Don't dismiss the presented view controller if it's already being dismissed to clear a warning about dismissing from a view controller while a dismiss is in progress.
Remove the sample buffer delegate when capture is being stopped to avoid a crash when a delegate method is sent to a deallocated object.
BUG=webrtc:4734
R=jiayl@webrtc.org, tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54669004.
Patch from Jon Hjelle <hjon@andyet.net>.
Cr-Commit-Position: refs/heads/master@{#9430}
- Set Camera.ErrorCallback callback when opening camera to
receive camera server error notifications.
- Allow user to provide interface for handling camera errors
happening on camera thread.
- Run camera observer on camera thread and monitor camera fps
and amount of callback buffers, print statistics and report error
if camera stops generating frames.
- Query camera formats starting from front camera instead of back
camera to detect camera failures as fast as possible.
- Change all DCHECK to CHECK in androidvideocapturer.cc to detect
camera error on release builds.
- Plus adding some extra logging.
R=hbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52519004
Cr-Commit-Position: refs/heads/master@{#9221}
Introduces a new capture class derived from cricket::VideoCapturer that
provides the ability to switch cameras and updates AppRTCDemo to use it.
Some future work pending to clean up AppRTCDemo UI.
BUG=4070
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48279005
Cr-Commit-Position: refs/heads/master@{#9137}
It breaks integration with upstream re2 library on Chromium.
Without patching re2 library, with this define, it produces the
following error:
../../third_party/re2/re2/re2.h:254:5: error: expected identifier
POSIX, // POSIX syntax, leftmost-longest match
As we define POSIX on the command line, the C preprocessor changes
RE2::POSIX to nothing and thus break the compilation. :(
See chromium-dev mailing list for this discussion in
https://groups.google.com/a/chromium.org/d/topic/chromium-dev/UXCHnX7pV44/discussion
BUG=None
TEST=ninja -C out/Debug, everything compiles as before
R=sergeyu@chromium.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46049004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#9032}
- Do not handle more than one camera switch request at a time
to avoid blocking camera thread with multiple switch requests.
- Add a callback to notify when camera switch has been done.
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46859004
Cr-Commit-Position: refs/heads/master@{#8978}
Also prevents that we try to restore audio mode when it has not been changed.
TBR=glaznev
BUG=NONE
TEST=AppRTCDemo and verify that volume control switches from "Ringtone to Phone" mode when call starts and switches back to Ringtone mode when call ends.
Review URL: https://webrtc-codereview.appspot.com/46879004
Cr-Commit-Position: refs/heads/master@{#8975}
- Make PeerConnectionClient a singleton.
- Fix crash in CpuMonitor.
- Remove reading constraints from room response.
- Catch and report camera errors.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43059004
Cr-Commit-Position: refs/heads/master@{#8930}
This should fix the TODO in that header.
BUG=None
TEST=ninja -C out/Debug still compiles everything.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47919004
Patch from Thiago Farina <tfarina@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8921}
AppRTCDemo is failing to cleanly exit a room because it sends a GET request to /bye. The request to /bye should be a POST request. Because the /bye request is failing, the room is still marked as "full" and rejoining will fail.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47759004
Patch from Chuck Hays <haysc@webrtc.org>.
Cr-Commit-Position: refs/heads/master@{#8860}
- Add HUD fragment with HUD related controls and more
HUD statistics.
- Create and set all peer connection constraints in
PeerConnectionClient class.
- Handle registration request in web socket class internally
once web socket connection is opened.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44669004
Cr-Commit-Position: refs/heads/master@{#8762}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8762 4adac7df-926f-26a2-2b94-8c16560cd09d
Relevant changes:
* src/third_party/android_tools: fd5a8ec..98a4345
Details: 00e438c..8d51d96/DEPS
This required updating our Android projects to API level 22,
as third_party/android_tools dropped support for API level 21.
Command used:
perl -pi -e "s/android-21/android-22/g" `find . -name project.properties`
Using 'android update project' would also work but that changes the
ANDROID_SDK_ROOT -> ANDROID_HOME, which the Chromium build toolchain
doesn't set properly when build/android/envsetup.sh is sourced.
Clang version was not updated in this roll.
R=henrika@webrtc.org, perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42779004
Cr-Commit-Position: refs/heads/master@{#8728}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8728 4adac7df-926f-26a2-2b94-8c16560cd09d
The AppRTCDemoTest is failing if the Android device lacks
an Internet connection (e.g. is in flight mode).
This change makes it benefit from the work done in
https://review.webrtc.org/36769004/ to work around that
limitation for loopback tests.
R=phoglund@webrtc.orgTBR=glaznev@webrtc.org
BUG=4421
TESTED=Successful run on Nexus 7 (2013) in flight mode using:
ninja -C out/Release
. build/android/envsetup.sh
adb install -r out/Release/apks/AppRTCDemo.apk
webrtc/build/android/test_runner.py instrumentation --test-apk AppRTCDemoTest --verbose --release
Review URL: https://webrtc-codereview.appspot.com/45649004
Cr-Commit-Position: refs/heads/master@{#8714}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8714 4adac7df-926f-26a2-2b94-8c16560cd09d
- Updated Java VideoRenderer removes setSize() from video renderer interface.
Remove no longer valid test, which requires setSize() call before any
frame can be rendered.
- test_runner.py tries to run private member of InstrumentationTestCase class.
Workaround it by renaming private loopback test method.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47549004
Cr-Commit-Position: refs/heads/master@{#8707}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8707 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add front/back camera switch test.
- Add video source stop and restart test to simulate
application going into background.
- Add a loopback test for 3 video codecs - VP8, VP8, H.264.
- Add a loopback test for voice only call.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43459004
Cr-Commit-Position: refs/heads/master@{#8560}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8560 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL makes sure the project compiles, it does not ensure the parameters now used are correct!
There may be something strange going on with the build files. I was previously able to recompile the whole project despite of the incorrect code, until I changed the file and tried again.
The changes made are just so that it will compile. The code should likely be updated by someone who knows what he/she is doing.
TBR=glaznev@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45369004
Cr-Commit-Position: refs/heads/master@{#8526}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8526 4adac7df-926f-26a2-2b94-8c16560cd09d
Peer connection may generate answer and ICE candidates before
websocket client is registered. Remove check from sendAnswer()
and sendLocalIceCandidate() functions and allow websocket client
to accumulate messages and send them later once it will be
registered.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44379004
Cr-Commit-Position: refs/heads/master@{#8508}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8508 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add a setting option to disable outgoing video in a call.
- Add an option to select audio codec.
- Add an option to specify audio bitrate for Opus codec.
- Plus add an option to select H.264 as default video codec.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42449004
Cr-Commit-Position: refs/heads/master@{#8468}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8468 4adac7df-926f-26a2-2b94-8c16560cd09d
- Allow to configure MediaCodec Java wrapper to use VP8
and H.264 codec.
- Save H.264 config frames with SPS and PPS NALUs and append them to every key frame.
- Correctly handle the case when one encoded frame may generate several output NALUs.
- Add code to find H.264 start codes.
- Add a flag (non configurable yet) to use H.264 in AppRTCDemo.
- Improve MediaCodec logging.
R=wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43379004
Cr-Commit-Position: refs/heads/master@{#8465}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8465 4adac7df-926f-26a2-2b94-8c16560cd09d
