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Author SHA1 Message Date
andrew@webrtc.org
3d9981d58a Remove unused ThreadData struct. 
TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/4949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5216 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 17:13:47 +00:00
andrew@webrtc.org
3054ba6bb2 Remove the long disabled WEBRTC_SVNREVISION define. 
BUG=500
TESTED=git try
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5215 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 17:00:44 +00:00
andresp@webrtc.org
5b51ebc179 Removing DropDeltaAfterKey functionality which is unused. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4939004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5214 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 15:53:24 +00:00
sprang@webrtc.org
71f055fb41 Add send frame rate statistics callback 
BUG=2235
R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 15:09:27 +00:00
asapersson@webrtc.org
9e5b0342f6 Added a delay measurement, measures the time between an incoming captured frame until the frame is being processed. Measures the delay per second. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5212 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 13:47:44 +00:00
stefan@webrtc.org
79b63206b9 Fixes a crash in fullstack tests introduced with r5209. 
TBR=mflodman@webrtc.org
BUG=1812

Review URL: https://webrtc-codereview.appspot.com/4689005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5211 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 13:34:28 +00:00
henrik.lundin@webrtc.org
b477fa6d21 Small fixes to plot_neteq_delay.m 
Fixing problems with wrap-arounds and other small things. Adding an
extra output value.

Review URL: https://webrtc-codereview.appspot.com/4929004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 12:28:47 +00:00
stefan@webrtc.org
7e9315b42e Adds support for sending redundant payloads over RTX. 
TEST=trybots
BUG=1812
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 10:24:26 +00:00
henrik.lundin@webrtc.org
9523b55826 Fix a typo in neteq.gypi 
This CL is for NetEq3. The #define for iSAC-fb was wrong on one
line. It did not affect the defualt use case, but resulted in
errors if 48 kHz mode was enabled.

TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4909004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5208 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-04 08:24:49 +00:00
andrew@webrtc.org
d7696c4ed1 Compile-out functions only used by the bit-exact test. 
Causes errors on platforms where the test is unused.

TBR=bjornv

Review URL: https://webrtc-codereview.appspot.com/4869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5207 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 23:39:16 +00:00
fischman@webrtc.org
d3865e9124 Don't HANDLE_EINTR(close). Use IGNORE_EINTR(close). 
It is incorrect to wrap close in HANDLE_EINTR on Linux.

BUG=chromium:269623
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4759004

Patch from Mark Mentovai <mark@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5206 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 19:10:20 +00:00
solenberg@webrtc.org
812dd11f8c Add baseline generation/verification to BWE test framework. 
Updating resource file separately, once LGTM. Generates ~628k of files for current tests, highly compressable, once/if we need that.

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

Review URL: https://webrtc-codereview.appspot.com/4639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5204 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 15:11:14 +00:00
sprang@webrtc.org
499631c1e4 Utility class for reading/writing network-byte-ordered integers. 
BUG=
R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2151008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5203 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 13:22:48 +00:00
sprang@webrtc.org
37968a9be7 Change BitrateStats to more generalized RateStatistics 
BUG=2656
R=holmer@google.com, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5202 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 10:31:59 +00:00
pbos@webrtc.org
b613b5ab2b Set local SSRC for VideoReceiveStream. 
As a bonus, also removes GenerateRandomSsrc, which only worked on sender
configs. There's no point to generate random SSRCs in tests.

BUG=2691
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5201 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 10:13:04 +00:00
henrik.lundin@webrtc.org
5ecdef11cc Do not use recursive calling in NetEq test tools 
This CL removes recursive calling in:
- NETEQTEST_DummyRTPpacket::readFromFile,
- NETEQTEST_RTPpacket::readFromFile.

The files currently exist for both NetEq3 and NetEq4, and all are
changed with this CL.

