henrike@webrtc.org
8b0b21161a
Revert 7093: "Implementing ICE Transports type handling in libjingle transport."
...
TBR=mallinath@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/28419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7112 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 22:46:28 +00:00
brettw@chromium.org
519c9e207d
Convert GN visibility to be a list.
...
GN visibility currently allows either string or list types, but this is causing
some problems for some templates. I'm going to require it to be lists, so am
changing all callers before pushing the new binary.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25439004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7111 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 22:45:18 +00:00
pbos@webrtc.org
7118e61669
Finish work queue in SctpDataMediaChannelTest.
...
Always finishing the work queue prevents memory leak detected in
LeakSanitizer (packet is deleted on the receiver side).
R=jiayl@webrtc.org
BUG=3608,chromium:375154
Review URL: https://webrtc-codereview.appspot.com/28399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7110 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 21:44:07 +00:00
jiayl@webrtc.org
0e52772aa9
Fix a bot-breaking memory leak from early returning in ParseMediaDescription.
...
BUG=3791
R=henrike@webrtc.org , pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7109 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 21:43:43 +00:00
jiayl@webrtc.org
c172320bd2
Revert "Fixes two issues in how we handle OfferToReceiveX for CreateOffer:" because it broke content_browsertests on Android.
...
This reverts commit r7068.
TBR=kjellander@webrtc.org
BUG=2108
Review URL: https://webrtc-codereview.appspot.com/23539004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7108 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 20:44:36 +00:00
andrew@webrtc.org
17454f79dc
Add ctors to ChannelBuffer to enable copying on construction.
...
Also:
- Fix the constness of some parameters.
- Add more const overloads.
- Use DCHECK in place of assert.
- Removed an unnecessary memset.
R=claguna@google.com
Review URL: https://webrtc-codereview.appspot.com/24469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7107 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 20:27:04 +00:00
buildbot@webrtc.org
fd42f9dd6f
(Auto)update libjingle 74955991-> 75042522
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7106 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 19:45:36 +00:00
sprang@webrtc.org
1272ee59b3
Suppress uninitialized read warning in cricket::VideoFrame::Validate
...
BUG=3789
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7105 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 14:00:38 +00:00
henrik.lundin@webrtc.org
c64246f42c
Set a default speech type in iSAC wrapper
...
If the decoder encounters an error, it may leave the speech type
unassigned, leading to a use-of-uninitialized-value in subsequent lines.
BUG=crbug/411162
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23519004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7104 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:40:58 +00:00
henrik.lundin@webrtc.org
ed8bcd3ac5
Starting to implement the new ACM API
...
The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.
This is work in progress.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7103 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:13:19 +00:00
houssainy@google.com
9600519147
Adding the ability to test on Chrome for Android.
...
use "android-chrome" as type in rtcbot running command.
Example: node test.js android-chrome
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20329004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7102 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 13:01:40 +00:00
bjornv@webrtc.org
37c39f3784
audio_processing: Removed use of macro WEBRTC_SPL_UMUL_16_16
...
The macro replaced is a trivial multiplication after explicit casts to uint16_t and uint32_t. This CL replaces its use with "*" and adds explicit casts if necessary.
Affected components:
* AECMobile
* AGC
* Noise Suppression (fixed point version)
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7101 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 11:21:56 +00:00
bjornv@webrtc.org
0d394f3609
video_processing: Removed usage of WEBRTC_SPL_UMUL_16_16
...
The trivial macro WEBRTC_SPL_UMUL_16_16 is nothing but plain mutliplication of casted values. This CL explicitly use "*" at place and casts if necessary.
BUG=3348,3353
TESTED=locally on linux and trybots
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7100 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 11:19:39 +00:00
houssainy@google.com
c77e4d6aef
- Adding AndroidDeviceManager to botManager.js to help in selecting devices, in case running test on Android devices.
...
- Select BotType using nodeJs terminal command.
