418 Commits

Author SHA1 Message Date
perkj@webrtc.org
1a38a51119 Add default implementation to VideoSourceInterface of Stop and Restart.
This is to make sure Chrome does not break when rolling. This should be reverted once
Chrome has been updated.

Please see:
http://build.chromium.org/p/chromium.webrtc.fyi/builders/Mac/builds/16556/steps/compile/logs/stdio

BUG=4303
R=magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35229004

Cr-Commit-Position: refs/heads/master@{#8391}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8391 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 14:51:43 +00:00
perkj@webrtc.org
8f605e8911 Add VideoSource::Stop and Restart methods.
The purpose is to make sure that start and stop is called on the correct thread on Android. It also cleans up the Java VideoSource implementation.

BUG=4303
R=glaznev@webrtc.org, magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39989004

Cr-Commit-Position: refs/heads/master@{#8389}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8389 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 13:54:42 +00:00
minyue@webrtc.org
f9b5c1b3d0 Removing CELT.
CELT is not supported in WebRTC/Libjingle. There are a few left-over in our code base. They are cleaned up in this CL.

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

Review URL: https://webrtc-codereview.appspot.com/36099004

Cr-Commit-Position: refs/heads/master@{#8385}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8385 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 12:37:14 +00:00
pthatcher@webrtc.org
3341b401cc Fix bug parsing media descriptions: the final field isn't a codec type for any of DTLS/SCTP, SCTP, or SCTP/DTLS.
BUG=none
TEST=none
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34029004

Cr-Commit-Position: refs/heads/master@{#8369}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8369 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 21:14:44 +00:00
perkj@webrtc.org
96e4db9bea Split peerconnection_jni.cc into separate files.
For now:
java_helpers - JNI convenience functions etc. Can in theory be moved to libjingle / webrtc general one day.
classreferenceholder - app/webrtc specific Java class loader.
androidvideocapturer_jni - the jni part of the video capturer I added.
peerconnection_jni - all the rest.

This also move all jni specifics into ns webrtc_jni to avoid naming collision.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38099004

Cr-Commit-Position: refs/heads/master@{#8363}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8363 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 12:47:21 +00:00
solenberg@webrtc.org
40fdb8ab96 Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway.
BUG=3871
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41879004

Cr-Commit-Position: refs/heads/master@{#8359}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8359 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-13 11:09:43 +00:00
solenberg@webrtc.org
503c33666f Re-enabling LocalP2PTestAnswerVideo and LocalP2PTestAnswerAudio test cases in peerconnection_unittest.
BUG=2288
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39919004

Cr-Commit-Position: refs/heads/master@{#8350}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8350 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 13:13:47 +00:00
andresp@webrtc.org
ff689be3c0 Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35079004

Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 11:55:32 +00:00
phoglund@webrtc.org
006521d5bd Makes libjingle_peerconnection_android_unittest run on networkless devices.
PeerConnectionTest.java currently works, but only on a device with
network interfaces up. This is not a problem for desktop, but it is a
problem when running on Android devices since the devices in the lab
generally don't have network (due to the chaotic radio environment in
the device labs, devices are simply kept in flight mode).

The test does work if one modifies this line in the file
webrtc/base/network.cc:

bool ignored = ((cursor->ifa_flags & IFF_LOOPBACK) ||
                IsIgnoredNetwork(*network));

If we remove the IFF_LOOPBACK clause, the test starts working on
an Android device in flight mode. This is nice - we're running the
call and packets interact with the OS network stack, which is good
for this end-to-end test. We can't just remove the clause though since
having loopback is undesirable for everyone except the test (right)?
so we need to make this behavior configurable.

This CL takes a stab at a complete solution where we pass a boolean
all the way through the Java PeerConnectionFactory down to the
BasicNetworkManager. This comes as a heavy price in interface
changes though. It's pretty out of proportion, but fundamentally we
need some way of telling the network manager that it is on Android
and in test mode. Passing the boolean all the way through is one way.

