Using the following works fine
gst-launch-0.10 filesrc location=temp.wav ! wavparse ! audioconvert ! alsasink device=hw:0
as does
gst-launch-1.0 filesrc location=temp.wav ! wavparse ! audioconvert ! alsasink device=hw:0
When I use hw:2, I get a device is busy error if a player is using it and an Internal data flow error otherwise.
If I enable debugging
GST_DEBUG=2
I get the following
alsa pcm_hw.c:1693:_snd_pcm_hw_open: alsalib error: Invalid value for card
basetransform gstbasetransform.c:1414:gst_base_transform_setcaps:<audioconvert0> transform could not transform audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int)2, channel-mask=(bitmask)0x0000000000000003, rate=(int)44100 in anything we support
basetransform gstbasetransform.c:1414:gst_base_transform_setcaps:<audioconvert0> transform could not transform audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int)2, channel-mask=(bitmask)0x0000000000000003, rate=(int)44100 in anything we support
basetransform gstbasetransform.c:1414:gst_base_transform_setcaps:<audioconvert0> transform could not transform audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int)2, channel-mask=(bitmask)0x0000000000000003, rate=(int)44100 in anything we support
basetransform gstbasetransform.c:1414:gst_base_transform_setcaps:<audioconvert0> transform could not transform audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int)2, channel-mask=(bitmask)0x0000000000000003, rate=(int)44100 in anything we support
wavparse gstwavparse.c:2249:gst_wavparse_loop:<wavparse0> error: Internal data flow error.
wavparse gstwavparse.c:2249:gst_wavparse_loop:<wavparse0> error: streaming task paused, reason not-negotiated (-4)
basetransform gstbasetransform.c:1414:gst_base_transform_setcaps:<audioconvert0> transform could not transform audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int)2, channel-mask=(bitmask)0x0000000000000003, rate=(int)44100 in anything we support
ERROR: from element /GstPipeline:pipeline0/GstWavParse:wavparse0: Internal data flow error.
I do get sound from both of the following
gst123 -a alsa=hw:2 temp.wav
aplay -D hw:2 temp.wav
Am I missing some additional arguments to gst-launch?
Any help would be much appreciated!
While writing up the question I found the answer, but I figured I'd go ahead and post the question in case anyone else ever encounters the same. There was indeed a missing argument... audioresample.
gst-launch-1.0 filesrc location=temp.wav ! wavparse ! audioconvert ! audioresample ! alsasink device=hw:2
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
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 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 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.