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• OpenCV FFMPEG RTSP Camera Feed Errors

  4 avril, par trn2020

  I'm getting these errors at random times when saving frames from an rtsp camera feed. The errors happen at different times, usually after 100-200 images have been saved, and the errors themselves are not always exactly the same. They cause the images that are saved at the time of the error to be distorted either to the point of being completely grey or contain distorted pixels.

  


  #Frame_142 - [hevc @ 0c3bf800] The cu_qp_delta 29 is outside the valid range [-26, 25].

  


  #Frame_406 - [hevc @ 0b6bdb80] Could not find ref with POC 41

  


  I've tried implementing the code in both python and c++ with the same result. Also tried saving as .png instead of .jpg. The rtsp feed works fine when using imshow to display the camera, the problem only appears to happen when trying to save the frames. From what I can gather the errors have to do with ffmpeg but google isn't much help for these types of errors.

  


  #include <iostream>&#xA;#include &#xA;#include <chrono>&#xA;#include <thread>&#xA;&#xA;using namespace std;&#xA;using namespace cv;&#xA;&#xA;int main() {&#xA;&#xA;    VideoCapture cap("rtsp://admin:admin@192.168.88.97/media/video1");&#xA;    if (!cap.isOpened())&#xA;        return -1;&#xA;&#xA;    for (int i = 0; i &lt; 500; i&#x2B;&#x2B;)&#xA;    {&#xA;        Mat frame;&#xA;        cap >> frame;&#xA;        imwrite("C:\\Users\\Documents\\Dev\\c&#x2B;&#x2B;\\OpenCVExample\\frames\\frame" &#x2B; std::to_string(i) &#x2B; ".png", frame);&#xA;        cout &lt;&lt; i &lt;&lt; "\n";&#xA;        std::this_thread::sleep_for(std::chrono::milliseconds(10));&#xA;&#xA;    }&#xA;&#xA;    return 0;&#xA;}&#xA;</thread></chrono></iostream>

  


• How do you get exactly equal segments when splitting an audio file using ffmpeg ? [duplicate]

  13 septembre 2022, par dev404

  I'm trying to split a WAV audio file into several 3 second segments.

  


  I tried the following, but I get files with irregular sizes that are slightly above or under 3 seconds :

  


  ffmpeg -i input.wav -c:a libvorbis -segment_time 3 -f segment output%d.ogg


  


  My files look like this :

  


  Duration: 00:00:03.01, start: 0.000000, bitrate: 125 kb/s
Duration: 00:00:03.01, start: 3.008435, bitrate: 132 kb/s
Duration: 00:00:02.99, start: 6.021224, bitrate: 129 kb/s
Duration: 00:00:02.99, start: 9.010794, bitrate: 130 kb/s
Duration: 00:00:03.01, start: 12.000363, bitrate: 128 kb/s
Duration: 00:00:03.00, start: 15.005896, bitrate: 134 kb/s
Duration: 00:00:03.02, start: 18.005624, bitrate: 135 kb/s
Duration: 00:00:02.98, start: 21.031474, bitrate: 131 kb/s


  


  Unfortunately the difference is significant enough that when I later try to chain them together, the gaps between some of them are clearly perceivable.

  


  I found these other two questions that mention similar problems, but they're converting video files not audio.

  


  ffmpeg not splitting into EXACT same length chunks

  


  How to get equal segment times using ffmpeg ?

  


  Is there something that could be used for audio files instead ?

  


• Batch mixing audio, given timestamps. Multiple offsets, only two sounds. How to do it efficiently ?

  3 août 2021, par Evil

  I have two stereo sounds, 1.wav and 2.wav, these sounds are less than 1 second long and list of timestamps (miliseconds from start of recording). Recording of pure video (recording.mp4) is several hours long and there are thousands (20 000 - 30 000) of timestamps per sounds.

  


  I want to convert list of timestamps and sounds into one recording, merging it with video. The part of merging audio with video is easy with ffmpeg, so this is not part of the question.

  


  The list of timestamps is tsv, for example :

  


  


  1201\t1.wav
  
1501\t2.wav
1603\t1.wav
  
and so on, up to 50 000

  


  


  I can convert it to anything, I am generating this file.

  


  I have seen mixing sound with padding and mixing audio to existing video, but I have to batch process a lots of samples, running sox that many times is not feasible. Mere constructing input for ffmpeg or sox is a cumbersome task.

  


  


  sox -M f2.wav f3.wav f1.wav out.wav delay 4 4 8 8 remix 1,3,5 2,4,6
  
(assuming stereo), or

  


  


  


  sox -m f1.wav "|sox f2.wav -p pad 4" "|sox f3.wav -p pad 8" out.wav

  


  


  Cool for three files. Not feasible for 50 000+. First one needs to read file multiple times (even if it is the same one) and remix channels. Second executes 50 000 sox invocations, also reading the same two files (1.wav, 2.wav) over and over.

  


  I do not use any effects on sounds. There is no explicit support in sox to take one input and play it multiple times (echo / echos destroys the material). Also creating padding or delay takes a lot of time. FFMPEG also needs long query to make it happen.

  


  Since muxing two files is easy, I have tried to record two sounds separately, but still it takes a lot of time to process.

  


  Is there simpler / faster way ?

  


  Taking advice from fdcpp, since wav is PCM coded I also consider writing C program to parse it. I will update code, when I am done.
  
This extends question : is there way to encode offsets in wav format ?