Another way might be to put the loopback filter behind an ifdef and
link a custom libjingle_peerconnection.so with the test. That is hacky
but doesn't pollute the interfaces. Not sure how to solve that in GYP
but it could mean some duplication between the production and
test .so files.

It would have been perfect to use flags here, but then we need to
hook up gflags parsing to some main() somewhere to make sure the
flag gets parsed, and make sure to pass that flag in our tests.
I'm not sure how that can be done.

Making the loopback filtering conditional is exactly how we solved the
equivalent problem in content_browsertests in Chrome, and it worked
great.

That's all I could think of.

BUG=4181
R=perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36769004

Cr-Commit-Position: refs/heads/master@{#8344}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8344 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 09:24:25 +00:00
perkj@webrtc.org
83bc721c7e Add Android specific VideoCapturer.
The Java implementation of VideoCapturer is losely based on the the work in webrtc/modules/videocapturer.

The capturer is now started asyncronously.
The capturer supports easy camera switching.

BUG=
R=henrika@webrtc.org, magjed@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30849004

Cr-Commit-Position: refs/heads/master@{#8329}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8329 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 11:27:22 +00:00
henrika@webrtc.org
62f6e75673 Refactoring WebRTC Java/JNI audio recording in C++ and Java.
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:

- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup

Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).

BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33969004

Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 08:39:19 +00:00
glaznev@webrtc.org
bc35703694 Add a method to remove an existing renderer from the internal list of Android renderers.
BUG=4290
R=jiayl@webrtc.org, mquiros@google.com

Review URL: https://webrtc-codereview.appspot.com/36089004

Cr-Commit-Position: refs/heads/master@{#8320}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8320 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-10 23:23:47 +00:00
glaznev@webrtc.org
44ae4c8b07 Support using VP9 video codec in AppRTCDemo.
- Add peer connection Java API to initialize field trial string.
- Add setting option to select VP8 or Vp9 as default video codec.
- Minor code clean up and allowing 720p portrait encoding.

R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39899004

Cr-Commit-Position: refs/heads/master@{#8303}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8303 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 23:26:41 +00:00
andresp@webrtc.org
53d9012faf Clean kForever from basictypes and move it to the interfaces that actually have it.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33269004

Cr-Commit-Position: refs/heads/master@{#8296}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8296 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 14:19:39 +00:00
pbos@webrtc.org
8cf9bdb3fa Remove USE_WEBRTC_DEV_BRANCH.
talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.

R=bjornv@webrtc.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/39849004

Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:17:12 +00:00
guoweis@webrtc.org
57ac2c84dd Default destination used by c line should be IPv4 only to avoid parsing error in legacy client.
Make sure the IP family overwrites the preference of candidates. Also,
make sure only UDP is used as default destination.

BUG=4269
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36009004

Cr-Commit-Position: refs/heads/master@{#8258}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8258 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-06 00:45:43 +00:00
glaznev@webrtc.org
f6932297e7 Fix Android video renderer to support video frames
with stride > width.

Recent libvpx update generates output video frames with stride
value greater than width, which was not supported by Android OpenGL
video renderer (Android GLES2 doesn't have GL_UNPACK_ROW_LENGTH
to provide stride information for buffer in glTexImage2D call).

Fix it by implementing native frame copying for Java
VideoRenderer.I420Frame implementation.

BUG=4248
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/40639004

Cr-Commit-Position: refs/heads/master@{#8252}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8252 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-05 17:30:17 +00:00
pthatcher@webrtc.org
877ac765ad Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
  confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.

This is a re-roll of 8237 (https://webrtc-codereview.appspot.com/39699004) with a default GetOption implementation.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38909004

Cr-Commit-Position: refs/heads/master@{#8245}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8245 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 22:03:41 +00:00
bjornv@webrtc.org
520a69e8ea Revert 8238 "Add RefCounting for TransportProxies"
Failing on Mac64_Debug

> Add RefCounting for TransportProxies
> 
> BUG=1574
> R=pthatcher@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/37869004

TBR=decurtis@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37159004

Cr-Commit-Position: refs/heads/master@{#8243}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8243 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 12:46:13 +00:00
bjornv@webrtc.org
c5f697135e Revert 8237 "Cleanup and prepare for bundling."
libjingle_peerconnection_objc_test consistently failing on Mac64 Debug.

