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Author SHA1 Message Date
fischman@webrtc.org
e3a628997f Roll libvpx 264320:267596 
BUG=3038
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-02 00:04:20 +00:00
fischman@webrtc.org
14ea7e8922 AppRTCDemo(android): added a Heads-Up Display for bandwidth estimation. 
- tap display to toggle visibility
- increased getStats frequency to 1hz.

R=glaznev@google.com

Review URL: https://webrtc-codereview.appspot.com/19419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 20:57:55 +00:00
fischman@webrtc.org
dd92feb6dd AppRTCDemo(android): send the created SDP, not the local description after setting it 
This is required to allow explicit filtering of ICE candidates.

BUG=3288
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 19:06:18 +00:00
turaj@webrtc.org
560dce5d48 Pull openmax into third_party. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6037 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 18:53:51 +00:00
jiayl@webrtc.org
9c16c39e61 Sets the SCTP port codec in the native SessionDescription. 
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.

BUG=3141
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 18:30:30 +00:00
jiayl@webrtc.org
53d82350c5 Ignore identical remote fingerprint in DtlsTransportChannelWrapper::SetRemoteFingerprint. 
Trying to set the same remote fingerprint could happen during renegotiation and should not fail.

BUG=crbug/362431
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-01 00:00:19 +00:00
andrew@webrtc.org
e44a84d851 Only clamp to 16 kHz when AECM is enabled. 
Otherwise we could needlessly downsample to 16 kHz (rather than 32 kHz)
when HW AEC is used.

BUG=3259
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 18:58:23 +00:00
tkchin@webrtc.org
ff2733204d Implement ObjC DataChannel wrapper 
R=fischman@webrtc.org
BUG=3112

Review URL: https://webrtc-codereview.appspot.com/16369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 18:32:33 +00:00
andrew@webrtc.org
65f933899b Fix constness of AudioBuffer accessors. 
Don't return non-const pointers from const accessors and deal with the
spillover. Provide overloaded versions as needed.

Inspired by kwiberg:
https://webrtc-codereview.appspot.com/12379005/

R=bjornv@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 16:44:13 +00:00
buildbot@webrtc.org
740e6b339a (Auto)update libjingle 65843899-> 65880186 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 15:33:45 +00:00
fischman@webrtc.org
7c82adae61 AppRTCDemo was blocking the main thread for network requests. This fixes it by making the background queue serial instead of using @synchronize to make the background operations serial. 
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16379004

Patch from Bridger Maxwell <bridgeyman@gmail.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-30 00:17:47 +00:00
turaj@webrtc.org
9bd49becc1 Fix a data race in ACM1 when audio is pulled. 
BUG=chromium:348511
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 20:27:45 +00:00
fischman@webrtc.org
a86c42c424 libjingle_unittest now compiles and passes on iOS! (reland of r5986) 
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && \
  ~/src/ios-deploy/ios-deploy -d -u -v -b \
    ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app

Note that the test's use of signals means that lldb will break in the middle
of the suite.  To ignore these signals tell lldb:

pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue

BUG=3241
R=kjellander@webrtc.org, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 18:37:29 +00:00
buildbot@webrtc.org
681f787cc4 (Auto)update libjingle 65752960-> 65813736 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:55:26 +00:00
henrike@webrtc.org
f2aafe4355 Added include of assert.h for files calling assert but missing the include. 
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19409005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6022 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:54:17 +00:00
fischman@webrtc.org
f04a6ea733 MediaCodecVideoEncoder: limit MediaCodec bitrate to 95% of requested to avoid overshoot. 
BUG=3194
R=noahric@google.com

Review URL: https://webrtc-codereview.appspot.com/17379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6021 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:53:30 +00:00
henrike@webrtc.org
82d3cb68cd Made common_types.h PacketTime declaration match https://code.google.com/p/webrtc/source/browse/trunk/talk/base/asyncpacketsocket.h#65 
BUG=N/A
R=mallinath@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:50:47 +00:00
henrike@webrtc.org
ceffdbc371 Fixed r5373-related regressions in VideoFramesQueue::FrameToRecord() 
R=henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 17:11:07 +00:00
pbos@webrtc.org
0300939484 Disable failing GoogleWifiTrace3Mbps. 
Disables BweFeedbackTest.GoogleWifiTrace3Mbps instead of
BweSimulation.GoogleWifiTrace3Mbps.

