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• gstreamer vaapih264enc generated ts segment dont play on AVPlayer

  21 juin 2022, par Guru Govindan

  I have a pipeline that transcodes an rtsp stream into hls segments. The manifest is playable in browsers(with hlsjs) and ffplay and VLC. However when I play the stream or individual ts segments in Quicktime player or my IOS App that uses avplayer, it doesnt work.

  


  This seems to be an issue only with vaapih264enc. When I try the same with software encode like x264enc it works fine.

  


  The following is a simple pipeline where this is reproducible.

  


  GST_DEBUG=3 gst-launch-1.0 rtspsrc location='rtsp://user:pass@10.10.10.12:554/' name=rtpsrc0 \
 rtpsrc0. ! rtph264depay ! queue ! decodebin ! vaapih264enc ! mpegtsmux name=mux ! filesink location=mymux.ts \
 rtpsrc0. ! decodebin ! queue ! fdkaacenc ! mux.


  


  when I run the following command to copy the encoded stream as is from the generated ts segment it works

  


  ffmpeg -i mymux.ts -c copy mymux_ffmpeg.ts


  


  Is the ffmpeg adding some additional header information that the quicktime player is happy with ?

  


  I appreciate any help with this.

  


• Best Web Video Encoding Practices for IOS (FFMpeg)

  21 juin 2012, par MagicMushroom

  I am working on an online video repository system for a client, written mostly in PHP. At the moment I am building a mobile version of our desktop website. Our desktop site allows users to watch videos in the browser, much like YouTube.

  My client uploads videos through the manager interface I have created, and my application uses FFmpeg on the server to transcode his videos into several resolutions and bitrates. I am no expert on FFmpeg, and while I do not know the ins and outs of each individual setting, I do understand how it works as a whole. Right now, we are using the mp4 container format with the h.264 codec to encode our videos. Our command looks like :

  ffmpeg -y -i "INPUT FILE.mov" -f mp4 -s 640x480 -vcodec libx264 -preset fast -maxrate 1500 -bitrate 1000 -bufsize 4096 -acodec libfaac -ab 192 -ac 2 "OUTPUT_FILE.mp4" >> "FILE.log" 2>&1 &

  I'm hoping to gain information about best practices with encoding video for web streaming on IOS and other mobile devices using FFmpeg. What resolutions and settings make for good mobile streaming video ? How can I ensure maximum compatibility across the sea of Android devices ?

• How to play video Media Source Extensions when the audio start is delayed ? Or how to fix it with ffmpeg ?

  11 décembre 2020, par sheodox

  I have a video that I'm splitting the individual video/audio streams out then dashing with MP4Box, then I'm playing them with Media Source Extensions and appending byte ranges to video/audio source buffers from the MPD files. It's all working nicely, but one video I have has audio that is delayed by about 1.1 second. I couldn't get it to sync up and the audio would always play ahead of the video.

  


  Currently I'm trying to set the audioBuffer.timestampOffset = 1.1 and that gets it to sync up perfectly. The issue I'm running into now though is the video refuses to play unless the audio source buffer has data. So the video stalls right away. If I skip a few seconds in (past the offset) everything works because both video/audio are buffered.

  


  Is there a way to get around this ? Either make it play without the audio loaded, somehow fill the audio buffer with silence (can I generate something with the Web Audio API) ? Add silence to the audio file in ffmpeg ? Something else ?

  


  I first tried adding a delay in ffmpeg with ffmpeg -i video.mkv -map 0:a:0 -acodec aac -af "adelay=1.1s:all=true" out.aac but nothing seemed to change. Was I doing something wrong ? Is there a better way to demux audio while keeping the exact same timing as when it was in the container with the video so I don't have to worry about delays/offsets at all ?