I'm having an issue with pulling audio and video from an RTSP stream using gstreamer.
The command I am using to test is as follows:
gst-launch-1.0 rtspsrc location=rtsp://192.168.50.160/whp name=src src. ! queue ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! x264enc bitrate=10000 ! rtph264pay ! udpsink host=192.168.50.164 port=8004 src. ! queue ! fakesink
The result of the above is that the pipe follows through for the first (video) stream. The second stream however is untouched and seems to sit in the rtspsrc plugin.
The way I am finding this is by looking at the resultant dot file:
If I am reading this right it looks like the queue connects correctly to rtpsession0, but seems to ignore rtpsession1 and the second queue doesn't connect to anything resulting in audio from my stream being completely ignored.
Am I reading this incorrectly? If not am I missing something in my pipeline command that would rectify this issue?
I am happy to provide any more information necessary
Thanks
Related
I'm a newbie to gstreamer so i would be appreciated if you could help me.
I'm trying to listen to a pipeline and record frames to a file.
I have tried the following pipeline:
gst-launch-1.0 udpsrc port=5600 do-timestamp=true ! application/x-rtp, payload=96 ! rtph264depay ! avdec_h264 ! clockoverlay ! jpegenc ! avimux ! filesink location=stream.avi
I want to record whole timeline even if the sender doesn't provide any frame data.
In default, recorder appends the frames when pipeline receive some valid frames. But I want to see some black frames when sender doesn't send data.
I experimented a bit and I don't think you'll be able to do this with a plain gst-launch command. Unfortunately what it would probably involve is to write an application that detects when packets/buffers are not coming in any more, and then modifying the pipeline. If you want to give it a go I'd suggest the input-selector element in something like this:
gst-launch-1.0 videotestsrc pattern=black ! video/x-raw ! input-selector name=selector ! clockoverlay ! jpegenc ! avimux ! filesink location=stream.avi
Then I'd create a method to attach the stream to the input-selector:
udpsrc port=5600 do-timestamp=true ! application/x-rtp, payload=96 ! rtph264depay ! avdec_h264 ! identity name=buffer-checker
To detect no packets coming in, you can listen for the handoff signal on the identity element, and then remove the stream when it times out and switch over to the black test pattern from the videotestsrc by using the active-pad property on the input-selector.
Using the videomixer element almost works, but I don't believe it will handle multiple stops and starts of the stream.
Anyway, hope someone else comes up with a better idea. You could also re-analyze your top level approach and see if there is a way you can work with multiple video clips instead of the one.
I would like to store a file which has AAC audio frames,
For that i used the below pipeline,
gst-launch-1.0 filesrc location=Test_44100Hz_2ch_s16le.wav ! "audio/x-raw,rate=44100,format=s16le,channels=2" ! audioparse format=raw raw-format=s16le rate=44100 channels=2 ! faac ! aacparse ! queue ! filesink location=a1
While reading that file again to pulsesink using below pipeline,
gst-launch-1.0 filesrc location=a1 ! aacparse ! faad ! audioconvert ! audioresample ! pulsesink
I am Receiving below error, I used GST_DEBUG=3, but i am not able find the solution.
0:00:00.031924804 3379 0x2231d60 WARN basesrc gstbasesrc.c:3483:gst_base_src_start_complete:<filesrc0> pad not activated yet
Pipeline is PREROLLING ...
0:00:00.033044700 3379 0x2231050 WARN baseparse gstbaseparse.c:3255:gst_base_parse_loop:<aacparse0> error: No valid frames found before end of stream
ERROR: from element /GstPipeline:pipeline0/GstAacParse:aacparse0: No valid frames found before end of stream
Additional debug info:
gstbaseparse.c(3255): gst_base_parse_loop (): /GstPipeline:pipeline0/GstAacParse:aacparse0
ERROR: pipeline doesn't want to preroll.
Can anybody help me, To solve this? I need to store AAC audio frames and need to stream that file as AAC audio stream.
