I am attempting to demux a live recording from a MiniDV camera using the dv1394src element and then transcode it into a vorbis/theora ogg file. My pipeline below stalls after a few seconds. I think I have queue elements in the right space.
gst-launch -e dv1394src ! dvdemux name=demux \
oggmux name=mux ! queue ! filesink location=/tmp/test.ogg \
demux. ! queue ! audioconvert ! vorbisenc ! queue ! mux. \
demux. ! queue ! dvdec ! ffmpegcolorspace ! theoraenc ! queue ! mux.
If I remove the muxer and add filesink end points to the video and audio paths then it does not stall, but that creates two files I have to mux afterwards. I would rather do it in one pipeline.
You could try using a multiqueue after the demuxer. The multiqueue might be able to balance the amount of queued data better.
Related
I'm new to GStreamer and I'm trying to create a pipeline to display a video and record it at the same time. I've managed to make the display part using:
ss << "filesrc location=/home/videos/video1.avi ! avidemux name=demux demux.video_0 ! mpeg4videoparse ! avdec_mpeg4 ! nvvidconv ! video/x-raw,format=I420 ! appsink name=mysink";
Also, I've read that filesink location=somepath is used for saving data into a file but I don't know how combine it with the rest of the pipeline.
So, how do I use appsink and filesink in the same pipeline?
GStreamer offers a tee element for such cases. Note however that in most cases you will want a queue after each branch of a tee to prevent deadlocks. E.g.
filesrc location=/home/videos/video1.avi ! avidemux name=demux demux.video_0 ! mpeg4videoparse ! avdec_mpeg4 ! nvvidconv ! video/x-raw,format=I420 ! tee name=mytee ! queue ! appsink name=mysink mytee. ! queue ! filesink location=out.raw
I am a beginner with gstreamer so bear with me.
I have a working pipeline where audio and video from a test source is sent to the webrtcbin element used to send out offer. Pipeline is as follows:
PIPELINE_DESC = '''
webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302
audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
videotestsrc is-live=true pattern=ball ! video/x-raw,width=320,height=240 ! videoconvert ! queue ! x264enc ! rtph264pay !
queue ! application/x-rtp,media=video,encoding-name=H264,payload=97 ! sendrecv.
'''
However doing this consumes a lot of CPU/Memory as gstreamer has to encode audio/video. Hence I was to use a pre-recorded file to lower the resource usage.
I want to use a sample file (sample.mp4) to send audio and video to the webRTCbin element. The mp4 file has H264 video and AAC audio. I have tried a lot of combinations of elements but it is not working. Could you please help me correct my pipeline?
PIPELINE_DESC = '''
webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302
filesrc location=sample.mp4 ! decodebin ! audioconvert ! sendrecv.
filesrc location=sample.mp4 ! decodebin ! videoconvert ! sendrecv.
'''
Many thanks in advance.
mp4 file is a container file format and it needs to be demultiplexed to get video and audio. For that purpose, you can use GStreamer's qtdemux element.
Considering above, example pipeline could be something like this
PIPELINE_DESC = '''
filesrc location=test.mp4 ! qtdemux name=demux
webrtcbin name=sendrecv stun-server=stun://stun.l.google.com:19302
demux.audio_%u ! aacparse ! rtpmp4apay !
queue ! application/x-rtp,media=audio,encoding-name=MP4A-LATM,payload=96 ! sendrecv.
demux.video_%u ! h264parse ! rtph264pay !
queue ! application/x-rtp,media=video,encoding-name=H264,payload=97 ! sendrecv.
'''
I'm trying to use this command to create multiple files from stream but they have no audio playback, I think decodebin should be dealing with it, what am I doing wrong?
gst-launch-1.0 -e filesrc location=video.mp4 ! queue ! decodebin ! queue ! videoconvert ! queue ! timeoverlay ! x264enc key-int-max=10 ! h264parse ! splitmuxsink location=videos/test%02d.mp4 max-size-time=1000000000000
Why do you make that assumption that decodebin will handle it? decodebin will decode the audio track to raw audio and exposes an audio pad. If you don't make use of that pad it will not make itself into the file.
Since you transcode you will have to re-encode the audio too:
gst-launch-1.0 -e filesrc location=video.mp4 ! queue ! decodebin ! queue ! \
videoconvert ! queue ! timeoverlay ! x264enc key-int-max=10 ! h264parse ! \
splitmuxsink location=videos/test%02d.mp4 max-size-time=1000000000000 \
decodebin0. ! queue ! voaacenc ! aacparse ! splitmuxsink0.
If you don't want to re-encode but passthrough the audio decodebin is the wrong way. parsebin may be a better fit in that case.
I am using the following pipeline to convert an flv file to mp4.
gst-launch-1.0 -vvv -e filesrc location="c.flv" ! flvdemux name=demux \
demux.audio ! queue ! decodebin ! audioconvert ! faac bitrate=32000 ! mux. \
demux.video ! queue ! decodebin ! videoconvert ! video/x-raw,format=I420 ! x264enc speed-preset=superfast tune=zerolatency psy-tune=grain sync-lookahead=5 bitrate=480 key-int-max=50 ref=2 ! mux. \
mp4mux name=mux ! filesink location="c.mp4"
The problem is when (for example) audio is missing, the pipeline gets stuck. (Same thing happens if just hooking a fakesink to demux.audio).
I need a way for the filters to ignore missing tracks, or produce empty tracks.
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.