I am trying to record a stream here on my machine to study ffmpeg library,
but with(out) success.
I have a file watcher to clean up bugged streams, that cleanup each 3 minutes files that have been not changed less then 3 minutes.
The real problem is, if I use the command below:
/usr/bin/ffmpeg -i http://sysrad.net:10090/ -y test.mp3
this command doesn't have any kind of codec or audio transformation, so my target file (test.mp3) become 256k quickly, but, if I use this command below:
/usr/bin/ffmpeg -i http://sysrad.net:10090/ -y -b:a 8k -ac 1 -ar 11025 test.mp3
My target file (test.mp3) keep 0k until the record has 256k, I am not sure if this is an Unix problem or ffmpeg problem.
Other information, if I run in loop:
while true; do wc -l teste.mp3; sleep 0.5; done;
test.mp3 file keeps 0 rows, until has 256k size...
I have no idea how to workaround that, to get the real time file size for each 1k that ffmpeg get from stream with those codecs, does you guys have any idea how can I handle that?
Thanks!!!!
Related
I'm trying to run ffmpeg mp3 stream with segmentation for each hour. Everything is working perfectly, except for one thing: when i run the command, the file size doesn't grow in real-time as i need, it only grows in packages of 256k.
Is there a way to turn a "real-time mode"?
I'm using ubuntu 18.04 with ffmpeg 3.4.6
This is the code i'm trying to run on linux terminal:
ffmpeg -i http://radiocentova.conectastm.com:8363/stream -y -acodec libmp3lame -b:a 16k -ac 1 -ar 11025 -vn -strftime 1 -f segment -segment_time 3600 -flush_packets 1 #test_%Y%m%d%H%M%S+00.mp3
Recording with segment:
Recording without segment:
The flush packets option has to be directed to the child muxer (mp3 in this case), so
-segment_format_options flush_packets=1 instead of -flush_packets 1.
I need to merge audio and video using ffmpeg so that, it should result in a video with the same duration as of audio.
I have tried 2 commands for that requirement in my linux terminal. Both the commands work for a few of the input videos; but for some other input_videos, they produce output same as the input video, the audio doesn't get merged.
The commands, I have tried are -
ffmpeg -i wonders.mp4 -i Carefull.mp3 -c copy testvid.mp4
and
ffmpeg -i wonders.mp4 -i Carefull.mp3 -strict -2 testvid.mp4
and
ffmpeg -i video.mp4 -i audio.wav -c:v copy -c:a aac -strict
experimental output.mp4
and these are my input videos -
samplevid.mp4
https://vid.me/z44E
duration - 28 seconds
size - 1.1 MB
status - working
And
wonders.mp4
https://vid.me/gyyB
duration - 97 seconds
size - 96 MB
status - not working
I have observed that the large size (more than 2MB) of the input video is probably the issue.
But, still I want the fix.
I want to capture the RTSP stream from some IP cameras, and after looking around I found 2 great tools to do this: avconv and openRTSP
openRTSP -u user password rtsp://10.48.34.125/axis-media/media.amp
avconv -i "rtsp://user:password#10.48.34.125/axis-media/media.amp" -vcodec copy -f mp4 10.48.34.125.mp4
but for some voodoo reason when I need to use URLs without an specific extension, such as:
rtsp://user:password#10.48.34.46/
avconv returns 401 Unauthorized
so I'm stuck with openRTSP at the moment...
The thing is, unlike avconv, openRTSP outputs a raw file which is encoded to 25fps, which made some of my videos look like they where in fast-forward. I found a (cpu expensive) way to re-encode the file to a closer frame rate to what I need:
avconv -r 7 -i video-H264-1 -r 24 -f mp4 10.48.34.28.mp4
(in this example I'm forcing the frame rate of the raw file to be 7, and the frame rate of the output file to be 24. I tried using openRTSP build-in flags, but the output file still had a frame rate of 25: openRTSP -f 7 -u user password rtsp://10.48.34.145/mpeg4/media.3gp)
Sadly the video looks odd at certain points, and that's because the original stream sometimes has a variable frame rate (for example at night).
My question is, is there some way to deactive this default encondig to 25fps?
And why 25? I mean, isn't the norm 24?
try:
avconv -rtsp_transport tcp -i rtsp://server -an -vcodec copy -f mp4 10.48.34.28.mp4
if you want to change original video rate to 24 you must transcode it:
avconv -rtsp_transport tcp -i rtsp://server -an -vcodec libx264 -r 24 -f mp4 10.48.34.28.mp4
I want to live stream video from webcam and sound from microphone from one computer to another but there is some problems.
When I use this command line:
ffmpeg.exe -f dshow -rtbufsize 500M -i video="Camera":audio="Microphone" -c:v mpeg4 -c:a mp2 -f mpegts udp://127.0.0.1:1234
FFmpeg console starts filling with yellow color messages and stream becomes unstable: http://s16.postimg.org/qglcgr345/Untitled.png
To solve this problem I have added new parameter to the command line to set the frame rate -r 25:
ffmpeg.exe -f dshow -rtbufsize 500M -r 25 -i video="Camera":audio="Microphone" -c:v mpeg4 -c:a mp2 -f mpegts udp://127.0.0.1:1234
After I added -r 25 problem with yellow color messages disappears but then appears another problem. When I fresh start FFmpeg with this command line video and sound looks synchronous but after one or two minutes appears ~25 seconds lag between video and sound, sound goes behind video. I have tried that with different protocols UDP, TCP, RTP but problems are the same. Please help me!
I found answer for my problem with "-r" and asynchronous audio and video. Who is interested answer is here: https://trac.ffmpeg.org/wiki/DirectShow (in paragraph "Specifying input framerate").
This is definitely a strange question but I'm looking for a way to split an mp3 mix of 60 minutes into 60 separate 1 minute long wav files to use with an audio fingerprinting API like Echonest.
Is this possible in a single ffmpeg command or would I have to run multiple iterations of ffmpeg with a the following values:
-ss is the startpoint in seconds.
-t is the duration in seconds.
You can use the segment muxer in ffmpeg:
ffmpeg -i input.mp3 -codec copy -map 0 -f segment -segment_time 60 output%03d.mp3
For a 4 minute input this results in:
$ ls -m1 output*.mp3
output000.mp3
output001.mp3
output002.mp3
output003.mp3
Since -codec copy enables stream copy mode re-encoding will be avoided. See the segment documentation for more information and examples.