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Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir -
Les formats acceptés
28 janvier 2010, parLes commandes suivantes permettent d’avoir des informations sur les formats et codecs gérés par l’installation local de ffmpeg :
ffmpeg -codecs ffmpeg -formats
Les format videos acceptés en entrée
Cette liste est non exhaustive, elle met en exergue les principaux formats utilisés : h264 : H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 m4v : raw MPEG-4 video format flv : Flash Video (FLV) / Sorenson Spark / Sorenson H.263 Theora wmv :
Les formats vidéos de sortie possibles
Dans un premier temps on (...) -
Ajouter notes et légendes aux images
7 février 2011, parPour pouvoir ajouter notes et légendes aux images, la première étape est d’installer le plugin "Légendes".
Une fois le plugin activé, vous pouvez le configurer dans l’espace de configuration afin de modifier les droits de création / modification et de suppression des notes. Par défaut seuls les administrateurs du site peuvent ajouter des notes aux images.
Modification lors de l’ajout d’un média
Lors de l’ajout d’un média de type "image" un nouveau bouton apparait au dessus de la prévisualisation (...)
Sur d’autres sites (10178)
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Facebook LIVE with ffmpeg error - av_interleaved_write_frame() : Connection reset by peer
22 décembre 2016, par above14I’m trying to stream to Facebook a Live Video from an Amazon EC2 instance (c4.xlarge). I installed
ffmpeg
and stream with a command like the followingffmpeg -re -f concat -i /home/ec2-user/playlist.txt -f lavfi -re -i anullsrc=channel_layout=stereo:sample_rate=44100 -s 854x480 -c:a aac -strict -2 -ar 44100 -vcodec libx264 -f flv "{rtmp_link}"
Everything works fine. I had LIVE videos that worked for up to 2 days in a row. Sometimes, without a reason, I got one of these videos interrupted with this error message :
av_interleaved_write_frame(): Connection reset by peer
I haven’t been able to figure out a reason for this error yet. The funny thing is that maybe I have 3 streaming going on in the same machine at the same time, and just one of them get stopped by this message. Other times this error happens where there’s just one streaming on the EC2 instance.
Any ideas about how to find a solution ?
EDIT
We have installed a newer version of FFMPEG and the error now is more detailed. We have this message
RTMP send error 104
I haven’t found a lot about this issue online. Does someone know what error 104 means ?
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AWS Lambda subprocess OSError : [Errno 2] No such file or directory
11 septembre 2016, par LevI’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')
returnduring 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 directoryWhen I use the same sp.call() with the same ffmpeg binary on my ec2 instance it works fine.
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ffmpeg stream chrome kiosk mode ubuntu 16.04 server
21 décembre 2016, par RaulI 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_RTMPNote : 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 msAt this point, here’s what we observed :
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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
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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 :
- Why would ffmpeg have so much lag if it’s started right after chrome ?
- 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 ?
- 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.
- 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.
- 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 :
- 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.
- What could cause the initial audio/video out of sync issue and then catching up ?
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