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Revolution of Open-source and film making towards open film making
6 octobre 2011, par
Mis à jour : Juillet 2013
Langue : English
Type : Texte
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Sur d’autres sites (2940)
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Best way to diagnose VideoCapture not opening the rtmp stream
8 janvier 2021, par Greg0ryI am pulling my hair off for a few days and I'm out of ideas.


I have two rtmp streams


- 

- first stream is transcoded directly by myself (I use ffmpeg to transcode) and then I attach to that stream with opencv - VideoCapture can open the stream with no problem
- second stream is transcoded by 3rd party system (this system captures video through WebRTC and then encodes to h264) - this stream cannot be opened by VideCapture no matter what I do






I can attach with pure ffmpeg to that dodgy stream and I can restream - but this is not ideal as introduces extra network traffic and latency.


I was playing with
OPENCV_FFMPEG_CAPTURE_OPTIONSenvironmental variable (I was trying to remove audio stream, change the video codec, playing with rtmp options like thisOPENCV_FFMPEG_CAPTURE_OPTIONS="loglevel;debug" python my_script.py) - no joy

So I figured I am trying to solve this problem from wrong end. I should somehow collect underlying ffmpeg logs that should be available when calling VideoCapture. So I tried to set
OPENCV_FFMPEG_CAPTURE_OPTIONS="v;debug"but I can see no ffmpeg output when calling VideoCapture.

This is very simple python3 script I was using during tests :


import cv2 as cv
dodgy_cap = cv.VideoCapture()
dodgy_cap.open('rtmp://my_local_ip_address/rtmp/dodgy_stream_name')
print(dodgy_cap.isOpened()) # always returns False
healthy_cap = cv.VideoCapture()
healthy_cap.open('rtmp://my_local_ip_address/rtmp/healthy_stream_name')
print(healthy_cap.isOpened()) # always returns True


I collected information about both streams with ffprobe, but even though they look different I cannot see what would be the difference that prevents opencv from opening VideoCapture for dodgy stream..


This is a fragment from (very) verbose log for healthy stream :


RTMP_ClientPacket, received: notify 254 bytes 
(object begin) 
Property: 
Property: 
(object begin) 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
(object end) 
(object end) 
Metadata:
 duration 0.00
 width 2048.00
 height 1536.00
 videodatarate 0.00
 framerate 6.00
 videocodecid 7.00
 title Session streamed by "preview"
 comment h264Preview_01_main
 encoder Lavf58.20.100
 filesize 0.00

(... raw network packets ...)

Input #0, flv, from 'rtmp://my_local_ip_address/rtmp/healthy_stream_name':
 Metadata:
 title : Session streamed by "preview"
 comment : h264Preview_01_main
 encoder : Lavf58.20.100
 Duration: 00:00:00.00, start: 159.743000, bitrate: N/A
 Stream #0:0, 41, 1/1000: Video: h264 (High), 1 reference frame, yuv420p(progressive), 2048x1536, 0/1, 6 fps, 6 tbr, 1k tbn



And this is the fragment for dodgy stream :


RTMP_ClientPacket, received: notify 205 bytes 
(object begin) 
Property: 
(object begin) 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
Property: 
(object end) 
(object end) 
RTMP_ReadPacket: fd=3 

(... raw network packets ...)

Input #0, flv, from 'rtmp://my_local_ip_address/rtmp/dodgy_stream_name':
 Duration: N/A, start: 4511.449000, bitrate: N/A
 Stream #0:0, 41, 1/1000: Video: h264 (High), 1 reference frame, yuv420p(progressive, left), 640x480, 0/1, 15.17 fps, 15.08 tbr, 1k tbn, 30 tbc
 Stream #0:1, 124, 1/1000: Audio: aac (LC), 48000 Hz, mono, fltp



Both streams don't require any authentication (they are not exposed to the outside world)


Dodgy stream contains audio but I don't think it is source of problem as I was able to connect to other healthy rtmp streams that contained audio..


I have no more ideas how can I debug this problem, please help..



I found in VideoCap documentation that I can enable exception mode, however it did not help much (it says where it failed but it does not say why) :


>>> dodgy_stream = cv.VideoCapture()
>>> dodgy_stream.setExceptionMode(True)
>>> dodgy_stream.open("rtmp://my_local_ip_address/rtmp/dodgy_stream_name")
Traceback (most recent call last):
 File "<stdin>", line 1, in <module>
cv2.error: OpenCV(4.5.0) /tmp/pip-req-build-s_nildlw/opencv/modules/videoio/src/cap.cpp:177: error: (-2:Unspecified error) could not open 'rtmp://my_local_ip_address/rtmp/dodgy_stream_name' in function 'open'
</module></stdin>

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FFMPEG : Use expression calculated from timestamp to drawtext [closed]
26 décembre 2020, par NicolasVUsing ffmpeg and draw text, I need write on the video something like that :


text=1+floor(BPM * mod(1000*t-StartCounter ;480000/BPM)/60000).


I don't know if I need use PTS or t.


StartCounter would be a variable in milliseconds, the text would appear at this point.


BPM is an integer variable between 100 and 180.


I would draw for instance with StartCounter = 3256 and BPM = 120.


Any help appreciated, thanks


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Convert stream to a stream on the fly with ffmpeg/avconv
6 juillet 2014, par ApacciI’m trying to convert flv stream to another mp3 or ogg stream. The point is I want to use mp3/ogg stream on HTML5 page, because flv stream can’t be played with HTML5. So, I configured avserver.conf :
Port 8090
BindAddress 0.0.0.0
MaxHTTPConnections 2000
MaxClients 1000
MaxBandwidth 1000
CustomLog -
<feed>
File /tmp/feed1.ffm
FileMaxSize 200K
ACL allow 127.0.0.1
</feed>
<stream>
Feed feed1.ffm
Format ogg
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</stream>
<stream>
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</stream>
<redirect>
URL http://www.libav.org/
</redirect>the command I use :
avconv -i http://some_stream_service?format=flv -acodec mp3 http://localhost:8090/feed1.ffmconsole output :
avconv version 9.13-6:9.13-0ubuntu0.14.04.1+fdkaac, Copyright (c) 2000-2014 the Libav developers
built on May 10 2014 17:26:31 with gcc 4.8 (Ubuntu 4.8.2-19ubuntu1)
[flv @ 0x126e020] max_analyze_duration reached
Input #0, flv, from 'http://some_stream_service?format=flv':
Metadata:
encoder : Lavf52.87.1
Duration: 00:00:00.00, start: 0.000000, bitrate: N/A
Stream #0.0: Audio: aac, 44100 Hz, stereo, fltp, 31 kb/s
Output #0, ffm, to 'http://localhost:8090/feed1.ffm':
Metadata:
encoder : Lavf54.20.4
Stream #0.0: Audio: libmp3lame, 44100 Hz, stereo, fltp
Stream mapping:
Stream #0:0 -> #0:0 (aac -> libmp3lame)
Press ctrl-c to stop encoding
size= 1080kB time=66.14 bitrate= 133.8kbits/sIt looks like working, but I keep getting 0 bytes on web page
http://localhost:8090/test1.ogg. Does anybody have ideas what I’m doing wrong ?
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