamer-insertbin-1.0.pc
- gstreamer-mpegts-1.0.pc
- gstreamer-player-1.0.pc
- gstreamer-plugins-bad-1.0.pc
gst-plugins-ugly packages urpmi gstreamer1.0-plugins-ugly
files
gst-python packages
files
gst-libav packages
files
gstreamer-editing-services
packages
urpmi lib64ges1.0-devel files
- libges-1.0.so
- GES-1.0.typelib
- GES-1.0.gir
- gst-editing-services-1.0.pc
-
- From git
- From tar files
-
- gst-build
(NOTE: replaced by mono
repo; NOTE: this is the default method
from version ==1.18)
- Getting started with GStreamer's gst-build (Collabora)
- meson
- list available options
meson configuremeson configure gst-python
- get present values for options
meson configure build
- set options
meson -D...=... build- ...
- list available options
- ninja
- build all
ninja -C build
- clean all
ninja -C build clean
- install
sudo ninja -C build install
- uninstall
sudo ninja -C build uninstall
- build all
- Dependencies
-
Project name CentOS 8 ...
dnf -y install ...
(repos) dnf -y install epel-release
dnf config-manager --set-enabled powertools
dnf -y install --nogpgcheck https://mirrors.rpmfusion.org/free/el/rpmfusion-free-release-8.noarch.rpm https://mirrors.rpmfusion.org/nonfree/el/rpmfusion-nonfree-release-8.noarch.rpm
All GStreamer modules dnf install git python3 gcc-c++ ninja-build
pip3 install --user meson
orc -
gstreamer glib2-devel cmake flex bison gtk3-devel
gst-plugins-base zlib-devel opus-devel libogg-devel libvorbis-devel libtheora-devel qt5-devel SDL2-devel iso-codes-devel(meson -Dexamples=disabled -Dtests=disabled) (qt5-devel breaks compilation on AWS EC2 instance)
- gl-headers -
- graphene
- mutest (?)
gst-plugins-good libsoup-devel libjpeg-turbo-devel nasm libnice-devel libvpx-devel pulseaudio-libs-devel
libnice openssl-devel
gst-plugins-bad libdrm-devel libass-devel librtmp-devel opencv-devel gobject-introspection-devel libmicrodns-devel libva-devel ...
- libdrm
- libavtp libbs2b-devel lcms2-devel libcurl-devel libdca-devel faac-devel faad2-devel fdk-aac-devel srt-devel libsrtp-devel x265-devel libexif-devel
- dssim
- microdns
- openh264 gtest-devel
- gtest
- openjp2 wxBase3-devel wxGTK3-devel libtiff-devel
gst-plugins-ugly libmpeg2-develx264-devel
gst-libav
- FFmpeg libgcrypt-devel twolame-libs
gst-rtsp-server libcgroup
gst-devtools json-glib-devel
- json-glib
gst-integration-testsuites
gst-editing-services
pygobject pygobject3-devel
- pycairo python3-cairo
platform-python-devel
gst-python "Python dynamic library path could not be determined"
Solution:
meson.build:
- ['gst-python', { 'option': get_option('python')}],
+ ['gst-python', { 'option': get_option('python'), 'libpython-dir':'/usr/lib/'}],
or:
meson -Dgst-python:libpython-dir=/usr/lib/ build
gst-examples
- CentOS
- CentOS 8
# gstreamer
sudo dnf install ninja-build git gcc-c++ libmount-devel flex bison glib2-develpython3-cairo libpng-devel libpciaccess-devel nasm cairo-devel wxBase3-devel
pip3 install --user meson- #
gst-plugins-base
sudo dnf install ... - #
gst-plugins-base
... - #
gst-plugins-good
sudo dnf installlibsoup-devel - #
gst-plugins-bad
... - #
gst-plugins-ugly
... - #
gst-python
sudo dnf installpygobject3-devel
- CentOS 7
sudo yum install python3 ninja-build git- Gstreamer module dependencies
- CentOS 8
- Mageia
sudo urpmi git python3 meson ninja orc- Gstreamer module dependencies
pip3 install --user meson
-
- Passos / Steps
cd ~/srcgit clone https://gitlab.freedesktop.org/gstreamer/gst-buildcd gst-buildmeson build --buildtype=debugninja -C buildsudo ninja -C build install
- Checkout
another branch using worktrees
cd ~/srcgit clone https://gitlab.freedesktop.org/gstreamer/gst-buildcd gst-build- ./gst-worktree.py add gst-build-1.18 origin/1.18
- cd gst-build-1.18
- meson
build
- meson -Dgst-python:libpython-dir=/usr/lib/ -Dtests=disabled -Dexamples=disabled build
- ninja -C build
sudo pip3 install mesonsudo ninja -C build installsudo sudo sh -c 'echo "/usr/local/lib64" >/etc/ld.so.conf.d/local.conf'sudo ldconfig- uninstall
- cd gst-build-1.18
- sudo ninja -C build uninstall
- Optional Installation
- Problemes / Problems
syntax error, unexpected ';' in ...- syntax error when ninja -C builddir
- Solució
- ...
- gst-uninstalled
- use gst_build instead
- https://gstreamer.freedesktop.org/src/
- if you want to be able to access GStreamer by using PyGObject (applications made in Python):
-
- Dependencies
-
- Mageia
-
urpmi lib64girepository-devel
- CentOS
-
sudo yum install gobject-introspection-devel
- Check config.log
-
HAVE_INTROSPECTION_TRUE=''
INTROSPECTION_COMPILER='/usr/bin/g-ir-compiler'
...
- Check that these files exist after compilation:
-
/usr/local/lib/girepository-1.0/Gst*.typelib
- In order to access Gst from your applications, you
will need to set environment variable (to avoid
error:
ValueError: Namespace Gst not available): -
export GI_TYPELIB_PATH=/usr/local/lib/girepository-1.0
gstreamer_install.sh1.16.2- gstreamer_install.sh
-
#!/bin/bash -e
# defaults
modules="gstreamer gst-plugins-base gst-plugins-good gst-plugins-bad gst-plugins-ugly gst-python gst-libav gstreamer-editing-services"
EXPECTED_ARGS=2
if (( $# != $EXPECTED_ARGS ))
then
cat <<EOF
Usage: `basename $0` [get,install,uninstall] version
Modules that will be built:
${modules}
Dependencies:
- gstreamer: automake autoconf gettext-devel libtool bison flex gtk-doc yasm glib2-devel gcc-c++ freetype freetype-devel
- plugins-base: libogg-devel libtheora-devel libvorbis-devel opus-devel wayland-devel
- plugins-good: libvpx-devel>=1.4.0 pulseaudio-libs-devel libsoup-devel
- plugins-bad: http://li.nux.ro/download/nux/dextop/el7/x86_64/nux-dextop-release-0-5.el7.nux.noarch.rpm librtmp-devel libsrtp-devel>=2.1.0 libcurl-devel>=7.35.0 fdk-aac>=2.0.0 libxml-2.0-devel>=2.9.2 libnice-devel>=0.1.14
- plugins-ugly: libx264-devel
- gst-python: python-devel pygobject3-devel
- gstreamer-editing-services: libxml2-devel
Examples:
- `basename $0` get 1.16.2
- `basename $0` install 1.16.2
- `basename $0` uninstall 1.16.2
EOF
exit 1
fi
# parameters
action=$1
version=$2
# update ldconfig
sudo sudo sh -c 'echo "/usr/local/lib" > /etc/ld.so.conf.d/local.conf'
sudo ldconfig
mkdir -p gst-${version}
cd gst-${version}
for module in $modules
do
src_name="${module}-${version}"
echo "============================================================= $src_name ============================================================="
tar_filename="${src_name}.tar.xz"
case $action in
get)
url=https://gstreamer.freedesktop.org/src/${module}/${tar_filename}
echo " getting from: ${url}"
curl -s -L -O ${url}
;;
install)
echo " installing from: ${tar_filename}"
tar xJf ${tar_filename}
export XDG_DATA_DIRS="/usr/local/share/:/usr/share/"
cd ${src_name}
./autogen.sh PKG_CONFIG_PATH=/usr/local/lib64/pkgconfig/:/usr/local/lib/pkgconfig/ --disable-gtk-doc
make
sudo make install
sudo ldconfig
cd ..
;;
uninstall)
echo " uninstalling: ${src_name}"
cd ${src_name}
sudo make uninstall
cd ..
;;
esac
done
exit 0
- Compilation problems
- plugins-good
gstvp9dec.c: In function 'gst_vp9_dec_get_valid_format':
gstvp9dec.c:148:14: error: 'vpx_image_t' has no member named 'cs'
if (img->cs == VPX_CS_SRGB)- Solució / Solution:
- install libvpx-devel >=1.4.0
- vpx_image.h
- CentOS from repositories: libvpx-devel 1.3.0-5.el7_0
- install libvpx-devel >=1.4.0
- Solució / Solution:
- plugins-bad:
- gstsrtp.c:
In function 'set_crypto_policy_cipher_auth':
gstsrtp.h:68:28: error: 'AES_128_GCM' undeclared (first use in this function)
# define SRTP_AES_GCM_128 AES_128_GCM- Solució / Solution
- srtp: Add support for GCM (RFC 7714)
- install libsrtp-devel
>=2.1.0 (1.5.0)
- CentOS from repositories: 1.4.4-10.20101004cvs.el7
- Solució / Solution
- gstsrtp.c:
In function 'set_crypto_policy_cipher_auth':
- plugins-good
- gstreamer will be installed in:
-
/usr/local/lib/gstreamer-1.0/
- gst-build
(NOTE: replaced by mono
repo; NOTE: this is the default method
from version ==1.18)
- Incidències / Issues (GitLab Issues) (others: gstreamer-devel list, Bugs)
-
- Issues
- HLS seek
- 1.16.0
- install wayland-devel
- autotools
gtk-doc scanner build failure: undefined reference
to `gst_gl_display_wayland_get_type'
- Solució / Solution
./autogen.sh PKG_CONFIG_PATH=/usr/local/lib/pkgconfig/ --disable-gtk-doc
- Solució / Solution
.../tmp-introspectAx4R5G/.libs/lt-GstMpegts-1.0: error while loading shared libraries: libgstvideo-1.0.so.0: cannot open shared object file: No such file or directory- Solution
sudo ldconfig
- Solution
(gst-plugin-scanner:21672): GStreamer-WARNING **: 15:56:30.108: Failed to load plugin '/usr/local/lib/gstreamer-1.0/libgstpango.so': /lib64/libcairo.so.2: undefined symbol: FT_Get_Var_Design_Coordinatesgst-inspect-1.0 timeoverlay- Solució / Solution
- update freetype to 2.8
- Opus audio encoder not found
-
gst-plugins-bad/ext/opus/opusenc.cwas present until version 1.6; then it was moved to plugins-base- Solució / Solution
-
- install system-wide Opus devel libraries and reconfigure and make plugins-base
Ús / Usage
- Edició / Edit
- Info
- Eines / Tools
- ges-launch
- gst-launch
- Desenvolupament / Development
- ...
- Acceleració / Acceleration
- Raspberry Pi
-
- Hardware Video Encoding progess with the Raspberry Pi
- GStreamer 1.2 on the Raspberry Pi
- Using Gstreamer with OpenMax in a Raspberry Pi
- Hardware-accelerated video playback on the Raspberry Pi
- Streaming with nginx-rtmp-module: GStreamer and Raspberry Pi
- Using GStreamer as transcoder for Dreambox or TVheadend
- Usage of hardware-accelerated encoding/decoding:
-
- decoding:
-
- omxmpeg2videodec (license needed)
- omxh264dec
- encoding:
-
- omxh264enc
-
gst-inspect-1.0 omxh264enc- Using GStreamer as transcoder for Dreambox or TVheadend
-
- Bug 720031 - omxh264enc: key frame interval missing
- solution to missing SPS/PPS
-
- http://pastebin.com/b5avcxpm (already in the latest git version)
- Tools
(CLI)
- How do I use the GStreamer command line interface ?
- Command line tools
- gst-validate
- ges-launch
-
- GES in Python
- GES development
- Install
- Mageia
urpmi gstreamer1.0-nle gstreamer1.0-editing-services
- Ubuntu
- sudo apt-get install ges1.0-tools ...
- Mageia
- Help
ges-launch-1.0 -h
- ges-videocrop-effect.sh
- Sintaxi / Syntax
-
ges-launch-1.0 --help-all
exemples python equivalent Project Options
-l, --load=<path>-s, --save=<path>- --save-only=<path>
-p --sample-path-r --sample-path-recurse
Rendering Options
-o --outputuri=<uri>()-f --format=<profile>(specified serialized encoding-profile; if not specified: application/ogg:video/x-theora:audio/x-vorbis)-e --encoding-profile=<profile-name>(from a preset file)--smart-rendering
--format=...
--smart-renderingpipeline.set_mode(GES.PipelineFlags.SMART_RENDER)
Playback Options
-v --videosink=<videosink>-a --audiosink=<audiosink>-m --mute
Informative Options
--inspect-action-type=--list-transitions
Application Options
--disable-mixing-r --repeat=-t, --track-types=<track-types>- --video-caps
- --audio-caps
--set-scenario
--track-types="audio"
--video-caps="video/x-raw,width=640,height=272"
- --disable-mixing
tracks = timeline.get_tracks()
for track in tracks:
track.set_mixing(False)
- --video-caps=...
- set_restriction_caps
- NOTE: Restriction caps are not taken into account when using mode=#GES_PIPELINE_MODE_SMART_RENDER.
- set_restriction_caps
+clip<path|uri>name=track-types=inpoint[i]=(referred to clip)duration[d]=start[s]=(referred to timeline)layer[l]=set-alphaposxposywidthheightvolumemute
+clip toto.mp4- +clip toto.mp4 track-types=audio
+clip toto.mp4 set-alpha 0.9
+test-clip<test_clip_pattern>
smptecircular- ...
name=start=duration=inpoint=layer=
+effect+effect "agingtv"- +effect "agingtv" set-dusts false
- +effect "audiopanorama panorama=-1.00"
set-
+title<text>start=
duration[d]=- inpoint=
- track-types=
- layer=
+title "abc" duration=2.0
+track<track_type>restrictions=
+keyframes<property_name>- binding-type=
- interpolation-mode=
- ...
set-...
+clip /path/to/media +effect "agingtv" set-dusts false
-
- Exemples / Examples
- help
ges-launch-1.0--help-all
- play a clip from second 4.0 to second 6.0:
ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0
- play a clip from second 4.0 to second 6.0 and then
another clip from the beginning:
ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 +clip sintel_720p.mp4
- play a clip with a logo during 10 seconds on top
right:
ges-launch-1.0 +clip bbb_720p.mp4 +clip logo.jpeg s=0 d=10 set-alpha 0.8 set-width 200 set-height 100 set-posx 1000 set-posy 20
- play a title during 2 seconds, then a video, then
a title during 3 seconds:
ges-launch-1.0 +title "begin" duration=2.0 +clip toto.mp4 +title "end" duration=3.0
- play a clip with a simultaneous title
- ...
- play to a window with the same dimensions as clip
(otherwise, a 1280x720 window is created, the
default for VideoTrack)
- see issue #139
- if you know the dimensions of the source (e.g.
640x272)
ges-launch-1.0--disable-mixing--video-caps="video/x-raw,width=640,height=272" +clip video.mp4
- GESSmartMixer
- GstCompositor
- save project to play a clip from second 4.0 to
second 6.0:
ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 --save bbb.xges
- play according to project (can also be generated
by pitivi):
ges-launch-1.0 --loadbbb.xges
- render to file according to project:
# output is forced to 720p
ges-launch-1.0 --loadbbb.xges -o toto.mp4# output preserves input resolution
ges-launch-1.0 --loadbbb.xges--smart-rendering-o toto.mp4
- launch pitivi
with this project:
pitivibbb.xges
- render without reencoding (fast) (NOTE: not
working with HLS):
- get_smart_profile
ges-launch-1.0 --smart-rendering +clip bbb_720p.mp4 i=4.0 d=2.0 -o bbb.mp4- Problemes / Problems
- with m3u8 input
sys:1: Warning: g_source_remove: assertion 'tag > 0' failed- ...
- with m3u8 input
- render to Ogg - Theora - Vorbis (default encoding profile):
ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 -o bbb.oggges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 -o bbb.ogg -f "video/ogg:video/x-theora:audio/x-vorbis"
- render to WebM - VP8 - Vorbis:
ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 -o bbb.webm -f "video/webm:video/x-vp8:audio/x-vorbis"- Problemes / Problems
ERROR from element qtdemux1: Internal data stream error.- Solució / Solution:
urpmi gstreamer1.0-vp8
- Solució / Solution:
- render to MP4 - H.264 - MP3:
ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 -o bbb.mp4 -f "video/quicktime,variant=iso:video/x-h264:audio/mpeg,mpegversion=1,layer=3"
- render to MP4 - H.264 - AAC encoding
profile (Mageia: gstreamer1.0-plugins-bad...tainted):
ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 -o bbb.mp4 -f "video/quicktime,variant=iso:video/x-h264:audio/mpeg,mpegversion=4"- Problemes / Problems
Invalid format specified: video/quicktime,variant=iso:video/x-h264:audio/mpeg,mpegversion=4- Solution
urpmigstreamer1.0-x264urpmigstreamer1.0-fdkaac
- Solution
ERROR from element qtdemux1: Internal data stream error.- Solució / Solution
urpmigstreamer1.0-x264
- Solució / Solution
- specify output video dimensions, as a canvas
(clips will not be rescaled) (NOTE: this is needed
for gstreamer 1.18; if not specified, output will be
1280x720 (TBC))
ges-launch-1.0 ...-o bbb.mp4-f "video/quicktime,variant=iso:video/x-h264,width=1920,height=1080:audio/mpeg,mpegversion=1,layer=3"
- specify input video dimensions (clips will be
rescaled and output dimensions will be those of the
first clip)
ges-launch-1.0 +clip clip_1.mp4 set-width 1280 set-height 720 +clip clip_2.mp4 set-width 1280 set-height 720 ...
