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• Encoding audio_common messages to OPUS

  14 juin 2023, par djangbahevans

  ---

  


  I am trying to stream microphone and camera data to Amazon KVS WebRTC. I'm able to make video work using this package (adapted for noetic) however I am struggling to make audio work. I'm using the audio_capture package to get mp3 frames. I'm trying to convert this to OPUS frames before streaming to KVS, but I'm unsure how to do this. I wrote this bit of code based on the small resources I can find on using ffmpeg, but it's not working. avcodec_fill_audio_frame is returning -22.

  


  #include "opus_encoder.h"

OPUSEncoder::OPUSEncoder() {
  av_register_all();
  codecContext == nullptr;
}

OPUSEncoder::~OPUSEncoder() {
  if (codecContext != nullptr) {
    avcodec_free_context(&codecContext);
  }
}

int OPUSEncoder::Initialize(int Fs, int channels) {
  AVCodec *codec = avcodec_find_encoder(AV_CODEC_ID_OPUS);
  if (!codec) {
    printf("Codec not found\n");
    return -1;
  }

  codecContext = avcodec_alloc_context3(codec);
  if (!codecContext) {
    printf("Could not allocate audio codec context\n");
    return -1;
  }

  codecContext->sample_fmt = AV_SAMPLE_FMT_S16;
  codecContext->bit_rate = 128000;
  codecContext->sample_rate = Fs;
  codecContext->channel_layout = av_get_default_channel_layout(channels);
  codecContext->channels = channels;

  if (avcodec_open2(codecContext, codec, nullptr) < 0) {
    printf("Could not open codec\n");
    return -1;
  }

  return 0;
}

int OPUSEncoder::Encode(const uint8_t *audio_data, int frameSize,
                        uint8_t *out) {
  AVPacket pkt;
  av_init_packet(&pkt);
  pkt.data = nullptr;
  pkt.size = 0;

  AVFrame *frame = av_frame_alloc();
  frame->nb_samples = frameSize;
  frame->format = codecContext->sample_fmt;
  frame->channel_layout = codecContext->channel_layout;

  int ret = avcodec_fill_audio_frame(frame, codecContext->channels,
                                     codecContext->sample_fmt, audio_data,
                                     frameSize * 2, 0);
  if (ret < 0) {
    printf("Error filling audio frame: %d\n", ret);
    return -1;
  }

  ret = avcodec_send_frame(codecContext, frame);
  if (ret < 0) {
    printf("Error sending the frame to the encoder\n");
    return -1;
  }

  while (ret >= 0) {
    ret = avcodec_receive_packet(codecContext, &pkt);
    if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
      return 0;
    } else if (ret < 0) {
      printf("Error encoding audio frame\n");
      return -1;
    }

    memcpy(out, pkt.data, pkt.size);
    out += pkt.size;
    av_packet_unref(&pkt);
  }

  av_frame_free(&frame);

  return 0;
}


  


• How to get the thumbnail of base64 encoded video file in Nodejs ?

  3 octobre 2018, par Wai Yan Hein

  I am developing a web application using Nodejs. I am using Amazon S3 bucket to store files. What I am doing now is that when I upload a video file (mp4) to the S3 bucket, I will get the thumbnail photo of the video file from the lambda function. For fetching the thumbnail photo of the video file, I am using this package - https://www.npmjs.com/package/ffmpeg. I tested the package locally on my laptop and it is working.

  Here is my code tested on my laptop

  var ffmpeg = require('ffmpeg');
  
  
  
  module.exports.createVideoThumbnail = function(req, res)
  
  {
  
      try {
  
          var process = new ffmpeg('public/lalaland.mp4');
  
          process.then(function (video) {
  
  
  
              video.fnExtractFrameToJPG('public', {
  
                  frame_rate : 1,
  
                  number : 5,
  
                  file_name : 'my_frame_%t_%s'
  
              }, function (error, files) {
  
                  if (!error)
  
                      console.log('Frames: ' + files);
  
                  else
  
                      console.log(error)
  
              });
  
  
  
          }, function (err) {
  
              console.log('Error: ' + err);
  
          });
  
      } catch (e) {
  
          console.log(e.code);
  
          console.log(e.msg);
  
      }
  
      res.json({ status : true , message: "Video thumbnail created." });
  
  }

  The above code works well. It gave me the thumbnail photos of the video file (mp4). Now, I am trying to use that code in the AWS lambda function. The issue is the above code is using video file path as the parameter to fetch the thumbnails. In the lambda function, I can only fetch the base 64 encoded format of the file. I can get id (s3 path) of the file, but I cannot use it as the parameter (file path) to fetch the thumbnails as my s3 bucket does not allow public access.

  So, what I tried to do was that I tried to save the base 64 encoded video file locally in the lambda function project itself and then passed the file path as the parameter for fetching the thumbnails. But the issue was that AWS lamda function file system is read-only. So I cannot write any file to the file system. So what I am trying to do right now is to retrieve the thumbnails directly from the base 64 encoded video file. How can I do it ?

• ffmpeg h.264 invalid cutting

  1er mai 2012, par E.Ar

  I have an s3 bucket with several hundreds video files.
  Those files were created by cutting parts of larger video files using ffmpeg.
  I wrote a script for this, which downloads the original video file from another bucket, runs ffmpeg to cut the file, and uploads the new file to it's bucket.
  For downloading and uploading from/to s3 i used this php library.
  The ffmpeg syntax I used :

  ffmpeg -y -vsync 2 -async 1 -ss [time-in] -t [duration] -i [large-input-video.mp4] -vcodec copy -acodec copy [short-output-video.mp4]

  Which should just cut the original file between the specified times, without any changes to the a/v codecs.
  All the original video files are encoded in h.264, and this is also the required encoding for the new files (which will be streamed through a CDN to the clients' flash players).

  My problem is that only a small part of the new files are coming out as encoded in h.264, but most of them aren't (h.264 is a must, otherwise the files wont play on the clients' side).
  I can't trace the problem to the original videos, since when i use the same ffmpeg command manually, with the same parameters and on the same files, the output files come out just fine. It seems arbitrary.

  I use ffprobe to get information about the files' codecs.
  For example :
  ffprobe of one of the large (original) video files :

  ...
  Stream #0.0(und) : Video : h264, yuv420p, 640x352, 499 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc
  ...

  ffprobe of the corresponding new cut file :

  ...
  Stream #0.0(und) : Video : mpeg4, yuv420p, 640x352 [PAR 1:1 DAR 20:11], 227 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
  ...

  As can be seen, the difference is in 'mpeg4' vs. 'h264'.

  Any insights on what can cause the new files to come out in the wrong encoding would be greatly appreciated.

  Thanks !

  Edit : Problem Resolved
  After analyzing all the files, I noticed that about two thirds of them are coming out in the wrong codec.
  Since I used three machines for the cutting process (three separate EC2 servers), it occurred to me that on two of them there is a bad installation of ffmpeg (as @LordNeckbeard suggested in his answer).
  I ran the process again, only on the invalid files, on the third machine alone - which produced the desired result.