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1342 Commits

Author SHA1 Message Date
jiayl@webrtc.org
cceb166a3f Fix a use-after-free when sending queued messages is aborted for blocked channel. 
BUG=4187
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8119 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-22 00:55:10 +00:00
tommi@webrtc.org
4fb7e25843 Update StatsReport and by extension StatsCollector to reduce data copying. 
Summary of changes:
* We're now using an enum for types instead of strings which both eliminates unecessary string creations+copies and further restricts the type to a known set at compile time.
* IDs are now a separate type instead of a string, copying of Values is not possible and values are const to allow grabbing references outside of the statscollector.
* StatsReport member variables are no longer public.
* Consolidated code in StatsCollector (e.g. merged PrepareLocalReport and PrepareRemoteReport).
* Refactored methods that forced copies of string (e.g. ExtractValueFromReport).
* More asserts for thread correctness.
* Using std::list for the StatsSet instead of a set since order is not important and updates are more efficient in list<>.

BUG=2822
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 11:36:18 +00:00
braveyao@webrtc.org
fedb9ea6bc Correct the class name in peerconnection_jni.cc. 
BUG=4194
TEST=Manual Test
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-21 07:57:06 +00:00
jlmiller@webrtc.org
5f93d0a140 Update libjingle license statements at top of talk files for consistency 
BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 21:36:13 +00:00
kjellander@webrtc.org
853049fa30 Move internal capture+render to build_with_chromium==0 condition 
This will avoid errors related to DirectX not being found
for Chromium bots (mainly GN, but it's safest to do the same
changes for GYP since they also make sense there as GYP generation
go slightly faster without having to process those targets).

Thanks to vchigrin@yandex-team.ru for originally suggesting
this being fixed in
https://webrtc-codereview.appspot.com/37639004/

TESTED=
Successfully ran:
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dbuild_with_chromium=1
and trybots.

R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8102 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-20 11:40:45 +00:00
tommi@webrtc.org
8e327c45d0 Update StatsCollector's interface in preparation of more changes. 
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8095

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8097 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 20:41:26 +00:00
tommi@webrtc.org
43e54e36bf Revert 8095 "Update StatsCollector's interface in preparation of..." 
> Update StatsCollector's interface in preparation of more changes.
> 
> This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.
> 
> The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.
> 
> The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.
> 
> BUG=2822
> R=perkj@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/36829004

TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8096 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 17:34:23 +00:00
tommi@webrtc.org
5b76fd79df Update StatsCollector's interface in preparation of more changes. 
This CL is the first of three and this one contains interface additions (not deletion for backwards compatibility) as well as a few necessary updates to internal code.

The next CL will be in Chromium to consume the new new methods and remove dependency on the old ones.

The third CL will then contain the bulk of the updates and improvements and be compatible with this interface.

BUG=2822
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 16:49:33 +00:00
phoglund@webrtc.org
f9d3555ec3 Fixing LD_LIBRARY_PATH, improving safety for libjingle java unit test. 
The was was really, really difficult to run before because you needed
a custom env with both LD_PRELOAD and library path. Now the script will
set up the correct library path and be more transparent about what it
requires.

BUG=None
TESTED=locally
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 13:57:59 +00:00
sprang@webrtc.org
ff9462eb54 Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan. 
Tests are flaky on tsan, disabling for now.

BUG=4135
R=kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-19 12:06:35 +00:00
decurtis@webrtc.org
487a444215 Add stats collection for the data channel. 
BUG=1805
R=bemasc@chromium.org, hta@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8083 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 22:55:07 +00:00
tkchin@webrtc.org
ef2a5dd398 Update AppRTCDemo UI. 
- Removed log box. Debug logs still available through lldb.
- Remote video displayed in aspect fill format.
- Provide a hangup button.
- Added Default-568.png so we display properly on iPhone5+.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 22:38:21 +00:00
guoweis@webrtc.org
61c1247224 Fix a case where empty candidate id is used 
BUG=4161
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-15 06:53:07 +00:00
pthatcher@webrtc.org
fd630a50d2 Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior. 
R=decurtis@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35759004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8067 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 23:19:06 +00:00
pbos@webrtc.org
f1c8b90520 Remove WebRtcVideoEncoderFactory2. 
This interface is no longer required and just adds complexity.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/33009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 17:29:27 +00:00
pbos@webrtc.org
f18fba2f7b Implement SimulcastEncoderAdapter support. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/37589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:26:23 +00:00
henrik.lundin@webrtc.org
8315d7de85 Remove dual stream functionality in VoiceEngine 
This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.

BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 16:07:26 +00:00
mflodman@webrtc.org
b4e5d1b34e Remove RTX SSRC when deleting the default receive stream. 
BUG=crbug 448632
TEST=New unittest hitting assert without this change.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 15:07:07 +00:00
kwiberg@webrtc.org
2ebfac5649 Remove COMPILE_ASSERT and use static_assert everywhere 
COMPILE_ASSERT is no longer needed now that we have C++11's
static_assert.

R=aluebs@webrtc.org, andrew@webrtc.org, hellner@chromium.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 10:51:54 +00:00
andresp@webrtc.org
86e1e487e7 Move system_wrappers.gyp files to the proper directory. 
Build targets should not refer to non-subpaths as was happening before when
 source/system_wrappers.gyp refers to ../interface/ files.

R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 09:30:52 +00:00
phoglund@webrtc.org
ef090927f4 No longer asserting in mocks, split first test case in two methods. 
This way assertions will be caught in the test runner instead of crashing other Android threads.

BUG=None
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-14 08:56:06 +00:00
kwiberg@webrtc.org
3df38b442f Unify the two copies of compile_assert.h 
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.

R=aluebs@webrtc.org, andrew@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-13 11:37:48 +00:00
pkasting@chromium.org
16825b1a82 Use int64_t more consistently for times, in particular for RTT values. 
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 21:51:21 +00:00
glaznev@webrtc.org
be40eb0579 Allow 720x1280 frames encoding on Android. 
Current maximum encoder width and height for Android is
hard-coded to 1280x720, so if device is rotated to portrait
orientation only part 720x1280 camera frame is extracted and
scaled to 1280x720. Increasing maximum height to 1280 allows
feeding video encoder with rotated 720x1280 frames directly
without scaling.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 17:55:47 +00:00
perkj@webrtc.org
81134d019d Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. 
In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is  specified in the call to CreatePeerConnectionFactory.

This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread.

Note that both Chrome and the Android implementation use an external signaling thread.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-12 08:30:16 +00:00
andrew@webrtc.org
8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." 
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.

> Support associated payload type when registering Rtx payload type.
> 
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
> 
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26259004
> 
> Patch from Changbin Shao <changbin.shao@intel.com>.

TBR=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 20:22:46 +00:00
glaznev@webrtc.org
80452d70cb Sync Android AppRTCDemo with internal repo. 
- Fixed some Lint warnings.
- Switch to OPUS by default.
- Add check to WebSocket connection that public methods are called
on correct thread.

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

Review URL: https://webrtc-codereview.appspot.com/36669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8032 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 19:34:06 +00:00
pthatcher@webrtc.org
9657265f39 Revert "Accept incoming pings before remote answer is set to reduce connection latency." 
This reverts r7980.

It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.

Review URL: https://webrtc-codereview.appspot.com/41429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 19:08:27 +00:00
pbos@webrtc.org
2a169640a3 Support associated payload type when registering Rtx payload type. 
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.

BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26259004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 15:16:10 +00:00
decurtis@webrtc.org
2ead571fb6 Hard define the GUID for AudioEndpoint to avoid conflicts during compile. 
BUG=3996
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 19:18:01 +00:00
pbos@webrtc.org
59062d5aef Rename SendAndReceiveH264SvcQqvga to VP8 instead. 
This looks like it's been incorrect for a while, this test configures
VP8 in QQVGA.

BUG=
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 19:21:18 +00:00
decurtis@webrtc.org
8af11042cb Avoid reading past end of string in GetLine. 
BUG=3881
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8017 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 19:15:51 +00:00
pbos@webrtc.org
bab79951ca Convert FileMediaEngineTest to use more expects. 
Allows pinpointing more precisely where a failure occurs.

BUG=4144
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 18:01:29 +00:00
kjellander@webrtc.org
07c83a1385 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) 
In https://webrtc-codereview.appspot.com/35669004/ the wrong
define was used (OS_WIN only exists in Chromium code).

BUG=4135
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 10:36:53 +00:00
tkchin@webrtc.org
4e5115ae73 RTCPeerConnectionFactory: Explicitly create new worker and signaling threads. 
There should be no change in behavior, since this is what the default
constructor does.

BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 06:35:18 +00:00
glaznev@webrtc.org
f6a9714760 Remove peer connection and signaling calls from UI thread. 
- Add separate looper threads for peer connection and websocket
signaling classes.
- To improve the connection speed start peer connection factory
initialization once EGL context is ready in parallel with the room
connection.
- Add asynchronious http request class and start using it in
webscoket signaling and room parameters extractor.
- Add helper looper based executor class.
- Port some of henrika changes from
https://webrtc-codereview.appspot.com/36629004/ to fix sensor
crashes on non L devices - will remove the change if CL will
be submitted soon.

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

Review URL: https://webrtc-codereview.appspot.com/41369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-06 22:24:09 +00:00
kjellander@webrtc.org
d95435c17a Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win 
These tests have turned out to be flaky on Windows.

BUG=4135
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-06 11:01:35 +00:00
kjellander@webrtc.org
cbe7ca8796 Roll chromium_revision 8e72e1d..271c6cc (307131:309333) 
This enables OpenSSL by default for Windows, see
8e72e1d..271c6cc/build/common.gypi
which required libjingle_tests.gyp to be updated since the
targets in third_party/nss/nss.gyp was moved into a condition in
https://codereview.chromium.org/694643002.

