I am trying to stream a video loop to justin.tv using FFmpeg? I have managed to loop an image sequence and combine it with line in audio:
ffmpeg -loop 1 -i imageSequence%04d.jpg -f alsa -ac 2 -ar 22050 -ab 64k \
-i pulse -acodec adpcm_swf -r 10 -vcodec flv \
-f flv rtmp://live.justin.tv/app/<yourStreamKeyHere>
Is it possible to do this with a video file?
Definitely possible. In the recent versions of ffmpeg they have added a -stream_loop flag that allows you to loop the input as many times as required.
The gotcha is that if you don't regenerate the pts from the source, ffmpeg will drop frames after the first loop (as the timestamp will suddenly go back in time). To avoid this, you need to tell ffmpeg to generate the pts so you get an increasing timestamp between loops. This is done with the +genpts call (it has to be before the -i arg).
Here's an example ffmpeg call (replace $F with your input file). This example generates two output streams and the -stream_loop -1 argument tells ffmpeg to continuously loop the input. The output in this case is for a similar stream broadcast ingest (MetaCDN), adjust accordingly to your requirements.
ffmpeg -threads 2 -re -fflags +genpts -stream_loop -1 -i $F \
-s 640x360 -ac 2 -f flv -vcodec libx264 -profile:v baseline -b:v 600k -maxrate 600k -bufsize 600k -r 24 -ar 44100 -g 48 -c:a libfdk_aac -b:a 64k "rtmp://publish.live.metacdn.com/2050C7/dfsdfsd/lowquality_664?hello&adbe-live-event=lowquality_" \
-s 1920x1080 -ac 2 -f flv -vcodec libx264 -profile:v baseline -b:v 2000k -maxrate 2000k -bufsize 2000k -r 24 -ar 44100 -g 48 -c:a libfdk_aac -b:a 64k "rtmp://publish.live.metacdn.com/2050C7/dfsdfsd/highquality_2064?mate&adbe-live-event=highquality_"
Sinclair Media has found a solution by using the lavfi filter and appending :loop=0 to the file name :
This is untested:
ffmpeg -f lavfi -re -i movie=StreamTest.avi:loop=0 \
-acodec libfaac -b:a 64k -pix_fmt yuv420p -vcodec libx264 \
-x264opts level=41 -r 25 -profile:v baseline -b:v 1500k \
-maxrate 2000k -force_key_frames 50 -s 640×360 -map 0 -flags \
-global_header -f segment -segment_list index_1500.m3u8 \
-segment_time 10 -segment_format mpeg_ts \
-segment_list_type m3u8 segmented.ts
But it should create a local "index_1500.m3u8" file that streams the video in "StreamTest.avi".
I just reuse the Rob's answers with a few of modifications in order to provide a file to live streaming
ffmpeg -threads 2 -re -fflags +genpts -stream_loop -1 -i gvf.mp4 -c copy -f mpegts -mpegts_service_id 102 -metadata service_name=My_channel -metadata service_provider=My_Self -max_interleave_delta 0 -use_wallclock_as_timestamps 1 -flush_packets 1 "udp://233.0.0.1:1001?localaddr=10.60.4.237&pkt_size=188"
Related
I'll update this question
ffmpeg -version
ffmpeg -version
ffmpeg version 4.3.1-4ubuntu1 Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 10 (Ubuntu 10.2.0-9ubuntu2)
I run this command to use ffmpeg to stream to youtube ;
ffmpeg -y -threads 12 \
-loop 1 -framerate 30 -re \
-i ./1280x720.jpg \
-i ./audio.mp3 \
-video_size 1280x720 \
-vcodec libx264 -pix_fmt yuv420p \
-b:v 4500k -maxrate 5500k -bufsize 22000k \
-preset ultrafast -crf 23 -tune stillimage \
-b:a 128k -ar 44100 -ac 2 -acodec aac \
-filter_complex "dynaudnorm=f=150:g=15" \
-r 30 -g 60 \
-f flv rtmp://a.rtmp.youtube.com/live2/xxxx 2>&1 | tee _LOG
The stream is excellent for 45-53 minutes then i'll get an error like this from ffmpeg:
[flv # 0x56077027cd80] Delay between the first packet and last packet in the muxing queue is 10034000 > 10000000: forcing output
then youtube starts to say, no data being received and the stream will end, which it does.
This is the full log: http://0x0.st/-zUH.txt
Your MP3 duration is 00:49:57.42 so the stream messes up after it ends. Loop the audio with -stream_loop -1 and add -re for real-time reading of the input:
ffmpeg -y \
-loop 1 -framerate 30 -re -i ./1280x720.jpg \
-re -stream_loop -1 -i ./audio.mp3 \
-c:v libx264 -pix_fmt yuv420p \
-b:v 4500k -maxrate 5500k -bufsize 22000k \
-preset ultrafast -tune stillimage \
-b:a 128k -ar 44100 -ac 2 -c:a aac \
-filter_complex "dynaudnorm=f=150:g=15" \
-g 60 -f flv rtmp://a.rtmp.youtube.com/live2/xxxx
Alternatively, remove -re -stream_loop -1 and add the output option -shortest if you want the stream to end when the audio ends.
Unrelated changes:
No need to set -threads. Let ffmpeg auto choose.
-video_size 1280x720 is an input option for certain demuxers and does nothing in your command. Removed. Your input is already 1280x720 anyway: otherwise, see Resizing videos with ffmpeg to fit a specific size.
-b:v and -crf are mutually exclusive. In your case -b:v is being ignored. For streaming you probably want to use -b:v. Removed -crf.
You already set the frame rate with -framerate 30 so -r 30 is not needed. Removed.