- ...
- help
- gst-transcoder
gst-transcoder-1.0 [OPTION?] <source uri> <destination uri> <encoding target name>[/<encoding profile name>]- Exemples / Examples
- Create a target file called device/mp4target.gep
gst-transcoder-1.0 <input_file> <output_file>.mp4 mp4target/mp4gst-transcoder-1.0 input.mp4 output.mkv matroska
- Problemes / Problems
WARN: ... no such element factory "uritranscodebin"!
- gst-discoverer
- gst-inspect
-
- list of all plug-ins
-
gst-inspect-1.0
- available properties for a specified plugin
-
gst-inspect-1.0 videoconvert...
- ...
- gst-launch (wp)
-
gst-launch-1.0 ... ! ... ! ...gst-launch-1.0 ... ! ... !...demux name=mydemux...mux name=mymux! ... ! ... mydemux. ! ... ! mymux. mydemux. ! ... ! mymux.-
input + demuxmux + outputaudiovideo
- Options
-
-e: end of stream on shutdown-f, --no_fault: ...--help: ...-q, --quiet: ...-m, --messages: ...-o FILE, --output=FILE: ...-t, --tags: ...-T, --trace: ...-v: verbose : ...--gst-debug-level=2
- Verbose messages (
-v, -q): sent to stdout (1)/<element>:<name>/<element>:<name>.<subelement>:<name>: <property>=<value>, <property>=<value> ...- gst_format_logs.sh
#!/bin/bash
input_path=$1
awk -F'\\\\ ' 'BEGIN {OFS="\n";ORS="\n\n"} $1 ~ /^\/GstPipeline/ {$1=$1;print $0}' ${input_path}
exit 0
- Debug: sent to stderr (2)
-
- Debugging tools
export GST_DEBUG=1 # default
export GST_DEBUG="*:2"
export GST_DEBUG=WARN,udpsrc:INFO,videodecoder:DEBUGexport GST_DEBUG=3,rtpjitterbuffer:3,rtpbasedepayload:6,videodecoder:4- export GST_DEBUG=3,hlsdemux:5 # to see the retrieved ts files
-
number
name
1
ERROR
2
WARNING
3
FIXME
4
INFO
5
DEBUG
6
LOG
7
TRACE
8
9
MEMDUMP
-
- categories (modules)
export GST_DEBUG=3,basesrc:4,basesink:4,...-
grep GST_DEBUG_CATEGORY_INIT -R gstreamer
examples (use grep to detect them) adapter
audio
audioencoder - opusenc0
audioresample - audioresample0
basesrc - fdsrc0
- shmsrc0
basesink - udpsink0
- udpsink1
- udpsink2
- udpsink3
basetransform - audioconvert0
- audioresample0
- capsfilter0
- capsfilter1
- videoconvert0
- videoscale0
bin
bufferpool - videobufferpool0
capsfilter - capsfilter0
- capsfilter1
- capsfilter2
default
hlsdemux
libav
opusenc - opusenc0
queue_dataflow - queue0
- queue1
query
rtpbin
rtpsession - rtpsession0:send_rtp_sink
- rtpsession1:send_rtp_sink
videodecoder
videoconvert - videoconvert0
videopool - videobufferpool0
videoscale - videoscale0
- Gstreamer pipeline diagram
-
- How to generate a Gstreamer pipeline diagram (graph)
- Dependencies
-
- Mageia
-
urpmi graphviz
- Utilització / Usage
- Sintaxi / Syntax
-
-
element:
ELEMENTTYPE [PROPERTY1 ...]
elements can be put into bins:
[BINTYPE.] ( [PROPERTY1...] PIPELINE-DESCRIPTION )
property:
NAME=*[(TYPE)]*VALUEin lists and ranges:*[(TYPE)]*VALUE- range:
[VALUE,VALUE] - list:
{VALUE[,VALUE...]}
-i int-f float-4 fourcc-b bool boolean-s str string-fraction
link:
[[SRCELEMENT].[PAD1,...]] ! [[SINKELEMENT].[PAD1,...]] [[SRCELEMENT].[PAD1,...]] ! CAPS ! [[SINKELEMENT].[PAD1,...]]
caps:
MIMETYPE [,PROPERTY[,PROPERTY...]]] [;CAPS[;CAPS...]]
- range:
-
input
demuxer
decoder
filter
encoder muxer
demux
buffer
parse
(to get specific packets from demuxer)
decode
filter
encode codec parameters
(CAPS)
parse
(to prepare specific packets for muxer)mux
output
file
- filesrc location=videofile
- uri=file:///path/to/test.ts
devices
- dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 pids=111:112
network
unix
- fdsrc fd=...
- shmsrc
socket-path=...
is-live=... - ... do-timestamp=...
decodebin name=decoder
(does not include sdpdemux)- corresponding parameters (caps) can be grouped, but must appear after filter declaration (except videoconvert: videoconvert ! <caps> ! videoconvert)
- if a value in caps must be used, the corresponding filter must be present. If the value in caps is not used, because the input already has this value, the filter is not needed
filter
description
mimetype, comma separated key=value
video
video/x-raw videoscale
width=360 height=288
pixel-aspect-ratio=1/1
videorate
framerate=25/1 videoconvert
format=BGRA ?
interlace-mode=progressive
audio
audio/x-raw
audiorate
Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream
tolerance=...
...
audioresample
Resamples audio
rate=48000
audioconvert Convert audio to different formats
format=S16LE
channels=2
layout=interleaved
- video
-
- x264enc bitrate=512 (kb/s)
- omxh264enc target-bitrate=100000 control-rate=variable inline-header=true periodicty-idr=250
- vp8enc target-bitrate=512000 (b/s)
- vp9enc target-bitrate=512000 (b/s)
- audio
- video
-
- video/x-h264,
profile=baseline
- video/x-h264,
- video
-
- h264parse
- audio
-
- mpegaudioparse
- aacparse
- mpegtsmux name=mux
- flvmux streamable=true name=mux
- mp4mux faststart=true
file
- filesink location=music.ogg
devices
- autovideosink
- xvimagesink
- autoaudiosink
- omxanalogaudiosink
network
- udpsink host=192.168.0.8 port=5004 sync=false
- rtmpsink location=rtmp://rtmp_server:1935/app/stream
unix
demux
specific stream (source Element Pads)
- demux.
matroskademux
- demux.audio_%u
- demux.video_%u
- demux.subtitle_%u
qtdemux name=demux - demux.video_0
- demux.audio_0
- demux.audio_1
- ...
sdpdemux name=demux - demux.stream_0
- demux.stream_1
- ...
tsdemux program-number=805 name=demux - demux.audio_xxxx
flvdemux name=demux
- audio
- video
...
- queue [leaky=...]
- multiqueue
- video
-
- mpegvideoparse
- h264parse
- audio
-
- mpegaudioparse
- aacparse
- decodebin
- video
-
- omxmpeg2videodec
- omxh264dec
- audio
-
- avdec_mpeg2video
- video
-
- videotestsrc [pattern=snow] [is-live=1] ...
- audio
-
- audiotestsrc [wave=5] [volume=0.8] [freq=200]
- video
-
- video/x-raw, framerate=25/1, width=640, height=360,
- if format is specified, videoconvert must be specified after it
-
- format=...,
...
- format=...,
- audio
-
- audio/x-raw,
format=...,
layout=...,
rate=...,
channels=...
- audio/x-raw,
-
-
sdpdemux
rtpbin
udpsrc
rtpsession
rtpssrcdemux
rtpjitterbuffer
rtpptdemux
- Sources and sinks
-
- GstBaseSrc
-
- do-timestamp
-
- it has to be specified at source if we want lipsync at the output: sync=true
- GstBaseSink
- fd (file descriptor) (see Snowmix audio)
-
- fdsink
-
- ...
- fdsrc
-
- ...
- Exemple / Example
- send audio to file descriptor 3 and listen
to it. Verbose and warning logs are shown in
terminal
export GST_DEBUG=WARN
AUDIOCAPS="audio/x-raw,format=S16LE,layout=interleaved,rate=44100,channels=2"
gst-launch-1.0 -v audiotestsrc wave=5 ! volume volume=0.1 ! ${AUDIOCAPS} ! fdsink fd=3 3>&1 1>&2 | gst-launch-1.0 fdsrc ! ${AUDIOCAPS} ! autoaudiosink
- send audio to file descriptor 3 and listen
to it. Verbose and warning logs are shown in
terminal
- shm (shared memory) (see Snowmix video)
-
- shmsink
-
gst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=25/1,width=640,height=480,format=BGRA! videoconvert !shmsink socket-path=/tmp/feed1 shm-size=`echo 640*480*4*22 | bc` wait-for-connection=0gst-launch-1.0 -vvideotestsrc is-live=true do-timestamp=true ! video/x-raw,framerate=25/1,width=640,height=480,format=BGRA ! videoconvert ! clockoverlay halignment=right valignment=top shaded-background=true font-desc="Sans, 24" ! shmsink socket-path=/tmp/feed1 shm-size=`echo 640*480*4*22 | bc` wait-for-connection=1 sync=truegst-launch-1.0 filesrc location=sintel_timecode_640x272_44100_stereo.mp4 ! qtdemux name=demux demux. ! decodebin ! videoconvert ! videoscale ! videorate ! video/x-raw,width=320,height=136,format=BGRA ! shmsink socket-path=/tmp/feed1 shm-size=`echo 320*136*4*22 | bc -l` wait-for-connection=1 sync=true
- shmsrc
-
- you must specify:
width, height, framerate, format+videoconvert, and they shoud match values specified in shmsink gst-launch-1.0 -v shmsrc socket-path=/tmp/feed1 do-timestamp=true is-live=true ! video/x-raw,width=640,height=480,framerate='25/1',format=BGRA ! videoconvert ! autovideosink- to play at a framerate different from the input, specify a different framerate and add videorate
- you must specify:
du -h /dev/shmls -l /dev/shmnetstat -pena --unix
- shm (video) + fd (audio)
-
send
play
Notes
audio
feed_rate=44100
feed_channels=2
AUDIOCAPS="audio/x-raw,format=S16LE,layout=interleaved,rate=${feed_rate},channels=${feed_channels}"
gst-launch-1.0 \
audiotestsrc wave=5 ! volume volume=0.1 ! ${AUDIOCAPS} ! fdsink fd=3 3>&1 1>&2 \
| gst-launch-1.0 fdsrc ! ${AUDIOCAPS} ! autoaudiosink
video
ratefraction="25/1"
feed_width=320
feed_height=180
video_input_sar="1:1"
pixel_aspect_ratio=${video_input_sar/:/\/} # replace : -> /
VIDEOCAPS="video/x-raw,framerate=${ratefraction},width=${feed_width},height=${feed_height},pixel-aspect-ratio=${pixel_aspect_ratio}"
FORMAT_SHM="BGRA"
VIDEOCAPS_WITH_FORMAT="${VIDEOCAPS},format=${FORMAT_SHM}"
shm_socket_path=/tmp/shm_toto
FORMAT_DISPLAY="I420"
VIDEOCONVERT_DISPLAY="video/x-raw,format=${FORMAT_DISPLAY}"
FORMAT_SHM="BGRA"
VIDEOCONVERT_SHM="video/x-raw,format=${FORMAT_SHM}"
gst-launch-1.0 \
videotestsrc ! ${VIDEOCAPS_WITH_FORMAT} ! videoconvert ! shmsink socket-path=${shm_socket_path} shm-size=`echo ${feed_width}*${feed_height}*4*22 | bc` wait-for-connection=0 \# send video with tee to monitor
gst-launch-1.0 \
filesrc location=${video_path} ! qtdemux name=demux \
demux.video_0 ! queue ! decodebin ! videoscale method=5 ! queue ! videorate ! ${VIDEOCAPS} ! tee name=tv \
tv. ! queue ! videoconvert ! ${VIDEOCONVERT_SHM} ! shmsink socket-path=${shm_socket_path} shm-size=`echo ${feed_width}*${feed_height}*4*22 | bc` wait-for-connection=0 \
tv. ! queue ! videoconvert ! ${VIDEOCONVERT_DISPLAY} ! autovideosink sync=true \
3>&1 1>&2
gst-launch-1.0 shmsrc socket-path=${shm_socket_path} do-timestamp=true is-live=true ! ${VIDEOCAPS_WITH_FORMAT} ! videoconvert ! autovideosink
audio + video
gst-launch-1.0 \
videotestsrc ! ${VIDEOCAPS_WITH_FORMAT} ! videoconvert ! shmsink socket-path=${shm_socket_path} shm-size=`echo ${feed_width}*${feed_height}*4*22 | bc` wait-for-connection=0 \
audiotestsrc wave=5 ! volume volume=0.1 ! ${AUDIOCAPS} ! fdsink fd=3 3>&1 1>&2 \gst-launch-1.0 \\
filesrc location=${video_path} ! qtdemux name=demux \
demux.video_0 ! queue ! decodebin ! videoscale method=5 ! queue ! videorate ! videoconvert ! ${VIDEOCAPS_WITH_FORMAT} ! \
shmsink socket-path=${shm_socket_path} shm-size=`echo ${feed_width}*${feed_height}*4*22 | bc` wait-for-connection=0 \
demux.audio_0 ! queue ! decodebin ! audioconvert ! audioresample ! queue ! audiorate ! $AUDIOCAPS ! tee name=ta \
ta. ! queue ! fdsink fd=3 \
3>&1 1>&2# send audio and video with tee to monitor
gst-launch-1.0 \
filesrc location=${video_path} ! qtdemux name=demux \
demux.video_0 ! queue ! decodebin ! videoscale method=5 ! queue ! videorate ! ${VIDEOCAPS} ! tee name=tv \
tv. ! queue ! videoconvert ! ${VIDEOCONVERT_SHM} ! shmsink socket-path=${shm_socket_path} shm-size=`echo ${feed_width}*${feed_height}*4*22 | bc` wait-for-connection=0 \
tv. ! queue ! videoconvert ! ${VIDEOCONVERT_DISPLAY} ! autovideosink sync=true \
demux.audio_0 ! queue ! decodebin ! audioconvert ! audioresample ! queue ! audiorate ! ${AUDIOCAPS} ! tee name=ta \
ta. ! queue ! autoaudiosink sync=true \
ta. ! queue ! fdsink fd=3 \
3>&1 1>&2 \
| gst-launch-1.0 \
shmsrc socket-path=${shm_socket_path} do-timestamp=true is-live=true ! ${VIDEOCAPS_WITH_FORMAT} ! videoconvert ! autovideosink \
fdsrc ! ${AUDIOCAPS} ! autoaudiosink
- es queden fitxers creats
(/tmp/shm_toto, /dev/shm/shmpipe*)
quan:
- no arrenca
- s'atura amb la X de la finestra de la sortida
- s'esborren bé els fitxers creats
quan:
- s'atura amb CTRL-C
-
-
- Demux
-
- sdpdemux
-
- Play from SDP file
- Includes
- Parameters
-
latency(ms)-
INFO rtpjitterbuffer gstrtpjitterbuffer.c:3942:do_deadline_timeout:<rtpjitterbuffer0> got deadline timeout
- Codecs
-
- x264enc
-
aud=false-
- do not generate NALs of type 9 (Access Unit Delimiter) (they generate problems when playing WebRTC on Chrome)
- Bins
-
-
playbinuridecodebin typefind queue2
- max-size-buffers=0
- max-size-bytes=230656
- use-buffering=TRUE
decodebin typefind - flvdemux
multiqueue
- aacparse
- avdec_aac
- h264parse
capsfilter
- avdec_h264
input-selector playsink
tee
streamsynchronizer
bin
queue
playsinkaudioconvert
audioconvert
audioresample
- pulsesink
bin
videoconvert
deinterlace
bin
queue
playsinkvideoconvert
videoconvert
videoscale
videobalance
videoconvert
- xvimagesink
input-selector - decodebin3
- rtpbin
-
- GStreamer pipeline recipe: Stream file via RTP using rtpbin (gst-launch)
- How te retrieve stream statistics in Gstreamer? (Python)
- Includes
-
- rtpsession
-
- rtcp-min-interval
- rtpssrcdemux
- rtpjitterbuffer
- rtpptdemux
-
rtpbin
Examples
sink
src
transmission
video
rtppaysend_rtp_sink_0send_rtp_src_0udpsink
send_rtcp_src_0udpsink sync=false async=false (SR)audio
rtppaysend_rtp_sink_1send_rtp_src_1udpsink
send_rtcp_src_1udpsink sync=false async=false (SR)reception
video
udpsrcrecv_rtp_sink_0recv_rtp_src_0_<ssrc_0>_<pt_0>rtpdepayudprsc (SR)recv_rtcp_sink_0send_rtcp_src_0udpsinksync=false async=false(RR)audio
udpsrcrecv_rtp_sink_1recv_rtp_src_1_<ssrc_1>_<pt_1>rtpdepayudpsrc (SR)recv_rtcp_sink_1send_rtcp_src_1udpsinksync=false async=false(RR)
-
- Play
-
- general: using
playbin -
gst-launch-1.0 -v playbin uri=...- Problemes / Problems:
map_vaapi_memory: failed to make image current- Solució / Solution
- Temporarily
skipping vaapi decoders in
decodebin
LIBVA_DRIVER_NAME=fakedriver gst-launch-1.0 playbin uri=file:///tmp/file.mp4
- Temporarily
skipping vaapi decoders in
decodebin
- Solució / Solution
-
gst-launch-1.0 -v playbin uri=...