New Android dependencies are required due to being introduced in
build/android/pylib/remote/device/remote_device_test_run.py
of 5c49978f09

This should also fix Android test execution that started failing after
https://codereview.chromium.org/815213002 was submitted, since
it's based on e2a338fac9

Relevant other changes:
* src/buildtools: 535aff2..23a4e2f
* src/third_party/android_tools: 4f723e2..8fe116f
* src/third_party/boringssl/src: 00505ec..306e520
* src/third_party/icu: 53ecf0f..51c1a4c
* src/third_party/libvpx: 9fbec81..d3f3dce
* src/tools/swarming_client: 1d4965c..119b084
Details: 8e72e1d..271c6cc/DEPS

Clang version updated 218707:223108:
8e72e1d..271c6cc/tools/clang/scripts/update.sh
Due to this, we had to disable deadlock detection for TSan
due to a bug in Clang (see webrtc:

BUG=4106
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8003 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-06 07:24:27 +00:00
tkchin@webrtc.org
3a63a3c35d iOS AppRTC: First unit test. 
Tests basic session ICE connection by stubbing out network components, which have been refactored to faciliate testing.

BUG=3994
R=jiayl@webrtc.org, kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-06 07:21:34 +00:00
pbos@webrtc.org
c37e72e890 Make setting identical RTP extensions a no-op. 
Setting extensions are responsible for a lot of stream tear-downs
causing substantial slowdowns in SetRemoteDescription.

BUG=1788,4077
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 18:51:13 +00:00
wzh@webrtc.org
433006a6c2 Fixed style issues from lint and got rid of unused fields. 
BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 17:39:43 +00:00
glaznev@webrtc.org
8390c2762e Add two unit tests for Android AppRTCDemo. 
First unit test will create peer connection client, run
for a few second, close it and verify that there were
no any errors and local video was rendered.

Second unit test will run peer connection in a loopback mode.

To run the test from command line install AppRTCDemoTest.apk
and execute the command:
adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7991 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 19:51:12 +00:00
pbos@webrtc.org
896888b7e4 Remove min bitrate from simulcast streams. 
Bitrates are still set using SetBitrateConfig() either way, and this
code causes assertion failures in
VideoSendStream::ReconfigureVideoEncoder: Assertion
`streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps' failed.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/38529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 15:40:56 +00:00
pbos@webrtc.org
9eacb8cc59 Make P2PTestConductor use VirtualSocketServer. 
Permits running JsepPeerConnectionP2PTestClient in parallel.

TBR=juberti@webrtc.org
BUG=2598
TEST=third_party/gtest-parallel/gtest-parallel -w 128 -r 100 out/Debug/libjingle_peerconnection_unittest --gtest_filter=JsepPeerConnectionP2PTestClient.*

Review URL: https://webrtc-codereview.appspot.com/37459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 09:03:19 +00:00
pbos@webrtc.org
c62749fb47 Parallelize MediaRecorder unittests. 
Exchanging static filenames for temporary ones, permitting tests to be
run in parallel without conflicting parallel uses of the same filenames.

TBR=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 -r 100 out/Debug/libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/34589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 09:01:20 +00:00
jiayl@webrtc.org
27f5317560 Use the prod GAE server in AppRTCDemo for iOS. 
BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7985 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-31 00:26:20 +00:00
jiayl@webrtc.org
5eb71eb4f4 Fix style issues from lint. 
BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7984 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-30 22:44:11 +00:00
glaznev@webrtc.org
b2bda67497 Removing old channel code from a few more places. 
Plus adding peer connection close event.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7982 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-30 18:15:43 +00:00
jiayl@webrtc.org
c5fd66dcdf Accept incoming pings before remote answer is set to reduce connection latency. 
BUG=4068
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-29 19:23:37 +00:00
henrika@webrtc.org
b024da3122 Add support for audio device selection in AppRTCDemo. 
Summary:

- Creates a list of available (possible to select) audio devices.
- Automatically selects (routes audio) the "best/default" audio device.
- If possible, starts a proximity sensor that will switch between headset earpiece and speaker phone based on how close the a person's ear the mobile device is held.

TBR=glaznev

BUG=4103,4109

Review URL: https://webrtc-codereview.appspot.com/31239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7978 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-29 10:35:06 +00:00
pthatcher@webrtc.org
5ad4178137 Move the Jingle-specific network code into webrtc/libjingle. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-23 22:14:15 +00:00
sprang@webrtc.org
46d4d29a75 Add field trial for screenshare bitrates when using temporal layers. 
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-23 15:19:35 +00:00
braveyao@webrtc.org
086c8d5a02 Use a temporary buffer to scale a screencast in OnFrameCaptured 
BUG=3903
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/23909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7973 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-22 05:46:42 +00:00
pthatcher@webrtc.org
4c0544ab07 Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. 
Also, fix the includes and header guards of examples/call.

R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 22:29:55 +00:00
tkchin@webrtc.org
7ce4a584aa Add initWithCoder to RTCEAGLVideoView. 
Allows for proper OpenGL initialization if view is created from
storyboard.

BUG=3896
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7970 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 20:47:35 +00:00
jiayl@webrtc.org
a6f7ba6848 Add a AppRTCDemo setting to change the GAE server. 
BUG=4041
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7966 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 17:32:14 +00:00
stefan@webrtc.org
742386a136 Enable payload-based padding by default and remove the API. 
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:33:17 +00:00
pthatcher@webrtc.org
5647877b2d Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. 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/33679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 03:32:59 +00:00
pthatcher@webrtc.org
aacc23465b 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 the 3rd try)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 20:31:29 +00:00
jiayl@webrtc.org
16a05dddb8 Clean up the Channel code in AppRTCDemo and use GAE prod server for new signaling mode. 
BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7955 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 20:12:03 +00:00
pthatcher@webrtc.org
f5847d7746 Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 17:09:11 +00:00
pbos@webrtc.org
ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:50:16 +00:00
pbos@webrtc.org
a9cf079248 Rename external_hmac_ctx_t to ExternalHmacContext. 
_t types are reserved by POSIX.

R=juberti@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/33699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7944 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:12:21 +00:00
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
buildbot@webrtc.org
32ec0dd032 (Auto)update libjingle 81063831-> 81073932 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7774 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 17:57:36 +00:00
pbos@webrtc.org
273a414b0e Report encoded frame size in VideoSendStream. 
Implements reporting transmitted frame size in WebRtcVideoEngine2.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=4033

Review URL: https://webrtc-codereview.appspot.com/33399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-01 15:23:21 +00:00
tommi@webrtc.org
2c13f659c7 Add a platform specific typedef for SOCKET in the peerconnection_server example since it's not universally 'int'. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7763 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-28 10:37:31 +00:00
tkchin@webrtc.org
3e9ad26112 Refactor iOS AppRTC parsing code. 
Moved parsing code to JSON categories for the relevant objects.
No longer prefer ISAC as audio codec.

BUG=
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31989005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7755 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-27 00:52:38 +00:00
sprang@webrtc.org
a71bb6033b Revert 7750 "Don't reset sequence number for a stream on deactiv..." 
> Don't reset sequence number for a stream on deactivate/reactivate.
>
> BUG=chromium:431908
> R=pbos@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/32199004

TBR=sprang@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7752 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 19:33:15 +00:00
sprang@webrtc.org
31f7a0e710 Don't reset sequence number for a stream on deactivate/reactivate. 
BUG=chromium:431908
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7750 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 16:55:52 +00:00
perkj@webrtc.org
2faf7eea6f Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection."" 
This reverts commit 308e7ff613.

Original cl description:

This adds an Android apk for running tests on the Java layer of PeerConnection.

The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner

BUG=4031
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7748 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 07:35:37 +00:00
glaznev@webrtc.org
58edb83fd4 Add video encoder fps and bitrate statistics to 
Android AppRTCDemo UI.

BUG=4045
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7747 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-26 00:39:42 +00:00
pbos@webrtc.org
008731868a Implement settable min/start/max bitrates in Call. 
These parameters are set by the x-google-*-bitrate SDP parameters. This
is implemented on a Call level instead of per-stream like the currently
underlying VideoEngine implementation to allow this refactoring to not
reconfigure the VideoCodec at all but rather adjust bandwidth-estimator
parameters.
Also implements SetMaxSendBandwidth in WebRtcVideoEngine2 as it's a SDP
parameter and allowing it to be dynamically readjusted in Call.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/26199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7746 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-25 14:03:34 +00:00
glaznev@webrtc.org
dab5d92df6 Use mirror image for Android AppRTCDemo local preview. 
Similar to Chrome apprtc using mirror image for camera
local preview provides better experience when device
is rotated.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7741 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 17:31:01 +00:00
kjellander@webrtc.org
8562f23acb OWNERS: Remove tomasl@ and mallinath@ 
mallinath@ has left the team and tomasl@ says he doesn't
know why he's owner in webrtc/test/channel_transport

R=henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-24 10:05:05 +00:00
kjellander@webrtc.org
308e7ff613 Revert "This adds an Android apk for running tests on the Java layer of PeerConnection." 
This reverts r7732

Reason: Breaks compilation on Linux:
[813/818] LINK libjingle_media_unittest
FAILED: cd ../../talk; build/build_jar.sh /usr/lib/jvm/java-7-openjdk-amd64 ../out/Debug/libjingle_peerconnection_test.jar ../out/Debug/obj/talk/libjingle_peerconnection_test_jar.gen app/webrtc/javatests/src:../out/Debug/libjingle_peerconnection.jar:../third_party/junit/junit-4.11.jar app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java
build/build_jar.sh: Entering directory `/mnt/data/b/build/slave/linux64/build/src/talk'
app/webrtc/java/testcommon/src/org/webrtc/PeerConnectionTest.java:46:warning: [deprecation] Assert in junit.framework has been deprecated
import static junit.framework.Assert.*;
                             ^
app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java:36:error: cannot find symbol
  @Test
   ^
  symbol:   class Test
  location: class PeerConnectionTestJava
app/webrtc/javatests/src/org/webrtc/PeerConnectionTestJava.java:43:error: cannot find symbol
  @Test
   ^
  symbol:   class Test
  location: class PeerConnectionTestJava
2 errors
1 warning
ninja: build stopped: subcommand failed.

