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
Related
I'm using the following pipeline to listen to a RTSP stream and save a video file:
gst-launch-1.0 -q rtspsrc location=rtsp://<ip>:<port>/video ! decodebin ! autovideoconvert ! x264enc pass=5 quantizer=25 speed-preset=6 ! h264parse ! matroskamux ! filesink location=<filename>
But even though I can see the files generated, they lack the duration of the video when playing on VLC.
I can fix it by passing it through ffmpeg later, but I want to generate the video from gstreamer already completely valid. How can I fix this pipeline?
gst-launch-1.0 -e rtspsrc location=rtsp://<ip>:<port>/video ! decodebin ! videoconvert ! x264enc ! mp4mux ! filesink location="filename.mp4"
This will create a video with duration shown correctly
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
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.
I am trying to create a GStreamer pipeline (v 1.0) in order to record and play special file format.
For recording purpose I use the following pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw-yuv, format=\(fourcc\)I420, width=640, height=480 ! videoconvert ! x264enc byte-stream=1 ! queue ! appsink
In appsink (using new_sample() callback) I use a compression method to compress H264 stream and finally store in a output file.
I use the following pipeline to play the recorded file:
gst-launch-1.0 appsrc ! video/x-h264 ! avdec_h264 ! autovideosink
In appsrc I decompress H264 stream and send it to appsrc buffer (using push-buffer). The size of each buffer is 4095.
Unfortunately GStreamer after push 2 buffers print the following debug message:
Error: Internal data flow error.
Is there any way to fix the problem?
Add legacyh264parse or h264parse (depending on your version of gst components) before your decoder. You need to be able to send full frames to the decoder.
Post avdec_h264 it would be nice to have a ffmpegcolorspace to be able to convert the video format to your display requirements.
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.