Adding WebRTC-in-Chrome test case that includes an SDP offer with B=AS:xx

Adding WebRTC-in-Chrome test case that includes an SDP offer with B=AS:xx
Adding a test case to webrtc contect_browser test: - Test case will verify that we can setup a call with an SDP offer that includes B=AS:xx

BUG=316459
Committed: https://src.chromium.org/viewvc/chrome?view=rev&revision=244374

[elham1, 2013-12-11 22:48:01]

Hi,
Can you please review this

[wjia(left Chromium), 2013-12-12 01:56:20]

I have sent the patch to try jobs. I'd see the results first.

[phoglund_chromium, 2013-12-12 10:02:32]

lgtm with nits

https://codereview.chromium.org/113383002/diff/1/content/test/data/media/peer...
File content/test/data/media/peerconnection-call.html (right):

https://codereview.chromium.org/113383002/diff/1/content/test/data/media/peer...
content/test/data/media/peerconnection-call.html:658: offerSdp =
offerSdp.replace('a=mid:audio\r\n', 'a=mid:audio\r\n'+'b=AS:16\r\n');
Nit: line is too long.

[elham1, 2013-12-12 18:15:50]

On 2013/12/12 10:02:32, phoglund wrote:
> lgtm with nits
> 
>
https://codereview.chromium.org/113383002/diff/1/content/test/data/media/peer...
> File content/test/data/media/peerconnection-call.html (right):
> 
>
https://codereview.chromium.org/113383002/diff/1/content/test/data/media/peer...
> content/test/data/media/peerconnection-call.html:658: offerSdp =
> offerSdp.replace('a=mid:audio\r\n', 'a=mid:audio\r\n'+'b=AS:16\r\n');
> Nit: line is too long.

Done

[elham1, 2013-12-12 21:47:55]

Wei,
I just disabled the test on android, can you please review my cl.

[wjia(left Chromium), 2013-12-12 22:04:34]

https://codereview.chromium.org/113383002/diff/40001/content/browser/media/we...
File content/browser/media/webrtc_browsertest.cc (right):

https://codereview.chromium.org/113383002/diff/40001/content/browser/media/we...
content/browser/media/webrtc_browsertest.cc:371:
EXPECT_TRUE(ExecuteJavascript("negotiateOfferWithBLine();"));
It seems this test enables both audio and video. On Android, we need to force
using ISAC/16000, just like EstablishAudioVideoCallAndMeasureOutputLevel below.
Is it needed here, too?

[elham1, 2013-12-12 23:19:22]

On 2013/12/12 22:04:34, wjia wrote:
>
https://codereview.chromium.org/113383002/diff/40001/content/browser/media/we...
> File content/browser/media/webrtc_browsertest.cc (right):
> 
>
https://codereview.chromium.org/113383002/diff/40001/content/browser/media/we...
> content/browser/media/webrtc_browsertest.cc:371:
> EXPECT_TRUE(ExecuteJavascript("negotiateOfferWithBLine();"));
> It seems this test enables both audio and video. On Android, we need to force
> using ISAC/16000, just like EstablishAudioVideoCallAndMeasureOutputLevel
below.
> Is it needed here, too?

Done, and I have started the Android_fyi_dbg trybot.

[elham1, 2013-12-13 18:21:39]

Had to disable the test on Android since it failed with Timeout.

Wei, can you review it?

[wjia(left Chromium), 2013-12-13 18:24:18]

https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
File content/test/data/media/peerconnection-call.html (right):

https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
content/test/data/media/peerconnection-call.html:161: transformSdp =
addBandwithControl;
Shall you call forceIsac16k_ inside addBandwithControl? I am not sure about the
calling sequence on transformSdp.

[elham1, 2013-12-13 18:42:01]

On 2013/12/13 18:24:18, wjia wrote:
>
https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
> File content/test/data/media/peerconnection-call.html (right):
> 
>
https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
> content/test/data/media/peerconnection-call.html:161: transformSdp =
> addBandwithControl;
> Shall you call forceIsac16k_ inside addBandwithControl? I am not sure about
the
> calling sequence on transformSdp.

Done added the test back on android. Lets hope it passes this time

[phoglund_chromium, 2013-12-16 08:00:42]

On 2013/12/13 18:42:01, elham1 wrote:
> On 2013/12/13 18:24:18, wjia wrote:
> >
>
https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
> > File content/test/data/media/peerconnection-call.html (right):
> > 
> >
>
https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
> > content/test/data/media/peerconnection-call.html:161: transformSdp =
> > addBandwithControl;
> > Shall you call forceIsac16k_ inside addBandwithControl? I am not sure about
> the
> > calling sequence on transformSdp.
> 
> Done added the test back on android. Lets hope it passes this time

It seems no WebRTC calls manage to get up on Android ATM - they hang or crash on
the Chromium FYI bots. I'm trying to dig into why this is the case, since it
works fine locally for me. For now I'd disable the test on Android.

[elham1, 2013-12-16 21:01:04]

On 2013/12/16 08:00:42, phoglund wrote:
> On 2013/12/13 18:42:01, elham1 wrote:
> > On 2013/12/13 18:24:18, wjia wrote:
> > >
> >
>
https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
> > > File content/test/data/media/peerconnection-call.html (right):
> > > 
> > >
> >
>
https://codereview.chromium.org/113383002/diff/100001/content/test/data/media...
> > > content/test/data/media/peerconnection-call.html:161: transformSdp =
> > > addBandwithControl;
> > > Shall you call forceIsac16k_ inside addBandwithControl? I am not sure
about
> > the
> > > calling sequence on transformSdp.
> > 
> > Done added the test back on android. Lets hope it passes this time
> 
> It seems no WebRTC calls manage to get up on Android ATM - they hang or crash
on
> the Chromium FYI bots. I'm trying to dig into why this is the case, since it
> works fine locally for me. For now I'd disable the test on Android.

Done

[elham1, 2013-12-17 23:08:20]

Wei,
Can you review again, I have now added a new transform that both controls
Bandwidth and forces Isac_16k

[elham1, 2014-01-09 19:17:06]

Can you please review again. I have rebased and made the necessary changes.

[wjia(left Chromium), 2014-01-09 22:50:15]

lgtm

[phoglund_chromium, 2014-01-10 08:26:58]

lgtm

Make sure to test this on Android as well. You can do that by building clankium
and uploading to a device. If you don't have all that set up, just try
uncommenting the #ifdef ANDROID in MakeTypicalPeerconnectionCall and try to run
it on your workstation. That way you can make sure your SDP modifications plays
nice with the Android-required SDP modifications.

[commit-bot: I haz the power, 2014-01-11 00:18:10]

CQ is trying da patch. Follow status at
https://chromium-status.appspot.com/cq/elham@chromium.org/113383002/180001

[commit-bot: I haz the power, 2014-01-11 22:48:31]

Change committed as 244374