gst-launch-1.0 uridecodebin uri=... name=decoder
playsink name=sink decoder.src_0 ! sink.video_sink decoder.src_1 ! sink.audio_sink gst-launch-1.0 rtmpsrc location=${source} ! queue2 ! decodebin name=decoder playsink name=sink decoder.src_0 ! sink.video_sink decoder.src_1 ! sink.audio_sink - from testsrc
-
- videotestsrc
-
- source code
- subprojects/gst-plugins-base/gst/videotestsrc/
- gstvideotestsrc.h
- gstvideotestsrc.c
- videotestsrc.h
- videotestsrc.c
- videotestsrc_blend_line
- subprojects/gst-plugins-base/gst/videotestsrc/
gst-launch-1.0 -v videotestsrc! video/x-raw,framerate=25/1,width=1280,height=720 ! autovideosink- using OpenGL (e.g. with Nvidia cards)
gst-launch-1.0 -v videotestsrc! video/x-raw,framerate=25/1,width=1280,height=720 ! glimagesink
gst-launch-1.0 -v videotestsrc pattern=snow ! video/x-raw,framerate=12/1,width=1280,height=720 ! autovideosinkgst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=12/1,width=1280,height=720,format=BGRA ! videoconvert ! autovideosink- test audio/video sync
- apply patch:
- usage:
- ...
- test judder
gst-launch-1.0 -v videotestsrc pattern=bar horizontal-speed=-40 ! video/x-raw,framerate=25/1,width=1280,height=720 ! autovideosinkgst-launch-1.0 -v videotestsrc pattern=bar horizontal-speed=-40 ! video/x-raw,framerate=60000/1001,width=1280,height=720 ! autovideosink
- clock overlay
- time overlay
-
gst-launch-1.0 -v videotestsrc is-live=true ! timecodestamper ! timeoverlay shaded-background=true 'time-mode=time-code' font-desc="Sans, 24" ! autovideosinkgst-launch-1.0 -v videotestsrc is-live=true ! video/x-raw, framerate=25/1, width=640, height=360 ! timecodestamper ! timeoverlay halignment=right valignment=bottom text="Stream time:" shaded-background=true font-desc="Sans, 24" ! autovideosink
- clock + time overlay
-
gst-launch-1.0 -v videotestsrc is-live=true ! video/x-raw, framerate=25/1, width=640, height=360 ! timecodestamper ! timeoverlay halignment=left valignment=top shaded-background=true font-desc="Sans, 24" ! clockoverlay halignment=right valignment=top shaded-background=true font-desc="Sans, 24" ! autovideosink
- write 10s (25*10=250 buffers) to an mp4
file:
gst-launch-1.0 -v videotestsrc pattern=sync num-buffers=250 ! video/x-raw,framerate=25/1,width=1280,height=720 ! x264enc ! video/x-h264,profile=high ! mux. mp4mux name=mux ! filesink location=sync.mp4
- ...
pattern
parameters
example
- background-color
- foreground-color
horizontal-speed specific to pattern
0 smpte SMPTE 100% color bars 
1
snow
2
black
3
white
4
red
5
green
6
blue
7
checkers-1
8
checkers-2
9
checkers-4
10
checkers-8
11
circular
12
blink
13
smpte75
14
zone-plate BBC R&D Report 1978/23 
- k0
- kt
- kt2
- kx
- kx2
- kxt
- ky
- ky2
- kyt
- xoffset
- yoffset
-
videotestsrc pattern=zone-plate kx2=20 ky2=20 kt=1
15
gamut
16
chrome-zone-plate
17
solid-color
18
ball
- animation_mode={frames,wall-time,running-time}
- flip (boolean)
- motion={wavy,sweep,hsweep}
videotestsrc pattern=ball animation-mode=wall-time motion=sweep
19
smpte100
20
bar
21
pinwheel
22
spokes
23
gradient
24 colors
25 smpte-rp-219
- source code
- audiotestsrc
-
gst-launch-1.0 -v audiotestsrc ! autoaudiosink- white noise (
wave=5), stereo (channels=2) -
gst-launch-1.0 -v audiotestsrc is-live=true wave=5 ! 'audio/x-raw,format=S16LE,layout=interleaved,rate=48000,channels=2' ! autoaudiosink
- white nois during 10s (audio:
10s*48000samples/s * 1 buffer/1024 samples =
468.75buffers)
gst-launch-1.0 -v audiotestsrc is-live=true wave=5 num-buffers=469 ! 'audio/x-raw,format=S16LE,layout=interleaved,rate=48000,channels=2' ! autoaudiosink
- wave
wave
parameters 0 sine
1 square
2 saw
3 triangle
4 silence
5 white-noise
6 pink-noise
7 sine-table
8 ticks - marker-tick-period
- marker-tick-volume
- sine-periods-per-tick
- tick-interval (ns)
9 gaussian-noise
10 red-noise
11 blue-noise
12 violet-noise
- test video + audio
-
gst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=25/1,width=1280,height=720 ! autovideosink audiotestsrc ! autoaudiosink- gst-launch-1.0
-v videotestsrc pattern=ball
animation-mode=wall-time motion=sweep !
autovideosink
audiotestsrc wave=ticks ! autoaudiosink - gst-launch-1.0
-v videotestsrc pattern=sync !
autovideosink
audiotestsrc wave=ticks ! autoaudiosink - write to a TS file:
gst-launch-1.0 -v videotestsrc pattern=sync ! video/x-raw,framerate=25/1,width=1280,height=720 ! x264enc ! video/x-h264,profile=high ! mux. audiotestsrc wave=ticks ! avenc_aac ! mux. mpegtsmux name=mux ! filesink location=sync.ts- Problemes / Problems
- avenc_aac not found
- verify:
gst-inspect-1.0 | grep avenc- sudo
dnf install
gstreamer1.0-libav
- if you installed ffmpeg aac after gstreamer, reinstall gstreamer1.0-libav
- sudo
dnf install
gstreamer1.0-libav
- ffmpeg -codecs | grep aac
- verify:
- avenc_aac not found
- write 10s to an mp4 file (video:
10s*25f/s=250 buffers; audio:
10s*48000samples/s * 1 buffer/1024 samples =
468.75buffers):
gst-launch-1.0 -v videotestsrc pattern=sync num-buffers=250 ! video/x-raw,framerate=25/1,width=1280,height=720 ! x264enc ! video/x-h264,profile=high ! mux. audiotestsrc wave=ticks num-buffers=469 ! avenc_aac ! mux. mp4mux name=mux faststart=true ! filesink location=sync.mp4
- from DVB device
-
- only video from DVB device (mpegvideoparse is needed because if not, maybe teletext is taken; and error is shown):
-
gst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=805 ! queue ! mpegvideoparse ! decodebin ! autovideosink
- only audio:
-
gst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=805 ! queue ! mpegaudioparse ! decodebin ! autoaudiosinkgst-launch-1.0 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux demux. ! queue ! mpegaudioparse ! decodebin ! omxanalogaudiosink
- audio and video from program 805 in DVB input:
-
gst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=805 name="demux" \
demux. ! queue ! mpegaudioparse ! decodebin ! autoaudiosink \
demux. ! queue ! mpegvideoparse ! decodebin ! autovideosink
- from file
-
gst-launch-1.0 -v playbin uri=file:/absoulte/path/to/your_video_filegst-launch-1.0 \
filesrc location=${video_path} ! decodebin name=dec \
dec. ! queue ! autovideosink \
dec. ! queue ! autoaudiosink- MP4
-
gst-launch-1.0 -v playbinuri=file:/absoulte/path/to/toto.mp4- audio and video from an MP4 file (queue is needed when playin audio and video)
-
gst-launch-1.0 filesrc location=sintel-1024-stereo.mp4 ! qtdemux name=demux \
demux. ! queue ! decodebin ! autovideosink \
demux. ! queue ! decodebin ! autoaudiosink
- rescale an anamorphic video
gst-launch-1.0 filesrc location=toto_720x576_anamorphic.mp4 ! qtdemux name=demux demux. ! queue ! decodebin ! videoscale ! video/x-raw,width=176,height=140,pixel-aspect-ratio=64/45 ! autovideosink
- only audio from an MP4 file
-
gst-launch-1.0 filesrc location=sintel-1024-stereo.mp4 ! qtdemux name=demux \
demux.audio_0 ! decodebin ! autoaudiosink
- TS
-
- program in a TS file (first program found?):
-
gst-launch-1.0 -v playbinuri=file:/tmp/toto.ts
- only video from TS file (program_number=802)
-
gst-launch-1.0 -v filesrc location=/disc/videos/tvc/tvc_794_20140821_1709.ts ! tsdemux program-number=802 ! mpegvideoparse ! decodebin ! autovideosink
- Problemes
mpegts_packetizer_pts_to_ts_internal: No groups, can't calculate timestamp
mpegts_packetizer_pts_to_ts_internal: Not enough information to calculate proper timestamp
mpegts_packetizer_offset_to_ts: Not enough observations to return a duration estimate- How to fix mpegts_packetizer_pts_to_ts: Not enough information to calculate proper timestamp in gstreeamer
- Solució / Solution:
- solved in gstreamer version 1.20.4
- OGG
-
- audio and video from an OGG file (queue is needed when playin audio and video)
-
gst-launch-1.0 filesrc location=sintel_trailer-720p.ogv ! oggdemux name=demux \
demux. ! queue ! decodebin ! autovideosink \
demux. ! queue ! decodebin ! autoaudiosink
- WebM / Matroska
-
gst-launch-1.0 -v playbinuri=file:/absoulte/path/to/toto.webm- playbinaudio and video from a webm file:
-
gst-launch-1.0 -v \
filesrc location=/path/to/toto.webm ! matroskademux name=demux \
demux.video_0 ! queue ! decodebin ! autovideosink sync=true \
demux.audio_0 ! queue ! decodebin ! autoaudiosink sync=truegst-launch-1.0 -v \
filesrc location=/path/to/toto.webm ! matroskademux name=demux \
demux. ! queue ! vp8dec ! autovideosink sync=true \
demux. ! queue ! opusdec ! autoaudiosink sync=true
- SDP
-
- See also: play from RTP
- sdpdemux
- UDP buffer
-
- value is taken from kernel parameter net.core.rmem_default
GST_DEBUG=3,udpsrc:4 gst-launchfilesrc location=toto.sdp ! sdpdemux name=demux ...-
udpsrc gstudpsrc.c:1428:gst_udpsrc_open:<udpsrc0> have udp buffer of 212992 bytes
- audio and video from SDP file (RTP):
-
gst-launch-1.0 filesrc location=toto.sdp do-timestamp=true ! sdpdemux latency=1000 debug=true name=bin \
bin. ! "application/x-rtp, media=(string)video" ! decodebin ! autovideosink sync=true \
bin. ! "application/x-rtp, media=(string)audio" ! decodebin ! autoaudiosink sync=true# RTP + RTCP, using sdpdemux
gst-launch-1.0 \
filesrc location=$sdp_path do-timestamp=true ! sdpdemux latency=${sdpdemux_latency_ms} name=bin \
bin. ! "application/x-rtp, media=(string)video" ! queue ! decodebin ! videoconvert ! videoscale ! videorate ! autovideosink sync=true \
bin. ! "application/x-rtp, media=(string)audio" ! queue ! decodebin ! audioconvert ! audioresample ! audiorate ! autoaudiosink sync=true- H.264 + AAC
-
gst-launch-1.0 filesrc location=toto.sdp do-timestamp=true ! sdpdemux name=demux \
demux. ! queue ! rtph264depay ! decodebin ! autovideosink sync=true \
demux. ! queue ! rtpmp4gdepay ! decodebin ! autoaudiosink sync=true
- only video from SDP file (RTP). Make sure that video is the first stream specified in sdp file:
-
gst-launch-1.0 filesrc location=toto.sdp ! sdpdemux name=demux \
demux.stream_0 ! queue ! decodebin ! autovideosink
- only audio from SDP file (RTP). Make sure that audio is the second stream specified in sdp file:
-
gst-launch-1.0 filesrc location=toto.sdp ! sdpdemux name=demux \
demux.stream_1 ! queue ! decodebin ! autoaudiosink
- Problemes / Problems
-
- Pèrdua de paquets / Packet loss
-
- Pèrdua de paquets amb ffmpeg / Packet loss with ffmpeg
- Frame drop when playing RTP stream using SDP file
- Solució / Solution
-
- Augmenteu la mida del buffer de recepció UDP modificant el paràmetre del kernel parameter net.core.rmem / Increase UDP reception buffer size by modifying kernel parameter net.core.rmem:
-
sudo sysctl -w net.core.rmem_default=851968- potser també haureu d'augmentar el màxim / you may need to also increase the maximum:
-
sudo sysctl -w net.core.rmem_max=851968
export GST_DEBUG=WARN,udpsrc:INFO; gst-launch-1.0 ...-
udpsrc gstudpsrc.c:1428:gst_udpsrc_open:<udpsrc3> have udp buffer of 851968 bytes
- Manca de fluïdesa / Lack of smoothness
-
WARN rtpjitterbuffer rtpjitterbuffer.c:573:calculate_skew: delta - skew: 0:00:02.543695912 too big, reset skew- Solució / Solution:
- ...
- Solució / Solution:
videodecoder gstvideodecoder.c:2775:gst_video_decoder_prepare_finish_frame:<avdec_h264-0> decreasing timestamp (0:00:00.008558259 < 0:00:00.058900688)-
- Solució / Solution
-
- Increase latency parameter (default: 200 ms) for sdpdemux:
gst-launch-1.0 filesrc location=toto.sdp ! sdpdemux latency=400 name=demux ...
audiobasesink gstaudiobasesink.c:1787:gst_audio_base_sink_get_alignment:<autoaudiosink0-actual-sink-alsa> Unexpected discontinuity in audio timestamps of +0:00:00.131360544, resyncing-
- Solució / Solution
-
- ...
- “delayed linking failed”
-
- Gstreamer “delayed linking failed” with rtspsrc video and audio
- Solució / Solution
-
... "application/x-rtp, media=(string)audio" ! decodebin ...... "application/x-rtp, media=(string)video" ! decodebin ...
- lipsync
- from network
-
- RTP
-
- See also: Play from SDP file
- Problemes / Problems
-
- udpsrc
-
videodecoder gstvideodecoder.c:2775:gst_video_decoder_prepare_finish_frame:<avdec_h264-0> decreasing timestamp (0:00:45.183879818 < 0:00:45.188331700)
gst-launch-1.0 udpsrc address=127.0.0.1 port=5004 ! "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-level-id=(string)64001f, payload=(int)96" ! queue ! \
rtph264depay ! decodebin ! autovideosinkgst-launch-1.0 -v \
udpsrc address=${address} port=${video_port} do-timestamp=true ! queue ! "application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)\"Z2QAHqzZQKAv+XARAAADAAEAAAMAPA8WLZY\=\,aOvssiw\=\", profile-level-id=(string)64001E" ! rtph264depay ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${audio_port} do-timestamp=true ! queue ! "application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, profile-level-id=(string)1, mode=(string)AAC-hbr, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)121056E500" ! rtpmp4gdepay ! decodebin ! autoaudiosink sync=true-
- caps can be obtained e.g. by
executing:
gst-launch -v ... sdpdemux ...
