I have a gstreamer pipeline which works perfectly and takes a camera stream, encodes it as H.264 video, saves it to a file AND displays it on the screen as follows:
gst-launch-1.0 -v autovideosrc ! tee name = t ! queue ! omxh264enc !
'video/x-h264, stream-format=(string)byte-stream' ! h264parse ! qtmux !
filesink location=test.mp4 t. ! queue ! videoscale ! video/x-raw,
width=480,height=270 ! xvimagesink -e sync=false
Now, I am trying to do something even simple and just record the stream to a file (without displaying on screen) and this does not seem to work! It writes a file but cannot play it. What I have tried so far is:
gst-launch-1.0 -v autovideosrc ! queue ! omxh264enc ! 'video/x-h264,
stream-format=(string)byte-stream' ! h264parse ! qtmux ! filesink
location=test.mp4 sync=false
I can also remove the queue element but with the same result:
gst-launch-1.0 -v autovideosrc ! omxh264enc ! 'video/x-h264,
stream-format=(string)byte-stream' ! h264parse ! qtmux ! filesink
location=test.mp4 sync=false
It does not give any errors but just does not write a valid stream to my filesink, it seems.
How do you stop the stream? Will the camera correctly inject an EOS signal? If not and you just press ctrl-c to stop the operation the .mp4 file will missing important headers which are required for proper playback.
Add -e to your command line. In that case when you press ctrl-c the pipeline will not just stop but is being properly shut down by sending an EOS signal through the pipeline.
Related
Using the following 2 commands I can stream a videotestsrc source over SRT.
gst-launch-1.0 -v videotestsrc ! queue ! x264enc ! queue ! mpegtsmux alignment=7 ! identity silent=false ! queue leaky=downstream ! srtsink uri="srt://:8888" sync=false async=false
gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:8888" ! identity silent=false ! fakesink async=false
And play it in this way:
gst-play-1.0 srt://127.0.0.1:8888
Now I want to stream a rtsp source, and I get it in the following way:
gst-launch-1.0 rtspsrc location=rtsp://localhost:8554/main latency=100 ! queue ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! videoscale ! video/x-raw,width=640,height=480 ! srtsink uri="srt://:8888" sync=false async=false
gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:8888" ! identity silent=false ! fakesink async=false
However, when I when to playback I have this error:
gst-play-1.0 srt://127.0.0.1:8888
Press 'k' to see a list of keyboard shortcuts.
Now playing srt://127.0.0.1:8888
Pipeline is live.
ERROR Could not determine type of stream. for srt://127.0.0.1:8888
ERROR debug information: ../subprojects/gstreamer/plugins/elements/gsttypefindelement.c(999): gst_type_find_element_chain_do_typefinding (): /GstPlayBin:playbin/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind
Reached end of play list.
How can I solve it?
I'm trying to record audio and video from internal webcam and mic to segmented files with gstreamer.
It works to a single file by doing:
gst-launch-1.0 -e avfvideosrc !
video/x-raw ! vtenc_h264 ! h264parse ! queue !
mpegtsmux name=mux ! filesink location=test.mp4 osxaudiosrc !
decodebin ! audioconvert ! faac ! aacparse ! queue ! mux.
It doesn't work when doing:
gst-launch-1.0 -e avfvideosrc !
video/x-raw ! vtenc_h264 ! h264parse ! queue !
splitmuxsink
muxer=mpegtsmux
location=test%04d.mp4
max-size-time=1000000000
name=mux osxaudiosrc !
decodebin ! audioconvert ! faac ! aacparse ! queue ! mux.
saying erroneous pipeline: could not link queue1 to mux
I'm using gstreamer 1.12.3 on Mac OSX Sierra
Note: The H264/AAC encoding isn't necessary for what I want to achieve, so if there are solutions that only work with e.g. avimux, for whatever reason, that's fine.
EDIT: I've tried this on a windows machine with the same error.
gst-launch-1.0 -ev ksvideosrc ! video/x-raw !
videoconvert ! queue !
splitmuxsink max-size-time=1000000000 muxer=avimux name=mux
location=video%04d.avi autoaudiosrc !
decodebin ! audioconvert ! queue ! mux.
Just like on Mac, replacing splitmuxsink with avimux ! filesink works. I'm sure I'm just missing out on some 'pipeline' logic so any clarifiction that can push me in the right direction would be helpful.
I needed to send the audio stream to the audio track of the muxer like so: mux.audio_0
gst-launch-1.0 -ev ksvideosrc ! video/x-raw !
videoconvert ! queue !
splitmuxsink max-size-time=1000000000 muxer=avimux name=mux
location=video%04d.avi autoaudiosrc !
decodebin ! audioconvert ! queue ! mux.audio_O
This happens when the documentation should be clear but you're missing out on some basic knowledge on how to interpret it.
I have a gstreamer pipeline that streams using :
v4l2src ! x264enc ! rtph264pay pt=96 ! udpsink host=ip port=8554
And this pipeline that receives this stream :
/ queue ! avdec_h264 ! appsink
udpsrc ! capsfilter ! rtpjitterbuffer ! rtph264depay ! tee !
\ queue ! h264parse ! mp4mux ! filesink
Simplified receiver pipeline without the tee is :
gst-launch-1.0 udpsrc port=8080 caps="lots-of-caps" ! rtpjitterbuffer ! rtph264depay ! h264parse ! mp4mux ! filesink location=/home/rish/Desktop/recorded.264 -e
Question :
Is there a way to save the H264 encoded stream received from udpsrc without having to re-encode it? How do I correctly close the filesink?
