GStreamer problem with separating audio and video - audio

Question

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...

Answer 1

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.