Sur d’autres sites (6646)

• Convert HEVC/H.265 video to a more widely supported format using fluent-ffmpeg in Node.js

  21 juin 2021, par Daniel Loiterton

  My app allows users to upload videos from their phones. However, the latest iPhone default format (HEVC / H.265) does not seem to be supported by Chrome and other browsers, so some users cannot stream the videos.

  


  I'm already using fluent-ffmpeg on my Node.js backend to read metadata and extract thumbnails from the uploaded videos. How would I go about converting HEVC videos to something else ? What format/options would be appropriate for streaming to browsers ? Ideally I'd like to keep files fairly small, and I'm willing to sacrifice some quality to that end.

  


• FFMpeg - Segment audio to chunks

  8 mars 2015, par fr3sh

  I have tried this example in order to segment a given video file using ffmpeg into an m3u8 file and smaller chunks (.ts files). This actually worked great. Is it possible to do practically the same thing with audio input ?
  This was my most promising approach so far (capturing live audio on Windows OS) :

  ffmpeg -f dshow -i audio="<name of="of" input="input" device="device">" -acodec libmp3lame -ab 64000 | segmenter - 10 stream stream.m3u8 http:///stream/stream/ 5 1
  
  </name>

  But this returns this error :

  At least one output file must be specified.
  
  Could not open input file, make sure it is an mpegts file: -1

  I really would not know how to convert the live audio stream to an mpegts file.
  Could anyone please give me a hint ?

  Thanks a lot

• Muxing in audio to gstreamer RTMP stream kills both video and Audio

  1er avril 2015, par Adam

  I need some genius help here - I’m trying to set up a live stream for my upcoming wedding... and I have it ALMOST working - audio seems to be the problem.

  This is my setup

  • Raspberry Pi Model B+
  • Logitech C920 (with onboard h264 encoding that I am utilising)
  • on-camera (C920) microphone
  • USB wifi to iPhone 4G connection
  • gstreamer1.0
  • Amazon EC2 Wowza RTMP server

  I have it all set up, but as soon as I mux in the audio, the streams wont play by any player.

  What Works :
  - my gstreamer pipeline WITHOUT the audio muxed in
  - Wowza receives a consistent stream, no failures
  - The various Flash players / iOS / Android and VLC all play back the video

  What doesnt :
  - enabling audio in the mux (using the pipeline below)
  - BUT gstreamer doesnt complain
  - BUT Wowza receives a consistent stream, no failures
  - The various flash players fail to play both Audio and Video. some just display the first video frame
  - VLC plays 1 video frame, and about 100ms of audio, then stops

  Ideally I’d like the muxed audio/video FLV stored on the SD card too in case the network goes down - but if the ’tee’ needs to be sacrificed to make it work, so be it.

  This is my current FAILING pipeline - I assume there’s something really stupid in it because I know practically nothing about gstreamer.... The first frame loads in all the players (except iOS.. which never shows anything)

  # set camera resolution to 720p, and the data format to H264 (alternatives are YUV and JPG)
  
  v4l2-ctl --device=/dev/video0 --set-fmt-video=width=1280,height=720,pixelformat=1
  
  # set the frame rate
  
  v4l2-ctl --device=/dev/video0 --set-parm=10
  
  
  
  gst-launch-1.0 -v -e uvch264src initial-bitrate=300000 average-bitrate=300000 device=/dev/video0 name=src auto-start=true src.vidsrc \
  
                  ! queue \
  
                  ! video/x-h264,width=1280,height=720,framerate=10/1 \
  
                  ! h264parse \
  
                  ! flvmux streamable=true name=mux \
  
                  ! queue \
  
                  ! tee name=t \
  
                  ! queue \
  
                  ! filesink location=/home/pi/wedding.flv t. \
  
                  ! queue \
  
                  ! rtmpsink location='rtmp://wowzaserver/live/wedding live=1' >>/home/pi/wedding.log 2>&1

  Some of the things I can’t really afford to change at this late stage are the encapsulation (FLV) and wowza RTMP because I’ve built everything around that...

  Please Help !! Thanks !

  UPDATE

  Given that I am also saving the FLV file, I have found that if I use ffmpeg to send that FLV file (using audio copy, video copy) to the RTMP server, everything works (but obviously its not live) ! So I am now starting to believe this is a problem with the way Gstreamer encapsulates RTMP - and by putting ffmpeg in the middle it fixes it... but it’s not live of course.
  Is it possible to pipe my output to ffmpeg and using ffmpeg’s RTMP ?