GStreamer

Índex

General

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
      activate non-free and tainted for dnf
      • 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 bison lib64girepository-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
      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 install qt5-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
      • 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 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
      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
      1. echo "/usr/local/lib64" > /etc/ld.so.conf.d/local.conf
      2. mkdir -p ~/src && cd ~/src
      3. git clone https://gitlab.freedesktop.org/gstreamer/gstreamer.git
      4. cd gstreamer
      5. if you compiled ffmpeg:
        1. PKG_CONFIG_PATH=/usr/local/lib/pkgconfig meson -Dgpl=enabled -Dtests=disabled -Dexamples=disabled -Dgst-python:libpython-dir=/usr/lib/ -DFFmpeg:nonfree=enabled [--wipe] build
        2. PKG_CONFIG_PATH=/usr/local/lib/pkgconfig:$PWD/build/meson-private ninja -C build
      6. if you installed ffmpeg-devel:
        1. meson -Dgpl=enabled -Dtests=disabled -Dexamples=disabled -Dgst-python:libpython-dir=/usr/lib/ -DFFmpeg:nonfree=enabled [--wipe] build
        2. PKG_CONFIG_PATH=$PWD/build/meson-private ninja -C build
      7. sudo ninja -C build install
      8. sudo ldconfig
      9. to uninstall
        1. sudo ninja -C build uninstall
    • Problems
    • ...
  • 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-devel autoconf 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-devel libopus-dev libvorbis-dev libogg-dev libtheora-dev libxv-dev libpango1.0-dev
        gst-plugins-good


        libvpx-devel libsoup-devel
        libvpx-devel pulseaudio-libs-devel libsoup-devel libvpx-dev
        gst-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-devel libx264-dev
        gst-python


        python-gobject3-devel
        python-devel pygobject3-devel
        gst-libav






        gstreamer-editing-services

        gst-editing-services libxml2-devel
        libxml2-devel
      • Mageia
        • urpmi autoconf 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.rpm
            • yum -y install librtmp-devel
          • gst-plugins-ugly
            • yum install libx264-devel
    • 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 configure
            • meson configure gst-python
          • get present values for options
            • meson configure build
          • set options
            • meson -D...=... build
            • ...
        • ninja
          • build all
            • ninja -C build
          • clean all
            • ninja -C build clean
          • install
            • sudo ninja -C build install
          • uninstall
            • sudo ninja -C build uninstall
        • 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-devel x264-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-devel
                python3-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 install
                libsoup-devel
              • # gst-plugins-bad
                ...
              • # gst-plugins-ugly
                ...
              • # gst-python
                sudo dnf install
                pygobject3-devel
            • CentOS 7
          • Mageia
          • pip3 install --user meson
        • Passos / Steps
          1. cd ~/src
          2. git clone https://gitlab.freedesktop.org/gstreamer/gst-build
          3. cd gst-build
          4. meson build --buildtype=debug
          5. ninja -C build
          6. sudo ninja -C build install
        • Checkout another branch using worktrees
          1. cd ~/src
          2. git clone https://gitlab.freedesktop.org/gstreamer/gst-build
          3. cd gst-build
          4. ./gst-worktree.py add gst-build-1.18 origin/1.18
          5. cd gst-build-1.18
          6. meson build
            • meson -Dgst-python:libpython-dir=/usr/lib/ -Dtests=disabled -Dexamples=disabled build
          7. ninja -C build
          8. sudo pip3 install meson
          9. sudo ninja -C build install
          10. sudo sudo sh -c 'echo "/usr/local/lib64" >/etc/ld.so.conf.d/local.conf'
          11. sudo ldconfig
          12. uninstall
            1. cd gst-build-1.18
            2. sudo ninja -C build uninstall
        • Optional Installation
        • Problemes / Problems
      • 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.sh 1.16.2
      • 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
        • 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
      • gstreamer will be installed in:
        • /usr/local/lib/gstreamer-1.0/
    • Incidències / Issues (GitLab Issues) (others: gstreamer-devel list, Bugs)

Ús / Usage

  • Edició / Edit
  • Acceleració / Acceleration
  • Raspberry Pi
  • 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 ...
      • 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-rendering
          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"
          +clip
          • <path|uri>
          • inpoint[i]=
          • duration[d]=
          • start[s]=
          • layer[l]=
          • set-
            • alpha
            • posx
            • posy
            • width
            • height
            • volume
            • mute


          +test-clip
          • smpte
          • circular
          • ...


