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• How to save AVPacket if I have input information from online camera

  31 mars 2020, par Orest

  I am new to libav.
I have a online video camera and want save video from archive to the video file with libav

  



  Camera provides such data

  



  uint32_t frameType, // I frame or P frame

void *frame, //pointer to the frame

size_t frameSize, //size of the frame in bytes

uint64_t timeStamp, //time stamp in time_t units

uint32_t width, //frame width

uint32_t height, //frame heigh

uint32_t genTime, //I do not now what is this. allways 0

const char *encodingType //H264 or H265


  



  I tried this

  



  void writeHeader(){
mOutputFilePath = outputFilePath;
    int ret = 0;
    avformat_alloc_output_context2(&output_format_context, nullptr, nullptr, outputFilePath.c_str());

AVStream *out_stream;
        out_stream = avformat_new_stream(output_format_context, nullptr);

        out_stream->discard = AVDISCARD_DEFAULT;//не змінювати
        out_stream->codecpar->level = 42;//не змінювати
        out_stream->codecpar->profile = FF_PROFILE_H264_HIGH;//не змінювати
        out_stream->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;

        if(codecID == "H264") out_stream->codecpar->codec_id = AV_CODEC_ID_H264;
        else if(codecID == "H265") out_stream->codecpar->codec_id = AV_CODEC_ID_H265;

        out_stream->codecpar->format = AV_PIX_FMT_YUV420P;
        out_stream->codecpar->height = heightFrame;
        out_stream->codecpar->width = widthFrame;
        //        out_stream->codecpar->bit_rate = 2478235;
        //        out_stream->codecpar->bits_per_coded_sample = 24;
        //        out_stream->codecpar->bits_per_raw_sample = 8;
        out_stream->codecpar->sample_aspect_ratio.num = 0;
        out_stream->codecpar->sample_aspect_ratio.den = 1;
        out_stream->codecpar->color_primaries = AVCOL_PRI_UNSPECIFIED;//не змінювати

avio_open(&output_format_context->pb, mOutputFilePath.c_str(), AVIO_FLAG_WRITE);
avformat_write_header(output_format_context, &opt);
}

void writePacket(){
 AVPacket inputPacket;
        av_init_packet(&inputPacket);
        inputPacket.buf = NULL;
        inputPacket.pts = (int)timeStamp;
        inputPacket.dts = inputPacket.pts; 
        inputPacket.data = (unsigned char*)frame;
        inputPacket.size = (int)frameSize;

        if (frameType == KP2P_FRAME_TYPE_IFRAME)
        {
            inputPacket.flags = AV_PKT_FLAG_KEY;
        }
        inputPacket.duration = 0;
        inputPacket.pos = -1;
av_interleaved_write_frame(output_format_context, &inputPacket);
    av_packet_unref(&inputPacket);
}

void closeFile()
{
av_write_trailer(output_format_context);
    if (output_format_context && !(output_format_context->oformat->flags & AVFMT_NOFILE))
        avio_closep(&output_format_context->pb);
    avformat_free_context(output_format_context);
}


  



  in output file I have black vindow and time is not correct (input 30 seconds in out 2 seconds)
What am I doing wrong ?

  


• How to convert rtp codec in online to amr-nb/wb (octet-align = 0) by FFMPEG ?

  28 janvier 2019, par Mehdi Talebi

  rtp is received as input with different payload types such as amr-nb and WAVE.It is necessary to convert all of them to amr-wb but in bandwidth-align (NOT octet-align).
  I use this command to convert :

  ffmpeg -re -i rtp ://127.0.0.1:1235 -ar 16000 -ab 19.85k -acodec libvo_amrbenc -f rtp rtp rtp ://127.0.0.1:1236

  But the out put (SDP) is like this :

  1. v=0
  2. o=- 0 0 IN IP4 127.0.0.1
  3. s=No Name
  4. c=IN IP$ 127.0.0.1
  5. t=0.0
  6. a=tool:libvaformat 58.25.100
  7. m=audio 1236 RTP/AVP 97
  8. b=AS:19
  9. a=rtpmap:97 AMR-WB/16000/1
  10. a=fmtp:97 octet-align =1

  My problem is octet-align =1, i need octet-align = 0. Do you know any argument to set this in the command ?

• How to improve my current method of generating an audio transcript from an online video

  23 janvier 2019, par 5areductase

  I am working on a project that needs to download and analyze online videos. Most of the videos are hosted on YouTube but some are hosted on Facebook.

  My task is to generate an audio transcript for every video.

  My colleague was using a series of programs sequentially given some {link} :

  youtube-dl -f '(mp4)[height = 360][width = 640]' {link} -o '{out_1}.%(ext)s'
  
  
  
  ffmpeg -i {out_1} -vn {out_2}.wav
  
  
  
  sox {out_2} {out_3} channels 1 rate 16000
  
  
  
  pocketsphinx_continuous -infile {out_3} -samprate 16000 -hmm {ACOUSTICMODEL} -dict {DICTIONARY} -lm {LANGMODEL} -fwdflat yes -bestpath yes 2> error.log | tee {out_4}.log && rm error.log

  Note that there’s an extra ffmpeg step to extract the audio, instead of simply directly downloading it with youtube-dl because video is needed as well.

  Everything works correctly as far as I can tell, but I’ve never really dealt with audio before so I’m not sure if this is the best way to go about it.