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Author SHA1 Message Date
andrew@webrtc.org
a37bf2c4fe Hack clock_unittest fix for parallel execution. 
It's a bad idea to depend on timing constraints in unit tests, but
moving this from 5 -> 100 ms should allow it to fail only very rarely.

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7999 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 19:08:58 +00:00
aluebs@webrtc.org
e5a921a82d Use tmp files in file_utils_unittests 
The static file names were breaking when executing tests in parallel. This fixes it.

BUG=4138
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7997 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 18:45:22 +00:00
pbos@webrtc.org
76bc981b2d Use a temp file in FileLockTest. 
Permits running FileLockTests in parallel as the lock files don't
conflict with concurrent runs.

BUG=4137
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7996 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 17:56:33 +00:00
marpan@webrtc.org
c4ad157d8d Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9. 
BUG=4059

TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7994 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 17:31:34 +00:00
mflodman@webrtc.org
215bbbdcdd Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7993 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 14:56:02 +00:00
kjellander@webrtc.org
aeb0dd3079 Disable RelayServerTest.TestExpiration on Mac. 
The test is flaky on Mac.
BUG=4134
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7992 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-03 17:47:05 +00:00
bjornv@webrtc.org
bac0012120 Extend delay estimation window in AEC to 500 ms on all platforms 
On non-Android the delay estimator in audio_processing/aec has solely been used for logging purposes. The maximum possible observed delay has been 236 ms. We have seen longer delays for which the delay estimate at best ends up at 236 ms, but can also be 'random'. reported delays are clamped to 500 ms.
This cl extends the delay estimation window to match that.

BUG=4086, 3504, 4113
TESTED=locally on Linux and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7989 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-02 09:23:49 +00:00
bjornv@webrtc.org
3a70625caf audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877 
BUG=N/A
TESTED=Now it builds with aec_debug_dump=1 on Mac
TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-01 22:04:12 +00:00
andrew@webrtc.org
34ac956706 Do not use openmax_dl for MIPS64 platform. 
This fix is intended for MIPS64 Chromium Android builds, which has no openmax_dl
support at this moment.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31339004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7983 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-30 18:19:56 +00:00
pthatcher@webrtc.org
a9b1ec0247 Support for DTLS in OpenSSLAdapter 
1)  Added SetMode() to SSLAdapter and OpenSSLAdapter so the mode can be set to
     SSL_MODE_DTLS
 2)  OpenSSLAdapter overrides SendTo() and RecvFrom() to handle calls from
     TurnPort via AsyncUdpSocket
 3)  OpenSSLAdapter derives from MessageHandler to implement an internal DTLS
     timer
 4)  Updated SSLAdapter unit tests

BUG=
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7981 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-29 23:00:14 +00:00
jiayl@webrtc.org
c5fd66dcdf Accept incoming pings before remote answer is set to reduce connection latency. 
BUG=4068
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7980 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-29 19:23:37 +00:00
andrew@webrtc.org
84d84471f5 Minor fixes regarding accumulator usage on MIPS platforms. 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33729004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7979 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-29 17:08:44 +00:00
pthatcher@webrtc.org
5ad4178137 Move the Jingle-specific network code into webrtc/libjingle. 
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7977 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-23 22:14:15 +00:00
sprang@webrtc.org
46d4d29a75 Add field trial for screenshare bitrates when using temporal layers. 
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-23 15:19:35 +00:00
pbos@webrtc.org
53cb74107f Make RelayServerTest use VirtualSocketServer. 
Permits running the tests in parallel.

R=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w64 out/Debug/rtc_unittests --gtest_filter=RelayServerTest.*

Review URL: https://webrtc-codereview.appspot.com/38479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7974 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-22 07:56:42 +00:00
pthatcher@webrtc.org
4c0544ab07 Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. 
Also, fix the includes and header guards of examples/call.

R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7972 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 22:29:55 +00:00
tkchin@webrtc.org
ed1a48b0cd Fix mac video capture leak. 
BUG=3878
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7971 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 20:51:02 +00:00
aluebs@webrtc.org
ae643ce280 Wire up Beamformer in AudioProcessing 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7969 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 19:57:34 +00:00
stefan@webrtc.org
8817256373 Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator. 
BUG=chromium:444023
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7968 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 18:00:21 +00:00
stefan@webrtc.org
50f7db8a77 Remove unneccessary lock causing a potential deadlock. 
TBR=pbos@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7967 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 17:55:20 +00:00
pbos@webrtc.org
5570769210 Remove the last getters from VideoReceiveStream stats. 
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/32899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7965 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:45:03 +00:00
stefan@webrtc.org
742386a136 Enable payload-based padding by default and remove the API. 
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 15:33:17 +00:00
kwiberg@webrtc.org
aa21f2765b Unify the two copies of move.h 
This patch basically deletes webrtc/base/move.h (which is the more
outdated copy) and moves webrtc/system_wrappers/source/move.h to take
its place.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 14:35:57 +00:00
pbos@webrtc.org
d16e839c6d Rtp-Rtcp sender cleanup. 
Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions.

Also removed const on non-pointer/reference types for related files.