- ping_pong.js test added.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19159004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7099 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 10:36:11 +00:00
kjellander@webrtc.org
142bb9d870
Roll chromium_revision 94532b1..ea769fd
...
Summary of changes (git diff 94532b1..ea769fd DEPS):
* buildtools 2328da4..ea4dc0e
* third_party/android_tools 3186999..7fc902d
* third_party/boringssl 6c7aed0..7bdec13
* third_party/libjpeg_turbo 2ed5319..3963fbc
* third_party/libvpx 982d147..ceebbcc (r291730:291805)
* third_party/nss 90c5f9a..7b5b6ec
* third_party/usrsctp/usrsctplib e6e1833..8975bd5
BUG=3608
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7098 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 10:06:37 +00:00
stefan@webrtc.org
fe16167507
Fix RTT calculations for send-only channels.
...
As we don't know the SSRC of the other end in a send-only channel since we haven't received packets from that end, we are required to assume that the SSRC of the first report block is the correct SSRC to use for RTT calculations.
BUG=3781
R=mflodman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7097 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 08:45:25 +00:00
sprang@webrtc.org
c30e9e2300
Ignore FEC packet in stats, if it is first packet on ssrc.
...
BUG=chrome:410456
R=mflodman@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7096 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-08 08:20:18 +00:00
kjellander@webrtc.org
6d08ca6379
GN: Prefix WebRTC specific variables with "rtc_"
...
BUG=3441
TESTED=Trybots + Running GN in a Chromium checkout with
src/third_party/webrtc symlinked to the WebRTC checkout
with this CL applied, both with the default GN settings
and using: --args="os=\"android\" cpu_arch=\"arm\""
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/27379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7095 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:36:10 +00:00
kjellander@webrtc.org
f68cf93e1b
Add video_capture_tests_apk_target
...
In https://codereview.chromium.org/500423004/ the
target that was previously used to build the Android APK
tests was removed. When building these tests from a
standalone checkout, the video_capture_tests_apk target
was missing in the chain of targets that gets generated
into the 'all' target.
BUG=3764
TESTED=Trybots.
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7094 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 17:35:51 +00:00
mallinath@webrtc.org
7256d31d28
Implementing ICE Transports type handling in libjingle transport.
...
BUG=1179
R=juberti@webrtc.org , bemasc@webrtc.org , jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7093 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-07 04:08:44 +00:00
kjellander@webrtc.org
a781f68712
Fix rm command for class cleanup in r7091
...
In https://webrtc-codereview.appspot.com/20339004
the rm command was missing 'r' for recursive mode.
TBR=henrike@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/26379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7092 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-06 22:11:28 +00:00
kjellander@webrtc.org
9510022e1f
Cleanup temporary class files for OpenSlDemo
...
I've seen tryjobs failing when they shouldn't on
the Android trybots and I suspect this might have
something to do with it.
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7091 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-06 18:03:45 +00:00
thorcarpenter@google.com
cc060563f3
Remove unnecessary include from testutils.cc.
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7090 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 21:19:00 +00:00
buildbot@webrtc.org
992febb997
(Auto)update libjingle 74873066-> 74873164
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7089 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:39:08 +00:00
thorcarpenter@google.com
a3344cfda4
Fix webrtcvideoframe tests.
...
R=fbarchard@google.com , harryjin@google.com , henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7088 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:34:13 +00:00
jiayl@webrtc.org
ddb85ab85b
Updated SCTP SDP attributes according to draft-ietf-mmusic-sctp-sdp-07
...
- SDP sctpmap attribute replaced with fmtp attribute
- SDP sctp-port attribute is newly added
BUG=3592
R=jiayl@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7087 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 16:31:56 +00:00
henrik.lundin@webrtc.org
8f073c5054
Create a new interface for AudioCodingModule
...
This is a first draft of the interface, and is work in progress.
