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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 !

• Files dissapearing with ffmpeg recursive conversion

  13 août 2014, par CaRoXo

  I started in askubuntu, asking for a way to convert recursively more than 14K of wma to mp3 extracting the wma files path from a txt file.
  There was an answer that could cover my needs, but a bug appears. The first run with some hundreds worked ok. The second, some wma albums got converted, others entirely deleted. There were some modifications. And last time completely, deleted all wma without converting.

  this was the original script

  #!/usr/bin/env bash
  
  
  
  readarray -t files < wma-files.txt
  
  
  
  for file in "${files[@]}"; do
  
      out=`echo $file | sed "s:wma:mp3:"`
  
      probe=`avprobe -show_streams "$file" 2>/dev/null`
  
      rate=`echo "$probe" | grep "bit_rate" | sed "s:.*=\(.*\)[0-9][0-9][0-9][.].*:\1:"`
  
      avconv -i "$file" -ab "$rate"k "$out"
  
      rm "$file"
  
  done

  Then the adaptation with ffmpeg

  #!/usr/bin/env bash
  
  
  
  readarray -t files < wma-files.txt
  
  
  
  for file in "${files[@]}"; do
  
      out=`echo $file | sed "s:wma:mp3:"`
  
      probe=`avprobe -show_streams "$file" 2>/dev/null`
  
      rate=`echo "$probe" | grep "bit_rate" | sed "s:.*=\(.*\)[0-9][0-9][0-9][.].*:\1:"`
  
      ffmpeg -i "$file" -ab "$rate"k "$out" && rm "$file"
  
  done

  With the first one I converted many files. Other just get deleted. The ones deleted were always the same release (so, all tracks from a release). I can listen, and even convert them with Soundkonverter.

  Both of them produces "no such file of directory" and when this happens, everything get deleted.

  The partition where files are stored is a usb HDD ntfs, but also happens in my ext4 internal HD.
  Im under Xubuntu 14.04

  Here the script running with avconv (wich what i managed to convert some, but other get dissapeared) http://pastebin.com/w5weqEws and with ffmpeg (that didn’t convert any) http://pastebin.com/3QkaPzvW

  I can’t find differences between successfully and deleted original wma’s. But for example, while other progs like beets read and write the tags, puddletag and mp3tag (under wine) don’t, until I converted them with soundkonverter.

  As the person trying to help me there redirect me here on the original post http://askubuntu.com/questions/508278/how-to-use-ffmpeg-to-convert-wma-to-mp3-recursively-importing-from-txt-file/508304#508304
  Im here asking for any help to make run an script like this. Or any to use ffmpeg to convert recursively the audio files. My capacity of understanding is short for being able to make something working just reading the docs.

  So I ask a help to run this. If I miss any relevant information, just tell me.

  NOTE : I want to add that doing the conversion with

  for file in "${files[@]}"; do
  
      out=`echo "$file" | sed s:wma:mp3:`
  
      avconv -i "$file" -ab 192k "$out"
  
      rm "$file"
  
  done

  It works in the same files (the ones that are deleted with the other). Only that it makes everything to 192k, so not good if Im converting lower bitrate ones. And get this error "Application provided invalid, non monotonically increasing dts to muxer in stream 0" that seems something typical from avconv in 14.04. With ffmpeg I cant try becouse I don’t find the way how to use it, even out of the script. I really don’t understand the docs seems
  .

  NOTE2 : This is a mediainfo exit from :

  1- A typical wma that get dissapeared always with the script

  Audio
  
  ID                                       : 1
  
  Format                                   : WMA
  
  Format version                           : Version 2
  
  Codec ID                                 : 161
  
  Codec ID/Info                            : Windows Media Audio
  
  Description of the codec                 : Windows Media Audio 9 - 128 kbps, 44 kHz, stereo 1-pass CBR
  
  Duration                                 : 2mn 25s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 44.1 KHz
  
  Bit depth                                : 16 bits
  
  Stream size                              : 2.21 MiB (99%)
  
  Language                                 : English (US)

  2- A Wma that got succesfully converted (yes Im using copies now, I cant risk specially some rares audios that I got on the road)

  Audio
  
  ID                                       : 1
  
  Format                                   : WMA
  
  Format version                           : Version 2
  
  Codec ID                                 : 161
  
  Codec ID/Info                            : Windows Media Audio
  
  Description of the codec                 : Windows Media Audio 9 - 128 kbps, 44 kHz, stereo 1-pass CBR
  
  Duration                                 : 4mn 35s
  
  Bit rate mode                            : Constant
  
  Bit rate                                 : 128 Kbps
  
  Channel(s)                               : 2 channels
  
  Sampling rate                            : 44.1 KHz
  
  Bit depth                                : 16 bits
  
  Stream size                              : 4.21 MiB (99%)
  
  Language                                 : English (US)

  So, as I don’t see difference, but maybe, I’m losing any data to look into ?

• Using Qt Media Player on Raspberry Pi 1

  18 mai 2015, par Mauker

  I have a project built using Qt5 which has to play a video. Just like in the videowidget sample code.

  I’ve followed these instructions to build qt5 on my Pi. And it went just fine. But when I try to run any qt program that uses QMediaPlayer, I get the error message :

  defaultServiceProvider::requestService(): no service found for - "org.qt-project.qt.mediaplayer"

  Which means I don’t have a backend to play the video, right ? Is there any one that I can use which will with Qt, like ffmpeg ? And how can I get it to work ? Specially for h264 videos.

  I’ve tried to install gstreamer as is told on this link, but it’s not working. Will I have to rebuild the entire qt5 again ?

  P.S. : I have the raspberry pi 1 model B with raspbian installed.

  Edit : As mentioned by Greenflow, I checked the ./configure log and saw that the GStreamer was compiled in, but the video apps are still not working...

  The message on the log was like this :

  GStreamer .............. yes (0.10)

  And the message on Greenflow’s log was like this :

  GStreamer .............. yes (1.0)

  Clearly it’s another version of GStreamer, but is it the problem ?

  I’ve also found this post which says QtMultimedia on the Pi is rather useless, but the post is from 2013, so I’m not sure if it’s really relevant. I’d like to have this app playing hardware accelerated videos on my Raspberry Pi, but I’m almost dropping the idea.

  Anyways, thanks Greenflow for the head start.

  Edit 2 : Found this thread on the Qtcentre. Damn, this thing is not going to be easy to solve, I guess...