This is it, tested working:
gst-launch-1.0 filesrc location=WAV_44_16bit.wav ! decodebin ! audioconvert ! queue ! voaacenc ! aacparse ! queue ! mp4mux ! filesink location=aac.mp4
gst-launch-1.0 filesrc location=aac.mp4 ! decodebin ! audioconvert ! audioresample ! alsasink
In container there are metadata information stored.. without them the decoder does not know how to process the data.
AAC Audio streams require a container in order to be useful within gstreamer
For decoder initialization it is necessary to know sampling frequency and Audio Object. In gstreamer we are unable to pass this metadata directly to the parser or the decoder. The parser collects this data instead from the mp4 header then the encoder inherits the frame structure/size and sample rate. So this is a deficiency in either aacparse(parser) or avdec_aac/faad(decoder), none of which have exposed parameters to specify frame size of a raw file, the afore mentioned metadata. That being said, I haven't found a compelling reason why anyone would need to do this. I found myself trying to do it before I discovered the aac simply needed to be muxed into an MP4(mp4mux) or another container to work and be portable. The container/framing only adds a small amount of data to the stream.
everyone
The version of GStreamer I use is 1.x. I've spent a lot of time in searching a way to delete a tee branch.
In an active pipeline, a recording bin is created as below and inserted into this pipeline by branching the tee element.
"queue ! video/x-h264, width=800, height=600, framerate=10/1, stream-format=(string)byte-stream ! h264parse ! mp4mux ! filesink location=/xxxx"
It works perfectly except that I want to dynamically delete the recording bin and get a playable mp4 file. According to some discussion and tutorial, to get a correct mp4 file , we need to handle something about EOS. After trying some methods, I always got broken mp4 files.
Does anyone have sample code written in C to show me ? I'd appreciate your help.
Your best bet for cases like this may be to create two processes. The first process would run the video, and half of the tee it has would deliver h264 data to the second process through whatever means.
Here are two pipelines demonstrating the concept using UDP sockets.
gst-launch-1.0 videotestsrc ! x264enc ! tee name=t ! h264parse ! avdec_h264 ! videoconvert ! ximagesink t. ! queue ! h264parse ! rtph264pay ! udpsink host=localhost port=8888
gst-launch-1.0 udpsrc port=8888 num-buffers=300 ! application/x-rtp,media=video,encoding-name=H264 ! rtph264depay ! h264parse ! mp4mux ! filesink location=/tmp/264.mp4
The trick to getting that clean mp4 is to make sure an EOS event is delivered reliably.
Instead of dynamically adding it you just have it in the pipeline by default, and add a probe callback at the source pad of the queue in the probe callback you have to do the trick either to pass the buffer or not (GST_PAD_PROBE_DROP drops the buffer and GST_PAD_PROBE_OK passes on the buffer to next element) so when you get an event to start/stop recoding you just need to return appropriate values. And filesink you can use multifilesink instead so as to write to different files everytime you start/stop.
Note the queue which drops the buffers needs before the mux element otherwise the file would be corrupt.
Hope that helps!
Finally, I came up with a solution.
Let's say that there is an active pipeline including a recording bin.
"udpsrc port=4444 caps=\"application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H264 ! rtph264depay !
tee name=tp tp. ! queue ! video/x-h264, width=800, height=600,
framerate=10/1 ! decodebin ! videoconvert ! video/x-raw, format=RGBA !
autovideosink"
recording bin:
"queue ! video/x-h264, width=800, height=600, framerate=10/1,
stream-format=(string)byte-stream ! h264parse ! mp4mux ! filesink
location=/xxxx"
After a period of time, we want to stop recording and save as a mp4 file, and video media is still streaming.
First, I use a blocking probe to block the src pad of tee. In this blocking probe callback, I use an event probe to catch EOS in the sink pad of filesink and do a busy waiting.
*if EOS is catched in the event probe callback
self->isGotEOS = YES;
*busy waiting in the blocking probe callback
while (self->isGotEOS == NO) {
usleep(100000);
}
Before entering the busy waiting while loop, an EOS event is created and sent to the sink pad of recording bin.
After the busy waiting is done:
usleep(200000);
[self destory_record_elements];
I think usleep(200000) is a trick. Without it, a non-playable mp4 file is usually the result. It would seem that 200ms is long enough handling the EOS.
