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3163 Commits

Author SHA1 Message Date
guoweis@webrtc.org
19e4e8d751 Add support for trying alternate server (STUN 300 error message) on TCP 
BUG=3774
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-10 02:41:32 +00:00
pthatcher@webrtc.org
0ba1533fdb Added support for an Origin header in STUN messages. 
For WebRTC there are instances where it may be desirable to provide
information to the STUN/TURN server about the website that initiated
a peer connection. This modification allows an origin string to be
included in the MediaConstraints object provided by the browser, which
is then passed as a STUN header in communications with the server.
A separate change will be submitted to the Chromium project that
uses and is dependent on this change, implementing IETF draft
http://tools.ietf.org/html/draft-johnston-tram-stun-origin-02

Originally a patch from skobalt@gmail.com.

(https://webrtc-codereview.appspot.com/12839005/edit)

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8035 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-10 00:47:02 +00:00
aluebs@webrtc.org
2693a54614 Add WEBRTC_BEAMFORMER define to BUILD.gn 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8034 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 23:26:13 +00:00
andrew@webrtc.org
8f27fcce79 Revert 8028 "Support associated payload type when registering Rt..." 
Reasons for revert:
1. glaznev discovered potentially related problems using the Android AppRTCDemo.
2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky.

> Support associated payload type when registering Rtx payload type.
> 
> Major changes include,
> - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
> - Receiver: Restore RTP packets by the new RTX-APT map.
> - Sender: Send RTP packets by checking RTX-APT map.
> - Add RTX payload type for RED in the default codec list.
> 
> BUG=4024
> R=pbos@webrtc.org, stefan@webrtc.org
> TBR=mflodman@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/26259004
> 
> Patch from Changbin Shao <changbin.shao@intel.com>.

TBR=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 20:22:46 +00:00
pthatcher@webrtc.org
9657265f39 Revert "Accept incoming pings before remote answer is set to reduce connection latency." 
This reverts r7980.

It was causing the ICE connected state to happen while still in the new state rather than going through the checking state, which was causing an ASSERT to fire, which was causing a crash.

Review URL: https://webrtc-codereview.appspot.com/41429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 19:08:27 +00:00
andrew@webrtc.org
f3fd8e7cdf Add NEON intrinsics version for transform_neon 
WebRtcIsacfix_Time2SpecNeon and WebRtcIsacfix_Spec2TimeNeon are added.
TransformTest in modules_unittests is passed on ARM32/ARM64 platform.

Initially reviewed here:
https://webrtc-codereview.appspot.com/36449004/

BUG=4002
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I0920ff66a0a0f529707fd7e6619f91e271a47019

Review URL: https://webrtc-codereview.appspot.com/31309004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 18:29:37 +00:00
pbos@webrtc.org
2a169640a3 Support associated payload type when registering Rtx payload type. 
Major changes include,
- Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType.
- Receiver: Restore RTP packets by the new RTX-APT map.
- Sender: Send RTP packets by checking RTX-APT map.
- Add RTX payload type for RED in the default codec list.

BUG=4024
R=pbos@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26259004

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

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-09 15:16:10 +00:00
kjellander@webrtc.org
8649fed1b8 GN: Fix Windows build. 
This required a tiny include fix in
src/third_party/winsdk_samples/src
which was committed in
https://code.google.com/p/webrtc/source/detail?r=7951

This incorporates contribution from vchigrin@yandex-team.ru
in https://webrtc-codereview.appspot.com/29299004/

BUG=261,1348,4105
R=pbos@webrtc.org
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 21:22:01 +00:00
bjornv@webrtc.org
758d6d431e audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 17:52:56 +00:00
bjornv@webrtc.org
dec649cbab audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with * 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 17:34:33 +00:00
bjornv@webrtc.org
5e5b32706a audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc 
The macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definition on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_armv7.h and common_audio/signal_processing/include/spl_inl_mips.h)

The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)

BUG=3348, 3353
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 17:25:34 +00:00
pbos@webrtc.org
124b9c70f9 Suppress races in event tracing code. 
Due to lack of atomics our tracing code is broken and triggering real
errors in ThreadSanitizer.

R=kjellander@webrtc.org
BUG=2497
TEST=out-tsan/out/Debug/libjingle_media_unittest --gtest_filter=WebRtcVideoMediaChannelTest.GetStatsMultipleRecvStreams + verifying that "race:*trace_event_unique_catstatic*" exists in the list of matched suppressions.

Review URL: https://webrtc-codereview.appspot.com/35719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8022 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 12:38:42 +00:00
asapersson@webrtc.org
823c9b8e36 Add histograms stats for sent/received fraction loss for a stream: 
- "WebRTC.Video.SentPacketsLostInPercent"
- "WebRTC.Video.ReceivedPacketsLostInPercent"

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/37419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-08 07:50:56 +00:00
andrew@webrtc.org
d730b288c8 Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon 
This function isn't used anymore. The file and header are also removed.

BUG=4002,3273
R=andrew@webrtc.org

Change-Id: I4b65dec57e6adc2ac2253031501f3b6de6937fac

Review URL: https://webrtc-codereview.appspot.com/35519004

Patch from Yang Zhang <yang.zhang@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8019 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 21:34:23 +00:00
pbos@webrtc.org
3663fb08ff Reenable dlclose() for InternalUnloadDll on TSan. 
Upstream TSan bug has been fixed and dlclose() no longer needs to be
excluded.

R=henrika@webrtc.org
BUG=3895

Review URL: https://webrtc-codereview.appspot.com/30099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 18:02:39 +00:00
pthatcher@webrtc.org
69472e711c Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS. 
Also, update SslSocketFactory to fail if StartSSL fails.

R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 18:01:07 +00:00
henrike@webrtc.org
c10eceab6e Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const. 
The BIO_METHODs ought to be const so they can go into rodata; BoringSSL makes
BIO_new take a const BIO_METHOD *, so there's no need for it to be non-const.
Also set SRTP_PROTECTION_PROFILE as const so we can constify those within
BoringSSL (https://boringssl-review.googlesource.com/#/c/2720/)

BUG=none
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 17:59:28 +00:00
pthatcher@webrtc.org
dfef02824c Ignore virtual box interfaces. 
BUG=3918
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8012 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-07 17:20:52 +00:00
andrew@webrtc.org
4796cb93dc Disable flaky RelayServerTest.TestExpiration on all platforms. 
BUG=4134
TBR=pthatcher

Review URL: https://webrtc-codereview.appspot.com/37529004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8001 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 23:56:19 +00:00
aluebs@webrtc.org
fb7a039e9d Use array geometry in Beamformer 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2015-01-05 21:58:58 +00:00
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