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• Ream-time watermarking with MPEG-DASH

  14 juillet 2016, par Calvin W.

  In the system, I want to add a unique watermark (e.g. IP address of client and time stamp) into the video that he/she want to watch.

  But when I handled it with OpenCV, it spent 25 minute with a 15-min video. And I need to transcode to mp4 with ffmpeg.

  Now I’m trying the watermark function of ffmpeg, bit it still needs some time.

  It it possible to send the video to client side with MPEG-DASH while transcoding it with ffmpeg ?

  System spec :(Amazon EC2 c3.xlarge)
  Intel Xeon E5-2680 v2 (Ivy Bridge) - 4 vCPU
  7.5G RAM
  40GB SSD
  Ubuntu 14.04 LTS
  OpenCV2.4.13
  ffmpeg 3.1.1
  

  code :

  import cv2
  
  import sys
  
  import time
  
  from datetime import datetime as dt
  
  
  
  # frame of input video
  
  fps = float(sys.argv[4])
  
  # encode to AVC
  
  fourcc = cv2.cv.CV_FOURCC('A', 'V', 'C', '1')
  
  # transparency of text
  
  alpha = 0.1
  
  beta = 1 - alpha
  
  
  
  # input video
  
  cap = cv2.VideoCapture(sys.argv[3])
  
  
  
  # current frame index, start from 0
  
  frameIndex = 0
  
  
  
  # get input video's width/height
  
  width = int(cap.get(cv2.cv.CV_CAP_PROP_FRAME_WIDTH))
  
  height = int(cap.get(cv2.cv.CV_CAP_PROP_FRAME_HEIGHT))
  
  
  
  # config output (error using .mp4)
  
  out = cv2.VideoWriter('output.avi', fourcc, fps, (width, height))
  
  
  
  # access time
  
  timeStr = dt.fromtimestamp(time.time()).strftime('%Y-%m-%d %H:%M:%S')
  
  
  
  requestIP = sys.argv[1]
  
  username = sys.argv[2]
  
  text = "%s %s %s" % (requestIP, username, timeStr)
  
  
  
  
  
  # start loading video
  
  while(cap.isOpened()):
  
      ret, frame = cap.read()
  
      if ret:
  
          # add text between 10s - 20s
  
          if frameIndex > time10 and frameIndex < time20:
  
              # clone a new frame to add text
  
              overlay = frame.copy()
  
              cv2.putText(overlay, text, (100, 100), cv2.FONT_HERSHEY_PLAIN, 0.5, (255, 255, 255))
  
              # combine both frame and make text transparent
  
              cv2.addWeighted(overlay, alpha, frame, beta, 0, frame)
  
          # write frame to output
  
          out.write(frame)
  
          frameIndex += 1
  
      # wait for next frame
  
      if cap.get(cv2.cv.CV_CAP_PROP_POS_FRAMES) == cap.get(cv2.cv.CV_CAP_PROP_FRAME_COUNT):
  
          break
  
  # End of video
  
  # release
  
  cap.release()
  
  out.release()

• 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;
}


  


• ffmpeg file conversion AWS Lambda

  10 avril 2021, par eartoolbox

  I want a .webm file to be converted to a .wav file after it hits my S3 bucket. I followed this tutorial and tried to adapt it from my use case using the .webm -> .wav ffmpeg command described here.

  


  My AWS Lambda function generally works, in that when my .webm file hits the source bucket, it is converted to .wav and ends up in the destination bucket. However, the resulting file .wav is always 0 bytes (though the .webm not, including the appropriate audio). Did I adapt the code wrong ? I only changed the ffmpeg_cmd line from the first link.

  


  import json
import os
import subprocess
import shlex
import boto3

S3_DESTINATION_BUCKET = "hmtm-out"
SIGNED_URL_TIMEOUT = 60

def lambda_handler(event, context):

    s3_source_bucket = event['Records'][0]['s3']['bucket']['name']
    s3_source_key = event['Records'][0]['s3']['object']['key']

    s3_source_basename = os.path.splitext(os.path.basename(s3_source_key))[0]
    s3_destination_filename = s3_source_basename + ".wav"

    s3_client = boto3.client('s3')
    s3_source_signed_url = s3_client.generate_presigned_url('get_object',
        Params={'Bucket': s3_source_bucket, 'Key': s3_source_key},
        ExpiresIn=SIGNED_URL_TIMEOUT)
    
    ffmpeg_cmd = "/opt/bin/ffmpeg -i \"" + s3_source_signed_url + "\" -c:a pcm_f32le " + s3_destination_filename + " -"
    
    
    command1 = shlex.split(ffmpeg_cmd)
    p1 = subprocess.run(command1, stdout=subprocess.PIPE, stderr=subprocess.PIPE)

    resp = s3_client.put_object(Body=p1.stdout, Bucket=S3_DESTINATION_BUCKET, Key=s3_destination_filename)

    return {
        'statusCode': 200,
        'body': json.dumps('Processing complete successfully')
    }