I tried to convert a UDP stream (that genrated from DVB signal) to HLS m3u8 file with this code:
ffmpeg -i udp://239.1.2.1:60001 -acodec aac -strict -2 -vcodec libx264 -hls_wrap 100 -f hls /var/www/html/ts/1.m3u8
and output m3u8 file is :
#EXTM3U
#EXT-X-VERSION:3
#EXT-X-TARGETDURATION:3
#EXT-X-MEDIA-SEQUENCE:66
#EXTINF:0.960000,
21.ts
#EXTINF:2.080000,
22.ts
#EXTINF:2.400000,
23.ts
#EXTINF:1.760000,
24.ts
#EXTINF:2.080000,
20.ts
I tried to change the target duration to 10 in this file with the option -segment_time 10 but the target duration doesn't change in m3u8 file, can anybody help me with this problem?
I solve this problem with code:
ffmpeg -i udp://239.1.2.4:60004?fifo_size=50000000 -acodec copy -vcodec copy -preset ultrafast -flags -global_header -f hls -hls_time 20 -hls_wrap 5 /var/www/html/ts/4.m3u8
Add this option to ffmpeg fix problem
-preset ultrafast -flags -global_header
Related
I am trying to encode a video to webm for playing through a HTML5 video tag. I have these settings...
ffmpeg -i input.mp4 -c:v libvpx-vp9 -b:a 128k -b:v 1M -c:a libopus output.webm
The results aren't great, video has lost lot's of it's sharpness. Looking at the original file I can see the bitrate is 1694kb/s.
Are there any settings I can add or change to improve the output? Would maybe a 2 pass encode improve things?
Try with
ffmpeg -i input.mp4 -c:v libvpx-vp9 -crf 30 -b:v 0 -b:a 128k -c:a libopus output.webm
Adjust the CRF value till the quality/size tradeoff is ok. Lower values produce bigger but better files.
Try to run two passes:
ffmpeg -i file.mp4 -b:v 0 -crf 30 -pass 1 -an -f webm -y /dev/null
ffmpeg -i file.mp4 -b:v 0 -crf 30 -pass 2 output.webm
From - https://trac.ffmpeg.org/wiki/Encode/VP9
THE INPUT FILES
An overlay image that has is being updated every 5 seconds by a Python script
A small MP4 file that will be looped by a concat input
An MP3 file as audio source
THE COMMAND (UPDATED)
This is the command I'm currently using to combine and stream the inputs.
ffmpeg -re -i music.mp3 -f concat -i videoincludes.txt
-r 1 -loop 1 -f image2 -i overlay.png
-c:v libx264 -c:a aac -shortest -crf 23 -pix_fmt yuv420p
-maxrate 2500k -bufsize 2500k -preset ultrafast -r 30 -g 60 -b:v 2000k -b:a 192k -ar 44100
-filter_complex "[1:v][2:v] overlay=0:0" -map 0:a -strict -2
-f flv rtmp://a.rtmp.youtube.com/live2/{key}
Als tried using -framerate 1 instead of -r 1
THE ISSUE
So the issue is that the image doesn't always update. Sometimes it does update every couple seconds at the start but it stops updating after 10-20 seconds without any difference in log output and sometimes it just doesn't update.
I can however confirm that the image is being updated by the Python script but FFmpeg is just not picking this up.
I read setting the input format of the image to image2 should allow it to update so I am not sure what is wrong or what I can do to improve it.
I'm working on the same task, and finally, I think, I found the answer.
Because streams different from each other we must reset their timestamps with setpts=PTS-STARTPTS to have them begin in the same zero timestamp . And, also, try to use image2pipe instead of image2.
This is your code with timestamp reset:
ffmpeg -re -i music.mp3 -f concat -i videoincludes.txt
-r 1 -loop 1 -f image2pipe -i overlay.png
-c:v libx264 -c:a aac -shortest -crf 23 -pix_fmt yuv420p
-maxrate 2500k -bufsize 2500k -preset ultrafast -r 30 -g 60 -b:v 2000k -b:a 192k -ar 44100
-filter_complex "[1:v]setpts=PTS-STARTPTS[out_main]; [2:v]setpts=PTS-STARTPTS[out_overlay]; [out_main][out_overlay]overlay=0:0" -map 0:a -strict -2
-f flv rtmp://a.rtmp.youtube.com/live2/{key}
p.s and I think, there is no need in -r or -framerate anymore
i am having issue, to concat video, it looses the audio sync and audio started from previous video.
