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• Muxing in audio to gstreamer RTMP stream kills both video and Audio

  1er avril 2015, par Adam

  I need some genius help here - I’m trying to set up a live stream for my upcoming wedding... and I have it ALMOST working - audio seems to be the problem.

  This is my setup

  • Raspberry Pi Model B+
  • Logitech C920 (with onboard h264 encoding that I am utilising)
  • on-camera (C920) microphone
  • USB wifi to iPhone 4G connection
  • gstreamer1.0
  • Amazon EC2 Wowza RTMP server

  I have it all set up, but as soon as I mux in the audio, the streams wont play by any player.

  What Works :
  - my gstreamer pipeline WITHOUT the audio muxed in
  - Wowza receives a consistent stream, no failures
  - The various Flash players / iOS / Android and VLC all play back the video

  What doesnt :
  - enabling audio in the mux (using the pipeline below)
  - BUT gstreamer doesnt complain
  - BUT Wowza receives a consistent stream, no failures
  - The various flash players fail to play both Audio and Video. some just display the first video frame
  - VLC plays 1 video frame, and about 100ms of audio, then stops

  Ideally I’d like the muxed audio/video FLV stored on the SD card too in case the network goes down - but if the ’tee’ needs to be sacrificed to make it work, so be it.

  This is my current FAILING pipeline - I assume there’s something really stupid in it because I know practically nothing about gstreamer.... The first frame loads in all the players (except iOS.. which never shows anything)

  # set camera resolution to 720p, and the data format to H264 (alternatives are YUV and JPG)
  
  v4l2-ctl --device=/dev/video0 --set-fmt-video=width=1280,height=720,pixelformat=1
  
  # set the frame rate
  
  v4l2-ctl --device=/dev/video0 --set-parm=10
  
  
  
  gst-launch-1.0 -v -e uvch264src initial-bitrate=300000 average-bitrate=300000 device=/dev/video0 name=src auto-start=true src.vidsrc \
  
                  ! queue \
  
                  ! video/x-h264,width=1280,height=720,framerate=10/1 \
  
                  ! h264parse \
  
                  ! flvmux streamable=true name=mux \
  
                  ! queue \
  
                  ! tee name=t \
  
                  ! queue \
  
                  ! filesink location=/home/pi/wedding.flv t. \
  
                  ! queue \
  
                  ! rtmpsink location='rtmp://wowzaserver/live/wedding live=1' >>/home/pi/wedding.log 2>&1

  Some of the things I can’t really afford to change at this late stage are the encapsulation (FLV) and wowza RTMP because I’ve built everything around that...

  Please Help !! Thanks !

  UPDATE

  Given that I am also saving the FLV file, I have found that if I use ffmpeg to send that FLV file (using audio copy, video copy) to the RTMP server, everything works (but obviously its not live) ! So I am now starting to believe this is a problem with the way Gstreamer encapsulates RTMP - and by putting ffmpeg in the middle it fixes it... but it’s not live of course.
  Is it possible to pipe my output to ffmpeg and using ffmpeg’s RTMP ?

• AWS Lambda function for modify video

  4 février 2017, par Gold Fish

  I want to create a Lambda function that invoked whenever someone uploads to the S3 bucket. The purpose of the function is to take the uploaded file and if its a video file (mp4) so make a new file which is a preview of the last one (using ffmpeg). The Lambda function is written in nodejs.
  I took the code here for reference, but I do something wrong for I get an error saying that no input specified for SetStartTime :

  //dependecies
  
  var async = require('async');
  
  var AWS = require('aws-sdk');
  
  var util = require('util');
  
  var ffmpeg = require('fluent-ffmpeg');
  
  
  
  // get reference to S3 client 
  
  var s3 = new AWS.S3();
  
  
  
  
  
  exports.handler = function(event, context, callback) {
  
      // Read options from the event.
  
      console.log("Reading options from event:\n", util.inspect(event, {depth: 5}));
  
      var srcBucket = event.Records[0].s3.bucket.name;
  
      // Object key may have spaces or unicode non-ASCII characters.
  
      var srcKey    =
  
      decodeURIComponent(event.Records[0].s3.object.key.replace(/\+/g, " "));  
  
      var dstBucket = srcBucket;
  
      var dstKey    = "preview_" + srcKey;
  
  
  
      // Sanity check: validate that source and destination are different buckets.
  
      if (srcBucket == dstBucket) {
  
          callback("Source and destination buckets are the same.");
  
          return;
  
      }
  
  
  
      // Infer the video type.
  
      var typeMatch = srcKey.match(/\.([^.]*)$/);
  
      if (!typeMatch) {
  
          callback("Could not determine the video type.");
  
          return;
  
