I'm trying to come up with a pipeline to aggregate few audio buffers before having the appsink callback executed.
I have tried the following:
gst-launch-1.0 udpsrc name=udpsrc address="192.168.1.33" retrieve-sender-address=false reuse=false port=16384 caps="application/x-rtp, media=(string)audio, payload=0, clock-rate=(int)8000" timeout=10000000000 ! rtppcmudepay ! rtppcmupay min-ptime=3200000000 max-ptime=3200000000 mtu=30000 ! rtppcmudepay ! udpsink host=192.168.1.8 port=16386
And that seemed to do the trick if I use gstreamer 1.20 or 1.18.
But when I run this pipeline under 'load' ~300 concurrent pipelines, I do have streams that are waking up the callback with 160 bytes rather than the 25600.
Is there any other way that I can achive that ?
Related
Dear gstreamer community,
I am running gstreamer (1.20.3) on ubuntu 22.04 LTS with decklinkdrivers (12.4)
After building (and playing around with gstreamer, wathing tutorials etc) the following pipeline I am able to decode a high quality HD SRT Stream (udp streaming) and am outputting it to SDI (in 1080i50), works very well.
gst-launch-1.0 -v srtsrc uri=srt://x.x.x.x:xxxx latency=200 ! tsdemux name=demux demux. ! h264parse ! video/x-h264 ! avdec_h264 ! queue ! videoconvert ! video/x-raw,format=UYVY ! decklinkvideosink mode=1080i50 sync=false demux. ! avdec_aac ! queue ! audioconvert ! audio/x-raw, format=S32LE, channels=2 ! decklinkaudiosink
Audio to Videosync is stable to each other for hours (didn't test for days), but after testing the encoder to decoder end to end on my gstreamer pipeline audio comes a little too early (about 60ms early).
I tried to only change buffersize in audiopart of the pipeline to correct the timing on the audiosite e.g.
queue max-size-buffers=0 max-size-time=0 max-size-bytes=0 min-threshold-time=60000000
but audio to video offset didn't change here trying several different min-threshold-times.
for the decklinkaudiosink there is no ts-offset cap to change the timing here and also changing the buffer-time property here didn't change anything.
Can anybody please help me here how to correct the audio timing or audio latency to accurate videodecoding on my pipeline!?
Thanks!
I have gstreamer pipeline which starts
udpsrc port=50000 caps='application/x-rtp' ! rtpopusdepay ! decodebin ! queue audioconvert ...
It was made in C++ code.
How can I get info about media data? Like sampling rate, mono/stereo and etc.
I have two AXIS IP cameras streaming H264 stream over RTSP/RTP. Both cameras are set to synchronize with same NTP server so I assume both cameras will have same exact clock (may be minor diff in ms).
In my application, both cameras are pointing to same view and its required to process both camera images of same time. Thus, I want to synchronize the image capture using GStreamer.
I have tried invoking two pipelines separately on different cmd prompts but the videos are 2-3 seconds apart .
gst-launch rtspsrc location=rtsp://192.168.16.136:554/live ! rtph264depay ! h264parse ! splitmuxsink max-size-time=100000000 location=cam1_video_%d.mp4
gst-launch rtspsrc location=rtsp://192.168.16.186:554/live ! rtph264depay ! h264parse ! splitmuxsink max-size-time=100000000 location=cam2_video_%d.mp4
Can someone suggest a gstreamer pipeline to synchronize both H264 streams and record them into separate video files?
Thanks!
ARM
I am able to launch a pipeline using gst-launch as shown below. It shows good improvement on captured frame synchronization compare to lanuching two pipelines. Most times they differ by 0-500 msec. Though, I still want to synchronize them less than 150 msec accuracy.
rtspsrc location=rtsp://192.168.16.136:554/axis-media/media.amp?videocodec=h264 \
! rtph264depay ! h264parse \
! splitmuxsink max-size-time=10000000000 location=axis/video_136_%d.mp4 \
rtspsrc location=rtsp://192.168.16.186:554/axis-media/media.amp?videocodec=h264 \
! rtph264depay ! h264parse \
! splitmuxsink max-size-time=10000000000 location=axis/video_186_%d.mp4
Appreciate if someone can point other ideas!
~Arm
What do you mean synchronize? if you record to separate video files you do not need any synchronization.. as this is going to totaly separate them.. each RT(S)P stream will contain different timestamps, if you want to align them somehow to the same time (I mean real human time.. like "both should start from 15:00") then you have to configure them this way somehow (this is just idea)..
Also you did not tell us whats inside those rtp/rtsp streams (is it MPEG ts or pure IP.. etc). So I will give example of mpeg ts encapsulated rtp streams.
We will go step by step:
Suppose this is one camera just to demonstrate how it may look like:
gst-launch-1.0 -v videotestsrc ! videoconvert ! x264enc ! mpegtsmux ! rtpmp2tpay ! udpsink host=127.0.0.1 port=8888
Then this would be reciever (it must use rtmp2tdepay. We are encapsulating metadata inside MPEG container):
gst-launch-1.0 udpsrc port=8888 caps=application/x-rtp\,\ media\=\(string\)video\,\ encoding-name\=\(string\)MP2T ! rtpmp2tdepay ! decodebin ! videoconvert ! autovideosink
If you test this with your camera .. the autovideosink means that new window will popup displaying your camera..
Then you can try to store it inside file.. we will use mp4mux..
