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• Is there a way to use ffmpeg audio filters to automatically synchronize 2 streams with similar content

  29 mai 2015, par user3741412

  I have a situation where I have a video capture of HD content via HDMI with audio from a sound board that goes through a impedance drop into a microphone input of a camcorder. That same signal is split at line level to a ’line in’ jack on the same computer that is capturing the HDMI. Alternatively I can capture the audio via USB from the soundboard which is probably the best plan, but carries with it the same issue.

  The point is that the line in or usb capture will be much higher quality than the one on HDMI because the line out -> impedance change -> mic in path generates inferior quality in that simply brushing the mic jack on the camera while trying to change the zoom (close proximity) can cause noise on the recording.

  So I can do this today :

  • Take the good sound and the camera captured sound and load each into
    audacity and pretty quickly use the timeshift toot to perfectly fit
    the good audio to the questionable audio from the HDMI capture and
    cut the good audio to the exact size of the video. Then I can use
    ffmpeg or other video editing software to replace the questionable
    audio with the better audio.

  But while somewhat quick and easy, it always carries with it a bit of human error and time. I’d like to automate this if possible as this process is repeated at least weekly throughout the year.

  Does anyone have a suggestion if any of these ideas have merit or could suggest another approach ?

  1. I suspect but have yet to confirm that the system timestamp of the start time may be recorded in both audio captured with something like Audacity, or the USB capture tool from the sound board as well as the HDMI mpeg-2 video. I tried ffprobe on a couple audacity captured .wav files but didn’t see anything in the results about such a time code, but perhaps other audio formats or other probing tools may include this info. Can anyone advise if this is common with any particular capture tools or file formats ?

     • if so, I think I could get best results by extracting this information and then using simple adelay and atrim filters in ffmpeg to sync reliably directly from the two sources in one ffmpeg call. This is all theoretical for me right now— I’ve never tried either of these filters yet— just trying to optimize against blind alleys by asking for advice up front.

  2. If such timestamps are not embedded, possibly I can use the file system timestamp for the same idea expressed in 1a, but I suspect the file open of the two capture tools may have different inherant delays. Possibly these delays will be found to be nearly constant and the approach can work with a built-in constant anticipation delay but sounds messy and less reliable than idea 1. Still, I’d take it, if it turns out reasonably reliable

  3. Are there any ffmpeg or general digital audio experts out there that know of particular filters that can be employed on the actual data to look for similarities like normalizing the peak amplitudes or normalizing the amplification of the two to some RMS value and then stepping through a short 10 second snippet of audio, moving one time stream .01s left against the other repeatedly and subtracting the two and looking for a minimum ? Sounds like it could take a while, but if it could do this in less than a minute and be reliable, I suspect it could work. But I have only rudimentary knowledge of audio streams and perhaps what I suggest is just not plausible— but since each stream starts with the same source I think there should be a chance. I am just way out of my depth as to how to go down this road, so if someone out there knows such magic or can throw me some names of filters and example calls, I can explore if I can make it work.

  4. any hardware level suggestions to take a line level output down to a mic level input and not have the problems I am seeing using a simple in-line impedance drop module, so that I can simply rely on the audio from the HDMI ?

  Thanks in advance for any pointers or suggestinons !

• Capturing PCM audio data stream into file, and playing stream via ffmpeg, how ?

  11 avril 2015, par icarus74

  Would like to do following four things (separately), and need a bit of help understanding how to approach this,

  1. Dump audio data (from a serial-over-USB port), encoded as PCM, 16-bit, 8kHz, little-endian, into a file (plain binary data dump, not into any container format). Can this approach be used :

     $ cat /dev/ttyUSB0 > somefile.dat

  Can I do a ^C to close the file writing, while the dumping is in progress, as per the above command ?

  2. Stream audio data (same as above described kind), directly into ffmpeg for it to play out ? Like this :

     $ cat /dev/ttyUSB0 | ffmpeg

  or, do I have to specify the device port as a "-source" ? If so, I couldn’t figure out the format.

  Note that, I’ve tried this,

  $ cat /dev/urandom | aplay

  which works as expected, by playing out white-noise..., but trying the following doesn’t help :

  $ cat /dev/ttyUSB1 | aplay -f S16_LE

  Even though, opening /dev/ttyUSB1 using picocom @ 115200bps, 8-bit, no parity, I do see gibbrish, indicating presence of audio data, exactly when I expect.

  3. Use the audio data dumped into the file, use as a source in ffmpeg ? If so how, because so far I get the impression that ffmpeg can read a file in standard containers.

  4. Use pre-recorded audio captured in any format (perhaps .mp3 or .wav) to be streamed by ffmpeg, into /dev/ttyUSB0 device. Should I be using this as a "-sink" parameter, or pipe into it or redirect into it ? Also, is it possible that in 2 terminal windows, I use ffmpeg to capture and transmit audio data from/into same device /dev/ttyUSB0, simultaneously ?

