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• Deploying on google App Engine : An error occurred : ffmpeg was killed with signal SIGABRT Error : ffmpeg was killed with signal SIGABRT

  27 août 2020, par Jérémy Gachon

  I wrote a node-js api, with node-js and fluent-ffmpeg :

  


  'use strict';
require('babel-register');
const path = require('path');    
const ffmpeg = require('fluent-ffmpeg');


  


  [...]

  


  var infs = new ffmpeg

infs.addInput(doc.data().url).outputOptions([
            '-preset slow', '-g 48', '-sc_threshold 0',
            '-map 0:0', '-map 0:1', '-map 0:0', '-map 0:1',
            '-s:v:0 1280x720', '-c:v:0 libx264', '-b:v:0 2000k',
            // "-var_stream_map", "'v:0,a:0 v:1,a:1'",
            '-master_pl_name ./' + req.params.id + '/master' + req.params.id + '.m3u8',
            '-f hls', '-hls_time 6', '-hls_list_size 0',
            '-hls_segment_filename ./' + req.params.id + '/fileSequence|' + req.params.id + '|%d|v%v.ts',
            '-max_muxing_queue_size 1024',
        ]).output('./' + req.params.id + '/video' + req.params.id + '.m3u8')
            .on('start', function (commandLine) {
                console.log('Spawned Ffmpeg with command: ' + commandLine);
            })
            .on('error', function (err, stdout, stderr) {
                console.log('An error occurred: ' + err.message, err, stderr);
            })
            .on('progress', function (progress) {
                console.log('Processing: ' + progress.percent + '% done')
            })
            .on('end', function (err, stdout, stderr) {

                console.log('Finished processing!' /*, err, stdout, stderr*/)
            })
            .run()
        res.status(200).send('GG').end();
    } 
   });


  


  [...]

  


  That work with

  


  


  node app.js

  


  


  on my macbook pro, but when i do

  


  


  gcloud app deploy

  


  


  and I call the public url, I have this logs :

  


  Processing: undefined% done
An error occurred: ffmpeg was killed with signal SIGABRT Error: ffmpeg was killed with signal SIGABRT 


  


  Here is my app.yaml :

  


      # Copyright 2017, Google, Inc.
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
#    http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.

# [START gae_flex_quickstart_yaml]
runtime: nodejs
env: flex

# This sample incurs costs to run on the App Engine flexible environment.
# The settings below are to reduce costs during testing and are not appropriate
# for production use. For more information, see:
# https://cloud.google.com/appengine/docs/flexible/nodejs/configuring-your-app-with-app-yaml
manual_scaling:
  instances: 1
resources:
  cpu: 1
  memory_gb: 6
  disk_size_gb: 30

# [END gae_flex_quickstart_yaml]


  


  

  


  How can I do to deploy correctly my node-js api on google app engine ?

  


  Thank's in advance.

  


  Jérémy.

  


• Best way to stream FFMPEG conversion to HTML5 video element ?

  4 août 2021, par SamTheFam

  Context

  


  Me and my friend are currently working on a media player desktop application using Electron (which implements website code into a desktop application). For the video element we are using the HTML5 video element, and loading a video from the user's local machine. Most browsers only support MP4, OGG, and WebM (Mozilla Specification), but we would like to support more formats to make it applicable to more users. For this reason, we decided to use the FFMPEG video converter. As FFMPEG takes some time to convert videos, we would like to have a streaming implementation that converts the video while playing the HTML5 video.

  


  The Problem

  


  The problem is that we cannot figure out the best way to stream our video whilst it is converting to another format. The primary issue is that we are unsure how we can implement skipping in the video, where we convert that specfic point if skipped to.

  


  For further clarification of the issue, if a user provided a .mkv file, we would start the FFMPEG conversion to .mp4, and would convert enough for the video to start playing. While the video is being watched, FFMPEG would continue converting before the user has reached that point in the video. If a user wanted to skip through the video, we would need a way to fast-forward the conversion to a specific point in the video, thus allowing the user to watch that section of the video, if it has not already been converted.

  


  Ideally we would like the conversion to be faster than the playback of the video, but if this has a cost in quality, we would prefer to make it optional to the end user.

  


  The first solution that came to our heads was to convert small sections of the video, and produce many small (3-5 second) videos, that would be loaded into the video element respectively. The issue here is that the video element takes quite a lot of time to load new source files and as a result make an awful user experience, with the video going black every x seconds.

  


  Another possible solution we stumbled accross was mentioned here. If we were to implement this, we would start a HTTP server on the user's local machine. The issue we had with this is that it could be a rather bloated solution and could be tedious to maintain. Along with this, it would likely slow down the start time for the application, and the runtime as well.

  


  Thank you in advance, we look forward to your responses.

  


• ffmpeg and Red5 Issue : Increase in number of ffmpeg simultaneous streams to Red5 resulting in packet loss

  30 octobre 2014, par kajarigd

  I have a screen sharing app written in flex, using which one person can share his screen with another person via Red5 server (Version : 1.0.3). Platform is Windows Server 2008. Now, I want to load test this Red 5 server to find out maximum how many simultaneous screen sharing session it can allow, without any quality compromise. By quality I mean, speed of transmission and no data loss during transmission. I simulated the load using ffmpeg command.

  For this, instead of transmitting a live captured screen, I am transmitting (uploading) a FLV file stored in my local to the Red5 server using ffmpeg command. In the receiving client side, I am starting to download (transmitting) this same FLV file after 5 secs since the upload has started. This is working fine when I am running this test for less than 10 pairs of upstreaming-downstreaming sessions. But, when the number is increasing beyond 10, I am observing significant packet loss in transmission.

  Here are the commands I am running in a loop. The loop count is the number of streaming pairs.

  1. upstreaming : ffmpeg -re -i  -f flv -ar 22050 "rtmp://" -report
  2. downstreaming : ffmpeg -re -i "rtmp:// live=1"  -report

  The and are set in such a way, that in the downstream I will download the same uploaded file. "rtmp ://" are the same in both the cases. I am not doing the upstream in record mode, hence, no physical file is getting saved in the server side. When I am analyzing the file I received in the receiving client side, it is a poor quality video due to frame loss. Uploading and downloading machines are two different machines. I ran the test for many hours, repeating the same 10 simultaneous streaming sets. Each set is consistently giving the same results.

  What is puzzling me is, this is working fine without any packet loss for less that 10 simultaneous streaming. I searched about it in various forums, but none of the answers were applicable for this scenario. For a while I was thinking that Red5 has limited capacity, but I found many posts saying Red5 can easily scale up to take very big load. Does that mean, the problem is in my configuration ? I am not sure which are to focus on.

  An example log snippet :

  Lots of missing data at downstream side. For e.g. between frames 101 and 102 there is a difference of 25 sec. On replaying the video there is a stoppage for this much time.In this time gap all the frames are lost.

  frame=  101 fps=1.0 q=14.5 size=    2650kB time=00:01:41.00 bitrate= 214.9kbits/s
  
  frame=  102 fps=1.0 q=13.2 size=    2763kB time=00:02:06.00 bitrate= 179.6kbits/s

  Any help is appreciated !