Webrtc-H264Capturer

Workarounds to use external H.264 video encoders in WebRTC Native C++ source code.
The main purpose of this project is to allow using different kinds of video sources with WebRTC. Hence, Webrtc-H264Capturer makes it possible:

• To use any video sources (not only webcams).
• To use any H.264 encoder (not only OpenH264), especially hardware implementations.

The proposed approach does not require transcoding => low CPU usage. A fake YUV capturer is used to write the H.264 elementary stream. Then, H.264 encoding (OpenH264) is bypassed and the elementary stream is directly packetized.

A web client is also provided in order to perform video streaming between WebRTC Native C++ and a web browser.

Useful references:
https://github.com/mpromonet/webrtc-streamer
https://github.com/zhanghuicuc/WebRTC-VideoEngine-Demo

Environment

Tested with Ubuntu 14.04.3, Kernel 3.16.0-59-generic.
Prerequisites: git, curl, python, "build-essential"...

Build

Get WebRTC (https://webrtc.org/native-code/development/):

mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync --with_branch_heads
export WEBRTC_DIR=`pwd`

Checkout the revision before patching:

cd $WEBRTC_DIR/src
git checkout branch-heads/62
python setup_links.py --force
gclient sync

If needed (see terminal outputs):

python setup_links.py --force
gclient sync

Apply the patch and generate ninja files:

cd $WEBRTC_DIR/src
patch -p1 < ../../webrtc_patch.diff
gn gen out/Default --args='rtc_use_h264=true is_component_ffmpeg=true rtc_include_tests=false  rtc_libvpx_build_vp9=false  rtc_build_opus=false rtc_include_opus=false rtc_build_openmax_dl=false rtc_use_openmax_dl=false rtc_use_gtk=false'

Third Party Libraries:

• FFmpeg

   cd $WEBRTC_DIR/src/third_party/ffmpeg
   mkdir build
   ./configure --enable-shared --disable-programs --disable-doc --prefix=$WEBRTC_DIR/src/third_party/ffmpeg/build --sysroot=$WEBRTC_DIR/src/build/linux/debian_jessie_amd64-sysroot --cc=$WEBRTC_DIR/src/third_party/llvm-build/Release+Asserts/bin/clang --ld=$WEBRTC_DIR/src/third_party/llvm-build/Release+Asserts/bin/clang
   make && make install
   cp $WEBRTC_DIR/src/third_party/ffmpeg/build/lib/*   $WEBRTC_DIR/src/out/Default
  

• Add shared libraries linking in Ninja files

   cd $WEBRTC_DIR/src
   grep -rl "solibs" ./out --include="peerconnection_client.ninja" | xargs sed -i '/solibs/s|./libffmpeg.so|./libavcodec.so ./libavformat.so ./libavutil.so ./libswscale.so|g'
  

H.264 capturer support:

cp H264_capturer/* $WEBRTC_DIR/src/webrtc/examples/peerconnection/client   

H.264 external encoding support is enabled by including h264videocapturer.h in conductor.cc (see webrtc_patch.diff).

cd $WEBRTC_DIR/src
find out/ -name "peerconnection_client.ninja" | xargs sed -i 's|include_dirs =|include_dirs = -I../../third_party/ffmpeg/build/include |g'

Build peerconnection_server/client:

ninja -C out/Default peerconnection_server
ninja -C out/Default peerconnection_client

Usage

Launch the server:

$WEBRTC_DIR/src/out/Default/peerconnection_server

Launch the client with the external H.264 stream URL, e.g.:

$WEBRTC_DIR/src/out/Default/peerconnection_client  --video_url "rtsp://192.168.123.53/Channel1"

The original GUI (gtk) of peerconnection_client has been removed in order to ease the integration in embeded system (e.g. Raspberry Pi: see Raspberrypi_cross_compile.md ). Therefore the SDP offer made in conductor.cc is sendonly.

Connect with a second client, e.g.:

firefox Web_client/index.html

To allow the peerconnection to be initiated from the web client, firefox must be configured (about:config) with (integer) media.navigator.video.preferred_codec = 126

Examples of H.264 video sources

• File (with H.264 elementary stream)

    $WEBRTC_DIR/src/out/Default/peerconnection_client  --video_url big_buck_bunny_4s.264
  

• Webcam logitech C920 (H.264 HW encoder)+ gstreamer

    gst-launch-1.0 -v -e uvch264src device=/dev/video0 name=src auto-start=true  iframe-period=1000 initial-bitrate=3000000 minimum-bitrate=2000000 maximum-bitrate=2500000 average-bitrate=2000000  src.vidsrc !  'video/x-h264,width=1280,height=720,framerate=30/1' !  queue  !  h264parse  !  rtph264pay ! 'application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96'  ! udpsink host=127.0.0.1 port=1234
    $WEBRTC_DIR/src/out/Default/peerconnection_client  --video_url test.sdp
  

• Webcam with MJPEG support + Raspberry Pi and gstreamer omxh264enc (see Raspberrypi_cross_compile.md to embed the application on the board)

    gst-launch-1.0 v4l2src device=/dev/video0 !  image/jpeg,width=1280,height=720,framerate=30/1  ! jpegdec ! video/x-raw,width=1280,height=720,framerate=30/1  !  queue  ! omxh264enc target-bitrate=2000000 control-rate=variable periodicty-idr=30 interval-intraframes=30 !  video/x-h264, width=1280, height=720, framerate=30/1  ! queue ! h264parse ! rtph264pay  ! udpsink host=127.0.0.1 port=1234
    peerconnection_client  --video_url test.sdp