Recherche avancée

Sur d’autres sites (6342)

• Mixing audio stream into video stream using ffmpeg while retaining original audio from the video stream as background [duplicate]

  18 mars 2020, par Core7s

  I have a live video stream and a live audio stream. The audio stream has to get mixed into the video stream and the resulting stream is pushed to YouTube. I have this working ffmpeg -i videostream -i audiostream -map 0:1 -map 1:0 -shortest -c:a copy -c:v copy -f flv streamout-youtube;
  However I am losing the original audio in the videostream. I want to have the original audio playing at maybe 20% of the original volume.
  I looked around a lot and found maybe I can modify the above command to use -filter_complex to get ffmpeg -i videostream -i audiostream -filter_complex "volume=0.2" -map 0:1 -map 1:0 -shortest -c:a copy -c:v copy -f flv streamout-youtube;

  However, when I add the -filter_complex option, the stream goes dead meaning it doesn’t get pushed to youtube anymore. I have verified the stream keys are correct. This seems to be a rather simple thing but I can’t put my finger on what’s wrong.

  I am using rtmp module of nginx to receive, mix and push out streams.

• dnxhddec : Decode and use interlace mb flag

  26 septembre 2015, par Christophe Gisquet

  dnxhddec : Decode and use interlace mb flag
  This bit is 1 in some samples, and seems to coincide with interlaced
  
  mbs and CID1260. 2008 specs do not know about it, and maintain qscale
  
  is 11 bits. This looks oversized, but may help larger bitdepths.
  Currently, it leads to an obviously incorrect qscale value, meaning
  
  its syntax is shifted by 1. However, reading 11 bits also leads to
  
  obviously incorrect decoding : qscale seems to be 10 bits.
  However, as most profiles still have 11bits qscale, the feature is
  
  restricted to the CID1260 profile (this flag is dependent on
  
  a higher-level flag located in the header).
  The encoder writes 12 bits of syntax, last and first bits always 0,
  
  which is now somewhat inconsistent with the decoder, but ends up with
  
  the same effect (progressive + reserved bit).
  Signed-off-by : Vittorio Giovara <vittorio.giovara@gmail.com>
  

  • [DH] libavcodec/dnxhddec.c

• Not outputting Opus raw audio

  3 mars 2016, par Dustin

  I’m currently writing a small script that coverts an MP4 to Opus audio on the fly and sends it to Discord in golang. Initially my script would pass an MP4 as it was downloading to ffmpeg through stdin and then pass stdout to an Opus encoder, then to Discord (exactly like this). After learning I could build ffmpeg with Opus, I’d like to cut out the opus encoder I previous had and pass ffmpeg’s output directly to Discord.

  Previous, my ffmpeg command looked like this (with using the second opus encoder)

  ffmpeg -i - -f s16le -ar 48000 -ac 2 pipe:1

  Now, without the encoder and letting ffmpeg do all the work, this is what I’ve come up with so far.

  ffmpeg -i - -f s16le -ar 48000 -ac 2 -acodec libopus -b:a 192k -vbr on -compression_level 10 pipe:1

  With this command however the audio doesn’t get accepted by Discord’s server, meaning I’m suspecting opus audio isn’t coming out the other end. No errors outputted. Have I done something wrong with ffmpeg that could of caused this ?