I am following a video tutorial online to stream low latency video and audio using gstreamer.
Here is the video link: https://youtu.be/mNQTORvhQ6Q
I have installed all the gstreamer dependencies and plugins on both the client and server and the rtsp package on the server also. The server runs with no issues but when I try run the client it has an error and ends the pipeline. I have tried some video only examples and it does indeed work so it's something to do with the pipeline I am using.
Here is the server pipeline running from a Raspberry Pi 4:
Ran from inside the /gst-rtsp-server-1.14.4/examples folder:
./test-launch --gst-debug=0 "( alsasrc device=hw:2,0 ! "audio/x-raw,channels=1,rate=48000" ! audioconvert ! opusenc ! rtpopuspay name=pay1 pt=97 v4l2src device=/dev/video0 ! "image/jpeg,width=800,height=600,frame-rate=30/1" ! rtpjpegpay name=pay0 pt=96 )"
Here is the pipeline on the client, which is a Ubuntu PC:
gst-launch-1.0 rtspsrc latency=0 location=rtsp://192.168.127.219:8554/test name=src src. ! "application/x-rtp, channels=1, media=audio, rate=48000, encoding-name=OPUS" ! rtpjitterbuffer ! rtpopusdepay ! opusdec ! audioconvert ! jackaudiosink src. ! "application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)JPEG" ! rtpjitterbuffer ! rtpjpegdepay ! jpegdec ! videoconvert ! autovideosink
It has these errors:
Setting pipeline to PAUSED ...
ERROR: Pipeline doesn't want to pause.
ERROR: from element /GstPipeline:pipeline0/GstJackAudioSink:jackaudiosink0: Jack server not found
Additional debug info:
gstjackaudiosink.c(355): gst_jack_ring_buffer_open_device (): /GstPipeline:pipeline0/GstJackAudioSink:jackaudiosink0:
Cannot connect to the Jack server (status 17)
Setting pipeline to NULL ...
Freeing pipeline ...
I have tested the output of jackaudiosink on its own with a test tone and it also works fine, so I assume it's specifically something about this pipeline that I haven't got quite right :(
Any help is much appreciated :)
Have you tried to put 'autoaudiosink' instead of 'jackaudiosink'? Like that:
gst-launch-1.0 rtspsrc latency=0 location=rtsp://192.168.127.219:8554/test name=src src. ! "application/x-rtp, channels=1, media=audio, rate=48000, encoding-name=OPUS" ! rtpjitterbuffer ! rtpopusdepay ! opusdec ! audioconvert ! autoaudiosink src. ! "application/x-rtp, media=(string)video, payload=(int)96, clock-rate=(int)90000, encoding-name=(string)JPEG" ! rtpjitterbuffer ! rtpjpegdepay ! jpegdec ! videoconvert ! autovideosink
Related
I'm using mediasoup create plaintransport then I forward from udpsrc to tcpserversink like this:
gst-launch-1.0 udpsrc port=57616 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,payload=(int)100,encoding-name=(string)OPUS,ssrc=(uint)613744965" ! rtpopusdepay ! opusdec ! tcpserversink port=23333 host=0.0.0.0
on Client:
gst-launch-1.0 tcpclientsrc port=23333 host=11.22.33.44 ! rawaudioparse ! decodebin ! audioconvert ! audioresample ! autoaudiosink
the problems is audio stream always delay 2,3 and increase time by time. and I have warning like this
gstrtpbasedepayload.c(505): gst_rtp_base_depayload_handle_buffer (): /GstPipeline:pipeline0/GstRTPOpusDepay:rtpopusdepay0:
Received invalid RTP payload, dropping
WARNING: from element /GstPipeline:pipeline0/GstRTPOpusDepay:rtpopusdepay0: Could not decode stream.
please help me solve this and improve stream delay problems
Is it possible to write rtpopus to a file, then read it back with gstreamer? It seems simple but I'm getting nowhere with it and can't seem to find any information online. Here is my attempt:
gst-launch-1.0.exe audiotestsrc ! opusenc ! rtpopuspay ! filesink location=test.opus
Then, close and run:
gst-launch-1.0.exe filesrc location="test.opus" ! rtpopusdepay ! fakesink dump=true
gstreamer fails with:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
ERROR: from element /GstPipeline:pipeline0/GstFileSrc:filesrc0: Internal data stream error.
Additional debug info:
../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstFileSrc:filesrc0:
streaming stopped, reason error (-5)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
Freeing pipeline ...
I don't think it could work. RTP is related to UDP packetization so it would work when streaming over UDP.
You'd better use a file container supporting opus audio such as matroskamux:
gst-launch-1.0 -e audiotestsrc ! audioconvert ! opusenc ! matroskamux ! filesink location=test.mkv
# Let play for 5s and stop with Ctrl-C
# Replay:
gst-launch-1.0 filesrc location=test.mkv ! matroskademux ! opusdec ! audioconvert ! autoaudiosink
I have a working gstreamer pipeline, using videobalance to adjust the contrast and brightness of a camera stream, the output of which is displayed on screen:
gst-launch-1.0 nvarguscamerasrc sensor-id=0 saturation=0 !
"video/x-raw(memory:NVMM),width=1920,height=1080,framerate=30/1" !
nvvidconv ! videobalance contrast=1.5 brightness=-0.3 ! nvoverlaysink
I want to do the same again, but this time record the camera stream to a file. I tried adding the videobalance element to the pipeline suggested by the authors of the drivers I'm using (which works fine otherwise):
gst-launch-1.0 nvarguscamerasrc sensor-id=0 saturation=0 !
