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• Python and ffmpeg audio sync and screen recording issues

  9 août 2020, par odddollar

  I'm using ffmpeg and python to record my desktop screen. When the program is run, it starts recording, then when I press a key-combo it cuts off the last x amount of seconds and saves it then starts recording again ; similar to the "record that" functionality of windows game bar.

  


  I have it working so it records video just fine, but then I change the ffmpeg command to record audio from my desktop and I get an error saying ValueError: could not convert string to float: 'N/A' occurring when I try to calculate the length of the recorded video. It appears as though the recording isn't being stopped until after I try to calculate the video length, even though this exact same code works fine when not recording audio.

  


  Additionally, I also have an issue when recording audio in that the audio is a couple hundred milliseconds in front of the video. It's not a lot but it's enough to be noticeable.

  


  What I'm overall asking, is there a way I can modify the ffmpeg command to prevent the audio desync issues, and what might be causing the problems I'm getting when attempting to find the length of the video with audio ?

  


  import keyboard, signal
from os import remove
from os.path import isfile
from subprocess import Popen, getoutput
from datetime import datetime
import configparser

class Main:
    def __init__(self, save_location, framerate, duration):
        self.save_location = save_location
        self.framerate = int(framerate)
        self.duration = int(duration)
        self.working = self.save_location + '\\' + 'working.avi'
        self.start_recording()

    def start_recording(self):
        if isfile(self.working):
            remove(self.working)

        # start recording to working file at set framerate
        self.process = Popen(f'ffmpeg -thread_queue_size 578 -f gdigrab -video_size 1920x1080 -i desktop -f dshow -i audio="Stereo Mix (Realtek High Definition Audio)" -b:v 7M -minrate 4M -framerate {self.framerate} {self.working}')
        #self.process = Popen(f'ffmpeg -f gdigrab -framerate {self.framerate} -video_size 1920x1080 -i desktop -b:v 7M -minrate 2M {self.working}')

    def trim_video(self):
        # stop recording working file
        self.process.send_signal(signal.CTRL_C_EVENT)

        # call 'cause I have to
        getoutput(f"ffprobe -i {self.working}")

        # get length of working video
        length = getoutput(f'ffprobe -i {self.working} -show_entries format=duration -v quiet -of csv="p=0"')

        # get time before desired recording time
        start = float(length) - self.duration

        # get save location and title
        title = self.save_location+'\\'+self.get_time()+'.avi'

        # cut to last amount of desired time
        Popen(f"ffmpeg -ss {start} -i {self.working} -c copy -t {self.duration} {title}")
        getoutput(f"ffprobe -i {self.working}")

        self.start_recording()

    def get_time(self):
        now = datetime.now()
        return now.strftime("%Y_%m_%d#%H-%M-%S")


if __name__ == "__main__":
    config = configparser.ConfigParser()
    config.read("settings.ini")
    config = config["DEFAULT"]

    run = Main(config["savelocation"].replace("\\", "\\\\"), config["framerate"], config["recordlast"])
    keyboard.add_hotkey("ctrl+shift+alt+g", lambda:run.trim_video())

    while True:
        try:
            keyboard.wait()
        except KeyboardInterrupt:
            pass


  


  The contents of the settings.ini file are listed below

  


  [DEFAULT]
savelocation = C:\
framerate = 30
recordlast = 10


  


  In the code block, the first line with self.process = Popen is the one that records audio and has the issues, the second line (the commented out one below) is the one that works fine.

  


• avutil/channel_layout : add 22.2 layout

  11 juin 2020, par Jan Ekström

  avutil/channel_layout : add 22.2 layout
  Requires some extraneous top side and bottom front channels to be
  
  defined.
  According to STD-B59v2, the defined channel layout is :
  
   FL
  
   FR
  
   FC
  
   LFE1
  
   BL
  
   BR
  
   FLc
  
   FRc
  
   BC
  
   LFE2
  
   SiL
  
   SiR
  
   TpFL
  
   TpFR
  
   TpFC
  
   TpC
  
   TpBL
  
   TpBR
  
   TpSiL
  
   TpSiR
  
   TpBC
  
   BtFC
  
   BtFL
  
   BtFR
  

  • [DH] doc/APIchanges
  • [DH] libavutil/channel_layout.c
  • [DH] libavutil/channel_layout.h
  • [DH] libavutil/version.h

• Ffmpeg inaccurate cutting with ts and m3u8 files despite resamping audio filter

  31 juillet 2020, par Lemon Sky

  I need to accurately seek and cut a video. Some online sources say put -ss in front or after the source. The result is the same for me. In the examples below, the start time is accurate but the duration is not accurate.

  


  ffmpeg -y -ss 00:00:05 -t 00:00:05 -i output.ts 5s.wav
ffprobe 5s.wav

Duration: 00:00:04.74

ffmpeg -y -i output.ts -ss 00:00:05 -t 00:00:05 5s.wav
ffprobe 5s.wav

Duration: 00:00:04.74


  


  Sometimes the starting point is not accurate but the duration is. This is clearly audible by cutting the ts file and cutting an uncompressed wav file, and listening to both.

  


  ffmpeg -y -i output.ts -ss 00:00:15 -t 00:00:05 5s.wav

ffmpeg -y -i output.wav -ss 00:00:15 -t 00:00:05 5s-reference.wav


  


  What fixes the starting time is if I use an m3u8 file that contains the byte offset for every keyframe AND I put the -ss option in front of the source file (if I put it after the source, the start time is inaccurate but the duration is accurate) :

  


  ffmpeg -y -ss 00:00:15 -t 00:00:05 -i output.m3u8 5s.wav


  


  This fixes the start time but the duration is at the location that I would get had I used no m3u8 file (duration is just 4.47s).

  


  It seems different timestamps are involved, and sometimes one or the other gets used.

  


  The ts file was generated by capturing a UDP stream and storing it with ffmpeg and -codec:v copy.

  


  Is ffmpeg broken, or the ts file ? How do I work around this issue or fix the ts file ? What I realize is that the video starts later than the audio, probably because the video does not start with a keyframe. Can I get ffmpeg to start the -codec:v copy after the first keyframe ? What I also notice is that using ffprobe reports "start : 1.400000". Can I force it to start at 0 ?

  


  Any hints would be appreciated.

  


  I tried both ffmpeg 4.3.1 and ffmpeg git-2020-07-24-21442a8.

  


  The output.* files were generated as follows. The statement "af aresample=async=1" should fill missing audio according to Duration of source video and subtracted audio are different. Adding this statement makes no difference in terms of accuracy or duration. The question is not a duplicate.

  


  ffmpeg -i udp://example:port ^
-af aresample=async=1 ^
-codec:v copy ^
-codec:a aac -ac 2 -ar 44100 -b:a 160k ^
-hls_time 4 -hls_flags single_file -hls_list_size 0 -hls_segment_filename output.ts -hls_segment_type mpegts output.m3u8 ^
-codec:a pcm_s16le -bitexact -ar 11025 -ac 1 output.wav