TBR=stefan@webrtc.org
BUG=3277

Review URL: https://webrtc-codereview.appspot.com/20389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6017 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 15:25:59 +00:00
pbos@webrtc.org
9353e6bc55 Disable GoogleWifiTrace3Mbps. 
Breaks bots, according to stefan@ there's a missing file for this test
to run.

TBR=stefan@webrtc.org
BUG=3277

Review URL: https://webrtc-codereview.appspot.com/13409005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 14:49:56 +00:00
stefan@webrtc.org
dfe2a1c995 Adding BweFeedbackTest which tracks BWE performance over a set of simulated scenarios. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11019006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 14:21:42 +00:00
mflodman@webrtc.org
f223746521 Upping start bitrate to min, if set to a lower value i SetSendCodec. 
BUG=3276
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 12:38:42 +00:00
pbos@webrtc.org
f5433753b8 Suppress DrMemory allocator mismatch errors. 
Bot is flooded with these errors, suppressing them to get back to a
normal state, contacting DrMemory team.

R=kjellander@webrtc.org
BUG=3275
TEST=git try -b win_drmemory_light -r 6012

Review URL: https://webrtc-codereview.appspot.com/13419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 12:24:45 +00:00
kjellander@webrtc.org
97e67cb476 Fix iOS assembly compile error. 
In the roll of
https://webrtc-codereview.appspot.com/13369007
the fix in transform_neon.S was incorrectly removed
assuming it was only affecting Android when rolling to
265795. This CL fixes the iOS build when rolled to
266514.

Error looks like:
[893/2157] CC obj/webrtc/modules/audio_coding/codecs/isac/main/source/iSAC.entropy_coding.o
FAILED: /Volumes/data/b/build/goma/gomacc ../../third_party/llvm-build/Release+Asserts/bin/clang -MMD -MF obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o.d -DV8_DEPRECATION_WARNINGS -DBLINK_SCALE_FILTERS_AT_RECORD_TIME -DDISABLE_NACL -DCHROMIUM_BUILD -DUSE_LIBJPEG_TURBO=1 -DENABLE_CONFIGURATION_POLICY -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DENABLE_EGLIMAGE=1 -DCLD_VERSION=1 -DENABLE_SPELLCHECK=1 -DDISABLE_FTP_SUPPORT=1 -DWEBRTC_RESTRICT_LOGGING -DWEBRTC_MODULE_UTILITY_VIDEO -DWEBRTC_ARCH_ARM -DWEBRTC_ARCH_ARM_V7 -DWEBRTC_ARCH_ARM_NEON -DWEBRTC_POSIX -DWEBRTC_MAC -DWEBRTC_IOS -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -DNDEBUG -DNVALGRIND -DDYNAMIC_ANNOTATIONS_ENABLED=0 -DNS_BLOCK_ASSERTIONS=1 -D_FORTIFY_SOURCE=2 -I../.. -I../.. -I../../webrtc -I../../webrtc/common_audio/resampler/include -I../../webrtc/common_audio/signal_processing/include -I../../webrtc/common_audio/vad/include -isysroot /Applications/Xcode.app/Contents/Developer/Platforms/iPhoneOS.platform/Developer/SDKs/iPhoneOS7.1.sdk -Os -gdwarf-2 -fvisibility=hidden -Werror -Wnewline-eof -miphoneos-version-min=6.0 -arch armv7 -Wall -Wendif-labels -Wextra -Wno-unused-parameter -Wno-missing-field-initializers -Wheader-hygiene -Wno-c++11-narrowing -Wno-char-subscripts -Wno-unneeded-internal-declaration -Wno-covered-switch-default -Wstring-conversion -Wno-deprecated-register -Wno-absolute-value -Wno-selector-type-mismatch -std=c99 -fcolor-diagnostics  -c ../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S -o obj/webrtc/modules/audio_coding/codecs/isac/fix/source/isac_neon.transform_neon.o
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:45:11: error: immediate expression for mov requires :lower16: or :upper16
  mov r6, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
          ^
../../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_neon.S:458:11: error: immediate expression for mov requires :lower16: or :upper16
  mov r2, #(WebRtcIsacfix_kSinTab1 - WebRtcIsacfix_kCosTab1)
in
http://build.chromium.org/p/client.webrtc/builders/iOS%20Release/builds/911/steps/compile/logs/stdio