So for same camera input we do:
gst-launch-1.0 -e udpsrc port=8888 caps=application/x-rtp\,\ media\=\(string\)video\,\ encoding-name\=\(string\)MP2T ! rtpmp2tdepay ! tsdemux ! h264parse ! mp4mux ! filesink location=test.mp4
Explanation: We do not decode and reencode(waste of processing power) so I will just demux the MPEG ts stream and then instead of decoding H264 I will just parse it for the mp4mux which accepts video/x-h264.
Now you could use the same pipeline for each camera.. or you can just copypaste all elements into the same pipeline..
Now as you did not provide any - at least partial - attempt to make something out this is going to be your homework :) or make yourself more clear about the synchronization as I do not understand it..
UPDATE
After your update to question this answer is not very useful, but I will keep it here as reference. I have no idea how to synchronize that..
Another advise.. try to look at timestamps after udpsrc.. maybe they are synchronized already.. in that case you can use streamsynchronizer to synchronize two streams.. or maybe video/audio mixer:
gst-launch-1.0 udpsrc -v port=8888 ! identity silent=false ! fakesink
This should print the timestamps (PTS, DTS, Duration ..):
/GstPipeline:pipeline0/GstIdentity:identity0: last-message = chain ******* (identity0:sink) (1328 bytes, dts: 0:00:02.707033598, pts:0:00:02.707033598, duration: none, offset: -1, offset_end: -1, flags: 00004000 tag-memory ) 0x7f57dc016400
Compare PTS of each stream.. maybe you could combine two udpsrc in one pipeline and after each udpsrc put identity (with different name=something1) to make them start reception together..
HTH
I'm really running out of ideas. Here is my problem: I need to stream on demand mp4 (H264) through the network. I'm new with gstreamer and after lot of tries with versions > 1.0 I decided to use 0.10 because seems to be most promising so far.
Command below works perfect ( I see window with my movie )
gst-launch filesrc location=/home/zuko/sintel_trailer-368p.mp4 ! decodebin2 name=dec ! queue ! ffmpegcolorspace ! autovideosink dec. ! queue ! audioconvert ! audioresample ! autoaudiosink
Now I'm trying to build TCP stream using commands (so far on localhost only):
Server side:
gst-launch filesrc location=/home/zuko/sintel_trailer-368p.mp4 ! decodebin2 name=dec ! tcpserversink host=127.0.0.1 port=5000
Client side:
gst-launch tcpclientsrc host=127.0.0.1 port=5000 ! decodebin2 name=dec ! queue ! ffmpegcolorspace ! autovideosink dec. ! queue ! audioconvert ! audioresample ! autoaudiosink
But response from the "client side" command is following:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstDecodeBin2:dec/GstTypeFindElement:typefind: Could not determine type of stream.
Additional debug info:
gsttypefindelement.c(813): gst_type_find_element_chain_do_typefinding (): /GstPipeline:pipeline0/GstDecodeBin2:dec/GstTypeFindElement:typefind
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
What is missing, or what I'm doing wrong?
I'm testing on: VirtualBox 4.3.12 with Ubuntu 14.04, kernel 3.13.0-24-generic #47-Ubuntu SMP Fri May 2 23:30:00 UTC 2014 x86_64 x86_64 x86_64 GNU/Linux
Full error with (GST_DEBUG_NO_COLOR=1 GST_DEBUG=*:3 ) attached here:
https://app.box.com/s/4ntyk6am2ibg0pohtg9h
First off, using 0.10 is an absolutely bad idea, you should really stick to 1.0, for which you will have community support.
Second, to your problem itself, you are trying to stream the decoded stream over the network ("decodebin2 ! tcpserversink") and to decode it again on the other side ("tcpclientsrc ! decodebin2"). Not only is it very wrong with respect to bandwidth usage, it also straight up won't work.
I'll advise you to have a look at the rtp plugins provided by gstreamer.
Using gstreamer 1.0 the server side can share h264 streams with:
gst-launch-1.0 filesrc location="C:\\Videos\\videotestsrc.avi" ! decodebin ! x264enc ! mpegtsmux ! queue ! tcpserversink host=127.0.0.1 port=8080
While the client side receives with:
gst-launch-1.0 tcpclientsrc host=127.0.0.1 port=8080 ! decodebin ! videoconvert ! autovideosink sync=false
Alternatively, the client could be simulated with VLC through:
Media >> Open Network Stream >> tcp://127.0.0.1:8080 >> Play
I'm trying to get UPNP streaming to work. Rygel runs fine, however, all I get is a mono stream, even if the input is stereo. Doing some debugging, I replicated Rygel's gstreamer pipeline with
gst-launch-1.0 pulsesrc device=upnp.monitor num-buffers=100 ! audioconvert ! lamemp3enc target=quality quality=6 ! filesink location=test.mp3
where the problem is also apparent:
mp3info -x test.mp3
...
Media Type: MPEG 1.0 Layer III
Audio: Variable kbps, 44 kHz (mono)
...
Where does this pipeline lose the second channel? How can I debug this?
You never ask for stereo:
gst-launch-1.0 pulsesrc device=upnp.monitor num-buffers=100 ! "audio/x-raw,channels=2" ! audioconvert ! lamemp3enc target=quality quality=6 ! filesink location=test.mp3
Add a -v to the launch-line to see all the caps negotiated on all pads of the pipeline. Look for "channels" and see where it goes from 2 to 1.