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
glaznev@webrtc.org
80452d70cb
Sync Android AppRTCDemo with internal repo.
...
- Fixed some Lint warnings.
- Switch to OPUS by default.
- Add check to WebSocket connection that public methods are called
on correct thread.
R=jiayl@webrtc.org , wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8032 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 19:34:06 +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
kjellander@webrtc.org
1592df78ef
PRESUBMIT: Add GN trybots for Windows and Mac.
...
Add to the default set since they're now green.
BUG=4105
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34689004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8029 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-09 15:38:29 +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
decurtis@webrtc.org
2ead571fb6
Hard define the GUID for AudioEndpoint to avoid conflicts during compile.
...
BUG=3996
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8026 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 19:18:01 +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
kjellander@webrtc.org
5f09564354
Suppress AsyncHttpRequestTest.TestCancel leak for LSan
...
This test fails when run with gtest-parallel on the
ASan bot (that also runs LSan):
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20Asan%20%28parallel%29/builds/27
I'm unable to reproduce this locally but it's
obviously failing on the bots.
BUG=4149
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8021 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-08 10:45:59 +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
59062d5aef
Rename SendAndReceiveH264SvcQqvga to VP8 instead.
...
This looks like it's been incorrect for a while, this test configures
VP8 in QQVGA.
BUG=
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8018 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 19:21:18 +00:00
decurtis@webrtc.org
8af11042cb
Avoid reading past end of string in GetLine.
...
BUG=3881
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8017 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 19:15:51 +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
pbos@webrtc.org
bab79951ca
Convert FileMediaEngineTest to use more expects.
...
Allows pinpointing more precisely where a failure occurs.
BUG=4144
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40399004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8015 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 18:01:29 +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
tina.legrand@webrtc.org
25dd754fff
Excluding a flaky test from DrMemory
...
The excluded tests occassionally fails on the Win DrMemory bot: http://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Full/builds/1529 .
BUG=3318
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33809004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8011 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 14:25:55 +00:00
kjellander@webrtc.org
7fbf278f3f
Suppress memcheck error in video_engine_tests
...
BUG=4147
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33789004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8010 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 11:04:57 +00:00
pbos@webrtc.org
1777880f54
Roll gtest-parallel.
...
Includes a method for setting additional arguments to binaries.
BUG=4142
R=kjellander@webrtc.org
TBR=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/32969004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8009 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 11:03:19 +00:00
kjellander@webrtc.org
07c83a1385
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2)
...
In https://webrtc-codereview.appspot.com/35669004/ the wrong
define was used (OS_WIN only exists in Chromium code).
BUG=4135
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33799004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8008 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 10:36:53 +00:00
tkchin@webrtc.org
4e5115ae73
RTCPeerConnectionFactory: Explicitly create new worker and signaling threads.
...
There should be no change in behavior, since this is what the default
constructor does.
BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39419004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8007 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 06:35:18 +00:00
glaznev@webrtc.org
f6a9714760
Remove peer connection and signaling calls from UI thread.
...
- Add separate looper threads for peer connection and websocket
signaling classes.
- To improve the connection speed start peer connection factory
initialization once EGL context is ready in parallel with the room
connection.
- Add asynchronious http request class and start using it in
webscoket signaling and room parameters extractor.
- Add helper looper based executor class.
- Port some of henrika changes from
https://webrtc-codereview.appspot.com/36629004/ to fix sensor
crashes on non L devices - will remove the change if CL will
be submitted soon.
R=jiayl@webrtc.org , wzh@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8006 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 22:24:09 +00:00
kjellander@webrtc.org
2ec50f2b0f
Memcheck suppression for uninitalized memory in WebRtcIsac_Decode
...
BUG=4143
TBR=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8005 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 11:33:08 +00:00
kjellander@webrtc.org
d95435c17a
Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win
...
These tests have turned out to be flaky on Windows.
BUG=4135
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35669004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8004 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 11:01:35 +00:00
kjellander@webrtc.org
cbe7ca8796
Roll chromium_revision 8e72e1d..271c6cc (307131:309333)
...
This enables OpenSSL by default for Windows, see
8e72e1d..271c6cchttps://codereview.chromium.org/694643002 .
New Android dependencies are required due to being introduced in
build/android/pylib/remote/device/remote_device_test_run.py
of 5c49978f09https://codereview.chromium.org/815213002 was submitted, since
it's based on e2a338fac98e72e1d..271c6cc8e72e1d..271c6ccR=pbos@webrtc.org , pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8003 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 07:24:27 +00:00
tkchin@webrtc.org
3a63a3c35d
iOS AppRTC: First unit test.
...
Tests basic session ICE connection by stubbing out network components, which have been refactored to faciliate testing.
BUG=3994
R=jiayl@webrtc.org , kjellander@webrtc.org , phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28349004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8002 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-06 07:21:34 +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
pbos@webrtc.org
c37e72e890
Make setting identical RTP extensions a no-op.
...
Setting extensions are responsible for a lot of stream tear-downs
causing substantial slowdowns in SetRemoteDescription.
BUG=1788,4077
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40389004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7998 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-05 18:51:13 +00:00
aluebs@webrtc.org
e5a921a82d
Use tmp files in file_utils_unittests
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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.
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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
wzh@webrtc.org
433006a6c2
Fixed style issues from lint and got rid of unused fields.
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BUG=
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41379004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7995 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-05 17:39:43 +00:00
marpan@webrtc.org
c4ad157d8d
Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9.
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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.
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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.
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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
glaznev@webrtc.org
8390c2762e
Add two unit tests for Android AppRTCDemo.
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First unit test will create peer connection client, run
for a few second, close it and verify that there were
no any errors and local video was rendered.
Second unit test will run peer connection in a loopback mode.
To run the test from command line install AppRTCDemoTest.apk
and execute the command:
adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner
R=jiayl@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33609004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7991 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 19:51:12 +00:00
pbos@webrtc.org
896888b7e4
Remove min bitrate from simulcast streams.
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Bitrates are still set using SetBitrateConfig() either way, and this
code causes assertion failures in
VideoSendStream::ReconfigureVideoEncoder: Assertion
`streams[i].target_bitrate_bps >= streams[i].min_bitrate_bps' failed.
R=pthatcher@webrtc.org
BUG=1788
Review URL: https://webrtc-codereview.appspot.com/38529004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7990 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 15:40:56 +00:00
bjornv@webrtc.org
bac0012120
Extend delay estimation window in AEC to 500 ms on all platforms
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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
pbos@webrtc.org
9eacb8cc59
Make P2PTestConductor use VirtualSocketServer.
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Permits running JsepPeerConnectionP2PTestClient in parallel.
TBR=juberti@webrtc.org
BUG=2598
TEST=third_party/gtest-parallel/gtest-parallel -w 128 -r 100 out/Debug/libjingle_peerconnection_unittest --gtest_filter=JsepPeerConnectionP2PTestClient.*
Review URL: https://webrtc-codereview.appspot.com/37459004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7988 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:03:19 +00:00
pbos@webrtc.org
c62749fb47
Parallelize MediaRecorder unittests.
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Exchanging static filenames for temporary ones, permitting tests to be
run in parallel without conflicting parallel uses of the same filenames.
TBR=juberti@webrtc.org
BUG=2597
TEST=third_party/gtest-parallel/gtest-parallel -w 64 -r 100 out/Debug/libjingle_p2p_unittest
Review URL: https://webrtc-codereview.appspot.com/34589004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7987 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-02 09:01:20 +00:00
