avconv: getting aac to work. -strict experimental doesn't work - aac

I am trying to get the following screencast command to work:
avconv -f alsa -ar 44100 -ac 2 -i default -acodec aac -strict experimental -ab 320k -f x11grab -s 1024x600 -r 24 -i :0.0 -vcodec rawvideo screencast.mp4
But I still get the following error:
encoder 'aac' is experimental and might produce bad results.
Add '-strict experimental' if you want to use it
Other sites suggest making sure that the -strict experimental appears immediately after the aac parameter, which I have done, to no effect.

Move both the -acodec aac and -strict experimental to somewhere after the last -i parameter in the command line, before the output file name.
Parameters to avconv are parsed as "avconv [input1 options] -i input1 [input2 options] -i input2 [output options] outputfile", so when you added these parameters before the second -i they were interpreted as options to the second input, not to the output.

Related

Mediasoup inject stream freezes

I am using the following ffmpeg command to inject an rtmp stream to mediasoup.
ffmpeg \
-re \
-v info \
-stream_loop -1 \
-i rtmp://3.126.121.45:1935/live/stream \
-map 0:a:0 \
-acodec libopus -ab 128k -ac 2 -ar 48000 \
-map 0:v:0 \
-c:v libvpx -minrate 2500k -maxrate 2500k -b:v 2500k -r 30 -g 60 -max_delay 0 -bf 0 -deadline realtime -cpu-used 1 \
-f tee \
"[select=a:f=rtp:ssrc=11111111:payload_type=101]rtp://52.29.30.225:41299?rtcpport=40612&pkt_size=1300|[select=v:f=rtp:ssrc=22222222:payload_type=102]rtp://52.29.30.225:44083?rtcpport=48791&pkt_size=1300"
But the video seems to freeze randomly and plays again. Any idea how I can fix this? Tried the solution given here and here with no luck.
Update: Seems like it is the problem of RTP retransmission when some packets are lost. Unfortunately, ffmpeg doesn't fare well with RTP streaming as mentioned here. Meaning ffmpeg doesn't support retransmission mechanism like nack, pli etc. So considering gstreamer instead as suggested in the mediasoup discourse discussion.

tcpdump with -w -C -G and -z options

I'm trying to take continuous traces which are written to files that are limited by both duration (-G option) and size (-C option). The files are automatically named with the -w option, and finally the files are compressed with the -z gzip option. Altogether what I have is:
tcpdump -i eth0 -w /home/me/pcaps/MyTrace_%Y-%m-%d_%H%M%S.pcap -s 0 -C 100 -G 3600 -Z root -z gzip &
The problem is that with the -C option, the current file count is appended onto the name, so I wind up with files ending in: .pcap2.gz .pcap3.gz .pcap4.gz, etc. I would much prefer to have them end as: _2.pcap.gz _3.pcap.gz _4.pcap.gz, etc.
But if I remove .pcap from the -w option, I wind up with 2.gz 3.gz 4.gz
This could work if I could include options in the "-z" command like -z "gzip -S .pcap.gz" so that gzip itself appends the .pcap or if I could use an alias like pcap_gzip="gzip -S .pcap.gz" and then -z pcap_gzip, but neither option seems to be working, the latter producing this error: compress_savefile:execlp(gzip -S pcap.gz, /home/me/pcaps/MyTrace_2018-08-07_105308_27): No such file or directory
I encountered the same problem today, In CentOS6. I found your problem, but the answer did not work to me.
In fact, it only needs to be adjusted slightly, that is, the absolute path of the saved file name and the name of the script to be executed is written, for example
tcpdump -i em1 ... -s 0 -G 10 -w '/home/Svr01_std_%Y%m%d_%H%M%S.pcap' -Z root -z /home/pcapup2arcive.sh
I found out that although the alias doesn't work, I was able to put the same commands in a script and invoke the script via tcpdump -z.
pcap_gzip.sh:
#!/bin/bash
gzip -S .pcap.gz "$#"
Then:
tcpdump -i eth0 -w /home/me/pcaps/MyTrace_%Y-%m-%d_%H%M%S -s 0 -C 100 -G 3600 -Z root -z pcap_gzip.sh &

ffmpeg image sequence specify input framerate

I am trying to set the input framerate of a sequence of images (many folders):
if I am working with a single image sequence everything works properly:
ffmpeg -framerate 30 -i folder01/img%05d.jpeg -filter:v "crop=640:360" -r 30 outfilm.mp4
then, because I have more folders (and I was unable to get the -i concat:filesequence1|filesequence2 working) I tried to use:
ffmpeg -framerate 30 -f concat -safe 0 -i filelist.txt -filter:v "crop=640:360" -r 30 outfilm.mp4
but I receive an error:
Option framerate not found.
then if I omit the -framerate 30, everything runs smoothly, but ffmpeg defaults to a 25 fps value for the input image sequences.
Any ideas on how to fix this?
Use
ffmpeg -f concat -safe 0 -r 30 -i filelist.txt -filter:v "crop=640:360" -r 30 outfilm.mp4
When -r is used as an input option, it generates new timestamps at the given rate and sets that as the input framerate.

How to capture screen and audio input and push to rtmp server?

I use avconv on ubuntu,I found this command
avconv -f alsa -i pulse -f x11grab -r 25 -s 1280x720 -i :0.0+0,0 -acodec libfaac -vcodec libx264 -pre:0 lossless_ultrafast -threads 0 video.mkv
to save as a file, and this command
avconv -i ./test.m4v -re -c copy -f flv "rtmp://localhost/livestream"
to push live stream.
How can I combine them together?
Firstly, you should ask such questions on video.stackexchange.com and not here.
Secondly, let's take apart the two commands that you have found:
-f alsa - format for the input is alsa
-i pulse - you are reading pulse (the pulseaudio driver)
-f x11grab - planning to read from the screen on x11
-r 25 -s 1280x720 - rate and size of the incoming video stream
-i :0.0+0,0 - this selects where the incoming video comes from
-acodec libfaac - here the output options start, you're setting audio code to libfaac, or at least trying to... since this option has been deprecated long time ago, currently -c:a would be used
-vcodec libx264 - setting video code, except that you should be using -c:v
-pre:0 lossless_ultrafast -threads 0 - some sort of parameters about how encoding should be done
video.mkv - this is the output file
And the second one
-i ./test.m4v - the file you're reading
-re - "Read input at native frame rate"
-c copy - do not reencode, but simply pipe as is
-f flv - the container format
"rtmp://localhost/livestream" - where you're planning to write all that.
When you understand that, it should be clear that what you are planning to do is to use the input and encoding part from the first command, and the format and output from the second one.
Here i didn't have time to check that everything that you found is working, you should do that yourself.

Wrong frame rate when saving camera feed to a file using FFMPEG

I'm trying to save the live feed from an IP camera to a file but the resulting file always plays much faster than the original speed.
I have tried with the following commands:
ffmpeg -i http://171.22.3.47/image -vcodec copy -an -t 900 c:\output.mp4
ffmpeg -i http://171.22.3.47/image -c:v libx264 -an c:\output.mp4
Does anybody know what I'm missing? Both commands create the file and I can use Windows Media Player to play them, but they run much faster.
Try forcing output framerate by adding -r key
ffmpeg -i http://171.22.3.47/image -c:v libx264 -an -r 30 c:\output.mp4
You can also try to slow down the resulting video as an option. This will make output.mp4 2 times slower:
ffmpeg -i output.mp4 -filter:v "setpts=2.0*PTS" -c:v libx264 -an output-slow.mp4