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• I have a m3u8 file where the individual files don't have any .ts format, Is there a way to cocnat them to a single mp4 file

  6 septembre 2020, par Suhail Hussain

  Here is a snippet of the m3u8 file

  


  #EXTM3U
#EXTINF:1,0
0
#EXTINF:1695,0c9c3bf590e32dcb8c4b83222056838b
0c9c3bf590e32dcb8c4b83222056838b
#EXTINF:1,1
1
#EXTINF:4,2
2
#EXTINF:3,3
3
#EXTINF:4,4
4
#EXTINF:3,5
5
#EXTINF:3,6
6
#EXTINF:4,7
7
#EXTINF:4,8
8
#EXTINF:3,9
9
#EXTINF:4,10
10


  


  This goes on for some 500 files. I am able to open the folder in vlc as a playlist but it is just a collection of 500 files that play one after the another. I checked online and found that ffmpeg can concatenate a m3u8 file to a mp4. That unfortunately did not work. After trying a few different syntaxes that I found on different forums which also did not work, I tried "ffplay" on the file name which once again gave the same error message as before - Invalid data found when processing input:=    0B f=0/0

  


  So this made me believe perhaps ffmpeg is unable to open the file while vlc is able to. Any way to combine these files to a single file is appreciated

  


• How do I use FFMPEG/libav to access the data in individual audio samples ?

  15 octobre 2022, par Breadsnshreds

  The end result is I'm trying to visualise the audio waveform to use in a DAW-like software. So I want to get each sample's value and draw it. With that in mind, I'm currently stumped by trying to gain access to the values stored in each sample. For the time being, I'm just trying to access the value in the first sample - I'll build it into a loop once I have some working code.

  


  I started off by following the code in this example. However, LibAV/FFMPEG has been updated since then, so a lot of the code is deprecated or straight up doesn't work the same anymore.

  


  Based on the example above, I believe the logic is as follows :

  


  

  1. get the formatting info of the audio file

  


  2. get audio stream info from the format

  


  3. check that the codec required for the stream is an audio codec

  


  4. get the codec context (I think this is info about the codec) - This is where it gets kinda confusing for me

  


  5. create an empty packet and frame to use - packets are for holding compressed data and frames are for holding uncompressed data

  


  6. the format reads the first frame from the audio file into our packet

  


  7. pass that packet into the codec context to be decoded

  


  8. pass our frame to the codec context to receive the uncompressed audio data of the first frame

  


  9. create a buffer to hold the values and try allocating samples to it from our frame

  


  


  From debugging my code, I can see that step 7 succeeds and the packet that was empty receives some data. In step 8, the frame doesn't receive any data. This is what I need help with. I get that if I get the frame, assuming a stereo audio file, I should have two samples per frame, so really I just need your help to get uncompressed data into the frame.

  


  I've scoured through the documentation for loads of different classes and I'm pretty sure I'm using the right classes and functions to achieve my goal, but evidently not (I'm also using Qt, so I'm using qDebug throughout, and QString to hold the URL for the audio file as path). So without further ado, here's my code :

  


  // Step 1 - get the formatting info of the audio file
    AVFormatContext* format = avformat_alloc_context();
    if (avformat_open_input(&format, path.toStdString().c_str(), NULL, NULL) != 0) {
        qDebug() << "Could not open file " << path;
        return -1;
    }

// Step 2 - get audio stream info from the format
    if (avformat_find_stream_info(format, NULL) < 0) {
        qDebug() << "Could not retrieve stream info from file " << path;
        return -1;
    }

// Step 3 - check that the codec required for the stream is an audio codec
    int stream_index =- 1;
    for (unsigned int i=0; inb_streams; i++) {
        if (format->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
            stream_index = i;
            break;
        }
    }

    if (stream_index == -1) {
        qDebug() << "Could not retrieve audio stream from file " << path;
        return -1;
    }

// Step 4 -get the codec context
    const AVCodec *codec = avcodec_find_decoder(format->streams[stream_index]->codecpar->codec_id);
    AVCodecContext *codecContext = avcodec_alloc_context3(codec);
    avcodec_open2(codecContext, codec, NULL);

// Step 5 - create an empty packet and frame to use
    AVPacket *packet = av_packet_alloc();
    AVFrame *frame = av_frame_alloc();

// Step 6 - the format reads the first frame from the audio file into our packet
    av_read_frame(format, packet);
// Step 7 - pass that packet into the codec context to be decoded
    avcodec_send_packet(codecContext, packet);
//Step 8 - pass our frame to the codec context to receive the uncompressed audio data of the first frame
    avcodec_receive_frame(codecContext, frame);

// Step 9 - create a buffer to hold the values and try allocating samples to it from our frame
    double *buffer;
    av_samples_alloc((uint8_t**) &buffer, NULL, 1, frame->nb_samples, AV_SAMPLE_FMT_DBL, 0);
    qDebug () << "packet: " << &packet;
    qDebug() << "frame: " <<  frame;
    qDebug () << "buffer: " << buffer;


  


  For the time being, step 9 is incomplete as you can probably tell. But for now, I need help with step 8. Am I missing a step, using the wrong function, wrong class ? Cheers.

  


• FFmpeg batch file - combine individual set files with randomized selection from another set of files

  4 août 2018, par Siampu

  I need to combine a specific set of files with a randomized selection from another set of files ; for more specific context, voice clips followed by a randomized walky-talky beep. At the moment, I’ve managed to assemble this so far from searching around :

  setlocal EnableDelayedExpansion
  
  cd beeps
  
  set n=0
  
  for %%f in (*.*) do (
  
      set /A n+=1
  
      set "file[!n!]=%%f"
  
  )
  
  set /A "rand=(n*%random%)/32768+1"
  
  cd ..
  
  for %%A IN (*.ogg) DO ffmpeg -y -i radio_beep.wav -i "%%A" -i "beeps\!file[%rand%]!" -filter_complex "[0:a:0][1:a:0][2:a:0]concat=n=3:v=0:a=1[outa]" -map "[outa]" "helper\%%A"

  At the moment, this will only run the randomization once and use that selection for every file. How can I have it do the randomization for each .ogg in the folder, and get that into FFmpeg as an input ?