BUG=3520
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17299004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7085 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 13:16:23 +00:00
buildbot@webrtc.org
af5fa95258
(Auto)update libjingle 74857067-> 74860820
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7084 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 13:03:50 +00:00
buildbot@webrtc.org
7e3bd3d7de
(Auto)update libjingle 74851128-> 74857067
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7083 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 11:45:42 +00:00
buildbot@webrtc.org
bc6fa1876e
(Auto)update libjingle 74825992-> 74851128
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7082 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 11:08:01 +00:00
pbos@webrtc.org
287e9614b3
Disable TestDrain test on memcheck bots.
...
P2PTransportChannelMultihomedTest.TestDrain is flaky on memcheck bots,
likely the test timeout is insufficient for memcheck which incurs a
serious slowdown.
BUG=2409,3447
R=minyue@webrtc.org
TBR=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17339004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7081 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 10:11:24 +00:00
pbos@webrtc.org
cdb48dbc23
Enable VideoAdapterTest.BlackOutput on DrMemory.
...
DrMemory r2061 fixes how the instruction psrlw's shadow is mirrored ->
this false positive is now gone.
R=kjellander@webrtc.org
BUG=3754
Review URL: https://webrtc-codereview.appspot.com/25399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7080 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 09:46:34 +00:00
kjellander@webrtc.org
fed47dc205
Drop buildbot_tests.py script
...
This is no longer used since the buildbots have moved
over to recipes (where these arguments are configured).
See https://code.google.com/p/chromium/codesearch#chromium/tools/build/scripts/slave/recipe_modules/webrtc/api.py&l=73
for details.
This is essentially a revert of
https://webrtc-codereview.appspot.com/1021006
BUG=None
TESTED=None
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7079 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 08:25:38 +00:00
kjellander@webrtc.org
a2da031dc0
Remove use_relative_paths from DEPS
...
This makes it possible for us to migrate to using the bot_update step
on our buildbots. That would mean they'd use a Git checkout, which
brings stability, speed and best of all: re-enables the
DEPS-second-sync capability on our trybots that we've been lacking.
bot_update currently doesn't support the use_relative_paths variable
so the synced deps end up in the wrong path with it enabled.
Since Chromium doesn't use it, and it doesn't pollute our
DEPS file that much, I think we should switch.
NOTICE: Any custom_deps entries for the solution in .gclient have to be
updated to support this change, which includes the entry normally present
for Valgrind binaries. The bots will need to be updated as well at the
same time as landing this.
BUG=3534
TESTED=Verified a local sync works.
R=andrew@webrtc.org , niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7078 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 08:25:24 +00:00
henrik.lundin@webrtc.org
bcf75e32a3
Modifying audio_coding/codecs/OWNERS
...
Adding myself.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7077 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 07:18:50 +00:00
bjornv@webrtc.org
c2c4117477
common_audio: Replaced WEBRTC_SPL_LSHIFT_U32 with << in audio_processing
...
Affected components:
* AECMobile
- Added a help function since the same operation was performed several times.
* Auto Gain Control
* Noise Suppression (fixed point)
BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7076 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 06:01:53 +00:00
kjellander@webrtc.org
2c03a97d37
Roll chromium_revision f0a439d..94532b1
...
Cr-Commit-Position changes: 292861:293188
Changes:
* third_party/drmemory to r2062
* third_party/icu 527ea2d..8983113
* tools/gyp 1970:1972
BUG=3754
TESTED=Local compile and trybots.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7075 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 05:33:31 +00:00
buildbot@webrtc.org
818b7b3ac9
(Auto)update libjingle 74825084-> 74825992
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7074 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 00:14:03 +00:00
jiayl@webrtc.org
dfbcf8161e
Fix an issue in MediaStreamSignaling that a remotely create DataChannel is added to the list twice.
...
BUG=3778
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7073 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-05 00:01:12 +00:00
henrike@webrtc.org
f1427c6731
Revert 7070 "TurnPort should retry allocation with a new address on error
...
