I'm trying to mux audio & video to rtspclientsink.
My pipeline :
gst-launch-1.0 -v libcamerasrc ! video/x-raw, width=640, height=480, framerate=30/1 ! videoconvert ! videoscale ! clockoverlay time-format="%D %H:%M:%S" ! x264enc speed-preset=ultrafast bitrate=600 key-int-max=40 ! queue ! qtmux0. autoaudiosrc ! voaacenc ! qtmux ! rtspclientsink location=rtsp://localhost:8554/mystream
WARNING: erroneous pipeline: could not link qtmux0 to rtspclientsink0
I'm feeding rtsp-simple-server docker image to which handle the rtsp server side.
Could you help me how to fix my pipeline?
Many thanks for the help.
Only video pipeline is working:
gst-launch-1.0 libcamerasrc ! video/x-raw, width=640, height=480, framerate=30/1 ! videoconvert ! videoscale ! clockoverlay time-format="%D %H:%M:%S" ! x264enc speed-preset=ultrafast bitrate=600 key-int-max=40 ! queue ! rtspclientsink location=rtsp://localhost:8554/mystream
but missing the audio obviously.
Related
I have googled it all but I couldn't find solution to my problem. I will be happy if anyone had similiar need and resolved somehow.
I do stream to RTMP server by following command. It captures video from HDMI Encoder, crops, rotates video.
gst-launch-1.0 -e v4l2src device=/dev/v4l/by-path/platform-fe801000.csi-video-index0 ! video/x-raw,format=UYVY,framerate=20/1 ! videoconvert ! videoscale ! video/x-raw, width=1280,height=720 ! videocrop top=0 left=0 right=800 bottom=0 ! videoflip method=counterclockwise ! omxh264enc ! h264parse! flvmux name=mux streamable=true ! rtmpsink sync=true async=true location='rtmp://XXXXX live=true'
and I want to add Audio to existing microphone on Raspberry. For example I can record microphone input to wav file by below pipeline.
gst-launch-1.0 alsasrc num-buffers=1000 device="hw:1,0" ! audio/x-raw,format=S16LE ! wavenc ! filesink location = a.wav
My question is; how can I add audio to my existing command line which streams to RTMP Server? And also, when I capture audio to file, there is a lots of Noise. How can I avoid?
Thank you
I have combined Audio & Video. But I have still Noise on Audio.
gst-launch-1.0 -e v4l2src device=/dev/v4l/by-path/platform-fe801000.csi-video-index0 ! video/x-raw,format=UYVY,framerate=20/1 ! videoconvert ! videoscale ! video/x-raw, width=1280,height=720 ! videocrop top=0 left=0 right=800 bottom=0 ! videoflip method=counterclockwise ! omxh264enc ! h264parse! flvmux name=mux streamable=true ! rtmpsink sync=true async=true location='rtmp://XXXXXXXXXXXXXXXX' alsasrc device="hw:1,0" ! queue ! audioconvert ! audioresample ! audio/x-raw,rate=44100 ! queue ! voaacenc bitrate=128000 ! audio/mpeg ! aacparse ! audio/mpeg, mpegversion=4 ! mux.
I have kind of resolved Noise by following code. but still not so good.
"ffmpeg -ar 48000 -ac 1 -f alsa -i hw:1,0 -acodec aac -ab 128k -af 'highpass=f=200, lowpass=f=200' -f flv rtmp://XXXXX.XXXXXXX.XXXXX/LiveApp/"+ str(Id) + "-" + str(deviceId)+"-Audio"
I cant save audio from stream I get only video in file. I suspect that I do not need two filesink in pipeline or there is some problem two different mux.
I tried to use autoadiosink and autovideosink and they works successfully.
autoadiosink and autovideosink pipeline:
gst-launch-1.0 rtspsrc location=rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov latency=0 droponlatency=1 name=rtp_source ! queue ! rtph264depay ! decodebin ! videoconvert ! autovideosink rtp_source. ! queue ! decodebin ! autoaudiosink
Save to file filesink pipeline:
gst-launch-1.0 rtspsrc location=rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov latency=0 droponlatency=1 name=rtp_source ! queue ! rtph264depay ! decodebin ! vp8enc ! webmmux ! filesink location=BigBuckBunny_115k.webm rtp_source. ! "application/x-rtp, media=(string)audio" ! queue ! decodebin ! vorbisenc ! oggmux ! filesink location=BigBuckBunny_115k.webm
I want to get also audio in resulting file.
