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• Optimize script commands FFMPEG with Limit CPU usage

  30 novembre 2019, par Arsal Imam

  I am working on a audio/video uploading and streaming project.

  Process :

  I have 2 audio files and a video file which I am processing with FFMPEG. I need to process these audio and video files in the below mentioned way :

  1. Generating thumbnail from the video file.

  2. Merging audio files with each other.

  3. then, merging the mixed audio file with the video file.

  4. after that, adding watermark over the video file.

  5. at last, splits video file into small chunks for streaming

  ISSUE :

  Everything is working fine but the issue is that all this process consuming very high CPU and TIME to complete. Is there anyway to optimize the FFMPEG commands to run in fastly. Right now, it is very slow !!

  CODE :

  I have created a bat file to complete the above mentioned process. Below is the source of the batch file which is executing over nodejs server.

  @echo off
  
  
  
  echo "########################################### Uploading Video #############################################"
  
  
  
  :GETOPTS
  
   if /I %~1 == --app-path set APP_PATH=%2& shift
  
   if /I %~1 == --video-name set VIDEO_NAME=%2& shift
  
   if /I %~1 == --do-merge set IS_DO_MERGE=%2& shift
  
   if /I %~1 == --audio-name set AUDIO_NAME=%2& shift
  
   if /I %~1 == --bg-beat-name set BEAT_NAME=%2& shift
  
   shift
  
  if not (%1)==() goto GETOPTS
  
  
  
  set USER_DIR=%APP_PATH%data\content\userContent\
  
  set BEAT_FILE_URI=%APP_PATH%data\content\beats\%BEAT_NAME%
  
  set FFMPEG=%APP_PATH%data\cmd\ffmpeg\bin\ffmpeg.exe
  
  set VIDEO_PATH=%USER_DIR%%VIDEO_NAME%
  
  set MERGED_AUDIO_PATH=%VIDEO_PATH%_mix_audio.aac
  
  set RESULTANT_PATH=%VIDEO_PATH%_finalized.mp4
  
  set WATER_MARKED_PATH=%VIDEO_PATH%_finalized_watermark.mp4
  
  
  
  echo "Taking screenshot - %VIDEO_PATH%"
  
  %FFMPEG% -threads 3 -loglevel panic -y -ss 00:00:01 -i %VIDEO_PATH% -vframes 1 -filter:v scale="500:-1" %VIDEO_PATH%_screenshot.jpg
  
  if "%IS_DO_MERGE%" == "true" (
  
  
  
    echo "Combining audio with music file"
  
    echo "OUTPUT: %MERGED_AUDIO_PATH%"
  
    %FFMPEG% -threads 3 -loglevel panic -y -i %USER_DIR%%AUDIO_NAME% -i %BEAT_FILE_URI% -filter_complex amix=inputs=2:duration=shortest:dropout_transition=3 -shortest %MERGED_AUDIO_PATH%
  
  
  
    echo "Merging Audio & video"
  
    %FFMPEG% -threads 3 -loglevel panic -y -i %VIDEO_PATH% -i %MERGED_AUDIO_PATH% -c:v libx264 -preset ultrafast -crf 22 -tune zerolatency -c:a aac -strict experimental -map 0:v:0 -map 1:a:0 -shortest %RESULTANT_PATH%
  
  
  
    echo "Adding watermark"
  
    %FFMPEG% -threads 3 -loglevel panic -y -i %RESULTANT_PATH% -i %APP_PATH%data\assets\watermark.png -filter_complex "overlay=x=(main_w-overlay_w)/2:y=30" -movflags +faststart %WATER_MARKED_PATH%
  
  
  
    #ffmpeg -i input.mp4 -profile:v baseline -level 3.0 -s 640x360 -start_number 0 -hls_time 10 -hls_list_size 0 -f hls index.m3u8
  
  
  
    #%FFMPEG% -threads 3 -loglevel panic -y -i %WATER_MARKED_PATH% -i %APP_PATH%data\assets\watermark.png -filter_complex "overlay=x=(main_w-overlay_w)/2:y=30" %WATER_MARKED_PATH%
  
  
  
    echo "--------------------------------- process completed with no errors ----------------------------------"
  
  
  
  ) else (
  
    echo "No need to merge as --do-merge is false"
  
  
  
    echo "Adding watermark"
  
    %FFMPEG% -threads 3 -loglevel panic -y -i %VIDEO_PATH% -i %APP_PATH%data\assets\watermark.png -filter_complex "overlay=x=(main_w-overlay_w)/2:y=30" %WATER_MARKED_PATH%
  
  )
  
  echo "######################################  Uploading Video Ended  ##########################################"

• avcodec/put_bits : Make skip_put_bits() less dangerous

  31 juillet 2020, par Andreas Rheinhardt

  avcodec/put_bits : Make skip_put_bits() less dangerous
  Before c63c303a1f2b58677d480505ec93a90f77dd25b5 (the commit which
  
  introduced a typedef for the type of the buffer of a PutBitContext)
  
  skip_put_bits() was as follows :
  static inline void skip_put_bits(PutBitContext *s, int n)
  
  
  
      s->bit_left -= n ;
  
      s->buf_ptr  -= 4 * (s->bit_left >> 5) ;
  
      s->bit_left &= 31 ;
  
  
  If s->bit_left was negative after the first subtraction, then the next
  
  line will divide this by 32 with rounding towards -inf and multiply by
  
  four ; the result will be negative, of course.
  The aforementioned commit changed this to :
  static inline void skip_put_bits(PutBitContext *s, int n)
  
  
  
      s->bit_left -= n ;
  
      s->buf_ptr  -= sizeof(BitBuf) * ((unsigned)s->bit_left / BUF_BITS) ;
  
      s->bit_left &= (BUF_BITS - 1) ;
  
  
  Casting s->bit_left to unsigned meant that the rounding is still towards
  
   inf ; yet the right side is now always positive (it transformed the
  
  arithmetic shift into a logical shift), so that s->buf_ptr will always
  
  be decremented (by about UINT_MAX / 8 unless n is huge) which leads to
  
  segfaults on further usage and is already undefined pointer arithmetic
  
  before that. This can be reproduced with the mpeg4 encoder with the
  
  AV_CODEC_FLAG2_NO_OUTPUT flag set.
  Furthermore, the earlier version as well as the new version share
  
  another bug : s->bit_left will be in the range of 0..(BUF_BITS - 1)
  
  afterwards, although the assumption throughout the other PutBitContext
  
  functions is that it is in the range of 1..BUF_BITS. This might lead to
  
  a shift by BUF_BITS in little-endian mode. This has been fixed, too.
  
  The new version is furthermore able to skip zero bits, too.
  Signed-off-by : Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
  

  • [DH] libavcodec/put_bits.h

• Audio equalizer using FFmpeg

  28 septembre 2015, par Evgeniy Kharchenko

  I’m developing audio player using FFmpeg. I want to add audio equaliqer to my app, but I don’t know how to do it. I know that FFmpeg has a FFT functions, but I have count of samples that not mutch with any power of 2. I also tried to use other FFT libraries, but I have some noise on my audio after equalization.
  Can anybody help me to understand how to use FFmpeg’s FFT functions or any oter ways to implement audio-eq using FFmpeg ?