STUN_ERROR_ALLOCATION_MISMATCH."
TBR=jiayl@webrtc.org
BUG=N/A
Review URL: https://webrtc-codereview.appspot.com/15359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7072 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 22:21:33 +00:00
glaznev@webrtc.org
4b234044d5
Reduce maximum video resolution for Android.
...
HW video encoder and decoder can not be initialized
with 3840x2160 resolution.
BUG=3757,3738
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7071 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 19:50:07 +00:00
jiayl@webrtc.org
574f2f60fe
TurnPort should retry allocation with a new address on error STUN_ERROR_ALLOCATION_MISMATCH.
...
BUG=3570
R=juberti@webrtc.org , mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20999004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7070 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 19:11:34 +00:00
aluebs@webrtc.org
021e76fd39
Add support for WAV output in audioproc
...
The default output is a WAV file, except if the --pcm_output flag is set.
BUG=webrtc:3359
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7069 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 18:12:00 +00:00
jiayl@webrtc.org
52055a276d
Fixes two issues in how we handle OfferToReceiveX for CreateOffer:
...
1. the options set in the first CreateOffer call should not affect the result of a second CreateOffer call, if SetLocalDescription is not called after the first CreateOffer. So the member var options_ of MediaStreamSignaling is removed to make each CreateOffer independent.
Instead, MediaSession is responsible to make sure that an m-line in the current local description is never removed from the newly created offer.
2. OfferToReceiveAudio used to default to true, i.e. always having m=audio line even if no audio track. This is changed so that by default m=audio is only added if there are any audio tracks.
BUG=2108
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7068 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 17:12:25 +00:00
brettw@chromium.org
afa77cd803
Add direct_dependent_config to desktop_capture in GN build.
...
This allows us to remove some configs in the Chrome build that should come
automatically from depending on this target.
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14309004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7067 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 17:00:55 +00:00
pbos@webrtc.org
ceb956b29d
Abort Negotiate() if DoCreateOffer() fails.
...
Addressing crash in test.
R=jiayl@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/19239004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7066 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 15:27:49 +00:00
kjellander@webrtc.org
d57c95fde4
Change Chromium .gclient to not use Managed mode.
...
Since the sync_chromium.py script always passes --revision
to the gclient sync command, we don't need to have
managed=True in the .gclient file.
This will avoid a warning that confuses our developers.
BUG=3776
TESTED=Removed my chromium/.last_sync_chromium and performed
a gclient sync with this patch applied. No warning complaining
about Managed mode appears.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/23469004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7065 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 14:58:55 +00:00
andresp@webrtc.org
fa822b940f
Fix strange owners files with comments that crashs "git cl presubmit"
...
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14359004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7064 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 14:25:07 +00:00
kjellander@webrtc.org
79ee97cf43
[MIPS] Fix gn gen failure for MIPS in webrtc
...
Fixes the following failure for mips:
"ERROR at //third_party/webrtc/BUILD.gn:136:7: Undefined variable for +=.
cflags += [ "-mhard-float" ]
^-----
I don't have something with this name in scope now."
BUG=3441
TEST=In Chromium. Passing compile locally on Linux using:
gn gen out-gn/mips --args="is_debug=false os=\"android\" cpu_arch=\"mipsel\"" --verbose && ninja -C out-gn/mips all
gn gen out-gn/arm --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" --verbose && ninja -C out-gn/arm all
gn gen out-gn/x86-linux --args="is_debug=false os=\"linux\"" --verbose && ninja -C out-gn/x86-linux webrtc
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15349004
Patch from Gordana Cmiljanovic <Gordana.Cmiljanovic@imgtec.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7063 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 14:10:49 +00:00
houssainy@google.com
38ef664418
Moving the api.js and bot.js to /rtcbot/bot/ to be shared between
...
/borwser and /android
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7062 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-04 13:44:47 +00:00