- caps can be obtained e.g. by
executing:
- common variables:
-
# ffmpeg -re -i easylife.mp4 -c:v copy -an -f rtp -cname toto rtp://234.1.2.3:5004 -vn -c:audio copy -f rtp -cname toto rtp://234.1.2.3:5006 -sdp_file /mnt/nfs/sdp/toto.sdp
address=234.1.2.3
video_rtp_port=5004
video_rtcp_port=$(( video_rtp_port + 1 ))
VIDEOCAPS="application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)\"Z2QAHqzZQKAv+XARAAADAAEAAAMAPA8WLZY\=\,aOvssiw\=\", profile-level-id=(string)64001E"
audio_rtp_port=$(( video_rtp_port + 2 ))
audio_rtcp_port=$(( video_rtp_port + 3 ))
AUDIOCAPS="application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, profile-level-id=(string)1, mode=(string)AAC-hbr, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)121056E500"
- RTP
-
# RTP
gst-launch-1.0 \
udpsrc address=${address} port=${video_rtp_port} do-timestamp=true ! "$VIDEOCAPS" ! \
rtph264depay ! queue ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! \
rtpmp4gdepay ! queue ! decodebin ! autoaudiosink sync=true
- RTP (using rtpbin)
-
# RTP, using rtpbin
gst-launch-1.0 \
rtpbin name=bin \
udpsrc address=${address} port=${video_rtp_port} do-timestamp=true ! "$VIDEOCAPS" ! bin.recv_rtp_sink_0 \
bin. ! rtph264depay ! queue ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! bin.recv_rtp_sink_1 \
bin. ! rtpmp4gdepay ! queue ! decodebin ! autoaudiosink sync=true
- RTP with RTCP (using rtpbin)
-
# RTP + RTCP, using rtpbin
gst-launch-1.0 \
rtpbin name=bin \
udpsrc address=${address} port=${video_rtp_port} do-timestamp=true ! "$VIDEOCAPS" ! bin.recv_rtp_sink_0 \
bin. ! "application/x-rtp, media=(string)video" ! queue ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${video_rtcp_port} ! "application/x-rtcp" ! bin.recv_rtcp_sink_0 \
udpsrc address=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! bin.recv_rtp_sink_1 \
bin. ! "application/x-rtp, media=(string)audio" ! queue ! decodebin ! autoaudiosink sync=true \
udpsrc address=${address} port=${audio_rtcp_port} ! "application/x-rtcp" ! bin.recv_rtcp_sink_1
# RTP + RTCP, using rtpbin
gst-launch-1.0 \
rtpbin name=bin \
udpsrc address=${address} port=${video_rtp_port} do-timestamp=true ! "$VIDEOCAPS" ! bin.recv_rtp_sink_0 \
bin. ! rtph264depay ! queue ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${video_rtcp_port} ! "application/x-rtcp" ! bin.recv_rtcp_sink_0 \
udpsrc address=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! bin.recv_rtp_sink_1 \
bin. ! rtpmp4gdepay ! queue ! decodebin ! autoaudiosink sync=true \
udpsrc address=${address} port=${audio_rtcp_port} ! "application/x-rtcp" ! bin.recv_rtcp_sink_1
- Problemes / Problems
-
- lipsync
-
- source: specify do-timestamp=true
- sinks: specify sync=true
- sdpdemux: latency must be big enough. If stream has B images, try with latency=1000
- check with tshark that RTP flow
contains RTCP packets with
"Source Description" (SDES)
information
- ffmpeg
does not generate "Source
Description" in RTCP, by
default. Force it by
adding option -cname
with the same value in both
audio and video streams:
ffmeg -an ... -f rtp -cname user123456789@host-87654321 rtp://... ... -vn ... -f rtp -cname user123456789@host-87654321 rtp://...
- ffmpeg
does not generate "Source
Description" in RTCP, by
default. Force it by
adding option -cname
with the same value in both
audio and video streams:
- HTTP
-
- check that souphttpsrc is present
-
gst-inspect-1.0 | grep souphttpsrc
- if not present, compile it
-
- Dependencies
-
- CentOS
-
sudo yum install libsoup-devel
- Mageia
-
urpmi libsoup-devel
- gst-plugins-good
-
./configuremakesudo make install
gst-launch-1.0playbinuri=http://download.blender.org/peach/bigbuckbunny_movies/BigBuckBunny_320x180.mp4
- RTMP
-
gst-launch-1.0 -vplaybinuri=rtmp://nginx-server/myapp/mystreamgst-launch-1.0 -vplaybinuri=${source}- gst-launch-1.0 uridecodebin uri=${source} name=decoder playsink name=sink decoder.src_0 ! sink.video_sink decoder.src_1 ! sink.audio_sink
gst-launch-1.0 rtmpsrc location=${source} ! queue2 use-buffering=true ! decodebin name=decoder playsink name=sink decoder.src_0 ! sink.video_sink decoder.src_1 ! sink.audio_sink
gst-launch-1.0 rtmpsrc location=${source}do-timestamp=true! queue2use-buffering=true! decodebin name=decoder playsink name=sink decoder.src_0 ! sink.video_sink decoder.src_1 ! sink.audio_sink
gst-launch-1.0 rtmpsrc location=${source}do-timestamp=true! queue2use-buffering=true
gst-launch-1.0 -v \
rtmpsrc location=${source} do-timestamp=true ! queue2 ! decodebin name=mydecoder \
mydecoder. ! autovideosink sync=true \
mydecoder. ! autoaudiosink sync=true(not working?) source=rtmp://nginx-server/myapp/mystream
gst-launch-1.0 \
rtmpsrc location=${source} do-timestamp=true ! flvdemux name=demux \
demux.video ! queue ! decodebin ! autovideosink sync=true \
demux.audio ! queue ! decodebin ! autoaudiosink sync=true
- general: using
- Mux to
-
- test to TS
-
gst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=24/1,width=1280,height=720 ! videoconvert ! x264enc ! video/x-h264,profile=high ! mpegtsmux ! filesink location=toto.ts
- test to MP4
-
- NOTE:
stream-format=(string)byte-streamis not supported by MP4 gst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=24/1,width=1280,height=720 ! videoconvert ! x264enc ! video/x-h264,profile=high ! mp4mux ! filesink location=toto.mp4-
- Problem:
-
- moov atom not found
-
- Solution
-
- (?) mp4mux faststart=true
- NOTE:
- Captura / Capture
- capture a single frame to a png file
gst-launch-1.0 -v videotestsrcnum-buffers=1! pngenc ! filesink location=toto.png
- capture timestamped frames (BeagleCam)
-
gst-launch v4l2src num-buffers=1 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! jpegenc ! filesink location=$(date +"%s").jpg
- capture a single frame to a png file
- Transmux
-
- rtmp
-> rtmp
gst-launch-1.0 rtmpsrc location=rtmp://server.org/my_app/first do-timestamp=true ! queue2 ! flvdemux name=demux\
flvmux name=mux\
demux.video ! queue ! mux.video\
demux.audio ! queue ! mux.audio\
mux.src ! queue ! rtmpsink location=rtmp://server.org/my_app/second- Note: if using nginx-rtmp-module as destination, check nginx.conf configuration
- only video (mp4 -> ts):
-
gst-launch-1.0 filesrc location=sintel-1024-stereo.mp4 ! qtdemux name=demux \
mpegtsmux name=mux ! filesink location=toto.ts \
demux. ! queue ! h264parse ! mux.gst-launch-1.0 filesrc location=sintel-1024-stereo.mp4 ! qtdemux name=demux \
mpegtsmux name=mux ! filesink location=toto.ts \
demux. ! video/x-h264 ! queue ! h264parse ! mux.
- video and audio (mp4 -> ts)
-
gst-launch-1.0 filesrc location=sintel-1024-stereo.mp4 ! qtdemux name=demux \
mpegtsmux name=mux ! filesink location=toto2015.ts \
demux. ! queue ! h264parse ! mux. \
demux. ! queue ! aacparse ! mux.
- video and audio (mp4 -> flv)
-
gst-launch-1.0 -v filesrc location=sintel-1024-stereo.mp4 ! qtdemux name=demux \
flvmux streamable=true name=mux ! filesink location=toto.flv \
demux. ! queue ! h264parse ! mux. \
demux. ! queue ! aacparse ! mux.
- video and audio (flv ->mp4)
-
- ...
- video (H.264) (sdp -> ts)
-
gst-launch-1.0 -v filesrc location=toto.sdp ! sdpdemux name=demux \
mpegtsmux name=mux ! filesink location=toto.ts \
demux. ! queue ! rtph264depay ! mux.
- video (H.264) and audio (AAC) (sdp -> ts)
-
gst-launch-1.0 -v filesrc location=toto.sdp ! sdpdemux name=demux \
mpegtsmux name=mux ! filesink location=toto.ts \
demux. ! queue ! rtph264depay ! mux. \
demux. ! queue ! rtpmp4gdepay ! mux.
- video (sdp->mp4) (not working)
-
gst-launch-1.0 filesrc location=/tmp/bbb.sdp ! sdpdemux name=demux \
mp4mux name=mux ! filesink location=/tmp/toto.mp4 \
demux. ! rtph264depay ! h264parse ! mux.-
- problem
-
ffplay toto.mp4
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7f128c0008c0] moov atom not found
toto.mp4: Invalid data found when processing input- Solution?
-
- mp4mux faststart=true
- video and audio (sdp -> mp4) (not working)
-
gst-launch-1.0 -v filesrc location=toto.sdp ! sdpdemux name=demux \
mp4mux name=mux ! filesink location=toto.mp4 \
demux. ! queue ! h264parse ! mux. \
demux. ! queue ! aacparse ! mux.
- rtmp
-> rtmp
- Transcode
-
- only video, to file:
-
gst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=805 name="demux" \
demux. ! queue ! mpegvideoparse ! decodebin ! videoconvert ! x264enc ! video/x-h264,stream-format=byte-stream,profile=high ! h264parse ! \
mpegtsmux ! filesink location=/tmp/toto.ts
- and resize, stream:
-
gst-launch-1.0 -v filesrc location=tvc_20150604.ts ! tsdemux program-number=806 ! \
mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc ! h264parse ! \
mpegtsmux ! udpsink host=192.168.0.8 port=5004 sync=falsegst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 ! \
mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc ! h264parse ! \
mpegtsmux ! udpsink host=192.168.0.8 port=5004 sync=falsegst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=805 name="demux" \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! 'video/x-raw, width=360, height=288' ! videoconvert ! x264enc ! video/x-h264,stream-format=byte-stream,profile=main ! h264parse ! \
mpegtsmux ! udpsink host=192.168.0.8 port=5004 sync=false
- tee
- two windows from MP4 file:
gst-launch-1.0 \
filesrc location=${video_path} ! qtdemux name=demux \
demux. ! queue ! decodebin ! tee name=tv \
tv. ! queue ! autovideosink sync=true \
tv. ! queue ! autovideosink sync=true \
demux. ! queue ! decodebin ! tee name=ta \
ta. ! queue ! autoaudiosink sync=true \
ta. ! queue ! autoaudiosink sync=true
- two windows from SDP file:
gst-launch-1.0 \
filesrc location=${video_path} do-timestamp=true ! sdpdemux name=bin \
bin. ! "application/x-rtp, media=(string)audio" ! queue ! decodebin ! tee name=ta \
ta. ! queue ! autoaudiosink sync=true \
ta. ! queue ! autoaudiosink sync=true \
bin. ! "application/x-rtp, media=(string)video" ! queue ! decodebin ! tee name=tv \
tv. ! queue ! autovideosink sync=true \
tv. ! queue ! autovideosink sync=true
- two windows from MP4 file:
- Stream
-
- Introduction to network streaming using GStreamer
- TS over UDP
-
- UDP unicast stream only audio:
-
gst-launch-1.0 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
demux. ! queue ! mpegaudioparse ! decodebin ! audioconvert ! lamemp3enc ! \
mpegtsmux ! udpsink host=192.168.0.8 port=5004 sync=falsegst-launch-1.0 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
demux. ! queue ! mpegaudioparse ! decodebin ! audioconvert ! lamemp3enc ! mux. \
mpegtsmux name=mux ! udpsink host=192.168.0.8 port=5004 sync=falsegst-launch-1.0 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
mpegtsmux name=mux ! udpsink host=192.168.0.8 port=5004 sync=false \
demux. ! queue ! mpegaudioparse ! decodebin ! audioconvert ! lamemp3enc ! mux.
- Encode to H.264, mux to TS, UDP stream:
-
gst-launch-1.0 -e videotestsrc ! video/x-raw, framerate=25/1, width=640, height=360 ! x264enc ! \
mpegtsmux ! udpsink host=192.168.0.8 port=5004 sync=falsegst-launch-1.0 -v -e videotestsrc ! video/x-raw, framerate=25/1, width=640, height=360 ! x264enc bitrate=512 ! video/x-h264,profile=high ! h264parse ! \
mpegtsmux ! udpsink host=192.168.0.8 port=5004 sync=falsegst-launch-1.0 -e mpegtsmux name="muxer" ! udpsink host=192.168.0.8 port=5004 sync=false \
videotestsrc ! video/x-raw, framerate=25/1, width=640, height=360 ! x264enc bitrate=512 ! video/x-h264,profile=high ! h264parse ! muxer.
- Mux video and audio, UDP stream:
-
gst-launch-1.0 -e mpegtsmux name="muxer" ! udpsink host=192.168.0.8 port=5004 sync=false \
videotestsrc ! video/x-raw, framerate=25/1, width=640, height=360 ! x264enc bitrate=512 ! video/x-h264,profile=high ! h264parse ! muxer. \
audiotestsrc wave=5 ! audioconvert ! lamemp3enc ! muxer.gst-launch-1.0 dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
mpegtsmux name=mux ! udpsink host=192.168.0.8 port=5004 sync=false \
demux. ! queue ! mpegaudioparse ! decodebin ! audioconvert ! lamemp3enc ! mux. \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=50 ! h264parse ! mux.- Notes:
-
- default for omxh264enc is inline-header=true
- specification of periodicty-idr is needed by vlc to be played
- these parameters are available on latest version of gstreamer. Maybe you need to compile version 1.2.
- to RTP
-
- RTP and RTSP support
- gstreamer/gst-plugins-good/gst/rtp/README
- Streaming H.264 via RTP
- Play from SDP file
- SDP generation
-
- RTP
components
- GStreamer SDP
library
- Structure
GstSdp.SDPMessage
- gst-plugins-base / gst-libs / gst / sdp
- webrtcbidirectional.c
- gstwebrtcbin.h
-
media
caps
SDP ret, media = GstSdp.SDPMedia.new()
caps = pad.get_current_caps()
(all sets)
ret = GstSdp.SDPMedia.set_media_from_caps(caps, media)
media.as_text()
- media.set_port_info(5004, 1)
m=<caps.media> <media.port> <media.proto> <caps.payload>
m=video 5004 RTP/AVP 96
- media.set_information("my info\n")
i=<media.information>
i=my info
- media.add_connection("IN", "IP4", "1.2.3.4", 16, 1)
c=<media.connection.nettype> <media.connection.addrtype> <media.connection.address>
c=IN IP4 1.2.3.4
a=rtpmap:<caps.payload> <caps.encoding-name>/<caps.clock-rate>
a=rtpmap:96 H264/90000
(caps) a=fmtp:<caps.payload> ...
a=fmtp:96 ...
- mp4rtp.py
class Streamer(object):
def build_sdp(self, rtpbin, address, port, sdp_path, session_name=None):
"""
rtpbin: Gst.RtpBin element
address: destination address
port: initial destination port
sdp_path: created file with SDP
session_name: (optional) session name (s=...)
"""
ret, sdp_message = GstSdp.SDPMessage.new()
sdp_message.set_version('0')
ttl = 64
number_addresses = 1
sdp_message.set_connection("IN", "IP4", address, ttl, number_addresses)
if session_name:
sdp_message.set_session_name(session_name)
pads = rtpbin.iterate_pads()
while True:
ret, pad = pads.next()
if ret==Gst.IteratorResult.OK:
# only source pads
if pad.direction != Gst.PadDirection.SRC:
continue
# only pads with name send_rtp_src...
pad_name = pad.get_name()
if not pad_name.startswith('send_rtp_src'):
continue
print("pad: {0:s}".format(pad.name))
caps = pad.get_current_caps()
print(" {0:s}".format(caps.to_string()))
ret, media = GstSdp.SDPMedia.new()
if ret != GstSdp.SDPResult.OK:
print("Error")
return
media.set_port_info(port, 1)
port = port + 2
media.proto = "RTP/AVP"
ret = GstSdp.SDPMedia.set_media_from_caps(caps, media)
if ret != GstSdp.SDPResult.OK:
print("Error")
sdp_message.add_media(media)
elif ret==Gst.IteratorResult.DONE:
break
elif ret==Gst.IteratorResult.ERROR:
break
print(sdp_message.as_text())
f = open(sdp_path, 'w')
f.write(sdp_message.as_text())
def on_message(self, bus, msg, user_data):
t = msg.type
...
elif t == Gst.MessageType.STATE_CHANGED:
old, new, pending = msg.parse_state_changed()
if new == Gst.State.PAUSED:
if msg.src.name=="bin":
print("RtpBin")
rtpbin = msg.src
self.build_sdp(rtpbin, self.dst_address, self.initial_port, "output.sdp", None)
...
- Structure
- GStreamer SDP
library
- webrtcbin
- ...
- Bug
795404 - webrtcbin: h.264 streams work
with Chrome but not with Firefox
- "Inserting an fmtp attribute with
a profile-level-id value seems to
fix it."
- webrtc-test.cpp
- webrtc-unidirectional-h264.c
- "Inserting an fmtp attribute with
a profile-level-id value seems to
fix it."
