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• Trouble with converting webm into mp3 with pydub in python

  15 août 2020, par rc_marty

  so basically I want to convert song what I downloaded from youtube in webm and convert to into mp3

  


  when I wanted export song just with song.export("neco.mp3") it didn't work too

  


  I have in workfolder ffmpeg.exe and ffprobe.exe

  


  here is the code

  


  from pydub import AudioSegment

song = AudioSegment.from_file(downloaded.webm,"webm")
print("Loaded")
song.export("neco.mp3", format="mp3", bitrate="320k")
print("Converted and saved")


  


  here is the output of the console

  


  Loaded&#xA;Traceback (most recent call last):&#xA;  File "e:/martan/projekty/Python/programek na pisnicky/songDownloader.py", line 188, in <module>&#xA;    song.export("neco.mp3", format="mp3", bitrate="320k")&#xA;  File "C:\Users\BIBRAIN\AppData\Local\Programs\Python\Python38\lib\site-packages\pydub\audio_segment.py", line 911, in export&#xA;    raise CouldntEncodeError(&#xA;pydub.exceptions.CouldntEncodeError: Encoding failed. ffmpeg/avlib returned error code: 1&#xA;&#xA;Command:[&#x27;ffmpeg&#x27;, &#x27;-y&#x27;, &#x27;-f&#x27;, &#x27;wav&#x27;, &#x27;-i&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpo20ooz_z&#x27;, &#x27;-b:a&#x27;, &#x27;320k&#x27;, &#x27;-f&#x27;, &#x27;mp3&#x27;, &#x27;C:\\Users\\BIBRAIN\\AppData\\Local\\Temp\\tmpiqpl57g7&#x27;]&#xA;&#xA;Output from ffmpeg/avlib:&#xA;&#xA;ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers&#xA;  built with gcc 10.2.1 (GCC) 20200726&#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-libgsm --disable-w32threads --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. 51.100 / 56. 51.100&#xA;  libavcodec     58. 91.100 / 58. 91.100&#xA;  libavformat    58. 45.100 / 58. 45.100&#xA;  libavdevice    58. 10.100 / 58. 10.100&#xA;  libavfilter     7. 85.100 /  7. 85.100&#xA;  libswscale      5.  7.100 /  5.  7.100&#xA;  libswresample   3.  7.100 /  3.  7.100&#xA;  libpostproc    55.  7.100 / 55.  7.100&#xA;Guessed Channel Layout for Input Stream #0.0 : stereo&#xA;Input #0, wav, from &#x27;C:\Users\BIBRAIN\AppData\Local\Temp\tmpo20ooz_z&#x27;:&#xA;  Duration: 00:03:54.71, bitrate: 3072 kb/s&#xA;    Stream #0:0: Audio: pcm_s32le ([1][0][0][0] / 0x0001), 48000 Hz, stereo, s32, 3072 kb/s&#xA;Stream mapping:&#xA;  Stream #0:0 -> #0:0 (pcm_s32le (native) -> mp3 (mp3_mf))&#xA;Press [q] to stop, [?] for help&#xA;[mp3_mf @ 00000000004686c0] could not find any MFT for the given media type&#xA;[mp3_mf @ 00000000004686c0] could not create MFT&#xA;Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height&#xA;Conversion failed!&#xA;</module>

  


  I think it is something with codec but I have no idea what to do

  


• kmsgrab : Use GetFB2 if available

  5 juillet 2020, par Mark Thompson

  kmsgrab : Use GetFB2 if available
  The most useful feature here is the ability to automatically extract the
  
  framebuffer format and modifiers.  It also makes support for multi-plane
  
  framebuffers possible, though none are added to the format table in this
  
  patch.
  This requires libdrm 2.4.101 (from April 2020) to build, so it includes a
  
  configure check to allow compatibility with existing distributions.  Even
  
  with libdrm support, it still won't do anything at runtime if you are
  
  running Linux < 5.7 (before June 2020).
  

  • [DH] configure
  • [DH] libavdevice/kmsgrab.c

• FFMPEG Transcode VP8 to H264 from rtp stream

  5 août 2020, par Akil

  I have a rtp stream, the server is receiving audio and video on 2 separate ports, the video is in VP8 and the audio is in Opus.

  


  My ultimate goal is to convert the RTP stream to RTMP to stream to Youtube Live, but Youtube Live supports only H264 https://developers.google.com/youtube/v3/live/guides/ingestion-protocol-comparison so first i'm looking to transcode my RTP stream to H264.

  


  I've run the below command

  


  ffmpeg -analyzeduration 300M -probesize 300M -protocol_whitelist file,udp,rtp -i test.sdp -c:v libx264  -pix_fmt yuv420p -r 25 -c:a aac -f flv youtube_rtmp_url


  


  My sdp file

  


  v=0
o=- 0 0 IN IP4 127.0.0.1
s=RTP Video
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 55.2.100
m=audio 50000 RTP/AVP 111
a=rtpmap:111 OPUS/48000
m=video 50002 RTP/AVP 100
a=rtpmap:100 VP8/90000
a=fmtp:100 packetization-mode=1


  


  Where 50000 and 50002 are the ports which receive the rtp video and audio.

  


  Log output :

  


  ffmpeg version 4.3-3ubuntu1~18.04.sav0 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version='3ubuntu1~18.04.sav0' --toolchain=hardened 
--libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 
--enable-gpl --disable-stripping --enable-avresample --disable-filter=resample 
--enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray 
--enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d 
--enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi 
--enable-libgme               
--enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg 
--enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librsvg 
--enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr 
--enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame 
--enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp 
--enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi 
--enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 -- 
enable-pocketsphinx --enable-crystalhd --enable-libmfx --enable-libdc1394 --enable-libdrm -- 
enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 -- 
enable-shared
  libavutil      56. 51.100 / 56. 51.100
  libavcodec     58. 91.100 / 58. 91.100
  libavformat    58. 45.100 / 58. 45.100
  libavdevice    58. 10.100 / 58. 10.100
  libavfilter     7. 85.100 /  7. 85.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  7.100 /  5.  7.100
  libswresample   3.  7.100 /  3.  7.100
  libpostproc    55.  7.100 / 55.  7.100
  [sdp @ 0x5649eef4b8c0] Could not find codec parameters for stream 1 (Video: vp8, yuv420p): 
  unspecified size

  Consider increasing the value for the 'analyzeduration' and 'probesize' options
    Input #0, sdp, from 'test.sdp':
     Metadata:
       title           : RTP Video
       Duration: N/A, start: 0.000000, bitrate: N/A
         Stream #0:0: Audio: opus, 48000 Hz, mono, fltp
         Stream #0:1: Video: vp8, yuv420p, 90k tbr, 90k tbn, 90k tbc
      [rtmp @ 0x5649eefd87c0] Cannot open connection tcp://a.rtmp.youtube.com:1935
      rtmp://a.rtmp.youtube.com/live2/_______: Immediate exit requested


  


  I've increased 'analyzeduration' and 'probesize' values, error doesn't change.