- From: Dee Cee <deeprcee@gmail.com>
- Date: Fri, 22 Aug 2014 12:26:59 -0700
- To: Varun Singh <vsingh.ietf@gmail.com>
- Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>
- Message-ID: <CAPtn87NkZdD4kcWHFZ6+=9PnVSg2+HWv2j1X1iL0+gvm0j7dSQ@mail.gmail.com>
Varun - Thanks for your reply. I do see many statistics in statscollector. Some of them like bytes/packets sent are obvious but the other ones like bucket delay, playout delay etc aren't so obvious at first look. It requires some code digging which is going a bit slow. Is there any documentation around what each statistic really mean ? Thanks Dee On Wed, Aug 20, 2014 at 1:26 AM, Varun Singh <vsingh.ietf@gmail.com> wrote: > Hi, > > Since you are discussing about chrome, there are some additional stats > appended with a `goog` identifier in getStats() which may help (e.g., > frame playout time in ms compare that to the frame rate). > > There are additional stats discussed in the wiki > (https://www.w3.org/2011/04/webrtc/wiki/Stats) but AFAIK they are not > yet implemented by any browser. From those metrics, there are a few > that may help with discerning A/V sync -- for example framesDropped, > which are typically dropped from the jitter buffer because they will > not make the intended playout time. > > > -Varun > > On Tue, Aug 19, 2014 at 11:27 PM, Dee Cee <deeprcee@gmail.com> wrote: > > Hi, > > > > > > > > We have a small application that uses webrtc to setup a call for video > chat. > > > > Is there a good way to measure A/V sync issues, frame drops and audio > > glitches from webrtc code ? > > > > I looked into chrome://webrtc-internal but did not find a suitable graph. > > > > > > Thanks > > > > Dee > > > > -- > http://www.netlab.tkk.fi/~varun/ >
Received on Friday, 22 August 2014 19:27:26 UTC