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7733 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-23 21:23:00 +00:00
perkj@webrtc.org
2751f2ab4c This adds an Android apk for running tests on the Java layer of PeerConnection. 
The only testcase is currently the same test we run on Java standalone.
To run the test adb shell am instrument -w org.webrtc.test/android.test.InstrumentationTestRunner

R=kjellander@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7732 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-23 16:00:57 +00:00
thorcarpenter@google.com
88d14f483b Remove expensive and unnecessary memory alloc for sending black frames on video 
mute.

Remove old crusty is_black_ member var in webrtcvideoengine which was not adding value.

R=henrike@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/26229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7731 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-22 01:04:26 +00:00
magjed@webrtc.org
bdcf38c894 cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class 
There is also an implementation in Chromium that can be removed if/when this lands:
content/renderer/media/webrtc/webrtc_video_capturer_adapter.cc

R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7728 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-21 10:53:00 +00:00
pkasting@chromium.org
4591fbd09f Use size_t more consistently for packet/payload lengths. 
See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 22:28:14 +00:00
glaznev@webrtc.org
edc6e57a92 Support loopback mode and command line execution 
for Android AppRTCDemo when using WebSocket signaling.

- Add loopback support for new signaling. In loopback mode
only room connection is established, WebSocket connection is
not opened and all candidate/sdp messages are automatically
routed back.
- Fix command line support both for channek and new signaling.
Exit from application when room connection is closed and add
an option to run application for certain time period and exit.
- Plus some fixes for WebSocket signaling - support
POST (not used for now) and DELETE request to WebSocket server
and making sure that all available TURN server are used by
peer connection client.

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

Review URL: https://webrtc-codereview.appspot.com/24339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7725 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-20 21:16:12 +00:00
magjed@webrtc.org
f58b455cf7 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. 
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7702

Committed: https://code.google.com/p/webrtc/source/detail?r=7707

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7721 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-19 18:09:14 +00:00
henrik.lundin@webrtc.org
6f6ef72950 Add DCHECK to ensure that NetEq's packet buffer is not empty 
This DCHECK ensures that one packet was inserted after the buffer was
flushed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7719 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-19 13:02:24 +00:00
henrika@webrtc.org
2176db343c AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land) 
This CL was incorrectly reverted in r7647 by the libjingle sync bot.

TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7717 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-18 13:22:28 +00:00
guoweis@webrtc.org
930e004a81 Add jmi field for packets discarded due to network error 
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7693

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7713 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 19:42:14 +00:00
magjed@webrtc.org
c72a22c23d Add preliminary empty file videoframefactory.cc 
The purpose of this CL is to add a new file in libjingle without breaking Chromium in the process. The plan is to do the following:
1. Land a no-op videoframefactory.cc in webrtc (this file).
2. Wait for it to roll into Chromium.
3. Modify libjingle.gyp in Chromium to include this file.
4. Make the real change in webrtc with the real implementation of this file.
5. Wait for the change to roll into Chromium.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7712 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 16:34:00 +00:00
minyue@webrtc.org
4ef22d1d29 Setting Opus FEC as default 
BUG=3986
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7710 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-17 09:26:39 +00:00
tommi@webrtc.org
4ec19e306a Revert 7707 "cricket::VideoAdapter: Drop frames before spending ..." 
This didn't compile on the FYI bots.  Example error:

FAILED: E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-with-manifests environment.x86 True chrome_child.dll "E:\b\depot_tools\python276_bin\python.exe gyp-win-tool link-wrapper environment.x86 False link.exe /nologo /IMPLIB:chrome_child.dll.lib /DLL /OUT:chrome_child.dll @chrome_child.dll.rsp" 2 mt.exe rc.exe "obj\chrome\chrome_child_dll.chrome_child.dll.intermediate.manifest" obj\chrome\chrome_child_dll.chrome_child.dll.generated.manifest
content_renderer.lib(content_renderer.webrtc_video_capturer_adapter.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.peerconnectionfactory.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.videocapturer.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

libjingle_webrtc_common.lib(libjingle_webrtc_common.dummydevicemanager.obj) : error LNK2001: unresolved external symbol "public: virtual class cricket::VideoFrame * __thiscall cricket::VideoFrameFactory::CreateAliasedFrame(struct cricket::CapturedFrame const *,int,int,int,int)const " (?CreateAliasedFrame@VideoFrameFactory@cricket@@UBEPAVVideoFrame@2@PBUCapturedFrame@2@HHHH@Z)

chrome_child.dll : fatal error LNK1120: 1 unresolved externals


> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
> 
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
> 
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
> 
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
> 
> Committed: https://code.google.com/p/webrtc/source/detail?r=7702
> 
> Review URL: https://webrtc-codereview.appspot.com/29949004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7708 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-16 22:58:11 +00:00
magjed@webrtc.org
858dbbced2 cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. 
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7702

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7707 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-16 18:21:51 +00:00
henrike@webrtc.org
6a782c2a46 Revert 7693 "Add jmi field for packets discarded due to network error" breaks chromium's webrtc_cases. 
TBR=guoweis@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/25179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7706 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 22:33:13 +00:00
magjed@webrtc.org
a73d746562 Revert 7702 "cricket::VideoAdapter: Drop frames before spending ..." 
Rease for revert: failed internal test cases

> cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.
> 
> In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.
> 
> This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.
> 
> R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/29949004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7703 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 13:25:25 +00:00
magjed@webrtc.org
bbd8cad21f cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after. 
In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7702 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 12:10:46 +00:00
pbos@webrtc.org
ece3890d3a Report total bitrate for all streams in GetStats. 
This regression wasn't caught because I accidentally disabled multiple
streams for EndToEndTest.GetStats in a refactoring.

R=stefan@webrtc.org, xians@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/27179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-14 11:52:04 +00:00
magjed@webrtc.org
35c1ace185 Revert 7698 "WebRtcVideoMediaChannel::SetSendParams: Don't cap r..." 
Reason for revert is failed testcases:
WebRtcVideoEngineExtendedTestFake.ResetSimulcastSendCodecOnNewFrameSize
WebRtcVideoEngineExtendedTestFake.MultipleSendStreamsDifferentFormats

> WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution
> 
> BUG=3936
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/30039004

TBR=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7700 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 16:21:49 +00:00
kjellander@webrtc.org
a1f5b96351 Remove unnecessary copying of libjingle resource files. 
This copying has probably not been needed since
https://code.google.com/p/webrtc/source/detail?r=7088

BUG=398
TESTED=Removed the top-level talk directory and ran
libjingle_media_unittest from the following working directories:
* checkout-root/src/out/Debug
* checkout-root/src
* checkout-root

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7699 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 15:53:08 +00:00
magjed@webrtc.org
52da44b7e6 WebRtcVideoMediaChannel::SetSendParams: Don't cap resolution 
BUG=3936
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7698 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 15:43:11 +00:00
guoweis@webrtc.org
312614a438 Add jmi field for packets discarded due to network error 
Also included the total packets attempted to send.

BUG=427555

Copied from https://webrtc-codereview.appspot.com/25959004/

R=harryjin@google.com, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7693 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-13 03:38:05 +00:00
jiayl@webrtc.org
6ca6190be2 Fix a SCTP message reordering issue in datachannel.cc. 
Previously DataChannel::SendQueuedDataMessages continues the loop of sending queued messages if the channel is blocked, which will cause message reordering if the channel becomes unblocked during the loop, i.e. messages attempted after the unblocking will be sent earlier than the older messages attempted before the unblocking.

BUG=3979
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7690 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-12 17:28:40 +00:00
henrik.lundin@webrtc.org
8038d42749 Follow-up fixes for G722 
This CL addresses post-commit comments on r7662. See
https://webrtc-codereview.appspot.com/27089004/#ps40001.

BUG=3951
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7677 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-11 08:38:24 +00:00
henrike@webrtc.org
c4922316b4 Removes talk/xmllite, talk/xmpp and talk/p2p as they are no longer used by gyp/gn builds. 
TBR=niklas.enbom@webrtc.org
BUG=3379

Review URL: https://webrtc-codereview.appspot.com/30959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7670 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 15:31:24 +00:00
pbos@webrtc.org
d819803d45 Wire up DSCP support in WebRtcVideoEngine2. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/24249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7669 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 14:41:43 +00:00
pbos@webrtc.org
957e802fe0 Refactor SetDefaultEncoderConfig to work on existing codecs. 
Addresses issue where SetDefaultEncoderConfig modifies the codec list
rather than just the targeted codec. This was previously done just to
pass more unit tests rather than be done properly. This incidentally
addresses a TODO causing this to work with external codecs as well.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/32009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7667 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-10 12:36:11 +00:00
buildbot@webrtc.org
3c1970f9f3 (Auto)update libjingle 79414100-> 79428003 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7664 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 17:58:41 +00:00
andresp@webrtc.org
188d3b2245 Enable VP9 video codec support on webrtcvideoengine behind a field trial. 
BUG=chromium:431285
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7663 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 13:21:04 +00:00
henrik.lundin@webrtc.org
f85dbce041 Reapply "Advertise G722 as 8 kHz rather than 16 kHz"" 
This reverts r7653 and relands r7645. The reason for the original revert was that G722 disappeared from the SDP offer. This is now fixed. Also, a unit test was updated compared with the original change.

