DM365 rtsp/test-launch audio lost

Gina,

I'm not sure about the exact issue. But by looking at the log, I'm trying to understand what is the relation with MMC driver with this. Can you check whether MMC card is inserted and is the device writing the stream to MMC/SD card as well? If so, can you remove the card and try the same and see whether this error occurs?

Renjith,

Thank you.

I remove the card, no warning then, but audio buffers still lost after a while

how do I know is the device writing the stream to the card?

I add "audio/x-raw-int,endianness=1234,signed=true,width=16,depth=16,rate=22050,channels=2"

to set the rate of alsasrc, it decrease CPU loading from 40~ to 30~  (when I gst-launch alsasrc with fakesink )

./test-launch "( alsasrc ! audio/x-raw-int,endianness=1234,signed=true,width=16,depth=16,rate=22050,channels=2 ! TIAudenc1 codecName=aaclcenc  engineName=codecServer ! rtpmp4apay name=pay0 pt=96 v4l2src input-src=composite ! video/x-raw-yuv,format=(fourcc)NV12,width=352,height=240 ! TIVidenc1 codecName=h264enc engineName=codecServer ! rtph264pay name=pay1 pt=97 )"

maybe bacause remove the card and set the rate the audio stream work fine much longer but still break after a while

-Gina

Gina,

Can you upload the audio file with the issue? I don't think CPU load could be the issue here as you are seeing only 40% CPU load. 

Renjith,

When I said 40% down to 30% I meant alsasrc only 

I run the v4l2src it would be like 45%

so when I use gst-rtsp/example/test-launch stream both audio and video I assume the CPU wound overload

since I don't know how to let the test-launch to print out damiperf arm load

here is another pipeline I tried

------------------------------------------------------------ 

CLIENT_IP=$1
gst-launch -ev gstrtpbin name=rtpbin\
alsasrc buffer-time=100000\
! audio/x-raw-int,endianness=1234,signed=true,width=16,depth=16,rate=22050,channels=2\
! dmaiperf 'print-arm-load=true'\
! TIAudenc1 codecName=aaclcenc engineName=codecServer bitrate=128000\
! rtpmp4apay pt=97 max-ptime=84000 ! rtpbin.send_rtp_sink_0\
rtpbin.send_rtp_src_0 ! udpsink port=1224 host=$CLIENT_IP\
rtpbin.send_rtcp_src_0 ! udpsink port=1225 host=$CLIENT_IP sync=false\
async=false udpsrc port=1229 ! rtpbin.recv_rtcp_sink_0\
v4l2src input-src=composite\
! 'video/x-raw-yuv,format=(fourcc)NV12,width=352,height=240'\
! TIVidenc1 codecName=mpeg4enc engineName=codecServer\
! rtpmp4vpay ! rtpbin.send_rtp_sink_1\
rtpbin.send_rtp_src_1 ! udpsink port=1234 host=$CLIENT_IP\
rtpbin.send_rtcp_src_1 ! udpsink port=1235 host=$CLIENT_IP sync=false\
async=false udpsrc port=1239 ! rtpbin.recv_rtcp_sink_1

--------------------------------------------------

-------------------Log--------------------------

INFO:
Timestamp: 0:45:38.589751733; bps: 88179; fps: 209; CPU: 89;
INFO:
Timestamp: 0:45:39.595553525; bps: 87761; fps: 208; CPU: 86;
INFO:
Timestamp: 0:45:40.597800025; bps: 88023; fps: 209; CPU: 93;
INFO:
Timestamp: 0:45:41.602243109; bps: 88685; fps: 211; CPU: 91;
INFO:
Timestamp: 0:45:42.611001151; bps: 88333; fps: 210; CPU: 92;
WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough
Additional debug info:
gstbaseaudiosrc.c(822): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 2730 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.
INFO:
Timestamp: 0:45:43.616404817; bps: 86089; fps: 204; CPU: 95;
INFO:
Timestamp: 0:45:44.619093025; bps: 87604; fps: 208; CPU: 85;
INFO:
Timestamp: 0:45:45.626957901; bps: 88838; fps: 211; CPU: 88;
INFO:
Timestamp: 0:45:46.632062692; bps: 88179; fps: 209; CPU: 94;
INFO:
Timestamp: 0:45:47.636798442; bps: 87848; fps: 209; CPU: 89;
INFO:
Timestamp: 0:45:48.643064609; bps: 88508; fps: 210; CPU: 98;
INFO:
Timestamp: 0:45:49.648466567; bps: 88179; fps: 209; CPU: 94;

-------------------

I use  ./test-launch --gst-debug=2 ... ,so it printed out

------------------Log-----------------

vpfe-capture vpfe-capture: adjusted width = 352, height = 240, bpp = 1, bytesperline = 352, sizeimage = 126720

-Gina

Gina,

Are you seeing 100% CPU load when you run your Audio + Video applications? Most likely this issue is with the Audio driver by looking at the logs.

Renjith,

If is not the CPU loading cause the audio buffers lost on client's VLC, any ideas?

I stream to two computer, when one VLC show lost buffer, the another one didn't but after a while it did. 

here is a screen shot of VLC audio streaming

<Lost buffers> keep increase as the <Played buffers> and <Decoded blocks>  but no sound

maybe the buffers somehow imcomplete, so vlc can't played?

sorry, i don't really know how to solve this problem

thank you for answering

-Gina

Gina,

Its tough to analyze it over forum. If you can spare one board, then I can look into the issue.