Hello i am transcoding video with rtmp and i want to put some text...i put Arial.ttf in my home directory of the server but doesnt work.. this is my command..
-map 0:0 -map 0:1 -strict -2 -crf 26 -vcodec libx264 -preset superfast
-acodec aac -b:a 128k -vf scale=-1:720 -aspect 16:9 -g 50 -r 30 -ar 48000
-ac 2 -vf drawtext="fontfile=/home/Arial.ttf: text='TEXT': fontcolor=white: fontsize=24: box=1: boxcolor=black: x=10:y=10"
-f flv
can someone help me please?
thanks
Make sure you include absolute path to font and file is there:
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf:
text='Test Text': x=100: y=50: fontsize=24: fontcolor=yellow#0.2:
box=1: boxcolor=red#0.2"
Redirect output to a log file and see if command outputs any errors.
You can also include a watermark image instead:
-i /path/to/watermark.png -filter_complex "overlay=main_w-overlay_w-4:4"
Related
I want to merge multiple commands into one so that whole processing time would be fastest. What I am doing so far is
A video file is uploaded from a webform and and server upload into a temp directory.
if (move_uploaded_file($_FILES[$param]['tmp_name'], $filePath)) {
return $filePath;
}
add black borders around the video(through ffmpeg command) - video uploads again to a directory
$command_new = "ffmpeg -i $filePath -vf 'scale=720:1280:force_original_aspect_ratio=decrease,pad=720:1280:(ow-iw)/2:(oh-ih)/2,setsar=1' $video_file";
exec($command_new);
optimizing the video(ffmpeg command) - video uploads again to a directory
$cmd_new = "ffmpeg -i $video_file -c:v libx264 -crf 28 $optimizeResultFile";
merging the uploaded file with the existing video file present in the database into one and uploading that file into directory(through ffmpeg command)
$command_new = "ffmpeg -i $optimizeResultFile -i $second_video_path -filter_complex '[0:v]pad=iw*2:ih[int];[int][1:v]overlay=W/2:0[vid]' -map [vid] -c:v libx264 -crf 23 -preset veryfast $videomerge";
//add custom sound in that final merged video
$cmd_new = "ffmpeg -i $videomerge -i $audio -c:v copy -c:a aac -shortest -map 0:v:0 -map 1:a:0 $with_new_audio";
Is it possible to do all at one command and not in multiple commands. This process takes 1 min to upload a final file. want to make it faster
ffmpeg -i $database -i $uploaded -i $audio -filter_complex "[1:v]scale=720:1280:force_original_aspect_ratio=decrease,pad=720:1280:-1:-1,setsar=1[right];[0:v][right]hstack,format=yuv420p[v]" -map "[v]" -map 2:a -c:v libx264 -crf 23 -preset veryfast -c:a copy -shortest -movflags +faststart output.mp4
The hstack filter is used instead of pad + overlay.
This assumes $database is the same height as $uploaded.
This assumes $audio is AAC. Otherwise, change -c:a copy to -c:a aac.
In this configuration file
https://github.com/TareqAlqutami/rtmp-hls-server/blob/master/conf/nginx.conf#L24-L30
for each received stream, transcode for adaptive streaming This single
ffmpeg command takes the input and transforms the source into 4
different streams with different bitrates and qualities. # these
settings respect the aspect ratio.
How we can dynamically generate variants? i.e for 1080p input generate all variants, but for 240p input generate no variants
My startup works without error
You need to configure the log and see what error it gives.
But another solution is to check manually
You may be using a codec that is not installed
I will check your tank, maybe I can contribute here
application live {
live on; # Allows live input
exec ffmpeg -i rtmp://localhost/live/$name -threads 8
-c:v libx264 -profile:v baseline -b:v 768K -s 640x360 -vf "drawtext= fontcolor=red: fontsize=20: fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf: text='360': x=10: y=10:" -f flv -c:a aac -ac 1 -strict -2 -b:a 96k rtmp://localhost/show/$name_360
-c:v libx264 -profile:v baseline -b:v 1024K -s 960x540 -vf "drawtext= fontcolor=red: fontsize=20: fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf: text='480': x=10: y=10:" -f flv -c:a aac -ac 1 -strict -2 -b:a 128k rtmp://localhost/show/$name_480
-c:v libx264 -profile:v baseline -b:v 1920K -s 1280x720 -vf "drawtext= fontcolor=red: fontsize=20: fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf: text='720': x=10: y=10:" -f flv -c:a aac -ac 1 -strict -2 -b:a 128k rtmp://localhost/show/$name_720
-c:v libx264 -profile:v baseline -b:v 4000K -s 1920x1080 -vf "drawtext= fontcolor=red: fontsize=20: fontfile=/usr/share/fonts/truetype/dejavu/DejaVuSans.ttf: text='720': x=10: y=10:" -f flv -c:a aac -ac 1 -strict -2 -b:a 128k rtmp://localhost/show/$name_1080;
}
application show {
live on; # Allows live input from above
hls on; # Enable HTTP Live Streaming
# hls_fragment 5s;
# Pointing this to an SSD is better as this involves lots of IO
hls_path /dest;
#hls_variant _240 BANDWIDTH=288000;
hls_variant _360 BANDWIDTH=448000;
hls_variant _480 BANDWIDTH=1152000;
hls_variant _720 BANDWIDTH=2048000;
hls_variant _1080 BANDWIDTH=4096000;
}
I have an RTMP arut server inside which I call a python script to run an FFMPEG command and create HLS packaging.
