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Author SHA1 Message Date
henrike@webrtc.org
99b4162ccf Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base, apply diff manually) 
BUG=3379
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6217 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 20:42:17 +00:00
buildbot@webrtc.org
f9f1bfbdae (Auto)update libjingle 67686255-> 67689476 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 17:02:15 +00:00
henrike@webrtc.org
a148704b4b Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. 
BUG=3380
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:52:14 +00:00
buildbot@webrtc.org
ce4201df52 (Auto)update libjingle 67643194-> 67686255 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:22:51 +00:00
jiayl@webrtc.org
7ca277b574 Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. 
BUG=https://code.google.com/p/webrtc/issues/detail?id=3196
R=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/21529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:02:31 +00:00
henrike@webrtc.org
000658a138 Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot. 
BUG=N/A
TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21519006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 16:01:13 +00:00
mcasas@webrtc.org
3b7e282caa Disabling systematically failing 
WebRtcVideoMediaChannelTest.SendVp8HdAndReceiveAdaptedVp8Vga

TBR= pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 14:25:20 +00:00
mcasas@webrtc.org
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
mcasas@webrtc.org
c2213b6a0f Revert 6208 "Patch from henrike@webrtc.org" 
Wasn't enough. I'll have to revert the whole rev 6202.

> Patch from henrike@webrtc.org
> https://code.google.com/p/webrtc/source/detail?r=6202
> didn't work for at least one file and broke most of 
> the compile steps in the FYI bots. The file is reverted
> here.
> 
> TBR= henrike@webrtc.org, sergeyu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/17609004

TBR=mcasas@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 10:03:09 +00:00
mcasas@webrtc.org
86df8acc92 Patch from henrike@webrtc.org 
https://code.google.com/p/webrtc/source/detail?r=6202
didn't work for at least one file and broke most of 
the compile steps in the FYI bots. The file is reverted
here.

TBR= henrike@webrtc.org, sergeyu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 08:40:56 +00:00
braveyao@webrtc.org
1a79bb8d30 WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker 
BUG=3366
TEST=Manual Test
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 03:37:45 +00:00
buildbot@webrtc.org
49a6a27bf0 (Auto)update libjingle 67555838-> 67643194 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-21 00:24:54 +00:00
wu@webrtc.org
82c4b8531c Calculate capture ntp timestamp in local timebase for decoded audio frame. 
BUG=3111
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6205 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 22:55:01 +00:00
henrik.lundin@webrtc.org
48438c2c90 Enabling NetEq bit-exactness test for Win x64 
A new reference file (neteq4_universal_ref_win_64.pcm) was generated and
uploaded.

Also removing the old hack to have different reference files
for different version of Visual Studio. The test is now only supporting
VS 2012 and later (_MSC_VER >= 1700). This makes the windows 32-bit
output identical to the generic reference file
(neteq4_universal_ref.pcm), so the specialized one
(neteq4_universal_ref_win_32.pcm) could have been removed. However,
since the resources sync mechanism does not include removing of old
files, a client could pick up the old reference and fail. Therefore,
this cl also updates neteq4_universal_ref_win_32.pcm to be identical to
neteq4_universal_ref.pcm.

BUG=1458
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 16:07:43 +00:00
henrik.lundin@webrtc.org
aed31fe8ab Modifying WATCHLISTS 
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 16:05:47 +00:00
henrike@webrtc.org
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
stefan@webrtc.org
4059c2f579 Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. 
TBR=wu@webrtc.org
BUG=3374

Review URL: https://webrtc-codereview.appspot.com/14579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 15:12:29 +00:00
stefan@webrtc.org
70bb2d5755 Revert r6198 "Expose the original packet length in in the RTP play tools." 
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:25:48 +00:00
stefan@webrtc.org
83599cba77 Reenable WebRtcVideoEngineTestFake.SendReceiveBitratesStats under DrMemory. 
The uninitialized read has been fixed. Suppressing CL: https://code.google.com/p/webrtc/source/detail?r=6073

BUG=11288120
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:16:35 +00:00
stefan@webrtc.org
e208458643 Expose the original packet length in in the RTP play tools. 
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6198 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 13:09:16 +00:00
stefan@webrtc.org
be4ab99a53 Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8. 
BUG=3370
R=bjornv@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6197 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 12:42:01 +00:00
henrik.lundin@webrtc.org
a36db970bd Suppress GMOCK printouts from TestVideoSenderWithVp8 
Adding a missing EXPECT_CALL.