i have tried below two link/so answer by Mulvya, but none of them work :(
here is the code i am trying:
1: re-encode file a (1):
ffmpeg.exe -i "f:\1.avi" -af apad -vf scale=1280:720 -crf 15.0 -vcodec libx264 -acodec aac -ar 48000 -b:a 192k -coder 1 -rc_lookahead 60 -threads 0 -shortest -avoid_negative_ts make_zero -fflags +genpts 01.mp4
2: re-encode file b (2):
ffmpeg.exe -i "f:\2.mp4" -af apad -vf scale=1280:720 -crf 15.0 -vcodec libx264 -acodec aac -ar 48000 -b:a 192k -coder 1 -rc_lookahead 60 -threads 0 -shortest -avoid_negative_ts make_zero -fflags +genpts 02.mp4
3: Now contact using following command:
ffmpeg.exe -f concat -safe 0 -i "f:\files.txt" -c copy test.mp4
but audio is not synced with video ;(
i also uploaded both video (and also the output) on dropbox:
Video Sync Issue Source And Result Files
so, any help, how to make audio perfectly sync with video would be great :)
Some one else on another forum helped me to find the issue. So, i post it as answer here so that someone else may can get help.
I was concerned about same frame size, audio and video codec to concat properly, but i forgot about frame rate.
That first sample video (mentioned in my question) frame rate was 12, while 2nd video frame rate was 25; and that's what make the sync problem.
Now, i have set frame rate 25 to those two video (-r 25) and it's works like charms :)
below is the full conversion code:
ffmpeg.exe -i "f:\1.avi" -r 25 -af apad -vf scale=1280:720 -crf 15.0 -vcodec libx264 -acodec aac -ar 48000 -b:a 192k -coder 1 -rc_lookahead 60 -threads 0 -shortest -avoid_negative_ts make_zero -fflags +genpts 01.mp4
hope it may help some one like me in near future.
best regards
some other solution for syncing audio & video
use -bsf:v h264_mp4toannexb in your input files
ffmpeg.exe -i "1.avi" -af apad -vf scale=1280:720 -crf 15.0 -vcodec libx264 -acodec aac -bsf:v h264_mp4toannexb -ar 48000 -b:a 192k -coder 1 -rc_lookahead 60 -threads 0 -shortest -avoid_negative_ts make_zero -fflags +genpts output1.flv
and same for 2nd file
ffmpeg.exe -i "2.avi" -af apad -vf scale=1280:720 -crf 15.0 -vcodec libx264 -acodec aac -bsf:v h264_mp4toannexb -ar 48000 -b:a 192k -coder 1 -rc_lookahead 60 -threads 0 -shortest -avoid_negative_ts make_zero -fflags +genpts output2.flv
then files can be joined together
ffmpeg -f concat -safe 0 -i concat.txt -c copy -bsf:a aac_adtstoasc final.flv
where concat.txt file contains
file 'output1.flv'
file 'output2.flv'
I am using ffmpeg to create m3u8 playlist for a video (actually a live video stream). I am using the following command:
ffmpeg -i /home/ubuntu/Download/1459530099245.mkv -c:a aac -strict experimental -ac 2 -ar 48k -ab 64k -c:v libx264 -s 480x270 -aspect 16:9 -b:v 400k -r 15 -g 45 -profile:v baseline -level 3.0 -f hls -hls_time 9 -hls_list_size 0 /home/ubuntu/Download/New Playlist.m3u8
It produces m3u8 file as well as ts files.
Question: simply, how can we produce m3u8 playlist and TS files for a particular duration of source video? E.g., I want to get playlist only for first 20 seconds or so?
You can use command -to. Just add -to 00:00:20 after input path
In your variant:
ffmpeg -i /home/ubuntu/Download/1459530099245.mkv -to 00:00:20 -c:a aac -strict experimental -ac 2 -ar 48k -ab 64k -c:v libx264 -s 480x270 -aspect 16:9 -b:v 400k -r 15 -g 45 -profile:v baseline -level 3.0 -f hls -hls_time 9 -hls_list_size 0 /home/ubuntu/Download/New Playlist.m3u8
More information here: http://www.bogotobogo.com/FFMpeg/ffmpeg_seeking_ss_option_cutting_section_video_image.php
I am trying to make a .m3u8 hls playlist locally on my computer. I am using Windows 7, 32 bit.
I am a begginner of FFMPEG, and finally I combined some commands into this:
ffmpeg -re -rtbufsize 999999k -y -f dshow -video_size 640x360 -r 15 -i video="ManyCam Virtual Webcam" -c:v libx264 -crf 18 -profile:v main -maxrate 999999k -bufsize 99999k -pix_fmt yuv420p -flags -global_header -hls_time 10 -g 3 -hls_list_size 2 -hls_wrap 0 -start_number 1 D:\\stream\stream.m3u8 -tune zerolatency
The problem is that while playing the m3u8 playlist with VLC or a HLS player, the video speed is faster than normal.
If you need more info, please comment.
EDIT
If I play one of the .ts files, the problem is the same.
Thank you so much!