"video/x-raw(memory:NVMM),width=1920,height=1080,framerate=30/1" !
nvv4l2h264enc ! videobalance contrast=1.5 brightness=-0.3 ! h264parse !
mp4mux ! filesink location=test.mp4 -e
But, I get the error:
WARNING: erroneous pipeline: could not link nvv4l2h264enc0 to videobalance0
Any suggestions for where I'm going wrong and/or possible solutions would be greatly appreciated.
NVidea encoders use NVMM memory, so it can't directly connect encoder to videobalance. Just adding simple video convert will solve the problem:
gst-launch-1.0 nvarguscamerasrc sensor-id=0 saturation=0 !
"video/x-raw(memory:NVMM),width=1920,height=1080,framerate=30/1" !
nvv4l2h264enc ! videoconvert ! videobalance contrast=1.5 brightness=-0.3 ! h264parse !
mp4mux ! filesink location=test.mp4 -e
I have trouble streaming pulse audio monitor via rtp to an audio player like vlc or gst-launch with udpsrc
this command works and the file has audio that is currently being played
gst-launch-1.0 -v pulsesrc device = "alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" ! opusenc ! oggmux ! filesink location=test.ogg
but when i use this,
gst-launch-1.0 -v pulsesrc device="alsa_output.pci-0000_00_1b.0.analog-stereo.monitor" ! opusenc ! rtpopuspay ! udpsink host=0.0.0.0 port=4000
vlc (from an android phone) tells me that it cannot play the stream with uri rtp://ip-addr:4000
and gst-launch from same machine starts but the resulting file is empty.
gst-launch-1.0 -v udpsrc uri=rtp://0.0.0.0:4000 ! rtpopusdepay ! oggmux ! filesink location=test.ogg
gstreamer version is
$ gst-launch-1.0 --version
gst-launch-1.0 version 1.16.0
GStreamer 1.16.0
Just started this account, I wanted to add this as a comment, but couldn't because of rep limitations.
I'm not experienced in using vlc, but at least I get your GStreamer pipelines working if I add the caps, definitions of the rtp stream parameters, to the rtpopusdepay.
So instead of:
gst-launch-1.0 -v udpsrc uri=rtp://0.0.0.0:4000 ! rtpopusdepay ! oggmux ! filesink location=test.ogg
you'll need to use:
gst-launch-1.0 udpsrc uri=udp://0.0.0.0:4000 ! application/x-rtp,payload=96,encoding-name=OPUS ! rtpopusdepay ! opusdec ! autoaudiosink
for GStreamer. The mandatory parts are the payload and encoding-name, others you can find from gst-inspect-1.0 rtpopuspay/rtpopusdepay. You might need to change the numbers depending on what you define on the server side/what's the default on your machine.
So in conclusion, I got that GStreamer pipeline working by moving the rtp definitions to the caps for rtpopusdepay. As I said, I'm not familiar with vlc, so I don't know how to define those GStreamer caps there, if it even depends on those, but I hope this gives some insight on your work.
i am using g streamer-0.10 on Ubuntu os for streaming an web cam video on to an rtmp server i am getting an video output but their is a problem in audio . Below command used for streaming
gst-launch-0.10 v4l2src ! videoscale method=0 ! video/x-raw-yuv,width=852,height=480,framerate=(fraction)24/1 ! ffmpegcolorspace ! x264enc pass=pass1 threads=0 bitrate=900 tune=zerolatency ! flvmux name=mux ! rtmpsink location='rtmp://..../live/testing' demux. alsasrc device="hw:0,0" ! audioresample ! audio/x-raw-int,rate=48000,channels=2,depth=16 ! pulseaudiosink
Blockquote
by running the above command i got an error
gstbaseaudiosrc.c(840): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 13920 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.
Blockquote
so the audio is not audible.
Help me out to solve this problem.
Thanks in advance
Ameeth
I don't understand your pipeline. What is "demux." in the middle?
The problem you are facing is because you have not seperated your elements with queues. Keep a queue before your sinks and after your sources to give the rest all seperate threads to run. It should allow get rid of the issue.
Since I don't have pulse audio or rtmp reciever in my system i have tested out the following and it works.
gst-launch-0.10 v4l2src ! ffmpegcolorspace ! queue ! x264enc pass=pass1 threads=0 bitrate=900000 tune=zerolatency ! queue ! flvmux name=mux ! fakesink alsasrc ! queue ! audioresample ! audioconvert ! queue ! autoaudiosink
You can change it accordingly and use it. The only thing I had to do to make it work and remove the error your are facing is to add the queues.
For me (Logitech c920 on Raspberry Pi3 w/ GStreamer 1.4.4) I was able to get rid of the "Dropped samples" warning by using audioresample to set the sampling rate of the alsasrc to something that flvmux liked. From gst-inspect-1.0 flvmux, it looks like flvmux only supports 5512, 11025, 22050, 44100 sample rates for x-raw and 5512, 8000, 11025, 16000, 22050, 44100 for mp4. Here's my working pipeline
gst-launch-1.0 -v -e \
uvch264src initial-bitrate=800000 average-bitrate=800000 iframe-period=2000 device=/dev/video0 name=src auto-start=true \
src.vidsrc ! video/x-h264,width=864,height=480,framerate=30/1 ! h264parse ! mux. \
alsasrc device=hw:1 ! 'audio/x-raw, rate=32000, format=S16LE, channels=2' ! queue ! audioresample ! "audio/x-raw,rate=44100" ! queue ! voaacenc bitrate=96000 ! mux. \
flvmux name=mux ! rtmpsink location="rtmp://live-sea.twitch.tv/app/MYSTREAMKEY"
I was surprised that flvmux didn't complain about getting an audio source that was at an unsupported sampling rate. Not sure if that's expected behavior.