- Generating a SDP file from a streaming pipeline
- caps to SDP (README)
- RTP
components
-
gstreamer
sdp file
command
gst-launch-1.0 -v videotestsrc ! videoconvert ! x264enc ! rtph264pay config-interval=10 pt=96 ! udpsink host=234.1.2.3 port=5004
v=0
m=<media> <port> RTP/AVP <payload>
c=IN IP4 <host>
a=rtpmap:<payload> <encoding-name>/<clock-rate>a=fmtp:96 packetization-mode=<packetization-mode>; sprop-parameter-sets=<sprop-parameter-sets>; profile-level-id=<profile-level-id>
caps
(given by -v)
application/x-rtp,
media=(string)video,
clock-rate=(int)90000,
encoding-name=(string)H264,
packetization-mode=(string)1,
profile-level-id=(string)f4000d,
sprop-parameter-sets=(string)"Z/QADZGbKCg/YC1BgEFQAAADABAAAAMDyPFCmWA\=\,aOvsRIRA",
payload=(int)96,
ssrc=(uint)3934427744,
timestamp-offset=(uint)2187273080,
seqnum-offset=(uint)1602,
a-framerate=(string)30
media=(string)audio,
...
-
send
receive
from file
common code
#!/bin/bashinput_path=$1
sdp_path=/tmp/toto.sdp
dst_address=234.1.2.3
video_rtp_port=5004
video_rtcp_port=$(( video_rtp_port + 1 ))
video_media_subtype="H264"
rtp_video_payload_type=96
audio_rtp_port=$(( video_rtp_port + 2 ))
audio_rtcp_port=$(( video_rtp_port + 3 ))
audio_media_subtype="aac"
rtp_audio_payload_type=$(( rtp_video_payload_type + 1 ))
audio_rate=48000
audio_channels=2#!/bin/bashsdp_path=$1
address=234.1.2.3
video_rtp_port=5004
video_rtcp_port=$(( video_rtp_port + 1 ))
rtp_video_payload_type=96
VIDEOCAPS="application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, sprop-parameter-sets=(string)\"Z2QAHqzZQKAv+XARAAADAAEAAAMAPA8WLZY\=\,aOvssiw\=\", profile-level-id=(string)64001E"
audio_rtp_port=$(( video_rtp_port + 2 ))
audio_rtcp_port=$(( video_rtp_port + 3 ))
rtp_audio_payload_type=$(( rtp_video_payload_type + 1 ))
AUDIOCAPS="application/x-rtp, media=(string)audio, payload=(int)97, clock-rate=(int)44100, encoding-name=(string)MPEG4-GENERIC, encoding-params=(string)2, profile-level-id=(string)1, mode=(string)AAC-hbr, sizelength=(string)13, indexlength=(string)3, indexdeltalength=(string)3, config=(string)121056E500"SDP
function create_sdp {
local sdp_path=$1
# sdp
cat >$sdp_path <<EOF
v=0
c=IN IP4 ${dst_address}
m=video ${video_rtp_port} RTP/AVP ${rtp_video_payload_type}
a=rtpmap:${rtp_video_payload_type} ${video_media_subtype}/90000
m=audio ${audio_rtp_port} RTP/AVP ${rtp_audio_payload_type}
a=rtpmap:${rtp_audio_payload_type} ${audio_media_subtype}/${audio_rate}/${audio_channels}
EOF
if (( channels == 2 )) && [[ ${audio_media_subtype} == "opus" ]]
then
echo "a=fmtp:${rtp_audio_payload_type} sprop-stereo=1" >>${sdp_path}
fi
}
RTP
gst-launch-1.0 \
filesrc location=${input_path} ! qtdemux name=demux \
demux.video_0 ! queue ! rtph264pay pt=$rtp_video_payload_type ! \
udpsink host=${dst_address} port=${video_rtp_port} sync=true \
demux.audio_0 ! queue ! rtpmp4gpay pt=$rtp_audio_payload_type ! \
udpsink host=${dst_address} port=${audio_rtp_port} sync=truegst-launch-1.0 \
udpsrc address=${address} port=${video_rtp_port} do-timestamp=true ! "$VIDEOCAPS" ! \
"application/x-rtp, media=(string)video"! queue ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! \
"application/x-rtp, media=(string)audio"! queue ! decodebin ! autoaudiosink sync=trueRTP using rtpbin
gst-launch-1.0 \
rtpbin name=bin \
filesrc location=${input_path} ! qtdemux name=demux \
demux.video_0 ! queue ! rtph264pay pt=$rtp_video_payload_type ! bin.send_rtp_sink_0 \
bin.send_rtp_src_0 ! udpsink host=${dst_address} port=${video_rtp_port} sync=true \
demux.audio_0 ! queue ! rtpmp4gpay pt=$rtp_audio_payload_type ! bin.send_rtp_sink_1\
bin.send_rtp_src_1 ! udpsink host=${dst_address} port=${audio_rtp_port} sync=truegst-launch-1.0 \
rtpbin name=bin \
udpsrc address=${address} port=${video_rtp_port} do-timestamp=true ! "$VIDEOCAPS" ! bin.recv_rtp_sink_0 \
bin. !"application/x-rtp, media=(string)video"! queue ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! bin.recv_rtp_sink_1 \
bin. !"application/x-rtp, media=(string)audio"! queue ! decodebin ! autoaudiosink sync=trueRTP+RTCP using rtpbin
gst-launch-1.0 \
rtpbin name=bin \
filesrc location=${input_path} ! qtdemux name=demux \
demux.video_0 ! queue ! rtph264pay pt=$rtp_video_payload_type ! bin.send_rtp_sink_0 \
bin.send_rtp_src_0 ! udpsink host=${dst_address} port=${video_rtp_port} sync=true \
bin.send_rtcp_src_0 ! udpsink host=${dst_address} port=${video_rtcp_port} sync=false async=false \
demux.audio_0 ! queue ! rtpmp4gpay pt=$rtp_audio_payload_type ! bin.send_rtp_sink_1\
bin.send_rtp_src_1 ! udpsink host=${dst_address} port=${audio_rtp_port} sync=true \
bin.send_rtcp_src_1 ! udpsink host=${dst_address} port=${audio_rtcp_port} sync=false async=false \gst-launch-1.0 \
rtpbin name=bin \
udpsrc address=${address} port=${video_rtp_port} do-timestamp=true ! "$VIDEOCAPS" ! bin.recv_rtp_sink_0 \
bin. !"application/x-rtp, media=(string)video"! queue ! decodebin ! autovideosink sync=true \
udpsrc address=${address} port=${video_rtcp_port} ! "application/x-rtcp" ! bin.recv_rtcp_sink_0 \
udpsrc address=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! bin.recv_rtp_sink_1 \
bin. !"application/x-rtp, media=(string)audio"! queue ! decodebin ! autoaudiosink sync=true \
udpsrc address=${address} port=${audio_rtcp_port} ! "application/x-rtcp" ! bin.recv_rtcp_sink_1RTP+RTCP using sdpdemux
gst-launch-1.0 -v \
filesrc location=$sdp_path do-timestamp=true ! sdpdemux latency=${sdpdemux_latency_ms} name=bin \
bin. ! "application/x-rtp, media=(string)video" ! queue ! decodebin ! videoconvert ! videoscale ! queue ! videorate ! autovideosink sync=true \
bin. ! "application/x-rtp, media=(string)audio" ! queue ! decodebin ! audioconvert ! audioresample ! queue ! audiorate ! autoaudiosink sync=true
- encode webcam, UDP stream:
-
gst-launch v4l2src ! video/x-raw-yuv,width=128,height=96,format='(fourcc)'UYVY ! ffmpegcolorspace ! ffenc_h263 ! video/x-h263 ! rtph263ppay pt=96 ! udpsink host=192.168.1.1 port=5000 sync=false
- test VP8 / Opus to RTP (no RTPC) (WebRTC and Janus)
-
gst-launch-1.0 \
audiotestsrc is-live=true wave=5 ! audioresample ! audioconvert ! audio/x-raw,channels=2,rate=16000 ! opusenc bitrate=20000 ! rtpopuspay pt=97 ! udpsink host=127.0.0.1 port=5002 \
videotestsrc ! video/x-raw,width=320,height=240,framerate=15/1 ! videoscale ! videorate ! videoconvert ! timeoverlay ! vp8enc ! rtpvp8pay pt=96 ! udpsink host=127.0.0.1 port=5004- sdp
-
v=0
c=IN IP4 127.0.0.1
m=video 5100 RTP/AVP 96
a=rtpmap:96 VP8/90000
m=audio 5102 RTP/AVP 97
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1
- test VP8 / Opus to RTP (with RTCP, using rtpbin):
-
sdp_path=/tmp/toto.sdp
dst_address=225.4.3.2
video_rtp_port=5100
video_rtcp_port=$(( video_rtp_port + 1 ))
video_media_subtype="VP8"
rtp_video_payload_type=96
audio_rtp_port=$(( video_rtp_port + 2 ))
audio_rtcp_port=$(( video_rtp_port + 3 ))
audio_media_subtype="opus"
rtp_audio_payload_type=$(( rtp_video_payload_type + 1 ))
rate=48000
channels=2
# sdp
cat >$sdp_path <<EOF
v=0
c=IN IP4 $dst_address
m=video $video_rtp_port RTP/AVP $rtp_video_payload_type
a=rtpmap:$rtp_video_payload_type ${video_media_subtype}/90000
m=audio $audio_rtp_port RTP/AVP $rtp_audio_payload_type
a=rtpmap:$rtp_audio_payload_type ${audio_media_subtype}/${rate}/${channels}
EOF
if (( channels == 2 )) && [[ ${audio_media_subtype} == "opus" ]]
then
echo "a=fmtp:${rtp_audio_payload_type} sprop-stereo=1" >>${sdp_path}
fi
gst-launch-1.0 -v \
rtpbin name=bin \
videotestsrc ! video/x-raw,width=320,height=240,framerate=25/1 ! videoscale ! videorate ! videoconvert ! timeoverlay ! vp8enc ! rtpvp8pay pt=$rtp_video_payload_type ! bin.send_rtp_sink_0 \
bin.send_rtp_src_0 ! udpsink host=${dst_address} port=${video_rtp_port} sync=true \
bin.send_rtcp_src_0 ! udpsink host=${dst_address} port=${video_rtcp_port} sync=false async=false \
audiotestsrc is-live=true wave=5 ! audioconvert ! audioresample ! audio/x-raw,channels=${channels},rate=${rate} ! opusenc bitrate=64000 ! rtpopuspay pt=$rtp_audio_payload_type ! bin.send_rtp_sink_1 \
bin.send_rtp_src_1 ! udpsink host=${dst_address} port=${audio_rtp_port} sync=true \
bin.send_rtcp_src_1 ! udpsink host=${dst_address} port=${audio_rtcp_port} sync=false async=false
- test H.264 to RTP (no RTCP)
-
gst-launch-1.0 -v videotestsrc ! videoconvert ! x264enc ! rtph264pay config-interval=10 pt=96 ! udpsink host=234.1.2.3 port=5004- player:
-
- toto.sdp
-
v=0
m=video 5004 RTP/AVP 96
c=IN IP4 234.1.2.3
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1
ffplay -i toto.sdp
- from file (VP8, Opus) to RTP (no RTCP)
-
sdp_path=/tmp/toto.sdpdst_address=225.4.3.2
video_rtp_port=5100
video_rtcp_port=$(( video_rtp_port + 1 ))
video_media_subtype="VP8"
rtp_video_payload_type=96
audio_rtp_port=$(( video_rtp_port + 2 ))
audio_rtcp_port=$(( video_rtp_port + 3 ))
audio_media_subtype="opus"
rtp_audio_payload_type=$(( rtp_video_payload_type + 1 ))
rate=48000
channels=2
# sdp
cat >$sdp_path <<EOF
v=0
c=IN IP4 $dst_address
m=video $video_rtp_port RTP/AVP $rtp_video_payload_type
a=rtpmap:$rtp_video_payload_type ${video_media_subtype}/90000
m=audio $audio_rtp_port RTP/AVP $rtp_audio_payload_type
a=rtpmap:$rtp_audio_payload_type ${audio_media_subtype}/${rate}/${channels}
EOF
if (( channels == 2 )) && [[ ${audio_media_subtype} == "opus" ]]
then
echo "a=fmtp:${rtp_audio_payload_type} sprop-stereo=1" >>${sdp_path}
fi
gst-launch-1.0 -v \
filesrc location=/path/to/toto.webm ! matroskademux name=demux \
demux.video_0 ! queue ! rtpvp8pay pt=$rtp_video_payload_type ! udpsink host=${dst_address} port=${video_rtp_port} sync=true \
demux.audio_0 ! queue ! rtpopuspay pt=$rtp_audio_payload_type ! udpsink host=${dst_address} port=${audio_rtp_port} sync=true
- from file (H.264, AAC) to RTP
-
gst-launch \
filesrc location=${input_path} ! qtdemux name=demux \
demux.video_0 ! queue ! rtph264pay pt=$rtp_video_payload_type ! udpsink host=${dst_address} port=${video_rtp_port} sync=true \
demux.audio_0 ! queue ! rtpmp4gpay pt=$rtp_audio_payload_type ! udpsink host=${dst_address} port=${audio_rtp_port} sync=true
- from file (H.264, AAC) to RTP, using rtpbin
-
- ...
- from file (H.264, AAC) to RTP + RTCP, using rtpbin
-
- ...
- RTMP to:
-
- rtmpsink uses librtmp
- options from librtmp
- problems with rtmps and TLS v1.3 (e.g. Facebook)
- rtmp2sink (rtmp2, Alternative RTMP Implementation) (GStreamer>=1.18) does not use librtmp
- nginx-rtmp
-
- Video and audio:
-
- from test to H.264, AAC
-
gst-launch-1.0 -v flvmux name=mux ! rtmpsink location=rtmp://nginx-server/myapp/mystream \
videotestsrc ! video/x-raw, width=360, height=288 ! x264enc ! video/x-h264,profile=baseline,width=360,height=288 ! h264parse ! mux. \
audiotestsrc wave=5 ! audioconvert ! avenc_aac compliance=experimental ! aacparse ! mux.
- from test to H.264 (omx), MP3
-
gst-launch-1.0 -v flvmux name=mux ! rtmpsink location=rtmp://nginx-server/myapp/mystream \
videotestsrc ! video/x-raw, width=360, height=288 ! omxh264enc ! video/x-h264,profile=baseline,width=360,height=288 ! h264parse ! mux. \
audiotestsrc wave=5 ! audioconvert ! lamemp3enc ! mpegaudioparse ! mux.
- from DVB
-
gst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
flvmux name=mux ! rtmpsink location=rtmp://nginx-server/myapp/mystream \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=50 ! h264parse ! mux. \
demux. ! queue ! mpegaudioparse ! decodebin ! audioconvert ! avenc_aac compliance=experimental ! aacparse ! mux.
- video, audio with PID 0x7c:
-
gst-launch-1.0 -vvv dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
flvmux name=mux ! rtmpsink location=rtmp://192.168.0.8/myapp/mystream \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=50 ! h264parse ! mux. \
demux.audio_007c ! queue ! mpegaudioparse ! decodebin ! audioconvert ! avenc_aac compliance=experimental ! aacparse ! mux.
- video, test audio:
-
gst-launch-1.0 -vvv dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
flvmux name=mux ! rtmpsink location=rtmp://nginx-server/myapp/mystream \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=50 ! h264parse ! mux. \
audiotestsrc wave=5 ! audioconvert ! lamemp3enc ! mpegaudioparse ! mux.
- video, audio forced to 44100Hz (MP3 at 48000Hz is not supported by FLV) (AAC at 48000Hz is supported, though) (queue max-size-time must be increase from 1000000ns [1s] to ... )
-
gst-launch-1.0 -vvv dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
flvmux name=mux ! rtmpsink location=rtmp://nginx-server/myapp/mystream \
demux. ! queue max-size-time=4000000000 ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=50 ! h264parse ! mux. \
demux.audio_007c ! queue max-size-time=4000000000 ! mpegaudioparse ! decodebin ! audioconvert ! audioresample ! audio/x-raw,rate=44100 ! lamemp3enc ! mpegaudioparse ! mux.
- video, audio AAC at 48000Hz:
-
gst-launch-1.0 -vvv dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=806 name=demux \
flvmux name=mux ! rtmpsink location=rtmp://nginx-server/myapp/mystream \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=32, height=32 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=50 ! h264parse ! mux. \
demux.audio_007c ! queue ! mpegaudioparse ! decodebin ! audioconvert ! audioresample ! audio/x-raw,rate=48000 ! avenc_aac compliance=experimental ! aacparse ! mux.
- Problemes / Problems
-
- queues
-
- [gst-devel] DV capture pipeline frozen
- Solution:
-
export GST_DEBUG="*queue*:5"- disable maximum limits in queue:
-
queue max-size-bytes=0 max-size-buffers=0 max-size-time=0
- when sending to nginx-rtmp-module,
with nginx
logs at level info (
error_log /var/log/nginx/error.log info):RTMP in chunk stream too big: xx >= 32too big message ...- too big message: 1048576 #324
- Solució / Solution
- nginx.conf
rtmp {
server {
# compatibility with GStreamer:
publish_time_fix on;
...