BUG=
R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34469004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 13:49:55 +00:00
kjellander@webrtc.org
556caffb36 GN: Fix build for Mac 
BUG=4105
R=henrika@webrtc.org, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7961 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 13:28:37 +00:00
stefan@webrtc.org
11d8176cb3 Move updating nack bitrate inside UpdateNACKBitRate. 
BUG=
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 09:52:24 +00:00
pthatcher@webrtc.org
5647877b2d Breakup Transports and TransportParsers and move TransportParsers into webrtc/libjingle. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-19 03:32:59 +00:00
aluebs@webrtc.org
0c39e91cc8 Merge beamformer 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7958 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 22:22:04 +00:00
andrew@webrtc.org
1090a6eccf Remove obsolete target_arch == armv7. 
Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.

BUG=3906
R=kjellander@webrtc.org, tkchin@webrtc.org, zhongwei.yao@arm.com

Review URL: https://webrtc-codereview.appspot.com/38379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 21:36:18 +00:00
pthatcher@webrtc.org
aacc23465b Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
(This is the 3rd try)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7956 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 20:31:29 +00:00
pthatcher@webrtc.org
f5847d7746 Move session/tunnel to webrtc/libjingle. This is part of the ongoing effort to move Jingle-specific things out of WebRTC and into its own repository. I won't submit this until all other projects have moved off of compiling this as well. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7953 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 17:09:11 +00:00
asapersson@webrtc.org
cb79141eab Store the received report blocks map (mapped per remote ssrc) in a map per source ssrc. 
When using rtx, receiver reports with two report blocks are received. The report blocks have the same remote ssrc and therefore the first report block was overwritten by the second report block when stored in the ReportBlockInfoMap.

Removed unused function ResetRTT.

BUG=4114
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33659005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7952 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 14:30:32 +00:00
pbos@webrtc.org
ce4e9a3562 Refactor some receive-side stats. 
Removes polling of CName as well as receive codec statistics in favor of
internal callbacks keeping a statistics struct up to date.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:50:16 +00:00
pbos@webrtc.org
98c04b38a8 Get avg_delay_ms from DecoderTiming callback. 
R=stefan@webrtc.org
BUG=1667

Review URL: https://webrtc-codereview.appspot.com/28339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7949 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 13:12:52 +00:00
sprang@webrtc.org
9b79197c80 Suppress REMB in bitrate ctrl if it seems lika a short network glitch. 
BUG=4082
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7948 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 11:53:59 +00:00
pbos@webrtc.org
f832a6d090 Remove _t from function pointer typedefs. 
_t are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7947 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:56:09 +00:00
henrik.lundin@webrtc.org
eed7a22bbf Make an AudioEncoder subclass for iSAC redundant encoding 
Adding unit test, too.

BUG=3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36559005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7946 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:52:36 +00:00
pbos@webrtc.org
dd8f6f3d48 Rename rtpDumpPktHdr_t to RtpDumpPacketHeader. 
_t names are reserved in POSIX.

BUG=162
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7945 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:18:42 +00:00
pbos@webrtc.org
e468bc9e60 Rename _t struct types in audio_processing. 
_t names are reserved in POSIX.

R=bjornv@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7943 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 09:11:33 +00:00
henrik.lundin@webrtc.org
cab1291745 Fixing the memory leak in AudioEncoderCopyRedDeathTest.NullSpeechEncoder 
Re-enable the test and explicitly call delete on red, even though the
test should die in the AudioEncoderCopyRed cunstructor. Apparently,
things work a little differently under memcheck.

BUG=4108, 3926
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7942 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 06:58:42 +00:00
guoweis@webrtc.org
4fba293c87 Workaround for issue 3927 to allow localhost IP even if it doesn't match the local turn port 
BUG=3927
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7941 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 04:45:05 +00:00
pthatcher@webrtc.org
4cb3856a4d Revert "Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository." 
This reverts r7939 because it broke Chromium and other depedent projects that rely on certain logic remaining in p2p/base/session.cc and not in webrtc/libjingle/session.cc.

BUG=
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7940 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 02:28:25 +00:00
pthatcher@webrtc.org
536f999e58 Split up (Jingle)Session from BaseSession. This is part of an ongoing effort to move Jingle-specific code out of WebRTC and into its own repository. 
This is an un-revert of r7992 and r7993.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7939 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 01:22:02 +00:00
guoweis@webrtc.org
c51fb9348d Fix an assert failure caused by race condition 
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7938 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-18 00:30:55 +00:00
andrew@webrtc.org
0ab42bc3f6 Make safe_conversions suitable for rtc_base_approved. 
Since we want to use checked_cast in WavReader.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7937 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 22:56:09 +00:00
pthatcher@webrtc.org
bc03192560 Move jingle examples from talk/ into webrtc/libjingle. This is part of the effor to move Jingle out of WebRTC and into its own repository. 
R=juberti@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7936 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 22:15:11 +00:00
guoweis@webrtc.org
0eb6eec5cb Move VirtualSocket into the .h file to allow unit tests more control over behavior. 
BUG=3927
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7935 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 22:03:33 +00:00
aluebs@webrtc.org
6f10ae25ea Support block_size greater than chunk_size in Blocker 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7934 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 17:28:31 +00:00
pbos@webrtc.org
eb544460e4 Rename _t struct types in audio_coding. 
_t names are reserved in POSIX.

R=henrik.lundin@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/34509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7933 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 15:23:29 +00:00
pbos@webrtc.org
e728ee03ba Remove or rename typedefs with _t prefixes. 
_t prefixes are reserved for additional typenames in POSIX.

R=henrik.lundin@webrtc.org, hta@webrtc.org, stefan@webrtc.org
BUG=162

Review URL: https://webrtc-codereview.appspot.com/36559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7931 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-12-17 13:43:55 +00:00