[gst-devel] rtsp-server serves slowly

[Wes Miller]

Hi All,

Using the example code in rtsp-server  .../examples/test-video.c as a
template, I have this partially working pipeline:

      gst_rtsp_media_factory_set_launch( factory,
      "( "
        "v4l2src "
        " ! videorate "
        " ! ffmpegcolorspace  "
        " ! video/x-raw-yuv,width=1280,framerate=30/1 "
        " ! ffenc_mpeg4 "
        " ! rtpmp4vpay name=pay0 pt=96  "
        "alsasrc device=\"hw:0,0\" "
        " ! audio/x-raw-int,rate=16000,channels=1,depth=16 "
        " ! audioresample "
        " ! audioconvert "
        " ! alawenc ! rtppcmapay name=pay1 pt=97 "
      ")"
   );


The video has a 4 second delay to get to 127.0.0.1 (using totem) and the
audio turns everything into occasional clicks.  Everything video starts up
slowly and seems to be sampling way too slow.  Wiggled fingers are more blur
than fingers.  

If I just connect the camera to autovideosink I get better video quality,
less blur though it is still there.

   gst-launch-0.10 v4l2src ! videorate  ! ffmpegcolorspace !
video/x-raw-yuv,width=1280,framerate=30/1  ! autovideosink

And if I hook up the microphone directly, I get sound but it is delayed
about a half second:

     gst-launch-0.10 -v alsasrc device="hw:0,0" !
audio/x-raw-int,rate=16000,width=16,depth=16 ! audioresample ! audioconvert
! alsasink

     /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000
     /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000
     /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps =
audio/x-raw-int, endianness=(int)1234, signed=
     (boolean)true, width=(int)16, depth=(int)16, rate=(int)16000,
channels=(int)1


Sorry, I don't know the specs on the webcam, the maker doesn't seem to want
to share in his included docs.

The rtsp-server window says:

** (lt-test-video:13312): WARNING **: 0x97aa000: got warning Can't record
audio fast enough (gstbaseaudiosrc.c(822): gst_base_audio_src_create ():
/GstPipeline:media-pipeline/GstBin:bin0/GstAlsaSrc:alsasrc0:
Dropped 4000 samples. This is most likely because downstream can't keep up
and is consuming samples too slowly.)

Any suggestions for speeding up audio and video and getting better quality
to flow through the connection?

Thanks,

Wes

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