}
}
- nginx.conf
chunk stream too big-
- I got "RTMP in chunk stream too big: 64 >= 32" error when send rtmp stream to nginx #246
- Broadcasting a video playlist to nginx-rtmp-module
Solution: nginx.conf-
rtmp {
server {
# compatibility with GStreamer and Android from Samsung S5 using ...:
publish_time_fix off;
# only for publish-time-fix2 branch:
#publish_time_fix2 on;
...
}
}/usr/local/nginx/sbin/nginx -s reload
-
-
- RTMPS (e.g. Facebook)
- ...
- Problems
- Streaming to rtmps fails with "TLS_Connect failed"
- librtmp with TLS v1.3
- Solució / Solution
- use rtmp2sink (GStreamer >=1.18)
- Wowza
-
- Live Streaming from RaspberryPi using GStreamer - Help please?
-
- Incoming security / Flash Version String:
-
Wirecast/|FME/|FMLE/|Wowza GoCoder*|Gstreamer/|Gstreamer/*|Gstreamer*
- How to secure publishing from an RTMP encoder that does not support authentication (ModuleSecureURLParams)
- Streaming to a Flash Media Server using the rtmpsink element
-
- profile=baseline
- woking / not working
-
- working
-
- omxh264enc ! video/x-h264,profile=high
- omxh264enc ! video/x-h264,profile=baseline
- x264enc ! video/x-h264,profile=baseline
- not working
-
- x264enc ! video/x-h264,profile=high
- only video:
-
gst-launch-1.0 -v -e flvmux name=mux ! rtmpsink location=rtmp://wowza_server/application/stream \
videotestsrc ! video/x-raw, framerate=25/1, width=640, height=360 ! x264enc bitrate=512 ! video/x-h264,profile=baseline ! h264parse ! mux.gst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=805 name=demux \
flvmux name=mux ! rtmpsink location=rtmp://wowza_server/application/stream \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=360, height=288 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=1 ! video/x-h264,profile=baseline ! h264parse ! mux.gst-launch-1.0 -v dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=658000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! tsdemux program-number=805 name=demux \
flvmux name=mux ! rtmpsink location=rtmp://wowza-server/application/stream \
demux. ! queue ! mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=360, height=288 ! videoconvert ! omxh264enc inline-header=true periodicty-idr=1 ! video/x-h264,profile=high ! h264parse ! mux.gst-launch-1.0 -v flvmux name=mux ! rtmpsink location=rtmp://wowza-server/application/stream \
videotestsrc ! video/x-raw, width=360, height=288 ! omxh264enc ! video/x-h264,profile=high ! h264parse ! mux.
- video and audio:
-
gst-launch-1.0 -v -e flvmux name=mux ! rtmpsink location=rtmp://wowza-server/application/stream videotestsrc ! video/x-raw, framerate=24/1, width=1024, height=436 ! x264enc bitrate=800 ! video/x-h264,profile=baseline ! h264parse ! mux. audiotestsrc wave=5 ! audioconvert ! lamemp3enc ! mpegaudioparse ! mux.gst-launch-1.0 -v flvmux name=mux ! rtmpsink location=rtmp://wowza-server/application/stream videotestsrc ! video/x-raw, width=360, height=288 ! x264enc ! video/x-h264,profile=baseline ! h264parse ! mux. audiotestsrc wave=5 ! audioconvert ! lamemp3enc ! mpegaudioparse ! mux.gst-launch-1.0 -v flvmux name=mux ! rtmpsink location=rtmp://wowza-server/application/stream videotestsrc ! video/x-raw, width=360, height=288 ! omxh264enc ! video/x-h264,profile=baseline ! h264parse ! mux. audiotestsrc wave=5 ! audioconvert ! lamemp3enc ! mpegaudioparse ! mux.gst-launch-1.0 -v flvmux name=mux ! rtmpsink location=rtmp://wowza-server/application/stream videotestsrc ! video/x-raw, width=360, height=288 ! omxh264enc ! video/x-h264,profile=baseline ! h264parse ! mux.
- Flash Media Server
-
- Streaming to a Flash Media Server using the rtmpsink element
-
- audio/mpeg,mpegversion=4,stream-format=raw
- rtmpsink uses librtmp
gst-launch filesrc location=videofile ! decodebin name=decoder \
decoder. ! queue ! audioconvert ! audioresample ! osssink \
decoder. ! ffmpegcolorspace ! xvimagesink
- gstreamer dvb streaming
- Graphical editor
- gst-editor (only for gstreamer 0.8)
Desenvolupament / Development
- Documentació / Documentation
- Llenguatges / Languages
- C applications (using GObject and GLib)
-
- Tutorials
-
- Building applications using GStreamer
-
gcc basic-tutorial-1.c -o basic-tutorial-1 `pkg-config --cflags --libs gstreamer-1.0`
- Basic tutorials
-
- ...
- Basic tutorial 11: Debugging tools
- Basic tutorial 12: Streaming (same as basic-tutorial-1, but with buffer watch)
- ...
- Basic tutorial 14: Handy elements
-
- audiorate: it takes an incoming stream of time-stamped raw audio frames and produces a perfect stream by inserting or dropping samples as needed
- API
-
- GstMessage
- Python applications (using PyGObject)
-
- Configuració / Setup
- Python with PyGObject
- system
- or virtualenv
- GStreamer
- from packages
- ...
- or compiled and installed to /usr/local/bin
- check that you compiled GStreamer with PyGObject enabled
- you have to export:
export GI_TYPELIB_PATH=/usr/local/lib/girepository-1.0
- from packages
- Python with PyGObject
- PyGObject API
- Tutorials and examples
-
- Basic tutorials
- francesc.pinyol.m
/ python-gst-examples
- application_development
- base.py (common code)
- buffering.py
- data_probe.py
- gui_player.py (GUI player)
- helloworld.py
- media_stream_type_detection.py
- player.py (play a uri or file)
- tags.py
- ges
- ges_base_player.py (play a file)
- ges_base_renderer.py (render to a file)
- gst-transcoding
- base_target.py
- other
- tutorials
- application_development
- gkralik / python-gst-tutorial
-
- basic-tutorial-...py
- rubenrua / GstreamerCodeSnippets / Python /
-
- pygst-sdk-tutorials
-
- basic-tutorial-...py
- pygst-tutorial
- gst.Caps Example (programtalk)
- How te retrieve stream statistics in Gstreamer?
-
- RTP/RTCP server and client
gst_bus_add_signal_watch
- Problemes / Problems
-
- CTRL-C not working
- Solució / Solution
if __name__ == '__main__':
# to be able to use CTRL-C to quit
import signal
signal.signal(signal.SIGINT, signal.SIG_DFL)
...
- Solució / Solution
ValueError: Namespace Gst not available-
- Solució / Solution
-
- if you compiled gstreamer from source,
-
- check that typelib files where
installed (
lib64girepository-develmust be installed before compilation):
-
- /usr/local/lib/girepository-1.0/Gst*.typelib
- set environment variable:
-
export GI_TYPELIB_PATH=/usr/local/lib/girepository-1.0
- check that typelib files where
installed (
- aaaa
- aaaa
- CTRL-C not working
- GES
- CLI: ges-launch-1.0
- PyGObject API
- Exemples / Examples
- Configuració / Setup
- ...
- ...
- Resum / Summary
task
options
steps
C
Python (base.py)
methods
headers
#include <gst/gst.h>
#!/usr/bin/env python3
import sys
import gi
gi.require_version('Gst', '1.0')
from gi.repository import Gst, GObject, GLib
Initialize Gstreamer
/* init */
gst_init (&argc, &argv);
class Player(object):def __init__(self):# init GStreamer
Gst.init(None)
main GLib loop
(optional, but needed when using add_signal_watch)
GMainLoop *main_loop;
main_loop = g_main_loop_new (NULL, FALSE);
self.loop = GLib.MainLoop.new(None, False)Arguments
Usage /* check args */
if (argc != 2) {
g_print ("Usage: %s <filename>\n", argv[0]);
return -1;
}
# check input arguments
if len(sys.argv) != 2:
print("Usage: {0:s} <filename>".format(sys.argv[0]))
sys.exit(1)
Arguments
if Gst.Uri.is_valid(sys.argv[1]):
uri = sys.argv[1]
else:
uri = Gst.filename_to_uri(sys.argv[1])
print("uri: {0:s}".format(uri))Build the pipeline
option 1:
Build the pipeline by parsing (basic tutorial 1)
/* create a new pipeline with all the elements */
GstElement *pipeline;
pipeline = gst_parse_launch ("playbin uri=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm")
# create a new pipeline with all the elements (remove double quotes around caps, if any)
pipeline = Gst.parse_launch("playbin uri=https://www.freedesktop.org/software/gstreamer-sdk/data/media/sintel_trailer-480p.webm")
option 2:
Build the pipeline from elements
(basic tutorial 2)
Create the elements /* create the elements */
GstElement *pipeline, *source, *sink;
source = gst_element_factory_make ("videotestsrc", "source");
sink = gst_element_factory_make ("autovideosink", "sink");# create the elements
source = Gst.ElementFactory.make("videotestsrc", "source")
sink = Gst.ElementFactory.make("autovideosink", "sink")(or create the factory and then the element)
(Creating a GstElement, basic tutorial 6)
/* create factory and element */
factory = gst_element_factory_find ("fakesrc");
element = gst_element_factory_create (factory, "source");
# create factory and element
source_factory = Gst.ElementFactory.find("videotestsrc")
sink_factory = Gst.ElementFactory.find("autovideosink")
source = source_factory.create("source")
sink = sink_factory.create("sink")
Create the empty pipeline /* create an empty pipeline */
pipeline = gst_pipeline_new ("test-pipeline");# create an empty pipeline
pipeline = Gst.Pipeline.new("test-pipeline")Build the pipeline: add and link elements
/* add elements to the pipeline */
gst_bin_add_many (GST_BIN (pipeline), source, sink, NULL);
if (gst_element_link (source, sink) != TRUE) {
g_printerr ("Elements could not be linked.\n");
gst_object_unref (pipeline);
return -1;
}
# add elements to the pipeline
pipeline.add(source)pipeline.add(sink)
if not source.link(sink):
print("ERROR: Could not link source to sink")
sys.exit(1)
Modify properties /* set property of an element */
g_object_set (source, "pattern", 0, NULL);
# set property of an element
source.set_property("pattern", 0)
Connect element signal to a callback
def on_have_type(self, element, probability, caps, user_data):
# connect signal to a callback
typefind.connect("have-type", on_have_type, None)option 3:
Dinamically connect the elements in the pipeline
(basic tutorial 3)
Callback pad_added_handler(...)
def on_pad_added(self, ...):
Connect signal to callback g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);
# connect signal to a callbacksource.connect("pad-added", self.on_pad_added)Start playing
/* set the pipeline to playing state */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
# set the pipeline to playing state
pipeline.set_state(Gst.State.PLAYING)
Main loop
(two ways to use a bus)
Get bus associated to pipeline /* get the bus from the pipeline */
GstBus *bus;
bus = gst_element_get_bus (pipeline);# get the bus from the pipeline
bus = pipeline.get_bus()option 1:
Wait until error or EOS
(basic tutorial 1)
/* wait until error or end of stream */
GstMessage *msg;
msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, GST_MESSAGE_ERROR | GST_MESSAGE_EOS);
# wait until error or end of stream
terminate = False
while True:
try:
msg = bus.timed_pop_filtered(0.5 * Gst.SECOND, Gst.MessageType.ERROR | Gst.MessageType.EOS)
if msg:
terminate = True
except KeyboardInterrupt:
terminate = True
if terminate:
break
option 2:
GMain loop with callback
(basic tutorial 12)
start loop /* start main loop */
g_main_loop_run (main_loop);# start main loop
self.loop.run()data structure (to be available from callback)
typedef struct _CustomData {
gboolean is_live;
GstElement *pipeline;
GMainLoop *loop;
} CustomData;
CustomData data;
data.loop = main_loop;
data.pipeline = pipeline;self.loop
self.pipeline
option a: one single callback for all messages
static void cb_message (GstBus *bus, GstMessage *msg, CustomData *data) {
const GstStructure *structure;
structure = gst_message_get_structure (msg);
g_print ("Message name: %s\n", gst_structure_get_name (structure) );
switch (GST_MESSAGE_TYPE (msg)) {
...
}def on_message(self, bus, msg, user_data):
def on_sync_message(self, bus, msg, user_data):
/* message handler */
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message", G_CALLBACK (cb_message), &data);
# general message handler
bus.add_watch(GLib.PRIORITY_DEFAULT, self.on_message, None)
# another option:
# general message handler
bus.add_signal_watch()
bus.connect("message", self.on_message, None)
# sync message handler
bus.enable_sync_message_emission()
bus.connect('sync-message', self.on_sync_message, None)option b: each message has its callback
def on_error(self,bus, msg, user_data):
def on_eos(self,bus, msg, user_data):
def on_state_changed(self,bus, msg, user_data):
def on_application_message(self,bus, msg, user_data):
...
# individual message handler
bus.add_signal_watch()
bus.connect("message::error", self.on_error)
bus.connect("message::eos", self.on_eos)
bus.connect("message::state-changed", self.on_state_changed)
bus.connect("message::application", self.on_application_message)
...
quit loop
self.loop.quit()Free resources
/* free resources */
if (msg != NULL)
gst_message_unref (msg);
gst_object_unref (bus);
gst_element_set_state (pipeline, GST_STATE_NULL);
gst_object_unref (pipeline);
return 0;
# free resources
pipeline.set_state(Gst.State.NULL)
if __name__ == '__main__':
# to be able to use CTRL-C to quit
import signal
signal.signal(signal.SIGINT, signal.SIG_DFL)
p = Player()
- Application
Development
Manual
(pdf,
ps, html)
- About GStreamer
- What is GStreamer?
- Design principles
- Foundations
- Communication between application / bus / pipeline:
- buffers: streaming data between elements (downstream (sources->sinks)) (buffering)
- events: between elements or from the application to elements (upstream (sinks->sources), downstream (sources->sinks))
- messages: posted by elements on the pipeline's message bus (message types)
- queries: allow applications to request information such as duration or current playback position from the pipeline (upstream, downstream) (querying)
- Communication between application / bus / pipeline:
- Building
an
Application
- Initializing
GStreamer
-
C
Python #include <gst/gst.h>import gi
gi.require_version('Gst', '1.0')
from gi.repository import Gst, GObject, GLibgst_init (&argc, &argv);Gst.init(None)
-
- Elements
- Creating elements
C
Python option 1: factory and element
factory = gst_element_factory_find("fakesrc")
element = gst_element_factory_create(factory, "source")factory = Gst.ElementFactory.find("fakesrc")
source = factory.create("source")option 2: element
element = gst_element_factory_make("fakesrc", "source")source = Gst.ElementFactory.make("fakesrc", "source") - Accessing elements
pipeline = Gst.parse_launch("... ! queue name=cua_1 ... queue name=cua_2 ...")cua_1 = pipeline.get_by_name("cua_1")
- Properties
p1 = cua_1.get_property('propietat_1')cua_1.set_property(...)
- Signals
- signals defined by element:
def on_queue_running(queue, udata):
current_level_buffers = queue.get_property('current-level-buffers')
print("[on_cua_running] [{}] current-level-buffers: {}".format(udata, current_level_buffers))
cua_1 = pipeline.get_by_name('cua_1')
cua_1.connect('running', self.on_queue_running, "cua_1")cua_2= pipeline.get_by_name('cua_2')
cua_2.connect('running', self.on_queue_running, "cua_2")
signal on modified (set_property) property of element:myelement.connect('notify::name_of_the_property', on_mycallback)
- signals defined by element:
- Element States
name
C
Python description
gst_element_set_state (pipeline, GST_STATE_PLAYING);pipeline.set_state(Gst.State.PLAYING)
NULL
GST_STATE_NULL
Gst.State.NULL
the NULL state or initial state of an element. Transition to it will free all resources.
READY
GST_STATE_READYGst.State.READYthe element is ready to go to PAUSED
PAUSED
GST_STATE_PAUSEDGst.State.PAUSEDthe element is PAUSED, it is ready to accept and process data. Sink elements however only accept one buffer and then block
PLAYING
GST_STATE_PLAYINGGst.State.PLAYINGthe element is PLAYING, the clock is running and the data is flowing
- Creating elements
- Bins
-
C
Python
Bin
gst_bin_new()Gst.Bin.new()gst_bin_add()
gst_bin_remove()
gst_bin_get_by_name()
gst_bin_get_by_interface()
gst_bin_iterate_elements()
Pipeline
(special top-level type of bin)
gst_pipeline_new()pipeline = Gst.Pipeline.new("test-pipeline")
pipeline.add(...)
...
