gstreamer-1.0 v4l2src ( webcam) + alsasrc has audio_sample drop problem

[Ashish Kumar Mishra]

Hi Nicolas  , 

1) The pipeline is better and seems to be in sync when done the changes
mentioned by you in another thread 
    *
http://gstreamer-devel.966125.n4.nabble.com/Issue-in-v4l2src-capture-Fails-with-minimum-latency-bigger-than-maximum-latency-td4675242.html*
Could you please provide some more info / any links where i could understand
how "tune=zerolatency" effects to pipeline ?



2) Increasing buffer-time had not improved the problem. 
    I even changed latency-time parameter of alsasrc such that (buffer-time
= 2* latency-time)
    But i kept on getting below mentioned log
       New clock: GstAudioSrcClock
        /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time =
200000
        /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time =
10000

 *Doesn't these parameter indicates the value which i am setting , why does
it shows same value every iteration ?*

 
Below is the current modified pipeline ( Working with webcam & audio sync
from 2.3sec )
Could you let me know if there's any other issue in the pipeline :-

*gst-launch-1.0 -v v4l2src do-timestamp=true !
video/x-raw,width=960,height=720,framerate=30/1 ! timeoverlay
shaded-background=1 ! x264enc tune=zerolatency ! mux. alsasrc
do-timestamp=true , provide-clock=false !
audio/x-raw,width=16,depth=16,rate=44100,channels=2 ! queue ! mux.
matroskamux name=mux  ! filesink location=test.mkv
*


Thanks ,
Ashish Kumar Mishra



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