c6db88b0cf
Make it possible to build webrtc for arm64.
...
- Bump revision of protobuf lib
- Remove -Wextra for arm64 gcc targets (warnings in stlport)
- Add MemoryBarrier implementation in single_rw_fifo.cc.
- [pending 15619004]: Bump revision of /deps/tools/android to get md5sum_bin for arm64.
BUG=chromium:354405,chromium:354539
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15629004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6330 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 17:15:42 +00:00
88fbb2d86b
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
Same as https://webrtc-codereview.appspot.com/19519004 . The issue in
http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Linux ...
is solved by this change
http://src.chromium.org/viewvc/chrome/trunk/src/third_party/libjingle/libjing ...
(tested locally).
BUG=3380
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17619005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6218 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 21:18:46 +00:00
2fa7f79094
Revert 6202 "Switch to using base/constructormagic.h and remove ..."
...
> Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
>
> BUG=N/A
> R=andrew@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/19519004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14579007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6210 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 11:07:29 +00:00
125ffd709d
Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6202 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-20 15:20:44 +00:00
cb711f77d2
Add interface to propagate audio capture timestamp to the renderer.
...
BUG=3111
R=andrew@webrtc.org , turaj@webrtc.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6189 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-19 17:39:11 +00:00
14abcc7322
libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict.
...
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 16:54:44 +00:00
8f69330310
Replace scoped_array<T> with scoped_ptr<T[]>.
...
scoped_array is deprecated. This was done using a Chromium clang tool:
http://src.chromium.org/viewvc/chrome/trunk/src/tools/clang/rewrite_scoped_ar ...
except for the few not-built-on-Linux files which were updated manually.
TESTED=trybots
BUG=2515
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5985 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-25 23:10:28 +00:00
d59359af4d
Remove 44.1 kHz workaround from the iOS AudioDevice.
...
Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.
This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.
BUG=1395
R=fischman@webrtc.org , henrikg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-22 18:07:49 +00:00
ca539bbed0
iOS: baby steps to being able to include_tests=1
...
- pull iossim in DEPS even when on mac (because bug 2152)
- fix audio_device_test_api.cc's use of bool instead of bool* (!)
- move unused-on-mobile message to non-mobile-only section of
hardware_before_streaming_test.cc
BUG=3185
R=kjellander@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5914 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-15 20:26:41 +00:00
2c89b5cb27
Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
...
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done
(and then removed the talk/ impact)
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
5692531f18
Added a new OnMoreData() interface which will not feed the playout data to APM.
...
BUG=3147
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11059005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5895 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 10:50:37 +00:00
c7c432aa9b
Remove AudioDevice::{Microphone,Speaker}IsAvailable.
...
This was only used for logging, except on Mac, where the methods are
now private.
BUG=3132
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5831 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-02 16:49:26 +00:00
a789f3720a
VoiceEngine(iOS & Android): removed NOT_SUPPORTED
...
Also:
- removed underflow of a uint32 creating crazy-large delay values
- removed always-fail AudioDeviceIPhone::MicrophoneIsAvailable() impl (see
bug 3132)
- removed unnecessary exclusion of features from iOS & Android builds
BUG=2050,3132
R=andrew@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10909005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5820 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-01 00:16:35 +00:00
0e65fdaa3b
Fix "unreachable code" warnings (MSVC warning 4702) in webrtc.
...
BUG=chromium:346399
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10139004
Patch from Peter Kasting <pkasting@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5747 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-21 10:26:42 +00:00
66061992fb
ifdef the alsa code based on macro USE_X11
...
BUG=none
TEST=try bots
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5583 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 03:05:05 +00:00
c1e28038ba
Moved the new OnData interface to AudioTranport, and expose the AudioTransport pointer via voe_base
...
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7779004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-02 15:30:20 +00:00
32c26eb90b
Android, OpenSlDemo: moved to webrtc/examples/android/opensl_loopback
...
BUG=N/A
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5400 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-17 23:12:51 +00:00
ead202b973
Android, OpenSlDemo: fixes issue where app would crash as soon as the application is started.
...
BUG=2801
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7259005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5398 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-16 23:26:37 +00:00
79cf3acc79
Removes usage of ListWrapper from several files.
