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• Can I use ffmpeg to create multi-bitrate (MBR) MPEG-4 videos ?

  5 décembre 2011, par hoangbv15

  I am currently in a webcam streaming server project that requires the function of dynamically adjusting the stream's bitrate according to the client's settings (screen sizes, processing power...) or the network bandwidth. The encoder is ffmpeg, since it's free and open sourced, and the codec is MPEG-4 part 2. We use live555 for the server part.

  How can I encode MBR MPEG-4 videos using ffmpeg to achieve this ?

• Handbreak ffmpeg encode vs decode speed (x265)

  10 février 2023, par Demian

  I'm encoding videos in x265 to save same storage (Gopro, DJI, etc.). This is working well for me with Handbreak and I got some good CRF values for my use cases.

  


  My question is about the encoding preset. The slower it is the better Qualitiy per Bitrate you get, but the encoding time increases.

  


  But how the decoding time. Does it also consume more CPU power to watch (decode) a video which was created with a slow prestet then a fast preset ?

  


  Thank you !

  


• 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.