-
- Bus
-
C
Python get all messages
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));bus_watch_id = gst_bus_add_watch (bus, my_bus_callback, NULL);static gboolean my_bus_callback(GstBus *bus, GstMessage *msg, gpointer data)
...
bus = self.pipeline.get_bus()bus.add_watch(GLib.PRIORITY_DEFAULT, self.on_message, None)def on_message(bus, msg, user_data):
if (msg.type == ...):
...
elif (msg.type == ...):
...
else:
# Unhandled message
passget selected messages
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message::error", G_CALLBACK (cb_message_error), NULL);
g_signal_connect (bus, "message::eos", G_CALLBACK (cb_message_eos), NULL);bus = self.pipeline.get_bus()
bus.add_signal_watch()
bus.connect("message::error", self.on_error)
bus.connect("message::eos", self.on_eos)(needed loop)
loop = g_main_loop_new (NULL, FALSE);
g_main_loop_run (loop);
...
g_main_loop_unref (loop);loop = GLib.MainLoop.new(None, False)
loop.run()
...
loop.quit() - Tipus de
missatges / Message
types
-
C
Python
def on_message(bus, msg, user_data):
msg_structure_name = msg.get_structure().get_name()
defon_sync_message(bus, msg, user_data):
...message type GstBus Signals
to be specified in connect(...)
Gst.MessageType
message
Error (fatal problem),
warning (non-fatal problem),
information (not problem)
message::error
GST_MESSAGE_ERROR
gst_message_parse_error()
_parse_warning ()
_parse_info ()if (msg.type == Gst.MessageType.ERROR):
err, debug = msg.parse_error()
#msg.parse_warning()
# msg.parse_info()
print("Error: {0:s}".format(err.message))
self.loop.quit()End-of-stream
message::eos
GST_MESSAGE_EOSif (msg.type ==Gst.MessageType.EOS):
print("EOS")
self.loop.quit()Tags
GST_MESSAGE_TAG
gst_message_parse_tag()if (msg.type ==Gst.MessageType.TAG):
tags = msg.parse_tag()
tags.foreach(self.print_one_tag, None)
def print_one_tag(self, list, tag, user_data):
res, val = Gst.TagList.copy_value(list, tag)
print("%s: %s"% (Gst.tag_get_nick(tag), val))State-changes
GST_MESSAGE_STATE_CHANGE
gst_message_parse_state_changed ()if (msg.type ==
old, new, pending = msg.parse_state_changed()
print("State changed: {0:s} -> {1:s}".format(Gst.Element.state_get_name(old), Gst.Element.state_get_name(new)))message::async-done
GST_MESSAGE_ASYNC_DONEif (msg.type ==
...Buffering
message::buffering
GST_MESSAGE_BUFFERING
gst_message_parse_buffering (message, &percent);if (msg.type ==
percent = message.parse_buffering()
sync-message
Element
(specific to element; e.g. queue signals)
sync-message::element
if (msg.type ==
...Application-specific
gst_message_get_structure()
Threads
sync-message::stream-status
GST_MESSAGE_STREAM_STATUS
gst_message_parse_stream_status (message, &type, &owner);if (msg.type ==
type, owner = message.parse_stream_status()
-
-
- Pads
and
capabilities
- Pads
availability
examples
C
Python
pad = gst_element_get_static_pad(...)pad = gst_element_get_compatible_pad (mux, tolink_pad, NULL);pad = gst_element_get_request_pad (tee, "src%d");name = gst_pad_get_name (pad);
pad = convert.get_static_pad("sink")pad = mux.get_compatible_pad(tolink_pad)pad = tee.get_request_pad("src%d")name = pad.get_name()
always
dynamic (sometimes) pads
- demuxer
/* listen for newly created pads */
g_signal_connect (demux, "pad-added", G_CALLBACK (cb_new_pad), NULL);static void cb_new_pad(...)gst_element_set_state ()gst_element_sync_state_with_parent ()
# listen for newly created pads
self.demux.connect("pad-added", self.on_new_pad)def on_new_pad(self):...
request pads
(basic tutorial 3)
- multiplexer
- aggregator
- tee
- Capabilities
of
a pad
- GstCaps
- non-negotiated pad: one or more GstStructure
- negotiated pad: only one GstStructure (with fixed values)
- ...
- possible caps: obtained with gst-inspect
- allowed caps: subset of possible capabilities, depending on the possible caps of the peer pad
- negotiated caps:
- Types of caps:
-
type
Gst.Structure
values
empty
0
ANY
simple
1
variable field types
fixed
1
no variable field types
-
-
C
Python check type
gst_caps_is_fixed (caps)caps.is_fixed()get structure
str = gst_caps_get_structure (caps, 0);str = caps.get_structure(0)get value
gst_structure_get_int (str, "width", &width)width = str.get_int("width")creation of simple caps
caps = gst_caps_new_simple ("video/x-raw",
"format", G_TYPE_STRING, "I420",
"width", G_TYPE_INT, 384,
"height", G_TYPE_INT, 288,
"framerate", GST_TYPE_FRACTION, 25, 1,
NULL);caps = Gst.Caps.new_empty_simple("video/x-raw")
caps.set_value("format", "I420")caps.set_value("width", 384)caps.set_value("height", 288)caps.set_value("framerate", ...)
creation of full caps
caps = gst_caps_new_full (
gst_structure_new ("video/x-raw",
"width", G_TYPE_INT, 384,
"height", G_TYPE_INT, 288,
"framerate", GST_TYPE_FRACTION, 25, 1,
NULL),
gst_structure_new ("video/x-bayer",
"width", G_TYPE_INT, 384,
"height", G_TYPE_INT, 288,
"framerate", GST_TYPE_FRACTION, 25, 1,
NULL),
NULL);(unavailable) filtering using caps (internally creates a capsfilter)
link_ok = gst_element_link_filtered (element1, element2, caps);link_ok = element1.link_filtered(element2, caps) - Ghost pads
- "A ghost pad is a pad from some element in the bin that can be accessed directly from the bin as well."
-
C
Python gst_element_add_pad (bin, gst_ghost_pad_new ("sink", pad));ghost_pad = Gst.GhostPad.new("sink",pad)
- GstCaps
- Pads
- Buffers
and
events
- Buffers
- Events
- "Events are control particles that are sent both upstream (right to left) and downstream (left to right) in a pipeline along with buffers."
- Examples: seeking, flushes, end-of-stream notifications, ...
-
C
Python create
event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_NONE,
GST_SEEK_METHOD_SET, time_ns,
GST_SEEK_TYPE_NONE, G_GUINT64_CONSTANT (0));event = Gst.Event.new_seek(1.0, Gst.Format.TIME,
Gst.SeekFlags.NONE,
Gst.SeekType.SET, time_ns,
Gst.SeekType.NONE, 0)send
gst_element_send_event (element, event);element.send_event(event)
- Your first application
- Initializing
GStreamer
- Advanced
GStreamer
Concepts
- Position
tracking
and seeking (basic_tutorial_4)
-
C
Python Querying (queries)
static gboolean cb_print_position(GstElement *pipeline)gst_element_query_position (pipeline, GST_FORMAT_TIME, &pos)gst_element_query_duration (pipeline, GST_FORMAT_TIME, &len)
g_timeout_add (200, (GSourceFunc) cb_print_position, pipeline);g_timeout_add_seconds (1, (GSourceFunc) cb_print_position, pipeline);g_main_loop_run (loop);
from helper import format_ns
class Player(object):
def on_message(self, bus, msg, data):
if (msg.type == Gst.MessageType.ASYNC_DONE):
running_time = msg.parse_async_done()
# query_duration
if self.duration == Gst.CLOCK_TIME_NONE:
ret, duration = self.pipeline.query_duration(Gst.Format.TIME)
if ret:
self.duration = duration
print("ret: {}, duration: {}".format(ret, format_ns(duration)))
def cb_print_position(self):
ret, position = self.pipeline.query_position(Gst.Format.TIME)
if ret:
print("ret: {}, position: {}".format(ret, format_ns(position)))
return True
def __init__(self):
# media duration (ns)
self.duration = Gst.CLOCK_TIME_NONE
self.loop = GLib.MainLoop.new(None, False)
GLib.timeout_add(200, self.cb_print_position)
bus.add_signal_watch()
bus.connect("message", self.on_message, None)
self.loop.run()
def cb_print_position(pipeline)ret, pos = pipeline.query_position(Gst.Format.TIME)ret, len = pipeline.query_duration(Gst.Format.TIME)print("{0:f}/{1:f}".format(pos, len))
GLib.timeout_add(200, cb_print_position, pipeline)- the function is called repeatedly until it returns False
GLib.timeout_add_seconds(1, cb_print_position, pipeline)loop.run()
Events: seeking (and more)
gst_element_seek (pipeline, 1.0,
GST_FORMAT_TIME,
GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, time_nanoseconds,
GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE)gst_element_seek_simple (...)
pipeline.seek(1.0,
Gst.Format.TIME,
Gst.SeekFlags.FLUSH,
Gst.SeekType.SET, time_nanoseconds,
Gst.SeekType.NONE, Gst.CLOCK_TIME_NONE)pipeline.seek_simple(
Gst.Format.TIME,
Gst.SeekFlags.FLUSH | Gst.SeekFlags.KEY_UNIT,
30 * Gst.SECOND)
-
- Metadata
- Interfaces
- GstColorBalance
- GstVideoOverlay
- ...
- Clocks and synchronization in GStreamer
- Buffering
-
Stream buffering "Buffering up to a specific amount of data, in memory, before starting playback so that network fluctuations are minimized" buffer element: queue2
- low watermark
- high watermark
Download buffering "Download of the network file to a local disk with fast seeking in the downloaded data. This is similar to the quicktime/youtube players." buffering.py Timeshift buffering "Caching of (semi)-live streams to a local, on disk, ringbuffer with seeking in the cached area. This is similar to tivo-like timeshifting."
-
C
Python
buffering.py (download buffering)
messages
gst_message_parse_buffering (message, &percent);percent = message.parse_buffering()queries
query = gst_query_new_buffering (GST_FORMAT_TIME);
gst_element_query (pipeline, query)
gst_query_parse_buffering_percent (query, &busy, &percent);
gst_query_parse_buffering_range (query, NULL, NULL, NULL, &estimated_total);query = Gst.Query.new_buffering(Gst.Format.TIME)
pipeline.query(query)
busy, percent = query.parse_buffering_percent()
format, start, stop, estimated_total = query.parse_buffering_range()
-
- Dynamic
Controllable
Parameters
-
C
Python
GstControlSourceGst.ControlSourcecreate
csource = gst_interpolation_control_source_new ();
g_object_set (csource, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
attach to the gobject property
gst_object_add_control_binding (object, gst_direct_control_binding_new (object, "prop1", csource));
GstTimedValueControlSource *tv_csource = (GstTimedValueControlSource *)csource;
gst_timed_value_control_source_set (tv_csource, 0 * GST_SECOND, 0.0);
gst_timed_value_control_source_set (tv_csource, 1 * GST_SECOND, 1.0);
-
- Threads
- Scheduling in GStreamer
- pad can:
- push from upstream
- pull to downstream
- pad can:
- Configuring threads in GStreamer
- message:
STREAM_STATUS
- GST_STREAM_STATUS_TYPE_CREATE: when a new thread is about to be created -> you can configure a GstTaskPool in the GstTask
- when a thread is entered or left -> you can configure thread priorities
- when a thread starts, pauses and stops -> you can visualize the status of streaming in a gui application
- Boost priority of a thread
- message:
STREAM_STATUS
- When would you force a thread?
- queue2 (Stream buffering)
- queue
- Scheduling in GStreamer
- Autoplugging
- Playback components
- Media types as a way to identify streams
- capability: media types + set of properties
- Full list of Media Types and Properties
- Media stream type detection
- Dynamically autoplugging a pipeline
- Pipeline
manipulation
- Using probes
- Data probes
- Play a section of a media file
- Manually adding or removing data from/to a pipeline
- Forcing a format
- Dynamically changing the pipeline
- Using probes
- Position
tracking
and seeking (basic_tutorial_4)
- Higher-level interfaces for GStreamer applications
- Appendices
- ...
- About GStreamer
- Additional documentation
- GStreamer design documents
- Tutorials (source code in gst-docs/examples/tutorials)
(playback
tutorials are based on playbin)
...
C
Python Table of Concepts
Basic tutorials / Playback tutorials
Basic tutorials (lang=python) francesc.pinyol.m / python-gst-examples / tutorials
gkralik / python-gst-tutorial
GstreamerCodeSnippets
Python/pygst-sdk-tutorialsGstreamerCodeSnippets
Others/0.10/Python/pygst-sdk-tutorialsGstreamerCodeSnippets
pygst-tutorial
(class GTK_Main)
Basic tutorial 1: Hello world!
Basic tutorial 1: Hello world! basic-tutorial-1.py
basic-tutorial-1.py basic-tutorial-1.py
Bus
Elements
Links
Pipelines
Basic tutorial 2: GStreamer concepts
Basic tutorial 2: GStreamer concepts basic-tutorial-2.py
basic-tutorial-2.py
basic-tutorial-2-ex-vertigo.py
basic-tutorial-2.py
Pads
Signals
States
Basic tutorial 3: Dynamic pipelines
- CustomData data
- data.source
- data.convert
- data.sink
- g_signal_connect (data.source, "pad-added", G_CALLBACK (pad_added_handler), &data);
- basic-tutorial-3.py
basic-tutorial-3-ex-video.py
basic-tutorial-3.py
- class Player
- def __init__(self)
- self.source
- self.convert
- self.sink
- self.source.connect("pad-added", self.on_pad_added)
- def on_pad_added(self, src, new_pad)
- def __init__(self)
basic-tutorial-3.py
- def pad_added_handler(src, new_pad, data)
- data["source"]
- data["convert"]
- data["sink"]
- data["source"].connect("pad-added", pad_added_handler, data)
(you can jump to Playback tutorials)
-
Queries
Seeks
Basic tutorial 4: Time management
- basic-tutorial-4.py basic-tutorial-4.py
- class Player
- def __init__
- def play
- bus = self.playbin.get_bus()
- msg = bus.timed_pop_filtered(100 * Gst.MSECOND, (Gst.MessageType.STATE_CHANGED | Gst.MessageType.ERROR | Gst.MessageType.EOS | Gst.MessageType.DURATION_CHANGED))
- if msg: self.handle_message(msg)
- self.playbin.query_position
- self.playbin.query_duration
- self.playbin.seek_simple
- def handle_message
- query = Gst.Query.new_seeking(Gst.Format.TIME)
- if self.playbin.query(query):
basic-tutorial-4.py
GUI
Basic tutorial 5: GUI toolkit integration
- gst_bus_add_signal_watch (bus);
- basic-tutorial-5.py basic-tutorial-5.py
- class Player
- def __init__
- Gtk.init(sys.argv)
- Gst.init(sys.argv)
- # connect to interesting signals in
playbin
self.playbin.connect("video-tags-changed", self.on_tags_changed)
self.playbin.connect("audio-tags-changed", self.on_tags_changed)
self.playbin.connect("text-tags-changed", self.on_tags_changed) - # instruct the bus to emit signals for
each received message
# and connect to the interesting signals
bus = self.playbin.get_bus()
bus.add_signal_watch()
bus.connect("message::error", self.on_error)
bus.connect("message::eos", self.on_eos)
bus.connect("message::state-changed", self.on_state_changed)
bus.connect("message::application", self.on_application_message)
- def start
- GLib.timeout_add_seconds(1, self.refresh_ui)
- Gtk.main()
- def cleanup
- def build_ui
- def on_realize
- def on_play
- def on_pause
- def on_stop
- ...
- def on_tags_changed
- self.playbin.post_message
- def on_error
- def on_eos
- def on_state_changed
- ...