          +effect
          • <bin-description>
            • agingtv
            • videocrop
            • ...
          • +effect "agingtv"

          set-


          +title
          • <text>
          • duration[d]=
          • ...
          • +title "abc" duration=2.0

      • 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 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
        • 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 --load bbb.xges
        • render to file according to project:
          • # output is forced to 720p
            ges-launch-1.0 --load
            bbb.xges -o toto.mp4
          • # output preserves input resolution
            ges-launch-1.0 --load
            bbb.xges --smart-rendering -o toto.mp4
        • launch pitivi with this project:
          • pitivi bbb.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
                • ...
        • render to Ogg - Theora - Vorbis (default encoding profile):
          • ges-launch-1.0 +clip bbb_720p.mp4 i=4.0 d=2.0 -o bbb.ogg
          • ges-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
        • 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
                • urpmi gstreamer1.0-x264
                • urpmi gstreamer1.0-fdkaac
            • ERROR from element qtdemux1: Internal data stream error.
              • Solució / Solution
                • urpmi gstreamer1.0-x264
        • 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 ...
        • ...
    • 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/mp4
        • gst-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 + demux
        • mux + output
        • audio
        • video
      • 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:DEBUG
        • export 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
            • mkdir /tmp/dots
            • export GST_DEBUG_DUMP_DOT_DIR=/tmp/dots
            • gst-launch ...
            • cd /tmp/dots
            • to generate svg:
              • dot -Tsvg ...-gst-launch...PLAYING....dot >pipeline.svg
              • gwenview pipeline.svg
            • to generate png:
              • dot -Tpng ...-gst-launch...PLAYING....dot >pipeline.png
              • gwenview pipeline.png
      • Sintaxi / Syntax
        • element:
          ELEMENTTYPE [PROPERTY1 ...]
          elements can be put into bins:
          [BINTYPE.] ( [PROPERTY1 ...] PIPELINE-DESCRIPTION )
          property:
          NAME=*[(TYPE)]*VALUE in lists and ranges: *[(TYPE)]*VALUE
          • range: [VALUE,VALUE]
          • list: {VALUE[,VALUE...]}
          type:
          • -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 ...]]
        • 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
          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
            • video/x-h264,
              profile=baseline
          • video
            • h264parse
          • audio
          • mpegtsmux name=mux
          • flvmux  streamable=true name=mux
          • mp4mux faststart=true
          file
          • filesink location=music.ogg
          devices
          network
          • udpsink host=192.168.0.8 port=5004 sync=false
          • rtmpsink location=rtmp://rtmp_server:1935/app/stream
          unix
          • fdsink
          • shmsink socket-path=...
            shm-size=...
            wait-for-connection=...
          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
          flvdemux name=demux
          • audio
          • video
          ...


          • video
            • mpegvideoparse
            • h264parse
          • audio
            • mpegaudioparse
            • aacparse
          • decodebin
          • video
            • omxmpeg2videodec
            • omxh264dec
          • audio
            • avdec_mpeg2video





          • video
            • video/x-raw, framerate=25/1, width=640, height=360,
            • if format is specified, videoconvert must be specified after it
              • format=...,
                ...
          • audio
            • audio/x-raw,
              format=...,
              layout=...,
              rate=...,
              channels=...

