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Author SHA1 Message Date
pthatcher@webrtc.org
4cb3856a4d Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." 
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.

BUG=
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 02:28:25 +00:00
pthatcher@webrtc.org
536f999e58 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
This is an un-revert of r7992 and r7993.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 01:22:02 +00:00
pthatcher@webrtc.org
bc03192560 Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 22:15:11 +00:00
tommi@webrtc.org
209df9bf77 Change MockStatsObserver to grab values inside of OnComplete. 
This is done since StatsReportCopyable is going away and the list of
supported properties of the mock class is known.
StatsReports holds a list of pointers to objects that cannot be cached,
so this is a simple way to grab the values when they're available.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7932 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 14:09:05 +00:00
pbos@webrtc.org
e728ee03ba Remove or rename typedefs with _t prefixes. 
_t prefixes are reserved for additional typenames in POSIX.

R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/36559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 13:43:55 +00:00
guoweis@webrtc.org
950c518251 Add adapter_type into Candidate object. 
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Committed: https://code.google.com/p/webrtc/source/detail?r=7906

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7925 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 23:01:31 +00:00
pthatcher@webrtc.org
f050791ba0 Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." 
This reverts r7992.

It broke the Chromium build because the Chroumium build relies on the logic in webtc/libjingle/session.cc, but Chromium doesn't compile that file.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7923 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 22:28:03 +00:00
pthatcher@webrtc.org
4afb59903c Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7922 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 21:37:37 +00:00
pthatcher@webrtc.org
e2b7585bc2 Move ViewRequest and MediaStreams to streamparams.h, and remove dependency on mediasessionclient.h and mediamessages.h. This is part of the effort to remove Jingle-specific code from WebRTC and into its own repository. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7921 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 21:09:08 +00:00
guoweis@webrtc.org
55360ae402 Revert "Add adapter_type into Candidate object." 
This reverts commit aaf02cc2d4.

BUG=
TBR=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7908 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-16 05:28:10 +00:00
guoweis@webrtc.org
aaf02cc2d4 Add adapter_type into Candidate object. 
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7885

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7906 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 23:03:10 +00:00
pkasting@chromium.org
0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess. 
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 22:09:40 +00:00
tommi@webrtc.org
e2e199b894 Clean up StatsObserver's OnComplete methods (address TODOs). 
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7898 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 13:22:54 +00:00
buildbot@webrtc.org
032b802a8c (Auto)update libjingle 82121498-> 82126219 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7896 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 09:48:07 +00:00
tommi@webrtc.org
dd0601fbcf Remove unneeded ctor and add a more practical one 
The default constructor isn't necessary, so I'm removing it.
I'm adding another one so that we can (later) make |type| const.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7895 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 09:47:49 +00:00
tommi@webrtc.org
69bc5a300f Add thread asserts to StatsCollector. 
Also adding a "ForTest" postfix to a test-only method.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7894 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 09:44:48 +00:00
pbos@webrtc.org
fb108b5a28 Revert r7885. 
Breaks compile step of other code where network name of
cricket::Candidate is used.

TBR=guoweis@webrtc.org,juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/31229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7892 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 08:04:50 +00:00
pbos@webrtc.org
18a3896bd2 Revert r7886:7887. 
Broke build steps in other code that uses securetunnelsessionclient.cc
and others.

TBR=tommi@webrtc.org,pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/36439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7890 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-15 07:03:04 +00:00
magjed@webrtc.org
e575e9c40f Move WebRtcVideoRenderFrame from webrtcvideoengine2.cc to webrtcvideoframe.h 
The purpose of this CL is to be able to reuse the class WebRtcVideoRenderFrame in webrtcvideoengine.cc.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7888 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-14 11:09:23 +00:00
pthatcher@webrtc.org
dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle. 
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 21:04:42 +00:00
guoweis@webrtc.org
8c9d79a29d Add adapter_type into Candidate object. 
Expose adapter_type from Candidate such that we could add jmidata on top of this.

Created a new type of report just for Ice candidate. The candidate's id is used as part of report identifier. This code change only reports the best connection's local candidate's adapter type. There should be cleaning later to move other candidate's attributes to the new report.

This is migrated from issue 32599004

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7885 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 19:21:14 +00:00
tommi@webrtc.org
c57310b982 Switch kStatsValueName* constants to be enums instead of char*. 
This is to guard against potentially assigning a value name to an incorrect value, non-static string or otherwise assume they can be treated as strings.

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7884 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 17:41:28 +00:00
pthatcher@webrtc.org
40b276ea7b Cleanup little things found when refactoring. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/33519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7880 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-12 02:44:30 +00:00
pbos@webrtc.org
2b19f06312 Wire up RTT statistics to webrtc::Call. 
R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667,1788

Review URL: https://webrtc-codereview.appspot.com/32249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7876 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 13:26:09 +00:00
pbos@webrtc.org
13518951e3 Remove old_factory from WebRtcVideoEngine. 
Minor pending cleanup.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/28239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7875 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 13:14:30 +00:00
perkj@webrtc.org
128fabaf7b Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin..."" 
Original cl description:

Change Android PeerConnectionUnittest to build using Chrome macros.
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7874 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-11 12:25:57 +00:00
buildbot@webrtc.org
a85307737c (Auto)update libjingle 81702493-> 81755413 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7860 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-10 09:01:18 +00:00
tommi@webrtc.org
aa2c342c10 Add back a constructor to fix FYI build. 
TBR=perkj

Review URL: https://webrtc-codereview.appspot.com/24349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7854 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 20:23:06 +00:00
tkchin@webrtc.org
87776a8935 iAppRTCDemo: WebSocket based signaling. 
Updates the iOS code to use the new signaling model. Removes old Channel API
code. Note that this no longer logs messages to UI. UI update forthcoming.

