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1020 Commits

Author SHA1 Message Date
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