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• merging 2 editing videos without saving them using ffmpeg

  14 avril 2021, par Eswar T

  I want to combine 2 videos assume even they are edited among themselves

  


  How normally we do

  


  video 1 :

  


  ffmpeg -i 1.mp4 -filter:a "volume=0.0" test1.mp4


  


  video 2 :

  


  ffmpeg -i 2.mp4 -filter:a "volume=10.0" test2.mp4  


  


  now I can combine them using

  


  ffmpeg -i test1.mp4 -i test2.mp4 -filter_complex [0:v:0][0:a:0][1:v:0][1:a:0]concat=n=2:v=1:a=1[outv][outa] -map [outv] -map [outa] out.put.mp4


  


  So my question is, Is there a way to make this 3 steps process into 1 step and without saving files of step 1 and step 2

  


  I do know that we can combine into one using && but my main query is there a way to do without saving the files of video1 and video2 that edited files

  


  Hope I'm a bit clear with my query

  


  Question edited/added :

  


  ffmpeg -i test.mp4 -filter:a "volume=8.0,atempo=4.0" -vf "transpose=2,transpose=2,setpts=1/4*PTS" -s 640x480 test.mkv 


  


  can we do all these options also in the merge command(operations like change video Speed, resolution, rotation, framerate, and trim) ?

  


• 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.

  


• Revision c8ed36432e : Non-uniform quantization experiment This framework allows lower quantization bi

  4 mars 2015, par Deb Mukherjee

  Changed Paths :
   Modify /configure

  
   Modify /vp9/common/vp9_blockd.h

  
   Modify /vp9/common/vp9_onyxc_int.h

  
   Modify /vp9/common/vp9_quant_common.c

  
   Modify /vp9/common/vp9_quant_common.h

  
   Modify /vp9/common/vp9_rtcd_defs.pl

  
   Modify /vp9/decoder/vp9_decodeframe.c

  
   Modify /vp9/decoder/vp9_detokenize.c

  
   Modify /vp9/encoder/vp9_block.h

  
   Modify /vp9/encoder/vp9_encodemb.c

  
   Modify /vp9/encoder/vp9_encodemb.h

  
   Modify /vp9/encoder/vp9_quantize.c

  
   Modify /vp9/encoder/vp9_quantize.h

  
   Modify /vp9/encoder/vp9_rdopt.c

  
  
  Non-uniform quantization experiment

  This framework allows lower quantization bins to be shrunk down or
  expanded to match closer the source distribution (assuming a generalized
  gaussian-like central peaky model for the coefficients) in an
  entropy-constrained sense. Specifically, the width of the bins 0-4 are
  modified as a factor of the nominal quantization step size and from 5
  onwards all bins become the same as the nominal quantization step size.
  Further, different bin width profiles as well as reconstruction values
  can be used based on the coefficient band as well as the quantization step
  size divided into 5 ranges.

  A small gain currently on derflr of about 0.16% is observed with the
  same paraemters for all q values.
  Optimizing the parameters based on qstep value is left as a TODO for now.

  Results on derflr with all expts on is +6.08% (up from 5.88%).

  Experiments are in progress to tune the parameters for different
  coefficient bands and quantization step ranges.

  Change-Id : I88429d8cb0777021bfbb689ef69b764eafb3a1de