Sur d’autres sites (6567)

• AWS Lambda subprocess OSError : [Errno 2] No such file or directory

  11 septembre 2016, par Lev

  I’m trying to create a lambda function that makes collection of thumbnails from a video on amazon s3 using ffmpeg. ffmpeg binary is included into fuction package.

  function code :

  # -*- coding: utf-8 -*-
  
  
  
  import stat
  
  import shutil
  
  import boto3
  
  import logging
  
  import subprocess as sp
  
  import os
  
  import threading
  
  
  
  thumbnail_prefix = 'thumb_'
  
  thumbnail_ext = '.jpg'
  
  time_delta = 1
  
  video_frames_path = 'media/videos/frames'
  
  
  
  print('Loading function')
  
  logger = logging.getLogger()
  
  logger.setLevel(logging.INFO)
  
  
  
  lambda_tmp_dir = '/tmp'  # Lambda fuction can use this directory.
  
  
  
  # ffmpeg is stored with this script.
  
  # When executing ffmpeg, execute permission is requierd.
  
  # But Lambda source directory do not have permission to change it.
  
  # So move ffmpeg binary to `/tmp` and add permission.
  
  ffmpeg_bin = "{0}/ffmpeg.linux64".format(lambda_tmp_dir)
  
  shutil.copyfile('/var/task/ffmpeg.linux64', ffmpeg_bin)
  
  
  
  os.chmod(ffmpeg_bin, 777)
  
  
  
  # tried also:
  
  # os.chmod(ffmpeg_bin, os.stat(ffmpeg_bin).st_mode | stat.S_IEXEC)
  
  
  
  s3 = boto3.client('s3')
  
  
  
  
  
  def get_thumb_filename(num):
  
      return '{prefix}{num:03d}{ext}'.format(prefix=thumbnail_prefix, num=num, ext=thumbnail_ext)
  
  
  
  
  
  def create_thumbnails(video_url):
  
      i = 1
  
      filenames_list = []
  
      filename = None
  
      while i == 1 or os.path.isfile(os.path.join(os.getcwd(), get_thumb_filename(i-1))):
  
          if filename:
  
              filenames_list.append(filename)
  
          time = time_delta * (i - 1)
  
          filename = get_thumb_filename(i)
  
          print(ffmpeg_bin)
  
          if os.path.isfile(ffmpeg_bin):
  
              print('ok')
  
          sp.call(['sudo',
  
                   ffmpeg_bin,
  
                   '-ss',
  
                   str(time),
  
                   '-i',
  
                   video_url,
  
                   '-frames:v',
  
                   '1',
  
                   get_thumb_filename(i)])
  
          i += 1
  
      print(filenames_list)
  
      return filenames_list
  
  
  
  
  
  def s3_upload_file(file_path, key, bucket, acl, content_type):
  
      file = open(file_path, 'r')
  
      s3.put_object(
  
          Bucket=bucket,
  
          ACL=acl,
  
          Body=file,
  
          Key=key,
  
          ContentType=content_type
  
      )
  
      logger.info("file {0} moved to {1}/{2}".format(file_path, bucket, key))
  
  
  
  
  
  def s3_upload_files_in_threads(filenames_list, dir_path, bucket, s3path, acl, content_type):
  
      for filename in filenames_list:
  
          if os.path.isfile(os.path.join(dir_path, filename)):
  
              print(os.path.join(dir_path, filename))
  
          t = threading.Thread(target=s3_upload_file,
  
                               args=(os.path.join(dir_path, filename),
  
                                     '{0}/{1}'.format(s3path, filename),
  
                                     bucket,
  
                                     acl,
  
                                     content_type)).start()
  
  
  