BUG=
R=glaznev@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7852 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 19:32:35 +00:00
pthatcher@webrtc.org
0babb4a4e6 Fix a comment. 
R=juberti@webrtc.org, pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7851 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 19:01:45 +00:00
tommi@webrtc.org
c9d155faeb Move implementation of types in statstypes. to its cc file. 
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7850 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 18:18:06 +00:00
henrika@webrtc.org
a954c07ee1 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer 
BUG=4034
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7849 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 16:22:09 +00:00
tommi@webrtc.org
5c3ee4bce6 Add empty implementation file that will hold statstypes.h implementation. 
The implementation for the types currently in statstypes.h is split between statstypes.h and statscollector.cc.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7844 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 10:47:01 +00:00
glaznev@webrtc.org
eef85387ec Fix AppRTCDemo closing error for KK and JB Android devices. 
- Do not allow connection output when sending http delete
request to ws server - this causes IOException for KK and JB devices.
- Avoid creating dialog box with error message when activity
has been already closed / paused -
this causes resource leak error message for KK devices.
- Plus some code clean up to support async http messages in
websocket channel wrapper and use Handler for running
peerconnection client funcitons on UI thread.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31159004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7836 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-09 01:29:17 +00:00
andrew@webrtc.org
3b3c406908 Revert 7826 "Change Android PeerConnectionUnittest to build usin..." 
Broke gclient runhooks on internal bots. e.g.
http://chromegw/i/internal.client.webrtc/builders/Linux64%20Debug/builds/3575

> Change Android PeerConnectionUnittest to build using Chrome macros.
> The purpose is to be able to run the tests using Chromes buildbots. To run:
> CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest
> 
> This also add a new build target to build java PeerConnection using Chromes build macros.
> 
> BUG=4031
> R=kjellander@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/28189004

TBR=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7829 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 17:21:50 +00:00
perkj@webrtc.org
ed7824b1c0 Change Android PeerConnectionUnittest to build using Chrome macros. 
The purpose is to be able to run the tests using Chromes buildbots. To run:
CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py instrumentation --test-apk=libjingle_peerconnection_android_unittest

This also add a new build target to build java PeerConnection using Chromes build macros.

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7826 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-08 15:41:01 +00:00
glaznev@webrtc.org
e2a9261f3e Improve AppRTCDemo connection speed by sending all 
http POST requests asynchronously.

R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7820 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 20:11:06 +00:00
kjellander@webrtc.org
bd8cc0b914 Add codereview.settings to the /talk subdirectory 
With this, it will be possible to create CLs from
Git repos created using
https://chromium.googlesource.com/external/webrtc/trunk/talk
(which is what you get when working with the repo currently
put in Chrome's src/third_party/libjingle/source/talk).

TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7819 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 13:47:37 +00:00
kjellander@webrtc.org
599e299b9d cricket::VideoFrame int64 to int64_t. 
Needed for successful compile of ios arm64.

BUG=3898
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30359004

Patch from Zeke Chin <tkchin@webrtc.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-05 09:42:57 +00:00
bemasc@webrtc.org
9b5467e88d Fix assertion failure when closing data channel, and add a unit test. 
BUG=4066
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7816 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 23:16:52 +00:00
glaznev@webrtc.org
4b407aa985 Update AppRTCDemo README with information on 3-dot-apprtc server 
and new command line arguments.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7815 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 22:42:59 +00:00
guoweis@webrtc.org
7169afd9d5 With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior. 
BUG=411086
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30919005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7814 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 17:59:29 +00:00
glaznev@webrtc.org
369746bcb8 Support new WebSocket signaling format. 
- Support new GAE message format and new signaling
sequence, which allows connection to 3-dot-apprtc server.
- Add UI setting to switch between GAE / WebSockets signaling.
- Some clean ups to better support command line application
execution.

BUG=3937,3995,4041
R=jiayl@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7813 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-04 17:28:52 +00:00
pbos@webrtc.org
0fb6ad2004 Check if cpu_monitor_ exists before Stop(). 
R=asapersson@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7797 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 13:44:29 +00:00
asapersson@webrtc.org
d8aed6b321 Verify that cpu_monitor exists before calling Stop(). 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7795 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 12:37:47 +00:00
pthatcher@webrtc.org
eb0954248d Don't reset sequence number for a stream on deactivate/reactivate. 
BUG=chromium:431908
R=pbos@webrtc.org, sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7788 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-03 00:34:10 +00:00
glaznev@webrtc.org
d01955179a Change minimum video encoder initialization resolution to 
176x144 to ensure HW encoder can be initialized.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7787 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 23:41:18 +00:00
perkj@webrtc.org
beee9cec22 Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. 
The reason is that the desktop apprtcdemo only handle one MediaStream and this doesn't play audio if it receive two streams.

TEST=Test that a call with audio and video can be setup between an Android device and a desktop client using apprtc.appspot.com.
BUG=4051,3786
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7781 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-02 14:38:18 +00:00
pthatcher@webrtc.org
146e0fd30f Fix the build by putting in a typecast to avoid a comparison between 
signed and unsigned ints introduced in cl/81073932.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7776 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 20:07:52 +00:00
glaznev@webrtc.org
dea5173edf Add start bitrate and vp8 hw acceleration option to 
Android AppRTCDemo.

- Add an option to set VP8 encoder start bitrate
usig x-google-start-bitrate line in remote SDP.
- Allow to enabled/disable VP8 hw decoder and
encoder acceleration using appRTC settings.

BUG=4046
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7775 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 20:02:13 +00:00