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Author SHA1 Message Date
stefan@webrtc.org
19a40ff05b Ensure that no packet stays in the pacer queue for longer than 2 seconds. 
BUG=2682
TEST=trybots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5182 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-27 14:16:20 +00:00
sprang@webrtc.org
b3ea3afa60 Create default implementation to fix issue in libjingle 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5181 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-27 10:28:27 +00:00
sprang@webrtc.org
4070935f4f Implement and test EncodedImageCallback in new ViE API. 
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5179 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 11:41:59 +00:00
asapersson@webrtc.org
c7ff8f990a Added measure of encode time. Added encode time to the ViE CpuOveruseMeasure api. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5178 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 11:12:33 +00:00
kjellander@webrtc.org
bd51d9324e LSan suppressions for libjingle_peerconnection_unittest 
Some more leaks have entered the code a while ago so
I'm suppressing them to green up the bot again.

BUG=2528
TEST=Successful execution of the test using the steps described in webrtc:2528.
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5177 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 09:37:12 +00:00
sprang@webrtc.org
7f959980f8 Remove const in vie_rtp_rtcp, where there is conflict with 
mock defines in fakewebrtcvideoengine.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5176 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-26 09:30:04 +00:00
mikhal@webrtc.org
d89b52af80 Faster implementation of BitRateStats. 
Landing cl for hguihot.

At high bitrate, EraseOld() could account for a significant part of
the total CPU usage on certain platforms. The new implementation
eliminates per-packet memory allocations and records the number of
bytes in buckets (one bucket per millisecond in window).

R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5175 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 17:49:28 +00:00
elham@webrtc.org
326bcff879 Updated WebRTC version to 3.47 
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5173 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 17:19:46 +00:00
pbos@webrtc.org
4e3161dd55 Style-option file for clang-format. 
Specifies that clang-format as well as related editor plugins should use
Chromium style.

BUG=
TEST=formatting code with vim, running :ClangFormat before/after this change.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5172 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 15:30:37 +00:00
phoglund@webrtc.org
3260f109e3 Made video quality toolchain more configurable. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5171 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 14:10:20 +00:00
stefan@webrtc.org
47fadba750 Add include stdlib.h to files using abs. 
abs function is declared in stdlib.h

Committing for alextaran@chromium.org.
Reviewed here: http://review.webrtc.org/4239004/

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5170 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 12:03:56 +00:00
stefan@webrtc.org
4ab4fc0044 Add test for automatically disabling padding when no video is being captured. 
BUG=2648
TEST=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5169 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 11:54:24 +00:00
fbarchard@google.com
b5bc098e20 Clear empty video frames in unittest so DrMemory will allow them to be read without an uninitialized read error. 
BUG=libyuv:263
TESTED=drmemory out\Debug\modules_unittests.exe --gtest_filter=*PreprocessorLogic
R=kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-25 09:06:33 +00:00
wu@webrtc.org
aa74b5d690 Add success/error callback to set sdp calls. 
Add a workaround for crbug/322756 to append a line break to the end of sdp if needed.

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

Review URL: https://webrtc-codereview.appspot.com/4299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5167 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-23 00:37:50 +00:00
turaj@webrtc.org
5272eb8d83 Don't register iSAC-swb and iSAC-fb in NetEqDecodingTest. 
Android bots break due to r5164. This CL patches that issue.

BUG=
TEST=modules_unittests on local device.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5166 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-23 00:11:32 +00:00
sergeyu@chromium.org
e839da02c1 Fix MouseCursor to MouseCursorShape conversion in ScreenCapturerWin. 
BUG=crbug.com/322596
R=dcaiafa@chromium.org, wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/4279005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5165 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 20:39:16 +00:00
turaj@webrtc.org
78b41a09bd Fix issues with sequence number wrap-around in jitter statistics. 
Related CL for NetEq 3 is https://code.google.com/p/webrtc/source/detail?r=5150

Jitter statistics was not very sensitive to timestamp warp-around, and NetEqDecodingTest.TimestampWrap *DID NOT* fail before fixes applied. However, we still keep the test.

The criteria for the tests are not satisfied for first few packets, before any wrap-around happens. We could either relax the bound or ignore the first few packets. We chose the latter.