...
BUG=2164
R=andrew@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5373 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 15:21:30 +00:00
573a1b45b5
Android: Fixes crash when exiting WebRTCDemo.
...
BUG=2738
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5365 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-10 22:58:06 +00:00
000dde99c8
Android build: make it quiet on success and not overly noisy on failure.
...
- OpenSLDemo and WebRTCDemo get the sauce that AppRTCDemo got in r5271
- libjingle_peerconnection_jar is now silent on success
- Fix a bug introduced by r5271 which caused ant logs to be emitted to a subdir of talk/examples instead of in the gyp output directory.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6199005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5332 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 22:49:35 +00:00
f6acf98a46
Fix the android clang bot for compiling with thread annotations.
...
TBR=niklas.enbom@webrtc.org
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6279005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5330 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 21:54:26 +00:00
7fb75ecbd4
Add thread_annotations for clang targets.
...
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.
R=niklas.enbom@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6209004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 20:20:50 +00:00
179908c81c
JNI Audio: remove dead members.
...
BUG=2735
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6049004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5312 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-17 23:46:14 +00:00
9ee75e9c77
Enables mixing and matching Java and native audio. It is used for getting best of both worlds capabilities (AEC and low latency).
...
BUG=N/A
R=fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4189004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5270 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 21:42:44 +00:00
f9bdbe3619
Roll chromium_revision 232627:238260
...
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003
TEST=trybots passing
BUG=none
R=andrew@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
7ae8495779
Removed unnecessary Pulse init from VoE startup.
...
Saves 10% (~260ms) of the total PeerConnectionTest wallclock time.
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5254 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-10 21:01:34 +00:00
d3865e9124
Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close).
...
It is incorrect to wrap close in HANDLE_EINTR on Linux.
BUG=chromium:269623
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4759004
Patch from Mark Mentovai <mark@chromium.org >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5206 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-03 19:10:20 +00:00
a750044396
Fixes a crash in VoE when unregistering JNI hooks.
...
BUG=11695087
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3939004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5144 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-20 22:32:12 +00:00
b082ade3db
Hook up audio/video sync to Call.
...
Adds an end-to-end audio/video sync test.
BUG=2530, 2608
TEST=trybots
R=henrika@webrtc.org , mflodman@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5128 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-18 11:45:11 +00:00
b8cb85b348
Fix broken build on x86 Android
...
BUG=2545
R=fischman@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3019004
Patch from Lu Quiang <qiang.lu@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-04 19:06:08 +00:00
621df678c8
WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN.
...
Mostly to remove a long-standing TODO...
TESTED=trybots
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2369005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 10:27:23 +00:00
3555303cb0
Roll chromium_revision 226126:228675 and fix clang warnings
...
By request from thakis@chromium.org , I disabled the
-Wno-unused-const-variable setting that is set in Chromium's
common.gypi so we can prepare our code for it's removal.
This required some cleanup in order to get the code to compile
with Clang having the -Wunused-const-variable warning enabled.
TEST=all trybots passing
BUG=none
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2400004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4966 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-15 20:10:17 +00:00
05773e5a70
Android OpenSlDemo: remove some usages of deprecated APIs that is breaking the bots.
...
TBR=fischman@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/2395004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4956 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-14 16:25:11 +00:00
f53622d42e
WebRTCDemo: Fixes warning for devices with pre-17 API level. Also fixes broken build build.xml and project.properties.
...
BUG=2083
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2375004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4951 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-11 21:28:26 +00:00
3f9288f987
Add APK and isolate target for video_engine_tests
...
Add .isolate file and _run target for video_engine_tests.
Move tools/swarm_client to be untracked in all .isolate file,
so refactorings in swarm_client doesn't require us updating
all our .isolate files (similar to the changes for the
Chromium tests done in:
https://src.chromium.org/viewvc/chrome?view=rev&revision=218844 )
Update modules_unittests.isolate with new NetEq4 reference files
needed.
TEST=trybots passing
I also setup a Chromium workspace where I patched third_party/webrtc
with the changes in this CL, followed by compiling with the settings
described in
https://code.google.com/p/webrtc/issues/detail?id=1882#c11
I then verified that the video_engine_tests_apk dir was created
in the output folder.
