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• avcodec/simple_idct_template : change the idct coefficients so that they match the...

  12 janvier 2014, par Michael Niedermayer

  avcodec/simple_idct_template : change the idct coefficients so that they match the x86 code
  no changes in either standard deviation or PSNR is seen in any of the changed fate
  
  cases
  MSE changes from 0.05012422 to 0.04890000
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavcodec/simple_idct_template.c
  • [DH] tests/ref/fate/prores-422
  • [DH] tests/ref/fate/prores-422_hq
  • [DH] tests/ref/fate/prores-422_lt
  • [DH] tests/ref/fate/prores-422_proxy
  • [DH] tests/ref/fate/prores-alpha
  • [DH] tests/ref/fate/prores-alpha_skip
  • [DH] tests/ref/fate/prores-transparency
  • [DH] tests/ref/fate/prores-transparency_skip
  • [DH] tests/ref/vsynth/vsynth1-dnxhd-720p-10bit
  • [DH] tests/ref/vsynth/vsynth1-prores
  • [DH] tests/ref/vsynth/vsynth1-prores_ks
  • [DH] tests/ref/vsynth/vsynth2-dnxhd-720p-10bit
  • [DH] tests/ref/vsynth/vsynth2-prores
  • [DH] tests/ref/vsynth/vsynth2-prores_ks

• When creating a Xing or Info tag in an MP3, may I use any MP3 header or does it have to match other frames ?

  13 décembre 2019, par Alexis Wilke

  I have a set of bare MP3 files. Bare as in I removed all tags (no ID3, no Xing, no Info) from those files.

  Just before sending one of these files to the client, I want to add an Info tag. All of my files are CBR so we will use an Info tag (no Xing).

  Right now I get the first 4 bytes of the existing MP3 to get the Version (MPEG-1, Layer III), Bitrate, Frequency, Stereo Mode, etc. and thus determine the size of one frame. I create the tag that way, reusing these 4 bytes for the Info tag and determining the size of the frame.

  For those wondering, these 4 bytes may look like this :

  FF FB 78 04

  To me it felt like you are expected to use the exact same first 4 bytes in the Info tag as found in the other audio frames of the MP3, but when using ffmpeg, they stick an Info tag with a hard coded header (wrong bitrate, wrong frequency, etc.)

  My question is : Is ffmpeg really doing it right ? (LAME doesn’t do that) Could I do the same, skipping the load of the first 4 bytes and still have the greater majority of the players out there play my files as expected ?

  Note : since I read these 4 bytes over the network, it would definitely save a lot of time and some bandwidth to not have to load these 4 bytes on a HEAD request. Resources I could use for the GET requests instead...

• How to match ONVIF Camera services with FFMPEG restreaming

  2 janvier 2018, par Milen

  This is the situation,

  I have an IP Camera from which I can view the video associated on an rtsp video streaming.

  I also have a software, developed with ffmpeg which takes the camera video, makes some changes to the image and restreams another video, on another rtsp address. The output address rtsp :// :/stream.

  Finally, I have an NVR device which is configured to record any camera streaming or substreaming video. To make it work, you have to determine the camera IP and an ONVIF port.

  So what I want to do and I dont know how, is to add the camera to the NVR but change the rtsp address. Or maybe, simulate an ONVIF camera service, and associate it with my generated rtsp video. Then, add this ""camera"" to the NVR.

  Any answers that consist in recording the video in any other way than using the NVR, will not be useful for me

  EDIT :

  The first option which consists on adding the camera to the NVR and change the rtsp address didnt work, because the device doesn’t support that. So I want to know how to implement (any language or platform) an ONVIF server which copy the camera service but change the rtsp address.

  I downloaded a virtualbox machine which work as ONVIF server, but still didnt know how to change the protocol so It gives my rtsp address to the NVR, because the code was a very close one.