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• How do I install ffmpeg on one EC2 Amazon Linux instance that can stream a mp4 ? [closed]

  12 septembre 2020, par starpebble

  Good day. How can I install ffmpeg on an EC2 amazon linux machine that can stream a mp4 ?

  


  The goal : an ffmpeg install on EC2 Amazon Linux that can stream one mp4 to one rtmps endpoint. Then, create an integration test suite with it.

  


  Is it just me or is ffmpeg a little crippled on EC2 Amazon Linux ?

  


  Example :

  


  ffmpeg -re -i input.mp4 -c:v libx264 -b:v 6000K -maxrate 6000K -pix_fmt yuv420p -s 1920x1080 -profile:v main -preset veryfast -g 120 -x264opts "nal-hrd=cbr:no-scenecut” -acodec aac -ab 160k -ar 44100 -f flv rtmps:///app/


  


  Linux OS :

  


  Linux version 4.14.193-113.317.amzn1.x86_64 (mockbuild@koji-pdx-corp-builder-60005) (gcc version 7.2.1 20170915 (Red Hat 7.2.1-2) (GCC)) #1 SMP Thu Sep 3 19:08:08 UTC 2020


  


  The stackoverflow answer to similar questions fail to install a ffmpeg that can stream.

  


  An installation script such as Install FFMPEG Library on EC2 Server fail this year.

  


  The static downloads referenced on John Van Sickle-FFmpeg Static Builds fail to stream to IVS. I tried the i686 release, my first guess for an x86_64 instance.

  


  The git source tree compiled binary fails to stream. Example : The tip of the tree isn't what I expected because the binary fails to recognize switches like -preset.

  


  I'd love to be able to explain streaming to anyone. Thanks.

  


• PyDub : ffmpeg header missing

  7 février 2024, par Ray Siplao

  I made a simple Python program that add short pauses at the start of mp3 files

  


  import os

from pydub import AudioSegment

AUDIO_DIR = 'C:\\Users\\Ray\\Downloads\\sentence-audio'
OUT_DIR = 'C:\\Users\\Ray\\Downloads\\out-audio'

pause = AudioSegment.silent(duration=400)

for fname in os.listdir(AUDIO_DIR):
    print(f'Processing {fname}')
    newf = os.path.join(OUT_DIR, os.path.basename(fname))
    if os.path.isfile(newf):
        print(f'{fname} exists')
        continue
    audio = AudioSegment.from_mp3(os.path.join(AUDIO_DIR, fname))
    new = pause + audio
    new.export(newf, format='mp3')
    print(f'{fname} lengthened')


  


  While running this program, I received the below error message for a particular mp3 file

  


  Traceback (most recent call last):&#xA;  File "C:\Users\Ray\Downloads\lengthen.py", line 16, in <module>&#xA;    audio = AudioSegment.from_mp3(os.path.join(AUDIO_DIR, fname))&#xA;  File "C:\Python39\lib\site-packages\pydub\audio_segment.py", line 796, in from_mp3&#xA;    return cls.from_file(file, &#x27;mp3&#x27;, parameters=parameters)&#xA;  File "C:\Python39\lib\site-packages\pydub\audio_segment.py", line 773, in from_file&#xA;    raise CouldntDecodeError(&#xA;pydub.exceptions.CouldntDecodeError: Decoding failed. ffmpeg returned error code: 1&#xA;&#xA;Output from ffmpeg/avlib:&#xA;&#xA;ffmpeg version git-2020-04-03-52523b6 Copyright (c) 2000-2020 the FFmpeg developers&#xA;  built with gcc 9.3.1 (GCC) 20200328&#xA;  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf&#xA;  libavutil      56. 42.102 / 56. 42.102&#xA;  libavcodec     58. 77.101 / 58. 77.101&#xA;  libavformat    58. 42.100 / 58. 42.100&#xA;  libavdevice    58.  9.103 / 58.  9.103&#xA;  libavfilter     7. 77.101 /  7. 77.101&#xA;  libswscale      5.  6.101 /  5.  6.101&#xA;  libswresample   3.  6.100 /  3.  6.100&#xA;  libpostproc    55.  6.100 / 55.  6.100&#xA;[mp3float @ 0000016489995380] Header missing&#xA;    Last message repeated 71 times&#xA;[mp3 @ 000001648998b500] decoding for stream 0 failed&#xA;[mp3 @ 000001648998b500] Could not find codec parameters for stream 0 (Audio: mp3 (mp3float), 0 channels, fltp): unspecified frame size&#xA;Consider increasing the value for the &#x27;analyzeduration&#x27; and &#x27;probesize&#x27; options&#xA;Input #0, mp3, from &#x27;C:\Users\Ray\Downloads\sentence-audio\354.mp3&#x27;:&#xA;  Duration: N/A, start: 0.000000, bitrate: N/A&#xA;    Stream #0:0: Audio: mp3, 0 channels, fltp&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))&#xA;Press [q] to stop, [?] for help&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;[mp3float @ 0000016489995800] Header missing&#xA;Error while decoding stream #0:0: Invalid data found when processing input&#xA;...&#xA;</module>

  


  I thought at first that the mp3 file is corrupt, thus it is unplayable. But, when I played the file in a media player, it was fine.

  


  I don't know how to resolve this issue. Is there any workarounds ?

  


• 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