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
Related
I'm new to gstreamer, basically a newbie.
I want to receive an rtmp video, process the video, reencode the video, merge it with the sound from the received video and then send it out as a new rtmp-video. Somehow I can not get get the sound working:
Receiver:
"rtmpsrc location=rtmp://xx.yy.10.40:1935/orig/1 do-timestamp=true ! queue ! flvdemux name=demux demux.video ! h264parse ! video/x-h264 ! nvh264dec ! videoconvert ! appsink"
"demux.audio ! aacparse ! queue ! mp4mux streamable=true ! shmsink socket-path=/tmp/foo sync=true wait-for-connection=false shm-size=100000000"
Please note, I separated the 2 strings simply for better readability. Both strings together are the reveiver queue. I get no error or warning up to GST_DBG=3. I used mp4mux because some claim, that I need a container.
Sender:
"appsrc ! videoconvert ! nvh264enc ! h264parse ! queue ! mux.video"
" shmsrc socket-path=/tmp/foo ! qtdemux ! aacparse ! queue ! mux.audio"
" flvmux name=mux ! rtmpsink location=rtmp://xx.yy.10.50:1935/result/1"
Please note I separated the strings for better readability. Again I get no error. But reading the sound buffer from shared memory (shmsrc) simply stalls. If I remove this line everything seems to work perfectly well, stable even for hours.
Any ideas someone, because all the working solutions seem to use raw audio and caps. But actually I'm not interested in audio at all, I just need it copied to the sender...
so an update from our side:
We tried many things, but the answer is that this problem is inherent to the elements shmsink and shmsrc from gstreamer
When using the shm-communication between threads you loose all meta-data, basically the audio stream coming from shmsrc is not an audio stream any more. You can test this by using launch:
gst-launch-1.0 --verbose rtmpsrc location=rtmp://xx.yy.10.40:1935/ai/1 do-timestamp=true timeout=10 ! flvdemux name=demux demux.video ! h264parse ! video/x-h264 ! fakesink demux.audio ! queue ! faad ! shmsink socket-path=/tmp/foo sync=true wait-for-connection=false shm-size=100000
will produce a lot of info, especially the audio and video formats
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw, rate=(int)48000, channels=(int)2, codec_data=(buffer)1190 ...
/GstPipeline:pipeline0/GstShmSink:shmsink0.GstPad:sink: caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)48000, channels=(int)2
/GstPipeline:pipeline0/GstH264Parse:h264parse0.GstPad:src: caps = video/x-h264, stream-format=(string)avc, codec_data=(buffer)014d4029ffe10015674d402995900780227e5c04400000fa40002ee02101000468eb8f20, pixel-aspect-ratio=(fraction)1/1, width=(int)1920, height=(int)1080, framerate=(fraction)24000/1001, interlace-mode=(string)progressive, chroma-format=(string)4:2:0, bit-depth-luma=(uint)8, bit-depth-chroma=(uint)8, parsed=(boolean)true, alignment=(string)au, profile=(string)main, level=(string)4.1
When you do the same on the side of the shmsrc (pls note we only transfer the audio via shm)
gst-launch-1.0 --verbose shmsrc socket-path=/tmp/foo ! queue ! fakesink
you will get nothing, for gstreamer "nothing" looks like this:
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
If you want to use shmsrc, you need to set the meta-data via caps manually:
gst-launch-1.0 --verbose shmsrc socket-path=/tmp/foo ! 'audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)48000, channels=(int)2' ! queue ! fakesink
Will give you:
/GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)48000, channels=(int)2
Which is a correct, but totally useless solution. So we decided to move to libav instead of gstreamer.
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 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
I am using Gstreamer to record speech and transmit in real-time( so RTP and UDP ). I have the following code :
Receiver:
gst-launch-0.10 -v udpsrc port=5000 ! "application/x-rtp,media=(string)audio, clock-rate=(int)44100, width=16, height=16, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, channel-positions=(int)1, payload=(int)96" ! rtpL16depay ! audioconvert ! alsasink sync=false
Sender:
gst-launch-0.10 alsasrc ! audioconvert ! audio/x-raw-int,channels=1,depth=16,width=16,rate=44100 ! rtpL16pay ! udpsink host=localhost port=5000
This works perfectly but it transmits Mono Speech.I also know that my Mic input is Mono. So it means that i have to use the Line-in port which shall be connected by a double-Mic ( left and right, one end a double to single jack and the other end two microphones ).
Now my problem is that i cant seem to find a way to change the input signal source of alsasrc to Line-in. Is there any way to change this?
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.