BUG=2662
TEST=modules_unittests,trybots
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5164 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 20:27:07 +00:00
fbarchard@google.com
832bd74cfa libyuv r874 for build improvements on ios/android, and improved YUV scale performance. 
BUG=libyuv:288
TESTED=try bots
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4229005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5163 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 19:30:52 +00:00
wu@webrtc.org
b43202d839 Disable PeerConnectionEndToEndTest for tsanv2 build. 
BUG=1205
TEST=try
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5162 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 19:14:25 +00:00
turaj@webrtc.org
1e8c93c953 Distinguish instances of ACM1 from ACM2 by a version string. This is fpr testing purposes and will be removed when the experiment is done and ACM1 is fade out. 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5161 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 17:04:49 +00:00
pbos@webrtc.org
2ffb149c2c Replace VideoFrameI420 with I420VideoFrame. 
Gives one less struct/class for I420 video frames.

BUG=2657
R=mflodman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5160 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 13:10:13 +00:00
andrew@webrtc.org
b0ed8f8a08 Don't reset the AEC filter in extended mode. 
I don't believe I've witnessed this "feature" ever provide a benefit,
and have now collected some evidence of its harm when using the
extended filter mode. It can cause erroneous resets in two cases:

1. Some preprocessing noise suppression is enabled in the system (i.e.
"audio enhancements") that push the noise floor very low, possibly to
zero. If the filter is non-zero this condition can be triggered very
easily, and erroneously.

2. Non-zero energy in the filter before the peak impulse response can
cause a slight (and harmless) "pre-echo" in the error signal. This
becomes more significant as the peak is set further back in the filter.
This effect can cause needless resets during echo onsets.

In short, this isn't a great criterion for filter reset and has the
potential to cause serious harm. Ideally we would remove it entirely,
but in the interests of safety, can start with the extended mode.

BUG=1261
R=aluebs@webrtc.org, bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 06:39:42 +00:00
dwkang@webrtc.org
9e85c01ec8 Add exception handling when configuring MediaCodc in order to prevent break in the new sdk release. 
BUG=2603
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5158 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-22 02:49:17 +00:00
henrik.lundin@webrtc.org
9fe3603dc1 Renaming ViEEncoderObserver::VideoSuspended 
New name is ViEEncoderObserver::SuspendChange.

BUG=2436
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5157 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 23:00:40 +00:00
pbos@webrtc.org
484ee962b5 Protect reads of ViEEncoder::video_suspended_. 
Does not fix an immediate bug, since this is the only method writing to
it there are no concurrent writes, but this should be more future-proof
by protecting all accesses.

BUG=2606
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4109006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 18:44:23 +00:00
fischman@webrtc.org
1977960866 AppRTCDemo(ios): remove codesigning hack now that gyp signs by default. 
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4119005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5155 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 16:48:51 +00:00
stefan@webrtc.org
ef2d55461b Increase size of pacer window to 500 ms as that better matches the encoder. 
BUG=1812
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4129006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5154 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 14:37:11 +00:00
henrik.lundin@webrtc.org
331d4402fc Connect pacer/padding to SuspendBelowMinBitrate 
The suspend function must not be engaged unless padding is also enabled.
This CL makes the connection so that the pacer and padding is enabled
when SuspendBelowMinBitrate is.

Had to change the unit test to make it aware of the padding packets.

BUG=2606
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5153 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 14:05:40 +00:00
pbos@webrtc.org
ffe1b17b57 Lock access to ModuleRtpRtcpImpl::simulcast_. 
Fixes race between RegisterSendPayload and SendOutgoingData.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4099006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5152 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 13:53:13 +00:00
pbos@webrtc.org
2c46f8d854 Rename DestroyStream methods to include Video. 
Matches r5135 which renames CreateSendStream->CreateVideoSendStream for
instance.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 13:49:43 +00:00
henrik.lundin@webrtc.org
6f6ba6edee Fix issues with sequence number wrap-around in jitter statistics 
Wrap-arounds in sequence numbers (and in timestamps) were not always
treated correctly. This is fixed by introducing two helper functions
for correct comparisons, and by casting to the right word size.

Also added a new member variable to AutomodeInst_t. The new member keeps
track of when the first packet has been registered in the automode code.
This was previously done implicitly (and not very good) using the fact
that the lastSeqNo and lastTimestamp members were initialized to zero.

Two new unit tests were added as part of this CL.
NetEqDecodingTest.SequenceNumberWrap was failing before the fixes were
made; now it is ok.