  
  
  def lambda_handler(event, context):
  
      bucket = event['Records'][0]['s3']['bucket']['name']
  
      video_key = event['Records'][0]['s3']['object']['key']
  
      video_name = video_key.split('/')[-1].split('.')[0]
  
      video_url = 'http://{0}/{1}'.format(bucket, video_key)
  
      filenames_list = create_thumbnails(video_url)
  
      s3_upload_files_in_threads(filenames_list,
  
                                 os.getcwd(),
  
                                 bucket,
  
                                 '{0}/{1}'.format(video_frames_path, video_name),
  
                                 'public-read',
  
                                 'image/jpeg')
  
      return

  during the execution I get following logs :

  Loading function
  
  
  
  /tmp/ffmpeg.linux64
  
  
  
  ok
  
  
  
  [Errno 2] No such file or directory: OSError
  
  Traceback (most recent call last):
  
   File "/var/task/lambda_function.py", line 112, in lambda_handler
  
   filenames_list = create_thumbnails(video_url)
  
   File "/var/task/lambda_function.py", line 77, in create_thumbnails
  
   get_thumb_filename(i)])
  
   File "/usr/lib64/python2.7/subprocess.py", line 522, in call
  
   return Popen(*popenargs, **kwargs).wait()
  
   File "/usr/lib64/python2.7/subprocess.py", line 710, in __init__
  
   errread, errwrite)
  
   File "/usr/lib64/python2.7/subprocess.py", line 1335, in _execute_child
  
   raise child_exception
  
  OSError: [Errno 2] No such file or directory

  When I use the same sp.call() with the same ffmpeg binary on my ec2 instance it works fine.

• java.lang.NoClassDefFoundError : Could not initialize class on Linux (Works fine on Windows)

  26 décembre 2016, par Jake Miller

  I’m using a C++ FFmpeg wrapper for Java (org.bytedeco.javacpp). This works perfectly on a Windows machine (my development machine) but throws this error when ran on Linux (Amazon Web Services Elastic Beanstalk) :

  java.lang.NoClassDefFoundError: Could not initialize class org.bytedeco.javacpp.avutil
  
      at java.lang.Class.forName0(Native Method) ~[na:1.8.0_111]
  
      at java.lang.Class.forName(Class.java:348) ~[na:1.8.0_111]
  
      at org.bytedeco.javacpp.Loader.load(Loader.java:472) ~[javacpp-1.2.1.jar!/:1.2.1]
  
      at org.bytedeco.javacpp.Loader.load(Loader.java:417) ~[javacpp-1.2.1.jar!/:1.2.1]
  
      at org.bytedeco.javacpp.avformat$AVFormatContext.<clinit>(avformat.java:2819) ~[ffmpeg-3.2.1-1.3.jar!/:1.2.1]
  
      at org.bytedeco.javacv.FFmpegFrameGrabber.startUnsafe(FFmpegFrameGrabber.java:391) ~[javacv-1.3.jar!/:1.3]
  
      at org.bytedeco.javacv.FFmpegFrameGrabber.start(FFmpegFrameGrabber.java:385) ~[javacv-1.3.jar!/:1.3]
  
  </clinit>

  I’ve been troubleshooting for the past 2 days and have tried the following to fix the issue :

  • upgrade to Linux to 2.4
  • downgrading javacpp to 1.2.1
  • running mvn clean
  • running mvn -U
  • deleting contents of /.m2/ and redownloading dependencies
  • various combinations of dependency versions
  • git clone on a Linux VM & running mvn install there

  When looking further into the issue, I stumbled upon documentation for avformat$AVFormatContext as it’s in the stack trace posted above (6th line). The documentation for a C++ class named AVFormatContext. Whenever I attempt to view the class in Eclipse, it says Source Not Found.

  My question : could this problem possibly be caused by the C++ libraries on my Linux VM ? None of the above solutions fixed it so this is my only hypothesis as of now.

  Here’s my other Stack Overflow question regarding this subject : Java.lang.NoClassDefFoundError caused by FFmpeg when deployed on Linux as a packaged .war (Works on development machine)

• ffmpeg stream chrome kiosk mode ubuntu 16.04 server

  21 décembre 2016, par Raul

  I have a weird out-of-sync issue while using ffmpeg to stream to youtube live a chrome browser from an ub untu 16.04 server.

