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• Download Partial Video via HTTP (for Remote Thumbnailing)

  13 février 2012, par HuntedC

  I have videos hosted on Amazon S3. I encode them with Zencoder and store a thumbnail for the video then using Zencoder. However, I need a way to generate thumbnails at certain points in the video (i.e. 00:00:03, 00:10:32, 01:40:18) and store them either on S3 or my server.

  ffmpeg allows remote thumbnailing, however it takes a very long time (sometimes several minutes) to get a thumbnail from the middle of a file—I believe this is because it downloads the entire file up to that point to get the thumbnail.

  My plan is to somehow download the header of the video file via HTTP byte-range request, guesstimate the byte range where I should be looking for the thumbnail, download about a second of video from that part of the file via HTTP byte-range request, then save the header and tiny video locally. I pull the thumbnail from that using ffmpeg and delete the temporary video.

  I have no idea on how exactly this would work (I believe the H.264 MP4 files I'm working with have a dynamic length header, for another issue). Any suggestions or better ideas ?

  Edit : To clarify, Zencoder thumbnailing is great, but they only allow thumbnail creation in combination with transcoding. I don't want to transcode my video every time I create a new thumbnail, so I need to do this on my own without Zencoder.

• FFmpeg/libav Storing AVPacket data in a file and decoding them again in a different stream - How to prepare and send custom packets in the decoder ?

  21 avril 2022, par Hitokage

  I want to be able to open a stream with a video file and send in the decoder my own packets with data that I previously stored from a different stream with the same codec. So like forging my own packets in the decoder.

  


  My approach is that I encode frames into packets using H.265 and store the data in a file like this :

  


  AVPacket *packet;
std::vector data;
...encoding...
data->insert(data->end(), &packet->data[0], &packet->data[packet->size]);
...storing the buffer in a file...


  


  I also have one mkv video with H.265 stream of the same parameters. Now I want to get the stored packet data in the file, create a new packet, and send it into the decoding process of the mkv file. To make it easier I just copy the parameters of the first packet in the mkv stream into a new packet where I insert my data and send it to the decoder. Is this a right approach ?

  


  ...open decoder with mkv file...
auto packet = av_packet_alloc();
av_read_frame(formatContext, packet);
//here is the packet I will use later with my data
av_packet_copy_props(decodingPacket, packet);
decodingPacket->flags = 0;
//sending the packet from the mkv
avcodec_send_packet(codecContext, packet);
//the data contains the previously stored data
av_packet_from_data(decodingPacket, data.data(), data.size());
avcodec_send_packet(codecContext, decodingPacket);
...retrieving the frame...


  


  However, I am getting this error when I try to send the forget packet :

  


  Assertion buf_size >= 0 failed at libavcodec/bytestream.h:141


  


  I have tried to manually change the decodingPacket->buf->size but that is probably not the problem. I believe that I need to set up some other parameters in the forged packet but what exactly ? Maybe I can also somehow store not only the packet data but the whole structure ? Please let me know, if there is a better way to store and reload a packet and force it into an unrelated decoder with the same parameters.
Thanks !

  


  EDIT : Seems like the buf_size problem was related to my data which were wrongly retrieved. I can confirm that this works now but I am getting a new error :

  


  [hevc @ 0x559b931606c0] Invalid NAL unit size (0 > 414).
[hevc @ 0x559b931606c0] Error splitting the input into NAL units.


  


• samplerate conversion function fails to produce an audible sound but only small pieces of audio

  2 juillet 2014, par user3749290

  playmp3() using libmpg123

  if (isPaused==0 && mpg123_read(mh, buffer, buffer_size, &done) == MPG123_OK)
  
  {
  
      char * resBuffer=&buffer[0]; //22100=0,5s
  
      buffer = resample(resBuffer,22050,22050); // I think the result is 1/2 of audio speed
  
      if((ao_play(dev, (char*)buffer, done)==0)){
  
          return 1;
  
  }

  resample() Using avcodec from ffmpeg

  #define LENGTH_MS 500       // how many milliseconds of speech to store 0,5s:x=1:44100 x=22050 sample to store
  
  #define RATE 44100      // the sampling rate (input)
  
  #define FORMAT PA_SAMPLE_S16NE  // sample size: 8 or 16 bits
  
  #define CHANNELS 2      // 1 = mono 2 = stereo
  
  
  
  struct AVResampleContext* audio_cntx = 0;
  
  //(LENGTH_MS*RATE*16*CHANNELS)/8000
  
  
  
      void resample(char in_buffer[],int out_rate,int nsamples,char out_buffer[])
  
      {
  
          //char out_buffer[ sizeof( in_buffer ) * 4];
  
          audio_cntx = av_resample_init( out_rate, //out rate
  
              RATE, //in rate
  
              16, //filter length
  
              10, //phase count
  
              0, //linear FIR filter
  
              1.0 ); //cutoff frequency
  
          assert( audio_cntx && "Failed to create resampling context!");
  
          int samples_consumed;
  
          //*out_buffer = malloc(sizeof(in_buffer));
  
          int samples_output = av_resample( audio_cntx, //resample context
  
              (short*)out_buffer, //buffout
  
              (short*)in_buffer,  //buffin
  
              &samples_consumed,  //&consumed
  
              nsamples,       //nb_samples
  
              sizeof(out_buffer)/2,//lenout sizeof(out_buffer)/2
  
              0);//is_last
  
          assert( samples_output > 0 && "Error calling av_resample()!" );
  
          av_resample_close( audio_cntx );    
  
      }

  When I run this code, the application part, the problem is that I hear the sound jerky, why ?
  The size of the array I think is right, I calculated considering that in the second half should be 22050 samples from store.