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• swscale/aarch64 : add vscale specializations

  13 août 2022, par Swinney, Jonathan

  swscale/aarch64 : add vscale specializations
  This commit adds new code paths for vscale when filterSize is 2, 4, or
  
  8. By using specialized code with unrolling to match the filterSize we
  
  can improve performance.
  On AWS c7g (Graviton 3, Neoverse V1) instances :
  
                                   before   after
  
  yuv2yuvX_2_0_512_accurate_neon :  558.8    268.9
  
  yuv2yuvX_4_0_512_accurate_neon :  637.5    434.9
  
  yuv2yuvX_8_0_512_accurate_neon :  1144.8   806.2
  
  yuv2yuvX_16_0_512_accurate_neon : 2080.5   1853.7
  Signed-off-by : Jonathan Swinney <jswinney@amazon.com>
  
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libswscale/aarch64/output.S
  • [DH] libswscale/aarch64/swscale.c

• Merging input Streams with nodejs/ffmpeg

  14 septembre 2020, par jAndy

  I'm creating a very basic and rudimentary Video-Web-Chat. On the client side, I'm going to use a simple getUserMedia API call to capture the webcam data and send video-data as data-blob to my server.

  


  From There, I'm planning to either use the fluent-ffmpeg library or just spawn ffmpeg myself and pipe that raw data to ffmpeg, which in turn, does some magic and pushes that out as HLS stream to an Amazon AWS Service (for instance), which then gets actually displayed on a Web Browser for all participating people in the video chat.

  


  So far, I think all of this should be fairly easy to implement, but I keep my head spinning around the question, how I can create a "combined" or "merged" frame and stream, so the output HLS data from my server to the distributing cloud service has only to be one combined data stream to receive.

  


  If there are 3 people in that video chat, my server receives 3 data streams from those clients and combines these data streams (from the individual web-cam data sources) into one output stream.

  


  How could that be accomplished ?
Can I "create" a new frame with ffmpeg, so to speak ? I would be very thankful if anybody could give me a heads up here, maybe I'm thinking in a complete wrong direction.

  


  Another question which arises to me is, if I really can just "dump" any data, which I'm receiving from a binary blob created from getUserMedia or MultiStreamRecorder to ffmpeg or if I have to specify somewhere and somehow the exact codecs being used etc.?

  


• Transcoding fMP4 to HLS while writing on iOS using FFmpeg

  12 juillet 2017, par bclymer

  TL ;DR

  I want to convert fMP4 fragments to TS segments (for HLS) as the fragments are being written using FFmpeg on an iOS device.

  Why ?

  I’m trying to achieve live uploading on iOS while maintaining a seamless, HD copy locally.

  What I’ve tried

  1. Rolling AVAssetWriters where each writes for 8 seconds, then concatenating the MP4s together via FFmpeg.

     What went wrong - There are blips in the audio and video at times. I’ve identified 3 reasons for this.

     1) Priming frames for audio written by the AAC encoder creating gaps.

     2) Since video frames are 33.33ms long, and audio frames 0.022ms long, it’s possible for them to not line up at the end of a file.

     3) The lack of frame accurate encoding present on Mac OS, but not available for iOS Details Here

  2. FFmpeg muxing a large video only MP4 file with raw audio into TS segments. The work was based on the Kickflip SDK

     What Went Wrong - Every once in a while an audio only file would get uploaded, with no video whatsoever. Never able to reproduce it in-house, but it was pretty upsetting to our users when they didn’t record what they thought they did. There were also issues with accurate seeking on the final segments, almost like the TS segments were incorrectly time stamped.

  What I’m thinking now

  Apple was pushing fMP4 at WWDC this year (2016) and I hadn’t looked into it much at all before that. Since an fMP4 file can be read, and played while it’s being written, I thought that it would be possible for FFmpeg to transcode the file as it’s being written as well, as long as we hold off sending the bytes to FFmpeg until each fragment within the file is finished.

  However, I’m not familiar enough with the FFmpeg C API, I only used it briefly within attempt #2.

  What I need from you

  1. Is this a feasible solution ? Is anybody familiar enough with fMP4 to know if I can actually accomplish this ?
  2. How will I know that AVFoundation has finished writing a fragment within the file so that I can pipe it into FFmpeg ?
  3. How can I take data from a file on disk, chunk at a time, pass it into FFmpeg and have it spit out TS segments ?