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• Revision debe4e920f : Reduce max_partition_size for low resolutions at speed 5. For speed 5 real-time

  20 juin 2015, par Marco

  Changed Paths :
   Modify /vp9/encoder/vp9_encodeframe.c

  
  
  Reduce max_partition_size for low resolutions at speed 5.

  For speed 5 real-time mode, the selection of the partition size for
  superblocks on the segment (aq-mode=3) uses the non-rd recursive
  pick partition search, and can sometimes select 64x64.

  For low resolutions, visually better to limit this to 32x32.

  Change-Id : I69657a7ed8899f8b3cf8c9c318a2509c5c72c565

• [C++][Linux + ffmpeg + h264 + rtsp + client] and [Window + ffmpeg + play video real time + server]

  21 avril 2015, par QuestionGuy

  I have a problem with ffmpeg and I don’t know how to continue. I have 2 computers :

  Client :

  • Run Ubuntu 14.04
  • FFmpeg installed
  • Use c++ language
  • Features : use ffmpeg to encode h264 video data from webcam of client laptop, then real time streaming to server

  Server :

  • Running Windows 7
  • FFmpeg installed
  • Use c++ language (MFC)
  • Features : Get real time data from client and show it on screen.

  I’ve just connected client to server and they can chat text data together, and I don’t have any idea to make real time video work.

  And my questions are :

  1. [Client] How to get video from webcam on laptop (using ffmpeg code), save it to buffer (raw data), encode it and send to server ?

     I use ffmpeg to get video from wc but it save to file. I really don’t want it. Code is :

     ffmpeg -f v4l2 -framerate 25 -video_size 640x480 -i /dev/video0 output.mkv

  2. [Client] How to get raw data from client, decode it and play it ?

     I have an idea to play it by using directshow in MFC.

• Publish audio to an RTMP server for real time live streaming in C or C++

  19 mai 2021, par Antenaina

  I want to publish audio stream to an RTMP server, for real time audio live streaming, from mobile device (with Android for example).
  
Suppose the mobile device has a way to yield to me those datas in real time (ex : using Oboe library). Packet by packet (a packet contains a certain number of audio frames).
  
When live streaming, there are some really custom computations to those datas that requires that I must send then little by little (packet by packet ?) to the RTMP server.
  
I'm trying to use FFMPEG for that purpose, and have similar problem with this thread's question : How to publish self made stream with ffmpeg and c++ to rtmp server ?. But the answer there is not detailed is not enough for me.
  
I tried reading FFMPEG code source, with the help of the documentation, but there are still some challenges I must face since I'm new to the streaming domain. What I need to know is :

  


  

  • How to properly configure FFMPEG for that purpose ? (AVFormatContext ?)

  


  • What is the proper way to write the stream (AVStream) ? (I read somewhere that the packet needs to be of a specific size, and other stuffs too)

  


  


  For simplicity :

  


  

  • I can handle the audio packet by packet and encoded.

  


  • Audio is encoded as mp3.

  


  • Audio has default sample rate of 44100 Hz, 320kb/s bitrate and some other details already known so that FFMPEG doesn't need to guess it.

  


  


  Further informations :
  
I'm using react-native. For android : native modules to communicate with Java, JNI to communicate Java with C++, Oboe to record and play audio. For iOS : not a problem for the moment.
  
I use node-media-server as RTMP server.
  
Thanks !