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
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
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
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
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
- 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
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
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
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
