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the more outdated copy) and moves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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
55360ae402
Revert "Add adapter_type into Candidate object."
...
This reverts commit aaf02cc2d4f696345ce0e6d5715f2cfa22aea689.
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
2faf7eea6f
Revert "Revert "This adds an Android apk for running tests on the Java layer of PeerConnection.""
...
This reverts commit 308e7ff61327d64ba5c7761ce6b58cd1fbc4847e.
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