BUG=3951
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7662 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 12:25:00 +00:00
perkj@webrtc.org
d105cc81dc Change dummy address to use 0.0.0.0 instead of :: 
This is to not break compatiblity with FF.

https://code.google.com/p/chromium/issues/detail?id=430333

TBR=pthatcher@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7661 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 11:22:06 +00:00
pbos@webrtc.org
a2ef4fe9c3 Prevent a lot of VideoSendStream reconfigures. 
Checking whether we're setting the same configuration or not.
Experimentally this brings down underlying reconfigures from ~20 to
about 4-5.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 10:54:43 +00:00
andresp@webrtc.org
82775b1396 Refactor webrtcvideoengines to have the default list of supported codecs being generated in runtime. 
This will allow to plugin VP9 based on a field trial.

R=pbos@webrtc.org, pbos, pthatcher

Review URL: https://webrtc-codereview.appspot.com/27949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7658 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-07 09:37:54 +00:00
henrika@webrtc.org
5e160660a6 Reland Volume buttons in AppRTCDemo should affect output audio volume (part I). 
Second attempt to land https://webrtc-codereview.appspot.com/32399004/

TBR=perkj@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7657 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 20:35:13 +00:00
henrik.lundin@webrtc.org
dced5d7835 Revert "Advertise G722 as 8 kHz rather than 16 kHz" 
This reverts r7645.

TBR=pthatcher@webrtc.org
BUG=3951

Review URL: https://webrtc-codereview.appspot.com/24199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7653 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 15:27:43 +00:00
buildbot@webrtc.org
34bda43fa6 (Auto)update libjingle 79326895-> 79329222 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7652 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:44:55 +00:00
henrika@webrtc.org
e5421e9602 Volume buttons in AppRTCDemo should affect output audio volume. 
BUG=3279
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7651 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:19:19 +00:00
perkj@webrtc.org
fd0efb694a Remove deprecated PeerConnection APIs. 
Removes PeerConnectionObserver::OnError.
Removes MediaConstraints argument to PeerConnection::AddStream.
None of these have ever been implemented and have been removed from the spec.

R=tommi@chromium.org

Review URL: https://webrtc-codereview.appspot.com/24189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7650 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 12:16:36 +00:00
andresp@webrtc.org
19b4741004 Removing unused method GetDefaultVideoEncoderConfig. 
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7649 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 11:16:32 +00:00
buildbot@webrtc.org
0ef890a4ba (Auto)update libjingle 79285346-> 79320771 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7647 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:22:08 +00:00
mcasas@webrtc.org
6340acde68 AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. 
Also removed some unused "summary" ListPreference
fields.

The looks of it can be found in [1] (lowest row).

[1] https://drive.google.com/file/d/0By6DR2QIwc_ZQm9TMW5YVEpsMWc/view?usp=sharing

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7646 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 09:05:48 +00:00
henrik.lundin@webrtc.org
1dcca4028f Advertise G722 as 8 kHz rather than 16 kHz 
G722 is a 16 kHz (wideband) speech codec, but a "bug" in the RFC
has it listed as 8 kHz. This means that the codec should be
advertised as 8 kHz in SDP messages. This change fixes that.

R=juberti@google.com
TBR=pthatcher@webrtc.org
BUG=3951
TEST=Verify that the G722 is advertised as a=rtpmap:9 G722/8000, not /16000.

Review URL: https://webrtc-codereview.appspot.com/27879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7645 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-06 08:55:01 +00:00
tkchin@webrtc.org
ee9d61ce45 This fixes a small memory leak (found using Xcode/Instruments on iOS) in 
the ObjC bindings of PeerConnection. The generated session description has
to be released by the recipient

BUG=3985
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28959004

Patch from Matthias Liebig <matthias.gcode@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7636 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 22:01:53 +00:00
stefan@webrtc.org
0bae1fab4a Wire up bandwidth stats to the new API and webrtcvideoengine2. 
Adds stats to verify bandwidth and pacer stats.

BUG=1788
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 14:05:29 +00:00
buildbot@webrtc.org
a22a628356 (Auto)update libjingle 79205306-> 79244016 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7633 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 13:25:48 +00:00
buildbot@webrtc.org
795d003770 (Auto)update libjingle 79200114-> 79205306 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7627 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-05 00:14:02 +00:00
tkchin@webrtc.org
8125744a5f Cleanup RTCVideoRenderer interface. 
RTCVideoRenderer should be a protocol not a class. This change includes
an adapter for use with the C++ apis. The video views have been refactored
to implement that protocol.

BUG=3795
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7626 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 23:06:15 +00:00
buildbot@webrtc.org
45ecf4c092 (Auto)update libjingle 79169148-> 79192489 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7624 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 21:48:54 +00:00
mcasas@webrtc.org
8944c9d08b AppRTCDemoActivity: use differnet Themes for different API levels 
The current AndroidManifest.xml hardcodes a Theme that
is only available in Android L or later (Material). To
maintain backwards compatibility, and for better App
style, create a single Theme/Style and define it for
different APIs.

I tested this in two Nexus %, one with prerelease L
and another with a KK 4.4.2 and the Theme is indeed
automagically updated :)

Note that this is compatible with
https://webrtc-codereview.appspot.com/26979004/

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7619 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 17:26:22 +00:00
pbos@webrtc.org
fad9aecbf5 Remove protected files from talk/PRESUBMIT.py. 
All files may now be committed to.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7616 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 16:06:35 +00:00
pbos@webrtc.org
88ef632286 Falling back on single-stream on multiple SSRC. 
Instead of failing, use one stream. Also clamp video min bitrate.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 15:29:29 +00:00
perkj@webrtc.org
b5d045e94d ReAdd PeerConnectionInterface::AddStream to fix Chrome build. 
AddStream(MediaStreamInterface* stream, const MediaConstraintsInterface* constraints);
This will be removed once Chrome has been updated.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7608 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 13:01:33 +00:00
tommi@webrtc.org
18de6f9622 Change the PeerConnection proxy templates to use blocking method calls instead of using Thread::Send. 
The problem with Thread::Send is that it processes incoming pending messages and for the proxies,
this can mean that multiple incoming calls can concurrently run on the same thread, resulting in unexpected behavior.

See e.g. crbug.com/429740 (and more)

R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7607 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 12:08:48 +00:00
perkj@webrtc.org
c2dd5ee2c0 Prepare for removal of PeerConnectionObserver::OnError. 
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-04 11:31:29 +00:00
buildbot@webrtc.org
a663d90ae3 (Auto)update libjingle 79104430-> 79104922 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7602 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 22:29:18 +00:00
glaznev@webrtc.org
5f38c8d1b8 Android AppRTCDemo improvements: 
- Add a room list to ConnectActivity with buttons to add/remove rooms.
- Add loopback call button.
- Add option to toggle full screen / letterbox video.
- Add camera fps settings.
- Fix device to landscape orientation for HD video until issue 3936
will be fixed.
- Fix a few crashes by avoiding calling peer connection and
GAE signaling function while connection is closing.
- Better handling GAE channel error - catch channel exceptions and
display dialog with error messages.

BUG=3939, 3935
R=kjellander@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7601 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 22:18:52 +00:00
pbos@webrtc.org
96a93259b3 Implement external decoder support in WebRtcVideoEngine2. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7594 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 14:46:44 +00:00
henrik.lundin@webrtc.org
2236267b5e Disable PeerConnectionEndToEndTest.CreateDataChannelAfterNegotiate under MSan 
This test is flaky on MSan bots.

BUG=3980
TBR=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7591 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-11-03 13:38:50 +00:00
kjellander@webrtc.org
5072e0f6cd Update Android projects to API level 21. 
The update in https://webrtc-codereview.appspot.com/23309004
was not enough, so this updates to 21 instead.

This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 20.

Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-21 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-21 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-21 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.

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

Review URL: https://webrtc-codereview.appspot.com/25029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7587 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 23:26:10 +00:00
kjellander@webrtc.org
c2c94a9a9f Change default JVM location to /usr/lib/jvm/java-7-openjdk-amd64 
Given that OpenJDK 1.7 is the recommended Java SDK for
Chromium these days, we should get rid of linking to the old
non-standardized link referring to a Sun Java 1.6 SDK.

Instead of requiring all users to set JAVA_HOME, I prefer
have the most common path as default and and close webrtc:2113
as won't fix after this is submitted.

BUG=2113
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7584 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 19:01:41 +00:00
kjellander@webrtc.org
78c222bfae Update all .isolate files for the new format. 
R=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27809004

Patch from Marc-Antoine Ruel <maruel@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 18:08:09 +00:00
kjellander@webrtc.org
8a130c1084 Update Android projects to API level 20. 
This is required in order to roll chromium_revision to
keep up with Chrome, as third_party/android_tools have now
dropped support for API level 19.

Commands used:
third_party/android_tools/sdk/tools/android update project --name OpenSlDemo --target android-20 --path webrtc/examples/android/opensl_loopback
third_party/android_tools/sdk/tools/android update project --name WebRTCDemo --target android-20 --path webrtc/examples/android/media_demo/
third_party/android_tools/sdk/tools/android update project --name AppRTCDemo --target android-20 --path talk/examples/android/
Then I restored the changes of the ANDROID_SDK_ROOT -> ANDROID_HOME since it seems the Chromium build toolchain doesn't set it properly when
build/android/envsetup.sh is sourced.

