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
braveyao@webrtc.org
7cb4752184
WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process.
...
This cl is to teach videocapture android how to deinitialize and allow it to be re-initializable.
BUG=3284
TEST=ManualTest
R=fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15429004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6167 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 03:18:15 +00:00
wu@webrtc.org
54231f0662
Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log.
...
BUG=crbug/371714
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6166 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 23:06:23 +00:00
mallinath@webrtc.org
bb6201ae4b
TCP remote socket address should have both server hostname and IP address.
...
Hostname is necessary when we are creating TLS based socket, for certificate
verification.
BUG=https://code.google.com/p/chromium/issues/detail?id=306285
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6165 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 22:43:05 +00:00
fischman@webrtc.org
a150bc9bbf
PeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine.
...
Enables applications that don't want to pay the init/startup cost or request
extra permissions (e.g. audio-only app, or DataChannel-only app).
BUG=3234
Review URL: https://webrtc-codereview.appspot.com/15489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6164 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 22:00:50 +00:00
buildbot@webrtc.org
ef5a752c29
(Auto)update libjingle 67043374-> 67044055
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6163 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 21:35:19 +00:00
buildbot@webrtc.org
3e924683d4
(Auto)update libjingle 67037200-> 67043374
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6162 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 21:29:04 +00:00
jiayl@webrtc.org
4f5801494d
Drop the DataChannel message if it's received when the channel is not open.
...
It may happen when the JS has closed the channel on the signaling thread while messages are received on the worker thread and posted before the state change is pushed to the worker thread.
BUG=crbug/363005
R=mallinath@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19469005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6161 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 20:32:35 +00:00
buildbot@webrtc.org
372701a872
(Auto)update libjingle 67023528-> 67036361
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6160 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 20:27:59 +00:00
andrew@webrtc.org
21299d4e00
Remove the use of AudioFrame::energy_ from AudioProcessing and VoE.
...
We want to remove energy_ entirely as we've seen that carrying around
this potentially invalid value is dangerous.
Results in the removal of AudioBuffer::is_muted(). This wasn't used in
practice any longer, after the level calculation moved directly to
channel.cc
Instead, now use ProcessMuted() in channel.cc, to shortcut the level
computation when the signal is muted.
BUG=3315
TESTED=Muting the channel in voe_cmd_test results in rms=127.
R=bjornv@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6159 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 19:00:59 +00:00
buildbot@webrtc.org
688ed699e0
(Auto)update libjingle 67017551-> 67023528
...
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6158 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 18:26:09 +00:00
henrike@webrtc.org
c50bf7cbd0
Added namespace rtc to some base classes and functions. It was causing linker error in the FYI bots: http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Android%20Builder%20%28dbg%29/builds/1808/steps/compile/logs/stdio but also, not doing it pollutes the global namespace.
...
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6157 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 18:24:13 +00:00
kjellander@webrtc.org
3147b97f8c
LSan suppressions for libjingle tests (fix)
...
Apparently wildcards are needed for templates
in the call stack for matching suppressions.
TBR=wu@webrtc.org
BUG=2528, 3345
TEST=None, since I cannot reproduce the leaks on the bot
in my local environment (and we don't have a trybot for this yet).
Review URL: https://webrtc-codereview.appspot.com/18459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6156 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 18:05:15 +00:00
kjellander@webrtc.org
7c0f6e1b47
LSan suppressions for libjingle tests (more)
...
Additional suppressions after a new test run was completed
after https://webrtc-codereview.appspot.com/18429006 was
submitted.
TBR=wu@webrtc.org
BUG=2528, 3345
TEST=None, since I cannot reproduce the leaks on the bot
in my local environment (and we don't have a trybot for this yet).
Review URL: https://webrtc-codereview.appspot.com/14509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6155 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 17:46:23 +00:00
fischman@webrtc.org
2c98af7935
PeerConnection(Java): auto-WrapCurrentThread() when creating PeerConnectionFactory.
...
Various pieces of talk/ assume that the current Thread is ThreadManager'd
without checking this, so unconditionally wrap the caller's thread in case it
was created by Java code unbeknownst to ThreadManager.
BUG=2947
R=wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6154 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 17:33:32 +00:00
kjellander@webrtc.org
a70dff4427
LSan suppressions for libjingle tests.
...
Adding a few more suppressions to make the tests pass
LSan as it's about to be enabled in the main waterfall
at the ASan bot.
TBR=wu@webrtc.org
BUG=2528, 3345
TEST=None, since I cannot reproduce the leaks on the bot
in my local environment (and we don't have a trybot for this yet).
Review URL: https://webrtc-codereview.appspot.com/18429006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6153 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 17:12:01 +00:00
wu@webrtc.org
88abf11cad
Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe.
...
BUG=3111
TEST=try bots
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6152 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 16:53:51 +00:00
pbos@webrtc.org
4e545cc244
Update webrtcvideoengine2.cc to use DeliveryStatus.
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talk/ changes corresponding to https://review.webrtc.org/12289005/ .
BUG=3228
R=perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6151 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 13:58:13 +00:00
pbos@webrtc.org
caba2d2a37
Add DeliveryStatus enum to DeliverPacket().
...
Allows signalling why packet delivery failed. Especially enables
signaling that delivery fails because the incoming packet had an unknown
SSRC. This allows an application to react and create receivers for the
new streams.
R=mflodman@webrtc.org
BUG=3228
Review URL: https://webrtc-codereview.appspot.com/12289005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6150 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 13:57:12 +00:00
andresp@webrtc.org
581e2172af
Fix libjingle to provide a field_trial implementation.
...
This completes https://webrtc-codereview.appspot.com/14489004/ by updating libjingle rules.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6149 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 13:12:45 +00:00
kjellander@webrtc.org
01edf2e2af
Updating LSan third party suppressions.
...
Several new third party suppressions have been updated in
Chromium's suppressions file:
https://code.google.com/p/chromium/codesearch#chromium/src/tools/lsan/suppressions.txt
These will solve some of the errors we're seeing at
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20LSan%20%28and%20ASan%29/
which needs to be resolved before switching our
ASan bot to recipes (since the recipe ASan configuration
has LSan enabled by default).
BUG=2527,2528,3345,3346
TEST=Successfully ran the following tests under ASan+LSan locally:
libjingle_media_unittest
libjingle_p2p_unittest
libjingle_peerconnection_unittest
libjingle_unittest
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6148 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:28:38 +00:00
andresp@webrtc.org
a36ad6929d
Add webrtc field trials API.
...
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.
Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.
Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.
BUG=crbug/367114
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00