What I've tried so far : The discussion from this thread suggests the pipeline I've tried above but file is still corrupt. (not correctly closed).
This question asks a similar question. However I do not want to decode and re-encode. Another answer in the thread suggests using matroskamux element instead of mp4mux. This works, but I'd rather prefer using mp4mux (no particular reason, but I'd like to know why matroskamux works and mp4mux doesn't).
Your pipeline is already muxing without re-encoding, there is no encoder on your pipeline. h264parse is just a parser.
you've already got an answer on how to close the stream here: Sending EoS to filesink while removing branch from tee
I have been working on an application where I use rtspsrc to gather audio and video from one network camera to another. However I can not watch the stream from the camera and thereby cant verify that the stream works as intended. To verify that the stream is correct I want to record it on a SD card and then play the file on a computer. The problem is that I want the camera to do as much of the parsing, decoding, depayloading as possible since that is the purpose of the application.
I thereby have to separate the audio and video streams by a demuxer and do the parsing, decoding etc and thereafter mux them back into a matroska file.
The video decoder has been omitted since it is not done yet for this camera.
Demux to live playback sink(works)
gst-launch-0.10 -v rtspsrc location="rtsp://host:pass#192.168.0.91/XXX/XXXX?resolution=1280x720&audio=1&audiocodec=g711&audiosamplerate=8000&audiobitrate=64000" latency=0 name=d d. ! rtppcmudepay ! mulawdec ! audioresample ! audioconvert ! autoaudiosink d. ! rtph264depay ! ffdec_h264 ! queue ! ffmpegcolorspace ! autovideosink
Multiple rtspsrc to matroska(works)
gst-launch-1.0 -v rtspsrc location="rtsp://host:pass#192.168.0.91/XXX/XXXX?audio=1&audiocodec=g711&audiosamplerate=8000&audiobitrate=64000" latency=0 ! rtppcmudepay ! mulawdec ! audioresample ! audioconvert ! queue ! matroskamux name=mux ! filesink location=/var/spool/storage/SD_DISK/testmovie.mkv rtspsrc location="rtsp://root:pass#192.168.0.91/axis-media/media.amp?resolution=1280x720" latency=0 ! rtph264depay ! h264parse ! mux.
Single rtspsrc to matroska(fails)
gst-launch-1.0 -v rtspsrc location="rtsp://host:pass#192.168.0.91/XXX/XXXX?resolution=1280x720&audio=1&audiocodec=g711&audiosamplerate=8000&audiobitrate=64000" latency=0 name=d d. ! queue ! rtppcmudepay ! mulawdec ! audioresample ! audioconvert ! queue ! matroskamux name=mux d. ! queue ! rtph264depay ! h264parse ! queue ! mux. ! filesink location=/var/spool/storage/SD_DISK/testmoviesinglertsp.mkv
The last example fails with the error message
WARNING: erroneous pipeline: link without source element
Have i missunderstood the usage of matroska mux and why does the 2 above examples work but not the last?
The problem is here:
queue ! mux. ! filesink
You need to do
queue ! mux. mux. ! filesink
mux. means that gst-launch should select a pad automatically from mux. and link it. You could also specify manually a name, like mux.src. So syntactically you are missing another element/pad there to link to the other element.
I want to listen to an audiostream from alsasrc continuously, and at the same time be able to save snippets to file. I will push a button with 'record' or 'stop'.
I think I need the following gst-launch-0.10 command to work:
gst-launch-0.10 alsasrc do-timestamp=true ! tee name=t ! queue ! alsasink t. ! queue ! audioconvert ! wavenc ! output-selector name=s s. ! filesink location=test1.wav s. ! fakesink
Where I program the output-selector to switch between the filesink and the fakesink when I push the record/stop-button, I know gst-plugins-bad/tests/icles/output-selector-test.c example in the plugins, and want to hack that a bit.
Now the problem arises in the outputselector, it creates the file test1.wav but does not write to it. To focus on this problem I created:
gst-launch-0.10 audiotestsrc is-live=true do-timestamp=true ! wavenc ! output-selector name=s s. ! filesink location=test1.wav s. ! filesink location=test2.wav
and this also does not work (while gst-launch-0.10 audiotestsrc is-live=true do-timestamp=true ! wavenc ! filesink location=test1.wav works as expected). The 2 files are created but not written to. Can anybody point me in the right direction?
In posting "[gst-devel] how to link multiple filesinks to a output-selector before playing pipeline" I read that the 2nd sink is blocking on preroll. That why the example in output-selector-test.c uses a live-src as a trick, I also do that with audiotestsrc but it does not do the trick for me.
Prerolling is not your friend here, so set async=0 on both filesink and fakesink:
gst-launch-0.10 command to work: gst-launch-0.10 alsasrc do-timestamp=true ! tee name=t ! queue ! alsasink t. ! queue ! audioconvert ! wavenc ! output-selector name=s s. ! filesink location=test1.wav async=0 s. ! fakesink async=0
What about putting the wavenc element into the s. ! filesink location=test1.wav branch? wavenc will do some work when the stream starts and stops. output-selector can only ensures that this work if the elements sits behind it.