  My knowledge of digital audio recording/processing formats, codecs is somewhat limited, so not sure if what I am trying to do qualifies as working with ’raw’ audio or not ?

  If ffmpeg is unable to do what I am hoping to achieve, could gstreamer be the solution ?

  PS> If anyone thinks that the answer could be improved, please feel free to suggest specific points. Would be happy to add any detail requested, provided I have the information.

• tag refuses to play mp4 video

  5 janvier 2015, par MightyPork

  I have a video I want to embed using the html5 video tag :

  • https://dl.dropboxusercontent.com/u/64454818/TMP/P6135199.MOV

  I’ve converted it using ffmpeg :

  ffmpeg -i P6135199.MOV -vcodec libx264 -acodec aac helios.mp4

  FFMPEG output :

  [libx264 @ 0x22ac340] using cpu capabilities : MMX2 SSE2Fast SSSE3 SSE4.2
  [libx264 @ 0x22ac340] profile High 4:2:2, level 1.2, 4:2:2 8-bit
  [libx264 @ 0x22ac340] 264 - core 142 r2455 021c0dc - H.264/MPEG-4 AVC codec - Copyleft 2003-2014 - http://www.videolan.org/x264.html - options : cabac=1 ref=3 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=7 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=2 b_adapt=1 b_bias=0 direct=1 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=15 scenecut=40 intra_refresh=0 rc_lookahead=40 rc=crf mbtree=1 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00
  Output #0, mp4, to ’helios.mp4’ :
    Metadata :
      comment-eng : OLYMPUS DIGITAL CAMERA                           
      comment : OLYMPUS DIGITAL CAMERA                           
      encoder : Lavf56.15.102
      Stream #0:0(eng) : Video : h264 (libx264) ([33][0][0][0] / 0x0021), yuvj422p(pc), 320x240, q=-1—1, 15 fps, 15360 tbn, 15 tbc (default)
      Metadata :
        creation_time : 2008-06-13 10:47:16
        encoder : Lavc56.13.100 libx264
  Stream mapping :
    Stream #0:0 -> #0:0 (mjpeg (native) -> h264 (libx264))
  Press [q] to stop, [?] for help
  frame=  240 fps=124 q=-1.0 Lsize=     906kB time=00:00:15.86 bitrate= 467.7kbits/s    
  video:902kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead : 0.409100%
  [libx264 @ 0x22ac340] frame I:1     Avg QP:23.71  size :  7960
  [libx264 @ 0x22ac340] frame P:123   Avg QP:23.85  size :  5255
  [libx264 @ 0x22ac340] frame B:116   Avg QP:25.41  size :  2317
  [libx264 @ 0x22ac340] consecutive B-frames :  3.3% 96.7%  0.0%  0.0%
  [libx264 @ 0x22ac340] mb I  I16..4 : 15.7% 69.0% 15.3%
  [libx264 @ 0x22ac340] mb P  I16..4 :  2.2% 13.4%  0.8%  P16..4 : 43.8% 23.1% 14.0%  0.0%  0.0%    skip : 2.8%
  [libx264 @ 0x22ac340] mb B  I16..4 :  0.3%  1.3%  0.1%  B16..8 : 42.4%  9.9%  3.0%  direct:12.2%  skip:30.9%  L0:37.2% L1:38.2% BI:24.5%
  [libx264 @ 0x22ac340] 8x8 transform intra:80.8% inter:71.9%
  [libx264 @ 0x22ac340] coded y,uvDC,uvAC intra : 76.7% 96.8% 49.7% inter : 37.8% 60.9% 5.1%
  [libx264 @ 0x22ac340] i16 v,h,dc,p : 39%  4%  8% 49%
  [libx264 @ 0x22ac340] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu : 15% 14% 39%  6%  3%  4%  3%  6% 12%
  [libx264 @ 0x22ac340] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu : 25% 11% 18%  8%  7%  7%  8%  8%  8%
  [libx264 @ 0x22ac340] i8c dc,h,v,p : 63%  7% 20% 10%
  [libx264 @ 0x22ac340] Weighted P-Frames : Y:5.7% UV:0.8%
  [libx264 @ 0x22ac340] ref P L0 : 51.4% 18.3% 20.6%  9.3%  0.4%
  [libx264 @ 0x22ac340] ref B L0 : 80.8% 19.2%
  [libx264 @ 0x22ac340] kb/s:461.52
  

  And I try to embed it as follows :

  <video src="helios.mp4" controls="controls">Get a better browser!</video>

  However, Chrome doesn’t play the video, only offers to download it (same as it did before I converted the video). Firefox has the same problem.

  I’ve tested and other mp4 files (from my phone) play just fine.

  What’s the problem ? Did I convert it wrong ?