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• How to split a 1 hour video .mp4 file into 5 seconds segments for Media Source Extension API

  15 avril 2018, par Harpreet Singh

  I’m designing a web media player with the Media Source Extension API. I really like the process of how dash.js distributes the media by having the original media split into number of small files and be played in the web. I have visited the source storage of the bbb_30fps at https://dash.akamaized.net/akamai/bbb_30fps. And I looked at the 720p video quality directory at https://dash.akamaized.net/akamai/bbb_30fps/bbb_30fps_1280x720_4000k and the audio directory at https://dash.akamaized.net/akamai/bbb_30fps/bbb_a64k. I saw that the original 10 min video has been split into small segments of 720p quality excluding the audio with the file type of .m4v and small segments of the audio excluding the video with the file type of m4a.

  I want to have my 1 hour .mp4 video split into 720p quality with separate segments of duration 5 seconds excluding the audio and with the file type of .m4v. I also want to split the audio into separate segments of 5 seconds excluding the video with the file type of .m4a. If someone can help me please comment below. Thanks

  So far I have visited http://www.instructables.com/id/Making-Your-Own-Simple-DASH-MPEG-Server-Windows-10/ but it want not able to help me with my needs. And I have tried number of other methods with MP4Box but I can’t get the result I want.

• How can I play an MPG file using Electron on windows without converting it ? [closed]

  1er novembre 2024, par Indy Singh

  I am trying to play MPG files in an Electron app on Windows. The files contain mpeg1 encoded video. I am using the video element. I can play MP4 files, but the MPG files do not play. One requirement is to be able to play the video from some arbitrary start time.

  


  Electron uses the Chrome engine to render output. Google Chrome app also cannot play MPG files.

  


  Electron uses ffmpeg.dll to play the video file. As far as I can tell ffmpeg.dll contains the mpeg1 decoder, but the video element does not appear to make use of it.

  


  I know I can convert the video before playing, however conversion takes several minutes and is not an ideal user experience. As a fallback I can convert small segments as needed, which still takes a bit of time, so it is not ideal.

  


  Streaming the video from a server process with on-the-fly conversion is possible, but I believe it may not work well with the need to seek. I could be wrong here.

  


  Any other suggestions would be welcome.

  


  EDIT 1 :
I am developing an Electron App using the programming languages TypeScript, JavaScript and HTML5 and the tools Nodejs and Electron. I did not post a code snippet as I am looking for an approach, not a fix for code. Not sure why this question got flagged as being 'not about programming'.

  


  A Google search does not show many options other than those mentioned below.

  


  EDIT 2 :
Thanks to the suggestion from VC.One I am looking at the JSMPEG library. It does look like it could possibly do the job but with 'numerous limitations'. It also say the other alternative called Broadway is 'huge, compiled with emscripten and quite complicated'

  


  EDIT 3 :
I am now looking hls.js, which I think will likely work for my needs. It integrates well with using the html5 video element. I can use it with on-demand decoding of the MPG MPEG1 video file in small segments.

  


• armv6 : Accelerate ff_imdct_half for general case (mdct_bits != 6)

  11 juillet 2014, par Ben Avison

  armv6 : Accelerate ff_imdct_half for general case (mdct_bits != 6)
  The previous implementation targeted DTS Coherent Acoustics, which only
  
  requires mdct_bits == 6. This relatively small size lent itself to
  
  unrolling the loops a small number of times, and encoding offsets
  
  calculated at assembly time within the load/store instructions of each
  
  iteration.
  In the more general case (codecs such as AAC and AC3) much larger arrays
  
  are used - mdct_bits == [8, 9, 11]. The old method does not scale for
  
  these cases, so more integer registers are used with non-unrolled versions
  
  of the loops (and with some stack spillage). The postrotation filter loop
  
  is still unrolled by a factor of 2 to permit the double-buffering of some
  
  VFP registers to facilitate overlap of neighbouring iterations.
  I benchmarked the result by measuring the number of gperftools samples
  
  that hit anywhere in the AAC decoder (starting from aac_decode_frame())
  
  or specifically in ff_imdct_half_c / ff_imdct_half_vfp, for the same
  
  example AAC stream :
                    Before          After
  
                    Mean   StdDev   Mean   StdDev  Confidence  Change
  
  aac_decode_frame  2368.1 35.8     2117.2 35.3    100.0%      +11.8%
  
  ff_imdct_half_*   457.5  22.4     251.2  16.2    100.0%      +82.1%
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavcodec/arm/mdct_vfp.S