Problem is that when I stop the RTMP stream, FFMPEG still run processes in background.
FFMPEG string is like this:
ffmpeg -re -v verbose -i rtmp://localhost:1935/live/testlive -rw_timeout 500 -http_persistent 1 -method PUT -http_user_agent test -f hls -hls_list_size 5 -hls_flags discont_start+delete_segments -vf "scale=426:trunc(ow/a/2)*2" -c:a libfdk_aac -ar 48000 -c:v h264 -profile:v main -crf 24 -sc_threshold 0 -g 48 -keyint_min 48 -hls_time 4 -hls_playlist_type event -preset veryfast -b:v 300k -maxrate 856k -bufsize 1200k -b:a 128k -vcodec libx264 -hls_segment_filename http://mystream/123/v1/testlive/0_%03d.ts http://mystream/123/v1/testlive/index.m3u8
The RTMP server should send an NetStream.Play.Stop event to subscribers (or disconnect them) to signal end of stream. If you are using the nginx rtmp module, look at the play_restart or idle_streams directives.
I am attempting to grab the .m3u8 file from an nginx-rtmp server and pass it along to YouTube rtmp. I believe this to be possible (for example here: https://stackoverflow.com/a/11978820/1552594 although this is on the same host). The command I am using is:
ffmpeg -analyzeduration 0 -i \
http://source.rtmp.server/hls/stream.m3u8 -pix_fmt yuv420p \
-f flv rtmp://a.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx
However the output contains only audio and YouTube doesn't like it. The command produces following:
As you can see no Video stream in output metadata, stream mapping shows only audio and the trace shows 0kb of Video for 651kb of Audio
Any help much appreciated
MORE INFO
Improved version of the command lifted from this article:
https://judge2020.com/restreaming-a-m3u8-hls-stream-to-youtube-using-ffmpeg/
"Restreaming a m3u8 HLS stream to Youtube using FFMPEG" AKA exactly what I am trying to do.
The command I am sending is now:
ffmpeg -re -i "http://source.rtmp.server/hls/stream.m3u8" \
-strict -2 -c:v copy -c:a aac -ar 44100 -ab 128k -ac 2 -flags \
+global_header -bsf:a aac_adtstoasc -bufsize 3000k -f flv \
"rtmp://a.rtmp.youtube.com/live2/xxx-xxxx-xxxx-xxxx"
I got pretty much exactly the same response except with the Audio being read and output using aac codec.
MORE MORE INFO
I have found that adding a mapping can force the video stream into the output:
ffmpeg -re -i "http://source.rtmp.server/hls/stream.m3u8" \
-strict -2 -c:v copy -c:a -map 0:0 -map 0:1 -ar 44100 -ab 128k -ac 2 \
-flags +global_header -bsf:a aac_adtstoasc -bufsize 1000k \
-f flv "rtmp://a.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx"
This throws up the error that has presumably been resulting in the video stream being silently dropped:
Finally worked it out. The last issue above was a red herring and was due to missing a codec argument for the audio -c:a.
The complete working command is as follows:
ffmpeg -probesize 100M -analyzeduration 20M -re \
-i "http://source.rtmp.server/hls/stream.m3u8" -strict -2 -c:v \
libx264 -pix_fmt yuv420p -c:a aac -map 0:0 -map 0:1 -ar 44100 \
-ab 128k -ac 2 -b:v 2567k -flags +global_header -bsf:a aac_adtstoasc \
-bufsize 1000k -f flv "rtmp://a.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx"
The important parts are -probesize and -analyeduration - these need tweaking until they work. The -re flag is important to indicate restreaming. The various video codec declarations are also important - -c:v libx264 -pix_fmt yuv420p or it will throw errors about the output size being 0x0. Finally the map flags ensure that both streams are included in the output: -map 0:0 -map 0:1
I want stream to rtmp server. I try:
ffmpeg.exe -f dshow -i "myvideo.avi" -s 480x288 -r 10 -vcodec libx264 -b:v 30k -pix_fmt yuv420p -acodec libvo_aacenc -ac 1 -b:a 4k -f flv rtmp://ip/live/test
And did not work... Error:
http://i.stack.imgur.com/06Rcf.png
how to do? Thx