      • 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
          • async
          • sync
            • the source must be specified with do-timestamp=true
        • 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
        • 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=0
            • gst-launch-1.0 -v videotestsrc 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=true
            • gst-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
          • du -h /dev/shm
          • ls -l /dev/shm
          • netstat -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
      • Bins
      • Play
        • general: using playbin
        • 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
            • 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 ! autovideosink
            • gst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=12/1,width=1280,height=720,format=BGRA ! videoconvert ! autovideosink
            • test audio/video sync
            • test judder
              • gst-launch-1.0 -v videotestsrc pattern=bar horizontal-speed=-40 ! video/x-raw,framerate=25/1,width=1280,height=720 ! autovideosink
              • gst-launch-1.0 -v videotestsrc pattern=bar horizontal-speed=-40 ! video/x-raw,framerate=60000/1001,width=1280,height=720 ! autovideosink
            • clock overlay
              • gst-launch-1.0 -v videotestsrc is-live=true ! clockoverlay halignment=right valignment=top shaded-background=true font-desc="Sans, 24" ! autovideosink
            • time overlay
              • gst-launch-1.0 -v videotestsrc is-live=true ! timecodestamper ! timeoverlay shaded-background=true 'time-mode=time-code'  font-desc="Sans, 24" ! autovideosink
              • gst-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 pattern_0



              1
              snow

              pattern_1




              2
              black

              pattern_2




              3
              white

              pattern_3




              4
              red
              pattern_4




              5
              green
              pattern_5




              6
              blue
              pattern_6




              7
              checkers-1
              pattern_7




              8
              checkers-2
              pattern_8




              9
              checkers-4
              pattern_9




              10
              checkers-8
              pattern_10




              11
              circular
              pattern_11




              12
              blink
              pattern_12




              13
              smpte75
              pattern_13




              14
              zone-plate BBC R&D Report 1978/23 pattern_14


              • k0
              • kt
              • kt2
              • kx
              • kx2
              • kxt
              • ky
              • ky2
              • kyt
              • xoffset
              • yoffset
              • videotestsrc pattern=zone-plate kx2=20 ky2=20 kt=1
              15
              gamut
              pattern_15




              16
              chrome-zone-plate
              pattern_16




              17
              solid-color
              pattern_17




              18
              ball
              pattern_18


              • videotestsrc pattern=ball animation-mode=wall-time motion=sweep
              19
              smpte100
              pattern_19




              20
              bar

              pattern_20




              21
              pinwheel
              pattern_21




              22
              spokes
              pattern_22




              23
              gradient
              pattern_23




              24 colors
              pattern_24




              25 smpte-rp-219
              pattern_25




          • 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
                    • ffmpeg -codecs | grep aac
            • 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 ! autoaudiosink
            • 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 ! 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_file
          • gst-launch-1.0 \
                filesrc location=${video_path} ! decodebin name=dec \
                dec. ! queue ! autovideosink \
                dec. ! queue ! autoaudiosink
          • MP4
            • gst-launch-1.0 -v playbin uri=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 playbin uri=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
          • 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 playbin uri=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=true
              • gst-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-launch filesrc 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
              • 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:
                    • ...
                • 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”
              • 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 ! autovideosink
            • gst-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 ...
            • 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
          • 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
                • ./configure
                • make
                • sudo make install
            • gst-launch-1.0 playbin uri=http://download.blender.org/peach/bigbuckbunny_movies/BigBuckBunny_320x180.mp4
          • RTMP
            • gst-launch-1.0 -v playbin uri=rtmp://nginx-server/myapp/mystream
            • gst-launch-1.0 -v playbin uri=${source}
            • gst-launch-1.0 uridecodebin uri=${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 ! 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=${source} do-timestamp=true ! queue2 use-buffering=true ! decodebin name=decoder decoder.src_0 ! queue ! autovideosink sync=true decoder.src_1 ! queue ! audioconvert ! autoaudiosink sync=true
            • gst-launch-1.0 -v \
                                rtmpsrc location=${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=${source} do-timestamp=true ! flvdemux name=demux \
                  demux.video ! queue ! decodebin ! autovideosink sync=true \
                  demux.audio ! queue ! decodebin ! autoaudiosink sync=true
      • Mux to
        • test to TS
          • gst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=24/1,width=1280,height=720 ! videoconvert ! x264enc ! video/x-h264,profile=high ! mpegtsmux ! filesink location=toto.ts
        • test to MP4
          • NOTE: stream-format=(string)byte-stream is not supported by MP4
          • gst-launch-1.0 -v videotestsrc ! video/x-raw,framerate=24/1,width=1280,height=720 ! videoconvert ! x264enc ! video/x-h264,profile=high ! mp4mux ! filesink location=toto.mp4
            • Problem:
              • moov atom not found
                • Solution
                  • (?) mp4mux faststart=true
      • Captura / Capture
        • capture a single frame to a png file
          • gst-launch-1.0 -v videotestsrc num-buffers=1 ! pngenc ! filesink location=toto.png
        • capture timestamped frames (BeagleCam)
          • gst-launch v4l2src num-buffers=1 ! video/x-raw-yuv,width=640,height=480,framerate=30/1 ! ffmpegcolorspace ! jpegenc ! filesink location=$(date +"%s").jpg
      • 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.
      • 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=false
          • 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 ! \
            mpegvideoparse ! decodebin ! videoscale ! video/x-raw, width=320, height=320 ! videoconvert ! omxh264enc ! h264parse ! \
            mpegtsmux ! udpsink host=192.168.0.8 port=5004 sync=false
          • 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 ! 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
      • 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=false
            • 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 ! mux. \
              mpegtsmux name=mux ! udpsink host=192.168.0.8 port=5004 sync=false
            • 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.
          • 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=false
            • gst-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=false
            • 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.
          • 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
                • 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()


