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• Use ffmpeg/avconv to re-stram a combined stream ?

  28 novembre 2014, par Terminello

  Having two separate stream sources (audio and video) each from an IP-camera, I would like to

  • combine them into a single stream
  • save this into a file (using segmentation)
  • re-stream (specific word unknown for me, sorry) the combined stream so that other users may access it without having to combine the streams on their own

  The first two things : I have already accomplished like that :

  ffmpeg -use_wallclock_as_timestamps 1 -f mjpeg -i 'http://camera.ip.address/videostream.cgi?user=username&pwd=password' -i 'http://camera.ip.address/videostream.asf?user=username&pwd=password' -map 0:v -map 1:a -acodec copy -vcodec copy -f segment -segment_time 3600 -reset_timestamps 1 'out%03d.mkv'

  Credits go to santiago (foscam.com forums)

  But how to make this stream available on the LAN at the same time while being recorded and saved to HDD ?

  BTW : You might ask, why do I need this : Because many clients do not have capabilitys to combine a stream from multiple sources, so I set up a server which does this task and clients do not go directly to the audio/video source (IP-camera) but catch the newly generated stram = : the idea.

• find matching files using a.bat script and assign them to a variable ?

  2 octobre 2019, par A Person

  I am trying to assign a file to a variable in batch-file and then also assign anoter 2 files into anoter variable.

  However, i am having an issue.

  From research, i found how i can do the assigning but does anyone know how i can do the below.

  From a folder or text file (either is fine) find the .m2v video file and assign that to Var1 then find matching audio in .wav and put that in Var2 and the third is also an audio .wav with mathcing name and assign that to Var3.

  Problem i am having is that trying to find the matching 2 audio file to the video.

  The video file is named as PAV_PRG_13683Highc450277201906251802090353.m2v

  Audio 1 is : PAV_PRG_13683High01c450211201906251802090376.wav

  Audio 2 is : PAV_PRG_13683High00c450211201906251802090368.wav

  the file name matches until it sees the word High. Everything after High is not needed and is random string so trying to match is an issue.

  is there a way to find the match by comparing everything before High.

  Also as i will be using the variable and putting them through ffmpeg to merge, is there way to do it so that when the ffmpeg command is done, it moves to the next matching files and assigns them to the variable.

  files are store in 2 folders

  One folder has all the video files *.m2v

  and another folder has all the *.wav audio files in pairs of 2. Each video has exactly 2 audios (left right audio)

  is there any hel pon this subject, i have already come up empty in my research and have been checking for this over the last week spent almost 30 hours.

• 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.