The synchronization access is unnecessary for rtc::Thread::WrapCurrent (called from JingleThreadWrapper) since JingleThreadWrapper never calls rtc::Thread::Stop or rtc::Thread::Join. Failing to get the access caused crashes in Chrome since rtc::Thread::Current will be NULL when rtc::Thread::WrapCurrent fails.
rtc::ThreadManager::WrapCurrentThread still requires the synchronization access, since I am not sure if the callers (e.g. the plugin) depends on it.
BUG=crbug/413853
R=juberti@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7224 4adac7df-926f-26a2-2b94-8c16560cd09d
The .gclient_entries file is written after a successful
gclient sync operation and contains paths mapped to URLs for
all DEPS entries that have been synced.
This has been causing problems for users when switching from
the legacy Subversion based checkouts to the new DEPS approach
using a Chromium Git checkout combined with symlinks.
Also it has been discovered that when entries have been
removed from the Chromium DEPS file, subsequent gclient sync
operations fail when it's trying to process those directories.
This CL changes so that .gclient_entries is wiped for the WebRTC
checkout when moving from the legacy SVN to Git.
It also wipes the chromium/.gclient_entries file when a new Chromium
revision is about to be synced, to avoid problems when DEPS entries
have been removed.
BUG=415219
R=agable@chromium.org
Review URL: https://webrtc-codereview.appspot.com/28509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7222 4adac7df-926f-26a2-2b94-8c16560cd09d
Targets must now link with implementation of their choice instead of at "gyp"-time.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common
Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests
GN changes:
- Not many since there is almost no test definitions.
Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.
Re-enable android tests by reverting 7026 (some tests left disabled).
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.orgTBR=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
into its own targets. Dependencies must link directly with the desired one.
Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.
Targets linking with default/external capture implementation:
- anything dependent on webrtc_test_common
- anything dependent on video_engine_core
Targets linking with internal capture implementation:
- vie_auto_test
- anything dependent on webrtc_test_renderer
GN changes:
- Not many since there is almost no test definitions.
TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3768
R=glaznev@webrtc.orgTBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
Replacing #include of "base/basictypes.h" in
overrides/webrtc/base/constructormagic.h with "base/macros.h". Our
version of constructormagic.h is not meant to include the base types,
only DISALLOW_COPY_AND_ASSIGN etc.
This fix is also a workaround for our overrides in Chromium seemingly
including the wrong things for certain webrtc targets like
audio_processing, so it looks like this #include "base/basictypes.h"
didn't include Chromium's base/basictypes.h but webrtc/base/basictypes.h
somehow, hence DISALLOW_COPY_AND_ASSIGN wasn't defined, causing the
revert in r7151.
R=henrike@webrtc.org, tommi@webrtc.org
BUG=3070
TEST=Chromium still builds locally with this change.
Review URL: https://webrtc-codereview.appspot.com/27509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7204 4adac7df-926f-26a2-2b94-8c16560cd09d
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7200
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7201 4adac7df-926f-26a2-2b94-8c16560cd09d
1) return the first global temporary and non-deprecrated ones.
2) if #1 not available, return global one.
3) if #2 not available, use ULA ipv6 as last resort.
ULA stands for unique local address. They are only useful in a private
WebRTC deployment. More detail: http://en.wikipedia.org/wiki/Unique_local_address
BUG=3808
At this point, rule #3 actually won't happen at current
implementation. The reason being that ULA address starting with 0xfc 0r 0xfd will be grouped into its own Network. The result of that is WebRTC will have one extra Network to generate candidates but the lack of rule #3 shouldn't prevent turning on IPv6 since ULA should only be tried in a close deployment anyway.
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7200 4adac7df-926f-26a2-2b94-8c16560cd09d
This is based on webrtc/build/merge_libs.gyp, with a dependency on
voice_engine.gyp instead and suitable name changes.
Executing:
$ rm -rf out/
$ ./webrtc/build/gyp_webrtc -Denable_video=0 -Denable_protobuf=0
-Drelease_optimize=s webrtc/build/merge_libs_voice.gyp
$ ninja -C out/Release merged_lib_voice
results in a minimially sized voice engine lib at:
out/Release/librtc_voice_merged.a
Linux: 6.4 MB
Mac: 3.7 MB
R=braveyao@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23659004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7199 4adac7df-926f-26a2-2b94-8c16560cd09d
These optimizations were originally committed in r6860, but reverted in r6861, since it broke a bitexactness test (ApmTest.Process) in modules_unittests. That test has now been updated in r7149, hence this CL now pass the test.
BUG=3767
TESTED=manually on linux and trybots
TBR=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7189 4adac7df-926f-26a2-2b94-8c16560cd09d
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)
This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.
The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.
BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
- Change hw video decoder wrapper to allow to feed multiple input
and query for an output every 10 ms.
- Add an option to decode video frame into an Android surface object. Create
shared with video renderer EGL context and external texture on
video decoder thread.
- Support external texture rendering in Android renderer.
- Support TextureVideoFrame in Java and use it to pass texture from video decoder
to renderer.
- Fix HW encoder and decoder detection code to avoid query codec capabilities
from sw codecs.
BUG=
R=tkchin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7185 4adac7df-926f-26a2-2b94-8c16560cd09d
When writing to wav files in the low level flag aec_debug_dump incorrect sample rates were used for recordings using rates from 32 kHz and above. This since internally inside the AEC we process the data using 16 kHz. Any upper band is processed and combined later on.
This CL adds the correct sample rate to the recording.
BUG=3359
TESTED=locally on 44.1 kHz recordings on Linux
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7182 4adac7df-926f-26a2-2b94-8c16560cd09d
This requires two fixes:
1. Use DCHECK instead of assert in ThreadChecker's unittest.
2. Activate DCHECK when DCHECK_ALWAYS_ON in enabled.
Both these modifications are in line with Chromium's implementation.
The ThreadChecker unittest was changed to use assert instead of DCHECK
on the initial import (since WebRTC did not have a DCHECK back then).
BUG=3803
TEST=local out/{Debug,Release}/rtc_unittests built with and without DCHECK_ALWAYS_ON
R=andrew@webrtc.org, pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7178 4adac7df-926f-26a2-2b94-8c16560cd09d