You just reuse the existing mux - so that the vorbis is put into the webmmux too:
gst-launch-1.0 rtspsrc location=rtsp://184.72.239.149/vod/mp4:BigBuckBunny_115k.mov \
latency=0 droponlatency=1 name=rtp_source ! queue ! rtph264depay ! decodebin ! \
vp8enc ! webmmux name=mux ! filesink location=BigBuckBunny_115k.webm rtp_source. ! \
application/x-rtp, media=(string)audio" ! queue ! decodebin ! vorbisenc ! mux.
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 am trying to construct a RTSP pipeline on the client side to receive audio and video streams on android platform
Only video pipeline works fine
data->pipeline = gst_parse_launch("rtspsrc location=rtsp://192.168.1.100:8554/ss ! gstrtpjitterbuffer ! rtph264depay ! h264parse ! amcviddec-omxtiducati1videodecoder ! ffmpegcolorspace ! autovideosink",&error);
I need to receive audio streams also, so I tried with below pipeline
gst-launch rtspsrc location=rtsp://192.168.1.100:8554/ss demux. ! queue ! rtph264depay ! h264parse ! ffdec_h264 ! autovideosink demux. ! queue ! rtpmp4gdepay ! aacparse ! ffdec_aac ! audioconvert ! autoaudiosink
Gstreamer throws error saying no element "demux"
Please let me know proper rtsp pipeline to receive audio and video streams on android
Please try this, (tested):
gst-launch rtspsrc location=rtsp://192.168.1.100:8554/ss name=demux. ! queue ! rtph264depay ! h264parse ! ffdec_h264 ! autovideosink demux. ! queue ! rtpmp4gdepay ! aacparse ! ffdec_aac ! audioconvert ! autoaudiosink
I'm using the following pipeline (SIMPLIFIED) in Gstreamer OSS Build 0.10.7 on Win 7 x64:
udpsrc ! application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96 !
gstrtpjitterbuffer latency=200 ! rtph264depay ! tee name=h264Tee
h264Tee. ! queue ! h264parse ! mux.
matroskamux name=mux ! filesink location=rec.mkv sync=false // same for avimux/mp4/qt
h264Tee. ! queue ! ffdec_h264 ! tee name=videoTee
//.videoTee ! queue ! dx9videosink
//.videoTee ! queue ! appsink
//udpsrc ! queue ! directsoundsink
audiotestsrc ! mux. //only for testing, should be connected to udpsrc
The pipeline is launched via Gstreamer-Sharp.
Here's the console output of the pipeline:
WARN default xoverlay.c:354:gst_x_overlay_set_xwindow_id:<videoSink> Using deprecated gst_x_overlay_set_xwindow_id()
ERROR d3dvideosink d3dvideosink.c:2204:gst_d3dvideosink_release_swap_chain: Direct3D device has not been initialized
WARN bin gstbin.c:2378:gst_bin_do_latency_func:<pipeline0> failed to query latency
WARN matroskamux matroska-mux.c:970:gst_matroska_mux_video_pad_setcaps:<mux> pad video_0 refused caps 05370C40
Both video and audio are playing just fine as long as I leave out the muxer. When include the muxer in the pipeline, the video freezes immedeately and no sound can be heard. What's wrong why does the muxer refuse the caps?
Ok solved it my self:
The video caps above don't contain sprop-parameter-sets which aren't needed for playback. For encoding however they are needed since various properties of the stream are encoded within these:
udpsrc !
application/x-rtp, media=(string)video, clock-rate=(int)90000,
encoding-name=(string)H264,
sprop-parameter-sets= (string)\"Z0LADdkBQfsBEAAAAwAQAAADAyjxQqSA\\,aMuMTIA\\=\",
payload=(int)96,
ssrc (uint)2332354585,
clock-base=(uint)1158355497,
seqnum-base=(uint)10049 !
gstrtpjitterbuffer latency=200 ! rtph264depay ! tee name=h264Tee
...