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

Review URL: https://webrtc-codereview.appspot.com/23309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7582 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 17:13:37 +00:00
pbos@webrtc.org
b7ed7799e7 Implement conference-mode temporal-layer screencast. 
Renames VideoStream::temporal_layers to temporal_layer_thresholds_bps to
convey that it contains thresholds needed to ramp up between them (1
threshold -> 2 temporal layers, etc.).

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788,1667

Review URL: https://webrtc-codereview.appspot.com/23269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7578 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 13:08:10 +00:00
pbos@webrtc.org
3bf3d238c8 Configure A/V sync in WebRtcVideoEngine2. 
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/23249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 12:59:34 +00:00
minyue@webrtc.org
2dc6f3154d Adapting bitrate according to maxplaybackrate for Opus. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7575 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 05:33:10 +00:00
tkchin@webrtc.org
14146e40aa arm64 iOS build. 
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.

BUG=3898
R=kjellander@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-31 00:14:39 +00:00
jiayl@webrtc.org
50ca986bc1 Improve the logging when a TCP connection is deleted. 
BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7572 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 23:50:54 +00:00
minyue@webrtc.org
8219529b98 Cleaning up r7562-7567. 
Wrongly used git svn dcommit for committing a CL.

Then two reverts were applied.

Still something needs to be cleaned.

BUG=

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7568 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 08:23:54 +00:00
buildbot@webrtc.org
879fac81d1 (Auto)update libjingle 78822708-> 78823675 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7567 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:50:13 +00:00
minyue@webrtc.org
5f73a37597 Revert 7563 "before rebase" due to wrong submission 
> before rebase

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7566 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:49:58 +00:00
minyue@webrtc.org
c11cc8d947 Revert 7564 "to submit" due to wrong submission 
> to submit

TBR=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7565 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:46:47 +00:00
minyue@webrtc.org
de386bf67b to submit 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7564 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:20:09 +00:00
minyue@webrtc.org
c673bb9f29 before rebase 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:19:57 +00:00
minyue@webrtc.org
0b62672576 adding default rates 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7562 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-30 07:19:49 +00:00
pbos@webrtc.org
776e6f289c Use external VideoDecoders in VideoReceiveStream. 
Removes direct VideoCodec use from the new API, exposes VideoDecoders
through webrtc/video_decoder.h similar to VideoEncoders.

Also includes some preparation for wiring up external decoders in
WebRtcVideoEngine2 by adding AllocatedDecoders that specify whether they
were allocated internally or externally.

Additionally addresses a data race in VideoReceiver that was exposed with this change.

R=mflodman@webrtc.org, stefan@webrtc.org
TBR=pthatcher@webrtc.org
BUG=2854,1667

Review URL: https://webrtc-codereview.appspot.com/27829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7560 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 15:28:39 +00:00
buildbot@webrtc.org
1abc146aa5 (Auto)update libjingle 78738075-> 78738103 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7554 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:14:14 +00:00
perkj@webrtc.org
7998089789 ApprtDemo Android: Switch between front and back camera. 
This adds a UI icon for switching between the front and back camera.
This cl adds the possibility to change between the front and back camera while in a call
or before the other end have connected.

BUG=3786
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7553 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 08:10:03 +00:00
minyue@webrtc.org
2623695dfb Renaming bandwidth to bitrate in webrtcvoiceengine. 
"bandwidth" is usually a misleading mentioning. It can mean network throughput, audio frequency contents, etc.

This is to remove the confusion inside webrtcvoiceengine

BUG=
R=juberti@webrtc.org, pbos@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7551 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-29 02:27:08 +00:00
henrike@webrtc.org
269fb4bc90 move xmpp and p2p to webrtc 
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 22:20:11 +00:00
buildbot@webrtc.org
ae694effd8 (Auto)update libjingle 78642371-> 78680406 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7545 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 17:37:17 +00:00
buildbot@webrtc.org
fbd55cb27d (Auto)update libjingle 78616359-> 78642371 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7540 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-28 05:35:35 +00:00
tommi@webrtc.org
f15dee6980 Check if a datachannel in the current local description is an sctp channel before assuming rtp. 
When generating an offer from a local description when 'sctp' is not explicitly set in the
media session options, we were generating an offer with an RTP datachannel even though the
channel in the local description was already sctp.

R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7539 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 22:15:04 +00:00
glaznev@webrtc.org
243eb8e9af Adding setting screen to AppRTCDemo. 
- Move server URL from connection screen
to the setting screen.
- Add setting for local video resolution.
- Auto save last entered room number.
- Use full screen mode in video renderer and fix
texture offsets recalculation when rendering type is
dynamically changed.

BUG=3935,3953
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7534 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 17:22:15 +00:00
buildbot@webrtc.org
068b529f46 (Auto)update libjingle 78583324-> 78583691 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7532 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 16:20:42 +00:00
pthatcher@webrtc.org
2e7ee4b28b Fix the SrtpFilter crash caused by two local offers. 
BUG=http://crbug.com/421774
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 16:10:29 +00:00
pbos@webrtc.org
efc82c2c73 Implement screencast settings for WebRtcVideoEngine2. 
Adds support for screencast_min_bitrate and sets content type
corresponding to the capture type.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7529 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 13:58:00 +00:00
braveyao@webrtc.org
1732df6129 Use flags set by the port allocator. 
Currently, port allocator flags are ignored. This is inconvenient if you
want to have your own PortAllocatorFactory subclass.

BUG=webrtc:3958
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7524 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-27 03:01:37 +00:00
buildbot@webrtc.org
3f7bcc126d (Auto)update libjingle 78430441-> 78445452 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7522 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 17:26:28 +00:00
buildbot@webrtc.org
c7ed8db7fd (Auto)update libjingle 78427027-> 78430441 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7521 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 12:59:08 +00:00
perkj@webrtc.org
470988742a Add HD support to Android if we detect a hardware video encoder that can be used. This Change the internal class MediaCodecVideoEncoder to have a one public method for checking if the platform is supported. It also adds &hd=true to the reqest url a hardware encoder is detected. 
BUG=3934
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7520 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-24 11:38:19 +00:00
pthatcher@webrtc.org
c9d6d14020 patch from issue 25469004 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7517 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 23:37:22 +00:00
buildbot@webrtc.org
8fe75ee234 (Auto)update libjingle 78381351-> 78389679 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7516 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 23:07:23 +00:00
buildbot@webrtc.org
fb5e9fc44e (Auto)update libjingle 78344087-> 78381351 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7515 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 21:36:17 +00:00
asapersson@webrtc.org
580d367b14 Add macros and APIs for webrtc histograms. 
BUG=crbug/419657

Code that links system_wrappers.gyp:system_wrappers should either:
- provide implementations for the APIs, or
- link with default implementations in system_wrappers.gyp:system_wrappers_default.

R=andresp@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7508 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 12:57:56 +00:00
buildbot@webrtc.org
9d446f2e16 (Auto)update libjingle 78296920-> 78342456 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7507 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-23 12:22:06 +00:00
buildbot@webrtc.org
a9f0898e7d (Auto)update libjingle 78273470-> 78296920 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7501 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 22:02:00 +00:00
glaznev@webrtc.org
7bb4a9881d Merging Henrik's and Peter's changes for AppRTCDemo 
from https://github.com/hkjellander/AppRTCDemo.

Description of changes:
- Add connect screen with an option to enter room number or select loopback mode.
- Add 'hangup' and 'WebRTC statistics' buttons to AppRTCDemo activity.

BUG=3938
R=kjellander@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7500 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 17:43:37 +00:00
buildbot@webrtc.org
fb5410a8b7 (Auto)update libjingle 78262388-> 78262615 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7496 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 15:45:17 +00:00
pbos@webrtc.org
eacc6e4657 Remove some disabled tests in WebRtcVideoEngine2. 
Removes some tests that shouldn't have to be implemented or have already
been through other tests.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/25929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7495 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-22 15:36:54 +00:00
buildbot@webrtc.org
a5c36b397a (Auto)update libjingle 78193292-> 78199328 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7485 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:44:16 +00:00
guoweis@webrtc.org
b6173abe59 Fix local address leakage when IceTransportsType is relay 
As part of implementing IceTransportsType constraint, we should hide the raddr which is the mapped address to prevent local address leakage.

BUG=1179
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7484 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 20:40:21 +00:00
buildbot@webrtc.org
1288cbb704 (Auto)update libjingle 78106439-> 78193292 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7482 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-21 19:29:16 +00:00
glaznev@webrtc.org
a8c0edd29f Avoid using EGLContext class for Android 4.1 and below. 
Support for this class was added in Android 4.2, so
disable surface decoding for lower Android versions.

BUG=3901
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7478 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 19:08:05 +00:00
pbos@webrtc.org
fa553ef605 Set up start bitrate in WebRtcVideoEngine2. 
R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/27789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7476 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-20 11:07:07 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." 
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 22:03:39 +00:00
buildbot@webrtc.org
7992b40994 (Auto)update libjingle 77953038-> 77970462 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7471 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 21:20:28 +00:00
glaznev@webrtc.org
58202946a7 Cleaning up Android AppRTCDemo. 
- Move signaling code from Activity to a separate class
and add interface for AppRTC signaling. For now
only pure GAE signaling implements this interface.
- Move peer connection, video source and peer connection
and SDP observer code from Activity to a separate class.
- Main Activity class will do only high level calls and
event handling for peer connection and signaling classes.
- Also add video renderer position update and use full
screen for local preview until the connection is established.