Recommend using the slowest -preset that still encodes fast enough.
I want to stream my video to 4 destinations. My input signal needs to be recoded to "H.264 AAC", so I want to send it to my server. This works already.
Client -> Server with ffmpeg -> Destinations
Now I have a performance problem: One should get the stream in 1080p and two in 720p.
So it would make sense to first get the stream in the desired formats H.264 1080p and AAC with 30 FPS and then calculate the stream once, send it 1:1 to the two HD targets.
and create a 720p stream in parallel and send it to the two remaining destinations.
What is the best way to do this on a Ubuntu 16.04 machine?
My previous approach:
ffmpeg -i rtmp://livestream.domain.example/live/<key> \
-threads 2 -s hd1080 -preset veryfast -f flv rtmp://destination1.example/live2/<key> \
-threads 2 -s hd1080 -preset veryfast -f flv rtmp://destination2.example/live2/<key> \
-threads 1 -s hd720 -c:v libx264 -c:a aac -preset veryfast -r 30 -g 60 -b:v 3000k -f flv rtmp://destination3.example/x/<key> \
-threads 1 -s hd720 -c:v libx264 -preset veryfast -c:a aac -f flv 'rtmps://destination4.exmple/rtmp/<key>'
You can see the repetitions in the code. :-/
Use the tee muxer:
ffmpeg -i rtmp://livestream.domain.example/live/<key> \
-filter_complex "[0:v]scale=-2:1080,fps=30,split=outputs=2[1080a][1080b];[0:v]scale=-2:720,fps=30,split=outputs=2[720a][720b]" \
-map "[1080a]" -map "[1080b]" -map "[720a]" -map "[720b]" -map 0:a \
-c:v libx264 -c:a aac -preset veryfast -g 60 -b:v 3000k -maxrate 3000k -bufsize 6000k -f tee \
"[select=\'v:0,a\':f=flv:onfail=ignore]rtmp://destination1.example/live2/<key>| \
[select=\'v:1,a\':f=flv:onfail=ignore]rtmp://destination2.example/live2/<key>| \
[select=\'v:2,a\':f=flv:onfail=ignore]rtmp://destination3.example/live2/<key>| \
[select=\'v:3,a\':f=flv:onfail=ignore]rtmp://destination4.example/live2/<key>"
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
How can I make a two-pass convert while using .ts chunks output?
I use the following command inside bash script for chunks generation (I think all variables are clear enough for understanding):
ffmpeg -i $1 -threads 1 -b:v ${selected_bitrate} -b:a ${audio_bitrate} -s ${selected_width}x${selected_height} -r ${framerate} -preset fast -level ${level} -vcodec libx264 -f ssegment -segment_list b${selected_bitrate}.m3u8 -segment_time 9 -force_key_frames "expr:gte(t,n_forced*9)" -y b${selected_bitrate}_%05d.ts
I want to try two-pass because I need to match desired bitrate more accurate. Right now when I use for example 200k bitrate for video stream, it results ~380k in ts chunks (of course without audio).
Just call the libx264 with "-pass 1" like the following:
ffmpeg -i $1 -threads 1 -ar -b:v ${selected_bitrate} -s ${selected_width}x${selected_height} -r ${framerate} -preset fast -level ${level} -vcodec libx264 -pass 1 -f null -
Then repeat your command with "-pass 2":
ffmpeg -i $1 -threads 1 -b:v ${selected_bitrate} -b:a ${audio_bitrate} -s ${selected_width}x${selected_height} -r ${framerate} -preset fast -level ${level} -vcodec libx264 -pass 2 -f ssegment -segment_list b${selected_bitrate}.m3u8 -segment_time 9 -force_key_frames "expr:gte(t,n_forced*9)" -y b${selected_bitrate}_%05d.ts
That should give you what you want.
I'm looking for a way to record a video UDP stream using ffmpeg but in 10mn chunks.
I currently use the following to get 10mn of video (with h264 transcoding).
"ffmpeg -i udp://239.0.77.15:5000 -map 0:0 -map 0:1 -s 640x360 -vcodec libx264 -g 100 -vb 500000 -r 25 -strict experimental -vf yadif -acodec aac -ab 96000 -ac 2 -t 600 -y /media/test.m4 "
My problem is that using command line ffmpeg needs time to resync with the udp stream loosing 2 seconds of video each time. Is it normal ?
Any idea if there is a way to do it in command line or should I tried to use the ffmpeg API ?
Thanks in advance
Ok found it.
Recently ffmpeg add a segmenter, here is the syntax:
-f segment: tell ffmpeg to use the segmenter
-segment_time: chunk size in second
You can use autoincrement file name with something like %03d (000,001,002,003...).
Here is my line to transcode a UDP MPEGTS stream, into H264+AAC and save it to file chunk (60 seconds):
ffmpeg -i udp://239.0.77.15:5000 -map 0:0 -map 0:1 -s 640x360 -vcodec libx264 -g 60 -vb 500000 -strict experimental -vf yadif -acodec aac -ab 96000 -ac 2 -y -f segment -segment_time 60 "xxx-%03d.ts"
This is a better way:
ffmpeg -re -i udp://10.1.1.238:1234?fifo_size=1000000 -vcodec libx264 -vb 500000 -g 60 -vprofile main -acodec aac -ab 128000 -ar 48000 -ac 2 -vbsf h264_mp4toannexb -b 1000k -minrate 1000k -maxrate 1000k -strict experimental -f stream_segment -segment_format mpegts -segment_time 5 -segment_atclocktime 1 -reset_timestamps 1 -strftime 1 d:/%H%M%S.mp4
By this code ffmpeg makes series of output files using current system time.