BUG=2654
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 13:17:29 +00:00
pbos@webrtc.org
b3cc78de28 Add -Wnon-virtual-dtor warning for C++ code. 
BUG=2659
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4119006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5149 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 11:42:02 +00:00
sprang@webrtc.org
72964bd4fb Make interface destructor virtual 
In summary, do this:

-  ~FrameCountObserver() {}
+  virtual ~FrameCountObserver() {}

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4099005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5148 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 09:09:54 +00:00
asapersson@webrtc.org
8d02f5dc71 Added API for enabling/disabling RTCP Receiver Reference Time extension. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 08:57:04 +00:00
asapersson@webrtc.org
54a05518e2 Increase run-time for full stack test for the rtt to be added reliably to the delay measurement. 
BUG=2592
R=holmer@google.com, phoglund@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3719004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5146 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 07:45:08 +00:00
braveyao@webrtc.org
425e1d0fb9 Typo in vie_autotest_win.cc 
BUG=2637
TEST=AutoTest
R=mflodman@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5145 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-21 02:17:01 +00:00
henrike@webrtc.org
a750044396 Fixes a crash in VoE when unregistering JNI hooks. 
BUG=11695087
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5144 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 22:32:12 +00:00
wu@webrtc.org
364f204d16 Update talk to 56698267. 
TBR=sergeyu@chromium.org

Review URL: https://webrtc-codereview.appspot.com/4119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5143 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 21:49:41 +00:00
sprang@webrtc.org
dc50aaeaa8 Interface changes to old api, for use by new api transition. 
BUG=2589
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 16:47:07 +00:00
asapersson@webrtc.org
b24d33565c Added ViE API for getting overuse measure. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3129005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 13:51:40 +00:00
pbos@webrtc.org
d29d4e9c08 Deliver I420VideoFrames from VideoRender module. 
Performance issue and simplicity, this implementation skips conversion
to VideoEngine's frame format and then back again to I420VideoFrame.

BUG=2526
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 13:19:54 +00:00
asapersson@webrtc.org
1ae1d0c471 Add possibility to get the last processed RTT from the call stats class (to be used by RTP/RTCP module). 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2383004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5139 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 12:46:11 +00:00
pbos@webrtc.org
27326b6a42 Rename newapi::Transport::SendRTP()->SendRtp(). 
Also fit rampup_tests.cc to use internal::TransportAdapter instead of
implementing its own.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5138 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 12:17:04 +00:00
pbos@webrtc.org
ce90eff345 Rename RTP-extension constants. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5137 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 11:48:56 +00:00
pbos@webrtc.org
53c8573525 Rename video streams' start/stop methods. 
{Start,Stop}{Send,Receive}() -> {Start,Stop}{Sending,Receiving}().

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5136 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 11:36:47 +00:00
pbos@webrtc.org
5a63655ab0 Rename Call::Create{Receive,Send}Stream(). 
Renaming the methods to include Video. Long-term there will hopefully be
AudioSendStream/AudioReceiveStreams as well.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3439004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5135 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-20 10:40:25 +00:00
aluebs@webrtc.org
0b72f5863b Add experimental noise suppression dummy API. 
Add this flag to the voe_cmd_test.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-19 15:17:51 +00:00
sergeyu@chromium.org
5d85819dd2 Fix DesktopAndCursorComposer to restore frames to the original state. 
Screen capturers may reuse frame buffers and they expect that the
frame content isn't changed by the frame consumer.
DesktopAndCursorComposer draws mouse cursor on generated frames and
it was releasing the frames with the mouse cursor on them. Fixed
it to restore frame content erasing mouse cursor before returning
desktop frames.

BUG=crbug.com/316297
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/3899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5133 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-19 02:15:47 +00:00
turaj@webrtc.org
7a05ae5c69 Adding back main() to the test. Now it is possible to choose between ACM1 and ACM2, furthermore, the test can simulate a channel with packet loss and FEC can be activated. Packet loss pattern is based on channel implementation in Channel{.cc,.h}, which currently is a determenistic pattern with 1 every 3rd packet is discarded. 
The main() was deleted in r4731.

BUG=
R=andrew@webrtc.org, minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5132 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 18:16:53 +00:00
pbos@webrtc.org
9c5fb76662 Exclude AV-sync test from Valgrind platforms. 
Test is performance-dependent and was observed to never sync on the
linux_memcheck bot.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5131 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-18 16:22:50 +00:00