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• Merge commit ’715f139c9bd407ef7f4d1f564ad683140ec61e6d’

  23 mars 2017, par Clément Bœsch

  Merge commit ’715f139c9bd407ef7f4d1f564ad683140ec61e6d’
  * commit ’715f139c9bd407ef7f4d1f564ad683140ec61e6d’ : (23 commits)
  
    vp9lpf/x86 : make filter_16_h work on 32-bit.
  
    vp9lpf/x86 : make filter_48/84/88_h work on 32-bit.
  
    vp9lpf/x86 : make filter_44_h work on 32-bit.
  
    vp9lpf/x86 : make filter_16_v work on 32-bit.
  
    vp9lpf/x86 : make filter_48/84_v work on 32-bit.
  
    vp9lpf/x86 : make filter_88_v work on 32-bit.
  
    vp9lpf/x86 : make filter_44_v work on 32-bit.
  
    vp9lpf/x86 : save one register in SIGN_ADD/SUB.
  
    vp9lpf/x86 : store unpacked intermediates for filter6/14 on stack.
  
    vp9lpf/x86 : move variable assigned inside macro branch.
  
    vp9lpf/x86 : simplify ABSSUM_CMP by inverting the comparison meaning.
  
    vp9lpf/x86 : remove unused register from ABSSUB_CMP macro.
  
    vp9lpf/x86 : slightly simplify 44/48/84/88 h stores.
  
    vp9lpf/x86 : make cglobal statement more conservative in register allocation.
  
    vp9lpf/x86 : save one register in loopfilter surface coverage.
  
    vp9lpf/x86 : add ff_vp9_loop_filter_[vh]_44_16_sse2,ssse3,avx.
  
    vp9lpf/x86 : add ff_vp9_loop_filter_h_48,84_16_sse2,ssse3,avx().
  
    vp9lpf/x86 : add an SSE2 version of vp9_loop_filter_[vh]_88_16
  
    vp9lpf/x86 : add ff_vp9_loop_filter_[vh]_88_16_ssse3,avx.
  
    vp9lpf/x86 : add ff_vp9_loop_filter_[vh]_16_16_sse2().
  
    ...
  All these commits are cherry-picks from FFmpeg. Maybe some slight
  
  differences sneaked in but the Libav codebase still differs too much
  
  with our own to make a proper diff. This merge is a noop.
  Merged-by : Clément Bœsch <u@pkh.me>
  

• setting bit rates in creating video from images in ffmpeg not working

  2 mai 2014, par mast kalandar

  I have a HQ video of one second

  Some information of this video is as below

  Dimensions : 1920 x 1080    
  
  Codec : H.264    
  
  Framerate : 30 frames per second    
  
  Size : 684.7 kB (6,84,673 bytes)
  
  Bitrates : 5458 kbps

  I have extracted frames from video

  ffmpeg -i f1.mp4 f%d.jpg

  All images are of 1920 x 1020 pixels by default 30 frames are generated (f7_1.jpg, f7_2.jpg,.....,f7_30.jpg)

  I have added some texts and objects to these images (without changing dimensions of any image, all 30 images are still of 1920 x 1020 pixels)

  Now I am trying to merge all these images to create single video (of 1 second)

  I referred this official document, I have run below command

  ffmpeg -f image2 -i f7_%d.jpg -r 30 -b:v 5458k foo_5458_2.mp4

  Video created is also of one second, thing is its bit rates are higher then the original one. New video has 6091 kbps bit rates, while I expect are 5458 kbps only.

  Because of higher bits, its gets finish very quickly compare to original video in video player.

  Is there any thing I missing ??

  And I don’t know what is exact meaning and job of -f image2 option, when I run command without this option, I am getting same video.

• Convert encoded audio file to text with signal values

  15 août 2017, par Niccolò Cirone

  I’ve been programming in c for the first time with audio files. I found this code which supposedly should read an audio file and then write a csv file containing several info in order to analyse the audio waves,that in case will be a simple voice : i’m interested in amplitude of the waves, in the timbre of the voice and its hight and extension.

             main () {   
  
             // Create a 20 ms audio buffer (assuming Fs = 44.1 kHz)
  
             int16_t buf[N] = {0}; // buffer
  
             int n;                // buffer index
  
  
  
            // Open WAV file with FFmpeg and read raw samples via the pipe.
  
            FILE *pipein;
  
            pipein = popen("ffmpeg -i whistle.wav -f s16le -ac 1 -", "r");
  
            fread(buf, 2, N, pipein);
  
            pclose(pipein);
  
  
  
            // Print the sample values in the buffer to a CSV file
  
            FILE *csvfile;
  
            csvfile = fopen("samples.csv", "w");
  
            for (n=0 ; n<n></n>code>

  Could someone explain me in detail how can I read an audio file so that I could extract from it the info I need ? Referring to this code, could someone explain me the meaning of the pipe at line 8

  pipein = popen("ffmpeg -i whistle.wav -f s16le -ac 1 -", "r");

  p.s. I already know how to read the header of the audio file, which contains a lot of useful info, but I also want to analyse the entire audio file, sample by sample.