Refactor RtcDataChannelHandler.

Refactor RtcDataChannelHandler.
This change splits the WebRtc observer functionality of RtcDataChannelHandler
from the WebKit related functionality. The former belongs with libjingle's
signaling thread and the latter with the main thread. As is, these threads
are one and the same, but this will change later this week.

BUG=369796
Committed: https://crrev.com/832aa011fa8f98972f04468216dd2fbbd3e7ca42
Cr-Commit-Position: refs/heads/master@{#301831}

[tommi (sloooow) - chröme, 2014-10-28 23:35:41]

tommi@chromium.org changed reviewers:
+ perkj@chromium.org

[perkj_chrome, 2014-10-29 08:02:17]

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
File content/renderer/media/rtc_data_channel_handler.cc (right):

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
content/renderer/media/rtc_data_channel_handler.cc:80:
scoped_ptr<webrtc::DataBuffer> new_buffer(new webrtc::DataBuffer(buffer));
There is a memcpy here as well:
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjin...

OnMessage should be changed to OnMessage(rtc::scopedptr<webrtc::DataBuffer>
buffer) I think.

Can you file a defect for this and add number here in the TODO so it gets fixed?

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
content/renderer/media/rtc_data_channel_handler.cc:220: DVLOG(1) <<
"OnStateChange " << state;
Add thread check

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
File content/renderer/media/rtc_data_channel_handler.h (right):

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
content/renderer/media/rtc_data_channel_handler.h:32: // that may be occur
before we've fully connected with webkit.
s/maybe

[tommi (sloooow) - chröme, 2014-10-29 12:20:16]

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
File content/renderer/media/rtc_data_channel_handler.cc (right):

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
content/renderer/media/rtc_data_channel_handler.cc:80:
scoped_ptr<webrtc::DataBuffer> new_buffer(new webrtc::DataBuffer(buffer));
On 2014/10/29 08:02:17, perkj wrote:
> There is a memcpy here as well:
>
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjin...
> 
> OnMessage should be changed to OnMessage(rtc::scopedptr<webrtc::DataBuffer>
> buffer) I think.
> 
> Can you file a defect for this and add number here in the TODO so it gets
fixed?

I don't know if webrtc's scoped_ptr supports move semantics.

In any case, I filed a bug for myself in the webrtc tracker:
https://code.google.com/p/webrtc/issues/detail?id=3967

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
content/renderer/media/rtc_data_channel_handler.cc:220: DVLOG(1) <<
"OnStateChange " << state;
On 2014/10/29 08:02:17, perkj wrote:
> Add thread check

Done.

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
File content/renderer/media/rtc_data_channel_handler.h (right):

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
content/renderer/media/rtc_data_channel_handler.h:32: // that may be occur
before we've fully connected with webkit.
On 2014/10/29 08:02:17, perkj wrote:
> s/maybe

Done.

[tommi (sloooow) - chröme, 2014-10-29 12:20:43]

Address comments

[perkj_chrome, 2014-10-29 12:54:11]

The CQ bit was checked by perkj@chromium.org

[perkj_chrome, 2014-10-29 12:54:11]

lgtm

lgtm

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
File content/renderer/media/rtc_data_channel_handler.cc (right):

https://codereview.chromium.org/643693007/diff/1/content/renderer/media/rtc_d...
content/renderer/media/rtc_data_channel_handler.cc:80:
scoped_ptr<webrtc::DataBuffer> new_buffer(new webrtc::DataBuffer(buffer));
On 2014/10/29 12:20:16, tommi wrote:
> On 2014/10/29 08:02:17, perkj wrote:
> > There is a memcpy here as well:
> >
>
https://code.google.com/p/chromium/codesearch#chromium/src/third_party/libjin...
> > 
> > OnMessage should be changed to OnMessage(rtc::scopedptr<webrtc::DataBuffer>
> > buffer) I think.
> > 
> > Can you file a defect for this and add number here in the TODO so it gets
> fixed?
> 
> I don't know if webrtc's scoped_ptr supports move semantics.
> 
> In any case, I filed a bug for myself in the webrtc tracker:
> https://code.google.com/p/webrtc/issues/detail?id=3967
It does.

[commit-bot: I haz the power, 2014-10-29 12:55:05]

CQ is trying da patch. Follow status at
 https://chromium-cq-status.appspot.com/patch-status/643693007/20001

[commit-bot: I haz the power, 2014-10-29 14:29:30]

Committed patchset #2 (id:20001)

[commit-bot: I haz the power, 2014-10-29 14:30:15]

Patchset 2 (id:??) landed as
https://crrev.com/832aa011fa8f98972f04468216dd2fbbd3e7ca42
Cr-Commit-Position: refs/heads/master@{#301831}

[csharp, 2014-10-29 15:58:54]

On 2014/10/29 14:30:15, I haz the power (commit-bot) wrote:
> Patchset 2 (id:??) landed as
> https://crrev.com/832aa011fa8f98972f04468216dd2fbbd3e7ca42
> Cr-Commit-Position: refs/heads/master@{#301831}

This looks like it broke the content browser test,
WebRtcBrowserTest.CanSetupLegacyCall and WebRtcBrowserTest.CallWithDataAndMedia,
on the Win7 test Dbg bot.