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20529005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6196 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 11:16:10 +00:00
bjornv@webrtc.org
f3e1341da7 VoEVolumeTest: Enabled Linux flaky tests 
Fixed error checks only on Linux to be able to turn on flaky tests. The cause of flaky failures is due to late values in pulse audio.

Related (deleted) CLs:
https://webrtc-codereview.appspot.com/19469007/
https://webrtc-codereview.appspot.com/19469004/

BUG=367
TESTED=trybots, voe_auto_test repeated
R=henrikg@webrtc.org, tina.legrand@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 10:43:42 +00:00
asapersson@webrtc.org
a826006132 Add NACK and RPSI packet types to RTCP packet builder. 
Fixes bug found when parsing received RPSI packet.

BUG=2450
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 09:53:51 +00:00
minyue@webrtc.org
2db9f45038 Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size 
BUG=webrtc:2925

TEST=passed_all_trybots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-20 08:33:30 +00:00
tkchin@webrtc.org
1732a591e7 Add a UIView for rendering a video track. 
RTCEAGLVideoView provides functionality to render a supplied RTCVideoTrack using OpenGLES2.

R=fischman@webrtc.org
BUG=3188

Review URL: https://webrtc-codereview.appspot.com/12489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 23:26:01 +00:00
tkchin@webrtc.org
7ca1edb31d Remove IOKit linkage from iOS builds. 
IOKit has been removed in iOS7, so link fails. iOS build succeeds after removing this setting and the corresponding one in talk/libjingle.gyp. Presubmit script tells me that CLs aren't allowed to touch both talk/ and webrtc/ at the same time so doing this separately.

BUG=
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 21:05:10 +00:00
fischman@webrtc.org
40bc7779aa talk_base: remove lock inversion between MessageQueue and MessageQueueManager. 
Removes the concept of a MessageQueue being "active" in favor of considering all
live MQ's to be active.
(previously a MQ was active starting from the first Post to it and stopped being
active in its dtor).

BUG=3230
R=sriniv@google.com

Review URL: https://webrtc-codereview.appspot.com/21489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 17:58:04 +00:00
wu@webrtc.org
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
pbos@webrtc.org
ebb467fdc8 Avoid NACK-list flush error on keyframe packets. 
Receiver code used to indicate a flush error even if the incoming packet
is a keyframe, forcing a request of a keyframe. Now it takes this
keyframe into account and doesn't error as the stream is decodable from
this point.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15549005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 15:28:02 +00:00
stefan@webrtc.org
64339a7069 Don't crash if a frame returned from the decoder is too old. 
BUG=crbug/371805
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 13:31:35 +00:00
michaelbai@google.com
725e582461 Use the new gyp_var_prefix local variable set by gyp instead of the 
global GYP_VAR_PREFIX set by the makefiles, since the latter is not
guaranteed to still be the same value at the time the command is
executed. Also, use abspath instead of realpath to convert paths to
absolute, since realpath expands to the empty string if the target file
doesn't exist, complicating build debugging.

BUG=
R=andrew@webrtc.org, torne@chromium.org

Review URL: https://webrtc-codereview.appspot.com/12559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 17:56:10 +00:00
henrike@webrtc.org
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
bjornv@webrtc.org
a3b5673879 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 
This macro was only used on two lines in iSACfix and I replaced those with the operations the macro performed.

BUG=3348
TESTED=trybots, manual unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 12:11:20 +00:00
pbos@webrtc.org
1e019d10b8 Fix delivery error-checking missed in r6151. 
Gets rid of quite a bit of false-warning logging in WebRtcVideoEngine2.

BUG=3228
R=perkj@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6183 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 11:38:45 +00:00
solenberg@webrtc.org
57e060251a Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. 
Flakiness was caused by a race condition between two atomic integers shared by two threads. Fixed by counting bad packets (those not containing the expected extension) instead of the good packets.