                  m=<caps.media> <media.port> <media.proto> <caps.payload>
                  m=video 5004 RTP/AVP 96

                  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)
                    ...
            • webrtcbin
            • Generating a SDP file from a streaming pipeline
            • caps to SDP (README)
          • 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/bash input_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/bash sdp_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=${audio_rtp_port} sync=true
            gst-launch-1.0 \
                udpsrc address=${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=true
            RTP 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=${audio_rtp_port} sync=true
            gst-launch-1.0 \
                rtpbin name=bin \
                udpsrc address=${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=${address} port=${audio_rtp_port} do-timestamp=true ! "$AUDIOCAPS" ! bin.recv_rtp_sink_1 \
                bin. !
            "application/x-rtp, media=(string)audio" ! queue ! decodebin ! autoaudiosink sync=true
            RTP+RTCP using rtpbin
            gst-launch-1.0 \
                rtpbin name=bin \
                filesrc location=${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_1
            RTP+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.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 \
              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
          • 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
          • RTMPS (e.g. Facebook)
          • 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
      • 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
    module submodule
    GitLab C Reference Manual
    PyGObject API reference
    symbol mapping (C - Python)
    CLI tool
    gtk


    GLib
    Symbol Mapping

    gstreamer gstreamer gstreamer
    GStreamer 1.0 Core Reference Manual

    Gst
    Symbol Mapping
    gst-launch-1.0
    GstBase

    GstCheck

    GstController

    GstNet


    gstreamer-libs
    gstreamer / libs
    GStreamer 1.0 Library Reference Manual



    gstreamer-plugins gstreamer / plugins
    GStreamer Core Plugins 1.0 Plugins Reference Manual



    gst-plugins-base gst-plugins-base-libs gst-plugins-base /  gst-libs GStreamer Base Plugins 1.0 Library Reference Manual
    GstAllocators


    GstApp


    GstAudio


    GstPbutils

    GstRtp

    GstRtsp

    GstSdp Symbol Mapping
    GstTag

    GstVideo

    GstGL

    gst-plugins-base-plugins gst-plugins-base / gst
    GStreamer Base Plugins 1.0 Plugins Reference Manual



    gst-plugins-good gst-plugins-good gst-plugins-good GStreamer Good Plugins 1.0 Plugins Reference Manual



    gst-plugins-ugly gst-plugins-ugly gst-plugins-ugly GStreamer Ugly Plugins 1.0 Plugins Reference Manual



    gst-plugins-bad gst-plugins-bad-libs gst-plugins-bad / gst-libs
    GStreamer Bad Plugins 1.0 Library Reference Manual GstInsertBin