I had similar problem previously, my pipeline
videotestsrc do-timestamp="TRUE" ! videoflip method=0 ! tee name=t
t. ! queue ! videoconvert ! glupload ! glshader ! autovideosink async="FALSE"
t. ! queue ! identity drop-probability=1 ! videoconvert name=conv2 ! openh264enc ! h264parse ! avimux ! multifilesink async="FALSE" post-messages=true next-file=4
Then I just change drop-probability property on identity element
drop-probability = 1 + gst_pad_send_event(conv2_sinkpad, gst_event_new_eos()); - stop recording
drop-probability = 0 - resume recording
I used the following GStreamer pipeline to store my encoded stream in a binary file:
gst-launch v4l2src ! videorate ! video/x-raw-yuv, framerate=\(fraction\)10/1 \
! videoscale ! video/x-raw-yuv, format=\(fourcc\)I420, width=640, height=480\
! ffmpegcolorspace ! x264enc ! fdsink > vid.bin
Now i want to play previously recorded files in GStreamer using the following pipeline:
cat vid.bin | gst-launch fdsrc ! ffdec_h264 ! autovideosink
But then it gives the following error:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/ffdec_h264:ffdec_h2640: Internal GStreamer error: negotiation problem. Please file a bug at http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
Additional debug info:
gstffmpegdec.c(2804): gst_ffmpegdec_chain (): /GstPipeline:pipeline0/ffdec_h264:ffdec_h2640:
ffdec_h264: input format was not set before data start
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
I know that the best way to capture video is using Muxers but is there any way to play my previous files?
Thanks
Not sure your pipeline is right.
If you want to write to a file why not simply use filesink and filesrc.
fdsink > vid.bin will not work fine because if you see the prints by gstreamer gst-launch will also go into the file. [Just open vid.bin in an text editor and you will see what I mean].
Also for x264 stream to be stored without a muxer you need to use byte-stream=1 in your x264enc to store it in annexb format so that it is decodable.
To play back raw x264 stream you need to have a color space convertor before the video sink
gst-launch filesrc location=inputfile ! legacyh264parse ! ffdec_h264 ! queue ! ffmpegcolorspace ! autovideosink
plays just fine here at my end
Or, to playback a raw h264 file with gstreamer 1.0:
gst-launch-1.0 filesrc location=/tmp/video.h264 ! h264parse ! avdec_h264 ! autovideosink
I'm having a problem trying to record audio+video from my webcam to a file. If I use videotestsrc and autoaudiosrc I get everything right (read as in I get a file with audio recorded from the webcam's mic, and test-video image), but as soon as I replace videotestsrc with v4l2src (or autovideosrc) I get Error starting streaming on device '/dev/video0'.
The command I'm using:
gst-launch-0.10 videotestsrc ! queue ! ffmpegcolorspace! theoraenc ! queue ! oggmux name=mux autoaudiosrc ! queue ! audioconvert ! vorbisenc ! queue ! mux. mux. ! queue ! filesink location = test.ogg
Why is that happening? What am I doing wrong?
EDIT:
In fact, something as simple as
gst-launch-0.10 autovideosrc ! autovideosink autoaudiosrc ! autoaudiosink
is failing with the same error (Error starting streaming on device '/dev/video0')
Replacing autovideosrc with videotestsrc gives me test image + real audio.
Replacing autoauidosrc with audiotestsrc gives me real image + test audio.
I'm starting to think that this is some kind of limitation of my webcam. Is that possible?
EDIT:
GST_DEBUG=2 log here: http://pastie.org/4755009
EDIT 2:
GST_DEBUG="v4l2*:5" (gstreamer 0.10): http://pastie.org/4810519
GST_DEBUG="v4l2*:5" (gstreamer 1.0): http://pastie.org/4810502
Please do a
gst-launch-1.0 v4l2src ! videoscale ! videoconvert ! autovideosink
Does that run? If not repeat as
GST_DEBUG="v4l2*:5" GST_DEBUG_NO_COLOR=1 gst-launch 2>debug.log ...
and check the log for errors. You also might want to run v4l-info (install v4l-conf under debian/ubuntu) and report what formats your camera supports.