Output:
[ RUN      ] WebRtcBrowserTest.CallWithDataAndMedia
[4752:1116:1029/085556:1566343:INFO:CONSOLE(64)] "Looking at video in element
remote-view-1", source: http://127.0.0.1:52407/media/webrtc_test_utilities.js
(64)
[4752:1116:1029/085556:1566359:INFO:CONSOLE(64)] "Looking at video in element
remote-view-2", source: http://127.0.0.1:52407/media/webrtc_test_utilities.js
(64)
[4752:1116:1029/085556:1566406:INFO:CONSOLE(782)] "Negotiating call...", source:
http://127.0.0.1:52407/media/peerconnection-call.html (782)
[4752:160:1029/085556:1566406:ERROR:audio_manager_win.cc(454)]
GetPreferredAudioParameters failed: 88890004
[4752:1116:1029/085600:1570649:INFO:CONSOLE(805)] "Receiving offer...", source:
http://127.0.0.1:52407/media/peerconnection-call.html (805)
[4752:1116:1029/085600:1570930:INFO:CONSOLE(907)] "Receiving remote stream...",
source: http://127.0.0.1:52407/media/peerconnection-call.html (907)
[4752:1116:1029/085600:1571039:INFO:CONSOLE(883)] "Receiving answer...", source:
http://127.0.0.1:52407/media/peerconnection-call.html (883)
[4752:160:1029/085601:1571335:ERROR:audio_manager_win.cc(454)]
GetPreferredAudioParameters failed: 88890004
[4752:160:1029/085601:1571335:ERROR:audio_output_resampler.cc(208)] Unable to
open audio device in low latency mode.  Falling back to high latency audio
output.
[4752:160:1029/085601:1571335:ERROR:audio_output_resampler.cc(218)] Unable to
open audio device in high latency mode.  Falling back to fake audio output.
[4752:4464:1029/085601:1571413:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:4464:1029/085601:1571632:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:1116:1029/085601:1571678:INFO:CONSOLE(907)] "Receiving remote stream...",
source: http://127.0.0.1:52407/media/peerconnection-call.html (907)
[4752:4464:1029/085601:1571803:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:4464:1029/085602:1573114:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:4464:1029/085602:1573207:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:1116:1029/085604:1574970:INFO:CONSOLE(80)] "Done looking at video in
element remote-view-2", source:
http://127.0.0.1:52407/media/webrtc_test_utilities.js (80)
[4752:4464:1029/085604:1575032:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:4464:1029/085604:1575126:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:1116:1029/085606:1576374:INFO:CONSOLE(80)] "Done looking at video in
element remote-view-1", source:
http://127.0.0.1:52407/media/webrtc_test_utilities.js (80)
[4752:1116:1029/085606:1576405:INFO:CONSOLE(585)] "gSecondConnection received
data", source: http://127.0.0.1:52407/media/peerconnection-call.html (585)
[4752:1116:1029/085606:1576592:INFO:CONSOLE(585)] "gSecondConnection received
data", source: http://127.0.0.1:52407/media/peerconnection-call.html (585)
[4752:1116:1029/085606:1576608:INFO:CONSOLE(782)] "Negotiating call...", source:
http://127.0.0.1:52407/media/peerconnection-call.html (782)
[4752:1116:1029/085606:1576811:INFO:CONSOLE(805)] "Receiving offer...", source:
http://127.0.0.1:52407/media/peerconnection-call.html (805)
[4752:4464:1029/085606:1576951:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:1116:1029/085606:1577045:INFO:CONSOLE(605)] "Uncaught NetworkError: Failed
to execute 'send' on 'RTCDataChannel': Could not send data", source:
http://127.0.0.1:52407/media/peerconnection-call.html (605)
From javascript: Error: Error: Uncaught NetworkError: Failed to execute 'send'
on 'RTCDataChannel': Could not send data
Script: http://127.0.0.1:52407/media/peerconnection-call.html
Line: 605
Column: 23
StackTrace: NetworkError: Failed to execute 'send' on 'RTCDataChannel': Could
not send data
at failTest (http://127.0.0.1:52407/media/webrtc_test_utilities.js:39:15)
at window.onerror
(http://127.0.0.1:52407/media/peerconnection-call.html:13:31)c:\b\build\slave\...):
error: Failed
[4752:4464:1029/085606:1577060:INFO:socket_host_udp.cc(304)] sendto() has failed
twice returning a  transient error. Dropping the packet.
[4752:1116:1029/085606:1577060:WARNING:channel.cc(148)] WriteMessage() while
shutting down
[4752:1116:1029/085606:1577185:INFO:user_input_monitor_win.cc(164)]
RegisterRawInputDevices() failed for RIDEV_REMOVE: The parameter is incorrect.
(0x57)
[  FAILED  ] WebRtcBrowserTest.CallWithDataAndMedia, where TypeParam =  and
GetParam() =  (12909 ms)

[csharp, 2014-10-29 15:59:21]

A revert of this CL (patchset #2 id:20001) has been created in
https://codereview.chromium.org/686183002/ by csharp@chromium.org.

The reason for reverting is: Broke some webrtc tests on win7.

[tommi (sloooow) - chröme, 2014-10-29 21:12:03]

Remove retry loop for unreliable data channels