BUG=
R=braveyao@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7469 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-17 17:42:38 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. 
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-15 17:30:28 +00:00
buildbot@webrtc.org
81ddc78536 (Auto)update libjingle 77701902-> 77709729 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7450 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 22:39:24 +00:00
buildbot@webrtc.org
1ecbe45c7e (Auto)update libjingle 77689511-> 77696841 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7449 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 20:29:28 +00:00
pbos@webrtc.org
43336b6b9f Remove unused (no-op) VideoOptions. 
Removing VideoOptions: adapt_input_to_encoder, adapt_view_switch,
video_one_layer_screencast and video_high_bitrate.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7448 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 19:12:06 +00:00
henrike@webrtc.org
a4351a045d libjingle: use _stricmp instead of deprecated stricmp. 
BUG=N/A
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7447 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 17:07:41 +00:00
pbos@webrtc.org
7fe1e03dd6 Wire up external encoders. 
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30649005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 04:25:33 +00:00
buildbot@webrtc.org
f68cc0b0c3 (Auto)update libjingle 77554188-> 77629208 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7439 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-14 01:17:42 +00:00
henrike@webrtc.org
1e6a5dd14e Removes xmllite from talk/xmllite since webrtc/xmllite is used instead. 
BUG=3379
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/23039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7436 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 18:27:11 +00:00
buildbot@webrtc.org
3c16d8bd1c (Auto)update libjingle 77414393-> 77554188 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-13 06:35:10 +00:00
xians@webrtc.org
3cefbc99f4 Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. 
This also marks all virtual overrides of other classes in the same files. 

This will make a subsequent change I intend to do safer, where I'll change the 
argument types of the base Transport functions, by breaking the compile if I 
miss any overrides. 

This also highlighted a number of unused functions. I've removed some of these. 

TBR=mflodman@webrtc.org, pkasting@chromium.org
BUG=none 
TEST=none

Review URL: https://webrtc-codereview.appspot.com/28709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-10 09:42:53 +00:00
glaznev@webrtc.org
dae40dcde9 Change setting VP8 codec specific info values by HW VP8 encoder 
to follow SW implementation.

This fixes video freezing observed when connecting Android AppRtcDemo
on devices with hW encoder support with Chrome apprtc.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7414 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 17:53:09 +00:00
glaznev@webrtc.org
95bacfed08 Remove bad waiting code from video decoder release function. 
Instead keep surface texture object alive while video codec
is re-initialized with a different resolution.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7401 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-09 00:00:11 +00:00
buildbot@webrtc.org
97abeee282 (Auto)update libjingle 77263371-> 77296420 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7400 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 22:24:30 +00:00
pbos@webrtc.org
575d126a3d Protect send_/recv_streams_ in WebRtcVideoEngine2. 
Important as OnLoadUpdate() won't be called on the worker thread and the
list of streams can't be concurrently modified while delivering this
callback to all send streams.

R=stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/22959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7395 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-08 14:48:08 +00:00
jiayl@webrtc.org
742922b313 Make the media content send only if offerToReceive is false while local streams exist. 
We previously do not add the media content if offerToReceive is false.

BUG=3833
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7390 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 21:32:43 +00:00
pbos@webrtc.org
d6bda09503 Initialize sctp_paddrparams in OpenSctpSocket(). 
Addresses 'use-of-uninitialized-value' detected with MemorySanitizer.
params.spp_address.sa_family was used without being initialized before
when calling usrsctp_setsockopt with SCTP_PEER_ADDR_PARAMS.

R=jiayl@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/23909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7389 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 19:23:43 +00:00
glaznev@webrtc.org
46ffc70878 Temporary fix to allow Invoke() calls for VP8 HW encoder and decoder. 
BUG=
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7387 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 17:11:36 +00:00
pbos@webrtc.org
963b979510 Remove potential deadlock in WebRtcVideoEngine2. 
Fixes lock-order inversions between capturer's SignalVideoFrame and
WebRtcVideoSendStream. Additionally also removes all deadlock
suppressions for WebRtcVideoEngine2.

R=stefan@webrtc.org
TBR=kjellander@webrtc.org
BUG=1788,2999

Review URL: https://webrtc-codereview.appspot.com/26729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7386 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 14:27:27 +00:00
kjellander@webrtc.org
6ed1cf49f0 Isolate: Remove use of --ignore_broken_items 
BUG=chromium:395700
R=jam@chromium.org

Review URL: https://webrtc-codereview.appspot.com/30659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7383 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-07 09:17:35 +00:00
henrike@webrtc.org
528fc650d8 Fixing build issue with L-sdk 
Based on https://codereview.appspot.com/153000043/

BUG=https://code.google.com/p/chromium/issues/detail?id=420293
R=niklas.enbom@webrtc.org, serya@chromium.org, yfriedman@chromium.org

Review URL: https://webrtc-codereview.appspot.com/29659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7374 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-06 17:56:43 +00:00
pbos@webrtc.org
42684be21b Wire up CPU adaptation in WebRtcVideoEngine2. 
Includes clean-up work to be able to use the webrtc::Call::Config that's
set up. This introduced a CallFactory to spawn a FakeCall with the
config used and allowed removal of FakeWebRtcVideoChannel2.

BUG=1788
R=mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7370 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-03 11:25:45 +00:00
glaznev@webrtc.org
25cc745d6b Switch to SW video decoder on Android after getting 2 or more 
critical errors from HW decoder.

BUG=410730
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7368 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-02 16:58:05 +00:00
henrike@webrtc.org
4530b2ca48 Revert 7355 "Fix parallelization in libjingle_p2p_unittest." 
Breaks waterfall.

TBR=pbos@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/22909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7357 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 15:43:55 +00:00
pbos@webrtc.org
fd29205e6e Fix parallelization in libjingle_p2p_unittest. 
Adding VirtualSocketServers to SessionTest and RelayServerTest to avoid
contention on real ports.

R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 out/Debug/libjingle_p2p_unittest

Review URL: https://webrtc-codereview.appspot.com/26679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7355 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 12:31:31 +00:00
henrik.lundin@webrtc.org
4cebd84c79 Reland "Remove DTMF status methods from Voice Engine" r7276 
This reverts r7277.

TBR=henrika@webrtc.org,pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7353 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-10-01 08:23:21 +00:00
xians@webrtc.org
7aad5e5cce Revert 7338 "Fixed the android build by making the interface pur..." 
> Fixed the android build by making the interface pure virtual.
> 
> TBR=asapersson@webrtc.org, bjornv@webrtc.org,
> 
> Review URL: https://webrtc-codereview.appspot.com/24789004

TBR=xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7341 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:26:15 +00:00
xians@webrtc.org
90d1979d77 Fixed the android build by making the interface pure virtual. 
TBR=asapersson@webrtc.org, bjornv@webrtc.org,

Review URL: https://webrtc-codereview.appspot.com/24789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7338 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 15:15:22 +00:00
pbos@webrtc.org
1795c358fc Add default implementation of Add/RemoveObserver. 
Needed to remove Add/RemoveObserver from RTCVideoEncoderFactory in
Chromium before removing these completely. This is done to keep the
chromium.webrtc.fyi bots happy and to make rolling webrtc revisions
easier.

R=stefan@webrtc.org
BUG=3876

Review URL: https://webrtc-codereview.appspot.com/23839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7332 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 09:45:25 +00:00
kjellander@webrtc.org
8cad9432d5 Revert 7327 "Update isolate.gypi files + link to isolate_driver.py" 
Breaks debug compilation (didn't run all trybots when testing this).

> Update isolate.gypi files + link to isolate_driver.py
> 
> This updates the isolate.gypi copies we're forced to
> maintain in our code repo to Chromium revision c264a05.
> 
> Since isolated testing is now using a new launch script
> in tools: isolate_driver.py, that is added to our links
> script.
> 
> BUG=395700
> TESTED=Ran one of our tests with:
> ninja -C out/Release tools_unittests_run
> tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate
> 
> R=henrika@webrtc.org, jam@chromium.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26649004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7328 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 08:44:00 +00:00
kjellander@webrtc.org
02cd3067d2 Update isolate.gypi files + link to isolate_driver.py 
This updates the isolate.gypi copies we're forced to
maintain in our code repo to Chromium revision c264a05.

Since isolated testing is now using a new launch script
in tools: isolate_driver.py, that is added to our links
script.

BUG=395700
TESTED=Ran one of our tests with:
ninja -C out/Release tools_unittests_run
tools/isolate_driver.py run --isolated out/Release/tools_unittests.isolated --isolate webrtc/tools/tools_unittests.isolate

R=henrika@webrtc.org, jam@chromium.org

Review URL: https://webrtc-codereview.appspot.com/26649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7327 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-30 08:34:57 +00:00
glaznev@webrtc.org
359d720004 Allow Android apps to set video renderer scaling type. 
Also add type check for EGL context object received from apps and
switch to byte buffer video decoding if EGL context is incorrect

BUG=3851
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7326 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 23:07:08 +00:00
jiayl@webrtc.org
7dfb7fa189 Reland disallowing blocking calls on the worker thread. 
This fixed the issue that invoking the call when the thread is not started.

BUG=3559
R=juberti@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/24769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7325 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 22:45:55 +00:00
asapersson@webrtc.org
626624061e Disable flaky tests: 
JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined
JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined

BUG=3871
R=henrike@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7323 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-29 14:30:07 +00:00
pbos@webrtc.org
34f2a9ea72 Initialize SSL in unittest_main.cc. 
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/30549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-28 11:36:45 +00:00
jiayl@webrtc.org
bebc75e8bd Fix the duplicated candidate problem when using multiple STUN servers. 
BUG=3723
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7312 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 23:01:11 +00:00
thorcarpenter@google.com
a21d071607 Reverting part of 
https://webrtc-codereview.appspot.com/15089004/diff/140001/talk/session/media/channelmanager.cc?context=10&column_width=80
because of a major regression hanging the executable on start.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7309 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 17:19:14 +00:00
pbos@webrtc.org
05305116d6 Explicitly initialize SSL for tests. 
Adding missing SSL initialization/cleanups in
TransportDescriptionFactoryTest and MediaSessionTest.