      }
  
      var videoType = typeMatch[1];
  
      if (videoType != "mp4") {
  
          callback('Unsupported video type: ${videoType}');
  
          return;
  
      }
  
  
  
      // Download the video from S3, transform, and upload to a different S3 bucket.
  
      async.waterfall([
  
          function download(next) {
  
              // Download the video from S3 into a buffer.
  
              s3.getObject({
  
                      Bucket: srcBucket,
  
                      Key: srcKey
  
                  },
  
                  next);
  
              },
  
          function transform(response, next) {
  
          console.log("response.Body:\n", response.Body);
  
          ffmpeg(response.Body)
  
              .setStartTime('00:00:03')
  
              .setDuration('10')   //.output('public/videos/test/test.mp4')
  
          .toBuffer(videoType, function(err, buffer) {
  
                          if (err) {
  
                              next(err);
  
                          } else {
  
                              next(null, response.ContentType, buffer);
  
                          }
  
                   });
  
          },
  
          function upload(contentType, data, next) {
  
              // Stream the transformed image to a different S3 bucket.
  
              s3.putObject({
  
                      Bucket: dstBucket,
  
                      Key: dstKey,
  
                      Body: data,
  
                      ContentType: contentType
  
                  },
  
                  next);
  
              }
  
          ], function (err) {
  
              if (err) {
  
                  console.error(
  
                      'Unable to modify ' + srcBucket + '/' + srcKey +
  
                      ' and upload to ' + dstBucket + '/' + dstKey +
  
                      ' due to an error: ' + err
  
                  );
  
              } else {
  
                  console.log(
  
                      'Successfully modify ' + srcBucket + '/' + srcKey +
  
                      ' and uploaded to ' + dstBucket + '/' + dstKey
  
                  );
  
              }
  
  
  
              callback(null, "message");
  
          }
  
      );
  
  };

  So what am I doing wrong ?

• compile ffmpeg on ec2 linux instance - no libfdk-acc available

  11 mai 2022, par byc

  I’m compiling the ffmpeg on an ec2 amazon linux instance, but it threw me an error ERROR: libfdk_aac not found.

  


  I'm following this guide http://trac.ffmpeg.org/wiki/CompilationGuide/Centos to first install the dependencies and compile the package.

  


  yum install autoconf automake bzip2 bzip2-devel cmake freetype-devel gcc gcc-c++ git libtool make pkgconfig zlib-devel

mkdir ~/ffmpeg_sources

cd ~/ffmpeg_sources
curl -O -L https://ffmpeg.org/releases/ffmpeg-snapshot.tar.bz2
tar xjvf ffmpeg-snapshot.tar.bz2
cd ffmpeg
PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
  --prefix="$HOME/ffmpeg_build" \
  --pkg-config-flags="--static" \
  --extra-cflags="-I$HOME/ffmpeg_build/include" \
  --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
  --extra-libs=-lpthread \
  --extra-libs=-lm \
  --bindir="$HOME/bin" \
  --enable-gpl \
  --enable-libfdk_aac \
  --enable-libfreetype \
  --enable-libmp3lame \
  --enable-libopus \
  --enable-libvpx \
  --enable-libx264 \
  --enable-libx265 \
  --enable-nonfree
make
make install
hash -d ffmpeg


  


  and this is the error message I've got

  


  ec2-user@ip-xx-xxx-xx-xx ffmpeg]$ PATH="$HOME/bin:$PATH" PKG_CONFIG_PATH="$HOME/ffmpeg_build/lib/pkgconfig" ./configure \
>   --prefix="$HOME/ffmpeg_build" \
>   --pkg-config-flags="--static" \
>   --extra-cflags="-I$HOME/ffmpeg_build/include" \
>   --extra-ldflags="-L$HOME/ffmpeg_build/lib" \
>   --extra-libs=-lpthread \
>   --extra-libs=-lm \
>   --bindir="$HOME/bin" \
>   --enable-gpl \
>   --enable-libfreetype \
>   --enable-libmp3lame \
>   --enable-libopus \
>   --enable-libvpx \
>   --enable-libx264 \
>   --enable-libx265 \
>   --enable-nonfree

ERROR: libfdk_aac >= 3.98.3 not found


  


  I tried installing libfdk_aac but there's no package found. I don't seem to come across any posts discussing this issue, or at least recently. Appreciate any pointers. thanks !

  


  [ec2-user@ip-xx-xxx-xx-xx ffmpeg]$ sudo yum install libfdk-aac
Loaded plugins: extras_suggestions, langpacks, priorities, update-motd
No package libfdk-aac available.
Error: Nothing to do