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.
'''
Related
As title, how can I change this so it also plays the files audio too?
gst-launch-1.0 filesrc location='/usr/share/myfile.mp4' ! qtdemux ! h264parse ! imxvpudec ! imxipuvideosink framebuffer=/dev/fb2 &
The I can get the file to play with audio using
gst-launch-1.0 -v playbin uri=file:///path/to/somefile.mp4
But I need the output to be onto device fb2 like in the first example
Many thanks
I posted a link to this question into the gstreamer reddit and a hero called Omerzet saved the day.
The following is the solution:
gst-launch-1.0 filesrc location='/usr/share/myfile.mp4' ! qtdemux name=demux demux.video_0 ! queue ! h264parse ! imxvpudec ! imxipuvideosink framebuffer=/dev/fb2 demux.audio_0 ! queue ! decodebin ! audioconvert ! audioresample ! alsasink device="sysdefault:CARD=imxhdmisoc"
Where framebuffer diverts the video to device /dev/fb2.
And
alsasink device="sysdefault:CARD=imxhdmisoc"
Diverts the audio to my define sound card.
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'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'm working on gstreamer.Is there any way to store the output of mpegtsdemux element in a pipeline to a file as I'm interested in seperating audio,video ts packets into different files.
You can seperate video and audio track after mpegtsdemux. I hope this exemple will help you:
gst-launch filesrc location="source.ts" ! mpegtsdemux name=demux ! queue max-size-buffers=400000000 ! decodebin ! videorate ! videoscale ! ffenc_mpeg4 ! matroskamux ! filesink location="your_video_file.mkv" demux. ! queue max-size-buffers=400000000 ! decodebin ! audioconvert ! wavenc! filesink location="your_audio_file.wav"
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.