The CL also eliminates another possible flake by introducing a test fixture which doesn't automatically start sending audio packets when constructed.

BUG=3340,3356
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 11:27:09 +00:00
andresp@webrtc.org
60015d27ae Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} 
This allows use of webrtc field trials  and opens up the possibility to try the different code paths when running the unit tests by wiring them up to a --force_fieldtrials.

Tested: running a test target that links with the above with a flag --force_fieldtrials=invalid leads the test to crash.

BUG=crbug/367114
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 09:39:51 +00:00
bjornv@webrtc.org
1b21a57902 common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 
Macro was only mapping a function used in one place.

BUG=3348
TESTED=trybots, unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6180 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 06:40:31 +00:00
bjornv@webrtc.org
d83d607271 common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED 
* Moved the macro to randomization_functions and made it static const.
* Made WebRtc_IncreaseSeed() static, since it is not used outside this function.
* Style guide changes.

BUG=3348,3353
TESTED=trybots, common_audio_unittests, modules_unittests, modules_tests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-16 06:38:47 +00:00
wu@webrtc.org
75718cf80a * Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly. 
* Updated to ClockTest.NtpTime to verify the returned NTP is at least larger than kNtpJan1970.

Current implementation uses timeGetTime, which returns the time since windows is started, which can't be converted to NTP time.

BUG=3325
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 23:54:14 +00:00
henrike@webrtc.org
bf58a75dd9 removed webrtc_base_tests_utils from merge libs as it was breaking some builds. 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 21:45:09 +00:00
henrike@webrtc.org
508795f088 Made the presubmit script accept license headers back to 2003 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 18:21:17 +00:00
henrike@webrtc.org
cfdf420e21 Rebase webrtc/base 6129:6163 (svn diff -r 6129:6163 http://webrtc.googlecode.com/svn/trunk/talk/base apply diff manually) 
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 16:33:04 +00:00
buildbot@webrtc.org
6bfd6196ff (Auto)update libjingle 67052073-> 67134648 
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6174 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 16:15:59 +00:00
pbos@webrtc.org
6aeeac95ca Fix Windows debug compile of overrides/ logging. 
Compile error detected when trying to roll to chromium. Adding a cast
of base::PlatformThread::CurrentId() to base::subtle::Atomic32 to match
types in DCHECK_EQ().

See logging.cc error in:
http://build.chromium.org/p/tryserver.chromium/builders/win_chromium_compile_dbg/builds/19944/steps/compile%20%28with%20patch%29/logs/stdio

R=mflodman@webrtc.org, perkj@webrtc.org
TBR=henrike@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/17529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 13:56:56 +00:00
mflodman@webrtc.org
d5da25063c Revert "Revert "Audio processing: Feed each processing step its choice 
of int or float data"

This reverts commit 6142.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 11:17:21 +00:00
pbos@webrtc.org
024e4d5c6e Fix Win VideoSendStream::...::ToString() compiles. 
Removed an extra ::VideoSendStream in the method declarations.

BUG=3171
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 10:03:24 +00:00
pbos@webrtc.org
1e92b0a93d Add ToString() to VideoSendStream::Config. 
Adds ToString() to subsequent parts as well as a common.gyp to define
ToString() methods for config.h. VideoStream is also moved to config.h.

BUG=3171
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 09:35:06 +00:00
bjornv@webrtc.org
1aae6bf735 common_audio: Removes unused macros 
* WEBRTC_SPL_MUL_32_32_RSFT32BI
* WEBRTC_SPL_IS_NEG

BUG=3348
TESTED=trybots, common_audio_unittests
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 07:22:53 +00:00
henrik.lundin@webrtc.org
b4e80e095f Re-enable almost all NetEqDecodingTests for Android 
All but three tests in NetEqDecodingTest could be re-enabled without
any changes. Also making sure that the TestNetworkStatistics test exits
on first diff. (Otherwise, the log output gets flooded with error
messages.)

The tests that are still disabled are:
NetEqDecodingTest.TestBitExactness
NetEqDecodingTest.TestNetworkStatistics
NetEqDecodingTest.DecoderError

BUG=3343
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-15 07:14:00 +00:00