I'm trying to send a stream of video that's coming into a BlackMagic DeckLink Studio 4K capture card over a few different RTMP streams at once with FFmpeg. The command that I am doing it with is this:
ffmpeg -re -format_code Hi59 -f decklink -i 'DeckLink Studio 4K' -map 0 -flags +global_header -vcodec libx264 -crf 25 -preset medium -pix_fmt yuv422p -acodec aac -f tee "[f=flv]rtmp://ip1/live/test|[f=flv]rtmp://ip2/live/test.
However, whenever I send this video out, I just get color bars when looking at the stream. I tried using a different video source (the testsrc supplied by FFmpeg), and that sends out fine over RTMP to multiple stream destinations.
Is there something weird with how tee and the decklink stuff work in FFmpeg? Or is there an issue with my command?
If you see colors bars, that means that ffmpeg is connecting to the card and streaming fine but the card is giving the bars. Your command says that ffmpeg is expecting 1920X1080#29.97 interlaced, make sure that is the format into the Decklink. You can also try explicitly setting the connection type, example:
ffmpeg -re -format_code Hi59 -video_input sdi -f decklink -i 'DeckLink Studio 4K' -map 0 -flags +global_header -vcodec libx264 -crf 25 -preset medium -pix_fmt yuv422p -acodec aac -f tee "[f=flv]rtmp://ip1/live/test|[f=flv]rtmp://ip2/live/test
If you are still running into issues, make sure that the BlackMagic software can see the video signal, and it's the format you expect.
One last thing to check, if its an HDMI input make sure it is not HDCP; it's not supported.
Related
I create a webrtc peer connection with my server(only stun)
Using pion webrtc for the server
I write the received RTP packets as VP8 and opus streams, as described here, to two pipes (the writers; created with os.Pipe() in golang)
The read ends of these two pipes are received by ffmpeg as inputs (via exec.Command.ExtraFiles) for transcoding using libx264 and aac into a single stream. The command:
ffmpeg -re -i pipe:3 -re -r pipe:4 -c:a aac -af aresample=48000 -c:v libx264 -x264-params keyint=48:min-keyint=24 -profile:v main -preset ultrafast -tune zerolatency -crf 20 -fflags genpts -avoid_negative_ts make_zero -vsync vfr -map 0:0,0:0 -map 1:0,0:0 -f matroska -strict -2 pipe:5
The above command outputs to a pipe(:5) the read end of which is being taken as input by the following:
ffmpeg -hide_banner -y -re -i pipe:3 -sn -vf scale=-1:'min(ih,360)' -c:v libx264 -pix_fmt yuv420p -ca aac -b:a 128k -b:v 1400k -maxrate 1498k -bufsize 2100k -hls_time 1 -hls_playlist_type event -hls_base_url /workdir/streamID/360p -hls_segment_filename /workdir/streamID/360p/360_%%03d.ts -f hls /workdir/streamID/360p.m3u8
This works fine as long as there are no movements of my webcam. The moment that happens the video speed suddenly increases for a split second and audio delay gets introduced. This delay keeps increasing each time I shake my webcam.
The first command in point 4 above - if written to a file separately will be absolutely fine, in terms of a/v sync, even with vigorous camera shaking. The weird audio delay is only when transcoding for hls output irrespective of whether I'm actually viewing it live or playing it back later.
This is my first time working with ffmpeg/hls/webrtc - would be really helpful if I could be pointed in the correct direction at least to be able to debug this or even know why this happens. Any and all help is greatly appreciated
I am using ffmpeg to encode livestreams for use in a tvheadend server. ffmpeg and hls discontinuities dont work, but ive fixed that using streamlink to read the hls stream then pipe that into ffmpeg.
Sometimes the audio has gaps in the live stream and the audio goes out of sync from that point on, I have managed to fix this using aresample=async=1. ffmpeg inserts silence for the gaps and audio stays synced.
Tvheadend doesnt like dts discontinuities and the stream will freeze whenever one is encountered. I have also fixed this with dts_delta_threshold 1. With this option the stream plays seamlessly without any freezes
Here is where my problem comes in when using dts_delta_threshold 1 the aresample command no longer works, I assume because there are no more gaps so it cant insert the silence. Ive tried various combinations and ordering of options.
Is there any way to apply the aresample=async=1 and also the dts_delta_threshold 1 command after.