    GstMpegts

    GstPlayer

    GstWebRTC

    gst-plugins-bad-pugins gst-plugins-bad GStreamer Bad Plugins 1.0 Plugins Reference Manual



    gst-python





    gst-rtsp-server

    GStreamer RTSP Server Reference Manual GstRtspServer

    gst-validate


    GstValidate Reference Manual


    gst-validate-transcoding-1.0
    gst-editing-services
    gstreamer / gst-editing-services
    pitivi / gst-editing-services (mirror)
    GStreamer Editing Services 1.12.2 Reference Manual GES
    ges-launch-1.0
    pitivi (developer) gst-transcoding
    pitivi / gst-transcoding

    GstTranscoder

    gst-transcoder-1.0
  • Llenguatges / Languages
  • 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 callback source.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)
    • 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, GLib
          gst_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)
        • 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_READY Gst.State.READY the element is ready to go to PAUSED
          PAUSED
          GST_STATE_PAUSED Gst.State.PAUSED the element is PAUSED, it is ready to accept and process data. Sink elements however only accept one buffer and then block
          PLAYING
          GST_STATE_PLAYING Gst.State.PLAYING the element is PLAYING, the clock is running and the data is flowing
      • 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
                   pass
          get 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()

            def
            on_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_EOS if (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 == Gst.MessageType.STATE_CHANGED):
                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_DONE if (msg.type == Gst.MessageType.ASYNC_DONE):
                ...
            Buffering
            message::buffering
            GST_MESSAGE_BUFFERING
            gst_message_parse_buffering (message, &percent);
            if (msg.type == Gst.MessageType.BUFFERING
                 percent = message.parse_buffering()

            sync-message

            Element
            (specific to element; e.g. queue signals)
            sync-message::element
            if (msg.type == Gst.MessageType.ELEMENT):
                ...
            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 == Gst.MessageType.STREAM_STATUS):
                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);
          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)
      • 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
    • 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 (...)
      • Metadata
        • Types
          • stream tags: non-technical information (author, title, album ...)
          • stream-info: technical information (GstPad, GstCaps)

        • C
          Python
          Metadata reading

          tags.py
          Tag writing


      • 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

          GstControlSource Gst.ControlSource
          create
          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
        • 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
        • When would you force a thread?
      • Autoplugging
      • 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
    • Higher-level interfaces for GStreamer applications
      • Playback Components
    • Appendices
    • ...
  • 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-tutorials
    GstreamerCodeSnippets
    Others/0.10/Python/pygst-sdk-tutorials
    GstreamerCodeSnippets
    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 (buggy)
    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)
    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
    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



    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);
    - 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
    Threads
    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)



    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);
    Playback tutorial 3: Short-cutting the pipeline
    - basic-tutorial-8.py


    Discoverer
    Basic tutorial 9: Media information gathering
    • GstDiscoverer
    - basic-tutorial-9.py




    gst-discoverer-1.0
    gst-launch-1.0
    Tools
    Basic tutorial 10: GStreamer tools






    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





    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
      • capsfilter
      • typefind
    • Debugging
    -





    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
    • 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:
      • 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]
              ...
        • /usr/share/pitivi/gstpresets/*.prs,
        • ~/.local/share/gstreamer-1.0/presets/*.prs
        • GST_PRESET_PATH
      • ~/.local/share/gstreamer-1.0/presets/GstX264Enc.prs
        • [_presets_]
          version=0.10
          element-name=GstX264Enc

          [slow12mbps]
          speed-preset=slow
          bitrate=12288


          [slow700kbps]
          speed-preset=slow
          bitrate=700
      • <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")
  • gst-editing-services
    • GStreamer Editing Services (API Reference)
    • Exemples / Examples
    • 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
                • ...
            • 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
                • ...


      • 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)
        init GES



        # 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



        # create timeline with one audio track and one video track
        #timeline = GES.Timeline.new_audio_video() #
        VideoTrack is 1280*720timeline = GES.Timeline.new()
        video_track = GES.VideoTrack.new()
        # if update_restriction_caps is not called, VideoTrack is 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)
  • gst-transcoding
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