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Author SHA1 Message Date
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
andresp@webrtc.org
262e676a08 Reland rev 7041 with BUILD.gn files. 
Original description:
  Audio codecs to include webrtc/typedefs.h

  Will easy merge of webrtc/typedefs.h and webrtc/base/basictypes.h

  CL Generated with:
  $ git grep -l \"typedefs.h\" | xargs sed -i "s/typedefs.h/webrtc\/typedefs.h/g"

BUG=3777
R=henrik.lundin@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:28:48 +00:00
bjornv@webrtc.org
3cbd6c26c8 Fix MSVC warnings about value truncations, webrtc/common_audio/ edition. 
This changes some method signatures to better reflect how callers are actually
using them.  This also has the tendency to make signatures more consistent about
e.g. using int (instead of int16_t) for lengths of things like vectors, and
using int16_t (instead of int) for e.g. counts of bits in a value.

This also removes a couple of functions that were only called in unittests.

BUG=3353,chromium:81439
TEST=none
R=andrew@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7060 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 13:21:44 +00:00
henrik.lundin@webrtc.org
f6ab6f86e7 Rename Audio[Multi]Vector.CopyFrom to .CopyTo 
The name of the copy method was confusing. This change makes the
code easier to read where the method is used.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 10:58:43 +00:00
kjellander@webrtc.org
3c0aae17f0 Change gflags and gmock includes to be full paths. 
This will fix PRESUBMIT warnings developers will get due to
r7014 and r7020.

Also some minor style cleanup in:
webrtc/modules/audio_coding/main/test/RTPFile.cc
webrtc/modules/audio_coding/neteq/test/RTPjitter.cc

BUG=
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 09:55:40 +00:00
kwiberg@webrtc.org
51bb33cc18 ACMOpus: Remove useless member variable fec_enabled_ 
R=henrik.lundin@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7057 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 08:42:44 +00:00
henrik.lundin@webrtc.org
7825b1abf9 Add support for multi-channel DTMF tone generation 
This CL opens up support for DTMF tones to be played to multi-channel
outputs. The same tones are replicated across all channels. Unit tests
are updated.

Also adding a new method AudioMultiVector::CopyChannel.

BUG=crbug/407114
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:39:21 +00:00
pbos@webrtc.org
bcb6bcfe6c Remove HybridVideoEngine. 
This is currently unused dead code.

R=pthatcher@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/24409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:32:26 +00:00
asapersson@webrtc.org
9d453931c5 Change return value for number of discarded packets to be int. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7054 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 07:07:44 +00:00
stefan@webrtc.org
01581da711 Fix audio/video sync when FEC is enabled. 
Also improves the tests by adding a test case for FEC, and running the a/v sync
tests with NACK and simulated packet loss.

BUG=crbug/374104
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 06:48:14 +00:00
andresp@webrtc.org
bfd7a8c448 Fix compile errors on webrtc/base. 
R=fbarchard@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 04:59:52 +00:00
andresp@webrtc.org
0229cbae33 Remove ambiguous call to MakeCheckOpString. 
BUG=3777
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7051 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-04 04:53:29 +00:00
thorcarpenter@google.com
95c2458766 * Move test data assests required by video frame tests to be in libjingle instead of elsewhere and co-located with other libjingle test data files. 
"gcl try" fails to upload these large files so adding them independently.

R=andrew@webrtc.org, harryjin@google.com, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 23:17:36 +00:00
fbarchard@google.com
9328f39a3e cast return values in uint16_t RTPFile::Read() to uint16_t to avoid compile error 
BUG=3663
TESTED=ninja local build on windows.
R=andrew@webrtc.org, kwiberg@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/16229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-03 23:05:07 +00:00