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• ffserver streaming h.264 raw data

  11 novembre 2014, par Samuel David Hudson

  I have a c++/openGL application which generate a .h264 stream using a hardware encoder (nvidia grid). Im using ffserver to serve the stream to be consumed via a url.

  The application writes out the stream data to a file on disk, which gets added to as time goes by.

  ffserver.conf

  # Port on which the server is listening. You must select a different
  
  # port from your standard HTTP web server if it is running on the same
  
  # computer.
  
  HttpPort 8090
  
  
  
  <stream>
  
      File "/home/ec2-user/demo/movie.h264"
  
      NoAudio
  
      Format mpegts
  
  </stream>

  The problem i am having is that whenever you view the stream, it starts from the begining of the data, rather than from ’now’. I want the stream to be as realtime as possible.

  Does anyknow know how i might achieve this ?

• How can i use live data from an api to livestream using FFMPEG

  17 août 2022, par Steve Mimshak

  I would like to use live data from an api in a live stream video in real time how i did not find any information about how this is possible.

  


  I want to use an ubuntu server with FFMPEG to create the livestream.

  


  Does anyone know how to do it.

  


• Encoding audio data using ffmpeg

  30 mai 2019, par Matin Kh

  I am receiving a byte array (int8_t*) and I would like to use FFMPEG to encode it into FLAC. All the examples I found are reading data from files, which is not the case for me. Following the original documents (see here), I came up with the following solution :

  #include "libavcodec/avcodec.h"
  
  
  
  // ...
  
  
  
  // params:
  
  //  audioData: original audio data
  
  //  len: length of the byte array (audio data)
  
  //  sampleRate: sample rate of the original audio data
  
  //  frameSize: frame size of the original data
  
  uint8_t* encodeToFlac(uint8_t* audioData, int len, int sampleRate, int frameSize) {
  
    uint8_t* convertedAudioData;
  
  
  
    // Context information
  
    AVCodecContext* context = avcodec_alloc_context();
  
    context->bit_rate = 64000;
  
    context->sample_rate = sampleRate;
  
    context->channels = 2;
  
    context->frame_size = frameSize;
  
  
  
    short* samples = malloc(frameSize * 2 * context->channels);
  
    int outAudioDataSize = len * 2;
  
    convertedAudioData = malloc(outAudioDataSize);
  
    int outSize = avcodec_encode_audio(c, convertedAudioData, outAudioDataSize, samples);
  
  
  
    return convertedAudioData;
  
  }

  I have two main issues with the above solution :

  1. I did not specify what the final encoding should be (for example, MP3, FLAC, etc), which makes me wonder if I’m using FFMPEG library correctly ?

  2. Do I have all the necessary information about the source - original audio data ? I am not certain if I have all the necessary information to perform the encoding.