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• FFMPEG color key filter applied, output to a transparent HEVC mov file, speed is different

  2 mars 2023, par Patrick Vellia

  I have an original recording that has the following ffprobe output :

  


  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'test.mp4':
  Metadata:
    major_brand     : mp41
    minor_version   : 538120216
    compatible_brands: mp41
    creation_time   : 2023-02-28T19:06:41.000000Z
    firmware        : H21.01.01.50.00
  Duration: 00:00:08.15, start: 0.000000, bitrate: 60160 kb/s
  Stream #0:0(eng): Video: hevc (Main) (hvc1 / 0x31637668), yuvj420p(pc, bt709), 3840x2160 [SAR 1:1 DAR 16:9], 59891 kb/s, 119.88 fps, 119.88 tbr, 120k tbn, 119.88 tbc (default)
    Metadata:
      creation_time   : 2023-02-28T19:06:41.000000Z
      handler_name    : GoPro H.265
      vendor_id       : [0][0][0][0]
      encoder         : GoPro H.265 encoder
      timecode        : 19:05:32:105
  Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 189 kb/s (default)
    Metadata:
      creation_time   : 2023-02-28T19:06:41.000000Z
      handler_name    : GoPro AAC  
      vendor_id       : [0][0][0][0]
      timecode        : 19:05:32:105
  Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s (default)
    Metadata:
      creation_time   : 2023-02-28T19:06:41.000000Z
      handler_name    : GoPro TCD  
      timecode        : 19:05:32:105
  Stream #0:3(eng): Data: bin_data (gpmd / 0x646D7067), 76 kb/s (default)
    Metadata:
      creation_time   : 2023-02-28T19:06:41.000000Z
      handler_name    : GoPro MET  
Unsupported codec with id 0 for input stream 2
Unsupported codec with id 100359 for input stream 3


  


  This video is chroma keyed, so I ran the following command the key out the green and export it to a HEVC mov container (the Apple HEVC supports alpha channel via the videotoolbox codec) :

  


  ffmpeg -i test.mp4 -vf colorkey=0x00b140:0.3:0.1 -vcodec hevc_videotoolbox -alpha_quality 0.5 -tag:v hvc1 output1.mov


  


  which results in this ffprobe :

  


  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'output1.mov':
  Metadata:
    major_brand     : qt  
    minor_version   : 512
    compatible_brands: qt  
    encoder         : Lavf58.76.100
  Duration: 00:00:08.15, start: 0.000000, bitrate: 366946 kb/s
  Stream #0:0(eng): Video: hevc (Main) (hvc1 / 0x31637668), yuv420p(tv, gbr/bt709/bt709, progressive), 3840x2160 [SAR 1:1 DAR 16:9], 367188 kb/s, 119.88 fps, 119.88 tbr, 120k tbn, 120k tbc (default)
    Metadata:
      handler_name    : GoPro H.265
      vendor_id       : FFMP
      encoder         : Lavc58.134.100 hevc_videotoolbo
      timecode        : 19:05:32:105
  Stream #0:1(eng): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 118 kb/s (default)
    Metadata:
      handler_name    : GoPro AAC  
      vendor_id       : [0][0][0][0]
  Stream #0:2(eng): Data: none (tmcd / 0x64636D74), 0 kb/s
    Metadata:
      handler_name    : GoPro H.265
      timecode        : 19:05:32:105
Unsupported codec with id 0 for input stream 2


  


  When the play the original video inside a web browser, it will play normally. As in, the video doesn't play in slo-mo but at "normal speed" which is what I actually want. User can slow it down and still have the frames necessary to keep the clarity when desired.

  


  However, when I play the second video, it is unnaturally slow even at 2x the speed.

  


  The only difference I can see above is that the export didn't include the second data stream (input 3).

  


  This was recorded on a GoPro Hero 10. I am using FFMPEG 4.6.

  


  Any help would be appreciated to fine tune the command.

  


• /bin/sh : ffmpeg : command not found while merge audio with with using ffmpeg aws lamda [duplicate]

  1er mai 2023, par bhavesh kaila

  I am trying to merge audio with video using the below command in AWS lambda using Python but the getting and below error.

  


  ffmpeg Command

  


  cmd = 'ffmpeg  -stream_loop -1 -i mayur.mp4 -i audio.mp3 -shortest -map 0:v:0 -map 1:a:0 -y out/output.mp4'
subprocess.call(cmd, shell=True) 


  


  Aws Lambda Error

  


  2023-05-01T16:08:18.591+05:30   INIT_START Runtime Version: python:3.10.v2 Runtime Version ARN: arn:aws:lambda:eu-north-1::runtime:7764dc7f3ff1fc45718f596be4cd03d7bca223f0586f3bfa5fe6584d6af81cd8

2023-05-01T16:08:19.631+05:30   START RequestId: 4557a174-e12c-4924-971a-ef8f852b106b Version: $LATEST

2023-05-01T16:08:22.286+05:30

Copy
/bin/sh: ffmpeg: command not found
/bin/sh: ffmpeg: command not found

2023-05-01T16:08:23.926+05:30   [ERROR] FileNotFoundError: [Errno 2] No such file or directory: '/tmp/output.mp4' Traceback (most recent call last):   File "/var/task/lambda_function.py", line 65, in lambda_handler     s3_upload_response = s3.put_object(Bucket=bucket, Body=open('/tmp/output.mp4', 'rb'), Key =filename)

