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• FFmpeg transcoding on Lambda results in unusable (static) audio

  17 mai 2020, par jmkmay

  I'd like to move towards serverless for audio transcoding routines in AWS. I've been trying to setup a Lambda function to do just that ; execute a static FFmpeg binary and re-upload the resulting audio file. The static binary I'm using is here.

  



  The Lambda function I'm using in Python looks like this :

  



  import boto3

s3client = boto3.client('s3')
s3resource = boto3.client('s3')

import json
import subprocess 

from io import BytesIO

import os

os.system("cp -ra ./bin/ffmpeg /tmp/")
os.system("chmod -R 775 /tmp")

def lambda_handler(event, context):

    bucketname = event["Records"][0]["s3"]["bucket"]["name"]
    filename = event["Records"][0]["s3"]["object"]["key"]

    audioData = grabFromS3(bucketname, filename)

    with open('/tmp/' + filename, 'wb') as f:
        f.write(audioData.read())

    os.chdir('/tmp/')

    try:
        process = subprocess.check_output(['./ffmpeg -i /tmp/joe_and_bill.wav /tmp/joe_and_bill.aac'], shell=True, stderr=subprocess.STDOUT)
        pushToS3(bucketname, filename)
        return process.decode('utf-8')
    except subprocess.CalledProcessError as e:
        return e.output.decode('utf-8'), os.listdir()


def grabFromS3(bucket, file):

    obj = s3client.get_object(Bucket=bucket, Key=file)
    data = BytesIO(obj['Body'].read())

    return(data)

def pushToS3(bucket, file):

    s3client.upload_file('/tmp/' + file[:-4] + '.aac', bucket, file[:-4] + '.aac')

    return


  



  You can listen to the output of this here. WARNING : Turn your volume down or your ears will bleed.

  



  The original file can be heard here.

  



  Does anyone have any idea what might be causing the encoding errors ? It doesn't seem to be an issue with the file upload, since the md5 on the Lambda fs matches the MD5 of the uploaded file.

  



  I've also tried building the static binary on an Amazon Linux instance in EC2, then zipping and porting it into the Lambda project, but the same issue persists.

  



  I'm stumped ! :(

  


• How do I deploy Whisper.cpp stream.wasm on an EC2 instance ? [closed]

  3 août, par tosUser

  How do I deploy the whisper.cpp stream.wasm demo on a EC2 instance ?

  


  The demo is available from the following git :
https://github.com/ggml-org/whisper.cpp/tree/master/examples/stream.wasm

  


  I am fairly far along, but the program hangs on Preparing ...
I do not know what is causing the issue.

  


  Here are the steps I have taken. (I am trying to make it brief, and provide more detail as necessary.)
I cloned the repository and built it locally.
It runs properly.

  


  I created an EC2 instance running Amazon Linux 2023, and uploaded the Whisper.cpp files to it.
I ssh into the server and installed nginx, it displays the default page to the server's public ip address.
I then configured nginx as a reverse proxy server :

  


      server {
        listen 80;
        server_name redactedPublicIpAddress;
        location / {
            proxy_pass http://127.0.0.1:8000;
        }
    }



  


  I enabled and restarted nginx.

  


  I then try to run whisper.cpp stream.wasm by running python3 examples/server.py
It indicates that it is serving home/ec2-user/whisper.cpp/build-em/bin to localhost:8000
This is analogous to the message when it is served locally.
Stream.wasm has a 301 response.
Helpers.js and coi-serviceworker.js have 200 responses.

  


  The index file displays normally at the public ip address/stream.wasm/

  


  However, all is not well. The js seems to hang on Preparing...
Other js functions on the page still function, I can download a model and the page indicates it was downloaded, etc. However, I cannot start recording. The start button is not active. It never properly initializes.

  


  This type of behavior is identical to trying to open the index file on my local computer without it being served.

  


  I think I am missing something simple, but do not know what it is.

  


  Here are some steps that I took afterwards that did not fix the issue :
I installed nodejs and npm so that I could install ffmpeg.wasm : npm install @ffmpeg/ffmpeg @ffmpeg/util

  


  The github page indicates that I need to put the files in the html path. So in trying to troubleshoot I put them in /user/shared/nginx/html which is where the nginx default index file is placed.

  


  I think it may be related to serving the python. The server.py file is not in a state to deploy it with uvicorn.

  


  Maybe there is some sort of cors issue.

  


  I simply do not know.

  


• WebRTC books – a brief review

  30 décembre 2013, par silvia

  I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

  Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

  Rob’s focus is very much on the features required in a typical Web application :

  • video calls
  • audio calls
  • text chats
  • file sharing

  In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

  Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

  —

  Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

  Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

  Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

  —

  Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.