- def analyze_streams
- def __init__
Capabilities
Basic tutorial 6: Media formats and Pad Capabilities
- static void print_pad_capabilities
- /* Retrieve negotiated caps (or acceptable
caps if negotiation is not finished yet) */
caps = gst_pad_get_current_caps (pad);
if (!caps)
caps = gst_pad_query_caps (pad, NULL);
- /* Retrieve negotiated caps (or acceptable
caps if negotiation is not finished yet) */
- basic-tutorial-6.py basic-tutorial-6.py
- def print_field
- def print caps(caps, pfx)
- structure = caps.get_structure(i)
- structure.foreach(print_field, pfx)
- def print_pad_templates_information(factory)
- pads = factory.get_static_pad_templates()
- for pad in pads:
- padtemplate = pad.get()
- if padtemplate.get_caps():
- print_caps(padtemplate.get_caps(), " ")
- def print_pad_capabilities(element, pad_name)
- pad = element.get_static_pad(pad_name)
- # retrieve negotiated caps (or acceptable caps
if negotiation is not yet finished)
caps = pad.get_current_caps()
if not caps:
caps = pad.get_allowed_caps() - print_caps(caps, " ")
- def main
- sink_factory = Gst.ElementFactory.find("autoaudiosink")
- print_pad_templates_information(sink_factory)
- sink = sink_factory.create("sink")
- print_pad_capabilities(sink, "sink")
Pad availability
- always
- sometimes
- on request
Basic tutorial 7: Multithreading and Pad Availability
- basic-tutorial-7.py basic-tutorial-7.py
- def main
- # manually link the tee, which has "Request"
pads
tee_src_pad_template = tee.get_pad_template("src_%u")
tee_audio_pad = tee.request_pad(tee_src_pad_template, None, None)
audio_queue_pad = audio_queue.get_static_pad("sink")
tee_audio_pad.link(audio_queue_pad)
tee_video_pad = tee.request_pad(tee_src_pad_template, None, None)
video_queue_pad = video_queue.get_static_pad("sink")
tee_video_pad.link(video_queue_pad)
- # manually link the tee, which has "Request"
pads
Buffers
- GstBuffer: chunk of data. Can contain multiple GstMemory (memory buffer)
Basic tutorial 8: Short-cutting the pipeline (same as 7, replacing audiotestsrc -> appsrc; adding a third branch appsink)
- appsrc
- appsink
- /* Configure appsrc */
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
audio_caps = gst_audio_info_to_caps (&info);
g_object_set (data.app_source, "caps", audio_caps, NULL);
g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data);
g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
- basic-tutorial-8.py
Discoverer
Basic tutorial 9: Media information gathering
- GstDiscoverer
- basic-tutorial-9.py
- class Discoverer
- def print_tag_foreach
- def print_stream_info
- def print_topology
- def on_discovered
- tags = info.get_tags()
- def on_finished
- self.loop.quit()
- def __init__
- self.loop = GLib.MainLoop.new(None, False)
- self.loop.run()
- get video size:
gst-discoverer-1.0
gst-launch-1.0
Tools
Basic tutorial 10: GStreamer tools
- gst-launch-1.0
- elements
- properties
- named elements
- pads
- caps filters
- gst-inspect-1.0
- gst-discoverer-1.0
Debugging
Basic tutorial 11: Debugging tools
Basic tutorial 12: Streaming
- Setting live streams to PAUSED succeeds, but returns GST_STATE_CHANGE_NO_PREROLL, instead of GST_STATE_CHANGE_SUCCESS to indicate that this is a live stream.
- basic-tutorial-12.py
- class Player
- def on_message
- if (msg.type == Gst.MessageType.ERROR):
- elif (msg.type == Gst.MessageType.EOS):
- elif (msg.type == Gst.MessageType.BUFFERING):
- elif (msg.type == Gst.MessageType.CLOCK_LOST):
- def __init__
- self.pipeline = Gst.parse_launch(...)
- bus = self.pipeline.get_bus()
- ret = self.pipeline.set_state(Gst.State.PLAYING)
print("ret: {0:d}".format(ret))
if ret == Gst.StateChangeReturn.FAILURE:
print("Unable to set the pipeline to the playing state")
sys.exit(1)
elif ret == Gst.StateChangeReturn.NO_PREROLL:
self.is_live = True - self.loop = GLib.MainLoop.new(None, False)
- bus.add_signal_watch()
bus.connect("message", self.on_message) - self.loop.run()
- def on_message
Basic tutorial 13: Playback speed
-
Basic tutorial 14: Handy elements
- Bins
- playbin
- uridecodebin
- decodebin
- File input/output
- filesrc
- filesink
- Network
- souphttpsrc
- Test media generation
- videotestsrc
- audiotestsrc
- Video adapters
- videoconvert
- videorate
- videoscale
- Audio adapters
- audioconvert
- audioresample
- audiorate
- Multithreading
- queue
- queue2
- multiqueue
- tee
- Capabilities
- Debugging
- fakesink
- identity
-
Basic tutorial 16: Platform-specific elements
Action signals
Audio switching
Tags
Playback tutorial 1: Playbin usage
- gst_bus_add_watch(bus);
Subtitles
Playback tutorial 2: Subtitle management
Playback tutorial 3: Short-cutting the pipeline
Playback tutorial 4: Progressive streaming
Playback tutorial 5: Color Balance
Playback tutorial 6: Audio visualization
Playback tutorial 7: Custom playbin sinks
Playback tutorial 8: Hardware-accelerated video decoding
Playback tutorial 9: Digital audio pass-through
- Encoding profiles and targets
- Exemples / Examples
- francesc.pinyol.m / python-gst-examples /
- ...
- Estructura / Structure
- ...
-
syntax
C
Python
Encoding target
- location of target files (*.gep)
$GST_DATADIR/gstreamer-GST_API_VERSION/encoding-profiles//usr/share/gstreamer-1.0/encoding-profiles/*.gep
$HOME/gstreamer-GST_API_VERSION/encoding-profiles/~/.local/share/gstreamer-1.0/encoding-profiles/<category>/<name>.gep$GST_ENCODING_TARGET_PATH/- Pitivi:
/usr/share/pitivi/gstpresets/*.gep
- $(target.category)/$(target.name).gep
[GStreamer Encoding Target]
name : <encoding_target_name>
category : <category>
\description : <description> #translatable
[profile-<profile1name>]
name : <encoding_profile_name>
\description : <description> #optional
format : <format>
preset : <preset>
[streamprofile-<id>]
parent : <encodingprofile.name>[,<encodingprofile.name>..]
\type : <type> # "audio", "video", "text"
format : <format>
preset : <preset>
restriction : <restriction>
presence : <presence>
pass : <pass>
variableframerate : <variableframerate>
- device/mp4target.gep
(encoding_target_name=mp4target,
encoding_profile_names=mp4, ...)
[GStreamer Encoding Target]
name=mp4target
category=device
description=MP4 (H.264, AAC) target
[profile-mp4]
name=mp4
type=container
description[c]=MP4 container profile
format=video/quicktime, variant=(string)iso
[streamprofile-mp4-0]
parent=mp4
type=video
format=video/x-h264
restriction=video/x-raw
presence=0
pass=0
variableframerate=false
[streamprofile-mp4-1]
parent=mp4
type=audio
format=audio/mpeg, mpegversion=(int)4
restriction=audio/x-raw
presence=0
[profile-(null)]
type=audio
format=audio/mpeg, mpegversion=(int)4
GstEncodingTarget
- gst_encoding_list_all_targets
GstPbutils.EncodingTarget
# create target with all profiles
# The name and category can only consist of lowercase ASCII letters for the first character, followed by either lowercase ASCII letters, digits or hyphens (‘-‘).
name = "mp4target"
category = GstPbutils.ENCODING_CATEGORY_DEVICE # "device"
description = "MP4 (H.264, AAC) target"
profiles = [container_profile, video_profile, audio_profile]
target = GstPbutils.EncodingTarget.new(name, category, description, profiles)# save target to ~/.local/share/gstreamer-1.0/encoding-profiles/<category>/<name>.gep
ret = target.save()# list targets for all categories
category = None
target_list = GstPbutils.encoding_list_all_targets(category)
print("target_list: {}".format(target_list))
Encoding profile
(gst-validate-transcoding)
- general syntax of serialized encoding profile:
mux_format:[video_restriction->]video_format[+video_preset][|video_presence]:[audio_restriction->]audio_format[+audio_preset][|audio_presence]
- element
factory:
- <muxer_factory_name>:<video_encoder_factory_name>:<audio_encoder_factory_name>
webmmux:vp8enc:vorbisenc
- caps:
- <muxer_source_caps>:<video_encoder_source_caps>:<audio_encoder_source_caps>
- WebM (VP8 + Vorbis):
video/webm:video/x-vp8:audio/x-vorbis - MP4 (H.264 + MP3):
video/quicktime,variant=iso:video/x-h264:audio/mpeg,mpegversion=1,layer=3 - MP4 (H.264 + AAC):
video/quicktime,variant=iso:video/x-h264:audio/mpeg,mpegversion=4 - OGG (Theora + Vorbis):
application/ogg:video/x-theora:audio/x-vorbis - MPEG-TS (H.264 + AC3):
video/mpegts:video/x-h264:audio/x-ac3
- caps
+ preset:
- location of preset files
video/webm:video/x-vp8+youtube-preset:audio/x-vorbis"video/quicktime,variant=iso:video/x-h264+Profile Main:audio/mpeg,mpegversion=1,layer=3""video/quicktime,variant=iso:video/x-h264+slow12mbps:audio/mpeg,mpegversion=1,layer=3""video/quicktime,variant=iso:video/x-raw,width=1920,height=1080->video/x-h264+slow12mbps:audio/mpeg,mpegversion=1,layer=3""video/quicktime,variant=iso:video/x-h264,width=1920,height=1080+slow12mbps:audio/mpeg,mpegversion=1,layer=3"- How to use the x264 encoding presets when rendering an XGES project
- How to use the x264 encoding presets when rendering an XGES project (April 12, 2018)
- caps + presence (number of times an encoding
profile can be used inside an encodebin; 0:
any):
video/webm:video/x-vp8|1:audio/x-vorbis
- caps + restriction:
- ...:restriction_caps->encoded_format_caps:...
"video/webm:video/x-raw,width=1920,height=1080->video/x-vp8:audio/x-vorbis"- "
video/quicktime,variant=iso:video/x-raw,format=I420->video/x-h264+Profile High:audio/mpeg,mpegversion=4" "matroskamux:x264enc,width=1920,height=1080:audio/x-vorbis"
- loading profile from encoding
target:
target_name[/profilename/category]/path/to/target.gep:profilename
GstEncodingProfile
GstPbutils.EncodingProfile
Container profile
GstEncodingContainerProfile
GstPbutils.EncodingContainerProfile
# container profile
name = "mp4"
description = "MP4 container profile"
#container_caps = "video/webm"
container_caps = "video/quicktime,variant=iso"
format = Gst.Caps(container_caps)
preset = None
container_profile = GstPbutils.EncodingContainerProfile.new(name, description, format, preset)
Video profile
GstEncodingVideoProfile
GstPbutils.EncodingVideoProfile
# video profile
#video_caps = "video/x-vp8"
video_caps = "video/x-h264"
format = Gst.Caps(video_caps)
preset = None
restriction = Gst.Caps("video/x-raw")
presence = 0 # allow any number of instances of this profile
video_profile = GstPbutils.EncodingVideoProfile.new(format, preset, restriction, presence)
container_profile.add_profile(video_profile)
Audio profile
GstEncodingAudioProfile
GstPbutils.EncodingAudioProfile
# audio profile
#audio_caps = "audio/x-vorbis"
audio_caps = "audio/mpeg,mpegversion=4" # AAC
format = Gst.Caps(audio_caps)
preset = None
restriction = Gst.Caps("audio/x-raw")
presence = 0 # allow any number of instances of this profile
audio_profile = GstPbutils.EncodingAudioProfile.new(format, preset, restriction, presence)
container_profile.add_profile(audio_profile)
Preset
- location of preset files (if a file with the
same name is found in more than one directory,
only the last one is taken into account):
- /usr/share/gstreamer-1.0/presets/*.prs
- /usr/local/share/gstreamer-1.0/presets/*.prs
- GstX264Enc.prs
...
[Profile Main]
...
- GstX264Enc.prs
- /usr/share/pitivi/gstpresets/*.prs,
- ~/.local/share/gstreamer-1.0/presets/*.prs
- GST_PRESET_PATH
- ~/.local/share/gstreamer-1.0/presets/GstX264Enc.prs
- <profile_name>.prs
- ...
Example: Using an encoder preset with a profile
preset = GST_PRESET (gst_element_factory_make ("theoraenc", "theorapreset"));
g_object_set (preset, "bitrate", 1000, NULL);
// The preset will be saved on the filesystem,
// so try to use a descriptive name
gst_preset_save_preset (preset, "theora_bitrate_preset");
gst_object_unref (preset);
Gst.Preset
preset = Gst.ElementFactory.make("theoraenc", "theorapreset")
preset.set_property("bitrate", 1000)
# save to ~/.local/share/gstreamer-1.0/presets/GstTheoraEnc.prs
preset.save_preset("theora_bitrate_preset")preset = Gst.ElementFactory.make("x264enc", "x264preset")
# will create a preset based on existing preset "Profile High"
preset.load_preset("Profile High")
preset.set_property("bitrate", 1000)# save to ~/.local/share/gstreamer-1.0/presets/GstX264Enc.prs
# (it will also include presets in /usr/local/share/gstreamer-1.0/presets/GstX264Enc.prs)
preset.save_preset("profile_high_1000")
- location of target files (*.gep)
- Exemples / Examples
- gst-editing-services
- GStreamer Editing Services (API Reference)
- Exemples / Examples
- ges-launch
- francesc.pinyol.m/python-gst-examples/ges
- Fun with video mixer (*)
- cut parts of a video using gstreamer/Python (gnonlin?)
- ...
- Resum / Summary
- Estructura / Structure (see Pitivi):
- pipeline
- timeline
- output (usually one video track and one
audio track)
- track_1 (can be GES.VideoTrack,
GES.AudioTrack)
- ...
- track_2
- ...
- track_1 (can be GES.VideoTrack,
GES.AudioTrack)
- input (several layers,
each one with several clips; each clip is a
fragment of an asset (uri), put at certain
position at the layer)
- layer_1
- clip_asset_1.1
- clip_asset_1.2
- ...
- layer_2
- clip_asset_2.1
- ...
- layer_1
- output (usually one video track and one
audio track)
- timeline
- pipeline
-
Python (see complete examples)
headers
import sys
import gi
gi.require_version('Gst', '1.0')
from gi.repository import Gst, GLib
init Gst
class Player(object):def __init__(self):# init GStreamer
Gst.init(None)GES Initialization
# workaround to avoid "g_array_append_vals: assertion 'array' failed" when importing GES before Gst.init (using python3)
gi.require_version('GES', '1.0')
from gi.repository import GES
# init GES
GES.init()main GLib loop
# create main glib loop
self.loop = GLib.MainLoop.new(None, False)create timeline (GESTimeline)
# create timeline with one audio track and one video track
#timeline = GES.Timeline.new_audio_video() #when using this helper function, VideoTrack is 1280*720
timeline = GES.Timeline.new()
video_track = GES.VideoTrack.new()
# if update_restriction_caps was not called, VideoTrack would be 1280*720
video_track.update_restriction_caps( Gst.Caps.from_string("video/x-raw,width=1920,height=1080") )
timeline.add_track(video_track)
audio_track = GES.AudioTrack.new()
timeline.add_track(audio_track)create asset/clip
# create asset
asset = GES.UriClipAsset.request_sync(uri)create layer in timeline
# create layer
layer = timeline.append_layer()put clips in layer
# put clip in layer
# start=0.0
start_on_timeline = 0
# inpoint=60.0
start_position_asset = inpoint * Gst.SECOND
# duration=5.0
duration = duration * Gst.SECOND
clip = layer.add_asset(asset, start_on_timeline, start_position_asset,
duration, GES.TrackType.UNKNOWN)create GES pipeline
# create GES pipeline
pipeline = GES.Pipeline()connect message bus to callback
# connect bus messages to callback
bus = pipeline.get_bus()
bus.add_signal_watch()
bus.connect("message", self.on_message, None)add timeline to pipeline
# add timeline to pipeline
pipeline.set_timeline(timeline)(optional: only render)
ges_base_renderer.py
containers for output format
# container profile
name = "mp4"
description = "MP4 container profile"#container_caps = "video/webm"
container_caps = "video/quicktime,variant=iso"format = Gst.Caps(container_caps)
preset = None
container_profile = GstPbutils.EncodingContainerProfile.new(name, description, format, preset)
# video profile
#video_caps = "video/x-vp8"
video_caps = "video/x-h264"
format = Gst.Caps(video_caps)
preset = None
restriction = Gst.Caps("video/x-raw")
presence = 0 # allow any number of instances of this profile
video_profile = GstPbutils.EncodingVideoProfile.new(format, preset, restriction, presence)
container_profile.add_profile(video_profile)
# audio profile
#audio_caps = "audio/x-vorbis"
audio_caps = "audio/mpeg,mpegversion=4"
format = Gst.Caps(audio_caps)
preset = None
restriction = Gst.Caps("audio/x-raw")
presence = 0 # allow any number of instances of this profile
audio_profile = GstPbutils.EncodingAudioProfile.new(format, preset, restriction, presence)
container_profile.add_profile(audio_profile)pipeline in render mode
# pipeline in render mode
pipeline.set_render_settings(output_uri, container_profile)
pipeline.set_mode(GES.PipelineFlags.RENDER)
# same as: ges-launch-1.0 --smart-rendering
# pipeline.set_mode(GES.PipelineFlags.SMART_RENDER)progress
# progress
GLib.timeout_add(300, self.duration_querier, pipeline)start
# start playing pipeline
pipeline.set_state(Gst.State.PLAYING)
self.loop.run()stop
# unset
pipeline.set_state(Gst.State.NULL)
- Estructura / Structure (see Pitivi):
- Problemes / Problems
- gst-transcoding
- CLI tool: gst-transcoder-1.0
- Exemples / Examples
- Create a target file called device/mp4target.gep
- gst-transcoder-1.0 <input_file> <output_file>
- ...
- ...
Projectes / Projects
- Web overlay
- Based on WPEWebKit
- Based on CEF (Chromium Embedded Framework)
- Live mixers
http://www.francescpinyol.cat/gstreamer.html
Primera versió: / First version: 27.X.2018
Darrera modificació: 29 de setembre de 2024 / Last update: 29th
September 2024
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