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• Ideal bitrates for different video resolutions

  15 octobre 2018, par Ramesh Navi

  I am building a Video-on-demand service for a closed community. I using FFMPEG for video processing and dash.js for adaptive bitrate player with custom resolution selector. Can somebody please suggest what ideal bitrates should I use while video/audio transcoding ?

  I am talking about -b:v and -ab option

  ffmpeg -i vid.mp4 -c:v libvpx-vp9 -keyint_min 150 \
  
  -g 150 -tile-columns 4 -frame-parallel 1  -f webm -dash 1 \
  
  -an -vf scale=144:-1 -b:v 120k -dash 1 video_1.webm \
  
  -an -vf scale=240:-1 -b:v 250k -dash 1 video_2.webm \
  
  -an -vf scale=360:-1 -b:v 500k -dash 1 video_3.webm \
  
  -an -vf scale=480:-1 -b:v 750k -dash 1 video_4.webm \
  
  -an -vf scale=720:-1 -b:v 1500k -dash 1 video_5.webm

  And

  ffmpeg -i vid.mp4 -vn -acodec libvorbis -ab 96k -dash 1 audio_96k.webm

  Any suggestions/hacks or examples to tackle real-world network situations are appreciated.

• How play AVI with sound by OpenCV on Win32

  4 janvier 2015, par Andrei Shumilov

  I need play AVI by OpenCV and any sound player (vfw/ffmpeg...) on Win32. It seems to me this is very secret or useless thing ’cos I found only one sample in the world : Audio output with video processing with opencv

  Mr. Karl Phillip wrote there :
  "On my Mac I compiled it with :
  g++ ffmpeg_snd.cpp -o ffmpeg_snd -D_GNU_SOURCE=1 -D_THREAD_SAFE -I/usr/local/include/opencv -I/usr/local/include -I/usr/local/include/SDL -Wl,-framework,Cocoa ...."

  But I must use MSVS2010 (I have working project). Ok, I installed mingw to try, but still don’t know what could replace "framework,Cocoa".

  Give me please links to WORKING examples OpenCV+anysoundplayer on Win32 or help me port Karl’s example at least.

• Transcode HLS Segments individually using FFMPEG

  27 mai 2013, par rayh

  I am recording a continuous, live stream to a high-bitrate HLS stream. I then want to asynchronously transcode this to different formats/bitrates. I have this working, mostly, except audio artefacts are appearing between each segment (gaps and pops).

  Here is an example ffmpeg command line :

  ffmpeg -threads 1 -nostdin -loglevel verbose \
  
     -nostdin -y -i input.ts -c:a libfdk_aac \
  
     -ac 2 -b:a 64k -y -metadata -vn output.ts

  Inspecting an example sound file shows that there is a gap at the end of the audio :

  

  And the start of the file looks suspiciously attenuated (although this may not be an issue) :

  

  My suspicion is that these artefacts are happening because transcoding are occurring without the context of the stream as a whole.

  Any ideas on how to convince FFMPEG to produce audio that will fit back into a HLS stream ?

  ** UPDATE 1 **

  Here are the start/end of the original segment. As you can see, the start still appears the same, but the end is cleanly ended at 30s. I expect some degree of padding with lossy encoding, but I there is some way that HLS manages to do gapless playback (is this related to iTunes method with custom metadata ?)

  
  

  ** UPDATED 2 **

  So, I converted both the original (128k aac in MPEG2 TS) and the transcoded (64k aac in aac/adts container) to WAV and put the two side-by-side. This is the result :

  
  

  I'm not sure if this is representative of how a client will play it back, but it seems a bit odd that decoding the transcoded one introduces a gap at the start and makes the segment longer. Given they are both lossy encoding, I would have expected padding to be equally present in both (if at all).

  ** UPDATE 3 **

  According to http://en.wikipedia.org/wiki/Gapless_playback - Only a handful of encoders support gapless - for MP3, I've switched to lame in ffmpeg, and the problem, so far, appears to have gone.

  For AAC (see http://en.wikipedia.org/wiki/FAAC), I have tried libfaac (as opposed to libfdk_aac) and it also seems to produce gapless audio. However, the quality of the latter isn't that great and I'd rather use libfdk_aac is possible.