2023-05-01T16:08:23.929+05:30   END RequestId: 4557a174-e12c-4924-971a-ef8f852b106b


  


  Full Code snippet

  


  import json
from gtts import gTTS
import openpyxl
import boto3
import base64
import openai
import os
import random
import ffmpeg
import subprocess

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

#from botocore.vendored import requests
language = 'en'
def lambda_handler(event, context):
    input1 = event["queryStringParameters"]['question']
    outputText=""
    
    # Define variable to load the dataframe
    dataframe = openpyxl.load_workbook('QNA/Book1.xlsx')
    
      #accesskey and secretkey for the S3 bucket
    accesskey = os.getenv("access_key")
    secretkey = os.getenv("secret_key")
    
    # Define variable to read sheet
    ws = dataframe['Sheet1']
    
    for row in ws.iter_rows(0,dataframe.active.max_row):
        for cell in row:
          #print(cell.value)
          if input1 in str(cell.value):
            outputText=row[1].value
  
    
    
      #Save Audio File
    audio = gTTS(text=outputText, lang=language, slow=False)
    audio.save("/tmp/audio.mp3")
    
    cmd='ffmpeg -stream_loop -1 -i QNA/mayur.mp4 -i tmp/audio.mp3 -shortest -map 0:v:0 -map 1:a:0 -y tmp/output.mp4'
    subprocess.run(cmd, shell=True)


  


  Note : I Download the static library from Here and added it to the python folder and generate python.zip with other dependencies and upload it on the aws layer and the AWS layer is linked with the python function.

  


  Note : Above code is working fine with google codelab and is able to generate a merged video working fine.

  


  Any help would be appriciated

  


• youtube stream from ffmpeg is buffering

  1er juin 2020, par Bartonsen

  I'm using ffmpeg running on a Raspberry Pi 3B with 1GB RAM to stream live video on youtube.
In the beginning the audio+video stream is excellent, but after some minutes I see error messages in YT studio, and video starts buffering.
After some more time (could be 30 mins or 1 hr) the youtube stream is gone, although ffmpeg is still running.

  



  ffmpeg is configured like this :

  



  ./configure --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi


  



  Running ffmpeg :

  



  pi@raspberrypi:~ $ ffmpeg -thread_queue_size 512 -rtsp_transport udp -i "rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream" -c:v copy -c:a aac -f flv rtmp://a.rtmp.youtube.com/live2/mykey
ffmpeg version git-2020-05-01-3c740f2 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 6.3.0 (Raspbian 6.3.0-18+rpi1+deb9u1) 20170516
  configuration: --arch=armel --target-os=linux --enable-gpl --enable-libx264 --enable-nonfree --enable-omx-rpi
  libavutil      56. 43.100 / 56. 43.100
  libavcodec     58. 82.100 / 58. 82.100
  libavformat    58. 42.102 / 58. 42.102
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 80.100 /  7. 80.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
[udp @ 0x3a8c370] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a9ea80] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3a8c3e0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
[udp @ 0x3abf4d0] attempted to set receive buffer to size 393216 but it only ended up set as 327680
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://10.x.x.x:554/user=user&password=password&channel=1&stream=0.sdp?real_stream':
  Metadata:
    title           : RTSP Session
  Duration: N/A, start: 0.000000, bitrate: N/A
    Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1920x1080, 20 fps, 20 tbr, 90k tbn, 180k tbc
    Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (pcm_alaw (native) -> aac (native))
Press [q] to stop, [?] for help
[aac @ 0x3ae9f00] Too many bits 8832.000000 > 6144 per frame requested, clamping to max
Output #0, flv, to 'rtmp://a.rtmp.youtube.com/live2/mykey':
  Metadata:
    title           : RTSP Session
    encoder         : Lavf58.42.102
    Stream #0:0: Video: h264 (Main) ([7][0][0][0] / 0x0007), yuvj420p(pc, bt709, progressive), 1920x1080, q=2-31, 20 fps, 20 tbr, 1k tbn, 90k tbc
    Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 8000 Hz, mono, fltp, 48 kb/s
    Metadata:
      encoder         : Lavc58.82.100 aac


  



  In youtube studio I see this :

  



  19:36 Good transmission. The quality is excellent.
19:39 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:39 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:39 Warning: The current bit rate (1974.24 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:41 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:41 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:41 Warning: The current bit rate (2151.41 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
19:43 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:43 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Suggestion: The current sampling rate is 0. Recommended sampling rates are 44.1 kHz and 48 kHz.
19:45 Suggestion: The current bitrate (0) of the audio stream is lower than the recommended bitrate. We recommend using a 128 Kbps bitrate for the audio stream.
19:45 Warning: The current bit rate (1737.61 kbps) is lower than the recommended bit rate. We recommend using a 4500 Kbps bitrate for the stream.
...
19:54: Error: YouTube does not receive enough video to maintain consistent streaming. Viewers will therefore experience buffering.


  



  What is the problem ? How can I get rid of the buffering ?
I've also tried the below two commands, but found the output to be worse...

  



  ffmpeg -i rtsp://... -c:v libx264  -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output
ffmpeg -i rtsp://... -c:v h264_omx -b:v 4000k -maxrate 4000k -bufsize 8000k -g 40 -preset ultrafast -vf format=yuv420p -c:a aac -f flv output