d6e84d9d13
Always copy processed audio to output buffer in ProcessStream.
...
In the old AudioFrame ProcessStream API, input and output buffers were shared.
Now that the buffers are distinct, the input must be copied to the
output even when no processing occurred.
R=andrew@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=78de5010d167d1e375e05d26177aad43c2e2de08
Review URL: https://webrtc-codereview.appspot.com/41459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8052 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-14 01:33:54 +00:00
c0da63c707
Optimize minimum delay in blocker
...
Could not hear any difference when running the beamformer_test, although sample-wise it changes because of the non-linear character of the processing.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35679004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8051 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 22:28:35 +00:00
af9d56f38c
Unify the two copies of template_util.h
...
This patch basically deletes webrtc/base/template_util.h (which is the
more outdated copy, although there are only cosmetical differences)
and moves webrtc/system_wrappers/source/template_util.h to take its
place.
The reunified header uses the rtc namespace like the old
webrtc/base/template_util.h, rather than the webrtc namespace like
webrtc/system_wrappers/source/template_util.h.
R=aluebs@webrtc.org , andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38579004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8050 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 20:32:04 +00:00
0b0c24177b
Only return Rtx mode in RTXSendStatus().
...
There is no need to return 'ssrc' and 'payloadtype' inside this function
since they are never used.
BUG=
R=pbos@webrtc.org , stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38569004
Patch from Changbin Shao <changbin.shao@intel.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 14:15:15 +00:00
3df38b442f
Unify the two copies of compile_assert.h
...
This patch basically deletes webrtc/base/compile_assert.h (which is
the more outdated copy) and moves
webrtc/system_wrappers/source/compile_assert.h to take its place.
R=aluebs@webrtc.org , andrew@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8048 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 11:37:48 +00:00
46323b3786
Remove useless AudioProcessing::Create() overload.
...
R=aluebs@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8046 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-13 06:48:06 +00:00
16825b1a82
Use int64_t more consistently for times, in particular for RTT values.
...
Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t. Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.
BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org , holmer@google.com , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:51:21 +00:00
a7add19cf4
audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter
...
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/37569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8044 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 21:12:29 +00:00
2a26734f04
Partial revert of r7396
...
This change reverts a small part of what was done in r7396. It seems
like that change uncovered another issue with NEON.
BUG=4177,chrome-os-partner:31534
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33849004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8043 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 20:52:21 +00:00
a525c98ca5
Fix parallelizability in ApmTests.
...
Using temporary filenames instead of fixed ones permits them to run in
parallel.
BUG=chromium:445880
R=andrew@webrtc.org , kjellander@webrtc.org
TEST=third_party/gtest-parallel/gtest-parallel -r100 -w100 out-asan/out/Debug/modules_unittests --gtest_filter=*ApmTest*:*CommonFormats*
Review URL: https://webrtc-codereview.appspot.com/35709004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8041 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 17:31:18 +00:00
45db7eefa2
Use Java based audio as default for WebRTC.
...
The work landed in 4034 (use of HW AEC in AppRTC) is currently not
active by default since we build for Open SL. I missed that when I
did my initial change (since I always disabled OpenSL by GYP_DEFINES).
This CL ensures that Java based audio is used as default in WebRTC.
It would be great if we could shift over to Open SL (to cut latency)
but that would (today) mean that we can't support the HW AEC.
Hence, we are not ready to do so yet.
BUG=4034
R=kjellander@webrtc.org , perkj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36699004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8040 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 14:27:23 +00:00
88a4298234
common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin()
...
In addition, expanded the unit test to verify both unstable and stable filters.
BUG=3353, 1132
TESTED=locally on Mac and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8038 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 05:53:43 +00:00
c14e3572c6
common_audio: Made input signal const in WebRtcSplFilterMAFastQ12()
...
BUG=3353, 1133
TESTED=locally on Mac and trybots
R=henrik.lundin@webrtc.org , kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37499004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8037 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 05:50:52 +00:00
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
3a70625caf
audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877
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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
34ac956706
Do not use openmax_dl for MIPS64 platform.
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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
a9b1ec0247
Support for DTLS in OpenSSLAdapter
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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
c5fd66dcdf
Accept incoming pings before remote answer is set to reduce connection latency.
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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
84d84471f5
Minor fixes regarding accumulator usage on MIPS platforms.
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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
5ad4178137
Move the Jingle-specific network code into webrtc/libjingle.
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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
46d4d29a75
Add field trial for screenshare bitrates when using temporal layers.
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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
53cb74107f
Make RelayServerTest use VirtualSocketServer.
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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
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.
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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
ed1a48b0cd
Fix mac video capture leak.
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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