> Cleanup and prepare for bundling.
> 
> - Add a GetOptions function. Needed for eventual bundle testing to
>   confirm that channel options are preserved.
> - Simplify unit tests and cleanup unused code.
> 
> BUG=1574
> R=pthatcher@webrtc.org, tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/39699004

TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34959004

Cr-Commit-Position: refs/heads/master@{#8241}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8241 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 10:22:43 +00:00
decurtis@webrtc.org
e2506670a4 Add RefCounting for TransportProxies
BUG=1574
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37869004

Cr-Commit-Position: refs/heads/master@{#8238}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8238 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 23:19:23 +00:00
pthatcher@webrtc.org
af01d93aa2 Cleanup and prepare for bundling.
- Add a GetOptions function. Needed for eventual bundle testing to
  confirm that channel options are preserved.
- Simplify unit tests and cleanup unused code.

BUG=1574
R=pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39699004

Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 23:14:18 +00:00
decurtis@webrtc.org
322a564f49 Fix datachannel stats id and timestamp.
Makes the id now be "datachannel_#####" where '####' is the id number for the datachannel.

Adds a timestamp to the data channel reports.

Implements unit tests to verify that the timestamp is set correctly.

BUG=1805
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33119004

Cr-Commit-Position: refs/heads/master@{#8236}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8236 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-03 22:10:13 +00:00
pkasting@chromium.org
005b6fffe6 Convert some EXPECTs to ASSERTs to avoid crashes when object creation fails.
BUG=none
TEST=none
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39649004

Cr-Commit-Position: refs/heads/master@{#8222}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8222 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-30 19:42:17 +00:00
tommi@webrtc.org
4161715e3f Remove ChangeUniqueID.
This fixes a two year old TODO of deleting dead code :)
In cases where the _id or id_ member variable is being used for tracing,
I changed the member to at least be const.

It doesn't look like id's are that useful anymore so maybe the next step is to get rid of them.

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

Review URL: https://webrtc-codereview.appspot.com/37849004

Cr-Commit-Position: refs/heads/master@{#8201}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8201 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-29 12:14:13 +00:00
glaznev@webrtc.org
8501ee632b Support VP8 HW decoding on devices with Exynos codec.
R=wzh@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8160 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-26 23:07:19 +00:00
jiayl@webrtc.org
dacdd9403d Reland r7980:
Accept incoming pings before remote answer is set, to reduce connection latency.
Set ICE connection state to 'checking' after setting the remote answer, so that it can transition into 'connected' if the peer reflexive connection is up before any remote candidate is set. See more details in crbug/446908

BUG=4068, crbug/446908
R=juberti@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8141 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 17:33:34 +00:00
kjellander@webrtc.org
a02d76845f Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due to flakiness
Disabling the test on all platforms since it's likely it can happen
on any platform, even if it's only been observed on Win x64 Release.

Running tests in parallel is a huge performance benefit to the team,
since it approximately reduces build cycle with 60-75%.

BUG=4219
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8138 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-23 14:34:52 +00:00
tommi@webrtc.org
586f2eda0d Change GetStreamBySsrc to not copy StreamParams.
This is something I stumbled upon while looking at string copying we do (in spades) and did a simple change to not be constantly copying things around needlessly. There's a lot more that can be done in these files of course so this is sort of a reminder for future code edits that it's possible to design interfaces/function in a way that's more performance aware and avoid forcing creation of copies, while still being very simple.  Also, we can use lambdas now :)

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

Review URL: https://webrtc-codereview.appspot.com/41589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8131 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 23:00:41 +00:00
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
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
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
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
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
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
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
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