BUG=1916,2462
R=andrew@webrtc.org , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2344007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4925 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 18:20:38 +00:00
ad2eb6f67d
Unbreaks Android build after r4915.
...
TBR=ajm@webrtc.org
BUG=Not filed
Review URL: https://webrtc-codereview.appspot.com/2348005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4921 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 14:21:23 +00:00
4e65e07e41
VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
...
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.
BUG=1407
R=henrike@webrtc.org , wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2334004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:23:13 +00:00
2a97317953
Fix include of isolate.gypi
...
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.
The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.
TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).
I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).
I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.
Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc
BUG=1916
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2338004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 19:31:16 +00:00
f8f78b1316
Android OpenSL: Fixes faulty assertion in jni-code.
...
BUG=2452
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2342004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4906 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 18:41:06 +00:00
1fdc51ae2a
APK for opensl loopback.
...
BUG=N/A
R=andrew@webrtc.org , fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2212004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4901 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 14:58:19 +00:00
6049787252
Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from
...
a different thread.
One example is the _playWarning can be changed in AudioDeviceLinuxALSA::Init, which is called on the application's thread. At the same time it can be read via PlayoutWarning() on the VoE's process_thread.
RISK=P2
TESTED=try bots and tsan test:
tools/valgrind-webrtc/webrtc_tests.sh --tool=tsan -t out/Debug/libjingle_peerconnection_unittest --gtest_filter=PeerConnectionFactoryTestInternal.CreatePCUsingInternalModules
BUG=1205
R=xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2315004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4866 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-27 18:19:25 +00:00
c8dea6a00f
Use the native sample rate for OpenSL recording.
...
BUG=N/A
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2219005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4771 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-17 18:44:51 +00:00
2553450959
Fix win trybot errors due to r4729.
...
The addition of logging.h in r4729 was causing the win trybot to fail
with "#pragma deprecated" errors in standard library headers. This
turned out to be due to including strsafe.h (via audio_device_config.h)
before sstream (via logging.h).
strsafe.h was only being included for the unused DEBUG_PRINT macro. I
removed all references to it.
This incidentally removes a bunch of other unneeded headers discovered
while trying to track the problem down.
This didn't show up in the commitbots; my guess is that the trybots are
using the VC10 toolchain and the commitbots the VC11 toolchain.
TBR=pbos
Review URL: https://webrtc-codereview.appspot.com/2204004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4738 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 00:02:13 +00:00
5eb997a2fd
Fix unsigned/signed comparison error due to r4729.
...
Fix it for real by switching to ints rather than casting.
TBR=xians
Review URL: https://webrtc-codereview.appspot.com/2191009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4730 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-12 01:01:42 +00:00
8f94013651
Reduce frequency of high audio delay warning logs.
...
This will log the warning every 5 seconds instead of every 10 ms.
BUG=b/10674993
TESTED=Ran voe_cmd_test with hard-coded high delay. Observed a log
every 5 seconds.
R=noahric@chromium.org , xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2184009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4729 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 22:35:00 +00:00
6138c5cfa4
OpenSl: fixes crashes externally reported in issue 2361 and 2362.
...
BUG=2361,2362
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4726 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 18:50:06 +00:00
835ef67d14
Remove repeated conditions key.
...
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2196007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4720 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-11 00:16:00 +00:00
82f014aa0b
OpenSL (not default): Enables low latency audio on Android.
...
BUG=1669
R=andrew@webrtc.org , fischman@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2032004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 18:24:07 +00:00
e141373b8a
Add isolate configuration for Android for all tests.
...
In https://code.google.com/p/webrtc/source/detail?r=4407
henrike@ added the path to the WebRTC resources and
data directories for Android that are required in order to
use isolate for test execution on Android.
This CL adds similar entries to the rest of the .isolate
files added in
https://code.google.com/p/webrtc/source/detail?r=4590 .
It also removes three accidentally added .isolate files that originated
from old test names:
* audio_device_test_api
* video_capture_module_test
* video_render_module_test
BUG=1882,1916
TEST=trybots passing.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2107004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4627 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 12:10:09 +00:00