TBR=ajm
TEST=ios trybots passing tryjob based on r6010.
BUG=

Review URL: https://webrtc-codereview.appspot.com/12439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6012 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 10:25:30 +00:00
henrik.lundin@webrtc.org
060b84b3bb Remove neteq_unittests from Android builds 
BUG=2996
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6011 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 09:49:51 +00:00
kjellander@webrtc.org
59343ee3d8 Roll chromium_revision 260462:266514 
Unfortunately needs to introduce yet another workaround
script for the Visual Studio toolchain download.

This will resolve the failures with our Dr Memory Full bot
(see https://code.google.com/p/chromium/issues/detail?id=366637#c2
for details). Long term, I'm considering a better approach
than using the added gclient solution pointing at
svn://svn-mirror.golo.chromium.org/chrome/trunk/deps/third_party/drmemory/drmemory.DEPS
i.e. add an entry that we roll separately in our DEPS file
instead. However, the Dr Memory team assured that changes
in their reporting format like this are rare.

Thanks fischman@ for the video_render.gypi fix!
Thanks kma@ for the transform_neon.S fix even if it turned out
not to be needed right now (probably will come back).

BUG=chromium:366637
TEST=git try -t compile
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13369007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6010 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 09:36:40 +00:00
henrik.lundin@webrtc.org
acf15dc90f Remove Version method from ACM1 
BUG=2996
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6009 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 09:25:21 +00:00
henrik.lundin@webrtc.org
70e53fa34d Remove ACM1 and NetEq3 related targets from modules.gyp 
Make necessary changes to compile.

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 08:58:46 +00:00
henrik.lundin@webrtc.org
fdf2053787 Remove AudioCodingModuleFactory 
These were no longer used anywhere in the code.

BUG=2996
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 08:22:14 +00:00
henrik.lundin@webrtc.org
0bc9b5a5a7 Add clock to ACM config struct 
The purpose is to clean up the ACM interface a bit. This is a
follow-up of a comment in http://review.webrtc.org/13379004/.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 08:09:31 +00:00
bjornv@webrtc.org
059488f2ea AEC: Startup phase only runs if reported_delay_enabled 
TESTED=trybots, modules_unittests
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-29 08:09:26 +00:00
buildbot@webrtc.org
af6640fce7 (Auto)update libjingle 65729829-> 65752960 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 21:31:51 +00:00
fischman@webrtc.org
874920f22e Disable WebRtcSpl_ScaleAndAddVectorsWithRoundNeon due to crash. 
The C (unoptimized) version of this function takes less than 1% of NetEq4 and
has been crashing since the NetEq3->4 move.

BUG=3243
R=henrik.lundin@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6003 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 20:09:09 +00:00
fischman@webrtc.org
82a045aae0 APM: limit native sample rate to 16kHz on mobile. 
Required by AECM which assert-fails on higher sample rates.

BUG=3259
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 17:26:32 +00:00
fischman@webrtc.org
f27fdeb9c9 AppRTCDemo(android): don't initialize process-globals more than once. 
BUG=3257
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 16:32:38 +00:00
michaelbai@google.com
497ff21fad Using realpath instead of android_src in Android webview 
BUG=367235
R=andrew@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/20369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 15:46:46 +00:00
andrew@webrtc.org
1152fe2247 Only download the VS toolchain if DEPOT_TOOLS_WIN_TOOLCHAIN=1. 
BUG=b/13171670
R=fbarchard@google.com, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5999 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 15:46:32 +00:00
pbos@webrtc.org
de1429e9ad Add thread annotations to Call API. 
Also constified a lot of pointers and reordered members to make
protected members more grouped together.