This is my current command
streamlink -l warning --ringbuffer-size 64M --hls-timeout 100000000 --hls-live-restart hls://192.168.10.1/play/$1.$2.m3u8 best -O | \
ffmpeg -loglevel fatal -err_detect ignore_err \
-f mpegts -i - \
-filter_complex "eq=contrast=${3:-1.0}" \
-c:v libx264 -crf 18 -preset superfast -tune zerolatency -pix_fmt yuv420p -force_key_frames "expr:gte(t,n_forced*2)" \
-c:a aac -b:a 256k -ac 2 -af aresample=async=1 \
-metadata service_provider=$1 -metadata service_name="$1.$2" -f mpegts pipe:1
Ive tried putting the dts_delta_threshold before and after input, same thing audio goes out of sync if there is a gap in audio. Ive tried putting async 1 before input but that doesnt work either
I am using MacOS with FFMPEG --enabled-decklink installed to capture from BlackMagic Ultrastudio Expressto SDI to Thunderbolt and streams it to youtube, it works perfectly fine, here is my code.
ffmpeg -f decklink -format_code 'Hi50' -i 'UltraStudio Express'
-deinterlace -b:v 1M -vcodec h264 -pix_fmt yuv420p -color_range 1 -acodec aac -f flv rtmp://a.rtmp.youtube.com/live2/*******
but I am kinda blind cuz I cannot see what I am capturing till its live on Youtube, is there away to preview the video locally on my Mac before I start streaming via ffmpeg?
I have a wifi camera that uses RTSP/ONVIF protocol and after reading FFMPEG docs and some threads at Google I am trying to broadcast the stream to Youtube. So I started a broadcast at youtube and in my computer in ffmpeg I executed this command:
ffmpeg -f lavfi -i anullsrc -rtsp_transport udp -i rtsp://200.193.21.176:6002/onvif1 -tune zerolatency -vcodec libx264 -t 12:00:00 -pix_fmt + -c:v copy -c:a aac -strict experimental -f flv rtmp://x.rtmp.youtube.com/live2/private_key
The command above looks like it's correct cause it ouputs constantly something like this:
The problem is that at YOUTUBE it still says I am offline. Why?
Try replace first part to: ffmpeg -re -i somefile.mp4, so you will to know, if here any problems with your camera or not.
ffmpeg and VLC very similar and even uses same code for codecs. But RTSP it handles differently. But try just ffmpeg -i rtsp://200.193.21.176:6002/onvif1 and nothing more as source.
I'm trying to capture video from a webcam, and I find that when I use the -vcodec copy option, it works really well (far better than any other software I've tried). However, I'd like my files to be a bit smaller, and it seems that every attempt I make to compress the video leads to extremely jumpy video. If, for example, I switch the output vcodec to mjpeg, it changes from reporting 15 fps to reporting between 3 and 4 fps. Am I doing something wrong?? Here is the call with -vcodec copy:
ffmpeg -y -f dshow -vcodec mjpeg -s 1184x656 -framerate 25 -i video="HD 720P Webcam" -vcodec copy test.avi
-- which gets me 15 fps. But if I change to mjpeg, I get only 3-4 fps:
ffmpeg -y -f dshow -vcodec mjpeg -s 1184x656 -framerate 25 -i video="HD 720P Webcam" -vcodec mjpeg test.avi
Experimental attempts to put -framerate 25 or -r 25 before test.avi also does nothing to help the situation. I'm not getting any smoother video when experimenting with mpeg4 or libx264 either. Only the copy option gives me smooth video (btw I'm filming my hands playing a piano, so there is a lot of fast motion in the videos).
Help!!!! And thank you...
I don't understand why the framerate drops so much, but you could try a 2 pass approach where you first record it using -vcodec copy (as you pasted in the question)
ffmpeg -y -f dshow -vcodec mjpeg -s 1184x656 -framerate 25 -i video="HD 720P Webcam" -vcodec copy test.avi
Then transcode it into mjpeg once it's done (something like this):
ffmpeg -i test.avi -vcodec mjpeg test.mjpeg
note: I haven't actually tested any of the above command lines.
Sounds like your webcam is outputting a variable frame rate stream. Try the below on one of your copy captured files.
ffmpeg -i test.avi -vcodec libx264 -r 30 test.mp4
(You should avoid capturing to AVI, use MKV instead)