R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5200 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-03 08:26:49 +00:00
fischman@webrtc.org
e0034557a7 RTCPeerConnection(objc): avoid leaking ICE candidate on addition. 
BUG=2670
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5199 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-12-02 18:49:54 +00:00
tina.legrand@webrtc.org
8418e9696b Fixing NetEq tests for new Opus version 
The new version of Opus doesn't generate the same number of bytes encoding the test vectors in audio_decoder_unittest. Therefore the test was updated not to check the length of the encoded packet, to prepare for the coming roll of Opus. Same change was applied to iSAC, which can also generate different number of bytes on different platforms.

BUG=1459
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5195 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-29 09:30:43 +00:00
braveyao@webrtc.org
54e8bfafba Apprtc demo: add DSCP support. 
BUG=2669
TEST=Manual Test
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5194 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-29 02:38:20 +00:00
phoglund@webrtc.org
03c7a35ac0 Fixing long lines in apprtc.py. 
These long lines causes the presubmit to get angry.

BUG=webrtc:2678
R=braveyao@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4369004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5193 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 17:45:08 +00:00
pbos@webrtc.org
e1fc3f22ea Disable check for all sent SSRCs being valid. 
Since the code for setting these up will set the codec before setting
SSRCs for the streams, any frames sent in between will be sent on
random-generated SSRCs.

This part should be added back during work on issue 1695.

BUG=1695
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5192 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 15:40:12 +00:00
bjornv@webrtc.org
bd41a84694 This CL adds an API to enable robust validation of delay estimates. 
Added is
- a member variable for turning robust validation on and off.
- API to enable/disable feature.
- API to check if enabled.
- unit tests for these APIs.

Not added is
- the actual functionality (separate CL), hence turning feature on/off has no impact currently.
- calls in AEC and AEC, where the delay estimator is used. This is also done in a separate CL when we know if it should be turned on in both components.

TESTED=trybots, module_unittest
BUG=
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 14:58:35 +00:00
stefan@webrtc.org
b627f676b3 Fixes a crash in the pacer where it fails to find a normal prio packet if there are no high prio packets, given that the queue has grown too large. 
BUG=2682
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4599005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5190 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 14:00:09 +00:00
pbos@webrtc.org
1f7c8d8b6a Lock frame in ViECapturer::IncomingFrameI420. 
r5160 explicitly assumed that IncomingFrameI420 was never called
sequentially. This assumption was found to be incorrect when some users
were changing beween existing capturers.

BUG=2657
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5189 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 13:26:33 +00:00
pbos@webrtc.org
13d38a13e3 Set up SSRCs correctly after switching codec. 
Before SSRCs were not set up correctly, as the old VideoEngine API
doesn't support setting additional SSRCs before a codec with as many
streams are set.

No test was in place to catch this, so two tests are added to make sure
that we send the SSRCs that are set, and also that we can switch from
using one to using all SSRCs, even though initially not all of them are
set up.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5188 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 11:59:31 +00:00
bjornv@webrtc.org
d1a1c353ac Recommit CL5184 
TBR=aluebs@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/4599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5187 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 11:45:05 +00:00
solenberg@webrtc.org
c8f76ddc19 Refactor Remote Estimators Test into a more reusable form. 
BUG=
R=andresp@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5186 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 10:11:35 +00:00
bjornv@webrtc.org
82eb3a690e Revert 5184 "Small refactoring change in delay_estimator." 
> Small refactoring change in delay_estimator.
> 
> This CL produce the bit exact output and is a preparing step for an upcoming robust validation scheme.
> 
> TESTED=trybots, module_unittest
> BUG=None
> R=aluebs@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/4549004

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5185 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 09:44:47 +00:00
bjornv@webrtc.org
eea079a376 Small refactoring change in delay_estimator. 
This CL produce the bit exact output and is a preparing step for an upcoming robust validation scheme.

TESTED=trybots, module_unittest
BUG=None
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4549004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5184 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-28 07:59:04 +00:00
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