These being missing prevent these tests from being run individually
without other tests preceding them that initialize SSL.

BUG=3860
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7300 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-25 15:50:26 +00:00
jiayl@webrtc.org
3987b6de50 Fix a problem in Thread::Send. 
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.

BUG=3559
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 17:14:05 +00:00
pbos@webrtc.org
d60d79a145 Thread annotation of rtc::CriticalSection. 
Effectively re-lands r5516 which was reverted because talk/-only
checkouts existed. This now resides in webrtc/base/, so no talk/-only
checkouts should be possible.

This change also enables -Wthread-safety for talk/ and fixes a bug in
talk/media/webrtc/webrtcvideoengine2.cc where a guarded variable was
read without taking the corresponding lock.

R=andresp@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7284 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 07:10:57 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/. 
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 06:05:00 +00:00
glaznev@webrtc.org
8166faeff3 Change Android video renderer to maintain video aspect 
ratio when displaying camera or decoded video frames.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7282 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 23:58:52 +00:00
glaznev@webrtc.org
90668b1633 Switch HW video decoder to output byte buffers if video 
renderer EGL context is not provided by app.

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7281 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 21:42:15 +00:00
buildbot@webrtc.org
1b7dcc1647 (Auto)update libjingle 76169599-> 76176062 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7280 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 17:41:48 +00:00
guoweis@webrtc.org
2c1bcea1bc Enable ipv6 by default for webrtc under a Finch experiment. 
Reapply 23529005 after fixing the build break issue (Chromium:582133002)

Committed: https://code.google.com/p/webrtc/source/detail?r=7253

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7278 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 16:23:02 +00:00
henrik.lundin@webrtc.org
3987f10c11 Revert "Remove DTMF status methods from Voice Engine" r7276 
This change caused some trouble.

TBR=henrika@webrtc.org,pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7277 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 13:15:14 +00:00
henrik.lundin@webrtc.org
bf7b9e0081 Remove DTMF status methods from Voice Engine 
These methods are not used.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7276 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 12:54:04 +00:00
pbos@webrtc.org
0a2087a711 Skeleton for registering external encoders/decoders. 
R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/31429005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7270 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-23 09:40:22 +00:00
pbos@webrtc.org
83f95ba9a6 Remove engine-level SetOptions. 
Already removed in WebRtcVideoEngine.

R=andresp@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 16:07:18 +00:00
henrik.lundin@webrtc.org
64a2f10f4b Remove Get/SetNetEQPlayoutMode APIs 
These are not used anymore.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-22 14:30:10 +00:00
guoweis@webrtc.org
97ed39344a Reapply 23529005 after fixing the build break issue (Chromium:582133002) 
Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 21:06:12 +00:00
buildbot@webrtc.org
ed5ca1f122 (Auto)update libjingle 75925673-> 75926712 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7252 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:30:44 +00:00
buildbot@webrtc.org
c98f217c65 (Auto)update libjingle 75924589-> 75925673 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:18:10 +00:00
buildbot@webrtc.org
0c9fe72b21 (Auto)update libjingle 75922684-> 75924589 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7250 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 20:05:02 +00:00
glaznev@webrtc.org
ebf2757339 Fix HW video decoder crash on some Android KK devices. 
Remove direct access to decoder Java output buffer memory
when HW decoder is configured to decode to surface.

-

R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30459005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7249 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 19:36:13 +00:00
thorcarpenter@google.com
c1eebfa107 Fix the libjingle_media_unittest failure in Windows build by modifying libjingle_tests.gyp and sctpdataengine_unittests.cc instead of ssladapter.cc. 
R=harryjin@google.com, pthatcher@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/22699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7245 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 17:54:00 +00:00
glaznev@webrtc.org
e65812427d Fixing compilation failure in peerconnection_jni.cc with WEBRTC_CHROMIUM_BUILD. 
Symbol LogcatTraceContext not defined.
Submitting on behalf of serya@.
Dup of https://webrtc-codereview.appspot.com/29529004/

TEST=Build target libjingle_peerconnection_javalib with applied CL https://codereview.chromium.org/551793003/
BUG=https://crbug.com/383418
R=serya@chromium.org

Review URL: https://webrtc-codereview.appspot.com/28529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7244 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 16:53:46 +00:00
pbos@webrtc.org
bbe0a8517d Config struct for VideoEncoder. 
Used for config parameters in common between multiple codecs as well as
the encoder-specific pointer. In particular this contains content mode
(realtime video vs. screenshare).

BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7239 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 12:30:25 +00:00
buildbot@webrtc.org
6e5c78422d (Auto)update libjingle 75875619-> 75878731 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7235 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 06:46:37 +00:00
buildbot@webrtc.org
b5a5c44ef7 (Auto)update libjingle 75865376-> 75875619 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7234 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 05:36:18 +00:00
buildbot@webrtc.org
d7acf11e8d (Auto)update libjingle 75854833-> 75865376 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7233 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-19 02:01:09 +00:00
buildbot@webrtc.org
ccb3e3f3db (Auto)update libjingle 75854418-> 75854833 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7232 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 23:31:03 +00:00
buildbot@webrtc.org
dcc1f0426b (Auto)update libjingle 75852725-> 75853560 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7231 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 23:14:12 +00:00
glaznev@webrtc.org
0b435ba597 A few fixes to avoid crash in HW codec on device orientation change. 
- Fix video encoder Reset() function to avoid setting codec
resolution to zero.
- Follow SW codec implementation and do not crash when frame
with the resolution different from the encoder resolution arrives.
Instead wait for at least 3 frames with new resolution and
re-initialize the codec. HW codec reset may take much longer
than SW codec, so these 3 frames threshold avoids resetting
codec when outstanding camera frame captured from previous device
orientation arrives.
- Plus some minor changes to make encoder reset/release
implementation closer to decoder implementation.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7230 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 23:01:03 +00:00
glaznev@webrtc.org
83af77bf3c Revert maximum video codec resolution on Android back to 720p again. 
Some low end Android devices still have problems with 1080p support.

BUG=3757
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7228 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 21:11:29 +00:00
buildbot@webrtc.org
933d88af58 (Auto)update libjingle 75818332-> 75837294 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7227 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 20:23:05 +00:00
jiayl@webrtc.org
42731bdded Avoid writing a double/float to a string to avoid a crash. 
BUG=crbug/367223
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7225 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 16:51:51 +00:00
pbos@webrtc.org
6cd6ba8ae0 Expose VP8/H264 defaults through video_encoder.h. 
Reduces code duplication quite a bit, these identical defaults were set
in quite a few different places.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=3070

Review URL: https://webrtc-codereview.appspot.com/19299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7220 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 12:42:28 +00:00
andresp@webrtc.org
ab071daab8 Split video_render_module implementation into default and internal implementation. 
Targets must now link with implementation of their choice instead of at "gyp"-time.

Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common

Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests

GN changes:
- Not many since there is almost no test definitions.

Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.

Re-enable android tests by reverting 7026 (some tests left disabled).

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-18 08:58:15 +00:00
guoweis@webrtc.org
369a637ac8 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Committed: https://code.google.com/p/webrtc/source/detail?r=7201

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 22:37:29 +00:00
glaznev@webrtc.org
3b67f8e0ca Enable HW video decoding on Qualcomm devices. 
Parallel decoding and encoding problem is fixed now
(b/16353967), so it is possible to start using Qualcomm
VP8 HW decoder. Bitrate overshoots should be fixed as well.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 21:25:51 +00:00
henrike@webrtc.org
4a5061fbff talk/p2p/base: removed unused variable "port_" 
BUG=N/A
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 12:33:07 +00:00
andresp@webrtc.org
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture) 
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:50:19 +00:00
andresp@webrtc.org
85ef770d92 Split video engine android initialization into each internal module initialization. 
This is to later on allow targets to pick at link time if to include the external or internal implementation. In order to do that the video_engine cannot compile different based on which option is picked later on.

BUG=3768,3770
R=glaznev@webrtc.org, stefan@webrtc.org
TBR=henrike@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 11:44:51 +00:00
pbos@webrtc.org
ab990ae43a Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."" 
Re-lands r7114 after landing r7204 to adress the compile error causing
the rollback in r7151.

BUG=3070
TBR=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 09:02:25 +00:00
buildbot@webrtc.org
6a9b155798 (Auto)update libjingle 75683337-> 75695882 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 08:08:38 +00:00
glaznev@webrtc.org
a59c501c99 Java VideoRenderer class may be backed by two different native 
classes depending on type of rendering.
Fix crash in AppRtcDemo by calling correct destructor on exit.

BUG=
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-17 03:26:59 +00:00
guoweis@webrtc.org
40c2aa36f2 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7200

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 20:29:41 +00:00
guoweis@webrtc.org
f8bff762d1 Implemented Network::GetBestIP() selection logic as following. 
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.

ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address

BUG=3808

At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.

R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 20:17:22 +00:00
pbos@webrtc.org
cddd17c0f8 Recreate VideoStreams when setting resolution. 
Instead of just changing resolution on the last stream streams are
reallocated to make sure that all streams are updated to match the
new input resolution.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/29469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 16:33:13 +00:00
pbos@webrtc.org
88e85ad64d Add pbos@webrtc.org (myself) to talk/media/webrtc/. 
Allows easier reviews of webrtcvideoengine2.cc and landing the new video
API on shorter review cycles.

R=pthatcher@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/30409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 16:14:51 +00:00
buildbot@webrtc.org
80132e4d70 (Auto)update libjingle 75610402-> 75610402 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 15:24:15 +00:00
kjellander@webrtc.org
595b23c66f Revert 7184 "Enable ipv6 by default for webrtc under a Finch exp..." 
Breaks Chrome build and prevents rolling WebRTC into Chrome DEPS.

