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• TCP Connection refused error when using FFMPEG for audio stream to HTTP on macOS

  26 novembre 2020, par freddy

  I'm trying to stream my microphone input via HTTP using ffmpeg, so I can stream it in HTML. I run the following ffmpeg command :

  


  ffmpeg -f avfoundation -i ":1" -c:a libmp3lame -f mp3 -r 30 http://localhost:809

  


  It, however, crashes with the following error message :

  


  ffmpeg version 4.3.1 Copyright (c) 2000-2020 the FFmpeg developers
  built with Apple clang version 12.0.0 (clang-1200.0.32.27)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.3.1_4 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libdav1d --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-libsoxr --enable-videotoolbox --disable-libjack --disable-indev=jack
  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
Input #0, avfoundation, from ':1':
  Duration: N/A, start: 3445.340045, bitrate: 22579 kb/s
    Stream #0:0: Audio: pcm_f32le, 44100 Hz, hexadecagonal, flt, 22579 kb/s
[tcp @ 0x7fa46ec96600] Connection to tcp://localhost:8090 failed: Connection refused
http://localhost:8090: Connection refused


  


  I've had success with streaming on that port using VLC, but it for some reason won't work using ffmpeg. Any ideas on how to fix this ?

  


• avformat : add DFPWM WAV container support

  8 mars 2022, par Jack Bruienne

  avformat : add DFPWM WAV container support
  This commit adds support for storing DFPWM audio in a WAV container.
  
  It uses the WAVEFORMATEXTENSIBLE structure, following these conventions :
  
  https://gist.github.com/MCJack123/90c24b64c8e626c7f130b57e9800962c
  
  The implementation is very simple : it just adds the GUID to the list of
  
  WAV GUIDs, and modifies the WAV muxer to always use WAVEFORMATEXTENSIBLE
  
  format with that GUID.
  This creates a standard container format for DFPWM besides raw data.
  
  It will allow users to transfer DFPWM audio in a standard container
  
  format, with the sample rate and channel count contained in the file
  
  as opposed to being an external parameter as in the raw format.
  This format is already supported in my AUKit library, which is the CC
  
  analog to libav (albeit much smaller). Support in other applications is TBD.
  Signed-off-by : Jack Bruienne <jackbruienne@gmail.com>
  

  • [DH] libavformat/riff.c
  • [DH] libavformat/riffenc.c
  • [DH] libavformat/version.h

• ffmpeg : improving MP4 to webm ogg conversions

  14 juillet 2017, par Randy

  (Edited to include some of the things I’ve tried)

  I’m a musician, and occasional web coder. I’ve been using video editing software (old version of Roxio Videowave from 2011) to build promotional videos from clips of some of my performances, and I’d like to put some of them on my own web pages in HTML5 video format. So that currently means I need MP4, WEBM, and OGG conversions. Fortunately the editing software churns out some very nice MP4 (H264) files, and has plenty of options for doing so. I purposely output the output size about 2X the likely display size, in hopes of offering more detail for better conversions. Specifically, the video output was AVC/H.264, 800 x 450, 30fps, variable bit rate, but with 600000 as a base line (that was the default for this setting anyway).

  Now I’m nowhere near expert at this stuff, and I probably left out some important data. But bottom line, the resulting MP4 looked very good. Unfortunately, to put it on my own web page means at least converting to WEBM and OGG formats. It would be nice if all browsers just supported MP4, but then there would be licensing fees, so conversions are needed. Sadly, I’ve been wasting days now trying to do this with ffmpeg. Its easy to do, its doing it WELL that is a mystery to me. Just letting ffmpeg work using its defaults (meaning I just specify an input and output file) results in pretty terrible video. But I’ve also tried most of the settings for better quality available, and the resulting conversions are nowhere near as good as youtube’s conversions.

  Based on the info about my original MP4 file, can someone suggest some better settings for ffmpeg conversions to WEBM and OGG ? Am I going about this all wrong ? The best I’ve done so far was with a string like this, which specified a high quality and a fairly robust bit rate...

  ffmpeg -i input-file.mp4 -c:v libvpx -crf 10 -b:v 1M -c:a libvorbis output-file.webm

  That was much better than the default settings, but still nowhere near the quality of YOUTUBE conversions. In my resulting WEBM video, you can pretty plainly see how the picture degrades, and will snap into focus every few seconds when a "key frame" comes up. These artifacts should not be so obvious. Thanks for any help.