  Issue : output video streamed to youtube has audio/video out of sync, sometimes with as much as 3s

  Current flow :

  1) start pulseaudio - we using something like this to start it :

  pulseaudio --start -vvv --disallow-exit --log-target=syslog --high-priority --exit-idle-time=-1 --daemonize

  2) start Xvfb

  Xvfb :0 -ac -screen 0 1920x1080x24

  3) start chrome linux in kiosk mode

  google-chrome --kiosk --disable-gpu --incognito --no-first-run --disable-java --disable-plugins --disable-translate --disk-cache-size=$((1024 * 1024)) --disk-cache-dir=/tmp/chrome/ --user-data-dir=/tmp/chrome/ --force-device-scale-factor=1 --window-size=1920,1080 --window-position=0,0 LOCATION_URL

  4) start ffmpeg

  ffmpeg -y \
  
    -thread_queue_size 8192 -rtbufsize 250M -f x11grab -video_size 1920x1080 -framerate 24 -i :0 \
  
    -thread_queue_size 8192 -channel_layout stereo -f alsa -i pulse \
  
    -c:v libx264 -pix_fmt yuv420p -c:v libx264 -g 48 -crf 24 -filter:v fps=24 -preset ultrafast -tune zerolatency \
  
    -c:a aac -strict -2 -channel_layout stereo -ab 96k -ac 2 -flags +global_header \
  
    -f flv YOUTUBE_LIVE_STREAMING_RTMP

  Note : this is running on an amazon ec2 instance, meaning there is no soundcard, so alsa and pulseaudio are creating a dummy audio card. However, the latency does not come from there. Logs :

  Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Adjust latency mode enabled, configuring sink latency to half of overall latency.
  
  Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Requested latency=23.22 ms, Received latency=23.22 ms
  
  Nov 25 06:14:22 ip-172-31-29-8 pulseaudio[26602]: [pulseaudio] protocol-native.c: Final latency 69.66 ms = 23.22 ms + 2*11.61 ms + 23.22 ms

  At this point, here’s what we observed :

  1. if we start ffmpeg exactly after issuing the command to start chrome, we see the DTS errors from ffmpeg. Audio is out of sync with the video and has delay of 3-5seconds AHEAD. We also noticed the out of sync remains the same for the full duration of the stream

  2. if we start ffmpeg after around 10seconds, audio and video are almost in sync. We then manually added a -itsoffset -0.125 to the ffmpeg command and everything is perfect.

  Questions :

  1. Why would ffmpeg have so much lag if it’s started right after chrome ?
  2. Is starting the ffmpeg after 10s or X seconds the expected behavior ? That is, is this because the system needs to wait for audio/video signals to be "ready" or something ?
  3. Is there a way to 100% calculate or know when Chrome is fully ready and start ffmpeg ? We found sometimes it takes 5s, sometimes 10. Depends on the URL we load.
  4. Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything. And a restart is required to "re-balance" the audio/video inputs and get them back in sync.
  5. Can pulseaudio be the problem in this scenario ?

  Thank you

  UPDATE Dec 20

  We were able to do some tricks to force chrome to start the audio on page load, and that will force connect to pulseaudio. Doing that, plus adding a 3s delay for ffmpeg to start, there is no more delay when ffmpeg starts.
  However, our app is a webRTC app, and we have a STRANGER thing happening : if we start the page with no webcam/audio, once the webcam/audio is enabled, ffmpeg (while showing no errors) has a delay of 2s or so. While keep talking, in about max 30s, that delay is GONE.

  So the new questions are :

  1. Besides the DTS error that ffmpeg throws, is there any other way to know if audio/video is out-of-sync ? as sometimes we have a delay of between 0.5 to 1s, but ffmpeg does not report anything.
  2. What could cause the initial audio/video out of sync issue and then catching up ?