> Enable ipv6 by default for webrtc under a Finch experiment.
> 
> BUG=413437 (chromium)
> https://code.google.com/p/chromium/issues/detail?id=413437
> 
> Review URL: https://webrtc-codereview.appspot.com/23529005

TBR=guoweis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 08:58:22 +00:00
andrew@webrtc.org
6ae5a6d7fe Add a target for the approved subset of rtc_base. 
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)

This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.

The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.

BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-16 01:03:29 +00:00
glaznev@webrtc.org
996784548d HW video decoding optimization to better support HD resolution: 
- Change hw video decoder wrapper to allow to feed multiple input
and query for an output every 10 ms.
- Add an option to decode video frame into an Android surface object. Create
shared with video renderer EGL context and external texture on
video decoder thread.
- Support external texture rendering in Android renderer.
- Support TextureVideoFrame in Java and use it to pass texture from video decoder
to renderer.
- Fix HW encoder and decoder detection code to avoid query codec capabilities
from sw codecs.

BUG=
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 17:52:42 +00:00
guoweis@webrtc.org
cd309e3168 Enable ipv6 by default for webrtc under a Finch experiment. 
BUG=413437 (chromium)
https://code.google.com/p/chromium/issues/detail?id=413437

Review URL: https://webrtc-codereview.appspot.com/23529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 16:31:13 +00:00
pbos@webrtc.org
000d86792d Make BW checks > 0 in peerconnection_unittest.cc. 
These checks (> 40k) fail on LSan FYI bots and the purpose of them seem
to be that we're getting non-zero BW reported.

R=stefan@webrtc.org
TBR=jiayl@webrtc.org, solenberg@webrtc.org
BUG=3817,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/29479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 14:38:07 +00:00
henrike@webrtc.org
7f826350e3 Stop building talk/xmllite since it is no longer used. 
BUG=3379
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-15 08:13:36 +00:00
buildbot@webrtc.org
a42a3ade54 (Auto)update libjingle 75390072-> 75428737 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-13 01:09:18 +00:00
fbarchard@google.com
7e31197cb2 Revert 7170 "Revert 7121 "ValidateFrame, When dumping the first ..." 
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*

> Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that."
> 
> Breaks other repos.
> 
> TBR=fbarchard@google.com
> BUG=N/A
> 
> Review URL: https://webrtc-codereview.appspot.com/23639004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-13 00:52:42 +00:00
glaznev@webrtc.org
192a54ff2f Temporary revert maximum video codec resolution back to 1080p. 
BUG=3757, 3738
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:40:35 +00:00
henrike@webrtc.org
3decd9b776 Revert 7121 "ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that." 
Breaks other repos.

TBR=fbarchard@google.com
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/23639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:31:29 +00:00
buildbot@webrtc.org
ea77334c30 (Auto)update libjingle 75302540-> 75327856 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 21:52:48 +00:00
henrike@webrtc.org
1d8f780779 Stop building talk/sound since it is no longer used. 
BUG=N/A
R=pbos@webrtc.org
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 17:16:56 +00:00
glaznev@webrtc.org
1d53f64b0f Disabling initializeAndroidGlobals when built with WEBRTC_CHROMIUM_BUILD. 
webrtc::VideoEngine::SetAndroidObjects and webrtc::VoiceEngine::SetAndroidObjects
are not compatible with WEBRTC_CHROMIUM_BUILD. Since neither VoiceEngine nor VideoEngine
are needed at the time it's better to disable it completely.

BUG=https://crbug.com/412276
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 16:58:25 +00:00
henrikg@webrtc.org
307d3dbdee Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h." 
Speculative revert, seems to be reason for flaky Win FYI bot compile break.

> Expose VideoEncoders with webrtc/video_encoder.h.
> 
> Exposes VideoEncoders as part of the public API and provides a factory
> method for creating them.
> 
> BUG=3070
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/21929004

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 09:48:30 +00:00
sprang@webrtc.org
c665dcb205 Revert 7145 "Stop building talk/sound since it is no longer used." 
> Stop building talk/sound since it is no longer used.
> 
> BUG=N/A
> R=pbos@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/22319004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 08:29:53 +00:00
henrik.lundin@webrtc.org
1972ff8a6e Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE. 
This will make a subsequent change I intend to do safer, where I'll change the
return type of one of the base Module functions, by breaking the compile if I
miss any overrides.

This also highlighted a number of unused functions (in many cases apparently
virtual "overrides" of no-longer-existent base functions).  I've removed some of
these.

This also highlighted several cases where "virtual" was used unnecessarily to
mark a function that was only defined in one class.  Removed "virtual" in those
cases.

BUG=none
TEST=none
R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 06:20:28 +00:00
henrike@webrtc.org
4c876453c8 Stop building talk/sound since it is no longer used. 
BUG=N/A
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 22:18:04 +00:00
glaznev@webrtc.org
3472dcd7b0 Fix frame rate selection for Android camera. 
- Android camera supports multiple fps values for a single video
resolution - change video source default video format selection
to pick up best available fps.
- Change fps range calculation to better match target fps value.

BUG=2622
R=tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 19:24:57 +00:00
henrike@webrtc.org
b2efb6771c Put base tests in webrtc_tests.gyp 
BUG=N/A
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 17:28:19 +00:00
jiayl@webrtc.org
b6d69282f5 Enable shared socket for TurnPort. 
In AllocationSequence::OnReadPacket, we now hand the packet to both the TurnPort and StunPort if the remote address matches the server address.

TESTED=AppRtc loopback call generates both turn and stun candidates.

BUG=1746
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 16:31:34 +00:00
buildbot@webrtc.org
5d639b3ef3 (Auto)update libjingle 75141932-> 75179475 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7129 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 07:57:12 +00:00
jiayl@webrtc.org
7d4891d3f1 Fixes two issues in how we handle OfferToReceiveX for CreateOffer: 
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.

2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.

BUG=2108
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7068

Review URL: https://webrtc-codereview.appspot.com/16309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 21:43:15 +00:00
fbarchard@google.com
54cf1505e2 ValidateFrame, When dumping the first 4 samples of a frame, first copy it to a temporary buffer that is zero padded, them use that. 
BUG=3789
TESTED=drmemory out\Debug\libjingle_media_unittest.exe --gtest_catch_exceptions=0 --gtest_filter=*Validate*
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7121 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 18:34:53 +00:00
jiayl@webrtc.org
22406fcc9b TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH. 
BUG=3570
R=juberti@webrtc.org, mallinath@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7070

Review URL: https://webrtc-codereview.appspot.com/20999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7120 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 15:44:05 +00:00
mallinath@webrtc.org
3d81b1b22a Relanding https://code.google.com/p/webrtc/source/detail?r=7093, after it got 
reverted due to some internal compile failures.

In this CL changes are done in portallocator_unittest.cc, in particular to EXPECT_EQ checking in new tests.

Original patch committed in https://code.google.com/p/webrtc/source/detail?r=7093

TBR=juberti@webrtc.org
BUG=1179

Review URL: https://webrtc-codereview.appspot.com/22329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7118 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 14:38:10 +00:00
andresp@webrtc.org
4d19e05ab2 Peerconnection_jni to use webrtc/base/checks.h instead of implementing its own. 
This needs to happen sooner or later as if webrtc/base/checks.h happens to be included transitively here it would collide.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7115 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 11:45:44 +00:00
pbos@webrtc.org
b420191743 Expose VideoEncoders with webrtc/video_encoder.h. 
Exposes VideoEncoders as part of the public API and provides a factory
method for creating them.

BUG=3070
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7114 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-09 10:40:56 +00:00
henrike@webrtc.org
8b0b21161a Revert 7093: "Implementing ICE Transports type handling in libjingle transport." 
TBR=mallinath@webrtc.org
BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/28419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 22:46:28 +00:00
pbos@webrtc.org
7118e61669 Finish work queue in SctpDataMediaChannelTest. 
Always finishing the work queue prevents memory leak detected in
LeakSanitizer (packet is deleted on the receiver side).

R=jiayl@webrtc.org
BUG=3608,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/28399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7110 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 21:44:07 +00:00
jiayl@webrtc.org
0e52772aa9 Fix a bot-breaking memory leak from early returning in ParseMediaDescription. 
BUG=3791
R=henrike@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 21:43:43 +00:00
jiayl@webrtc.org
c172320bd2 Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android. 
This reverts commit r7068.

TBR=kjellander@webrtc.org
BUG=2108

Review URL: https://webrtc-codereview.appspot.com/23539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 20:44:36 +00:00
buildbot@webrtc.org
fd42f9dd6f (Auto)update libjingle 74955991-> 75042522 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-08 19:45:36 +00:00
mallinath@webrtc.org
7256d31d28 Implementing ICE Transports type handling in libjingle transport. 
BUG=1179
R=juberti@webrtc.org, bemasc@webrtc.org, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-07 04:08:44 +00:00
thorcarpenter@google.com
cc060563f3 Remove unnecessary include from testutils.cc. 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 21:19:00 +00:00
buildbot@webrtc.org
992febb997 (Auto)update libjingle 74873066-> 74873164 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:39:08 +00:00
thorcarpenter@google.com
a3344cfda4 Fix webrtcvideoframe tests. 
R=fbarchard@google.com, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7088 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:34:13 +00:00
jiayl@webrtc.org
ddb85ab85b Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07 
- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added

BUG=3592
R=jiayl@webrtc.org, juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 16:31:56 +00:00
buildbot@webrtc.org
af5fa95258 (Auto)update libjingle 74857067-> 74860820 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-05 13:03:50 +00:00