R=kjellander@webrtc.org, stefan@webrtc.org
BUG=2770

Review URL: https://webrtc-codereview.appspot.com/15399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5998 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 13:00:21 +00:00
kjellander@webrtc.org
7d825e9b2c Revert "libjingle_unittest now compiles and passes on iOS!" 
This reverts commit r5986 as it fails compilation on Mac
(non-iOS). The failure was not discovered on the commitbots
since they don't clobber their builds.

BUG=3241
TBR=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 12:50:47 +00:00
pbos@webrtc.org
c891577e6d Disable flaky CaptureNtpTimeWithNetworkJitter. 
TBR=wu@webrtc.org
BUG=3271

Review URL: https://webrtc-codereview.appspot.com/19389005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 11:57:32 +00:00
bjornv@webrtc.org
494aa0e93d AEC: Moved delay buffer size enums from aec_core.h to aec_core_internal.h 
These enums are noly used internally in aec_core.c and it makes more sense to put them in aec_core_internal.h

TESTED=trybots
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19379005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5995 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 11:42:27 +00:00
pbos@webrtc.org
8dfe8ff590 Disable capture test for FrameRate on Windows. 
Flaky on Windows, has been for a while.

R=kjellander@webrtc.org
TBR=mflodman@webrtc.org
BUG=3270

Review URL: https://webrtc-codereview.appspot.com/19389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 11:27:36 +00:00
henrik.lundin@webrtc.org
e772c71743 Introduce a config struct for AudioCoding module 
The config struct currently contains the module ID, and the NetEq
config struct, but will be extended in the future. The purpose of this
change is to expose certain NetEq settings to the ACM interface.

BUG=3083
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 10:16:57 +00:00
pbos@webrtc.org
f043f79711 Disabling flaky CanReceiveFec. 
CanReceiveFec is flaky, likely to the test expecting the first FEC
packet to always be decoded and rendered.

R=stefan@webrtc.org
BUG=3269

Review URL: https://webrtc-codereview.appspot.com/19379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 09:00:50 +00:00
pbos@webrtc.org
69e9950469 Disable flaky RunsRtpRtcpTestWIthoutErrors. 
BUG=1790
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5991 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:49:07 +00:00
henrik.lundin@webrtc.org
12a34247a4 Fix the NetEq build 
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5990 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:36:35 +00:00
henrik.lundin@webrtc.org
116ed1d4f0 Include buffer size limits in NetEq config struct 
This change includes max_packets_in_buffer and max_delay_ms in the
NetEq config struct. The packet buffer is also no longer limited in
terms of payload sizes (bytes), only number of packets.

The old constants governing the packet buffer limits are deleted.

BUG=3083
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:20:04 +00:00
henrik.lundin@webrtc.org
b08bbf57a6 Add henrik.lundin as owner in AudioCoding module 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5988 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-28 08:15:35 +00:00
mallinath@webrtc.org
a0d3067575 Use CreatePeerConnection method which accepts port_allocator. 
Other method will be removed, in a different CL.

R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20369006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5987 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-26 00:00:15 +00:00
fischman@webrtc.org
95cd1551f8 libjingle_unittest now compiles and passes on iOS! 
Example run from cmd-line:
ninja -C out_ios/Debug-iphoneos libjingle_unittest && ~/src/ios-deploy/ios-deploy -d -u -v -b ~/src/wr/trunk/out_ios/Debug-iphoneos/libjingle_unittest.app
Note that the test's use of signals means that lldb will break in the middle of the suite.  To ignore these signals tell lldb:

pro hand -p true -s false -n false SIGINT
pro hand -p true -s false -n false SIGTERM
continue

BUG=3241
R=noahric@google.com, tkchin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-25 23:59:56 +00:00